| 00:29.48 | *** join/#asterisk lorsungcu (~anonymous@216.160.1.105) |
| 00:41.44 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.143) |
| 00:47.21 | *** join/#asterisk Vutral (ss@mirbsd/special/Vutral) |
| 01:11.29 | *** join/#asterisk DynamicFail (~Administr@198.99.129.159) |
| 01:11.50 | DynamicFail | ~log |
| 01:11.51 | infobot | All conversations are logged to http://infobot.rikers.org/%23asterisk/ Lines starting with spaces are not logged. Logs are updated daily. |
| 01:13.11 | DynamicFail | So I have been having a real issue trying to get some android phones to register to my box |
| 01:13.25 | DynamicFail | so I decided to not have them register |
| 01:13.32 | DynamicFail | I changed qualify=no |
| 01:13.36 | DynamicFail | allowguest=yes |
| 01:13.41 | DynamicFail | allowsubscribe=yes |
| 01:14.00 | DynamicFail | and in each phone's config in sip.conf defined it as host=<phone static ip> |
| 01:14.10 | DynamicFail | the phones were still not able to call each other |
| 01:18.52 | ChannelZ-Wk | you don't need allowguest just to use static IP, but nevermind |
| 01:18.58 | ChannelZ-Wk | What is "not able to call each other" |
| 01:22.02 | DynamicFail | well I dial the other's number and the phone doesn't ring |
| 01:22.17 | DynamicFail | I just stripped my entire plan to the most simple dialplan/config I could think of |
| 01:22.21 | DynamicFail | still issues |
| 01:22.25 | DynamicFail | one sec will pastebin |
| 01:23.00 | ChannelZ-Wk | Would need to see sip debug. |
| 01:29.44 | DynamicFail | http://pastebin.com/f94nDZYK |
| 01:29.49 | DynamicFail | ChannelZ-Wk, |
| 01:29.55 | DynamicFail | configs, and debug |
| 01:31.29 | ChannelZ-Wk | What are you typing into your phones for login information? They are not bothering to try to auth. |
| 01:34.41 | DynamicFail | Account Name: 3140, Caller ID, inTouch 3140 |
| 01:34.44 | DynamicFail | Server 192.168.3.1 |
| 01:34.49 | DynamicFail | Username 3140 |
| 01:34.52 | DynamicFail | password: xxx |
| 01:34.58 | DynamicFail | that's it |
| 01:35.27 | DynamicFail | I have tried with csipsimple |
| 01:35.32 | DynamicFail | and zoiper |
| 01:35.44 | DynamicFail | hmm this is odd |
| 01:35.49 | DynamicFail | when i do show dialplan |
| 01:35.53 | DynamicFail | it shows an old dialplan |
| 01:36.03 | DynamicFail | even though I have saved the sip.conf, and extensions.conf |
| 01:36.07 | DynamicFail | and done a reload in the CLI |
| 01:36.25 | Penguin | There was no call in that pastebin. |
| 01:36.25 | DynamicFail | and y astetcdir is /etc/asterisk |
| 01:36.46 | DynamicFail | Penguin, correct, I went back to just trying a very very simple config |
| 01:36.55 | DynamicFail | stripped it all down to the min, and the phones still don't register |
| 01:36.59 | DynamicFail | this is just registration |
| 01:37.08 | Penguin | It's going to be awfully hard to debug a bad call when there is no call. |
| 01:37.37 | ChannelZ-Wk | if it's not authing on register, it's not going to auth a call either. |
| 01:37.42 | DynamicFail | Well I thought step 1, would be to get them to register, you think I should debug the call instead first? |
| 01:38.20 | ChannelZ-Wk | You can of course set it up to not require auth, but.. you're treating a symptom |
| 01:38.23 | Penguin | That's two separate issues. |
| 01:38.36 | DynamicFail | correct |
| 01:38.45 | DynamicFail | I guess I would like to get them to regiser first |
| 01:39.03 | Penguin | That seems like a good step. |
| 01:39.48 | DynamicFail | so I think I found a major problem... when I do "dialplan show" it loads an old dialplan even though I have a new one saved in extensions.conf |
| 01:40.07 | DynamicFail | It coudl very easily be doing the same with the sip.conf file (although I don't know how to show that from cli) |
| 01:40.13 | Penguin | Then you need to run "dialplan reload" to load whatever is in extensions.conf. |
| 01:40.13 | ChannelZ-Wk | different problem |
| 01:40.29 | DynamicFail | doesn't seem to work |
| 01:40.31 | Penguin | But the dialplan (extensions.conf) isn't related to the registration issue. |
| 01:40.39 | DynamicFail | I have been doing a full "reload" |
| 01:40.39 | Penguin | core set verbose 3 |
| 01:40.43 | Penguin | dialplan reload |
| 01:40.44 | DynamicFail | "dialplan reload" didn't help either |
| 01:41.01 | Penguin | Then go read the first several lines of output. |
| 01:41.15 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.130) |
| 01:43.21 | DynamicFail | == Parsing '/etc/asterisk/asterisk.conf': == Found |
| 01:43.34 | DynamicFail | <PROTECTED> |
| 01:43.45 | Penguin | dialplan reload doesn't parse asterisk.conf, as far as I know. |
| 01:43.49 | DynamicFail | but it definitely isn't loading what is in extensions.conf |
| 01:44.29 | DynamicFail | astetcdir => /etc/asterisk (in asterisk.conf) |
| 01:45.01 | [TK]D-Fender | DynamicFail: pastebin "ls -la /etc/asterisk" |
| 01:45.06 | [TK]D-Fender | including the call itself |
| 01:46.17 | DynamicFail | [TK]D-Fender, http://pastebin.com/6hWuH0C6 |
| 01:46.56 | [TK]D-Fender | Now why am I not rtrusting that symlink... |
| 01:47.21 | Penguin | /tmp/OS/etc/asterisk/extensions.conf ? |
| 01:47.26 | Penguin | ls -l /tmp/OS/etc/asterisk/extensions.conf |
| 01:48.31 | DynamicFail | -rw-r--r-- 1 root root 106 Dec 20 18:52 /tmp/OS/etc/asterisk/extensions.conf |
| 01:48.38 | DynamicFail | Hmm maybe the show dialplan is actually working |
| 01:48.56 | DynamicFail | does asterisk automatically add a whole bunch of stuff not in your extensions.conf? |
| 01:49.08 | Penguin | No. |
| 01:49.23 | Penguin | It only adds what the config files tell it to add. |
| 01:50.39 | Penguin | And when you turn up the verbosity and then run dialplan reload, it will show you all the config files it is using for creating the dialplan. |
| 01:51.03 | Penguin | So there won't be surprises. |
| 01:51.53 | DynamicFail | http://pastebin.com/jiKuNivH |
| 01:52.23 | DynamicFail | ok so that wasn't an issue |
| 01:52.29 | DynamicFail | it was me not knowing how ot read logs |
| 01:52.35 | DynamicFail | it's parsing other files I'm not using |
| 01:52.39 | *** part/#asterisk gtxcomm (~gtxcomm@72.135.4.232) |
| 01:52.56 | Penguin | extensions.conf and users.conf are the only two files it is reading for dialplan. |
| 01:53.11 | Penguin | Those are the only two files it parsed at all. |
| 01:55.39 | DynamicFail | hmm alright |
| 01:56.42 | Penguin | I normally just delete users.conf, if there is one, to avoid any stupid stuff it tries to do. |
| 01:56.45 | Penguin | ~users.conf |
| 01:56.45 | infobot | [~users.conf] users.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable asterisk config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system. |
| 01:58.30 | DynamicFail | deleted |
| 01:59.04 | Penguin | Now when you "dialplan reload" you will only get what is in extensions.conf. |
| 02:00.24 | DynamicFail | http://pastebin.com/T3H0nv41 |
| 02:00.29 | DynamicFail | I still seem to get alot |
| 02:00.43 | DynamicFail | unless that bottom stuff is normal |
| 02:01.32 | Penguin | wtf |
| 02:01.50 | Penguin | It still has extensions from other files that it does not say that is is parsing. |
| 02:02.38 | Penguin | Run "core restart now" |
| 02:03.53 | [TK]D-Fender | <PROTECTED> |
| 02:04.29 | Penguin | What doesn't pbx_ael say "parsing extensions.ael"? |
| 02:04.50 | [TK]D-Fender | <PROTECTED> |
| 02:04.53 | [TK]D-Fender | And LUA... |
| 02:05.00 | [TK]D-Fender | I guess that's just because it's what it does... |
| 02:05.17 | [TK]D-Fender | -rw-r----- 1 root root 5253 Mar 21 2013 extensions.lua |
| 02:05.24 | [TK]D-Fender | -rw-r----- 1 root root 12776 Mar 21 2013 extensions.ael |
| 02:05.36 | [TK]D-Fender | It's not like we aren't sitting there with those right in front of us. |
| 02:06.39 | DynamicFail | So just delte those files? |
| 02:06.57 | Penguin | Those permissions and ownership also say asterisk runs under root, which is another thing to put on the list of items to fix. |
| 02:07.32 | Penguin | If you didn't configure anything in those files, delete them. |
| 02:08.03 | Penguin | From what you showed, you're just using the sample files. |
| 02:14.00 | DynamicFail | so I deleted everything except sip.conf, asterisk.conf, and extensions.conf |
| 02:14.01 | DynamicFail | http://pastebin.com/wbPTpNyE |
| 02:14.03 | DynamicFail | still getting alot |
| 02:15.07 | Penguin | I just don't get it. |
| 02:15.43 | Penguin | ael is still loading extensions. Where is it getting them? |
| 02:20.58 | DynamicFail | ? |
| 02:21.18 | DynamicFail | Do you think it could cause issues with the registration/calling which is what I'm really concerned about |
| 02:21.27 | Penguin | Nope. Totally unrelated. |
| 02:22.26 | DynamicFail | alright |
| 02:22.27 | DynamicFail | then I |
| 02:22.33 | DynamicFail | I'm find ignoring that problem for now |
| 02:22.42 | DynamicFail | what would be the next step in debugging the registration issues |
| 02:29.10 | *** join/#asterisk cmendes0101 (~cmendes01@pool-96-251-59-96.lsanca.fios.verizon.net) |
| 02:29.37 | [TK]D-Fender | look at the actual debug... |
| 02:30.02 | *** join/#asterisk DynamicFail (~Administr@198.99.129.159) |
| 02:30.03 | DynamicFail | hmm |
| 02:30.05 | DynamicFail | that was odd |
| 02:30.08 | DynamicFail | got disconnected |
| 02:30.22 | [TK]D-Fender | look at the actual debug... |
| 02:31.35 | *** join/#asterisk cmendes0101| (~cmendes01@pool-96-251-59-96.lsanca.fios.verizon.net) |
| 02:33.13 | DynamicFail | [TK]D-Fender, you mean the sip set debug on |
| 02:33.15 | DynamicFail | debug correct? |
| 02:34.04 | DynamicFail | hmm |
| 02:34.29 | DynamicFail | now my box doesn't even have sip loaded after I restarted |
| 02:35.11 | DynamicFail | do you need any files in /etc/asterisk/ in order for the sip module to load? |
| 02:35.30 | Penguin | How about modules.conf? |
| 02:35.52 | DynamicFail | lol |
| 02:35.56 | DynamicFail | my bad |
| 02:36.11 | DynamicFail | loaded sip with module load chan_sip.so |
| 02:36.44 | [TK]D-Fender | without modules.conf your DOA |
| 02:36.49 | [TK]D-Fender | you're* |
| 02:38.05 | DynamicFail | all fixed |
| 02:38.11 | DynamicFail | loaded the modules.conf reloaded asterisk |
| 02:38.17 | DynamicFail | now back to debugging the issue |
| 02:38.18 | DynamicFail | http://pastebin.com/2ua4Jrr3 |
| 02:41.50 | DynamicFail | It seems like the phone does the REGISTER, the Server responds with SIP/2.0 401 Unauthorized (expected) |
| 02:41.58 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.203) |
| 02:42.03 | DynamicFail | but then the phone never responds to the auth challenge |
| 02:42.19 | DynamicFail | shouldn't it ACK in some way |
| 02:42.33 | Penguin | Have you tried a different UA? |
| 02:42.38 | Penguin | It sounds broken. |
| 02:42.47 | DynamicFail | android app |
| 02:43.12 | DynamicFail | have tried built in android sip device, csipsimple (which I have used before) and currently testing with, and with zoiper |
| 02:43.19 | DynamicFail | they are also able to register with a different asterisk box |
| 02:43.30 | DynamicFail | I'm thinking some sort of network issue, but not sure what |
| 02:43.38 | Penguin | I use csipSimple. I like it. |
| 02:44.13 | Penguin | So the one you're using now on this particular debug session... it registers against a different asterisk? |
| 02:44.26 | DynamicFail | it can yes |
| 02:44.33 | DynamicFail | (not currently) |
| 02:44.54 | DynamicFail | I don't get this line: Sending to 192.168.3.140:54070 (NAT) |
| 02:44.59 | Penguin | I saw that your asterisk was outdated. Can you upgrade it? |
| 02:45.08 | DynamicFail | I have nat=no in the sip.conf file |
| 02:45.13 | DynamicFail | unfortunetly no |
| 02:45.25 | Penguin | The sip.conf that you showed us did not have that in it. |
| 02:45.25 | DynamicFail | I don't have a way to put the box online |
| 02:45.31 | DynamicFail | I added that |
| 02:45.37 | DynamicFail | been trying a few different things |
| 02:45.45 | DynamicFail | small changes |
| 02:46.28 | Penguin | How are you able to be online and also be connected to the asterisk box? |
| 02:46.46 | DynamicFail | manually modified my laptop routing table |
| 02:46.48 | Penguin | There's clearly some network connectivity somewhere. |
| 02:46.49 | DynamicFail | ssh'd in |
| 02:47.04 | DynamicFail | laptop is on wifi and wired network |
| 02:47.34 | DynamicFail | I removed the default route for wired 0.0.0.0 and manually added static routes for my private network |
| 02:47.50 | DynamicFail | the wired network is connected to my asterisk box |
| 02:49.26 | DynamicFail | lol |
| 02:49.31 | DynamicFail | no idea what is going on |
| 02:49.41 | DynamicFail | best guess is still a network issue |
| 02:49.53 | DynamicFail | although I have flushed iptables |
| 02:50.50 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 02:51.01 | [TK]D-Fender | I see no reason to have been messing with the core networking on your box |
| 02:51.10 | [TK]D-Fender | [21:45]DynamicFailI added that <- and the location of this matters |
| 02:51.16 | [TK]D-Fender | And you haven't shown us |
| 02:52.35 | *** join/#asterisk nickgaw (~nick@70-139-55-30.lightspeed.hstntx.sbcglobal.net) |
| 02:53.36 | [TK]D-Fender | <--- SIP read from UDP:192.168.3.138:56975 ---> From: "InTouch 3138" <sip:3138@192.168.3.1>;tag=LsBN6j0Vo7dCKQUqBNjnygXprlUSVD9Z |
| 02:53.50 | [TK]D-Fender | http://pastebin.com/f94nDZYK <- you don't even HAVE a 3138 in your sip.conf you've shown us |
| 02:54.04 | [TK]D-Fender | I see a *** 3038 *** hoever |
| 02:54.05 | nickgaw | Hi, I am planning on getting an asterisk system from my hosting company I use for web hosting. They say there is a GUI web interface I use to configure it if such a thing exists can I just ignore it and edit the configuration files by hand? |
| 02:54.10 | [TK]D-Fender | however* |
| 02:54.39 | [TK]D-Fender | nickgaw: How do we know if they give you a means of getting to the raw files? |
| 02:55.07 | nickgaw | they tell me I will have ssh shell access to the system. I should have mentioned this. |
| 02:55.27 | [TK]D-Fender | nickgaw: Depends what you have acess to. |
| 02:55.32 | [TK]D-Fender | Depends what their web interface does. |
| 02:55.39 | [TK]D-Fender | Depends what they'll "warantee". |
| 02:55.58 | [TK]D-Fender | Depends on sun-spots, shifts in the gravitational axis of the planet, and most of all, KITTENS |
| 02:56.53 | nickgaw | understandable but if I wish to have customizations done is it best to contact my hosting company or to ask on here for an asterisk technition for services? |
| 02:57.24 | *** join/#asterisk DynamicFail (~Administr@198.99.129.159) |
| 02:57.25 | [TK]D-Fender | nickgaw: It's best if you contact your hosting company and find our the TERMS of what they are selling you |
| 02:57.30 | DynamicFail | ~logs |
| 02:57.30 | infobot | All conversations are logged to http://infobot.rikers.org/%23asterisk/ Lines starting with spaces are not logged. Logs are updated daily. |
| 02:57.33 | [TK]D-Fender | nickgaw: You are asking the wrong people. |
| 02:57.36 | DynamicFail | sorry wifi has been shaddy |
| 02:58.27 | [TK]D-Fender | [21:53][TK]D-Fender<--- SIP read from UDP:192.168.3.138:56975 ---> From: "InTouch 3138" <sip:3138@192.168.3.1>;tag=LsBN6j0Vo7dCKQUqBNjnygXprlUSVD9Z |
| 02:58.29 | [TK]D-Fender | [21:53][TK]D-Fenderhttp://pastebin.com/f94nDZYK <- you don't even HAVE a 3138 in your sip.conf you've shown us |
| 02:58.30 | [TK]D-Fender | [21:54][TK]D-FenderI see a *** 3038 *** however |
| 02:58.51 | DynamicFail | h |
| 02:59.43 | nickgaw | What I was thinking was this I find out what they will provide me then possibley setup a local asterisk system on my own local network that does not have an outside phone number so I can do testing before I push changes to the main system. If I should do this and have a local setup do asterisk technitions exist to help me configure the system? |
| 03:00.27 | [TK]D-Fender | nickgaw: Who says they'll assist you? |
| 03:00.58 | nickgaw | well no one I was just asking if such people exist either here or elsewhere? |
| 03:01.32 | [TK]D-Fender | nickgaw: Go find out what you are even ALLOWED to do and if you can work around their GUI |
| 03:01.39 | [TK]D-Fender | nickgaw: We don't ahve those answers |
| 03:01.42 | DynamicFail | [TK]D-Fender, Penguin I must have changed it on accident, here is with the correct extensions, I still have the same issue |
| 03:01.44 | DynamicFail | http://pastebin.com/3H12AZBn |
| 03:03.37 | [TK]D-Fender | DynamicFail: Try another client |
| 03:03.46 | DynamicFail | I have tried 3 clients |
| 03:04.15 | DynamicFail | Is there a way to test the networking side |
| 03:04.19 | Penguin | dynamicfail: That's sip.conf, not extensions. |
| 03:04.19 | DynamicFail | I can ping the device |
| 03:04.23 | nickgaw | That sample asterisk extention that is installed with most asterisk systems like on my Debian system that does and IAX connection to that server is there an outside phone number I can call to connect to that extention using a normal phone line? |
| 03:04.37 | DynamicFail | Penguin, is the extensions.conf used in registration? |
| 03:04.52 | Penguin | dynamicfail: No. |
| 03:04.57 | Penguin | Just like the last time you asked. |
| 03:05.03 | [TK]D-Fender | nickgaw: No. |
| 03:05.03 | DynamicFail | I'm not yet trying to place a call, just register my csipsimple |
| 03:05.26 | Penguin | But you said something a moment ago about changing extensions and then pastebinned a sip.conf. |
| 03:05.35 | nickgaw | so it just supports IAX connections? |
| 03:05.38 | Penguin | Two totally different items. |
| 03:05.42 | [TK]D-Fender | DynamicFail: Undo your use of templates and remove the extra white-space |
| 03:05.54 | [TK]D-Fender | [22:05]nickgawso it just supports IAX connections? <- what does? |
| 03:06.04 | DynamicFail | whitespace in the sip.conf file? |
| 03:06.17 | [TK]D-Fender | <PROTECTED> |
| 03:08.26 | nickgaw | that sample Digium test server that has options for sales and training that server just supports an IAX connection no normal telephone connection? |
| 03:08.55 | Kobaz | heh |
| 03:09.08 | Kobaz | so what sort of fun asterisk stuff is everyone working on for the holidays |
| 03:09.24 | DynamicFail | [TK]D-Fender, changed: http://pastebin.com/n6wQxrKs |
| 03:09.34 | DynamicFail | Kobaz, making phone A and phone B register and call each other |
| 03:09.51 | DynamicFail | is there a way to test that the rtp ports etc are working correctly? |
| 03:10.15 | Penguin | Yeah, make a call. |
| 03:10.29 | Penguin | RTP also has nothing to do with registration. |
| 03:10.50 | *** join/#asterisk seraphie (~erin@24.96.64.240) |
| 03:11.23 | [TK]D-Fender | [22:05][TK]D-FenderDynamicFail: Undo your use of templates and remove the extra white-space |
| 03:12.10 | DynamicFail | [TK]D-Fender, I am not using a "template" and I removed the starting whitespace as you mentioned |
| 03:12.43 | Penguin | you ARE using templates. |
| 03:12.44 | DynamicFail | well I don't think I'm using a template, I just tried to strip the config to the most basic settings possible |
| 03:12.54 | DynamicFail | I don't know what that means I guess then |
| 03:12.59 | Kobaz | DynamicFail: oh yay |
| 03:13.02 | Penguin | Those parenthetical items after the device names... |
| 03:13.15 | Kobaz | DynamicFail: that's the easy part |
| 03:13.19 | nickgaw | That sample asterisk extention that is installed with most asterisk systems like on my Debian system that does and IAX connection to that server is there an outside phone number I can call to connect to that extention using a normal phone line? I am talking about that sample Digium server that has options for sales and training. |
| 03:13.21 | Penguin | That applies the template of the corresponding name. |
| 03:13.31 | [TK]D-Fender | DynamicFail: (!) < -------------------------- |
| 03:13.34 | DynamicFail | oh |
| 03:13.37 | DynamicFail | hahaha |
| 03:13.38 | DynamicFail | ok |
| 03:13.44 | DynamicFail | so add all that under each device config |
| 03:13.51 | Penguin | The ! is the template. The devices are using said template. |
| 03:14.54 | nickgaw | I got my answer the -custom.conf configuration files override the GUI. |
| 03:15.01 | Kobaz | yeah |
| 03:15.19 | Kobaz | if you're trying to just 'get stuff to work' then do everything longhand and dont use templates |
| 03:15.36 | Kobaz | i don't even use templates for my stuff. Although all my configs are db generated anyway |
| 03:15.51 | *** join/#asterisk nickgaw (~nick@70-139-55-30.lightspeed.hstntx.sbcglobal.net) |
| 03:15.56 | Penguin | I use a crap-ton of templates. |
| 03:16.16 | [TK]D-Fender | [22:14]nickgawI got my answer the -custom.conf configuration files override the GUI. <- says virtually nothing...... |
| 03:16.27 | Penguin | Cascading templates, too. |
| 03:16.31 | Kobaz | Penguin: i do insert into asterisk.devices_sip (stuff)... and then the db makes sip.conf for me :P |
| 03:17.32 | nickgaw | basically what they told me was if I edit files like asterisk-custom.conf I can put what I wish in there and if it is not setup already in the GUI then I can do all my manually coding I choose. |
| 03:18.37 | [TK]D-Fender | nickgaw: Doesn't sound quite right.... |
| 03:19.39 | nickgaw | why do you say this? I also read from playing around on Debian and trying to get freepbx to work that the custom.conf files you could do your own changes in and they work? |
| 03:19.52 | DynamicFail | [TK]D-Fender, I think to your specifications: http://pastebin.com/9t5LLQKC |
| 03:20.59 | [TK]D-Fender | DynamicFail: "sip show peer 3138" |
| 03:22.23 | DynamicFail | http://pastebin.com/sC8cpjZf |
| 03:23.56 | [TK]D-Fender | So far all I'm left with is to check your password and if that is 100%.... use another client |
| 03:24.09 | [TK]D-Fender | Doesn't look like this one is responding the 401 challenge |
| 03:25.45 | DynamicFail | hmm that's unfortunte |
| 03:25.55 | DynamicFail | well I have tried 3 different clients with different apps |
| 03:25.59 | DynamicFail | and confirmed my password |
| 03:27.02 | DynamicFail | [TK]D-Fender, would the response be on a different port or network settingt |
| 03:27.12 | DynamicFail | that could get hit by something like a firewall but the others wouldn't |
| 03:28.15 | [TK]D-Fender | check firewalls on your server & the other systems |
| 03:28.36 | nickgaw | [tk]d-fender, I take it you are an advanced asterisk user correct? |
| 03:29.00 | DynamicFail | haha not the question to ask |
| 03:29.29 | nickgaw | why do you say that? The advanced users are probably the ones who know the most right? |
| 03:30.18 | nickgaw | I meant nothing bad about asking that question. |
| 03:30.39 | *** join/#asterisk LiuYan (~LiuYan@222.125.137.217) |
| 03:35.12 | *** join/#asterisk zamba (marius@flage.org) |
| 03:35.50 | DynamicFail | Penguin, you all out of ideas as well? |
| 03:36.42 | Penguin | I can't think of anything different from what has already been addressed and suggested. |
| 03:37.20 | [TK]D-Fender | DynamicFail: Show us another client. |
| 03:38.02 | [TK]D-Fender | nickgaw: Vague question as to what it takes to constitue "expert", but I have been around for a long time, and configured a lot of different things. |
| 03:38.34 | Penguin | I can send and receive calls. Does that mean anything? |
| 03:42.41 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.223) |
| 03:46.59 | DynamicFail | hahaha using the "expert" mode with csipsimple I got the phone to say "registered" Penguin however doesn't appear to actually register |
| 03:47.02 | DynamicFail | according ot asterisk |
| 03:47.35 | Penguin | What are you doing to find out if asterisk thinks it is registered? |
| 03:48.05 | [TK]D-Fender | maybe you should show us your client setup..... |
| 03:49.26 | DynamicFail | well before I was using the basic setup: Account Name: 3138, User 3138, server 192.168.3.1, password: xxx |
| 03:49.48 | Penguin | You're saying that, according to asterisk, it is not registered. How did you make that determination? |
| 03:50.13 | Penguin | If the phone says it is registered, what makes you think asterisk says it is not? |
| 03:50.23 | DynamicFail | sip show peers |
| 03:50.32 | Penguin | And it says what? |
| 03:50.47 | DynamicFail | UNKNOWN (Unspecified) |
| 03:51.00 | [TK]D-Fender | sip debug is all we should care about |
| 03:51.17 | DynamicFail | says the same it was always saying |
| 03:51.24 | [TK]D-Fender | next client |
| 03:51.38 | DynamicFail | [TK]D-Fender, do you mean app or device |
| 03:51.41 | DynamicFail | I don't have any other devices |
| 03:51.51 | [TK]D-Fender | client = software |
| 03:51.51 | Penguin | He means UA. |
| 03:51.58 | [TK]D-Fender | indeed |
| 03:52.00 | [TK]D-Fender | User Agent |
| 03:52.38 | Penguin | And not the kind of UA you get when your boss thinks you're getting high at work. :/ |
| 03:52.57 | DynamicFail | ha, any suggestions for android? |
| 03:53.42 | [TK]D-Fender | 3cx |
| 03:54.00 | nickgaw | So before I go asking here for assistance with my asterisk system I should first check with my hosting company? |
| 03:54.11 | DynamicFail | 3cx doesn't have a regular sip app anymore [TK]D-Fender, it only works with their phone system now |
| 03:54.31 | [TK]D-Fender | nickgaw: ..... |
| 03:55.01 | [TK]D-Fender | nickgaw: before SIGNING UP you should actually have a clue what you are dealing with. |
| 03:55.22 | [TK]D-Fender | nickgaw: And that doesn't mean some loose claim as to how things work. that means SEEING it. |
| 03:55.34 | [TK]D-Fender | DynamicFail: Since when? |
| 03:55.51 | DynamicFail | fairly recently |
| 03:56.07 | DynamicFail | https://play.google.com/store/apps/details?id=com.tcx.sipphone12 |
| 03:56.13 | DynamicFail | Description |
| 03:56.13 | DynamicFail | 3CXPhone for 3CX Phone System requires 3CX Phone System v12!!! It will not work with a VoIP provider or any other PBX!!! |
| 03:56.42 | nickgaw | I know what I will be dealing with mainly asking about manually configuration outside of the GUI. Say the GUI does not do something I wish to do and I can write the same in asterisk configuration language I wish to be able to do this with out breaking the system. from what they told me yes what I wish to do is possible like coding in a voice menu. |
| 03:56.43 | Penguin | This is now the second time I have heard about it. |
| 03:57.25 | [TK]D-Fender | DynamicFail: Zoiper |
| 03:57.33 | [TK]D-Fender | DynamicFail: It has a good rep for working. |
| 03:59.30 | nickgaw | What I am setting up is a small system for my house where people can call and dial the extention of who they wish to talk with. I should have mentioned this before but did not want to get to detailed if this is not what you were wanting or here the more detail I can give the better? |
| 03:59.31 | [TK]D-Fender | nickgaw: "From what they told you" Also depends on what you asked and how they interpreted it. Either way.... we can't confirm anything for you. |
| 03:59.56 | [TK]D-Fender | nickgaw: that aside... it is (if you're lucky" yours to configure.... so do whatever you want (and are allowed access to) |
| 04:00.31 | [TK]D-Fender | nickgaw: You are telling us NOTHING. What kind of answers are you expecting? |
| 04:00.47 | [TK]D-Fender | nickgaw: Have you gone and learned how to use * already? |
| 04:01.02 | nickgaw | does not look that tricky to do as I already have the cell phones with SIP do they sell hardware based phones that do IAX? |
| 04:01.05 | [TK]D-Fender | nickgaw: Have you installed it, learned and tested thigns already? |
| 04:01.44 | nickgaw | played around with the GUI yes and setup a few SIP extentions. |
| 04:01.50 | [TK]D-Fender | nickgaw: Almost no IAX hardphones exist, and the few I've seen were cheap trash |
| 04:02.07 | *** join/#asterisk DynamicFail (~Administr@198.99.129.159) |
| 04:02.13 | DynamicFail | dahhhhhhhhhh stupid internet |
| 04:02.15 | DynamicFail | ~book |
| 04:02.15 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 04:02.19 | [TK]D-Fender | [23:01]nickgawplayed around with the GUI yes and setup a few SIP extentions. <- that does not constitute learning Asterisk in terms of raw configuration |
| 04:05.33 | Penguin | I don't remember if I ever tried sipdroid. |
| 04:06.33 | DynamicFail | [TK]D-Fender, Penguin I missed some getting disconnected did you have a client besides 3cx to recommend |
| 04:06.48 | Penguin | zoiper was suggested. |
| 04:06.50 | [TK]D-Fender | [22:57][TK]D-FenderDynamicFail: Zoiper |
| 04:06.52 | [TK]D-Fender | [22:57][TK]D-FenderDynamicFail: It has a good rep for working. |
| 04:06.59 | DynamicFail | had tried zoiper already |
| 04:07.01 | DynamicFail | same issue |
| 04:07.09 | Penguin | There's one called "sip phone" too. |
| 04:07.40 | Penguin | Media5 fone was okay on iPhone. |
| 04:07.54 | Penguin | Acrobits has a SIP phone on droid. |
| 04:08.42 | nickgaw | I have read the asterisk.conf files that the GUI plays with. |
| 04:09.06 | Penguin | What's your point? We don't support any GUI here. |
| 04:09.36 | nickgaw | understandable I was not talking about a GUI I was talking about raw configuration edits. |
| 04:09.57 | nickgaw | I have looked at the raw configuration files that are editted by the GUI. |
| 04:10.02 | Penguin | If you're going to admin asterisk without a GUI, don't use a GUI. |
| 04:10.15 | Penguin | The two don't mix. |
| 04:10.18 | nickgaw | so there is no middle method? |
| 04:10.36 | [TK]D-Fender | nickgaw: You talk about everything generically. |
| 04:10.39 | nickgaw | so either use the GUI or edit manually not both? |
| 04:10.56 | [TK]D-Fender | nickgaw: what that GUI lets you do ... WE DON'T KNWO |
| 04:11.15 | nickgaw | understandable |
| 04:11.19 | [TK]D-Fender | nickgaw: You've given NO NAMES. You have not confirmed precisely what level of control you REALLY ahve with it. |
| 04:11.37 | nickgaw | it is using freepbx. |
| 04:11.43 | [TK]D-Fender | ... |
| 04:11.46 | [TK]D-Fender | NOW you tell us? |
| 04:12.08 | [TK]D-Fender | We could have told you DEFINITIVE things long ago if we had known |
| 04:12.32 | nickgaw | earlier I mentioned configuration of freepbx on my Debian system. I should have said the same GUI was on my production system. I do still get ssh shell access. |
| 04:12.36 | [TK]D-Fender | <PROTECTED> |
| 04:12.41 | [TK]D-Fender | And the reverse |
| 04:12.54 | [TK]D-Fender | nickgaw: I'm asking about your PROVIDER |
| 04:13.14 | [TK]D-Fender | nickgaw: I don't even know what computer this "Debian system" you're talking about is. |
| 04:13.23 | [TK]D-Fender | nickgaw: is the YOUR PC? |
| 04:13.28 | [TK]D-Fender | nickgaw: is that your PROVIDER you're talkig about? |
| 04:14.23 | nickgaw | the Debian system I own is my desktop the asterisk system not totally sure what it is running I know it is a VPS platform but have not yet really looked into the shell access as I don't want to break things. |
| 04:15.40 | nickgaw | I tried installing asterisk and freepbx on my Debian system and did not want to give names of things as I don't want to bash or say one company is better then another. |
| 04:16.56 | [TK]D-Fender | nickgaw: You're asking for advice ont hings and you aren'g giving us any useful details. This is a complete waste of our time. |
| 04:18.01 | nickgaw | so what I am trying to find out is with using freepbx with asterisk is there any type of documentation on what I can and can not do manually so I don't break the system as I have to pay to have it reloaded? |
| 04:18.33 | nickgaw | or does freepbx have a backup module I can use? |
| 04:18.49 | [TK]D-Fender | nickgaw: Have you already gone through their WIKI? |
| 04:19.08 | [TK]D-Fender | nickgaw: And the rest of the guides on their site? |
| 04:19.30 | nickgaw | not all that much mainly read the internal information like in the extention setup where is their wiki located? |
| 04:19.38 | [TK]D-Fender | nickgaw: Asking if they have a backup module shows you haven't even looked at FreePBX's modules. |
| 04:20.57 | [TK]D-Fender | [23:19]nickgawnot all that much mainly read the internal information like in the extention setup where is their wiki located? <- on their website. |
| 04:21.17 | nickgaw | when playing around with it on my Debian system I remember seeing something about backing up configurations but not totally sure where I found it and this current system is a lot newer. With sshfs I can login as root what directory should I backup for things like extentions and other voicemails is it /etc/asterisk and something in /var? |
| 04:21.40 | [TK]D-Fender | Depends what you think you're backing up. |
| 04:22.30 | nickgaw | trying to backup my extention data and voicemail greetings and messages so if I need to restore things I can do so and not lose much data. |
| 04:22.57 | [TK]D-Fender | Do you know where *'s files go? |
| 04:23.04 | [TK]D-Fender | Do you know what FreePBX uses? |
| 04:23.25 | [TK]D-Fender | I'd guess not since you're asking..... |
| 04:23.28 | [TK]D-Fender | ~book |
| 04:23.29 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 04:23.41 | [TK]D-Fender | Go learn Asterisk, and go learn FreePBX |
| 04:23.47 | [TK]D-Fender | and for FreePBX questions.... |
| 04:23.49 | [TK]D-Fender | ~freepbx |
| 04:23.49 | 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 |
| 04:23.51 | [TK]D-Fender | ^^^ |
| 04:24.14 | *** join/#asterisk Vutral (ss@mirbsd/special/Vutral) |
| 04:24.53 | nickgaw | good resource this is what I need. |
| 04:26.19 | *** join/#asterisk gtjoseph (~gtjoseph@unaffiliated/gtj) |
| 04:26.26 | DynamicFail | [TK]D-Fender, http://pastebin.com/6MJq78e2 |
| 04:26.50 | DynamicFail | Using 3cx, I had an old apk saved |
| 04:26.57 | DynamicFail | same issue |
| 04:27.49 | [TK]D-Fender | DynamicFail: add "alwaysauthreject=no" under [general], and reload chan_sip.so |
| 04:28.02 | [TK]D-Fender | DynamicFail: Lets see if that helps |
| 04:29.23 | DynamicFail | negitive ghostrider |
| 04:29.32 | DynamicFail | same result |
| 04:30.59 | [TK]D-Fender | hrm |
| 04:31.00 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
| 04:31.42 | Penguin | |
| 04:33.28 | DynamicFail | only thing I can think is some sort of network issue where the REGISTER and the 401 are able to get through, but for some reason the ACK isn't |
| 04:43.31 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.171) |
| 04:59.38 | ChannelZ | YOUR NETWORK IS SCREW! |
| 04:59.55 | DynamicFail | lol |
| 05:00.03 | DynamicFail | good to see you again ChannelZ |
| 05:01.36 | ChannelZ | either that or there is some minute problem with the headers Asterisk is sending such that NONE of your phones understand it... but I'm guessing it's something else. |
| 05:02.47 | ChannelZ | The box your asterisk is running on, is it using iptables? |
| 05:07.01 | ChannelZ | see, I said his network was screw. |
| 05:09.13 | *** join/#asterisk xous (~xous@unaffiliated/xous) |
| 05:19.09 | *** join/#asterisk g_r_eek (~g_r_eek@176.92.223.118) |
| 05:30.15 | *** join/#asterisk bmg505 (~leon@196-209-99-62.dynamic.isadsl.co.za) |
| 05:40.57 | *** join/#asterisk hos7ein (~chatzilla@2.191.234.176) |
| 05:47.44 | *** join/#asterisk digilink (~digilink@unaffiliated/digilink) |
| 05:47.56 | *** join/#asterisk tahc (180e1ff2@gateway/web/freenode/ip.24.14.31.242) |
| 05:48.14 | tahc | Hi all, can someone help answer some VoIP questions I have? |
| 05:49.34 | ChannelZ | perhaps |
| 05:50.02 | tahc | I'm looking to get a hosted plan with Vocalocity but am having trouble understanding the concept of lines on VoIP |
| 05:50.37 | tahc | I have four main locations each with their own number, which I am going to keep. From what I understand, they allow for unlimited inbound / outbound calling (channels?) |
| 05:51.17 | tahc | so I'm looking at SIP phones now, if I purchase a 12 line Polycom SIP phone, will it mean that extension can make / receive up to 12 calls at once bandwidth dependant? |
| 05:58.02 | WIMPy | It can mean 12 calls or 12 accounts. Marketing speak. |
| 05:59.39 | tahc | So what is a Secondary line appearance? Is that related to the 12 lines that come with the phone? Or is that showing another number on the actual device? Thanks for your help, WIMPy |
| 06:00.10 | WIMPy | has no idea what that term could mean. |
| 06:00.29 | tahc | got it, thanks for your help |
| 06:04.34 | ChannelZ | Could mean for 'call waiting' type scenarios.. a second line key that refers to the same account |
| 06:09.25 | tahc | and account means line in this case? |
| 06:09.29 | tahc | and line means phone number? |
| 06:09.46 | ChannelZ | Not really |
| 06:09.59 | ChannelZ | Account means a SIP account. |
| 06:10.15 | ChannelZ | Which might have 50 different phone numbers that can come to it |
| 06:11.02 | tahc | gotcha |
| 06:11.03 | ChannelZ | A "line" is kind of a vague term in voip because it's all virtual. |
| 06:11.16 | *** join/#asterisk cmendes0101 (~cmendes01@pool-96-251-59-96.lsanca.fios.verizon.net) |
| 06:12.28 | ChannelZ | We tend to call them "channels", as it's a channel of communication between two places. How that maps to the public telephone system (what you might call "a line") -- if at all -- can get a bit ambiguous |
| 06:13.14 | tahc | so what is a phone number actually called? |
| 06:13.17 | WIMPy | It has been a little vague in the PSTN as well since ISDN. |
| 06:13.18 | tahc | Right now I have 15 POTS lines |
| 06:13.18 | ChannelZ | I don't know anything about Vocalocity, but generically speaking if you get a SIP account at some ITSP or another, you get 1 or 2 simultaneous channels |
| 06:13.36 | WIMPy | A phone number is called a phone number. |
| 06:14.46 | tahc | so I have three phone numbers, "unlimited" channels which are used by what the phone manufacturers call "lines" ? |
| 06:14.57 | ChannelZ | (per DID - phone number - I should say..) |
| 06:15.02 | WIMPy | Why would they limit the number of simultaneous calls unless you have a plan with flat rate calls. |
| 06:15.08 | WIMPy | ? |
| 06:15.19 | tahc | so if I had a 12 line polycom but only two channels, only two of those lines would work at any given time |
| 06:15.22 | ChannelZ | Bandwidth. |
| 06:15.32 | ChannelZ | Just depends on what they offer. |
| 06:15.55 | WIMPy | In >99% of the cases here, phone numbers are non-DID. |
| 06:17.31 | ChannelZ | tach: unless your phone can internally conference 12 calls, you're only going to be using one or two at a time anyway :) But again it depends on your ITSP |
| 06:17.57 | ChannelZ | If you have a more expensive account they might not care how many channels you use simultaneously. Basic ones, they sort of do. |
| 06:19.15 | tahc | What often ends up happening at my office is I get 4 or 6 calls coming in at once |
| 06:19.25 | tahc | I need to have my receptionist answer them all and place them all on hold |
| 06:19.35 | ChannelZ | I can't imagine an ITSP that has say a $15/mo unlimited account that would be pleased if you were making 25 simultaneous calls running your entire office. |
| 06:20.21 | WIMPy | Handling multiple calls at once is a PITA with most SIP phones. |
| 06:20.21 | tahc | They're charging $19.99 / extension for unlimited dialing |
| 06:20.40 | tahc | why is that, WIMPy? |
| 06:20.52 | WIMPy | The Digium ones are ok if you use only one account. With more accounts they are about as bad as the others. |
| 06:21.51 | WIMPy | Or actually not just ok, but good. |
| 06:22.43 | ChannelZ | Personally I don't like treating voip like analog, as if they were discreet lines like pots, with line keys mapped to each one |
| 06:22.59 | tahc | what do you prefer to do, ChannelZ ? |
| 06:23.14 | ChannelZ | Park them. |
| 06:23.15 | WIMPy | That's a prety bad concept indeed. |
| 06:23.35 | WIMPy | I prefer to have call keys. |
| 06:23.54 | tahc | now I'm a bit more confused, park them? call keys? |
| 06:24.03 | tahc | I need to spend a lot more time learning this stuff |
| 06:24.27 | ChannelZ | Parking = transferring a call into a "parking lot" where they get put on hold, and you pick them up by dialing a specific extension |
| 06:24.35 | WIMPy | A list of your calls on the display with a button for each call next to that line on the display. |
| 06:24.55 | tahc | how is park different than hold? |
| 06:25.14 | WIMPy | parking is a server based feature. |
| 06:25.21 | tahc | thanks for your help also guys |
| 06:25.35 | ChannelZ | It's not bound to the concept of a line. |
| 06:25.56 | ChannelZ | Mostly it's not any different from anyone's point of view except yours |
| 06:26.27 | WIMPy | Hold isn;t bound to lines, either. |
| 06:26.36 | WIMPy | At least not if done right. |
| 06:26.39 | ChannelZ | That is, your secretary can still say "pick up line 3" which refers to parking lot space 3. |
| 06:27.19 | WIMPy | So you can get that vintage feeling. |
| 06:27.24 | tahc | so she can continue to get tons of incoming calls and park them all until she has a chance to get to each one and transfer / answer questions appropriately ? |
| 06:27.42 | ChannelZ | sure |
| 06:27.50 | tahc | Maybe I should send out some memos letting everyone know to say "Pick up parking lot space xyz" from now on :p |
| 06:28.23 | tahc | excellent. I was considering setting up asterisk but seeing how foreign I am with all this VoIP stuff I thought it best to leave it to a hosted solution for a while |
| 06:28.37 | ChannelZ | well.. you can kind of set the whole thing up like WIMPy is saying to assign calls to keys |
| 06:29.05 | WIMPy | You can? |
| 06:29.11 | ChannelZ | I'm mostly saying try not to think about the line keys of your phone as physical phone lines like analog, because you're just boxing yourself in and kind of missing some of the point of voip and the flexibility it can provide |
| 06:29.12 | WIMPy | What phone will do that? |
| 06:30.06 | tahc | any suggestions on good 4 to 6 line SIP phones? |
| 06:30.29 | WIMPy | Ok, I can answer that myself. the Snoms can, but they don't display a full list. |
| 06:31.23 | tahc | also, just to clarify, if I purchase a two line phone, I can only make or receive two simultaneous calls from that phone regardless of parking / etc ? |
| 06:31.57 | ChannelZ | Not necessarily |
| 06:32.04 | tahc | why not? |
| 06:32.06 | WIMPy | If you park a call, it's ended from your phones point of view. |
| 06:33.04 | tahc | aah, If I get a call, park it, I still have two open lines |
| 06:33.30 | ChannelZ | Well it's kind of a vague question, besides a conference call, what use is having 2+ simultaneous calls going? Like you're still only one person with one handset. |
| 06:34.15 | tahc | yeah, I am still thinking in the POTS mindset where a call comes in, I place it on hold, that line is tied up until the call is dealt with |
| 06:34.26 | ChannelZ | That's not generally the case. |
| 06:34.50 | tahc | with VoIP, a call comes in, I park it, I open up two lines on the phone again |
| 06:34.54 | WIMPy | Ineed. Even POTS supports holding calls. |
| 06:35.20 | tahc | I have a nortel system setup right now, where hold means that line is no longer usable until someone disconnects it |
| 06:35.50 | WIMPy | It's a common issues of PBXs that they don't support holding calls. |
| 06:35.54 | ChannelZ | It takes up a line key on your phone perhaps, but otherwise the call isn't technically connected to the phone anymore |
| 06:36.18 | tahc | are there phone or server based limitations on how many calls I can park or how many calls I can see parked? (assuming I have unlimited channels) |
| 06:36.56 | ChannelZ | Not particularly no |
| 06:36.56 | WIMPy | Bandwidth |
| 06:37.03 | ChannelZ | just bandwidth |
| 06:37.13 | tahc | if I get 6 calls in at once and park them all, I'd like to be able to pick them up from anywhere in the office |
| 06:37.13 | WIMPy | And for seeing them, the number of buttons with LEDs. |
| 06:37.19 | tahc | you guys are excellent. |
| 06:38.03 | tahc | so why do they sell 4 line - 6 line - 12 line phones? |
| 06:38.10 | tahc | (VoIP phones) |
| 06:38.11 | ChannelZ | Parking is server-side so you can pick it up from across the country if you want |
| 06:38.23 | WIMPy | To have 12 accounts. |
| 06:39.13 | tahc | Why would one need more than one SIP account? |
| 06:39.46 | WIMPy | Multiple ITSPs/servers. |
| 06:40.00 | ChannelZ | I have a 4-line phone at home.. one connects to my server at home, one to my server at work |
| 06:40.01 | WIMPy | Or to signal calls to different numbers. |
| 06:40.45 | tahc | are you both running asterisk ? |
| 06:40.50 | ChannelZ | Right. Maybe you have 2 phone numbers, "main" and "tech support" and you want them to obviously show up separately. |
| 06:41.17 | ChannelZ | Yeah. (hint, you're in #asterisk :) |
| 06:41.27 | tahc | hah, I figured as much. |
| 06:41.27 | WIMPy | yes |
| 06:41.53 | tahc | I think that about covers everything for me |
| 06:41.58 | tahc | again thanks a ton |
| 06:42.22 | ChannelZ | think virtually |
| 06:42.43 | tahc | If you all have time / care to answer, suggestions for routers? I've got a DD-WRT setup right now but want to move to something more enterprise, either the Edge series by Ubiquiti or PfSense |
| 06:43.04 | tahc | so I can separate the data and telecom into their own VLANs |
| 06:43.08 | WIMPy | You do? |
| 06:43.44 | tahc | I do? |
| 06:43.59 | WIMPy | I'd stay with DD-WRT. |
| 06:44.53 | tahc | That was a third option, but it is running on a residential router which I don't think will handle the voice traffic so well |
| 06:45.08 | WIMPy | Why not? |
| 06:45.54 | tahc | it's already dropping packets and struggling under our current load |
| 06:46.06 | tahc | plus it isn't very reliable in terms of uptime |
| 06:46.15 | WIMPy | They usually perform badly when you try to let them encrypt your traffic. But oterwise they should be fine. |
| 06:46.26 | ChannelZ | How many ports are on the thing? |
| 06:46.29 | tahc | I have it set to restart twice a week which is helping some, but I'd like to try to avoid doing that too often |
| 06:46.33 | tahc | 6 ports |
| 06:46.42 | tahc | but it's sent to a 24 port switch |
| 06:46.47 | WIMPy | Then get some better hardware. |
| 06:46.54 | ChannelZ | oh so you've got some hubs strung together |
| 06:47.24 | ChannelZ | or wait |
| 06:47.26 | tahc | Comcast pipe -> Modem (bridged) -> DDWRT router -> Switch -> Machines & UniFi WLAN |
| 06:47.36 | ChannelZ | this is your internet ... yeah |
| 06:47.56 | tahc | VoIP go live is Jan 15th |
| 06:48.47 | ChannelZ | can't help you there, my router is a whole linux box, haven't run any appliances |
| 06:49.09 | tahc | Thinking about doing PfSense on a centos distro |
| 06:49.20 | tahc | easy enough to configure and looks super reliable |
| 06:49.40 | tahc | only drawback is power usage from having a machine run 24/7 vs a tiny router |
| 06:50.04 | tahc | should I assign the phones static IPs or let them stay DHCP? |
| 06:50.18 | WIMPy | You are concerned about power consumption, yet want to go VOIP? |
| 06:50.30 | ChannelZ | either way |
| 06:50.57 | tahc | good point I guess, WIMPy |
| 06:51.01 | WIMPy | DHCP is fine, if you let the phones register with the server, wich is the usual configuration. |
| 06:51.20 | tahc | but i'm 90% sure the VoIP phones will pull less power than the Nortel beast that's chugging away right now |
| 06:51.20 | WIMPy | VOIP phones are quite expensive to run. |
| 06:51.45 | ChannelZ | you just need a bigger hamster |
| 06:51.51 | WIMPy | I'm >99% sure the opposite is true. |
| 06:52.20 | tahc | 10 / 12 phones, 12 or 24V PoE at less than an amp each? |
| 06:52.36 | tahc | I've killed too many hamsters trying to keep all this alive. The nortel has seen better days |
| 06:53.00 | tahc | No onsite server for the VoIP so no power usage there |
| 06:53.01 | ChannelZ | By the way, all the cool functions we talked about may or may not be at your fingertips via a hosted service you are perhaps talking about |
| 06:53.36 | ChannelZ | Like the whole parking business. |
| 06:53.57 | tahc | I'll be sure to clarify with them before go live |
| 06:54.19 | tahc | Do you still have your analog lines running into your asterisk box or are you using SIP trunking (is that what it's called?) |
| 06:54.41 | WIMPy | hasn't had an analog line for >20 years. |
| 06:54.43 | ChannelZ | If it's a business solution type package they might have some web interface they let you configure some stuff with, to setup your internal extensions and stuff.. |
| 06:54.57 | ChannelZ | otherwise it might just be a generic DID account |
| 06:55.03 | WIMPy | Until about 3 months ago it was ISDN. |
| 06:55.23 | ChannelZ | At work I have 4 POTS going into asterisk and then all SIP phones on the local side |
| 06:55.53 | tahc | what phones are you running? |
| 06:56.02 | ChannelZ | At home it's all voip |
| 06:56.25 | WIMPy | The old ISDN phones and one Snom 360. |
| 06:56.38 | ChannelZ | I have a bunch of Linksys SPA922's (now Cisco SPA50x of some sort) |
| 06:57.35 | WIMPy | has several other SIP phones, but dosn't have them connected. |
| 06:57.46 | tahc | looks like parking is supported http://support.vocalocity.com/kb/call-park/ |
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| 06:59.13 | ChannelZ | ok so it's a more of a hosted businsess solution |
| 06:59.57 | tahc | yup, they market it towards small business. Seems like a solid company |
| 07:00.11 | ChannelZ | where are you? |
| 07:00.15 | tahc | Chicago |
| 07:00.21 | tahc | Cold, snowy, Chicago |
| 07:00.31 | tahc | you? |
| 07:01.06 | ChannelZ | Cold Colorado, not snowing at the moment as far as I know. |
| 07:02.13 | tahc | how long did it take for you to get comfortable with asterisk ? |
| 07:02.23 | tahc | & how did you learn it? Just messing around? |
| 07:02.44 | WIMPy | Define comfortable. |
| 07:02.50 | ChannelZ | Basically yes.. I was starting my own company and needed a phone system, didn't want to buy some monolithic thing |
| 07:03.32 | ChannelZ | I probably had it up and running in a week.. |
| 07:03.58 | tahc | I might give it a shot in a month or two, it seems like it has a ton more features and customizability |
| 07:04.05 | ChannelZ | Probably a month or two before I really understood what was going on under the hood |
| 07:04.08 | tahc | and can't beat free |
| 07:04.18 | ChannelZ | And as WIMPy suggests, it's an ongoing thing :) |
| 07:04.44 | WIMPy | "Free" can be rather expensive. |
| 07:05.00 | tahc | expensive but sounds like fun |
| 07:05.03 | tahc | at least once you get it working |
| 07:05.05 | ChannelZ | It's "how valuable is your time" |
| 07:05.16 | tahc | I'm sure there will be tons of ripping my hair out trying to get it to work |
| 07:05.19 | ChannelZ | But I'm a nerd and yeah it was fun |
| 07:05.32 | WIMPy | It's interesting. Fun is pretty debatable. |
| 07:05.51 | tahc | Do you have Fiber / T1 lines or just cable / DSL? |
| 07:06.00 | ChannelZ | Well I found it fun. |
| 07:06.13 | ChannelZ | Cable |
| 07:06.25 | WIMPy | DSL and one with cable. |
| 07:07.00 | tahc | any call quality issues? what is your DS/ US ? How many simultaneous calls at once on average? |
| 07:07.39 | WIMPy | Sure. All DSL are 16/1. |
| 07:08.14 | ChannelZ | Call quality can be tricky if your internet connection is not dedicated to the task... someone starts downloading a giant file and things can go to hell |
| 07:08.46 | tahc | QoS? |
| 07:08.50 | ChannelZ | I'd like to switch to voip at work but need to get some bandwidth shaping going first |
| 07:09.04 | WIMPy | You definitely need TC. |
| 07:09.22 | WIMPy | It it will only help for your local upload. |
| 07:09.30 | ChannelZ | Essentially. You can only truly control your outgoing, but can influence what comes in to a degree. |
| 07:10.21 | tahc | I think / hope with 50/10 we should be okay with at most 6 simul. calls at once |
| 07:10.33 | ChannelZ | To do it totally right you need the cooperation of your ISP but good luck with that on something like Comcast :) |
| 07:10.46 | ChannelZ | If you buy THEIR phone service they do it of course.. |
| 07:11.35 | tahc | I asked them about their VoiceEdge... they charge INSANE prices and lock you into a 3 year contract so there goes that idea |
| 07:11.41 | ChannelZ | The bandwidth for 6 calls isn't much of a problem, it's everything else |
| 07:12.56 | tahc | nothing too crazy going on. We primarily connect to a Saas server and pull a total of 2 - 3 gigs incoming total per day at most |
| 07:13.22 | tahc | users don't download anything as everything gets wiped nightly thanks to the wonders of DeepFreeze |
| 07:14.18 | ChannelZ | Well we're talking about realtime communications here. It's not an enormous amount of data, but it's small bits of it at a regular rate and has to leave and arrive consistently. |
| 07:15.21 | tahc | Guess we'll have to wait and see! Not much I can do about it |
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| 07:22.42 | ChannelZ | Well that was nice. |
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| 09:45.49 | ChannelZ | hmm. pjsip doesn't seem to want to bind |
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| 12:08.59 | SudanM38 | Hi All ,, |
| 12:09.34 | SudanM38 | Asking about vtiger sterisk Integration ,, any Guide |
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| 13:25.46 | edwin_quijada | Hi! |
| 13:26.52 | edwin_quijada | I am having a weird problem to do an IVR using AGI in perl get_Data doesnt read the numbers pushed |
| 13:27.48 | edwin_quijada | I test the others elements and works but I cant read the digits using get_data |
| 13:28.00 | edwin_quijada | I am using asterisk 1.8 |
| 13:28.03 | elenril | i want asterisk (sip) to listen both on ipv4 and ipv6, but seems the deny option only applies to one of those |
| 13:28.43 | elenril | so i can either deny ipv6 (then all ipv4 connections are accepted) or the reverse |
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| 18:24.42 | mcf3782 | WIMPy: Are you around? |
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| 18:24.54 | WIMPy | yes |
| 18:26.33 | mcf3782 | Cool. You were giving me a few pointers Wednesday night about an issue I was having with connecting my asterisk box through a TE220 to a Mitel box.. One of the things you said was that maybe asterisk had been built without libpri. |
| 18:26.49 | WIMPy | right |
| 18:27.02 | mcf3782 | I knew I'd built libpri, but I went down the path of checking that stuff anyway.. |
| 18:28.49 | mcf3782 | It turns out, that when I ran the "configure/make/make install" trifecta; libpri got installed in /lib64 on this Centos box.. Everything else went into /usr/lib/asterisk… So, in effect; asterisk was built without libpri, even though I technically did build libpri. |
| 18:29.00 | WIMPy | You need to configure and make Asterisk after installing libpri and dahdi. |
| 18:29.22 | WIMPy | Interesting. |
| 18:30.12 | mcf3782 | I'm still not sure why that happened.. |
| 18:31.10 | mcf3782 | But, I wanted to say Thank You. Your comment sent me down a debug path I wasn't thinking of at that moment. |
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| 18:36.15 | mcf3782 | dahdi_tool sees both spans on the TE220 now. One span is connected to the Comcast PRI. The other span is connected to the Mitel-3300. The plan is to do a migration off the Mitel in batches, as money becomes available to buy new desk sets. |
| 18:38.03 | mcf3782 | If I can figure out the Caller ID stuff, then I can leave the Asterisk box "in the middle", between Comcast and the Mitel, and pass calls through to the Mitel for users who haven't been migrated to new desk sets yet. |
| 18:39.50 | mcf3782 | I get the number of the calling party on the Mitel desk sets, but not the name. If I route an inbound call from the Comcast PRI to a SIP softphone (Zoiper), I get the number and the name just fine. |
| 18:45.04 | *** join/#asterisk fornax_ (~fornax@85.183.53.64) |
| 18:45.43 | fornax_ | Hi all, I have a dahdi problem, very bad noise but only one way, after I changed my main board, can someone help me identifying the cause? |
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| 18:48.25 | WIMPy | mcf3782: Do you get names from Comcast or doe the Mitel look them up in its directory and that doesn't work any more? |
| 18:49.25 | WIMPy | fornax_: Tell us the whole way your calls take in and out of Asterisk and where you get noise. |
| 18:50.16 | mcf3782 | WIMPy: They come from Comcast. |
| 18:51.24 | fornax_ | I have an internal network where I have SIP and SCCP devices, connected to asterisk. From there I connect to the outside using a Beronet Quad Bri. When I call someone or someone calls me, I have a brilliant sound but the partner has periodical cracking and distortion. |
| 18:51.44 | mcf3782 | I think I captured all of the relevant configs here… http://pastebin.com/v0nWTJsp |
| 18:52.07 | fornax_ | The way from my internal phones to my asterisk 11.6 on gentoo seems to work since i connect with sccp or sip and the problem occurs in both situations |
| 18:53.10 | fornax_ | dahdi_test gives me nothing worse than 99.98 but it is normally 99.995% |
| 18:54.33 | fornax_ | dahdi_speed gives me 596584, I have the problem that my IRQ is shared but this does not really explain, why I have noise only in one direction. Since it is a board with only one slot, I cannot change the PCI slot |
| 18:54.46 | mcf3782 | As an example, if I call one of the DIDs from my cell phone, with the Comcast PRI connected to directly to the Mitel, then what shows up on the desk set is "XXX-XXX-XXXX" "Cell Phone GA". If I put the asterisk box in the middle, and call that same DID, the only thing that shows up on the desk set is "XXX-XXX-XXXX" (Just my cell phone's phone number). |
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| 18:58.48 | fornax_ | WIMPy: here is my lspci -v http://dpaste.com/1516994/ if you think it is a interrupt problem since the card is shared with SATA and USB |
| 19:03.32 | fornax_ | no idea? |
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| 19:18.05 | WIMPy | mcf3782: How are the names transmitted? |
| 19:19.46 | WIMPy | fornax_: When/where do you have the noise? |
| 19:20.32 | fornax_ | WIMPy: I always have the noise, incoming and outgoing calls make no difference, the noise is only on the other side, I hear everything normally |
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| 19:21.14 | mcf3782 | WIMPy: Sorry. I don't understand your question. |
| 19:21.23 | WIMPy | Usually transmitting is less of an issue than receiving. So that seems to be something special. |
| 19:22.06 | WIMPy | mcf3782: Yu said you get the names from Comcast. How do they send them? |
| 19:22.13 | fornax_ | WIMPy: I think so, I never had such a problem before and I use asterisk for a very long time and zaptel / dahdi never my any problems in only one direction |
| 19:22.30 | fornax_ | WIMPy: Do you have an idea where I can continue searching? |
| 19:23.03 | WIMPy | fornax_: Not really, no. |
| 19:23.48 | fornax_ | WIMPy: :-( So do you think that i probably is an Interrupt problem? This would be hard to solve, but why only one direction? |
| 19:24.08 | WIMPy | No. |
| 19:24.25 | WIMPy | I'd expect receiving to suffer more from interrupt issues. |
| 19:25.20 | fornax_ | WIMPy: Okay, I will switch to a newer kernel, recompile dahdi and switch the port of the baronet, maybe some help, but I tried so many things and really have no more good ideas |
| 19:26.54 | WIMPy | It could always be some hardware issue. |
| 19:27.21 | mcf3782 | WIMPy: The data comes as part of the CallerID data. Since the Comcast switch is doing everything in National format, I presume that the CallerID data is BellCore standard format. |
| 19:28.34 | WIMPy | mcf3782: I'm not in to the national signalling. Have you tried pri debug and see what is received on one port and what is sent on the other? |
| 19:29.01 | fornax_ | WIMPy: I changed the mainboard, CPU, men and recompiled my system for the new hardware. I don't think that something really is broken. some compatibility problem, maybe |
| 19:29.48 | WIMPy | fornax_: The compatibility problem might still be in hardware. |
| 19:30.05 | WIMPy | Do you use the same versions as before? |
| 19:31.54 | mcf3782 | WIMPy: I have tried that. I didn't see anything that was any help to me. And it looks like I didn't save any captures of that data; so I will have to go back to the site, put the asterisk box inline again, and do that capture again. |
| 19:32.23 | WIMPy | That's unfortunate. |
| 19:32.53 | WIMPy | But if you get them, please use intense debug. |
| 19:33.06 | dar123 | hey guyz, finally my fxo issue resolved. It was a dahdi bug, i got the patch from digium |
| 19:33.12 | dar123 | happy today :) |
| 19:34.49 | mcf3782 | Will do. |
| 19:35.31 | mcf3782 | Since today is Saturday, and there are no regular users on the system, I can do some testing without much trouble. I'll head back over there and do that. |
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| 20:09.35 | navaismo | is this a valid transport configuration?--->http://pastebin.com/G0hZtaRK |
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| 20:18.54 | ChannelZ | looks like |
| 20:21.26 | file | ChannelZ, did you get pjsip binding? |
| 20:21.34 | navaismo | Thanks, file is explaining in #freepbx |
| 20:22.09 | ChannelZ | Yeah finally.. I think chan_sip and it were fighting and then I found the note about restarting asterisk to change transports |
| 20:22.31 | file | aye |
| 20:22.54 | ChannelZ | I haven't actually gotten it working yet but at least got it listening |
| 20:24.41 | navaismo | is there a table scheme for confbridge? |
| 20:28.21 | ChannelZ | I keep getting "AOR 'Office' not found for endpoint 'Office'" when I try to register a phone, not sure what is going on.. my config seems dead-simple |
| 20:28.52 | file | what is your config? |
| 20:33.40 | ChannelZ | http://pastebin.com/5jgXab8Q |
| 20:34.28 | file | your device is trying to register to an AOR named "Office" but you are using aor-single |
| 20:36.04 | ChannelZ | I'm misunderstanding how it all connects together then. What is the aors=xxx line for then for an endpoint? |
| 20:36.22 | file | it specifies the AORs that an endpoint is able to register to |
| 20:36.31 | file | when an endpoint registers it specifies the name of the AOR |
| 20:36.46 | file | generally for phones it uses the same name as what you have set it to authenticate as |
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| 20:43.49 | ChannelZ | so if I make a [Office] type=aor, but then don't specify an aors on the endpoint? |
| 20:44.27 | file | it should still reject it because you haven't given that endpoint permission to use that AOR |
| 20:47.30 | ChannelZ | is confused |
| 20:48.15 | ChannelZ | So when I say aors=something, that does not mean it's pointing to the [something] whose type=aor ? |
| 20:48.40 | file | it gives the endpoint permission to use that AOR, it does not mean "use this AOR" |
| 20:48.56 | file | when a REGISTER comes in the name of the AOR is in the request |
| 20:49.29 | file | REGISTER = "I want to add this way to contact me to the AOR named <blah>" |
| 20:50.23 | ChannelZ | Ok I see. |
| 20:51.03 | ChannelZ | My aor-single should really be more of a template. |
| 20:51.15 | file | that's what I've seen a few people do |
| 20:53.26 | ChannelZ | Thanks. I guess I was thinking that the endpoint/auth config for a peer were used and the AOR was more of an internal object that got created for me, and the aors= line was basically just giving it settings to use. |
| 20:53.37 | ChannelZ | Works now. |
| 20:53.52 | navaismo | thanks guys seems like almost all is working now, the ws have an odd behavior but all other stuff seems working, im going to test all this week |
| 20:53.56 | navaismo | see you |
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| 21:25.45 | mcf3782 | WIMPy: I've captured a call with pri intense debug turned on. Do you have a few minutes to look at the data and see if you spot what the problem is? If so, I can create a pastebin of the log file.. |
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| 21:33.47 | WIMPy | mcf3782: Do it. |
| 21:35.23 | mcf3782 | WIMPy: Is a PM to you OK with you? |
| 21:35.35 | WIMPy | yes |
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| 22:32.21 | *** join/#asterisk F-JJTH (~clement@car84-1-78-229-89-36.fbx.proxad.net) |
| 22:32.29 | F-JJTH | Good evening |
| 22:32.52 | F-JJTH | I'm trying to use my Asterisk server like a proxy |
| 22:33.06 | F-JJTH | For this I'm using Dial() |
| 22:33.43 | F-JJTH | I noticed that my proxy Asterisk is still part of the link between my client and the final Asterisk server |
| 22:33.59 | Penguin | That's because asterisk isn't a proxy. |
| 22:34.44 | F-JJTH | My question is: what kind of data is handled by my proxy server ? all (sync+voice data) ? only sync ? |
| 22:35.14 | Penguin | If all factors allow for it, you could enable directmedia between your UAs and the peered asterisk. |
| 22:35.49 | Penguin | Asterisk is not a proxy. Asterisk is a Back-to-Back User Agent. |
| 22:35.51 | F-JJTH | "directmedia" ? |
| 22:36.12 | F-JJTH | I understand that I use Asterisk in a special way |
| 22:36.16 | Penguin | In your sip.conf, enable directmedia=yes. |
| 22:36.29 | F-JJTH | I'm using IAX |
| 22:36.44 | F-JJTH | am I done ? :/ |
| 22:37.16 | Penguin | I don't think IAX2 allows for any type of reinvite. |
| 22:38.00 | Penguin | Since IAX2 carries both the signaling and the media, you can't go direct. |
| 22:38.30 | F-JJTH | it make sense |
| 22:38.49 | Penguin | If you were using SIP, you could break the media away from the signaling and allow media to go directly between the UA and the other asterisk, while only the signaling would remain on the intermediate asterisk. |
| 22:39.30 | WIMPy | IAX can rtedirect the media stream and/or signalling. |
| 22:40.30 | F-JJTH | hmmm indeed it's exactly what I need :) but I'm maintaining a project existing since many years using IAX and I wouldn't change for SIP else I will loose the backward compatibility |
| 22:41.04 | F-JJTH | WIMPy: ah interesting ! do you have a documentation ? |
| 22:41.49 | F-JJTH | looking at iax.conf I can't see anything related to "redirecting a media" |
| 22:42.05 | F-JJTH | hmmm transfer=mediaonly ! |
| 22:42.20 | WIMPy | Exactely. |
| 22:42.28 | F-JJTH | but isn't it related to the Transfer() instead of Dial() ? |
| 22:42.49 | WIMPy | No |
| 22:42.58 | F-JJTH | Nice ! |
| 22:43.09 | F-JJTH | I will play with that |
| 22:50.01 | F-JJTH | hmmm I got "Channel 'IAX2/192.168.0.253:4569-2507' unable to transfer" |
| 22:50.47 | F-JJTH | should I change something on my final server ? |
| 22:51.29 | F-JJTH | However the Dial() is working as expected (I'm using echo() for testing and I can hear my voice) |
| 22:54.25 | F-JJTH | hmmm is it true that MeetMe application can use _only_ ulaw codec ? Currently I'm using Speex and someone told me that MeetMe was converting automatically it in ulaw |
| 22:55.01 | WIMPy | MeetMe uses dahdi mixing, so it's at least restricted to G.711. |
| 22:55.56 | F-JJTH | G.711=ulaw |
| 22:56.34 | WIMPy | Or a-law |
| 22:56.59 | F-JJTH | so if I want reduce the CPU load on the server it would be better to use uLaw on client side, that way MeetMe has no convertion to do, right ? |
| 22:57.15 | WIMPy | yes |
| 22:57.56 | [TK]D-Fender | "less" |
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| 22:58.24 | F-JJTH | on the other hand, uLaw require 64kB/s while Speex require only 8kB/s .... I have to choose between CPU or bandwidth |
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| 23:00.58 | F-JJTH | is there a way to know why Asterisk is "unable to tranfer" ? |
| 23:01.31 | F-JJTH | I'm happy to know that asterisk can't transfer my call but I would be happier if he told me why :) |
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