| 00:00.18 | nightrid3r | sounds like it, |
| 00:01.25 | nightrid3r | voip uses tunneling and a vpn is also a tunnel which requires a very good config or all sorts of nasty will happen |
| 00:02.29 | *** join/#asterisk xpot-mobile (~james@166-70-100-198.ip.xmission.com) |
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| 00:22.26 | WIMPy | Are you sure it is using the tunnel and not trying to use a public IP? |
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| 00:35.51 | Beltechs | its pointed to the internal server ip |
| 00:37.54 | *** join/#asterisk kaushal (~kaushal@115.246.224.177) |
| 00:37.59 | kaushal | Hi |
| 00:39.22 | *** join/#asterisk binbash_ (~peter@a83-161-240-87.adsl.xs4all.nl) |
| 00:39.42 | kaushal | Can someone please guide me about my issue posted on http://lists.digium.com/pipermail/asterisk-users/2011-July/264304.html |
| 00:40.52 | jc319 | http://pythianpang.wordpress.com/2011/06/24/solution-bash-ulimit-open-files-cannot-modify-limit-operation-not-permitted/ |
| 00:41.04 | jc319 | have a look kaushal.. |
| 00:43.53 | kaushal | jc319: so set nofile to 65536 ? |
| 00:46.02 | jc319 | ulimit -Hn |
| 00:46.17 | jc319 | what do you have there? |
| 00:46.43 | kaushal | sure |
| 00:47.02 | kaushal | jc319: please give me a moment |
| 00:52.26 | p3nguin | That just gave me an idea. I should probably use a VPN and then I can use my SCCP phone easier from a remote location. |
| 00:52.51 | p3nguin | I was going to switch over to SIP and forego any special networking configuration. |
| 01:00.22 | p3nguin | VoIP uses tunneling, huh? How come mine doesn't use tunneling? |
| 01:01.19 | jc319 | kaushal: What is the output of ulimit -Hn ? |
| 01:01.44 | kaushal | jc319: i am not able to reach the server |
| 01:01.50 | kaushal | please give me a moment |
| 01:06.59 | p3nguin | Do you think if you keep waiting that you'll be able to reach it? |
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| 02:44.36 | p3nguin | That's the fastest poweroff and cold boot up of all time! |
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| 03:45.47 | atan | Anyone know what headset port the avaya phones have? Is there a difference between RJ9 and RJ10? I'm trying to add a headset to one but am not sure which port it has on it. =\ |
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| 04:28.14 | techknowlust | http://pastie.org/2153154 trying to get a very simple 2 user setup going on asterisk |
| 04:29.52 | techknowlust | but for some reason calls aren't working between the only two phones registered to the system. http://pastie.org/2153157 these are the config files |
| 04:30.01 | techknowlust | any suggestions would be very much appreciated |
| 04:31.27 | WIMPy | Define "not working". |
| 04:32.54 | techknowlust | the calls "time out" |
| 04:33.23 | techknowlust | I have a softphone on my laptop and the 2nd user configured on my android phone. |
| 04:33.45 | p3nguin | Not working. You know... doesn't work. |
| 04:33.51 | techknowlust | when one calls the other it doesn't receive the call and it just times out |
| 04:35.19 | p3nguin | Does either of those pastes show sip show peers? |
| 04:35.41 | techknowlust | the 1st |
| 04:37.05 | p3nguin | And have you watched sip debug when trying a call which fails? |
| 04:39.07 | techknowlust | I have asterisk running on verbose mode, though now that you mention this, that might not be enough |
| 04:40.15 | p3nguin | It's not even close. core set verbose 0 sip set debug on |
| 04:40.24 | p3nguin | At this point, you don't need the verbose at all. |
| 04:40.57 | p3nguin | The sip debug will show you what happens with the actual call, if anything. |
| 04:41.01 | techknowlust | ok. |
| 04:42.14 | techknowlust | hmm giving a 404 error ? |
| 04:42.26 | p3nguin | 404 is what? |
| 04:42.31 | techknowlust | not found |
| 04:42.56 | p3nguin | Okay, sounds like Asterisk doesn't know about the peer that you're calling. |
| 04:43.04 | techknowlust | weird. |
| 04:43.15 | p3nguin | Now we go back to verbose 4. |
| 04:43.48 | p3nguin | This time, I want you to make the call and capture the entire thing from start to finish, then paste the entire thing in the pastebin. |
| 04:44.03 | p3nguin | verbose and sip debug together will be fine. |
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| 04:44.53 | techknowlust | ok will do |
| 04:45.30 | p3nguin | My guess is a bad extension, but when I see the paste I might be more comfortable saying this with certainty. |
| 04:48.57 | techknowlust | http://pastie.org/2153192 |
| 04:51.34 | p3nguin | I didn't see any NAT settings at all in your sip.conf. |
| 04:51.49 | p3nguin | But you have private addresses floating around in your sip debug. |
| 04:52.06 | p3nguin | So at least some NAT settings will be required. |
| 04:52.23 | techknowlust | ok |
| 04:53.29 | p3nguin | You also have a crapload of unnecessary codecs being used. Consider changing that after you get calls going. |
| 04:53.31 | techknowlust | to give context this asterisk instance is running on a vps and both the phone and laptop are on a separate network |
| 04:53.57 | p3nguin | That's fine, as long as you have Asterisk's network settings right. |
| 04:55.48 | techknowlust | is there any particular guide to look to for this ? |
| 04:55.59 | p3nguin | ~book |
| 04:55.59 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 04:56.31 | nightrid3r | ~buybook |
| 04:56.31 | infobot | You can buy "Asterisk: The Definitive Guide" at http://oreilly.com/catalog/9780596517342 so go buy it SERIOUSLY |
| 04:59.24 | coppice | can it be bought FRIVOLOUSLY? |
| 05:05.56 | *** join/#asterisk suma (~pongada@c-98-245-176-77.hsd1.co.comcast.net) |
| 05:22.28 | techknowlust | hmm new problem. moved the asterisk instance to the local network to sort nat problems |
| 05:22.42 | techknowlust | calls go through but die as soon as they are answered |
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| 06:41.12 | KNERD | jingle anyone? |
| 07:10.40 | suma | knerd: what'z up ? |
| 07:11.13 | KNERD | jingle module not appearing |
| 07:11.25 | suma | you mean google talk ? |
| 07:11.36 | KNERD | j<tab> only show Jabber..no jingle |
| 07:11.42 | KNERD | corrsct GT |
| 07:11.51 | suma | which version of asterisk you are using ? |
| 07:12.10 | KNERD | 1.8.4.2 |
| 07:12.34 | KNERD | it is compiled into it |
| 07:12.43 | suma | hold on |
| 07:12.49 | suma | are you using linux ? |
| 07:12.55 | suma | or wrt ? |
| 07:13.03 | KNERD | linux |
| 07:14.10 | suma | is chan_gtalk is loaded ? |
| 07:14.49 | KNERD | yes...all that is..only jingle missing |
| 07:15.10 | KNERD | trying to get incoming calls |
| 07:16.17 | suma | what is the error you get ? |
| 07:17.23 | KNERD | no errors |
| 07:17.50 | KNERD | unless.. |
| 07:18.05 | suma | i don't see jingle in my module list |
| 07:18.12 | KNERD | No such command 'jingle reload' (type 'core show help jingle reload' for other possible commands) |
| 07:18.16 | suma | but i receive gtalk and make gtalk calls |
| 07:18.38 | suma | i think jingle protocol is included in chan_gtalk |
| 07:18.41 | KNERD | well I dont get calls and was told I need jingle |
| 07:19.31 | KNERD | and chan_jingle is a valid module |
| 07:19.57 | suma | good |
| 07:20.05 | suma | i don't use that one for making calls to google |
| 07:20.11 | suma | i use chan_gtalk |
| 07:20.18 | KNERD | i dont either. I want to RECEIVE calls |
| 07:20.45 | suma | do you have chan_gtalk and res_jabber loaded ? |
| 07:21.00 | KNERD | of course |
| 07:21.04 | KNERD | I make calls just fine |
| 07:21.23 | suma | may i see your gtalk.conf |
| 07:21.26 | suma | and jabber.conf ? |
| 07:21.32 | suma | you can mask the password |
| 07:21.50 | suma | and also extension.conf which it receives the call |
| 07:21.58 | suma | *i mean the context |
| 07:22.36 | suma | paste everything on pastebin |
| 07:22.45 | suma | ~pastebin |
| 07:22.45 | infobot | [~pastebin] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 07:23.13 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 07:24.01 | KNERD | jabber http://pastebin.com/kf6Xa6dB |
| 07:25.06 | KNERD | gtalk http://pastebin.com/tp6TWfx3 |
| 07:28.31 | suma | KNERD: gtalk and jabber seems ok |
| 07:28.40 | suma | may I have a look at the googlein context ? |
| 07:28.58 | suma | in extensions.conf |
| 07:30.00 | KNERD | Google In http://pastebin.com/Nu4kyfM9 |
| 07:30.33 | suma | found a problem on your gtalk.conf |
| 07:30.52 | suma | can you guess it ? |
| 07:31.59 | KNERD | ulaw? |
| 07:32.38 | suma | nope |
| 07:32.45 | suma | let me update the file |
| 07:32.50 | suma | and you can copy and paste it |
| 07:33.18 | KNERD | ok |
| 07:34.07 | suma | KNERD: http://pastebin.com/3kh0T3Fw |
| 07:34.20 | suma | you missed the very important connection |
| 07:34.50 | suma | possible shutdown and start asterisk |
| 07:35.10 | suma | and make a sample call |
| 07:36.59 | KNERD | trying now |
| 07:37.39 | KNERD | suma: thanks..that got it! |
| 07:38.09 | suma | knerd: glad I could make it to work |
| 07:38.49 | KNERD | strange i missed that |
| 07:39.09 | suma | It happens, with the copy and paste world |
| 07:39.33 | KNERD | yes it does |
| 07:40.50 | suma | KNERD: no plans for the long weekend ? |
| 07:41.56 | KNERD | no |
| 07:42.38 | KNERD | kinda tells you someone trying to use google voice for free incoming and out going calls is probably broke |
| 07:43.06 | suma | means ? |
| 07:43.29 | suma | I was trying to use G722 along with that |
| 07:44.07 | suma | I'm planning to upgrade my cisco phone |
| 07:44.16 | suma | since cisco 7960 does not support G722 |
| 07:45.19 | KNERD | no dinero |
| 07:45.38 | KNERD | upgrade? |
| 07:45.49 | KNERD | you mean newer phone or firmware? |
| 07:47.32 | suma | to a newer phone |
| 07:47.48 | suma | I doubt cisco 7960 has the ability to decode g722 |
| 07:48.01 | suma | planning to buy 7970G |
| 07:50.20 | KNERD | what you going to do with the old used up dinosaur? |
| 07:50.38 | suma | drop it in ebay for a sale |
| 07:50.50 | KNERD | $10 or something? |
| 07:50.50 | suma | Do you need one ? |
| 07:51.05 | suma | Current price is $90+ for cisco 7960 |
| 07:51.07 | KNERD | I wish I could have one.. I been using ATA for 3 years |
| 07:51.24 | suma | that is the price in ebay |
| 07:51.25 | KNERD | and I dont like it |
| 07:51.40 | suma | cisco offers beter sound quality, if you are more into voice quality ! |
| 07:51.58 | suma | Also the microphone is good as well. |
| 07:52.34 | KNERD | Well...Linksys claims to have a professional quality ATA, but I have not tested it. I got the consumer version for cheap |
| 07:53.10 | suma | KNERD: tried those cheap ones, will work fine for a year, after that I need to take it to bin. |
| 07:53.34 | suma | I almost tried every ata's and voip phones in the market |
| 07:53.37 | KNERD | mine has served me well for 3 years now |
| 07:53.54 | KNERD | but i would rather have a VoIP phone |
| 07:54.04 | KNERD | with all the neat features |
| 07:54.07 | suma | yep |
| 07:54.13 | suma | cheapy one could be from polycom |
| 07:54.21 | suma | which has HD as well |
| 07:54.59 | KNERD | Well I been studying that "HD" only a different codec that has been around for a while |
| 07:55.28 | KNERD | heard also it is heavy CPU usage |
| 07:55.34 | suma | g722 offers very good |
| 07:55.39 | suma | yes |
| 07:55.50 | suma | latest phones has g722 built in |
| 07:55.53 | KNERD | I have seen some cheap cisco on ebay for $25 |
| 07:56.14 | suma | basic phone ? |
| 07:56.20 | KNERD | VERY basic |
| 07:56.37 | KNERD | one line LCD |
| 07:56.54 | suma | it will not give alll the comfort for making calls |
| 07:57.02 | suma | checking voicemails |
| 07:57.04 | suma | etc., |
| 07:57.21 | suma | bigger screen |
| 07:57.24 | suma | etc., |
| 07:57.33 | suma | try a 7970G |
| 07:57.37 | suma | it is very good |
| 07:57.41 | KNERD | some year i will get one |
| 07:58.00 | suma | you think $150 is too costly for a phone ? |
| 07:58.17 | suma | *quality phone |
| 08:01.20 | KNERD | it is when you are a starving student |
| 08:02.01 | KNERD | overqualitifed for fast food, under for the profesion |
| 08:02.16 | suma | if you are a student then you must only softphone |
| 08:02.21 | suma | no hardphones ;) |
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| 08:03.55 | KNERD | i did until my logitech $35 USB headset broke down. |
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| 08:04.15 | KNERD | so I keep on using ATA |
| 08:04.48 | KNERD | besides, takes too long to answer phone on computer |
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| 08:32.21 | suma | KNERD: you must try iphone or cheap android phone and make the softphones on those ones |
| 08:51.00 | The_REV | suma: KNERD: android phones with gingerbread (2.3.X) have a built in SIP softphone works perfect with Asterisk |
| 08:51.42 | coppice | although only narrow band, and only over 802.11 |
| 08:52.04 | The_REV | not true coppice i use mine over 3g... |
| 08:52.14 | The_REV | it depends on the phone/carrier |
| 08:52.30 | coppice | which phone? |
| 08:53.21 | The_REV | Nexus phones... anything loaded with Cyanogenmod... |
| 08:53.44 | coppice | oh, well, if you change the software anything is possible |
| 08:54.06 | The_REV | but the Nexus phone I know all do it stock... |
| 08:54.16 | nightrid3r | and there's lots of sip clients in android market |
| 08:54.19 | coppice | it has nothing to do with the carriers, though. a retail nexus out of the box won't do SIP over 3G, or use wideband codecs |
| 08:55.17 | The_REV | well then flash it with a good ROM... :) |
| 08:57.41 | coppice | its an awful phone, so such minor things don't bother me that much |
| 08:58.35 | The_REV | otherwise the 3cx client for android and iOS works brilliantly... |
| 08:59.15 | coppice | the nice thing about the built in SIP in android is the level of integration |
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| 08:59.57 | The_REV | true... damn near seemless... |
| 09:00.20 | The_REV | *seamless |
| 09:02.56 | KNERD | Then someone please buy me one. I can't afford those toys |
| 09:03.12 | coppice | you wouldn't like it |
| 09:04.48 | KNERD | i would anything that is free |
| 09:05.47 | coppice | didn't anyone ever teach you that there's no such thing as a free phone? |
| 09:07.51 | KNERD | if someone buys it for you it is |
| 09:08.02 | KNERD | free softphones |
| 09:08.18 | coppice | how foolish |
| 09:10.49 | KNERD | not if you are a starving student |
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| 09:27.58 | KNERD | so who is going to get me an android? |
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| 09:50.13 | \DSAFEW\ | your mom |
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| 10:22.03 | KNERD | nope |
| 10:22.10 | KNERD | one of you |
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| 18:45.29 | drynish | Wow I'm unable to configure correctly my VOIP 5801 even if I try anything |
| 18:45.39 | drynish | I don't know where it's failing |
| 18:46.34 | drynish | What's the correct way of debugging even with everything at full debug, I don't see any SIP going on on the asterisk server |
| 18:46.37 | \DSAFEW\ | drynish, try sniffing the tcp stream |
| 18:46.52 | \DSAFEW\ | drynish, wireshark is nice |
| 18:47.04 | drynish | Ok but it's a small astlinux distro |
| 18:48.37 | drynish | So I cannot use a wireshark on it |
| 18:50.10 | drynish | I think it's just the fxo device I'm trying to use, it's very very difficult to configure |
| 18:50.44 | \DSAFEW\ | drynish, dahdi_tool give a green light? |
| 18:51.30 | drynish | It's a zoom 5801 |
| 18:51.34 | drynish | I don't know dahdi |
| 18:51.52 | drynish | How can I know more about asterisk |
| 18:52.26 | drynish | I've read the asterisk book future of telephony |
| 18:52.34 | drynish | but I think it's a bit outdated today |
| 18:53.18 | \DSAFEW\ | looks like a bad review: http://www.voip-info.org/wiki/view/Zoom+5801 |
| 18:53.38 | drynish | yes |
| 18:53.39 | drynish | a really bad |
| 18:53.54 | \DSAFEW\ | ~book |
| 18:53.54 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 18:55.18 | \DSAFEW\ | I really don't know anything about your voip device, so I can't say whether there are ways to export logs or check on status |
| 18:55.29 | \DSAFEW\ | sorry, I hope that link helped |
| 18:55.53 | drynish | I will try :) |
| 18:55.56 | drynish | to read more about it |
| 18:55.59 | drynish | and test drive it more |
| 18:56.06 | \DSAFEW\ | "Someone at Zoom has a sence of humor. I got a response to my sip command that stated it was a COFFEE POT not a sip server. :-o) LOL!!! " |
| 18:56.16 | drynish | Can you tell me why the dialplan by default is so full of ... ;) |
| 18:56.17 | \DSAFEW\ | :| |
| 18:56.23 | drynish | ? |
| 18:56.30 | drynish | Is it all usefull? |
| 18:57.29 | \DSAFEW\ | the default dialplan is a demo to test everything, it's good for a few things, but start from scratch when creating a real dialplan |
| 18:57.50 | \DSAFEW\ | it shows you the syntax, and has a monkey recording |
| 18:58.06 | \DSAFEW\ | maybe not in the zoom version |
| 18:58.33 | drynish | the zoom doesn't have a dialplan |
| 18:58.43 | drynish | All I really want is just making a voicemail of it ;) |
| 18:59.27 | \DSAFEW\ | well, you have a lot of reading to do :) |
| 19:00.13 | drynish | yeah :) |
| 19:00.15 | drynish | I will read all that |
| 19:00.35 | drynish | I've read a lot about asterisk around two-three years ago |
| 19:00.41 | drynish | I can't believe it has changed so much! |
| 19:00.47 | *** join/#asterisk AviMarcus (~avi@109.67.184.240) |
| 19:01.56 | AviMarcus | I've been doing residential & biz VoIP.. and calling cards.. but I realize I have some awesome (price + quality) routes for termination and origination in Israel. Can anyone clue me into the wholesale business, for international routes? |
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| 21:22.49 | jeffik | AviMarcus: do you know avout Arbinet? |
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| 23:22.39 | RickB17 | i'm attempting to get a backtrace and locate the core file; but cannot locate the core file. any suggestions? |
| 23:22.47 | RickB17 | trying to follow https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace |
| 23:47.02 | RickB17 | nevermidn, i found the core dump in /tmp. I was looking for "core", not core.* |