| 00:00.59 | *** part/#asterisk l2cache (~l2cache@c-98-213-134-102.hsd1.il.comcast.net) |
| 00:02.43 | *** join/#asterisk krynnotaur (~krynnotau@unaffiliated/krynnotaur) |
| 00:04.19 | *** join/#asterisk sawgood (~sawgood@173-13-158-25-sfba.hfc.comcastbusiness.net) |
| 00:18.37 | sawgood | In a nutshell, if I am running Asterisk 1.6.2.6 .... (build from src) and say in three weeks a 'newer' release of the 1.6.2.x src is released ... |
| 00:18.52 | sawgood | Is it as simple to 'upgrade' as doing a ./configure make then make install |
| 00:19.12 | sawgood | Of course paying attention to any new requirements the newer build might need ... |
| 00:22.56 | carrar | should be |
| 00:23.07 | carrar | but always wise to skim through the changelog |
| 00:25.19 | *** join/#asterisk romb_work (~romb@unaffiliated/romb-work/x-7222485) |
| 00:27.00 | sawgood | thank you |
| 00:28.08 | *** join/#asterisk rampage73 (~rampage73@173-165-2-121-Illinois.hfc.comcastbusiness.net) |
| 00:44.05 | cecil_lincher | Hi. If I'm entering a macro using the M parameter in the Dial command, is there any way to return a variable from the macro at all? I'm trying to have a called leg press a button, read what button is pressed, then have the caller leg branch depending on what the called pressed in the macro |
| 00:44.23 | cecil_lincher | oof. I'm sorry in advance if that made no sense. Let me know if i need to reiterate |
| 00:44.24 | *** join/#asterisk KingDavidNYC (~Chris1232@pool-71-190-78-39.nycmny.east.verizon.net) |
| 00:51.20 | KingDavidNYC | anybody here? |
| 00:51.38 | ChannelZ | nope |
| 00:56.01 | KingDavidNYC | most be the holidays |
| 00:56.15 | KingDavidNYC | probably everyone is on spring break |
| 01:11.20 | *** join/#asterisk romb_work (~romb@unaffiliated/romb-work/x-7222485) |
| 01:11.23 | ChannelZ | my newphew is.. he's over here playing Little Big Planet :) |
| 01:12.34 | KingDavidNYC | I should be on spring break too |
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| 01:40.33 | hardwire | any way to query outbound registration status? |
| 01:40.51 | TJNII | sip show registry |
| 01:58.33 | *** join/#asterisk hhkahya (hluesea@213.144.112.178) |
| 02:01.21 | hardwire | TJNII: yeh.. I was thinking from the dialplan |
| 02:01.24 | hardwire | sorry. |
| 02:08.51 | *** join/#asterisk QbY (~kelvin@c-24-126-145-123.hsd1.ga.comcast.net) |
| 02:09.22 | QbY | Is G.722 (HD Voice) supported in Asterisk? |
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| 02:13.45 | carrar | yes |
| 02:14.14 | Gugge | QbY: yes, and the first link on google for "asterisk g722" tells you what version supports it too |
| 02:19.22 | hardwire | yeh... quit wasting poor carrars talent! |
| 02:21.44 | hardwire | his poor talents! |
| 02:21.47 | hardwire | gone. |
| 02:26.16 | carrar | This would have been the appropriate response: http://tinyurl.com/yeyec7m |
| 02:27.18 | QbY | anyone know why "make menuselect" would say to install ncurses or newt when they are already installed |
| 02:29.45 | *** join/#asterisk voipmonk (~shido6@c-98-207-80-167.hsd1.ca.comcast.net) |
| 02:29.58 | jaytee | Qby, did you run ./configure first? |
| 02:30.04 | QbY | yes sor |
| 02:30.06 | QbY | yes sir |
| 02:30.10 | jaytee | what distro? |
| 02:30.15 | QbY | centos |
| 02:30.27 | QbY | even used --with-ncurses=/usr/lib/ncurses |
| 02:30.48 | jaytee | odd, never had to specify the path to either with RHEL or CentOS |
| 02:30.55 | QbY | me neither |
| 02:35.24 | QbY | checking for mandatory modules: NCURSES... ok ..... No menuselect user interface found. Install ncurses, newt or GTK libraries to build one and re-rerun 'make menuselect'. |
| 02:36.17 | *** join/#asterisk dockimble (~dockimble@77.227.1.24) |
| 02:37.20 | carrar | rpm -qa | egrep -i "ncurses|newt" |
| 02:37.31 | carrar | You have the devel installed? |
| 02:37.40 | QbY | yes.. |
| 02:37.50 | QbY | i just untarred the downlaoded again, configured, and it worked |
| 02:37.53 | QbY | is baffled |
| 02:38.03 | carrar | user error :) |
| 02:38.44 | QbY | i wouldn't deny it |
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| 03:27.44 | hardwire | install ncurses-dev |
| 03:27.54 | hardwire | err.. haha.. I'm totally in the backlog |
| 03:33.21 | Slugs_ | slaps hardwire around a bit with a large trout |
| 03:33.29 | hardwire | 404 trout denied |
| 03:33.35 | Slugs_ | lol |
| 03:33.38 | hardwire | sashimi grade pacific salmon only |
| 03:33.39 | Slugs_ | oh its there |
| 03:33.49 | Slugs_ | ummm i love sushi |
| 03:33.54 | Slugs_ | spicy tuna! |
| 03:33.58 | hardwire | yum |
| 03:34.28 | Slugs_ | what are you up 2 |
| 03:34.51 | hardwire | I'm headed to another building to install the PBX configuration I've been working on for a while |
| 03:35.07 | Slugs_ | fun |
| 03:35.07 | Slugs_ | :) |
| 03:35.11 | hardwire | and finishing off my tostaditas |
| 03:35.39 | Slugs_ | w/ salsa? |
| 03:35.50 | *** join/#asterisk jsjc (~chatzilla@219-90-149-184.ip.adam.com.au) |
| 03:35.51 | jsjc | hello! |
| 03:36.18 | jsjc | how can I do so extension users can transfer calls between them? |
| 03:36.32 | hardwire | salsa - yeh |
| 03:36.50 | hardwire | jsjc: eh? |
| 03:37.33 | jsjc | hardwire: well think I am john in extension 101 calling jenny in extension 102 and then jenny tells me I put you through with peter 103. so the call at the end it is 101 to 103 instead of 101 to 102 |
| 03:37.56 | jsjc | i dont know how to explain I am new to this and my terminology most likely will be incorrect. |
| 03:41.44 | hardwire | the answer is yes. |
| 03:41.48 | hardwire | sleep soundly. |
| 03:42.02 | hardwire | also.. for the potty read: |
| 03:42.03 | hardwire | ~book |
| 03:42.04 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
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| 03:58.28 | ChannelZ | jsjc: your phone/softphone probably has a transfer function; If not (you're on analog phones with ATAs) you can configure a DTMF sequence in features.conf |
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| 09:14.46 | ChannelZ | it's oh so quiet |
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| 09:35.39 | |amadeus| | sup |
| 09:36.36 | |amadeus| | i cant sleep |
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| 11:43.22 | sa000 | Hi! the rtp traffic flows directory after the connection is established or it always pass from asterisk server |
| 11:43.34 | sa000 | Computer A ---------- Asterisk Server ----------- Computer B |
| 11:43.46 | sa000 | directly **** |
| 12:15.41 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 12:22.13 | ariel_ | sa000: if you are talking sip and have canreinvite=yes then rtp will bypass the asterisk, but if it's not set or set to no it will always go through the asterisk, also if the codec used are different then they will also go through the asterisk. |
| 12:25.23 | sa000 | ariel_: I am talking about RTP packets |
| 12:25.39 | ariel_ | yes |
| 12:25.43 | ariel_ | so am i |
| 12:26.11 | sa000 | because i am using wireshark to sniff the RTP packets and i can only see computer A and asterisk server communication if i am running wireshark on computer A |
| 12:27.13 | *** join/#asterisk arl (~arl@e176008046.adsl.alicedsl.de) |
| 12:27.17 | arl | hi all |
| 12:27.37 | ariel_ | hello |
| 12:27.48 | arl | my english is not so good so I would like to know if there is an german asterisk channel? |
| 12:28.45 | WIMPy | arl: I don't know one, but you could try to open one. |
| 12:29.29 | arl | WIMPy: well I could try but du the fact that I am starting with asterisk today, I think this wouldn't make sense |
| 12:29.47 | arl | but I will try to ask in english in this channel here |
| 12:30.39 | arl | so I would like to know what is state of the art when building an asterisk for use with ISDN connection to the outside world and also an internal S0-bus |
| 12:30.48 | ariel_ | I also don't know of any, but I also have not looked as I don't speak German |
| 12:31.43 | ariel_ | as long as you have a supported ISDN card, no issue there all basic same, about the S0-Bus I have no idea there |
| 12:31.47 | WIMPy | arl: Depends (as usual). But the latest is to use LCR. http://www.linux-call-router.de/ |
| 12:32.53 | WIMPy | The old mISDN doesn't work any more. I personally dislike the idea to have two apps running, but it works quite well. |
| 12:33.19 | WIMPy | And both parts can work independant of each other. |
| 12:34.13 | arl | ok so LCR and asterisk are two diffrent software-projects? |
| 12:35.02 | WIMPy | Yes, but they work together. LCR will only do ISDN. You connect them using cahn_lcr for Asterisk. |
| 12:36.12 | arl | hmm looks like I need to learn a lot :-) |
| 12:36.31 | WIMPy | Using chan_misdn is a little easier, but won't work with a current kernel. |
| 12:37.29 | arl | so lcr will do the ISDN part (connecting the line and the phones) and asterisk would be the PBX? |
| 12:37.51 | WIMPy | chan_misdn uses the old misdn and LCR uses misdn2, which has been in the standard kernel for some versions. |
| 12:38.43 | DennisG | arl just try it :) my english is also not good but i try it at least :) |
| 12:38.53 | WIMPy | LCR can do the PBX as well. So if you only want ISDN you don't need asterisk. But you need it if you also want to use voip or some other features. |
| 12:39.46 | arl | can I start with LCR and if needed "connect" asterisk later? |
| 12:39.53 | WIMPy | Sure |
| 12:40.04 | jaytee | seems to me arl's command of the English language is better than most whose native language is English :-) |
| 12:40.33 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 12:40.41 | WIMPy | But if you want to connct it to Asterisk, you would probably want to configure call routing in Asterisk, not LCR. But it can be mixed, if you want. |
| 12:41.22 | WIMPy | Hauptgewinn: Freie Auswahl :-) |
| 12:41.36 | ariel_ | Yes it does seem his English is perfect |
| 12:41.58 | arl | DennisG: well I thought, that ISDN is a "german-thing" so I thought, that it would be better to ask the questions for the first steps in german. |
| 12:42.51 | WIMPy | Yes, that's a good reason to start a german channel, I guess. |
| 12:42.56 | arl | jaytee and ariel_ well I think i can speak and write english "ok" but my native language is german and some of my british collegues sometimes laughting at me when I speak english ;) |
| 12:43.15 | WIMPy | OTOH it might split things even further. |
| 12:43.24 | arl | OTOH? |
| 12:43.36 | WIMPy | On The Other Hand |
| 12:43.46 | arl | ah I see |
| 12:44.10 | WIMPy | Demands of people used to ISDN and other seem to differ quite dramatically. |
| 12:44.11 | arl | well I think I will try it the "english-way" |
| 12:44.23 | WIMPy | others |
| 12:44.24 | ariel_ | Well in my view your doing great, and I for one would never laugh at someones accent. We have many people here with many different accents. (South Florida) |
| 12:44.50 | jaytee | bets that arl's british colleagues can't speak any german and are just being dicks |
| 12:45.05 | WIMPy | I find getting overlap dial working in Asterisk quite challanging. |
| 12:45.37 | arl | no no they are great guys and they don't really mind my english - they can understand my so it's ok :-) |
| 12:46.18 | |amadeus| | outet sich dann auch mal °_° |
| 12:46.23 | arl | :) |
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| 12:53.27 | sa000 | ariel_: thanks |
| 12:57.13 | |amadeus| | goeie dag meesterarend |
| 12:57.33 | meesterarend | hi |
| 12:57.38 | meesterarend | hoe gaan? |
| 12:58.26 | |amadeus| | goed ek was besig om op die web te lees |
| 12:58.49 | meesterarend | ok |
| 13:00.53 | DennisG | he, hier zijn ook nederlanders :P |
| 13:01.59 | meesterarend | uhm seg wee |
| 13:02.55 | |amadeus| | dennisg, afrikaans :P |
| 13:03.00 | meesterarend | ;-) |
| 13:03.56 | meesterarend | kan net genoeg nedeland verstaan om innie moeilikheid te kom met dit |
| 13:04.05 | |amadeus| | *G* |
| 13:05.02 | DennisG | haha :P |
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| 13:10.58 | pokui | does anyone know what happened to afelio that was supposed to knock asterisk off mac/bsd based machines? |
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| 14:55.28 | xnixan | Hi, what is the best ip phone to run with asterisk, and if possible cost wise? |
| 14:57.37 | carrar | Polycom, cost varies depending if you buy new or used and what model of polycom |
| 14:58.34 | carrar | 601 for execs, 430 for users works out well |
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| 16:03.56 | QbY | After upgrading to 1.6 I've found that my Cisco handsets register without issue; however I have Polycoms that will not.. http://pastebin.com/Ugb63rUd -- Ideas? |
| 16:11.18 | DennisG | change JMPOLYCOM to 5001 :) |
| 16:12.50 | QbY | that did it |
| 16:12.52 | DennisG | if that's not the solution then add also this to your phone settings: username = 5001 |
| 16:12.55 | DennisG | ooh oke :) |
| 16:12.57 | DennisG | ghehe |
| 16:13.04 | QbY | must be a change from the old |
| 16:13.11 | DennisG | i think so |
| 16:13.48 | DennisG | but i always use [5001] with username 5001 for a sip peer |
| 16:14.04 | DennisG | that works in asterisk 1.2, 1.4 and 1.6 :P |
| 16:14.14 | manxpower | QbY: did you read UPGRADE*.txt, the one document that should have all the important changes listed? |
| 16:14.47 | QbY | bows his head in shame |
| 16:15.04 | QbY | i knew we were going to encounter a lot of dialplan issues--which i was ready to fight |
| 16:16.31 | manxpower | READ IT. Better late than never. |
| 16:16.48 | manxpower | and don't just read the 1.6 one, read at least the 1.4 one too |
| 16:17.53 | manxpower | Traditionally it works like this [incomingusername] username=outgoingusername |
| 16:17.59 | manxpower | not on the same line of course |
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| 17:14.41 | maisa | hello |
| 17:14.49 | douglasawh | hi maisa |
| 17:15.02 | maisa | can i have some help starting with asterisk |
| 17:15.18 | douglasawh | I'm guessing I know the answer to this, but would it be possible to run asterisk off a hosted site...like Dreamhost? |
| 17:15.48 | douglasawh | we just need video conferencing...not phone #s and such. I don't know how modular things are...clearly I'm a n00b |
| 17:15.55 | maisa | anyone here using 2n voiceblue lite? |
| 17:16.27 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 17:17.15 | bn-7bc | douglasawh: : hol on i will see if the provide vps |
| 17:17.48 | bn-7bc | douglasawh: : soory for the typos |
| 17:18.06 | douglasawh | bn-7bc: no worries. thanks for the help! |
| 17:21.04 | bn-7bc | douglasawh: no, seems like they only do webhosting |
| 17:22.50 | bn-7bc | douglasawh: what you need to rub asterisk on a server at some other location is eirg´her a virtual private server (vps) or a dedicated server |
| 17:25.10 | douglasawh | bn-7bc: yeah, I don't think anyone has the hardware available, but I know the guy who does video conferencing at my job (~3500 people), so maybe he can throw something up I can use on the weekend...he's given me test stuff before...but I'm guessing since our phone network is all Cisco that that wouldn't fly |
| 17:25.22 | bn-7bc | hmm, cone to think of it does asterisk actualy do video conf? |
| 17:25.43 | douglasawh | ha, well, that would be the first part of fail in my question if not :) |
| 17:26.51 | bn-7bc | douglasawh: well I have not tryed video cobf yet so i can not say |
| 17:26.58 | *** join/#asterisk mxmasster (~mxmasster@cpe-76-166-197-184.socal.res.rr.com) |
| 17:27.00 | mxmasster | hello |
| 17:27.56 | douglasawh | bn-7bc: cool, thanks. I'll poke around a bit and see if I can find a solution. the problem is it needs to work on the three major platforms, so something like Ekiga won't work |
| 17:28.48 | mxmasster | via the extensions.conf is it possible to send a call to multiple gateways for a sip provider? |
| 17:28.52 | mxmasster | i.e. i have this currently |
| 17:28.53 | mxmasster | exten => _1NXXNXXXXXX,1,Dial(SIP/12345${EXTEN}@x.x.x.x) |
| 17:29.13 | mxmasster | do i need to have multiple lines, or can i have it in the dial block? |
| 17:33.44 | mxmasster | second question - i'm looking for an international dialplan example without 011 |
| 17:45.39 | douglasawh | don't know if anyone else was interested in my video conference question, but I found this: http://sourceforge.net/projects/dimdim/forums/forum/613324/topic/3376398 <-- dimdim/asterisk integration |
| 17:46.58 | bn-7bc | mxmasster: hmm, just one q, doues youre sip provider alow you to send calls withot authenticatibg? |
| 17:47.23 | mxmasster | authentication is based on ip address and tech prefix |
| 17:47.38 | mxmasster | so no "authentication" in the username/password sense |
| 17:49.17 | bn-7bc | mxmasster: ok that one was new to me but if it works fine, what dou you men with multiple gar´tways |
| 17:49.37 | bn-7bc | mxmasster: like failover? |
| 17:50.25 | mxmasster | failover, load balancing, round robin |
| 17:50.35 | mxmasster | i'd actually like round robin |
| 17:50.41 | bn-7bc | mxmasster: I have never had that setup so I need to google |
| 17:50.53 | mxmasster | :) thanks i've been doing the same thing |
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| 18:10.06 | WIMPy | Where do I find a complete documentation on dahdi/system.conf and dahdi-channels.conf, not just the examples? |
| 18:11.57 | drmessano | The examples are often the best source of documentation |
| 18:12.14 | drmessano | Google helps too |
| 18:13.34 | WIMPy | The examples don't explain grouping lines and I haven't found much using google so far. But I'd really like to understand the config, nut just copy and try. |
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| 19:56.04 | sawgood | If I want a statement in my dialplan to ADD 1408 to any 7 digits entered ... would this work 1408+XXXXXXX |
| 19:56.30 | sawgood | I tried dialing just seven digits (figuring the 1408 would be added), but it did not work |
| 20:01.32 | TJNII | exten => _XXXXXXX,1,Dial(SIP/1408${EXTEN}@account) |
| 20:05.31 | sawgood | 9|1408+NXXXXXX |
| 20:05.35 | sawgood | This is not working for me |
| 20:05.52 | sawgood | thank you for your tip! |
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| 20:13.05 | *** join/#asterisk alexfranci (~shiva@host74-34-dynamic.183-80-r.retail.telecomitalia.it) |
| 20:13.27 | alexfranci | hello! i have big problems with new version of asterisk (i upgraded from 1.2 to 1.4 today) |
| 20:13.38 | alexfranci | it can not find extensions! |
| 20:14.01 | alexfranci | i can register all my devices (i can see in CLI), but asterisk can't find extensions |
| 20:14.14 | alexfranci | how can i do? please help me, it would be nice :) |
| 20:15.09 | alexfranci | Call from '5338779' to extension '301' rejected because extension not found. |
| 20:15.14 | alexfranci | it is an example |
| 20:18.45 | alexfranci | plese help |
| 20:23.34 | qxork | is extension 301 listing in the context that 5338779 can dial? |
| 20:23.47 | alexfranci | qxork, can u help me? |
| 20:24.09 | qxork | alexfranci: is extension 301 listing in the context that 5338779 can dial? |
| 20:24.29 | alexfranci | yes sorry... the extension i use is [default] |
| 20:24.44 | alexfranci | for example, this is a user of my config file |
| 20:24.51 | alexfranci | [ip301_alex] |
| 20:25.15 | alexfranci | (i will pastebin it) |
| 20:25.24 | qxork | i guess the best thing would be to post your extensions.conf and sip (or users) |
| 20:25.27 | qxork | brb |
| 20:25.43 | alexfranci | http://pastebin.com/hWZ61033 |
| 20:25.54 | alexfranci | ok |
| 20:26.58 | alexfranci | [default] |
| 20:26.58 | alexfranci | ; |
| 20:26.58 | alexfranci | ; By default we include the demo. In a production system, you |
| 20:26.58 | alexfranci | ; probably don't want to have the demo there. |
| 20:26.58 | alexfranci | ; |
| 20:26.59 | alexfranci | include => demo |
| 20:27.09 | qxork | do you want to past the extensions? |
| 20:27.22 | alexfranci | yep |
| 20:27.55 | alexfranci | www.the-root.org/files/extensions.conf |
| 20:29.05 | alexfranci | as you can see qxork, [default] include demo |
| 20:29.16 | alexfranci | but i can't call demo extensions from ip301_alex |
| 20:29.20 | qxork | neither has ext 301 |
| 20:29.52 | qxork | I'm certain you dialing 1234 will work |
| 20:29.58 | qxork | as that is listed there |
| 20:29.59 | alexfranci | [Apr 4 22:29:50] NOTICE[20217]: chan_sip.c:14441 handle_request_invite: Call from 'ip301_alex' to extension '500' rejected because extension not found. |
| 20:30.08 | qxork | exten=>1234,1,Dial(SIP/ip301_alex) |
| 20:30.26 | alexfranci | [Apr 4 22:30:18] NOTICE[20217]: chan_sip.c:14441 handle_request_invite: Call from 'ip301_alex' to extension '1234' rejected because extension not found. |
| 20:30.41 | qxork | is ip301_alex registered? |
| 20:30.46 | alexfranci | yep |
| 20:31.03 | alexfranci | ip301_alex/ip301_alex 192.168.1.81 D N 5085 Unmonitored |
| 20:31.07 | alexfranci | i can see it from CLI |
| 20:31.17 | qxork | what is a result of sip show peer ip301_alex |
| 20:31.27 | alexfranci | i will try to reboot peer |
| 20:31.45 | qxork | instead, what is a result of sip show peer ip301_alex |
| 20:31.53 | alexfranci | i will pastebin result |
| 20:32.02 | alexfranci | http://pastebin.com/q2gwyzD7 |
| 20:33.52 | qxork | dial 1000 |
| 20:34.14 | alexfranci | [Apr 4 22:34:09] NOTICE[20217]: chan_sip.c:14441 handle_request_invite: Call from 'ip301_alex' to extension '1000' rejected because extension not found. |
| 20:34.55 | qxork | ok. Your messages do not match the posted extensions.conf |
| 20:35.15 | qxork | are you sure that's whats on the system? |
| 20:35.39 | alexfranci | yes |
| 20:35.46 | alexfranci | /etc/asterisk/extensions.conf |
| 20:36.05 | *** join/#asterisk alexfranci__ (~shiva@host74-34-dynamic.183-80-r.retail.telecomitalia.it) |
| 20:36.33 | alexfranci | how can i check it? |
| 20:37.46 | qxork | core show dialplan |
| 20:38.03 | alexfranci | *CLI> core show dialplan |
| 20:38.04 | alexfranci | No such command 'core show dialplan' (type 'help core show' for other possible commands) |
| 20:38.05 | alexfranci | wtf? |
| 20:38.12 | qxork | sorry |
| 20:38.18 | qxork | dialplan show |
| 20:38.35 | alexfranci | i will pastebin it |
| 20:38.49 | alexfranci | http://pastebin.com/cMu4bgPD |
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| 20:39.27 | alexfranci | why 500 does not work? (for example) |
| 20:39.42 | alexfranci | [Apr 4 22:39:34] NOTICE[20322]: chan_sip.c:14441 handle_request_invite: Call from 'ip301_alex' to extension '500' rejected because extension not found. |
| 20:39.56 | qxork | right. |
| 20:40.25 | qxork | this pastebin doesn't match the extensions.conf you posted. |
| 20:40.25 | alexfranci | have you found the problem? |
| 20:40.34 | qxork | so try a dialplan reload |
| 20:40.55 | alexfranci | *CLI> dialplan reload |
| 20:40.55 | alexfranci | No such command 'dialplan reload' (type 'help dialplan reload' for other possible commands) |
| 20:41.07 | qxork | what version again? |
| 20:41.15 | alexfranci | serverone:/etc/asterisk# dpkg -l|grep asterisk |
| 20:41.16 | alexfranci | ii asterisk 1:1.4.21.2~dfsg-3+lenny1 Open Source Private Branch Exchange (PBX) |
| 20:41.42 | alexfranci | i can restart asterisk... |
| 20:41.42 | qxork | kk... not sure why you wouln't want to run the latest version of 1.4 |
| 20:41.51 | qxork | try extensions reload |
| 20:42.07 | alexfranci | not such command |
| 20:42.16 | qxork | just type reload |
| 20:42.20 | alexfranci | just reload works |
| 20:42.28 | qxork | any errors get listed? |
| 20:42.45 | alexfranci | some warnings |
| 20:42.47 | alexfranci | not errors |
| 20:42.57 | qxork | and they were... |
| 20:43.06 | alexfranci | [Apr 4 22:42:50] WARNING[20322]: chan_sip.c:16800 add_realm_authentication: Format for authentication entry is user[:secret]@realm at line 928 |
| 20:43.06 | alexfranci | *CLI> [Apr 4 22:42:50] WARNING[20322]: chan_sip.c:16800 add_realm_authentication: Format for authentication entry is user[:secret]@realm at line 951 |
| 20:43.06 | alexfranci | [Apr 4 22:42:50] WARNING[20322]: chan_sip.c:16800 add_realm_authentication: Format for authentication entry is user[:secret]@realm at line 980 |
| 20:43.11 | alexfranci | [Apr 4 22:42:50] WARNING[20322]: frame.c:1320 ast_parse_allow_disallow: Cannot allow unknown format 'g711u' |
| 20:43.11 | alexfranci | [Apr 4 22:42:50] WARNING[20322]: frame.c:1320 ast_parse_allow_disallow: Cannot allow unknown format 'g711a' |
| 20:43.35 | alexfranci | [Apr 4 22:42:50] WARNING[20306]: chan_zap.c:12594 process_zap: Ignoring switchtype |
| 20:43.35 | alexfranci | [Apr 4 22:42:50] WARNING[20306]: chan_zap.c:12594 process_zap: Ignoring signalling |
| 20:43.36 | alexfranci | [Apr 4 22:42:50] WARNING[20306]: chan_zap.c:12594 process_zap: Ignoring rxwink |
| 20:43.43 | qxork | how did you install asterisk? |
| 20:43.51 | alexfranci | apt-get |
| 20:45.35 | alexfranci | i fixed problems with auth |
| 20:45.56 | alexfranci | but same problems however |
| 20:46.49 | alexfranci | i fixed g711[u-a] problems |
| 20:47.07 | alexfranci | same problems with the extensions -.- |
| 20:47.13 | qxork | stand by |
| 20:47.15 | alexfranci | ok |
| 20:47.37 | alexfranci | in the mean time i wil pastebin to you my sip.conf |
| 20:49.04 | alexfranci | www.the-root.org/files/sip.conf |
| 20:49.46 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 20:50.55 | alexfranci | oh my god |
| 20:50.57 | alexfranci | it works |
| 20:51.11 | alexfranci | *CLI> show dialplan |
| 20:51.11 | alexfranci | [ Context 'ael-demo' created by 'pbx_ael' ] |
| 20:51.18 | alexfranci | why ael-demo? |
| 20:51.22 | alexfranci | and not only demo? |
| 20:51.34 | alexfranci | i changed context from demo to ael-demo and it works |
| 20:51.43 | alexfranci | what came ael-demo from? |
| 20:51.47 | alexfranci | *where |
| 20:54.03 | qxork | it's from ael extensions |
| 20:54.16 | qxork | you are not loading extensions.conf |
| 20:54.25 | qxork | may want to check the permissions and reload |
| 20:55.23 | qxork | you can also check /etc/asterisk.conf to see if it's checking /etc/asterisk for the files |
| 20:55.35 | qxork | I don't know where the apt-get installion happens. |
| 20:55.45 | qxork | Generally, I recommend installing from source |
| 20:56.01 | qxork | it's not difficult and will correct many issues since 1.4.21 |
| 20:56.44 | qxork | here's some examples. http://www.voip-info.org/tiki-index.php?page=Asterisk+Linux+Debian |
| 20:57.50 | alexfranci | i fixed permissions |
| 20:57.53 | alexfranci | but problems remains |
| 20:58.11 | alexfranci | http://pastebin.com/UmfEhLx8 |
| 20:58.17 | alexfranci | this is asterisk.conf |
| 20:58.38 | qxork | you had permission errors? |
| 20:59.14 | alexfranci | yes, i had. but not now |
| 20:59.32 | alexfranci | all files are readble by asterisk |
| 20:59.41 | qxork | ok. I would restart asterisk at the very least. And if I were you, I'd install from source. |
| 21:00.00 | alexfranci | okay, i will install from source |
| 21:00.12 | qxork | sounds like a plan. |
| 21:00.15 | alexfranci | i hope i won't have such problems |
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| 22:13.04 | grkblood13 | anyone know how to turn off the dial tone on an xlite? |
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| 22:30.02 | Whtsup | does anyone know how to rearrange trunk order in a2billing |
| 22:31.22 | Whtsup | ?? |
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| 23:35.04 | manxpower | perhaps if someone had an Asterisk question.... |
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| 23:36.15 | Slugs_ | that bored huh? |
| 23:43.04 | ariel_ | evening folks |
| 23:43.43 | Slugs_ | evening |
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| 23:58.42 | QbY | Trying to setup realtime. I have built Asterisk with Postgres support, configured extconfig.conf, res_pgsql.conf --but get this error: Realtime mapping for 'sippeers' found to engine 'pgsql', but the engine is not available |