| 00:16.22 | saxa | ~book |
| 00:16.22 | 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 |
| 00:18.34 | saxa | is this the latest book , covering *1.8 ? |
| 00:21.06 | saxa | yep, it is |
| 00:21.09 | saxa | :) |
| 00:21.28 | saxa | thx infobot |
| 00:25.47 | saxa | btw, anybody in here knows how can I get beack to me a wrongly transferred call ? |
| 00:26.38 | saxa | I mean, I transfer from my 1001 to 1003, but the call should go to 1002 , is there a way to retrive it back from 1003 and transfer it to 1002 ? |
| 00:26.47 | saxa | this is during blind transfer |
| 00:29.23 | godmachine-x6 | can someone here help me get SIP working with google voice on asterisk 1.8.4 ? i'd greatly appreciate it. i don't think it is nothing major i need fixed, just being new to asterisk i think i have just overlooked something in one of the config files. |
| 00:33.00 | saxa | godmachine-x6: i have no idea sorry, never tried it. |
| 00:33.40 | godmachine-x6 | well even with my sip configuration, in ekiga, it keeps saying could not register |
| 00:34.05 | godmachine-x6 | 1 sip peers [Monitored: 0 online, 1 offline Unmonitored: 0 online, 0 offline] |
| 00:34.45 | godmachine-x6 | i have ekiga set to : domain = localhost |
| 00:34.52 | godmachine-x6 | user = 101 |
| 00:34.58 | godmachine-x6 | authentication user = 101 |
| 00:44.55 | hardwire | whats the trick to defining an iax2 friend that uses a different inbound authentication username match than the peer name itself? |
| 00:45.09 | hardwire | I'd like to call my iax2 peer something other than the username. |
| 00:46.01 | WIMPy | Specify a username? |
| 00:51.11 | hardwire | thats already in there. |
| 00:51.29 | hardwire | When I make an outbound call it uses the username. Inbound appears to be matched to peer name vs username |
| 00:51.37 | hardwire | not sure if that's correct but when I make em match things work |
| 00:54.13 | WIMPy | Do you have type=user? |
| 00:54.21 | hardwire | type=friend |
| 01:37.10 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 02:00.44 | *** join/#asterisk n0tk (~n0tk@gateway/tor-sasl/n0tk) |
| 02:16.20 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 02:24.07 | *** join/#asterisk saliak (~kailas@ip68-9-228-184.ri.ri.cox.net) |
| 02:29.02 | *** join/#asterisk irieKen (~ken@wsip-24-249-206-198.sd.sd.cox.net) |
| 02:31.48 | godmachine-x6 | i have asterisk running, and trying to register SIP on ekiga |
| 02:31.58 | godmachine-x6 | ekiga keeps saying unable to register |
| 02:32.29 | godmachine-x6 | but while trying to connect from ekiga, the asterisk shell says registration successful for user 101 and 102 (the only two users i defined) |
| 02:32.41 | godmachine-x6 | when i try to dial something though it doesn't do anything |
| 02:32.47 | godmachine-x6 | can someone help with this? |
| 02:40.16 | irieKen | godmachine-x6: Hmm. Are you connecting from within your own network, or are the Asterisk server and Ekiga client on different networks? |
| 02:40.30 | godmachine-x6 | both on same machine |
| 02:40.39 | godmachine-x6 | just noticed this error |
| 02:40.46 | irieKen | And no firewall? |
| 02:40.59 | godmachine-x6 | Call from '' to extension '18889799949' rejected because extension not found in context 'default'. |
| 02:41.03 | godmachine-x6 | no firewall |
| 02:41.08 | godmachine-x6 | all this running on my gentoo install |
| 02:41.22 | irieKen | hmm. |
| 02:41.33 | irieKen | I'm hoping someone else will chime in;) |
| 02:41.43 | godmachine-x6 | i think it should be an easy fix |
| 02:41.48 | godmachine-x6 | im just overlooking something |
| 02:41.53 | godmachine-x6 | or didn't define something right somewhere |
| 02:42.11 | godmachine-x6 | i followed all the instructions for this on mario's blog |
| 02:42.19 | godmachine-x6 | do you know what im talking about? |
| 02:42.23 | irieKen | Nope. |
| 02:42.24 | godmachine-x6 | i can get the URL if you need it |
| 02:43.23 | godmachine-x6 | http://supermario-world.blogspot.com/2010/11/asterisk-18-and-native-google-voice.html |
| 02:44.35 | irieKen | Reading through it now. |
| 02:46.17 | irieKen | Did you assign a context for your extensions? |
| 02:46.27 | godmachine-x6 | i used the same config settings as he did on his files list |
| 02:47.00 | godmachine-x6 | ill paste a couple snippets for what i think should be "right" |
| 02:47.59 | godmachine-x6 | this is in my sip.conf |
| 02:47.59 | irieKen | k. |
| 02:48.06 | godmachine-x6 | [general] |
| 02:48.06 | godmachine-x6 | context=default |
| 02:48.33 | godmachine-x6 | here ill pastebin the whole config. i dont have a password for now since im testing |
| 02:48.58 | irieKen | k. |
| 02:49.34 | irieKen | I'm looking at Morio's config, and he has a default context for internal calls called mario-default |
| 02:50.05 | godmachine-x6 | http://paste.pocoo.org/show/433935/ |
| 02:51.20 | godmachine-x6 | here is my extensions.conf |
| 02:52.13 | godmachine-x6 | http://paste.pocoo.org/show/433938/ |
| 02:53.36 | irieKen | Honestly, this is beyond my area of expertise. |
| 02:53.43 | godmachine-x6 | thats ok man |
| 02:53.48 | godmachine-x6 | i appreciate you looking |
| 02:53.50 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 02:54.02 | godmachine-x6 | he uses a jabber.conf as well |
| 02:54.14 | godmachine-x6 | and thats the only place for your password to google voice |
| 02:54.22 | godmachine-x6 | im just wondering exactly how your even suppose to use this |
| 02:54.31 | godmachine-x6 | hell it may not be meant to be used by sipphone client |
| 02:54.38 | godmachine-x6 | does jabber clients have dialing abilities? |
| 02:56.30 | godmachine-x6 | look at this |
| 02:56.33 | godmachine-x6 | [default] |
| 02:56.33 | godmachine-x6 | exten => s,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})}) |
| 02:56.34 | godmachine-x6 | exten => s,n,Dial(SIP/101,10) |
| 02:56.34 | godmachine-x6 | exten => s,n, Hangup |
| 02:56.34 | godmachine-x6 | exten => 101, 1, Dial(SIP/101, 10) |
| 02:56.34 | godmachine-x6 | exten => 102, 1, Dial(SIP/102, 10) |
| 02:56.41 | irieKen | I believe that Google Talk does do audio signaling over XMPP. |
| 02:56.48 | godmachine-x6 | but look here |
| 02:57.06 | godmachine-x6 | Call from '' to extension '102' rejected because extension not found in context 'default' |
| 02:57.28 | godmachine-x6 | isn't that extension found right there in "default" ? |
| 02:58.32 | godmachine-x6 | i have 101 and 102 both logged in now and was just trying to test to see if they could call each other |
| 02:58.38 | godmachine-x6 | since it shouldn't have to use any google voice stuff |
| 02:59.29 | *** join/#asterisk seraphie (~erin@207.98.195.107) |
| 03:00.29 | irieKen | Looks that way, but again, I have no idea what's going on. |
| 03:00.35 | irieKen | :) |
| 03:07.15 | *** join/#asterisk war9407 (war@c-71-62-61-74.hsd1.va.comcast.net) |
| 03:08.02 | godmachine-x6 | wish someone was here that knew how to use asterisk pretty good |
| 03:12.05 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 04:18.43 | ChannelZ | godmachine-x6: the error is pretty self explanatory. Extension 102 doesn't exist in the context 'default' |
| 04:19.01 | godmachine-x6 | ChannelZ:: yea thats what i thought too |
| 04:19.11 | godmachine-x6 | but in the default context it does exist |
| 04:19.16 | godmachine-x6 | or at least it looks to me like it does |
| 04:19.25 | ChannelZ | dialplan show |
| 04:20.00 | godmachine-x6 | http://paste.pocoo.org/show/434038/ |
| 04:20.28 | godmachine-x6 | and i have modified that dialplan but it still shows 101.. i can't dial ANY number |
| 04:20.33 | godmachine-x6 | US telephone |
| 04:20.36 | godmachine-x6 | toll free |
| 04:20.37 | godmachine-x6 | 101 |
| 04:20.43 | godmachine-x6 | none of the numbers they all say that error |
| 04:20.43 | ChannelZ | Where's 102? |
| 04:20.55 | godmachine-x6 | its an asterisk user defined in sip |
| 04:21.03 | ChannelZ | so? |
| 04:21.10 | godmachine-x6 | forget it |
| 04:21.19 | godmachine-x6 | i didn't come here for an attitude |
| 04:21.27 | godmachine-x6 | been trying to mess with this all day |
| 04:21.28 | ChannelZ | I didn't come here to give one |
| 04:21.33 | godmachine-x6 | last thing im going to hear is someone being smart |
| 04:21.36 | *** part/#asterisk godmachine-x6 (~godmachin@h92.171.140.67.dynamic.ip.windstream.net) |
| 04:21.40 | ChannelZ | The error you posted |
| 04:21.42 | ChannelZ | well OK then. |
| 04:24.17 | ChannelZ | Maybe a dumb person will help him since I guess he doesn't want to hear from smart ones |
| 05:18.49 | *** join/#asterisk timahvo1 (~rogue@41.223.57.76) |
| 05:26.43 | *** join/#asterisk cerberus_za (~coert@196-210-218-150.dynamic.isadsl.co.za) |
| 05:48.15 | *** join/#asterisk g00gle (~thameema@c-98-248-232-219.hsd1.ca.comcast.net) |
| 05:48.34 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 05:48.56 | *** part/#asterisk g00gle (~thameema@c-98-248-232-219.hsd1.ca.comcast.net) |
| 05:50.06 | *** join/#asterisk WiretapSeven (~Wiretap@unaffiliated/wiretap) |
| 05:57.45 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 06:21.13 | *** join/#asterisk gravin (~gravin@113.210.255.138) |
| 06:22.15 | p3nguin | :) |
| 06:37.14 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 06:41.15 | jmls | I would be grateful if someone could give some advice on what phone to use. I want to be able to easily change the display on the phone when someone hotdesks. Prefably I'd like to be able to change it without rebooting the phone and via http |
| 06:41.18 | *** join/#asterisk radic (~radic@tmo-096-179.customers.d1-online.com) |
| 06:44.25 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 06:53.17 | *** join/#asterisk irroot (~irroot@41-132-209-115.dsl.mweb.co.za) |
| 07:32.41 | *** join/#asterisk timahvo1 (~rogue@41.223.57.77) |
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| 07:40.05 | godmachine-x6 | ChannelZ:: sorry for earlier was in a bad mood just wanted to appologize for showing my anger. it wasn't u |
| 08:19.52 | *** join/#asterisk bn-7bc (bjarne@pdpc/supporter/active/bn-7bc) |
| 08:38.22 | *** join/#asterisk hehol (~Adium@2a01:198:71d:0:21f:d0ff:fea1:568e) |
| 08:39.46 | *** join/#asterisk dieguito84 (~dieguito8@host131-132-dynamic.61-82-r.retail.telecomitalia.it) |
| 09:15.46 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 09:20.34 | godmachine-x6 | how exactly is asterisk suppose to work with google voice? |
| 09:21.03 | godmachine-x6 | am i not suppose to be able to just open up my SIP phone and dial out a norm 1xxx-xxx-xxxx number and go through ? |
| 09:21.17 | godmachine-x6 | these instructions online are so scattered i have racked my head all day |
| 09:21.24 | irroot | yes depending on the dial plan |
| 09:21.26 | godmachine-x6 | i can't even get any sounds when i try these dial plans |
| 09:21.37 | godmachine-x6 | irroot:: do you care to help me please? |
| 09:21.43 | irroot | i have not used it for ages |
| 09:22.00 | godmachine-x6 | well i dont understand what i need to be doing |
| 09:22.05 | godmachine-x6 | i have read read and read |
| 09:22.15 | irroot | patched it back in the day to work with DTMF |
| 09:22.21 | godmachine-x6 | tried tried and tried and none of it is making sense. |
| 09:22.45 | godmachine-x6 | even these sipgate instructions are crazy |
| 09:22.51 | godmachine-x6 | and they are a commercial company |
| 09:23.13 | godmachine-x6 | http://www.sipgate.com/faq/article/394/How_do_I_configure_Asterisk does that look like good instructions to you? |
| 09:24.22 | godmachine-x6 | im using gentoo and i don't want to have to set up a box dedicated just to asterisk. i was just wanting it to run for my one user to make / receive calls on google voice. since gizmo got shut down.. but this just doesn't make any sense |
| 09:24.42 | godmachine-x6 | i've got 3 configs i been messing with |
| 09:24.52 | godmachine-x6 | gtalk.conf, sip.conf, and extensions.conf |
| 09:24.56 | godmachine-x6 | no wait there are 4 |
| 09:24.59 | godmachine-x6 | jabber.conf too |
| 09:25.16 | irroot | jingle.conf |
| 09:25.21 | godmachine-x6 | i've tried every how-to i can find |
| 09:25.28 | godmachine-x6 | i've not read nothing about jingle.conf |
| 09:25.43 | godmachine-x6 | these google voice instructions aren't saying to put anything in that |
| 09:26.31 | godmachine-x6 | i never in my life have seen a layout of script / code like this asterisk stuf lol |
| 09:26.38 | irroot | http://www.voip-info.org/wiki/view/Asterisk+Google+Talk |
| 09:28.58 | godmachine-x6 | yea i have tried all that |
| 09:29.15 | godmachine-x6 | and i cant distinguish the implied changes of the code from the actual code in the configs |
| 09:29.17 | godmachine-x6 | for example |
| 09:29.45 | godmachine-x6 | exten => s,1,NoOp( Call from Gtalk ) |
| 09:29.45 | godmachine-x6 | exten => s,n,Set(CALLERID(name)="From Google Talk") |
| 09:29.45 | godmachine-x6 | exten => s,n,Dial(SIP/my_sip_phones) |
| 09:29.48 | irroot | https://reviewboard.asterisk.org/r/1312/ |
| 09:30.05 | godmachine-x6 | ok do i put my SIP username in there where it says SIP ? |
| 09:30.11 | godmachine-x6 | or is that a defined piece of code / variable |
| 09:30.22 | irroot | you put your name in |
| 09:30.39 | irroot | ~thebook might help |
| 09:30.45 | godmachine-x6 | name from what? the [definedname] from another config? |
| 09:30.49 | godmachine-x6 | or my actual gmail name? |
| 09:31.04 | irroot | ~thebook |
| 09:31.04 | 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 |
| 09:31.16 | godmachine-x6 | meh |
| 09:31.34 | godmachine-x6 | is this something im going to have to learn python and 2 or 3 other languages just to get this to work? |
| 09:31.58 | irroot | no dont think so its jut basic understanding of asterisk |
| 09:32.00 | godmachine-x6 | hell i'd give someone my username /pass to google voice if they'd just give me the configs that would work |
| 09:32.18 | godmachine-x6 | i've worked on this all day my eyes are crossing |
| 09:33.22 | godmachine-x6 | why can't there be a script to do all this. create the approrpiate setings in sip.conf, jabber.conf, extensions.conf, and gtalk.conf ? every script to automate it i have tried has required some modification in one of the other config files |
| 09:33.34 | godmachine-x6 | or i have found out it is for doing something entirely different than i expected |
| 09:33.51 | godmachine-x6 | all i want to do is receive/place calls through google voice using ekiga or x-lite |
| 09:34.01 | godmachine-x6 | i dont want nothing fancy.. |
| 09:34.28 | irroot | you need to patch asterisk see the RB link above there is a protocol change from google |
| 09:34.44 | godmachine-x6 | i have 1.8.4 with gentoo |
| 09:34.52 | godmachine-x6 | and i used the gtalk USE flag |
| 09:35.01 | godmachine-x6 | so it should have got patched through portage shouldn't it? |
| 09:35.12 | godmachine-x6 | its all compiled from source.. not using a binary or a premade package |
| 09:37.06 | irroot | well look at above patch |
| 09:37.17 | irroot | Gtalk may not work at all without it |
| 09:38.12 | *** part/#asterisk eject_ck (~eject_ck@62.205.134.210) |
| 09:46.51 | godmachine-x6 | does google change how the protocol works pretty often? |
| 09:47.36 | godmachine-x6 | because if i spend much more time on this and then they changge it im going to be really pissed if i have to wait for more people to start repairing this, becaue i have no idea what to do with this stuff. i just installed asterisk for the first time today lol |
| 09:48.11 | *** join/#asterisk war9407 (war@c-71-62-61-74.hsd1.va.comcast.net) |
| 09:48.46 | irroot | not sure i dont use it you can look at the change log |
| 09:50.34 | godmachine-x6 | google voice releases their changelog to the public? |
| 09:50.53 | irroot | no the chan_gtalk log |
| 09:50.55 | irroot | easier |
| 09:52.14 | godmachine-x6 | lol |
| 09:52.22 | godmachine-x6 | u mean to tell me this makes sense to you? |
| 09:52.25 | godmachine-x6 | https://issues.asterisk.org/jira/browse/ASTERISK-18084 |
| 09:52.26 | CaptainPants | [ASTERISK-18084] [Status: Triage] chan_gtalk only receives calls from Gtalk Android - https://issues.asterisk.org/jira/browse/ASTERISK-18084 |
| 09:53.10 | godmachine-x6 | hmm is that a bot? or is that someone pointing me to the same thing i was telling you to look at? |
| 09:54.01 | godmachine-x6 | he client android (gtalk) support voice and video, is ginglerbread 2.3.4, and version is 1.3. |
| 09:54.01 | godmachine-x6 | Asterisk normally receive voice calls, but do not for the client, text only. |
| 09:54.01 | godmachine-x6 | I think it has to do with Jingle capable: no, because the client's Web gtlak is yes, but the Android client is "NO" |
| 09:54.01 | godmachine-x6 | Android Client: |
| 09:54.23 | godmachine-x6 | is Yes is NO but NO is YES ? huh |
| 09:54.26 | godmachine-x6 | i give up |
| 09:54.27 | godmachine-x6 | lol |
| 09:54.53 | godmachine-x6 | ill wait for a nice gui app in about 5 yrs to come to ubuntu and ill just take their source package and compile it on my gentoo |
| 09:55.00 | godmachine-x6 | in the meantime ill have no phone haha |
| 09:56.42 | irroot | yeah might be |
| 09:58.38 | godmachine-x6 | i was under the impression at first that all i had to do was set my username info in the gtalk.conf |
| 09:58.45 | godmachine-x6 | set up the sip credentials in sip.conf |
| 09:58.56 | godmachine-x6 | add the dial plan to extensions.conf |
| 09:59.08 | godmachine-x6 | set up a jabber listener for gvoice in jabber.conf |
| 09:59.22 | godmachine-x6 | then use the account i created in sip.conf to connect via ekiga |
| 09:59.37 | godmachine-x6 | dial out a real land line / cell number for a US phone through ekiga and get connected |
| 09:59.47 | godmachine-x6 | guess it dont work like that though haha |
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| 12:31.21 | kuku | long live asterisk :) |
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| 13:05.06 | jasoncarter | My vps has the interfaces venet0:0 and venet0:1 bound to public IP addresses. Asterisk is set to listen on the IP of venet0:0 and I've added the default route to have the source IP of venet0:1. All trunks are registered and I've made a test call to one number and it enters the queue successfully but are there any problems that could occur in the future because of this? |
| 13:29.21 | *** join/#asterisk n0tk (~n0tk@gateway/tor-sasl/n0tk) |
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| 14:00.25 | fazendeiro | hi all! |
| 14:01.25 | fazendeiro | can someone tell me where i find a list off cell phones capable to be uses as a trunk via bluetooh with asterisk ? |
| 14:01.47 | fazendeiro | *list of |
| 14:02.24 | WIMPy | Why don't you use a propper gateway? |
| 14:03.45 | fazendeiro | what you mean? i'm new in this field. it's a device? |
| 14:03.52 | WIMPy | Yes |
| 14:05.14 | WIMPy | You can get them as SIP or ISDN with 1 to 32 channels. |
| 14:06.23 | fazendeiro | its just for mobile, rigth? the cell chips are connected in it? |
| 14:07.16 | WIMPy | You've got a BRI, PRI or ethernet on one end and an antenna at the other. |
| 14:09.49 | *** join/#asterisk timahvo1 (~rogue@41.223.57.73) |
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| 15:49.09 | *** join/#asterisk fraudory (~James@82.132.248.219) |
| 15:50.30 | fraudory | Hi all! |
| 15:58.46 | *** join/#asterisk vikapi (~vikapi@124.125.34.134) |
| 15:59.23 | *** join/#asterisk jmwpc (~jmwpc@c-76-103-168-194.hsd1.ca.comcast.net) |
| 16:06.04 | fraudory | I have a question re: asterisk call screening I was hoping someone might be able to give me a hand with? Is there any way so that when someone answers the call (to get the 'press 1 to...' menu) that the call still keeps ringing to other extensions until a point where someone actually accepts the call by pressing 1 (or responds to the menu in another way)? |
| 16:10.27 | WIMPy | Isn't that what happens? |
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| 16:11.38 | fraudory | If someone answers (and does or doesn't respond to the menu prompt) the call won't ring through to any more extensions |
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| 17:36.08 | raden | has anyone messed with A2billing ? |
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| 17:37.25 | WIMPy | yes |
| 17:37.54 | raden | how is it ? |
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| 22:43.57 | timeshell | jql Were you the one helping me yesterday with the 401 issues? |
| 22:44.15 | timeshell | And irroot |
| 22:47.21 | timeshell | Problem solved if you're interested in the fix. |
| 22:48.35 | p3nguin | Please share with the rest of us. |
| 22:59.37 | timeshell | I had to turn off VTP pruning on the switch between the asterisk server and the phones that were on the SGE2010P. |
| 23:00.16 | timeshell | Apparently the Cisco switch was pruning the ARP traffic from the phones |
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| 23:45.11 | kaushal | hi |
| 23:45.31 | kaushal | what does this command mean wanpipemon -i w1g1 -c Ta ? |
| 23:45.51 | kaushal | i know i means interface and -c means command |
| 23:46.04 | kaushal | not sure w1gi and Ta |
| 23:46.13 | kaushal | w1g1* |
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