| 00:01.21 | cweagans | drmessano, WIMPy: thank you for the advice! |
| 00:03.48 | *** join/#asterisk sputnick (~sputnick@unaffiliated/sputnick) |
| 00:03.54 | sputnick | hi there |
| 00:06.05 | sputnick | anyone can give me a short exemple to play "hello-world" to put in extension.conf when I press "4" when the connection was made ? ( and "Hangup" only after that ) |
| 00:16.56 | p3nguin | exten => 4,1,Playback(hello-world) |
| 00:20.20 | sputnick | p3nguin: that snippet just play "hello-world" if I call "4". Waht I mean is by exemple I call "303", then I listen a message, and I have to type a choice, let's say "4" or "5". Do you get me ? |
| 00:27.45 | *** join/#asterisk ariel_ (~chatzilla@c-24-127-196-248.hsd1.fl.comcast.net) |
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| 00:28.29 | sputnick | What I starts with : http://pastie.org/1019507 I "just" need to know how code line 2 |
| 00:29.13 | sputnick | this one is better : http://pastie.org/1019508 |
| 00:29.33 | Dunkirk | I'm going _crazy_. Anyone try to run a TDM400P under Ubuntu (10.04)? I can't get it to come up. |
| 00:29.42 | pabelanger-lap | sputnick: core show application WaitExten |
| 00:30.50 | sputnick | nice pabelanger-lap, thanks |
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| 00:35.56 | p3nguin | sputnick: I doubt line 3 is going to do what you want it to do. |
| 00:36.08 | sputnick | "exten => s,4,Read(TEST||10)" seems interesting |
| 00:39.02 | sputnick | Read(TEST|hello-world|1|s|1|10) |
| 00:40.00 | pabelanger-lap | sputnick: http://asterisk.pastebin.com/D07sXhLn |
| 00:41.32 | *** join/#asterisk jtrimmer (~jtrimmer@75-151-66-133-WestFlorida.hfc.comcastbusiness.net) |
| 00:44.15 | jtrimmer | Evening everyone. I'm trying to create a call file to run a custom context but it doesn't seem to be working. I think my problem is in the Channel: line I don't understand it completly. Channel: Local/1000@testing-context does 1000 have to be a real extension? |
| 00:44.42 | sputnick | thanks pabelanger-lap, I test it |
| 00:45.50 | pabelanger-lap | jtrimmer: no, but it has to be a valid exten within testing-context |
| 00:46.06 | pabelanger-lap | IE: [testing-context] |
| 00:46.20 | pabelanger-lap | exten => 1000,1,Verbose("Hello World") |
| 00:46.39 | jtrimmer | ohh I see |
| 00:48.54 | jtrimmer | so if I had say Channel: Local/milk@testing-context then it would in theory work if I had exten=> milk,1,Verbose("Hello World") ? |
| 00:49.25 | pabelanger-lap | jtrimmer: Yes |
| 00:50.39 | jtrimmer | ty very much that just made the fog in my head clear right up |
| 00:51.15 | *** join/#asterisk twanny796 (~twanny@78.133.65.141) |
| 00:51.37 | twanny796 | skype for asterisk module? |
| 00:51.49 | pabelanger-lap | twanny796: what about it? |
| 00:52.06 | twanny796 | pabelanger-lap: where can I get it from? |
| 00:52.16 | sputnick | pabelanger-lap: very good :) But if I hit "4" * understand immediatly to play "foo" but if I choose "4" I have to wait. |
| 00:52.17 | pabelanger-lap | ~skypeforasterisk |
| 00:52.18 | infobot | it has been said that skypeforasterisk is a Digium-made channel driver that allows Asterisk to connect to the Skype network using a Skype-supported connection method. Unlike all other solutions it does not require X11, kernel modules, or a Windows machine. See http://www.digium.com/skype for details |
| 00:52.32 | sputnick | ( wait to listen bar ) |
| 00:53.28 | pabelanger-lap | sputnick: Say that again? |
| 00:55.41 | sputnick | When I call 303, I listen "choose 4 or 5" sound, then if I choose "4", "foo" is played just after I type it and if I type "5", I need to wait some secondes to listen "bar" sound. |
| 00:56.46 | pabelanger-lap | sputnick: you must have some other exten in that context that start with 5. IE: exten => _510,1,blah() |
| 00:57.21 | pabelanger-lap | sputnick: your best to move that logic into another context and use goto(new_context,s,1) to access the menu |
| 00:57.35 | pabelanger-lap | sputnick: your delay is because of pattern matching |
| 00:57.46 | sputnick | yes I have 500. |
| 00:58.22 | pabelanger-lap | sputnick: So, WaitExten see the 500, because it is in the same context. And will timeout after 3 seconds. |
| 00:58.38 | sputnick | understand yes pabelanger-lap |
| 01:00.14 | pabelanger-lap | sputnick: When I design IVRs using asterisk, I will create a new context for each menu. IE: [AA-MainMenu] then when I need to move to a new menu, I use the goto command. Makes it easy and clean to read / understand |
| 01:01.07 | sputnick | pabelanger-lap: yes, that's what I try, I'm very new to * ;) |
| 01:02.04 | pabelanger-lap | ~book |
| 01:02.05 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 01:02.08 | pabelanger-lap | sputnick: ^^ |
| 01:02.33 | sputnick | thanks, I have already this book opened right here |
| 01:02.55 | pabelanger-lap | sputnick: Then you are on your way. |
| 01:03.02 | sputnick | dream of a French version of this book |
| 01:03.31 | sputnick | *dreams* |
| 01:04.09 | twanny796 | is there a compiled module of skypeforasterisk 1.4? |
| 01:06.24 | pabelanger-lap | twanny796: Yes, from Digium |
| 01:06.41 | pabelanger-lap | ~skypeforasterisk |
| 01:06.42 | infobot | i guess skypeforasterisk is a Digium-made channel driver that allows Asterisk to connect to the Skype network using a Skype-supported connection method. Unlike all other solutions it does not require X11, kernel modules, or a Windows machine. See http://www.digium.com/skype for details |
| 01:06.49 | pabelanger-lap | twanny796: ^^ |
| 01:08.40 | twanny796 | pabelanger-lap: but digium are selling it!! |
| 01:09.04 | pabelanger-lap | twanny796: Of course, they built it. |
| 01:10.43 | twanny796 | pabelanger-lap: ok, is there an open skype for sip? |
| 01:11.42 | pabelanger-lap | twanny796: no, Skype is a propriety protocol, not the same as SIP |
| 01:12.21 | pabelanger-lap | twanny796: There is no free skype channel driver. You have to buy one from Digium or write your own. |
| 01:16.09 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 01:16.48 | twanny796 | pabelanger-lap: what happened here? http://asterisk.tmcnet.com/topics/open-source/articles/61555-digium-rolls-out-skype-asterisk-open-beta-download.htm |
| 01:18.19 | pabelanger-lap | twanny796: Did you read the article? |
| 01:19.23 | twanny796 | pabelanger-lap: yep, I just read it ;( |
| 01:35.51 | sputnick | thanks all & especially pabelanger-lap. I find my way : http://pastie.org/1019558 |
| 01:43.21 | p3nguin | sputnick: That's still not the ideal way. |
| 01:43.45 | sputnick | why ? |
| 01:45.49 | pabelanger-lap | sputnick: If you use WaitExten, you can add some error handlers and digit timeouts. |
| 01:46.14 | pabelanger-lap | sputnick: IE: exten => i,1,playback(invalid) |
| 01:47.16 | pabelanger-lap | sputnick: But still a good first round |
| 01:48.47 | drmessano | Wait, Digium SELLS stuff? |
| 01:48.52 | drmessano | Those bastards |
| 01:49.17 | sputnick | ok pabelanger-lap |
| 01:49.32 | *** join/#asterisk Godfather_ (~Godfather@79.109.251.13.dyn.user.ono.com) |
| 01:58.38 | p3nguin | sputnick: Why? Because something similar to this makes more sense to most people: http://asterisk.pastebin.com/Q53sgrjy |
| 02:12.12 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 02:14.39 | sputnick | thanks p3nguin, I put it close to me to read when my brain will be more... cooler |
| 02:17.58 | ChannelZ | mmm dirrtay |
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| 02:26.00 | WIMPy | Hmm. Did I get something wrong about allowmultiplelogin=no in manager.conf? I would understand that you can login once per user only, but it seems I can only log in once in total. |
| 02:26.35 | *** join/#asterisk Ziaeon (~Alchemist@adsl-074-166-171-132.sip.mia.bellsouth.net) |
| 02:26.55 | WIMPy | I have two isers. One is logged in and when I try the other I get "Login Already In Use" |
| 02:27.00 | WIMPy | users |
| 02:29.09 | WIMPy | Got it. Was a typo, but it definitely gives te wrong error message. |
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| 04:31.34 | *** join/#asterisk Alton35 (~alton@69.45.116.128) |
| 04:34.05 | Alton35 | A question, if I may. By way of background, running Asterisk 1.6.1.2 right now, with phpagi.php and an "incoming.php" answering calls. |
| 04:34.40 | Alton35 | The problem is that if the caller hangs up, the program exits immediately. I had programmed a lot a couple of years ago with this same setup, and don't remember this behaviour. |
| 04:35.04 | Alton35 | It is something with version 1.6? Or just my memory failing me? |
| 04:35.04 | Alton35 | . |
| 04:36.02 | Alton35 | By "exits immediately", I mean that the program is interrupted, and I can't detect the hangup and write a CDR. |
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| 05:24.38 | [TK]D-Fender | Alton35: Your program isn't responsible for writing CDR, * does that already. And you rpogram gets a SIGHUP which you are capable of trapping |
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| 05:39.47 | Alton35 | Odd, I don't remember having this problem previously. It was a calling-card program and I always wrote my own CDRs. |
| 05:40.15 | Alton35 | I'll see about the SIGHUP though. Good tip. |
| 05:41.46 | *** join/#asterisk QubeZ (~nkasu@68.204.67.110) |
| 05:41.50 | QubeZ | hello all |
| 05:42.25 | QubeZ | what is the preferred method to failover or load balance an Asterisk server? Right now our border server servicing all of our 800/888 numbers is the sole system and we need to plan for failure. |
| 05:43.34 | Alton35 | Put another system in and have your provider send to both of them. |
| 05:44.41 | QubeZ | if sent to both, how do we control which one handles the call and routes? |
| 05:44.58 | Alton35 | They can control that. |
| 05:45.14 | Alton35 | I'd let both of them run, though, and if one goes down, well, you should figure that out and fix it. |
| 05:45.48 | Alton35 | Do you know what's sending to you? Acme Packet or the like? |
| 05:46.37 | QubeZ | yes, Time Warner Telecom |
| 05:46.49 | QubeZ | but we'd rather not have them control, maybe something like OpenSER would be useful here? |
| 05:47.09 | Alton35 | they can do it any way you want it, load balance between the two servers or just send to the 1st unless it fails |
| 05:47.32 | Alton35 | I don't know, I always wrote little switches in asterisk too, but whatever you put in there will just add to the "complication", you know what I mean. |
| 05:48.19 | Alton35 | I don't know how redundant you want to be, but different server, on a different circuit, different location if possible, can't hurt. |
| 05:48.40 | QubeZ | yup, so we were thinking to have an OpenSER in front of both proxy servers and it handles all the balancing BUT then what if the OpenSER server goes down hehe |
| 05:49.00 | Alton35 | that's what I mean, it's more complication on your end, let them handle it, they won't mind |
| 05:49.35 | Alton35 | well, duh, if it's Time Warner, they might be lucky to figure out how. :-) but it's easy |
| 05:49.44 | QubeZ | lol |
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| 06:34.03 | Alton35 | Hah, I finally ran across the "solution", well, what I was doing before, |
| 06:34.13 | QubeZ | Alton35: ? |
| 06:34.15 | Alton35 | which is to run DeadAGI() instead of AGI() |
| 06:34.23 | Alton35 | a question I posted before you arrived I think |
| 06:34.37 | Alton35 | how to keep asterisk from killing your program when the caller hangs up |
| 06:35.00 | Alton35 | Fender, I did find the code to catch the SIGHUP too, so thanks. |
| 06:35.52 | QubeZ | Alton35: we use DeadAGI also to gather some more info and record stuff into the DB after a caller hangs up |
| 06:36.57 | Alton35 | yeah, and I had done that a couple of years ago, I just bleepin' forgot |
| 06:37.08 | Alton35 | The docs keep saying not to use it, but everybody does. |
| 06:38.16 | Alton35 | well, it says deadagi() is deprecated in version 1.6 and will be removed some time, |
| 06:38.22 | Alton35 | not sure whether that's a shame or not :-) |
| 06:38.32 | Alton35 | There is this signal-handling code, here: |
| 06:39.06 | Alton35 | # Callers hanging up will cause SIGHUP, so trap that and do nothing for now. |
| 06:39.06 | Alton35 | declare(ticks = 1); |
| 06:39.06 | Alton35 | function signal_handler($signal_number) |
| 06:39.06 | Alton35 | { |
| 06:39.06 | Alton35 | } |
| 06:39.06 | Alton35 | pcntl_signal(SIGHUP,"signal_handler"); |
| 06:39.20 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-145.cablep.bezeqint.net) |
| 06:39.30 | Alton35 | I do nothing in the routine, to ignore the signal, since my code catches hangups elsewhere. |
| 06:41.47 | *** join/#asterisk coppice (~chatzilla@m121-202-44-106.smartone-vodafone.com) |
| 06:47.00 | Alton35 | I just tried out the SIGHUP-catching code above and it worked fine, for what it's worth. |
| 06:47.06 | Alton35 | This calls for a drink! :-) |
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| 07:13.21 | *** join/#asterisk Godfather_ (~Godfather@193.153.129.150) |
| 07:16.46 | Godfather_ | o/ |
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| 07:27.09 | *** join/#asterisk Mw3 (mw3@mw3.hu) |
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| 07:36.32 | *** join/#asterisk athom (~casper@ip-70-160.dobrich.net) |
| 07:36.33 | athom | I'm using AsteriskNOW 1.7, FreePBX 2.7 and followed this guide for adding extensions: http://www.freepbx.org/support/documentation/howtos/howto-setup-a-remote-sip-extension BTW, setted "NAT IPs from sip_nat.conf" and opened "IPTables port 5060" but I can't register to the extension that I want.. any help please? |
| 07:39.30 | Alton35 | What sort of extensions? IP phones? |
| 07:46.27 | Kyosh | iptables? is the pbx forward facing or behind the NAT? |
| 07:46.50 | athom | hmm |
| 07:47.03 | athom | yes, I just want to connect to the extensions with softphone |
| 07:47.04 | Kyosh | yea dude thats my first concern |
| 07:47.13 | Kyosh | is the pbx forward facing or behind the NAT? |
| 07:47.19 | Kyosh | simple question |
| 07:47.27 | athom | it's different PC |
| 07:47.32 | Kyosh | ok |
| 07:47.33 | athom | I mean, not local |
| 07:47.36 | Kyosh | let me spell it out for you |
| 07:47.41 | Kyosh | ok |
| 07:47.53 | Kyosh | does the pbx have a public IP or a NAT ip? |
| 07:48.04 | athom | public IP |
| 07:48.05 | athom | static |
| 07:48.08 | Kyosh | k |
| 07:48.10 | Kyosh | makes sense |
| 07:48.13 | athom | ohh |
| 07:48.17 | Kyosh | have you tried a softphone? |
| 07:48.31 | athom | I need to set the NAT ips from FreePBX |
| 07:48.34 | athom | not from sip_nat.conf |
| 07:48.35 | athom | :))) |
| 07:48.46 | Kyosh | well actually |
| 07:48.46 | athom | maybe that's the problem, I'll check it now |
| 07:48.54 | Kyosh | thats not the problem |
| 07:49.09 | Kyosh | you are setting static ip's for the sip phones in freepbx or sip_nat.conf? |
| 07:49.18 | athom | sip_nat |
| 07:49.18 | athom | :D |
| 07:49.22 | Kyosh | why? |
| 07:49.28 | Kyosh | leave the IP's for the phones blank |
| 07:49.31 | Kyosh | they will self register |
| 07:49.41 | Kyosh | and of course, i mean on the pbx |
| 07:49.42 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 07:51.02 | athom | now when I go to Asterisk SIP Settings |
| 07:51.04 | athom | it shows me error |
| 07:51.10 | athom | that I need to remove the lines from sip_nat.conf |
| 07:51.12 | athom | I removed them |
| 07:51.14 | Kyosh | cause you already went down the wrong path |
| 07:51.27 | athom | yeah |
| 07:51.37 | athom | I click on auto-configure and it set the IPs |
| 07:51.44 | athom | I'll try now to connect from softphone |
| 07:51.44 | athom | :) |
| 07:51.50 | Kyosh | what ip's? the ip's for the phones? |
| 07:51.57 | athom | no no |
| 07:51.58 | Kyosh | man ur vague |
| 07:51.59 | athom | my static IP |
| 07:52.06 | Kyosh | dude |
| 07:52.18 | Kyosh | sip_nat.conf should not be necessary since the pbx is NOT behind a NAT |
| 07:52.26 | Kyosh | leave that shit alone |
| 07:52.29 | Kyosh | leave it |
| 07:52.30 | athom | okay |
| 07:52.38 | Kyosh | good |
| 07:52.41 | Kyosh | :) |
| 07:52.43 | athom | so I need to make extension NAT disable? |
| 07:52.44 | athom | :) |
| 07:52.51 | Kyosh | extension nat? |
| 07:52.58 | athom | I mean |
| 07:53.08 | athom | in the extension shows: NAT: Yes |
| 07:53.13 | athom | I need to make it NAT: no, right? |
| 07:53.16 | Kyosh | dude |
| 07:53.27 | Kyosh | you seem to be confusing extensions with the pbx |
| 07:53.39 | athom | yeah :)) |
| 07:53.49 | Kyosh | in freepbx, under extensions, all that config is for the user phones, not the pbx settings |
| 07:54.01 | Kyosh | understand? |
| 07:54.06 | athom | yes |
| 07:54.15 | athom | so how I need to set it up? |
| 07:54.19 | Kyosh | so NAT=yes for the extensions, thats all for the phones |
| 07:54.29 | Kyosh | so if the phones are behind a nat, then yes. nat=yes |
| 07:54.43 | Kyosh | i would also suggest leaving the IP address for the phones blank |
| 07:55.01 | athom | ok, I did it |
| 07:55.04 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 07:55.09 | athom | but I still can't register with the softphone |
| 07:55.20 | athom | BTW, I'm on other PC |
| 07:55.23 | Kyosh | actually the host should say 'dynamic' |
| 07:55.38 | athom | the softphone is in the office, the FreePBX is in home |
| 07:55.40 | athom | :) |
| 07:56.01 | athom | I'm in the office now and I'm trying to connect with softphone to the FreePBX |
| 07:56.06 | athom | but it shows me time out.. |
| 07:56.19 | Kyosh | ok so on the freepbx machine, if at a console you do an ifconfig command, it will show the public ip? |
| 07:57.26 | athom | yep |
| 07:57.29 | athom | 217.79.80.213 |
| 07:57.29 | Kyosh | im just being thorough, i hope you understand |
| 07:57.32 | Kyosh | fine |
| 07:57.36 | athom | :) |
| 07:57.38 | Kyosh | now the iptables |
| 07:57.47 | athom | I opened port 5060 |
| 07:57.50 | athom | I think.. |
| 07:57.50 | Kyosh | are you familiar with configuring iptables? |
| 07:57.57 | athom | I saw a guide |
| 07:57.58 | athom | :D |
| 07:58.09 | Kyosh | oy |
| 07:58.13 | Kyosh | one sec |
| 07:58.29 | athom | I typed this |
| 07:58.30 | athom | iptables -A INPUT -p udp -m udp --dport 5060 -j ACCEPT |
| 07:58.36 | athom | then save and then reboot |
| 07:59.06 | Kyosh | one sec buddy |
| 07:59.10 | athom | okay :D |
| 08:00.16 | Kyosh | im actually gonna check to see if you opened it properly |
| 08:00.20 | Kyosh | if you dont mind |
| 08:00.21 | athom | okay |
| 08:00.25 | athom | no problem of course |
| 08:00.26 | athom | :) |
| 08:00.29 | Kyosh | a quick scan |
| 08:00.39 | fenrus | heh, that setting might not persist after a reboot ;) |
| 08:00.52 | athom | I saved the iptables |
| 08:00.57 | athom | and then stop/start |
| 08:00.59 | athom | and then reboot.. |
| 08:01.11 | athom | to be sure :D |
| 08:01.21 | athom | maybe this is the problem , I don't know |
| 08:01.38 | athom | I added a port 80 to iptables and now I can access to FreePBX |
| 08:01.45 | athom | but I can't access to 5060 maybe.. |
| 08:02.08 | athom | this is the log: 10:52:05 Registering user '8520@217.79.80.213' |
| 08:02.08 | athom | 10:52:38 Timeout registration for '8520@217.79.80.213' |
| 08:02.15 | athom | :[ |
| 08:03.11 | Kyosh | nothing is open |
| 08:03.20 | athom | oh my good :(( |
| 08:03.30 | athom | can you give me a command to open it please |
| 08:03.43 | Kyosh | hell if i know |
| 08:03.46 | Kyosh | i dont use iptables |
| 08:03.52 | athom | :D |
| 08:04.06 | Kyosh | lemme see if i can find one |
| 08:04.23 | athom | okay thanks |
| 08:04.29 | Kyosh | http://sipx-wiki.calivia.com/index.php/HowTo_configure_iptables |
| 08:04.29 | fenrus | http://pastebin.com/bi1Ug3N1 |
| 08:04.33 | Kyosh | http://www.voip-info.org/wiki/view/Asterisk+firewall+rules |
| 08:05.22 | Kyosh | yup he's locked |
| 08:06.13 | athom | what does it means? |
| 08:06.20 | Kyosh | try what i pasted |
| 08:06.27 | Kyosh | or |
| 08:06.27 | Kyosh | http://www.google.com/search?sourceid=chrome&ie=UTF-8&q=iptables+open+sip+port |
| 08:06.35 | athom | I used the second one from voip-info |
| 08:06.40 | Kyosh | k |
| 08:06.44 | Kyosh | try the first one |
| 08:06.52 | Kyosh | and remember you need to open RTP as well |
| 08:07.10 | athom | okay, I stopped the iptables |
| 08:07.11 | Kyosh | try them now and do not reboot |
| 08:07.38 | athom | but I think 5060 is still stopped.. |
| 08:08.09 | athom | what is RTP? :D |
| 08:08.53 | Kyosh | real time protocol |
| 08:08.59 | Kyosh | the proto used for the voice transit |
| 08:09.35 | athom | hmm |
| 08:09.42 | athom | okay, the iptables is stopped now |
| 08:09.58 | athom | but maybe port 5060 is still closed |
| 08:10.04 | athom | because I can't register to the extension |
| 08:11.22 | Kyosh | service asterisk restart |
| 08:11.30 | athom | ok |
| 08:11.47 | athom | can I amportal restart? |
| 08:12.12 | athom | Stopped Asterisk.. asterisk stopped.. Starting Asterisk.. asterisk started |
| 08:12.18 | athom | it's ready :) |
| 08:12.29 | athom | I still can |
| 08:12.33 | athom | *can't register |
| 08:12.47 | Kyosh | still filtered |
| 08:13.16 | Kyosh | why do you have the pbx at home and the connections from the office? |
| 08:13.24 | Kyosh | kinda backwards |
| 08:14.07 | athom | because I need set-up the .call files from windows and upload them with SSH to the server |
| 08:14.12 | athom | and then the server start calling |
| 08:14.30 | athom | but now I'm just trying to connect with a normal softphone |
| 08:14.36 | athom | for testing |
| 08:15.03 | athom | can I set the extension's port to 5061 |
| 08:15.11 | athom | and try to connect to port 5061? |
| 08:15.38 | athom | oh, BTW in the extension shows me: deny: 0.0.0.0/0.0.0.0, permit: 0.0.0.0/0.0.0.0 |
| 08:15.40 | athom | is this good? |
| 08:15.40 | Kyosh | why |
| 08:15.52 | Kyosh | dude |
| 08:15.59 | Kyosh | is this a home project or office project? |
| 08:16.07 | athom | home |
| 08:16.08 | athom | :D |
| 08:16.27 | Kyosh | oh so why do you want to connect from the office to your home? |
| 08:16.50 | athom | I don't |
| 08:16.53 | athom | I'm trying because |
| 08:16.59 | *** join/#asterisk AlHafoudh (~AlHafoudh@adsl-dyn190.78-98-189.t-com.sk) |
| 08:16.59 | Kyosh | you said that is what you are trying to do |
| 08:17.04 | athom | when I move .call file to outgoing/ path |
| 08:17.11 | athom | asterisk don't call |
| 08:17.24 | athom | and maybe there is some problem |
| 08:17.25 | Kyosh | i am lost |
| 08:17.29 | Kyosh | gives up |
| 08:17.32 | athom | :D |
| 08:17.46 | athom | okay, the problem is that I want to call to number 359898602211 |
| 08:17.58 | athom | and I maked a simple 359898602211.call file |
| 08:18.01 | Kyosh | you are a phisher |
| 08:18.04 | *** join/#asterisk Tim_Toady (~moi@178.128.16.115.dsl.dyn.forthnet.gr) |
| 08:18.11 | athom | and moved it /var/spool/asterisk/outgoing |
| 08:18.21 | athom | but asterisk don't make any moves and don't call to this number |
| 08:18.42 | athom | and one guy from here told me 1st try to connect with a softphone |
| 08:18.45 | athom | and look what will be happen |
| 08:18.57 | athom | SIP trunk is setted and working |
| 08:19.03 | athom | but I just can't set the extension.. |
| 08:20.43 | athom | IP Phones Online |
| 08:20.44 | athom | 0 |
| 08:20.44 | athom | IP Trunks Online |
| 08:20.44 | athom | 1 |
| 08:20.46 | athom | :( |
| 08:21.31 | Kyosh | http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out |
| 08:21.39 | Kyosh | make sure the data is in the .call file |
| 08:21.46 | Kyosh | check the bottom of that page |
| 08:21.59 | athom | this is the same guide |
| 08:22.03 | athom | that I was doing.. |
| 08:22.26 | athom | Channel: SIP/bgopen/359898602211 |
| 08:22.30 | athom | Application: Playback |
| 08:22.31 | athom | Data: hello-world |
| 08:22.33 | athom | that's it.. |
| 08:22.39 | athom | the file was test.call |
| 08:22.50 | athom | and I chown-ed to asterisk:asterisk and move the file.. |
| 08:22.51 | athom | but nothing.. |
| 08:23.23 | athom | the trunk is connected, everything looks fine but asterisk don't do anything.. |
| 08:24.29 | athom | oh my gooodd |
| 08:24.32 | athom | is working now |
| 08:24.36 | athom | :))) |
| 08:24.47 | athom | I don't know what I was doing last night.. |
| 08:24.56 | athom | maybe I didn't chown-ed it correctly |
| 08:25.35 | athom | I placed the .call file and it calls to my number |
| 08:25.36 | athom | :D |
| 08:25.48 | athom | excellent |
| 08:25.50 | athom | thanks alot! |
| 08:27.01 | Kyosh | i didnt do anything |
| 08:27.04 | Kyosh | :( |
| 08:27.29 | Kyosh | but glad you're happy now :) |
| 08:28.11 | athom | no no |
| 08:28.12 | athom | thanks a lot |
| 08:28.13 | athom | :)) |
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| 09:12.11 | Godfather_ | o/ |
| 09:15.10 | ChannelZ | :| |
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| 10:16.41 | D0HZ0R | morning |
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| 10:52.58 | postkonform | Hello, at all |
| 10:55.13 | *** join/#asterisk binbash_ (~peter@ip4da53781.direct-adsl.nl) |
| 10:57.28 | postkonform | I have a problem with the CDRs. When I redirect an incoming call with Goto to an extension which then initialises a call via Dial, no CDR is written into my mysql database. If i use Gosub instead it works. The problem now is, that I use the redirect action of the api to redirect incoming calls, and I need CDRs to be written, but Redirect seems to use goto instead of gosub |
| 10:57.47 | postkonform | ami , not api |
| 10:58.07 | *** join/#asterisk imox1234 (~imox1234@e179006119.adsl.alicedsl.de) |
| 10:59.36 | postkonform | the interesting thing is, it writes a CDR if no bridge was established (no answer, busy, fail) but not if the call was answered |
| 11:00.03 | postkonform | is this a bug in the goto app or am i doing something wrong? |
| 11:02.12 | postkonform | no suggestions? |
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| 11:25.12 | athom | I'm using AsteriskNOW 1.7, FreePBX 2.7, SIP trunk successfully connected and making calls from .call files. I want these calls to hang-up after 1 ring.. how can I make that kind of DialPlan in FreePBX? |
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| 12:39.21 | Godfather_ | how should i play a file durnig a conversation? |
| 12:40.35 | Godfather_ | for example, i want to play an announcement when presing some keys (*3), should i use features.conf? |
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| 12:55.59 | FutureWeb | does anyone know a good mp3 to ulaw convertor ? please (dont tell me use sox or whatever it is.. cause it doesnt wanna work :| ) |
| 12:57.49 | *** join/#asterisk gnomie (~anomie@cpc2-shef10-2-0-cust202.barn.cable.virginmedia.com) |
| 13:00.27 | gnomie | Is it possible to make an outgoing direct ip call from Asterisk without having a sip trunk? |
| 13:11.03 | *** join/#asterisk pwell (~pwell@ool-435255fc.dyn.optonline.net) |
| 13:11.19 | pwell | anyone know what system is the old school party lines use? |
| 13:11.23 | pwell | -is |
| 13:11.34 | pwell | the free one's with the 9 rooms and a lobby |
| 13:13.25 | athom | I'm using AsteriskNOW 1.7, FreePBX 2.7, SIP trunk successfully connected and making calls from .call files. I want these calls to hang-up after 1 ring.. how can I make that kind of DialPlan in FreePBX? |
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| 13:31.41 | gnomie | athom: inside the call files you can set a parameter Waittime for how long you want it to ring. See http://www.the-asterisk-book.com/unstable/call-file.html |
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| 13:43.07 | Godfather_ | can anyone explain me this? http://pastebin.com/Fz17d747 I tried both, defining that variable and not (dynamic_features), and in both cases i'm able to use the feature in the caller/called party? |
| 13:49.12 | troy42 | doc/tex/channelvariables.tex:${DYNAMIC_FEATURES} * The list of features (from the [applicationmap] section of |
| 13:49.21 | troy42 | you may already know this, but it looks like it may be specific to those |
| 13:49.24 | *** join/#asterisk nighty^ (~nighty@x122091.ppp.asahi-net.or.jp) |
| 13:49.46 | troy42 | (i haven't personally used that variable, i just grepped the source) |
| 13:50.00 | *** join/#asterisk sequencer (~something@81.10.125.87) |
| 13:50.07 | sequencer | morning all |
| 13:50.30 | sequencer | anyone available to help ? :) |
| 13:51.00 | troy42 | morning :) ask and see |
| 13:51.22 | sequencer | thnx! i am having trouble with my sip conf and x-lite :s |
| 13:52.53 | sequencer | Using SIP RTP CoS mark 5 |
| 13:53.04 | sequencer | idont even know what that means :s |
| 13:53.32 | troy42 | no idea whether anyone's around, it might be worth saying what problem you're having |
| 13:53.35 | troy42 | :O |
| 13:54.01 | sequencer | basically more like.. a first time installation |
| 13:54.13 | sequencer | need to configure extensions / SIPs |
| 13:54.38 | sequencer | then need to configure it to connect to another Server via IAX |
| 13:54.41 | troy42 | checked TFOT yet? |
| 13:54.55 | sequencer | whats that ? :s |
| 13:55.15 | sequencer | am not a guru.. just a 3 hours working on this :s |
| 13:55.16 | troy42 | pop over to http://cdn.oreilly.com/books/9780596510480.pdf |
| 13:55.21 | sequencer | i got this |
| 13:55.26 | troy42 | or actually http://www.google.com/search?q=asterisk+tfot |
| 13:55.53 | sequencer | i already have it |
| 13:56.03 | sequencer | but it doesnt seem to work with what am having |
| 13:56.25 | sequencer | exten => 1000,n,Dial(SIP/1000,30) |
| 13:56.39 | sequencer | i am trying a test extension |
| 13:56.47 | sequencer | but it doesnt make calls |
| 13:57.14 | fenrus | doesnt the n need to be 1 for the first entry? |
| 13:57.20 | troy42 | does it register? |
| 13:57.27 | troy42 | fenrus: and yes |
| 13:57.36 | *** join/#asterisk nighty^ (~nighty@x122091.ppp.asahi-net.or.jp) |
| 13:58.08 | *** join/#asterisk jmacz (~jmacz@190.25.5.200) |
| 13:59.12 | sequencer | http://pastebin.org/360190 |
| 13:59.24 | sequencer | ok heres my conf |
| 13:59.46 | troy42 | are you able to register? |
| 13:59.56 | sequencer | am not sure.. |
| 14:00.06 | sequencer | my xlite doesnt actually "register" |
| 14:00.14 | sequencer | it displays ready 1000 |
| 14:00.41 | sequencer | but only if i removed the setting "register with domain" |
| 14:00.59 | troy42 | i'd suggest getting on the asterisk console, running "sip set debug on", "core set verbose 999", and "core set debug 999" and seeing what happens when you start xlite |
| 14:01.34 | troy42 | or asterisk -vvvvvvv when starting the cli and flipping on sip debug |
| 14:01.47 | troy42 | makes breakfast |
| 14:02.20 | sequencer | ok here comes trouble.. |
| 14:02.52 | troy42 | i doubt it's broadly-applicable enough to be useful, but i've got screenshots of an xlite config at http://help.cloudvox.com/faqs/sip-phones/x-lite |
| 14:03.34 | sequencer | http://pastebin.org/360197 |
| 14:03.38 | sequencer | whopa ! |
| 14:03.55 | sequencer | 408 request time out :s |
| 14:05.26 | troy42 | hm, i see the unauthorized response with the realm to auth with, but no reply |
| 14:06.15 | sequencer | misconfig ? |
| 14:06.33 | *** join/#asterisk smooth_penguin (~smoove@59.95.15.9) |
| 14:06.36 | troy42 | potentially. i'd double-check your sip user settings, and make sure it's matching your user |
| 14:07.40 | troy42 | may need to tinker with the "insecure" option, although with what's there i don't know for sure |
| 14:07.52 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 14:07.59 | troy42 | and/or "host" to make sure your host matches or is dynamic |
| 14:08.09 | troy42 | bbiaf, bagel |
| 14:08.14 | sequencer | http://pastebin.org/360202 |
| 14:08.40 | sequencer | am not sure if this is right though |
| 14:10.02 | sequencer | i got sth wiered |
| 14:10.14 | sequencer | why does it say 192.168.0.157 |
| 14:10.30 | sequencer | shouldnt this be my global IP ? |
| 14:12.13 | fenrus | does that machine have any other ip-address on any other interface |
| 14:12.14 | fenrus | ? |
| 14:12.24 | sequencer | which machine ? |
| 14:12.45 | sequencer | the server or my client ? |
| 14:13.03 | fenrus | hm, sorry - did not read what you've written |
| 14:14.06 | *** join/#asterisk nighty^ (~nighty@x122091.ppp.asahi-net.or.jp) |
| 14:14.18 | troy42 | sequencer: you may need nat=yes |
| 14:14.25 | sequencer | in sip ? |
| 14:14.29 | troy42 | yeah |
| 14:14.42 | troy42 | multitasking here so may be off a bit, but check out the nat option |
| 14:14.51 | sequencer | lets try |
| 14:15.11 | sequencer | Jun 26 16:20:49] NOTICE[28957]: chan_sip.c:21639 handle_request_register: Registration from '"test"<sip:1000@mecmc.myvnc.com>' failed for '81.10.125.87' - Username/auth name mismatch |
| 14:15.11 | sequencer | Scheduling destruction of SIP dialog 'ODkyNDcyODc3ZjRhYTg0YTYzNzUxZTg2ZTNjMWY5OGQ.' in 32000 ms (Method: REGISTER) |
| 14:15.41 | sequencer | whoa |
| 14:15.44 | sequencer | regsitered! |
| 14:16.01 | sequencer | thanks man! |
| 14:16.46 | troy42 | congrats! |
| 14:16.47 | troy42 | np |
| 14:17.31 | sequencer | whoa |
| 14:17.32 | sequencer | ok |
| 14:17.42 | sequencer | bi called my self |
| 14:17.50 | sequencer | i couldnt hang up :s |
| 14:18.10 | troy42 | haha |
| 14:18.13 | sequencer | its fine though |
| 14:18.15 | troy42 | nice |
| 14:18.18 | sequencer | not a big deal :s |
| 14:18.37 | sequencer | now need to configure iax :s |
| 14:19.23 | sequencer | the echo test isnt working though :s |
| 14:21.38 | *** join/#asterisk troy42 (troy@fitzroy.yort.com) |
| 14:21.43 | sequencer | wb |
| 14:22.05 | troy42 | thx |
| 14:22.29 | troy42 | brews 49th parallel coffee |
| 14:25.42 | *** join/#asterisk valajbeg (~hamo@77.78.242.52) |
| 14:26.43 | sequencer | whew |
| 14:26.53 | sequencer | am having a trouble with getting in the echo test |
| 14:27.17 | sequencer | it just says calling.. :s |
| 14:27.31 | sequencer | then it times out and starts ringing |
| 14:30.39 | *** join/#asterisk garymc (~chatzilla@host81-148-64-72.in-addr.btopenworld.com) |
| 14:32.50 | *** join/#asterisk n3hxs (~HAMming@phx-69-171-161-19.evdo.leapwireless.net) |
| 14:36.46 | sequencer | troy? |
| 14:38.33 | p3nguin | Did you post your dialplan in pastebin.com? |
| 14:39.04 | garymc | anyone familiar with freepbx here as none around in other channel? |
| 14:42.55 | sequencer | p3nguin yeah i did |
| 14:43.08 | *** join/#asterisk nicoAMG (~nicoamg@201.237.49.131) |
| 14:43.20 | sequencer | http://pastebin.org/360190 |
| 14:48.02 | p3nguin | So you dial 500 on your phone, and what happens? |
| 14:51.20 | *** join/#asterisk mifadir (29891ea9@gateway/web/freenode/ip.41.137.30.169) |
| 14:56.03 | *** join/#asterisk Tim_Toady (~moi@178.128.16.115.dsl.dyn.forthnet.gr) |
| 14:57.50 | sequencer | pretty much nothing |
| 14:58.16 | sequencer | it just keep sayin Calling.. |
| 14:58.36 | sequencer | after a minute i start to hear that its ringing on the other side |
| 14:58.39 | sequencer | nothing else :s |
| 14:59.10 | p3nguin | the other side of what? |
| 14:59.16 | sequencer | i mean.. |
| 14:59.28 | sequencer | its not my phone is ringing like am recieving calls |
| 14:59.34 | sequencer | i hear a ring on the line |
| 14:59.39 | sequencer | like if am calling someone |
| 15:01.23 | *** join/#asterisk n3hxs (~HAMming@phx-69-171-161-240.evdo.leapwireless.net) |
| 15:01.50 | p3nguin | http://pastebin.org/360257 |
| 15:02.00 | p3nguin | This is how I prefer to use the echo test. |
| 15:02.39 | *** join/#asterisk mesfet (~mesfet@host165-3-static.25-87-b.business.telecomitalia.it) |
| 15:03.26 | sequencer | the call was picked up |
| 15:03.32 | sequencer | but i didnt hear anything |
| 15:03.37 | p3nguin | Using my way? |
| 15:03.40 | sequencer | yeah |
| 15:04.08 | p3nguin | Do you have NAT in between your phone and asterisk system? |
| 15:04.15 | *** part/#asterisk mesfet (~mesfet@host165-3-static.25-87-b.business.telecomitalia.it) |
| 15:04.17 | *** join/#asterisk bipolar (~bipolar@offsitesysadmin.com) |
| 15:04.18 | sequencer | whats that ? |
| 15:04.23 | p3nguin | ~nat |
| 15:04.24 | infobot | nat is, like, Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 15:04.43 | sequencer | i did set it up |
| 15:04.45 | sequencer | nat=yes |
| 15:04.53 | sequencer | wihtout it i couldnt register |
| 15:04.59 | p3nguin | Is Asterisk behind NAT, too? |
| 15:05.12 | sequencer | :s |
| 15:05.20 | sequencer | the nat is configered within asterisk |
| 15:05.25 | sequencer | in the sip.conf |
| 15:05.35 | p3nguin | Is Asterisk behind NAT, too? |
| 15:05.51 | sequencer | i am not sure what do you ask exactly |
| 15:06.00 | sequencer | asterisk is on a server |
| 15:06.05 | sequencer | with DMZ option |
| 15:06.54 | sequencer | if you can explain it better i might be able to answer |
| 15:07.01 | p3nguin | That's probably half the problem. |
| 15:07.07 | p3nguin | No one understands DMZ. |
| 15:07.56 | sequencer | oh |
| 15:07.57 | Gugge | sequencer: is the server behind some router running NAT (sharing a public ip with computers having private ips) |
| 15:07.59 | p3nguin | If the Asterisk system is behind NAT, you need to configure it accordingly. |
| 15:08.01 | p3nguin | ~sipnat |
| 15:08.02 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 15:08.07 | p3nguin | ^^^^^^^^ |
| 15:08.15 | sequencer | oh |
| 15:08.17 | sequencer | ok |
| 15:08.20 | sequencer | well.. yeah |
| 15:08.22 | sequencer | thats right |
| 15:08.27 | sequencer | although.. |
| 15:08.39 | sequencer | the server itself is set as DMZ through the routere |
| 15:08.47 | p3nguin | ANd that's half the problem. |
| 15:08.55 | sequencer | DMZ: Dimilished Zone |
| 15:09.41 | sequencer | the server basically has the same IP address as the connection |
| 15:09.42 | Gugge | demilitarized zone |
| 15:09.49 | sequencer | same same.. :S |
| 15:10.07 | Gugge | basically is not enough, the server does not know the external ip, unless you set it with externip |
| 15:10.13 | Gugge | when its running behind NAT |
| 15:10.19 | sequencer | hmmm |
| 15:10.37 | sequencer | does it have to be a statis ip ? |
| 15:10.45 | sequencer | static* |
| 15:10.46 | Gugge | no, you are allowed to change that setting |
| 15:10.51 | Gugge | but its gonna be boring |
| 15:11.03 | Gugge | you need to change it every time the public ip changes |
| 15:11.07 | sequencer | how about if i set it to a dynamic domain name? |
| 15:11.31 | sequencer | do i set this in the sip.config or extensions ? :d |
| 15:11.32 | p3nguin | Follow the guide listed above. |
| 15:11.57 | p3nguin | Come back when you're finished. |
| 15:12.15 | sequencer | ok .. but.. |
| 15:12.24 | p3nguin | Meanwhile, forget about DMZ, since no one seems to understand what it is nor the proper usage of it. |
| 15:12.38 | sequencer | ok |
| 15:13.09 | sequencer | whats the exact topic to use though ? |
| 15:13.15 | sequencer | heres the situation: |
| 15:13.45 | sequencer | clients within the network will be using asterisk to connect to a third party iax server |
| 15:13.46 | *** join/#asterisk btsteve (~tstevens@24-196-234-39.dhcp.gwnt.ga.charter.com) |
| 15:14.00 | p3nguin | "the network" |
| 15:14.05 | p3nguin | What network are you talking about? |
| 15:14.06 | sequencer | they will connect locally to the server |
| 15:14.13 | sequencer | LAN users |
| 15:14.16 | p3nguin | The one where Asterisk resides? |
| 15:14.21 | sequencer | exactly |
| 15:14.31 | sequencer | but as am on a remote side |
| 15:14.41 | p3nguin | okay |
| 15:14.42 | sequencer | am trying to configure and test it from a remote |
| 15:15.00 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 15:15.09 | p3nguin | Get the damn DMZ setting disabled and configure it in a more appropriate way. |
| 15:15.24 | sequencer | i have to set it through dmz |
| 15:15.37 | sequencer | or i will have to use port forwarding for all ports |
| 15:15.51 | p3nguin | Asterisk doesn't need "all ports." |
| 15:16.10 | p3nguin | It needs one port for incoming SIP and a small range of ports for RTP. That is all. |
| 15:16.20 | *** join/#asterisk jetlag (jetlag@pool-173-61-204-106.cmdnnj.east.verizon.net) |
| 15:16.25 | sequencer | i understand |
| 15:16.28 | sequencer | but its a hassle |
| 15:16.44 | sequencer | dmz basically gives the public ip to the server |
| 15:16.50 | Gugge | what happens when some other user behind that NAT router is using one of the ports asterisk tries to use for RTP? |
| 15:17.00 | Gugge | then that DMZ setting is useless |
| 15:17.02 | p3nguin | If you understood, you wouldn't be here arguing with me that you need to have it in DMZ, which is most definitely not implemented correctly in the first place. |
| 15:17.21 | sequencer | no its not :s |
| 15:17.22 | p3nguin | DMZ absolutely does not give a public IP address to the server. |
| 15:17.55 | Gugge | if it did, no other users behind that router would be able to access the internet :) |
| 15:18.00 | sequencer | it emulates it |
| 15:18.07 | p3nguin | No it doesn't. |
| 15:18.36 | sequencer | giving access to WAN users to be able to access the private server through the Public IP address |
| 15:18.45 | p3nguin | That's retarded. |
| 15:18.50 | sequencer | instead of assigning valid port forwarding |
| 15:18.54 | sequencer | maybe it is.. |
| 15:19.02 | p3nguin | Anyway, you feel like you know more about this, so I'm going to leave you to your own devices. |
| 15:19.10 | sequencer | but thats the setup we have here :s |
| 15:19.16 | p3nguin | Good luck! |
| 15:19.17 | sequencer | lol thanks |
| 15:19.23 | sequencer | i really need it! |
| 15:19.25 | sequencer | :) |
| 15:19.40 | p3nguin | If you're willing to do some networking the right way, I'm sure someone will be willing to help you. |
| 15:19.55 | sequencer | so, how would it be the right way? |
| 15:20.05 | sequencer | just setup port forwarding ? |
| 15:20.17 | p3nguin | 1) You don't understand what DMZ is, so stop trying to use it. |
| 15:20.30 | p3nguin | 2) Forward the ports that are necessary and nothing else. |
| 15:20.56 | garymc | Anyone know how to transfer calls to a que in freepbx? |
| 15:21.17 | p3nguin | ~freepbx |
| 15:21.17 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 15:22.00 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.168.251.dsl.dyn.forthnet.gr) |
| 15:24.20 | *** join/#asterisk DrCron (rszasz@saxonco.com) |
| 15:33.19 | FutureWeb | does anyone know a good mp3 to ulaw convertor ? please (dont tell me use sox or whatever it is.. cause it doesnt wanna work :| ) |
| 15:33.58 | *** join/#asterisk mesfet (~mesfet@host165-3-static.25-87-b.business.telecomitalia.it) |
| 15:34.02 | Chainsaw | FutureWeb: You could use lame to decode the MP3 to a regular WAV. It'll become easier from there. |
| 15:34.20 | FutureWeb | I tried wav etc.. none works |
| 15:34.54 | Chainsaw | FutureWeb: That's not what I'm saying. |
| 15:35.13 | Chainsaw | FutureWeb: I'm saying make it a two-stage process. MP3 -lame-> WAV -sox-> G.711 ulaw. |
| 15:36.49 | FutureWeb | Chainsaw: WAV -sox-> G.711 ula << its that part which I have no idea what img onna use to do it ? |
| 15:36.53 | FutureWeb | *im gonna |
| 15:37.03 | Tim_Toady | if you load format_mp3.so on asterisk you can use the mp3 files directly or use asterisk -rx "file convert foo.mp3 foo.ulaw" to convert them to ulaw |
| 15:37.07 | Gugge | FutureWeb: install sox, and mpt support for it, then itll work fine :) |
| 15:37.19 | Chainsaw | transfers FutureWeb's call to Tim_Toady |
| 15:37.38 | Gugge | s/mpt/mp3/ |
| 15:38.01 | FutureWeb | kk thanks let me try ;D |
| 15:38.01 | Tim_Toady | but i think using lame + sox might produce better quality output |
| 15:39.02 | Gugge | i think sox uses lame for its mp3 support actually :) |
| 15:41.56 | FutureWeb | [pdx.ftwb-networks.net moh]# asterisk -rx "file convert lg.mp3 lg.ulaw" |
| 15:41.56 | FutureWeb | Unable to open input file: lg.mp3 |
| 15:41.57 | FutureWeb | Command 'file convert lg.mp3 lg.ulaw' failed. |
| 15:42.13 | Tim_Toady | loaded format_mp3 module? |
| 15:42.23 | FutureWeb | I guess so let me re-check |
| 15:42.25 | Gugge | maybe try full path to the file |
| 15:43.11 | FutureWeb | in what conf file would that be exactly ? |
| 15:43.14 | FutureWeb | the moh one ? |
| 15:43.16 | Faithful | I can not get * to answer my SPA3000. It picks the call up but the call does not pass through to the extension that answered instead the SPA3000 says it is busy... and it's the same when you try to trunk out of it to pstn |
| 15:43.35 | Gugge | file convert /path/to/lg.mp3 /path/to/lg.ulaw |
| 15:43.55 | FutureWeb | ahaa that worked |
| 15:44.14 | Gugge | without the path asterisk has no idea where your files are |
| 15:48.30 | sequencer | hi again |
| 15:48.35 | sequencer | man what a mess! |
| 15:49.12 | sequencer | apparently.. i have the worst setup ever! |
| 15:49.18 | Alton35 | hah |
| 15:49.48 | Alton35 | I thought the DMZ thing would get you by, at least for testing. But I have no idea what sort of problem you have. |
| 15:50.07 | sequencer | asterisk server behind nat, using sip to register at a second server, clients connect to the server from LAN and from internet behind NAT |
| 15:50.26 | sequencer | it does get by though.. |
| 15:50.57 | sequencer | the only thing i can do is to bind the router mac address to the server's |
| 15:51.02 | Alton35 | ever consider a vpn? |
| 15:51.10 | Alton35 | I have zero problems getting stuff here and there, usually use openvpn. |
| 15:51.26 | sequencer | yeah.. but configuration is a hassle |
| 15:51.31 | Alton35 | not at all |
| 15:51.32 | sequencer | the server is used for everything.. |
| 15:51.36 | Alton35 | get into it, it's very simple. |
| 15:51.42 | sequencer | database, webserver..etc |
| 15:52.46 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 15:53.15 | sequencer | am trying to get ast. to register to another server but it fails :s |
| 15:53.43 | sequencer | am sure its because its behind nat |
| 15:53.54 | sequencer | i enabled nat=yes in the sip.conf |
| 15:55.44 | Alton35 | I dunno, just giving you something that's sure to work. It's good to get into the vpn thing for situations such as this. |
| 15:56.39 | sequencer | hmm.. |
| 15:56.50 | sequencer | you mean to tie both servers in a vpn ? |
| 16:06.11 | *** join/#asterisk Alton35 (~alton@69.45.116.128) |
| 16:06.36 | Alton35 | sequencer: yes, directly together |
| 16:07.52 | *** join/#asterisk WWGD (~WWGD@208.79.14.130) |
| 16:07.59 | Alton35 | I wouldn't expect to pump a huge amount of traffic through it, but for making things work, it's fine. |
| 16:08.00 | *** join/#asterisk ph8 (ph8@unaffiliated/ph8) |
| 16:08.26 | sequencer | thanks.. but it wont work in my case :s |
| 16:08.37 | ph8 | morning all, i'm trying to setup asterisk/freepbx - freepbx can't connect to the asterisk manager though and nmap says it's not listening for connections on localhost or elsewhere - do i have to explicitly enable it comehow? |
| 16:08.51 | sequencer | i only have a register access on the second server |
| 16:10.36 | Godfather_ | can anyone explain me this? http://pastebin.com/Fz17d747 I tried both, defining that variable and not (dynamic_features), and in both cases i'm able to use the feature in the caller/called party? |
| 16:13.37 | Kyosh | i have no idea what that pastebin post is in regards to |
| 16:13.47 | Kyosh | is there a specific file that belongs in? |
| 16:13.57 | Kyosh | any way to further expand on what you are trying to do? |
| 16:14.09 | *** join/#asterisk aidinb (~Aidin@71-95-223-217.dhcp.mtpk.ca.charter.com) |
| 16:16.12 | Kyosh | guess not |
| 16:17.16 | Godfather_ | Kyosh, its to me? |
| 16:18.47 | Kyosh | yup |
| 16:19.19 | Godfather_ | Kyosh, i've defined this feature: |
| 16:19.20 | Godfather_ | asterisk -rx "file convert /var/lib/asterisk/mohmp3/k2.mp3 /var/lib/asterisk/mohmp3/k2final.ulaw" |
| 16:19.25 | Godfather_ | Dynamic Feature Default Current |
| 16:19.25 | Godfather_ | --------------- ------- ------- |
| 16:19.25 | Godfather_ | antispam no def #9 |
| 16:20.16 | Godfather_ | and i read i need to define DYNAMIC_FEATURES=antispam# in extensions.conf |
| 16:20.18 | Godfather_ | to enable it |
| 16:20.45 | Godfather_ | i tried both, defining it in [globals] and not defining it, and both works |
| 16:20.53 | ph8 | anyone got any idea why my asterisk manager might not be listening? enabled = yes in manager.conf |
| 16:20.59 | Godfather_ | then i dont understand that paragraph in features.conf |
| 16:22.21 | Kyosh | yea dude that should be in your pastebin too |
| 16:24.32 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 16:24.46 | Kyosh | ph8: did you reload ast after enabling? |
| 16:24.56 | Kyosh | godfather, yea it doesnt make much sense |
| 16:24.59 | ph8 | Kyosh: yeh did a service restart |
| 16:25.26 | Kyosh | godfather: but there are samples, testfeature => #9,peer,Playback,tt-monkeys ;Allow both the caller and callee to play |
| 16:25.26 | Kyosh | ; ;tt-monkeys to the opposite channel |
| 16:25.36 | Kyosh | ph8, hmmm |
| 16:25.49 | Kyosh | ph8, did you assign users to the manager? |
| 16:26.00 | sequencer | ok so after i have this outrageous setup i decided to get a vps to run asterisk :s |
| 16:26.31 | Kyosh | godfather, http://www.voipuser.org/forum_topic_7787.html may help |
| 16:26.55 | Kyosh | as an example at ledast |
| 16:26.59 | Kyosh | least |
| 16:28.04 | Kyosh | brb |
| 16:28.37 | Godfather_ | Kyosh, i'll have a look at it, ty |
| 16:32.24 | ph8 | Kyosh: yeh the standard admin one looks like it's configured ok |
| 16:32.46 | ph8 | i have [admin] secret = 1234 permit=0.0.0.0/0.0.0.0 |
| 16:32.52 | ph8 | and the read/write lines |
| 16:36.41 | ph8 | Kyosh: ah it would appear asterisk crashes whenever i try and connect to the manager |
| 16:46.29 | *** join/#asterisk nighty^ (~nighty@x122091.ppp.asahi-net.or.jp) |
| 16:53.59 | Kyosh | yeesh |
| 16:54.01 | Kyosh | thats not good |
| 16:58.01 | *** join/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 16:58.34 | Godfather_ | Kyosh, did you understand my question? |
| 16:59.55 | Godfather_ | Kyosh, my application maps works fine, i dont know why works if i not set [global] dynamic_features=antispam# |
| 17:01.57 | pwell | anyone know what system is the old school party lines use? Defcon for example and other from back in the day. It was always the same setup so it must have been either one guy or propriatary equiptment. "Welcome to the Board.." "You are in the Lobby" "Room 3" |
| 17:04.45 | *** join/#asterisk BANSAL (~bansal@117.199.124.89) |
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| 17:20.55 | *** join/#asterisk sequencer (~something@81.10.125.87) |
| 17:20.59 | sequencer | hi all :) |
| 17:21.21 | sequencer | Alton35 :) |
| 17:35.45 | *** join/#asterisk MiserySoft (~MiserySof@host81-148-66-78.in-addr.btopenworld.com) |
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| 17:56.02 | *** part/#asterisk WWGD (~WWGD@208.79.14.130) |
| 18:02.02 | *** join/#asterisk voxter (~voxter@189.182.29.65) |
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| 18:20.12 | *** join/#asterisk athom (~asd@95-42-220-29.btc-net.bg) |
| 18:21.15 | athom | I'm using AsteriskNOW 1.7 with FreePBX 2.7, how to make automatically hang-up after 1 ring (dialplan) when making a call? |
| 18:21.57 | athom | I was using WaitTime: 1 but this is not after 1 ring, it's after 1 second.. and sometimes it even don't make the call.. |
| 18:22.43 | *** join/#asterisk MohsenSaeedi (~mohsen@unaffiliated/mohsensaeedi) |
| 18:22.52 | MohsenSaeedi | i have problem with elastix 1.6 and 2.0 rc3 too. i have openvox A400 and when i configured my voip system everything works very good. but when i restarted my system then i couldn't connect to outbound trunk . when i use 9 i got 503 eror |
| 18:30.05 | [TK]D-Fender | athom: .. |
| 18:30.09 | [TK]D-Fender | ~freepbx |
| 18:30.10 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 18:35.43 | athom | ok |
| 18:35.58 | *** part/#asterisk athom (~asd@95-42-220-29.btc-net.bg) |
| 18:46.32 | *** join/#asterisk [sr] (~Unknowned@pal-213-228-140-150.netvisao.pt) |
| 18:46.34 | [sr] | hi :) |
| 18:46.43 | *** join/#asterisk Alagar (~Administr@122.164.39.24) |
| 18:56.59 | *** join/#asterisk Alagar (~Administr@122.164.39.24) |
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| 19:22.43 | daemon | hey guys is there a way to rbing up a console or such for asterisk (debug purpose) |
| 19:23.34 | x-demon | daemon, asterisk -r |
| 19:23.38 | daemon | ty |
| 19:23.47 | x-demon | then core set verbose 10 for more verbosity |
| 19:30.17 | [sr] | one thing |
| 19:30.28 | [sr] | i don't know the name in english |
| 19:30.50 | [sr] | are you seeing the external rings?, that ring when a call is received? |
| 19:35.59 | *** join/#asterisk brezular (~brezular@adsl-dyn215.78-98-244.t-com.sk) |
| 19:48.11 | Alton35 | Goal! ;-) |
| 19:48.40 | Godfather_ | 1-1 |
| 19:48.42 | Godfather_ | :) |
| 19:49.14 | Alton35 | Those Ghana guys are all over the place. We're lucky to get anything. |
| 19:50.09 | Godfather_ | They're more athletics |
| 19:50.35 | Alton35 | yup |
| 19:51.02 | Alton35 | and so accomplished that the ref almost never calls a penalty on them ;-) |
| 19:51.11 | Godfather_ | btw, i was big surprised how usa played previous rounds, very well |
| 19:51.27 | *** join/#asterisk sgimeno (~chatzilla@95.122.8.54) |
| 19:51.41 | Alton35 | This isn't the sort of thing that we follow much here, so we're just as surprised. |
| 19:52.11 | Godfather_ | Here spain ^^ |
| 19:53.06 | Alton35 | here the US |
| 19:53.32 | Alton35 | mucho gusto |
| 19:53.39 | Godfather_ | n1 :-) |
| 19:55.09 | Entulho | If I have an asterisk behind NAT, do I solve the problems using a SIP proxy, or SIP proxy is not to solve this? |
| 20:00.13 | drmessano | ~sipnat |
| 20:00.14 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 20:00.31 | drmessano | Asterisk will work fine behind a NAT. Configure it properly. |
| 20:01.57 | *** join/#asterisk crazybyte (~crzp@unaffiliated/crazypenguin/x-000001) |
| 20:05.33 | Entulho | ok, thanks |
| 20:09.14 | *** join/#asterisk RypPn (~TuMbL@rosscom.co.uk) |
| 20:25.36 | *** join/#asterisk voxter (~voxter@189.182.29.65) |
| 20:28.10 | MohsenSaeedi | Goal! |
| 20:28.36 | MohsenSaeedi | I couldn't use dahdi show channels in asterisk! |
| 20:29.00 | MohsenSaeedi | but i installed my openvox card and it's detected by linux. what is wrong? |
| 20:29.28 | ChannelZ | did you rebuild asterisk after having built dahdi? (I assume you didn't have it before) |
| 20:29.59 | MohsenSaeedi | ChannelZ: i installed asterisk and dahdi with rpm |
| 20:30.38 | ChannelZ | then did you configure /etc/dahdi/system.conf and /etc/asterisk/chan_dahdi.conf appropriately? |
| 20:31.27 | MohsenSaeedi | ChannelZ: I configured it with dahdi_genconf |
| 20:31.54 | MohsenSaeedi | ChannelZ: I can paste it to paste bin if you want |
| 20:32.53 | ChannelZ | genconf makes a file which is not normally automatically read by asterisk |
| 20:33.21 | ChannelZ | I forget what it's called, dahdi_channels.conf or something |
| 20:33.58 | ChannelZ | it's NOT a complete chan_dahdi.conf file, the intention is that you setup some global options and then include the dahdi_channels.conf file |
| 20:34.05 | MohsenSaeedi | ChannelZ: you mean dahdi-channels.conf? |
| 20:34.49 | *** join/#asterisk nicoAMG (~nicoamg@201.237.49.131) |
| 20:35.02 | MohsenSaeedi | ChannelZ: I configured all of them with elastix. but when i reboot my system then asterisk couldn't detect dahdi! and i can't use dahdi show channels for example. |
| 20:35.04 | Godfather_ | MohsenSaeedi, maybe you forget to add to the end of chan_dahdi.conf "#include dahdi-channels.conf" |
| 20:35.48 | MohsenSaeedi | Godfather_: no. elastix add this line automatically |
| 20:37.41 | Godfather_ | MohsenSaeedi,what returns lsdahdi ? |
| 20:38.46 | MohsenSaeedi | Godfather_: http://fpaste.org/WpN4/ |
| 20:39.25 | Godfather_ | seems ok |
| 20:39.47 | MohsenSaeedi | Godfather_: yes. but asterisk doesn't work with dahdi! |
| 20:40.33 | MohsenSaeedi | Godfather_: i don't have deep knowledge for debugging this problem . please help me. that's very bad for me if i can't solve this problem |
| 20:44.19 | WIMPy | MohsenSaeedi: Is chan_dahdi loaded? Try to turn up verbose and debug and to load it manually. That should tell you what's going on. |
| 20:45.21 | MohsenSaeedi | WIMPy: should i run ls chan_dahdi on the asterisk console with high verbose? |
| 20:46.22 | WIMPy | Try module unload chan_dahdi.so then module load chan_dahdi.so. |
| 20:46.22 | MohsenSaeedi | WIMPy: which should i load manually? please help me. |
| 20:46.47 | Godfather_ | MohsenSaeedi, try this: service dahdi stop, service asterisk stop, service dahdi start, service asterisk start |
| 20:47.33 | MohsenSaeedi | Godfather_: I tested it. but didn't solve my problem |
| 20:48.07 | MohsenSaeedi | WIMPy: sir, my output is here http://fpaste.org/O8qw/ |
| 20:48.11 | *** join/#asterisk Z_God (~julius@88.128.90.143) |
| 20:48.46 | WIMPy | That looks good to me. |
| 20:49.13 | Godfather_ | MohsenSaeedi, dahdi show status ? |
| 20:49.17 | MohsenSaeedi | WIMPy: that's good? i see some error! Unable to load module chan_dahdi.so |
| 20:49.18 | MohsenSaeedi | Command 'module load chan_dahdi.so' failed. |
| 20:49.52 | *** join/#asterisk obnauticus (~obnauticu@about/windows/regular/obnauticus) |
| 20:50.03 | Godfather_ | thats not good.. :| |
| 20:50.22 | WIMPy | Errr. Indeed. Interesting. It obviouselt gets loded, however. |
| 20:50.25 | MohsenSaeedi | Godfather_: i don't have this command |
| 20:50.41 | Godfather_ | yep, i forgot sorry, you dont have dahdi at the moment |
| 20:50.59 | Godfather_ | re-install dahdi |
| 20:51.07 | MohsenSaeedi | Godfather_: my problem is very interesting :D |
| 20:51.17 | MohsenSaeedi | Godfather_: ok. i will do it |
| 20:53.21 | MohsenSaeedi | Godfather_: now system is reinstalling dahdi packages |
| 20:55.40 | [sr] | hi WIMPy |
| 20:55.55 | WIMPy | hi [sr] |
| 20:56.06 | *** join/#asterisk pabelanger-lap (~pabelange@CPE001fe2a8fd1d-CM0012254094b2.cpe.net.cable.rogers.com) |
| 21:04.45 | [sr] | WIMPy: got a question :) |
| 21:06.57 | MohsenSaeedi | WIMPy and Godfather_: i reinstalled dahdi but no change! |
| 21:07.33 | MohsenSaeedi | WIMPy: do you want to ssh to my box? |
| 21:09.24 | WIMPy | MohsenSaeedi: I'm sorry, but after that rather interesting paste I have reached the end of my knwledge. |
| 21:09.51 | Godfather_ | MohsenSaeedi, let me try |
| 21:10.05 | Godfather_ | but i've no idea how elastix works |
| 21:10.12 | MohsenSaeedi | Godfather_: ok, wait . |
| 21:10.28 | MohsenSaeedi | Godfather_: elastix is not important in this case |
| 21:10.32 | Godfather_ | i suppose. |
| 21:10.44 | MohsenSaeedi | this problem is exist between asterisk and dahdi module |
| 21:11.24 | Godfather_ | MohsenSaeedi, maybe you could install dahdi from sources instead of rpm |
| 21:11.33 | [sr] | WIMPy: i dont know the correct name but i'll try to explain, do you see the external rings, normally fire departsments have it, when the phone rings, that ring also rings so that everyone can listen the fone |
| 21:11.37 | MohsenSaeedi | Godfather_: maybe |
| 21:11.55 | [sr] | WIMPy: how could be integrated/configured with voip? is there any special deviced for voip ? |
| 21:11.56 | *** join/#asterisk TimeRider (steve@5ac3186e.bb.sky.com) |
| 21:12.30 | Godfather_ | [sr], a megaphone? |
| 21:12.49 | [sr] | Godfather_: dont know if thats the name, let me try to find a pic on google... sec |
| 21:12.50 | Godfather_ | this could be a funny riddle |
| 21:13.15 | MohsenSaeedi | Godfather_: do you received information about my box on private message? |
| 21:13.15 | Godfather_ | [sr], whats your primary language? |
| 21:13.24 | [sr] | Godfather_: PT |
| 21:13.33 | Godfather_ | [sr], in PT is..? |
| 21:13.38 | [sr] | portuguese :) |
| 21:13.46 | Godfather_ | i know |
| 21:13.48 | Godfather_ | the word |
| 21:14.32 | [sr] | Godfather_: well, campainha de telefone |
| 21:15.10 | [sr] | Godfather_: WIMPy, ring's like this one: http://www.fleshtel.com.br/loja/produtos/p_CAMPAINHA%20AL6_PQ.bmp |
| 21:15.10 | WIMPy | [sr]: So you just need an extra (extra loud) bell? |
| 21:15.22 | [sr] | WIMPy: that may be the correct word :) |
| 21:16.07 | [sr] | normally they are fixed on the wall |
| 21:16.09 | Godfather_ | jingle bells |
| 21:16.12 | Godfather_ | hehe |
| 21:16.12 | WIMPy | [sr]: There are one for POTS, but using Asterisk you could control round about anything via some script. |
| 21:16.44 | WIMPy | Godfather_: I guess it's more about decibells :-) |
| 21:17.27 | *** join/#asterisk Krolik13 (~Krolik13@5ac313cb.bb.sky.com) |
| 21:17.37 | [sr] | WIMPy: hum, doesn't exist any kind of this bell's, that have full integration with asterisk? only workarrounds? |
| 21:18.10 | WIMPy | I have not seen simple bells with SIP support. |
| 21:18.28 | [sr] | hum |
| 21:18.39 | [sr] | i may query my supplier about this |
| 21:18.39 | Krolik13 | sorry, i'm a little bit confused. I have two sip users, and one gateway with two FXO ports. How can i configure that each user's call to be routerd to spefixic FXO port. Let say, user1=FXO1, user2=FX02. |
| 21:18.51 | WIMPy | But you could recycle any phone with an optical ringin indication as an hardware interface. |
| 21:20.22 | WIMPy | [sr]: I guess I'd wire it to the servers parallel or serial port. |
| 21:20.22 | [sr] | WIMPy: hum, an ideia, but i have to study one thing 1st that is how the things work now with "normal" phones/PBX's |
| 21:20.49 | [sr] | WIMPy: hum...an interesting ideia, i know a few things about eletronic and could build something to make it ring |
| 21:21.29 | WIMPy | [sr] either via a pots line (directely in parallel to the phone) or via an extra alarm output. |
| 21:21.33 | Krolik13 | sorry, i'm a little bit confused. I have two sip users, and one gateway with two FXO ports. Asterisk in the middle. How can i configure that each user's call to be routerd to spefixic FXO port. Let say, user1=FXO1, user2=FX02. How to force user to get specific extension in dialplan? |
| 21:22.17 | Jomu | by his default context |
| 21:22.35 | Jomu | Krolik13 give user a context, and route him by that context |
| 21:22.38 | Krolik13 | Jomu> default context as i know works just for incomming calls... no? |
| 21:22.50 | [sr] | WIMPy: going to check about that :) |
| 21:23.32 | Jomu | Krolik13 default does... but give user his context, meaning his default |
| 21:24.26 | Krolik13 | ok, thanks |
| 21:24.27 | Krolik13 | let me try |
| 21:29.12 | *** join/#asterisk niekvlessert (~niek@82-171-252-6.ip.telfort.nl) |
| 21:34.46 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
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| 21:49.25 | Krolik13 | i've created a new extension content, and I thought that using exten => s,1, will cover all extensions by default. But unfortunately asterisk gives me a error that it doesn't find any extension. why? |
| 21:50.25 | Gugge | s is not a default extension |
| 21:50.30 | Gugge | its just an extension named s |
| 21:50.35 | Krolik13 | which one is default? |
| 21:50.40 | Gugge | none |
| 21:50.46 | Krolik13 | hmm |
| 21:50.54 | Gugge | you _can_ use i (invalid), but i wouldnt recommend it |
| 21:51.06 | Krolik13 | ooo, cool idea :D |
| 21:51.11 | Gugge | no its not |
| 21:51.15 | Krolik13 | why you don't receomment it? |
| 21:51.27 | WIMPy | Krolik13: Maybe you should take a look at the book. |
| 21:51.32 | WIMPy | ~book |
| 21:51.33 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 21:51.53 | Krolik13 | thanks for the link |
| 22:08.25 | *** join/#asterisk mifadir (29891eab@gateway/web/freenode/ip.41.137.30.171) |
| 22:14.00 | *** join/#asterisk voxter (~voxter@189.182.29.65) |
| 22:18.23 | pabelanger-lap | Krolik13: exten => _.,1,blah() |
| 22:18.37 | pabelanger-lap | Is a wildcard exten, but NOT recommend |
| 22:20.34 | p3nguin | Yeah, terrible idea. |
| 22:22.28 | p3nguin | gugge: The i extension is only "invalid" from interactive commands, such as voicemail, directory, BackGround, or WaitExten. It won't catch non-matching extensions otherwise. |
| 22:23.03 | *** part/#asterisk sekil (~Ognjen@80.93.247.26) |
| 22:23.17 | p3nguin | krolik13: You'd be better off defining the extension that you are trying to reach or at least a pattern that matches it. |
| 22:23.22 | WIMPy | p3nguin: It does. |
| 22:28.05 | p3nguin | wimpy: Are you trying to get me to believe that 'i' will match if there is no other extension in a context? 'Cause it won't. |
| 22:28.54 | WIMPy | It does for me. |
| 22:29.07 | p3nguin | I don't see how. |
| 22:29.40 | p3nguin | It should give you a busy tone and the CLI should show you that the call to exten <something other than i> was rejected because it was not found. |
| 22:29.54 | Godfather_ | WIMPy, i believe you are wrong |
| 22:30.08 | WIMPy | It goes to i, that's what it always did. |
| 22:30.19 | Godfather_ | try to do a Goto(asdf,1) |
| 22:30.27 | WIMPy | But maybe that's because I use overlap dialling. |
| 22:30.31 | p3nguin | It can't just magically go to 'i' |
| 22:30.45 | Godfather_ | in my case no. |
| 22:31.20 | p3nguin | If you have a single context where calls are routed, and the only exten in it is 'i' ... unless you dial 'i' there is no match, so the call fails. |
| 22:31.22 | WIMPy | Goto is quite different from dialling. |
| 22:31.24 | Krolik13 | what is the best one port E1 voice gateway? |
| 22:31.29 | WIMPy | (unfortunaletly) |
| 22:32.12 | WIMPy | I obviousely have other extensions, but if none is matched, it goes to i. |
| 22:32.24 | p3nguin | The CLI will show you call from <you> to extension <something> rejected because extension not found. |
| 22:32.54 | p3nguin | If it doesn't, your test isn't valid to begin with. |
| 22:33.03 | WIMPy | Seems to depend on the channel used. |
| 22:33.24 | p3nguin | Which tech are you using for the test? |
| 22:33.36 | WIMPy | lcr |
| 22:33.53 | WIMPy | misdn used to work the same. |
| 22:34.27 | WIMPy | sip does not, but IIRC SCCP did as well. |
| 22:35.07 | p3nguin | dialplan should be completely independent of the channel technology used, so I don't see how it could matter. |
| 22:35.32 | WIMPy | Looks like it isn't. |
| 22:36.22 | WIMPy | I think it worked with IAX as well, but I switched that link to dundi. |
| 22:36.48 | p3nguin | I'm going to test sccp. I'm sure it will behave the same as sip, though. |
| 22:37.53 | WIMPy | cant test sccp as that ceased to work some time. |
| 22:37.56 | p3nguin | Yep, tones. Sounded like congestion tone rather than busy, though. |
| 22:40.31 | p3nguin | I don't have an IAX2 phone handy right now, so I can't test it, but I bet it would behave exactly the same as SIP and SCCP do. |
| 22:41.51 | [sr] | well |
| 22:41.54 | [sr] | going to sleep |
| 22:41.56 | [sr] | see ya |
| 22:46.53 | Entulho | Using a client with STUN, and a SIP Proxy with RTPProxy, shoud be enough to make this client behind a NAT to connect to an external Asterisk?? |
| 22:47.46 | WIMPy | Nothing will usually do just as well. |
| 22:47.53 | WIMPy | ~sipnat |
| 22:47.54 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 22:54.04 | *** join/#asterisk timholum_droid (~AndChat@48.sub-72-111-46.myvzw.com) |
| 22:54.22 | timholum_droid | Hello everyone |
| 23:02.59 | *** join/#asterisk jksM (jks@193.189.93.254) |
| 23:14.44 | WIMPy | Just trying to get chan_skinny up again. |
| 23:15.20 | p3nguin | chan_skinny sucks; use chan_sccp instead. |
| 23:15.34 | WIMPy | Even if I copy the current example skinny.conf and uncomment the devices, skinny show devices stays empty. |
| 23:41.27 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 23:43.58 | *** part/#asterisk The_Blob (~blob@S010600c00cb01b8c.vs.shawcable.net) |
| 23:49.47 | WIMPy | Doesn't look like I have much luck with chan_sccp. |
