| 00:00.02 | kaushal | leifmadsen: I am still unable to resolve the Error 101 issue. Please suggest |
| 00:00.17 | leifmadsen | I don't know what "the Error 101 issue" is |
| 00:00.37 | kaushal | I have emailed you about my issue |
| 00:00.54 | leifmadsen | You're going to elaborate :) |
| 00:00.58 | kaushal | sure |
| 00:01.02 | *** part/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 00:01.05 | leifmadsen | to have to elaborate* |
| 00:02.33 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 00:02.40 | kaushal | I have 8 port Sangoma PRI Card attached to my server. I have 8 E1 ISDN PRI lines |
| 00:02.47 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-fvhsywkiaxhybfog) |
| 00:03.30 | kaushal | when i enable 8 * 30 channels, i am stuck with error 101 issue. whereas when i use 8 * 13 channels or 4 * 25 channels it works perfectly fine |
| 00:03.48 | kaushal | I am using Asterisk to make Outbound Call |
| 00:04.10 | kaushal | I am calling approximately 230 Phone numbers |
| 00:04.19 | kaushal | leifmadsen: Any clue ? |
| 00:04.23 | leifmadsen | nope |
| 00:04.34 | leifmadsen | I don't install PRI systems anymore, just SIP |
| 00:04.48 | leifmadsen | and I don't use Sangoma hardware, so there isn't much advice I can offer |
| 00:05.30 | kaushal | leifmadsen: where do i seek help ? |
| 00:06.26 | leifmadsen | you're in the right spot, but it will require patience. If you can't get help here, then I would suggest the asterisk-users mailing list. Failing that, you should seek out a professional consultant who has experience in the area you seek. I would suggest Core Telecom (coretel.ca) if you're looking for recommendations. |
| 00:08.44 | f2knight | p3nguin, ahh those dagnagget [] of asterisk! |
| 00:09.21 | f2knight | leifmadsen, off topic but just ordered your book, looking forward to a good read. |
| 00:09.41 | leifmadsen | f2knight: yay! hope you enjoy it! |
| 00:10.01 | *** topic/#asterisk by leifmadsen -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.0.0-beta1 (2011/07/22), 1.8.6.0 (2011/08/31), dahdi-linux 2.5.0 (2011/08/08), dahdi-tools 2.5.0 (2011/08/08), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 00:11.28 | f2knight | leifmadsen, I think I will I enjoyed your fist one (dog eared to death it is) new one I got as PDF , android tablets the best e-Reader yet. |
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| 00:18.53 | WIMPy | kaushal: Do you still have RNR and/or REJ frames? |
| 00:19.02 | nwxi | is there any way of compiling up dahdi without requiring kernel sources? i don't think I need dahdi, but im not sure if asterisk would function without it? |
| 00:19.38 | WIMPy | no, yes (but without meetme) |
| 00:20.41 | nwxi | ok, so meetme requires dahdi but nothing else. |
| 00:21.28 | kaushal | WIMPy: RNR or REJ ? |
| 00:21.33 | WIMPy | and page uses meetme. |
| 00:21.38 | WIMPy | But that's it. |
| 00:22.00 | nwxi | Ok. Cool. Thx :) |
| 00:22.05 | WIMPy | kaushal: The switch telling you to stop. |
| 00:22.07 | nwxi | Know anywhere I can get british english sounds? |
| 00:22.11 | p3nguin | You don't necessarily have to use MeetMe, either. There's always ConfBridge. |
| 00:23.27 | f2knight | Q: Trying to figure out the bestway. (or anyway) to have a caller, placed in a different context as a call comes to a duration. e.g. Caller is calling in, and when I dial the internal user, I use the L option to notify the caller that there is 1 min left on the call. What I want to do is have the caller press 1 to purchase more time, other wise they are disconnected. |
| 00:23.37 | nwxi | Ok. See, i'm currently running a trixbox server, but want to set up an asterisk server via just creating it from scratch as I learn the system. So I'm compiling up asterisk at the moment. What would be the best reference material (free ideally) |
| 00:25.38 | kaushal | WIMPy: i can share you the details |
| 00:25.45 | kaushal | Do you want me to pastebin ? |
| 00:26.09 | WIMPy | go ahead |
| 00:26.29 | kaushal | I mean the trace logs from telco and Sangoma PRI Card |
| 00:26.50 | f2knight | nwxi, http://ofps.oreilly.com/titles/9780596517342/ |
| 00:26.55 | WIMPy | yes |
| 00:27.32 | kaushal | WIMPy: can i pvt message you ? |
| 00:27.48 | WIMPy | yes |
| 00:31.04 | f2knight | leifmadsen, I have a quick question on a personal flavor ... FastAGI or AMI to do Dialplan/CallFlow? I am not a big fan of AMI (mostlly because I do not know it) but I do like xAGI because I can keep the logic off the server all together. |
| 00:31.43 | leifmadsen | f2knight: well they are really separate things :) One is Asterisk triggering a program/script, and the other is a program/script interacting and controlling asterisk |
| 00:31.49 | leifmadsen | so it's comparing apples and oranges |
| 00:32.57 | f2knight | leifmadsen, xAGI being asterisk calling a program/script. right. |
| 00:33.09 | leifmadsen | yes |
| 00:33.15 | leifmadsen | Asterisk dialplan triggers the AGI |
| 00:33.23 | leifmadsen | the script interacts by communicating with AMI |
| 00:33.32 | leifmadsen | Asterisk --> script |
| 00:33.35 | leifmadsen | script --> Asterisk |
| 00:33.46 | leifmadsen | former being AGI, latter being AMI |
| 00:34.25 | f2knight | leifmadsen, so with AMI. you can have a totally external process start/establish and execute asterisk actions. |
| 00:34.54 | f2knight | leifmadsen, where as with xAGI you have to have at least some dialplan logic and wait for asterisk to call the script to then execute other commands. |
| 00:34.56 | f2knight | ? |
| 00:35.19 | leifmadsen | yes, so incoming calls could be using AGI |
| 00:35.27 | leifmadsen | outgoing calls could be triggered via AMI |
| 00:35.55 | f2knight | so AMI would be an ideal tool to do something like polling a database for customer notifications. |
| 00:36.16 | leifmadsen | no, other way around |
| 00:36.30 | leifmadsen | AGI would be used to lookup customer data |
| 00:36.53 | f2knight | sorry... I mean lets say you wanted to call customers to remind them of an appointment. |
| 00:36.55 | leifmadsen | AMI could be used by a website to trigger a call between to end points for example |
| 00:37.03 | leifmadsen | yes, then you would use AMI to trigger the call setup |
| 00:37.08 | f2knight | AMI would work better then agi because you could cron up the process |
| 00:37.23 | f2knight | okay so apples and oranges in there proper bins, |
| 00:37.28 | leifmadsen | yep |
| 00:37.42 | f2knight | with AGI, can I actually establish a full dialplan logic? |
| 00:38.10 | f2knight | in other words, can I just do something like this.. exten 500 => AGI(...) |
| 00:38.31 | f2knight | and have the agi script handle everything ? (Mostly thinking IVR actions here,) |
| 00:41.16 | *** part/#asterisk nwxi (~nwxi@client-86-23-125-82.brhm.adsl.virginmedia.com) |
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| 02:26.06 | shmaltz | hi everyone |
| 02:29.05 | leifmadsen | f2knight: yes, I know some people who do exactly that |
| 02:29.21 | leifmadsen | just let the xAGI handle all the logic by just calling it at the start of the call |
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| 03:25.35 | f2knight | leifmadsen, Thats what I was thinking. What about performance though? does this slow the asterisk box down? I am thinking the dialplan part would be an agi call and an error message if not connected. nothing else. But I wonder if its faster to process by dialplan in astereisk or if xagi is just as fast. |
| 03:26.36 | f2knight | leifmadsen, the one part I seem to be having a problem with is they WaitExten part. After I stream a file or get the DTMF back. I am sure I am probably just missing something (more reading the guide) |
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| 03:30.38 | ChannelZ | I would think WaitExten would return control so to speak back to the dialplan. Background too, |
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| 03:44.29 | f2knight | Q: Ackward behavior with DIAL()... I have a macro defined in AEL. but when using DIAL(....M(mymacro)) asterisk complains about No such context 'macro-&acceptcall(300) |
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| 03:46.38 | f2knight | ChannelZ, in my 'limited' testing that is what I thought would happen too but what happened was that it left the FAGI script and jumped right to the next part of the the Dialplan. Thinking I could put a WaitExten in the dialplan didn't help it just passed right buy with out the proper background before it. Of course I think now that I might have been calling the playback wrong from the AGI as I was using STREAM FILE. |
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| 04:57.53 | f2knight | I have hunted and hunted. With no solution in sight. Has anyone implimented a Dial statement calling a macro written in AEL? I want to verify this is a bug and not just me being dense. |
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| 05:13.10 | kaldemar | f2knight: how are you calling the macro? |
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| 05:14.39 | f2knight | kaldemar, Dial(SIP/300,45,M(acceptcall) |
| 05:14.45 | f2knight | kaldemar, Dial(SIP/300,45,M(acceptcall)) |
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| 05:16.08 | kaldemar | f2knight: and you have a context called macro-acceptcall? |
| 05:16.44 | f2knight | kaldemar, in extensions.ael I have macro accpetcall () {...} yes |
| 05:16.59 | f2knight | kaldemar, this is all in AEL. I could pastebin |
| 05:17.58 | kaldemar | do you see it with dialplan show? |
| 05:19.17 | irroot | morning folks |
| 05:19.39 | f2knight | kaldemar, http://pastebin.com/u291SWW2 |
| 05:20.40 | f2knight | kaldemar, yes it says not found. but it is there. if I rename it to macro macro-acceptcall it sees it and says its need an s extension, but that does not work with the ael parser macro can not have an extension, it is assumed s already |
| 05:20.45 | f2knight | irroot, morning |
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| 05:22.23 | f2knight | kaldemar, I think it is is a bug. in the way the M() is executed in the Dial command, maybe hard coded to only look for macro-* names. |
| 05:22.50 | f2knight | kaldemar, I even tried calling it the way you call a macro in ael with a ¯oname |
| 05:23.15 | f2knight | kaldemar, then I get error about macro-¯oname not found, which is why I think its a bug |
| 05:23.35 | kaldemar | f2knight: yes it will look for macro-macroname |
| 05:24.28 | f2knight | kaldemar, So I am assuming this means you can not use the M() function of the Dial command in side of AEL... |
| 05:24.39 | f2knight | kaldemar, I was hoping others could confirm this |
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| 05:42.14 | kaldemar | f2knight: looks like you can't make a context in AEL that is caller macro-something, and M() is indeed hard codec to look for macro-something. |
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| 05:50.17 | f2knight | kaldemar, would you classify this as a bug? |
| 05:57.39 | kaldemar | f2knight: maybe, but don't expect anyone else to. you'll probably get a "don't mix AEL and .conf" if you file a bug report. |
| 05:59.06 | f2knight | kaldemar, I don't even use .conf! Heck I don't even load pbx_conf.so!!! |
| 05:59.57 | f2knight | kaldemar, I did confirm that after converting the macro to .conf format and activating pbx_conf, it calls the macro from the conf. |
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| 06:00.20 | f2knight | ~report bug |
| 06:00.33 | f2knight | do you know the bug link? |
| 06:01.06 | singler | bugtracker? |
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| 06:04.23 | f2knight | yes looks like i have to have an account. |
| 06:05.10 | singler | https://issues.asterisk.org/jira/ |
| 06:05.11 | f2knight | okay so now that I know there is a bug in the DIAL command for the M()... with AEL.. Anyone know how to put in a call screen feature using AEL? |
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| 06:22.30 | schmidts | good morning |
| 06:23.52 | irroot | lo there schmidts |
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| 06:44.53 | f2knight | morning schmidts |
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| 06:52.21 | f2knight | hey guys is the bugtracker down or is it just me? |
| 06:55.35 | schmidts | f2knight same for me |
| 06:56.16 | f2knight | schmidts, okay thanks. :( |
| 06:56.50 | kaldemar | it is up, just slow as hell. |
| 06:57.23 | schmidts | now it works |
| 06:57.25 | kaldemar | hm. now it seems ok again. |
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| 07:47.05 | Dovid | morning all |
| 07:50.03 | Maliuta | It's past beer'o'clock ... not morning at all :) |
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| 07:52.11 | irroot | Maliuta hehe im off to the tropical region this afternoon will get a frosty on the beach and prevent the monkeys getting it but tonight will be in the drakensburg picking out a surfing spot for dec 21 2012 |
| 07:52.52 | Maliuta | it's cold here ... I just sat out the back and watched the temp drop from +23c to +20c |
| 07:53.24 | Maliuta | irroot: when you live in a decent part of the world cold has a different meaning :) |
| 07:54.03 | irroot | i got me 23c at 10am |
| 07:54.39 | Maliuta | irroot: I like my summers, where it's 28c at 7am :) |
| 07:55.16 | Maliuta | although going +28c with 90% humidity to -35c in one plane trip is a bit screwed up |
| 07:55.31 | irroot | lol indeed |
| 07:55.45 | Maliuta | irroot: you need to experience South East Queensland for some time |
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| 07:56.28 | irroot | got a sheila for me :P |
| 07:58.17 | Maliuta | irroot: sure, you can take the one I have a restraining order against :P |
| 07:58.43 | Maliuta | oh, and you might want to remove the "root" from your nick ... root is something else here :P |
| 07:58.46 | irroot | im not talking about the sheep .... |
| 07:58.56 | Maliuta | Sheep is New Zealand |
| 07:58.59 | irroot | oh here its the same :P |
| 07:59.20 | Maliuta | ahh, .za |
| 07:59.27 | Maliuta | I know a few expats from there |
| 07:59.45 | irroot | 1/2 of your pop is expat .za's |
| 07:59.58 | Maliuta | pity your best cricketer is now the coach for england |
| 08:00.19 | Maliuta | irroot: that and south africans |
| 08:00.25 | irroot | hey lets rather talk religion or politics :P |
| 08:00.34 | Maliuta | rofl |
| 08:00.49 | Maliuta | although we did just kick sri lanka's butts |
| 08:01.04 | Maliuta | we're in a rebuilding stage |
| 08:01.06 | irroot | yeah its all rugby mode here at the moment |
| 08:01.19 | Maliuta | world cup in the land of sheep fuckers |
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| 08:03.49 | Maliuta | obviously no Kiwi's in here ... they're all over here and on welfare. Suprised there is anyone left to turn on the lights for the cup |
| 08:04.11 | irroot | hehe |
| 08:04.51 | Maliuta | so how is Rhodesia these days? |
| 08:05.02 | irroot | was up there 2 years back |
| 08:05.13 | irroot | doing well |
| 08:05.17 | Maliuta | One of my friends worked with the MDC for some time |
| 08:05.37 | Maliuta | he's been smuggled in and out more than once |
| 08:05.41 | irroot | i was in harare few years ago was up in chirundu/chinoyi |
| 08:06.09 | irroot | hehe they call it smuggling i call it a 100R bribe at the border if they spot you |
| 08:06.41 | Maliuta | does currency still have value? ;) |
| 08:06.41 | irroot | a bag of maize meal would get you armed escort |
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| 08:07.04 | irroot | they use USD only now have couple 100T zim dollar notes at hom e |
| 08:07.26 | irroot | were good for a round of beers and snacks when they abandoned it |
| 08:08.33 | Maliuta | wonder what the value of a .50cal round into "the man" would be worth |
| 08:09.29 | irroot | Maliuta the Zanu-PF new headquaters in town has a massive rooster there symbol painted on the top |
| 08:09.45 | irroot | its known as the bigest cock up in town |
| 08:10.50 | schmidts | irroot :D |
| 08:11.11 | Maliuta | irroot: that's just because I haven't unziped my fly for measuring ;P |
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| 08:21.29 | iulhk | hi all, will anybody guide how to use this dialplan " exten => _X.,n,VoiceMail(${EXTEN}@${EXTEN}) " extension in AGI ? |
| 08:22.14 | irroot | Exec ?? |
| 08:22.48 | iulhk | i am trying this way but getting error .."$AGI->exec('VoiceMail', "SIP/$callerid", "$callee@callee");" ? |
| 08:24.30 | kaldemar | ${EXTEN}@${EXTEN} does not match with 'VoiceMail', "SIP/$callerid", "$callee@callee" |
| 08:25.03 | kaldemar | you're trying to give two arguments to the VoiceMail app from AGI now. |
| 08:27.09 | iulhk | ok |
| 08:30.52 | *** join/#asterisk RZero (~Mick_Love@gateway-hq-uk.oxygen8.com) |
| 08:32.11 | *** join/#asterisk MrSmurf (~MrSmurf@unaffiliated/mrsmurf) |
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| 08:40.46 | *** join/#asterisk Polysics (~Luca@host162-75-dynamic.245-95-r.retail.telecomitalia.it) |
| 08:40.48 | Polysics | hello |
| 08:41.21 | Polysics | caller calls help desk, help desk parks him and finds an available translator, then ALL 3 get connected and start a call |
| 08:41.33 | Polysics | wha'ts the asterisk function flow there? |
| 08:41.43 | Polysics | just call parking + a conference room? |
| 08:42.41 | Polysics | asterisk 1.8 driven through AGI/AMI by Adhearsion, so i can basically do as i please :-D |
| 08:46.08 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
| 08:53.28 | *** join/#asterisk klashniv (~klashniv@langw.roketelkom.co.ug) |
| 08:55.12 | klashniv | hullo, am running asterisk 1.4.31 and have a quick question. how can I reset the callno usage for an IP as listed by 'iax2 show callnumber usage' |
| 08:55.20 | irroot | http://rss.slashdot.org/~r/Slashdot/slashdot/~3/wh40qC2XWrg/Why-the-Fax-Machine-Refuses-To-Die FYI /. |
| 08:56.37 | klashniv | I have a IP that has requirecalltoken=no and 70 current calls but is shown as having 2048 under callno usage |
| 08:58.55 | garymc | Hi I need help again desperatley, phones arnt hitting the server, but the the isdn is working as I can phone in from outside just here the hold music and welcome message. no phones in office ring though |
| 08:58.58 | garymc | and now no phone will initialize |
| 08:59.00 | garymc | HELP!!! |
| 09:00.54 | irroot | phones are sip ? |
| 09:01.05 | irroot | are they registered "sip show peers" |
| 09:01.09 | *** join/#asterisk Smiley (~Smiley@last.fm/resident-dj/Smiley) |
| 09:01.21 | irroot | is the server locked up "core show locks" |
| 09:01.40 | Smiley | is waving to M.L. |
| 09:01.53 | klashniv | anyone have any idea on how to reset callno usage counter? |
| 09:02.33 | RZero | is waving to Smiley o/ |
| 09:03.48 | irroot | klashniv use the power switch :P |
| 09:03.59 | irroot | but there is probably a better way |
| 09:04.03 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 09:04.42 | klashniv | I was trying not to restart the server as it has live calls, guess will have to wake up very early in the morning where there is low traffic |
| 09:04.48 | *** part/#asterisk RZero (~Mick_Love@gateway-hq-uk.oxygen8.com) |
| 09:05.02 | klashniv | probably a good time to upgrade asterisk as well |
| 09:05.23 | Dovid | irroot: Whats happening with your T.38 patch? |
| 09:05.44 | *** join/#asterisk garymc (~chatzilla@host81-148-107-162.in-addr.btopenworld.com) |
| 09:05.46 | irroot | Dovid the T.38 is in 10 and works |
| 09:05.55 | irroot | 1.8 is in my svn branch |
| 09:06.03 | klashniv | in the meantime, I have increased the callno limit for the peer |
| 09:08.09 | garymc | I need a hired help right NOW |
| 09:08.46 | kaldemar | garymc: you might want to describe the task... |
| 09:08.55 | Chainsaw | kaldemar: They're consultants. |
| 09:09.00 | Chainsaw | kaldemar: They don't care as long as you pay. |
| 09:09.42 | Chainsaw | enables the UK consultant call signal and watches 3 people drop out of the corner like ninjas |
| 09:09.44 | f2knight | Q: is it possible to do the equivalent of this Dial(SIP/300,${MAXRING},U(screen2,s,1)A(welcome)dL(60000:30000:2000)S(1210)Fgm) with an originate command or some other method over AGI? |
| 09:11.01 | f2knight | yea that was a brain twister. |
| 09:11.33 | kaldemar | does exec not do the trick for you? |
| 09:13.04 | f2knight | kaldemar, no I tried exec ( 'DIAL', ' all the options here ') but got an error of invalid command. |
| 09:13.24 | f2knight | kaldemar, oh I am using pyst. and called agi.execute() directly. |
| 09:16.22 | *** join/#asterisk kabanara (7ca81857@gateway/web/freenode/ip.124.168.24.87) |
| 09:16.40 | kabanara | going through the a channels sorry |
| 09:16.42 | *** part/#asterisk kabanara (7ca81857@gateway/web/freenode/ip.124.168.24.87) |
| 09:16.50 | kaldemar | f2knight: the library you're using is to blame. if you do it directly, it will work. |
| 09:17.14 | garymc | my Asterisk NOw box no longer has an eth0 connection. A server I unplugged must have been distributing the DHCP |
| 09:17.28 | garymc | the server I unplugged was a LTSP server |
| 09:17.35 | *** part/#asterisk nny (~SM@cpe-174-107-223-014.sc.res.rr.com) |
| 09:17.52 | f2knight | garymc, the simple answer would be to plug it back in :) |
| 09:17.52 | kaldemar | f2knight: i just tried "exec Dial SIP/300,10,U(screen2,s,1)A(welcome)dL(60000:30000:2000)S(1210)Fgm" |
| 09:18.40 | f2knight | kaldemar, phpagi? |
| 09:19.21 | f2knight | kaldemar, or pyst? |
| 09:19.39 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 09:20.03 | garymc | f2knight I have but it doesnt work |
| 09:20.20 | kaldemar | f2knight: i weren't using any library, just a shell script that issued the actual agi command directly. |
| 09:20.44 | f2knight | kaldemar, will try it over nc in a second. |
| 09:24.04 | *** join/#asterisk cyford (~cyford@c-24-99-17-185.hsd1.ga.comcast.net) |
| 09:27.48 | *** join/#asterisk alfadir (~alfadir@79.141.248.181) |
| 09:31.56 | f2knight | kaldemar, your right it worked. Looks like I will have to patch my AGI Class |
| 09:41.31 | f2knight | kaldemar, well that was easy.. on a whim I just added "EXEC" before the dial statement and it worked. |
| 09:58.10 | *** join/#asterisk e7e5 (~rudenko@188.134.2.33) |
| 09:59.06 | irroot | false ;while [ "$?" != 0 ];do ....;done <- how to quickly loop a function till it succeeds |
| 10:03.55 | *** join/#asterisk garymc (~chatzilla@host81-139-71-64.in-addr.btopenworld.com) |
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| 10:16.24 | *** join/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
| 10:21.40 | *** join/#asterisk merlin8282 (~merlin828@AStrasbourg-554-1-204-183.w83-203.abo.wanadoo.fr) |
| 10:21.50 | merlin8282 | Hello |
| 10:22.05 | merlin8282 | How can I see if a queuemember is paused or not ? |
| 10:22.39 | merlin8282 | nah, forget it, sorry |
| 10:24.10 | *** part/#asterisk f2knight (~ben@c-76-115-44-207.hsd1.or.comcast.net) |
| 10:25.58 | merlin8282 | So another question: what is the best place to report a "bug" (not really one) ? When doing completion on "queue " in the console, the "show" parameter is given twice. |
| 10:26.12 | merlin8282 | *CLI> queue |
| 10:26.13 | merlin8282 | add pause reload remove reset set show unpause show |
| 10:26.50 | *** part/#asterisk Smiley (~Smiley@last.fm/resident-dj/Smiley) |
| 10:31.14 | ollii | http://www.google.de/search?q=asterisk%20bug&num=50 |
| 10:33.25 | kaldemar | merlin8282: don't bother, that does not occur in 1.8.6.0. |
| 10:34.16 | merlin8282 | kaldemar: mmm : Asterisk 1.8.6.0-1digium1~squeeze built by pbuilder @ nighthawk on a x86_64 running Linux on 2011-08-31 21:25:15 UTC |
| 10:36.16 | kaldemar | i have a vanilla version built from source. |
| 10:36.57 | merlin8282 | ok, so maybe the problem is related to the packager, I suppose. |
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| 11:50.02 | leifmadsen | f2knight: one of the many reasons I avoid AEL |
| 11:52.26 | irroot | new cabinet been built here in office its pretty sweet data centre in a box |
| 11:53.01 | Dovid | iroot: Whats the jira ID to your T.38 patch ? |
| 11:53.35 | irroot | JIRA bug closed |
| 11:54.12 | irroot | http://svnview.digium.com/svn/asterisk/team/irroot/patches/t38gateway-1.8.patch?view=log |
| 11:59.30 | Dovid | irrot: Is it now included in >= 1.5 |
| 11:59.37 | Dovid | oops >= 1.8 |
| 12:00.23 | *** join/#asterisk oej (~olle@static-213-115-251-100.sme.bredbandsbolaget.se) |
| 12:03.33 | *** join/#asterisk nunne (~nunne@static-213-115-116-75.sme.bredbandsbolaget.se) |
| 12:03.40 | irroot | Dovid yes 10 has it |
| 12:04.56 | nunne | far end nat on a cisco 800-series.. anyone have any experience? i'm using qualify=yes and canreinvite=no, but they still cant call each other it seems (internally). i have disabled SIP ALG |
| 12:13.35 | *** join/#asterisk ollii (~risker@2001:470:1f15:1384:f27b:cbff:fe9d:f3e4) |
| 12:17.21 | Rufus | I need to set same sip username to be able to conenct from 2 locations: one from office, and a second one from say laptop @ home. The home connection needs to overrule the office connection ( meaning if home connection is active, office phone shouldn't be able to connect ). is this possible ? |
| 12:24.32 | *** join/#asterisk DND (~ctcarino@94.200.7.26) |
| 12:24.39 | DND | hi guys i wanted to know if there is a much simple cdr reporting tool that has call cost included? i dont like a2billing since its too complicated |
| 12:26.00 | *** join/#asterisk stix_ (~stix@193.89.191.209) |
| 12:39.07 | *** join/#asterisk MariusAgon (~MariusAgo@79.142.116.89) |
| 12:39.56 | *** join/#asterisk azv4 (~azv4@host-66-202-7-218.pit.choiceone.net) |
| 12:44.08 | MariusAgon | Hello, does Asterisk has a module to control, what data is going to asterisk console. I don't know whats going on now, becouse now I just can see agi.verbose outputs, warnings and notices. But I can't see, how the call runs through my dialplan. What can be wrong right here? Thank you. |
| 12:44.32 | wdoekes2 | Rufus: why can't you use 2 account and dial both? i.e. rufus-office and rufus-home, and when someone dials in, you Dial(SIP/rufus-office&SIP/rufus-home) |
| 12:44.40 | kaldemar | MariusAgon: "core set verbose 10" |
| 12:45.08 | wdoekes2 | or, alternatively, if you don't want rufus-office to ring when rufus-home is up, you can do 2 consecutive Dials and check for the former being "up" |
| 12:46.11 | MariusAgon | kaldemar: thank you very much! |
| 12:46.15 | Rufus | so I can check if home is up, and I dial office, to get the call forwarded to home ? |
| 12:46.35 | wdoekes2 | nunne: detailed sip debug logs help. if the devices work properly, it should work |
| 12:47.41 | wdoekes2 | Rufus: Dial(SIP/rufus-home) ExecIf($[${DIALSTATUS}=CONGESTION]?Dial(SIP/rufus-office) |
| 12:48.03 | wdoekes2 | something like that.. add an extra ) to the second statement |
| 12:48.21 | Rufus | think I get the picture. will try. thanks |
| 12:56.49 | *** join/#asterisk coppice (~chatzilla@116.92.16.50) |
| 13:00.30 | leifmadsen | MariusAgon: also you can use Verbose() to control your own messages that go to the console. If you use Verbose(1,This is my message) or Verbose(2,This is my message), then you can do 'core set verbose 1' |
| 13:00.50 | leifmadsen | (or 2) and get only your own messages (plus whatever else you have setup in logger.conf) |
| 13:05.22 | *** join/#asterisk serafie (~erin@nat/digium/x-ijntmnkpwrpxqfjq) |
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| 13:08.26 | *** join/#asterisk skrusty (~skrusty@83.166.171.74) |
| 13:09.26 | _ramaskez_ | Hi All. I was wondering if its possible to write applications and functions for asterisk 1.8 in C++? and if so is there anywhere in the documentation where this is covered? |
| 13:10.07 | *** join/#asterisk ollii (~risker@port-87-193-161-154.static.qsc.de) |
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| 13:13.57 | OldSmurf | I am using Asterisk 1.6.2.9-2ubuntu2.1 and trying to setup a realtime ODBC connection. The connection works and I can see it established in the server, but in asterisk I get "config.c:2014 find_engine: Realtime mapping for 'meetme' found to engine 'odbc', but the engine is not available". Anyone got any clues? |
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| 13:22.45 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 13:23.32 | atan | Anyone aware of a "Print to SIP fax" type software for Windows? |
| 13:25.23 | *** join/#asterisk jplank (~G_Bove@208-104-67-27.dyn.fttp.comporium.net) |
| 13:25.38 | atan | Also, is there a dial in number for 811 services? |
| 13:27.07 | *** join/#asterisk treborsux (~treborsux@75-144-117-117-Jacksonville.hfc.comcastbusiness.net) |
| 13:27.57 | Sylnai | atan: if you are using hylafax, there are a lot of clients which can talk to it |
| 13:28.14 | atan | I have no sweet clue what I am using to be honest. |
| 13:28.36 | atan | I could setup an ATA and connect an actual fax machine to it I suppose |
| 13:28.40 | leifmadsen | I've used PRI --> IAX2 --> hylafax on a couple of systems, and it works quite well |
| 13:29.15 | Sylnai | hylafax can expose itself as a shared samba printer iirc |
| 13:29.18 | leifmadsen | atan: as long as your incoming call isn't SIP, and you're using ulaw/alaw, that should work in most cases |
| 13:29.34 | atan | leifmadsen, sorry? What's the problem with SIP? |
| 13:29.48 | leifmadsen | Sylnai: ya I've been meaning to look at more of the extra features that hylafax can provide. So far I've only used it as a fax-to-email gateway. |
| 13:29.57 | *** join/#asterisk iulhk (carryoff@117.102.32.126) |
| 13:30.14 | leifmadsen | atan: you can't fax over the internet using SIP like that -- fax is HIGHLY sensitive to latency, jitter, and packet loss, all of which are prevalent on the internet |
| 13:30.36 | florz | no, fax is not sensitive to latency |
| 13:30.38 | Sylnai | actually, T38 isn't that terrible |
| 13:30.59 | florz | at least not HIGHLY ;-) |
| 13:31.09 | leifmadsen | florz: generalizations suck :) |
| 13:31.14 | atan | I suppose I will just scan + e-mail it for now :D will need to look in to it more soon though |
| 13:31.20 | leifmadsen | anyways, faxing over SIP across the internet is going to fail more often than you want |
| 13:31.44 | atan | Okay. Sounds good. |
| 13:31.47 | leifmadsen | Sylnai: ya I wasn't accounting for T.38 etc. things |
| 13:31.59 | florz | you cannot fax over SIP as SIP is a signalling protocol only (I have the feeling you should actually know that ... =:-) |
| 13:32.30 | leifmadsen | heh, ok RTP if you wish to be pedantic ) |
| 13:32.30 | leifmadsen | :) |
| 13:32.34 | leifmadsen | (as we probably should be) |
| 13:32.42 | leifmadsen | I have barely finished 1 cup of coffee though |
| 13:32.42 | florz | well, we should be, as t.38 is SIP, too |
| 13:32.56 | atan | is going to die with all the techy terms |
| 13:33.00 | florz | as in: can be used with SIP as the signalling protocol |
| 13:35.56 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 13:41.47 | iulhk | how could i use this extension " exten => _X.,n,VoiceMail(${EXTEN}@${EXTEN}) " in AGI will anybody guide, i am trying this way $AGI->exec(VoiceMail $phone); but failed , will anybody guide pls ? |
| 13:49.02 | *** join/#asterisk markitoxs (~markitoxs@fluency-gw.factotum.co.uk) |
| 13:49.23 | kaldemar | iulhk: what do you expect $phone's value to be? |
| 13:50.28 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 13:50.31 | markitoxs | hi, I was wondering how would you guys would debug a call that has been reported as dead? |
| 13:51.37 | DND | autopsy? |
| 13:51.43 | iulhk | <kaldemar>: dialed number |
| 13:51.43 | markitoxs | lol |
| 13:51.47 | Sylnai | capture the SIP traffic and analyse it in wireshark? |
| 13:51.50 | *** join/#asterisk Laurent_N (~chatzilla@LMontsouris-156-25-42-38.w80-13.abo.wanadoo.fr) |
| 13:51.52 | kaldemar | iulhk: and what library are you using? |
| 13:52.01 | Laurent_N | I everybody |
| 13:52.04 | markitoxs | But im referring to a call that has been reported dead, as in like an hour ago |
| 13:52.19 | markitoxs | so, that would require me to be constantly capturing traffic |
| 13:52.34 | Laurent_N | Can I ask a question about g729 codec outbound call problem ? |
| 13:52.36 | iulhk | <kaldemar>: perl |
| 13:52.46 | Sylnai | if you have a busy enough network, it may be worth caching a day worth of SIP traffic on a dedicated monitoring box |
| 13:53.09 | markitoxs | Sylnai, that might be a good idea I guess |
| 13:53.26 | markitoxs | Its quite a busy pbx, so it gets quite hard to exactly determine what happened to a call |
| 13:53.50 | markitoxs | I do log the dialplan execution, so I can always trace into /var/log/asterisk/messages |
| 13:53.58 | markitoxs | but usuaslly all i see is a hangup |
| 13:54.14 | markitoxs | and a spawn extension exited on non 1 status |
| 13:54.22 | kaldemar | iulhk: the syntax is $AGI->exec($app, $options) right? |
| 13:54.36 | Laurent_N | I get 5 channels licenses for g729 codec and they are functional on sip calls, but I can only make one call to a DAHDI channel. the Other ones get the error : No translator path exists for channel type DAHDI (native 0x4c) to 0x100 |
| 13:54.47 | markitoxs | Is there anyway sip set debug ip X.X.X.X will be saved to a specific file? |
| 13:55.09 | kaldemar | markitoxs: script, tee... |
| 13:55.18 | Laurent_N | did I miss something ? |
| 13:55.29 | iulhk | <kaldemar>: ok thanx , now i hv to find options |
| 13:55.42 | markitoxs | kaldemar, what you mean? |
| 13:56.19 | kaldemar | markitoxs: use one of those command line applications for example to save output to a file. |
| 13:57.40 | kaldemar | iulhk: options are the options for the VoiceMail application. which in your case would be ${EXTEN}@${EXTEN}. you were just missing a comma in the exec(). |
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| 13:57.50 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:57.54 | markitoxs | kaldemar, the problem is that that would require a session to be running with sip debug on |
| 13:58.10 | markitoxs | and if anyone requires access to asterisk console, they would be flooded with sip traces |
| 13:58.23 | markitoxs | ideally i'd like something similar to what you can do in logger.conf |
| 13:58.35 | *** join/#asterisk nix8n82-phone (~AndChat@65.161.180.230) |
| 13:58.45 | markitoxs | but i dont know if that is possible |
| 13:58.51 | Kobaz | really the tracing options need to be session specific |
| 13:58.57 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 13:58.57 | Kobaz | but right now the console session is one fd |
| 13:59.07 | Kobaz | and it's shared between all attachments |
| 13:59.38 | markitoxs | i find it really hard to come to the bottom of dropped calls |
| 13:59.50 | markitoxs | because obviouslly i would require to have that traffic captured |
| 13:59.51 | kaldemar | Kobaz: wasn't someone already working on that a while ago? |
| 13:59.57 | Kobaz | probably |
| 14:00.19 | markitoxs | i guess the best option really is what Sylnai suggested, to capture the sip traffic on a dedicated box |
| 14:00.26 | Kobaz | for finding which side hung up, i've found that hooking into the channel hangup functions is great |
| 14:01.06 | *** join/#asterisk bchia (~Adium@nat/digium/x-sxdeeuqkicandpel) |
| 14:01.30 | markitoxs | Kobaz, but how do you get then to recall the channel id for a call that was in the past? |
| 14:01.47 | Kobaz | you'll have to log that data |
| 14:02.15 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 14:02.24 | markitoxs | i see |
| 14:02.30 | Kobaz | add logging of whatever you want to sip_hangup() |
| 14:02.36 | Kobaz | in channels/chan_sip.c |
| 14:03.25 | Kobaz | the best way to really get what you want out of asterisk, is to dive right in and monkey with the code |
| 14:03.36 | Kobaz | and feel free to submit patches :) |
| 14:03.47 | _ramaskez_ | Hi All. I was wondering if its possible to write applications and functions for asterisk 1.8 in C++? and if so is there anywhere in the documentation where this is covered? I can see sections of code that do an IF based on cplusplus but Im not sure how to get my own app to compile as C++? |
| 14:03.47 | Kobaz | then the rest of us get to use the nifty new features |
| 14:04.58 | Kobaz | _ramaskez_: you'll have to basically write all your module registration stuff in c which calls your c++ stuff |
| 14:05.02 | Qwell | _ramaskez_: I don't think so, no. You can link against C++ stuff, but.. |
| 14:05.04 | markitoxs | Kobaz, reast reassured, will do if i get the chance |
| 14:05.12 | Qwell | see chan_h323 for examples of how it's done |
| 14:05.16 | markitoxs | what you guys think of freeswitch? |
| 14:05.24 | Kobaz | markitoxs: no unit testing |
| 14:05.55 | Kobaz | markitoxs: at least the last time i looked into it |
| 14:05.55 | _ramaskez_ | Kobaz, Qwell: Thank you |
| 14:07.23 | Kobaz | _ramaskez_: there's the specific helpers for registering things like dialplan apps and functions, manager commands etc... it'll be the easiest if you have a small stub of c functions that are hooks into your c++ stuff |
| 14:07.33 | Kobaz | and your linker will take care of linking the c and c++ |
| 14:09.19 | Kobaz | markitoxs: i think there's a much bigger active asterisk community than freeswitch |
| 14:09.49 | markitoxs | Kobaz, yes, i feel the same, but freeswitch seems to be... different |
| 14:10.20 | Kobaz | it's very different |
| 14:10.26 | markitoxs | so I think it will be worth logging network traffic |
| 14:10.34 | Kobaz | lots of xml, and lots of modularity |
| 14:10.52 | Kobaz | like it's cool they support several languages for dialplan |
| 14:10.54 | _ramaskez_ | So have a an app_{NAME}.c that does the normal loading and unloading of the application but have some called functions from the main app_exec( that use C++? |
| 14:11.27 | Kobaz | but in the practical sense, it's not a huge difference whether you have native language support for dialplan versus using something like agi |
| 14:12.05 | Kobaz | it's a bit less efficient to run agi versus native, but unless you're running very high usage systems you won't really notice |
| 14:12.37 | Kobaz | _ramaskez_: that would be my approach |
| 14:15.01 | atan | Does Canadian 811 have a regular dial in number? |
| 14:15.10 | *** join/#asterisk navaismo (~IceChat77@201.123.65.40) |
| 14:15.17 | atan | My VOIP provider doesn't play nice when "811" it seems |
| 14:15.20 | Kobaz | atan: there's the non-emergency numbers |
| 14:15.42 | Kobaz | atan: there's also local municipality emergency numbers, which should be in your phone book |
| 14:16.07 | atan | Yes but like, for 911 I have some 847 number or something that's marked as Northern 911 Operator... |
| 14:16.21 | atan | 911 calls forward there, but I don't know what the heck to do with 811 |
| 14:16.23 | *** join/#asterisk Elit3 (~Elit3@41.35.198.133) |
| 14:16.35 | Kobaz | ohm what's 811 do? |
| 14:16.37 | atan | I also don't own a local phone book. I figured the number would be a little better known |
| 14:16.41 | atan | 811 in Canada calls a nurse |
| 14:16.45 | atan | It's non-emergency though. |
| 14:16.57 | Elit3 | hey any one know a voip provider which allow CID manipulation ? |
| 14:17.03 | Kobaz | Elit3: most do |
| 14:17.23 | Elit3 | i tried buster but cant do.. |
| 14:17.32 | Laurent_N | anybody for my g729 problem ? |
| 14:18.16 | Elit3 | any suggestions? |
| 14:20.15 | Kobaz | i'm here, you don't have to message me |
| 14:20.25 | Kobaz | ~itsplist-usa |
| 14:20.29 | Elit3 | np |
| 14:20.47 | Kobaz | i know voicepulse does, vitelity |
| 14:20.53 | Kobaz | where's the bot |
| 14:20.56 | Kobaz | ~itsplist |
| 14:21.49 | *** join/#asterisk timahvo1 (~rogue@41.223.57.73) |
| 14:22.56 | atan | Okay so I'm wrong. 811 is Nova Scotia only I guess and no direct number yet. |
| 14:22.58 | atan | Hmm. Oh well. |
| 14:24.56 | markitoxs | Kobaz, one of the things we use FREESWITCH for is to have a dialplan driven via a database, and its very simple to do using mod_curl_xml |
| 14:25.02 | Elit3 | thx kobaz |
| 14:25.58 | markitoxs | it was so simple and straigh forward to set up |
| 14:26.28 | Elit3 | how to link my inbound route to my trunk? i configured my did but it cant reach my extension.. any suggestions? |
| 14:26.28 | Kobaz | markitoxs: you can do the same thing with asterisk |
| 14:26.35 | markitoxs | that easy ? |
| 14:26.48 | Elit3 | how to link my inbound route to my trunk? i configured my did but it cant reach my extension.. any suggestions? NB : i use elastix |
| 14:26.52 | markitoxs | i mean, is it via AGI? |
| 14:26.55 | Kobaz | the setup is a few lines in a config file |
| 14:27.20 | Kobaz | actually moving your dialplan to the db is of course dependent on your dialplan |
| 14:27.39 | Kobaz | agi is for having an external application control the dialplan |
| 14:27.45 | Elit3 | any tutortial to look at? |
| 14:28.03 | Kobaz | offhand, i don't know something specific, but check out the book |
| 14:28.26 | Kobaz | http://astbook.asteriskdocs.org/ |
| 14:29.23 | Kobaz | and theres some stuff on the voip wiki |
| 14:29.25 | Kobaz | http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions |
| 14:29.48 | *** join/#asterisk mort_gib (~mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 14:29.51 | *** join/#asterisk Luyt (~Luyt@gandhi.xs4all.nl) |
| 14:29.59 | leifmadsen | Kobaz: better off with the ofps link for the new version |
| 14:30.06 | leifmadsen | ~thebook |
| 14:30.07 | 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 |
| 14:30.30 | atan | Can you make a 'test' 911 call to confirm your dialplan + VOIP provider are working as expected? |
| 14:30.58 | Kobaz | oh, the bot was ignoring me |
| 14:31.07 | Luyt | Thanks for all the fish! ;-) I just installed Asterisk and I'm investigating whether I can use it for my use case, which I will describe shortly: |
| 14:31.15 | leifmadsen | atan: you have to schedule that with your 911 location in most cases -- you can't just call 911 randomly |
| 14:31.29 | Kobaz | leifmadsen: we do that all the time |
| 14:31.36 | atan | leifmadsen: who would I contact to schedule this? |
| 14:31.38 | Kobaz | leifmadsen: in the us, don't know about CA |
| 14:31.38 | leifmadsen | Kobaz: it depends on the area -- you can't do that in Toronto |
| 14:31.52 | leifmadsen | atan: usually the local police station or something like that |
| 14:32.06 | Kobaz | new system install, set up the routes, dial 911. say "hi this is a new phone system, we're testing 911, what address do you have on file?.... okay thanks" |
| 14:32.19 | atan | Now my voip provider is voip.ms but it uses a server in New York. Would that route the 911 call incorrectly? |
| 14:32.25 | leifmadsen | Kobaz: yep but not all areas are allowed to do that, so I never recommend someone to "just try it" |
| 14:32.33 | Kobaz | ah |
| 14:32.42 | leifmadsen | atan: it's all based on the PSAP information on file |
| 14:32.51 | atan | What is none is provided? :S |
| 14:32.58 | leifmadsen | then they get no information about your location |
| 14:33.12 | leifmadsen | you need to fill in a web form usually for each of the DIDs on file |
| 14:33.20 | *** part/#asterisk markitoxs (~markitoxs@fluency-gw.factotum.co.uk) |
| 14:33.22 | atan | ...and so I'm trying to figure out if they can route the call |
| 14:33.27 | leifmadsen | then that data is pushed up via SOAP to the PSAP database |
| 14:33.35 | leifmadsen | what do you mean route the call? |
| 14:33.38 | leifmadsen | route it where? |
| 14:33.52 | leifmadsen | if you don't have data on file for your address, it just goes to an operator |
| 14:33.52 | *** part/#asterisk sekil (~sekil@78.24.104.73) |
| 14:33.56 | atan | Well the voip numbers have no e911 information setup on them. If a user dials 911 I'm triyng to figure out what happens |
| 14:34.05 | Kobaz | without address information the 911 call tends to go to the regional operator for the area code |
| 14:34.05 | leifmadsen | just goes to an operator |
| 14:34.10 | Kobaz | which might be 100 miles away |
| 14:34.14 | leifmadsen | exactly |
| 14:34.16 | atan | And the operator redirects it to the correct location? |
| 14:34.21 | leifmadsen | how? |
| 14:34.32 | leifmadsen | what is the "correct location"? |
| 14:34.34 | Kobaz | after you waste 30 seconds telling them your address |
| 14:34.41 | Kobaz | they can forward the call |
| 14:34.41 | leifmadsen | assuming you can talk |
| 14:35.07 | Luyt | I now have 5 phones with 5 voip accounts, every phone connects to its own account at a provider somewhere. I want to have them connect to Asterisk instead, and still let them dial out (and receive incoming calls) from their respective accounts. Then I want to use the Manager Interface to track all the calls that take place on the 5 phones. Can this be done with Asterisk? |
| 14:35.23 | atan | Okay, thanks guys :-) |
| 14:35.26 | Kobaz | Luyt: yeap |
| 14:35.32 | Luyt | Kobaz: That's good news! |
| 14:35.56 | Kobaz | you want to have certain scripts fire off when people place calls? |
| 14:35.59 | Kobaz | or do you just want a history? |
| 14:36.17 | Luyt | Kobaz: Just a history, I'll use it for a popup thingy in our CRM app |
| 14:36.29 | atan | Luyt, I think you want the CDR |
| 14:36.54 | Luyt | Kobaz: The provider already does intricate things like letting multiple phones ring, fallback to other numbers etc... |
| 14:37.01 | Luyt | atan: But is CDR realtime? |
| 14:37.07 | Kobaz | semi-realtime |
| 14:37.12 | Kobaz | the record is written once the call hangs up |
| 14:37.19 | Luyt | Ah, that's too late. |
| 14:37.25 | leifmadsen | then ya, you'll need to use the AMI |
| 14:37.28 | atan | If you want account pops on a screen when the call starts I'm sticking around to hear how you are going to do it! :) |
| 14:37.30 | Kobaz | or dialplan |
| 14:37.32 | Luyt | As soon as a phone rings, the app should be notified. |
| 14:37.40 | leifmadsen | Kobaz: oh ya, you could do that via res_jabber :) |
| 14:37.42 | atan | There is exec() I think. |
| 14:37.44 | leifmadsen | poor mans screen pop |
| 14:37.51 | Kobaz | ODBC..(INSERT INTO FOO)... |
| 14:37.56 | Kobaz | Dial(SIP/123) |
| 14:38.02 | leifmadsen | atan: Exec() isn't what he wants -- I think you mean System() or SHELL() |
| 14:38.10 | Kobaz | 'h' exten => update foo set hangup... |
| 14:38.20 | Luyt | Well, I was already able to use 'nc' to login the AMI and see these call setup packets fly by |
| 14:38.38 | mort_gib | Hi, anyone fancy helping with custom dialtones on polycoms? |
| 14:38.59 | Luyt | So the instances of the CRM app will just open a TCP connection to Asterisk and sniff those call setup events. |
| 14:39.04 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 14:39.48 | Kobaz | Luyt: you could do it that way |
| 14:40.05 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 14:40.07 | Luyt | The only thing I have to find out is how to configure these 5 voip accounts into Asterisk and do it in such a way that nor the phones, nor the voip provider sees any change. |
| 14:40.21 | Luyt | I'll be lynched if I break the telephone system ;) |
| 14:40.39 | Kobaz | i doubt the voip provider will care |
| 14:41.02 | Kobaz | but definitely do your playing on a test system or on the weekends if you dont want to break anything |
| 14:41.07 | Luyt | Probably not, but if my setups breaks some existing functionality, I get angry users in my neck |
| 14:41.25 | Luyt | Yeah ;-) I have a nice setup at home |
| 14:42.00 | Luyt | I even installed a few network cards into a BSD machine so I'll be able to bridge and inspect every packet that flies on the wire, but that's probably not necessary now I have discovered the AMI |
| 14:42.35 | Luyt | My first idea was to sniff the port 5060 SIP UDP traffic, but that is fraught with problems |
| 14:42.52 | Luyt | Requires a switch with monitor port etc... |
| 14:43.12 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 14:46.32 | *** join/#asterisk Fritz09 (~Adium@pop1-3802.catv.wtnet.de) |
| 14:50.23 | Kobaz | there's always an easier way |
| 14:50.41 | Luyt | You have an idea? |
| 14:51.03 | Kobaz | oh i was agreeing with you that AMI is the way |
| 14:51.19 | Luyt | Ah yes, much more elegant. And I run the Asterisk somewhere in the cloud. |
| 14:51.44 | Kobaz | you'll have varying success with cloud voip |
| 14:52.23 | Luyt | Well, it only has to do this event logging, nothing else. |
| 14:53.48 | Kobaz | are you going to use reinvite? |
| 14:53.49 | Luyt | Or do you think it'll also start transcoding etc? |
| 14:53.59 | Kobaz | otherwise all your data is making an extra hop |
| 14:54.26 | Kobaz | and in the new asterisk 1.8, it's called directmedia |
| 14:54.33 | mort_gib | Luyt -You are asking for problems if you try to do that |
| 14:54.48 | Luyt | Kobaz: I'm not yet very well read into SIP terminology etc... |
| 14:55.06 | Luyt | mort_gib: You mean, running Asterisk in the cloud, or my whole idea? |
| 14:55.29 | mort_gib | I mean you need to implement the full system locally |
| 14:55.44 | Kobaz | in general cloud voip is pretty terrible |
| 14:55.55 | mort_gib | I have heard bad stuff about clouds and VOIP |
| 14:56.27 | mort_gib | But that apart you need to use your system, not just as an "event" collector |
| 14:56.28 | Luyt | Lets see.... I could also use a colocated data center server (a machine all for my own!) |
| 14:56.37 | *** join/#asterisk Igneous (~igneous@208.86.215.38) |
| 14:56.46 | mort_gib | Or a plugpc |
| 15:02.34 | *** join/#asterisk JustinCampbell (~justinCam@135.10.98.3) |
| 15:02.35 | Luyt | Hmmm, I should also investigate SIP proxies (which offer something similar to the AMI). |
| 15:02.56 | JustinCampbell | hello all |
| 15:03.02 | JustinCampbell | im having an issue with an inbound sip trunk |
| 15:03.10 | JustinCampbell | from a Nortel CS1K |
| 15:03.18 | mort_gib | Luyt -What you are trying to do is not that difficult at all |
| 15:03.31 | JustinCampbell | my asterisk installation plays inbound ringing audio, and the CS1K thinks the call has been picked up |
| 15:03.39 | JustinCampbell | so hunt groups dont work on their side |
| 15:03.51 | Luyt | mort_gib: That's good news, but since I'm new to this stuff I have no idea where to start (well, installing Asterisk was not that hard ;-) |
| 15:04.01 | JustinCampbell | ive tried setting progressinband no, never, and yes globally but it doesnt seem to change |
| 15:04.11 | *** join/#asterisk jblack (~jblack@pool-71-173-14-99.sctnpa.east.verizon.net) |
| 15:04.11 | mort_gib | No :-) But getting people happy is |
| 15:04.24 | mort_gib | And that that's not restricted to phone systems :-) |
| 15:04.53 | mort_gib | <JustinCampbell> -Do you Answer() or Dial() |
| 15:05.10 | JustinCampbell | mort_gib: good point |
| 15:05.15 | Luyt | mort_gib: Hence my very first question, whether I was on the right track using Aterisk for my problem in the first place |
| 15:05.18 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:05.20 | JustinCampbell | im using adhearsion and i think is Answers() |
| 15:05.29 | dandre | Hello, |
| 15:05.55 | *** join/#asterisk jkroon (~jkroon@dsl-241-232-76.telkomadsl.co.za) |
| 15:06.04 | mort_gib | Luyt I use asterisk for a LOT of different setups, more than 300 users on different sites |
| 15:06.28 | Luyt | It is a very versatile system, I already noticed ;-) |
| 15:06.48 | mort_gib | <JustinCampbell> You need to use Dial() |
| 15:07.01 | JustinCampbell | mort_gib: makes sense |
| 15:07.05 | JustinCampbell | mort_gib: thanks ill play with that a bit |
| 15:07.17 | mort_gib | :-) np |
| 15:07.24 | Luyt | For example, when I dial a '5', Miles Davis starts to play over the phone. But now I'm investigating more serious things. |
| 15:07.50 | mort_gib | Luyt Good start though, unless it's some of his spaced out stuff |
| 15:08.18 | Luyt | "Dig". Although I also like the spaced out stuff like 'On the Corner'. |
| 15:08.22 | dandre | I have a configuration where I have defined a sip peer and I must include @ character in the dialstring. For instance: |
| 15:08.22 | dandre | DIAL(SIP/peer/foo@bar) where foo@bar must be passed to peer as the dialed string. The format for SIP channel doesn't seem to allow this as @ is used to indicate the host. |
| 15:08.29 | dandre | Is there any solution? |
| 15:08.33 | *** join/#asterisk gxdssoft (~gxdssoft@201.230.220.101) |
| 15:08.49 | ollii | ~thebook |
| 15:08.49 | 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 |
| 15:09.55 | Luyt | dandre: I think you can quote characters in strings using the backslash. |
| 15:10.04 | leifmadsen | yep |
| 15:10.24 | dandre | ok I will try |
| 15:10.29 | Luyt | Ah, the nuggets of wisdom one gets by randomly reading docs. |
| 15:10.39 | Luyt | Not knowing where to start or what's important... |
| 15:10.59 | Luyt | I did download that O'Reilly book, btw |
| 15:11.29 | jkroon | hi guys, managed to get a trap on a generic_http_callback failing on fdopen. Asterisk 1.8.5.0, it seems authentication passes, and then this happens. |
| 15:12.49 | jkroon | the particular manager call on which this usually SIPPeers, after which around half of my SIP peers are simply no longer available, a simple sip reload almost always fixes this. |
| 15:13.48 | jkroon | dandre, that's a URI so you might be able to %?? escape the @ |
| 15:14.07 | jkroon | eg, foo%40bar |
| 15:15.39 | dandre | jkroon: I will try as \@ doesn't work |
| 15:18.14 | leifmadsen | dandre: which version of Asterisk? |
| 15:19.32 | dandre | 1.4 |
| 15:20.22 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 15:20.30 | jkroon | ok, back to my issue, snippet of code is this: http://pastebin.com/QqJsyzpH where char []template = "/tmp/ast-http-XXXXXX"; |
| 15:20.59 | JustinCampbell | mort_gib: is it safe to answer multiple times? |
| 15:21.14 | mort_gib | no |
| 15:21.15 | mort_gib | no |
| 15:21.17 | mort_gib | no |
| 15:21.19 | mort_gib | :-) |
| 15:21.19 | JustinCampbell | :) |
| 15:21.24 | mort_gib | Why would you do that?? |
| 15:21.49 | JustinCampbell | well my thought was to answer before playing back anything |
| 15:21.54 | JustinCampbell | and otherwise use dial |
| 15:21.57 | mort_gib | You have to |
| 15:22.01 | JustinCampbell | right |
| 15:22.29 | JustinCampbell | im just wondering if the system happens to call answer twice, because another function somewhere already did |
| 15:22.39 | JustinCampbell | otherwise, i could check if it was answered before answering |
| 15:24.12 | JustinCampbell | mort_gib: another question, do i need to answer before adding sip headers? |
| 15:24.22 | JustinCampbell | guess ill find out if i try and it doesnt work ;) |
| 15:24.24 | mort_gib | I don't |
| 15:24.40 | JustinCampbell | mort_gib: thanks for your help! |
| 15:24.48 | mort_gib | That's how I set ringtones, to say Miles Davis tunes |
| 15:24.51 | mort_gib | :-) |
| 15:27.55 | mort_gib | Only I can't get the damn things to work on Polycoms!!! |
| 15:28.03 | mort_gib | I should stick to Snoms |
| 15:30.37 | Elit3 | hey |
| 15:30.41 | Elit3 | i have problem with my DID |
| 15:30.57 | Elit3 | Sep 7 18:23:39 VERBOSE [3112] pbx.c: |
| 15:30.57 | Elit3 | -- Executing [17772215448@from-sip-external:1] NoOp("SIP/66.193.176.35-0000000f", "Received incoming SIP connection from unknown peer to 17772215448") in new stack |
| 15:30.57 | Elit3 | Sep 7 18:23:39 VERBOSE [3112] pbx.c: |
| 15:30.57 | Elit3 | -- Executing [17772215448@from-sip-external:2] Set("SIP/66.193.176.35-0000000f", "DID=17772215448") in new stack |
| 15:30.57 | Elit3 | Sep 7 18:23:39 VERBOSE [3112] pbx.c: |
| 15:30.57 | Elit3 | -- Executing [17772215448@from-sip-external:3] Goto("SIP/66.193.176.35-0000000f", "s,1") in new stack |
| 15:30.57 | Elit3 | Sep 7 18:23:39 VERBOSE [3112] pbx.c: |
| 15:31.05 | WIMPy | ~pb |
| 15:31.05 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 15:31.32 | *** join/#asterisk Elit3 (~Elit3@41.35.198.133) |
| 15:31.54 | WIMPy | ~pb |
| 15:31.54 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 15:31.56 | Elit3 | Sep 7 18:23:59 WARNING [2532] chan_sip.c: |
| 15:31.56 | Elit3 | Maximum retries exceeded on transmission 4436091-3524397819-622242@msw1.telengy.net for seqno 1 (Critical Response) -- See doc/sip-retransmit.txt. |
| 15:31.56 | Elit3 | <PROTECTED> |
| 15:31.56 | Elit3 | <PROTECTED> |
| 15:32.03 | Luyt | I think even one line of gibberish is a candidate for a pastebin. |
| 15:32.05 | ollii | <3 |
| 15:32.50 | Elit3 | this my DID asterisk log |
| 15:32.51 | Elit3 | http://pastebin.com/iYHVdyPK |
| 15:33.41 | ollii | thats far away from readable :D |
| 15:33.47 | *** join/#asterisk garymc (~chatzilla@host81-139-71-64.in-addr.btopenworld.com) |
| 15:34.07 | Elit3 | well the call reach my asteirsk |
| 15:34.12 | Elit3 | but never reach my extension |
| 15:34.54 | WIMPy | Elit3: Fix your dialplan or preferrably the sip peer for your provider. |
| 15:34.58 | *** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net) |
| 15:36.06 | WIMPy | Or rather both. Accepting guest calls but not them call your DID doesn;t seem to make much sense. |
| 15:40.51 | *** join/#asterisk BuenGenio (~Gene@m180-219-73-69.smartone-vodafone.com) |
| 15:41.21 | Luyt | Heh, I'm streaming mp3 music to three phones at once, and Asterisk consumes only 0.20% CPU (each mpg123 process consumes 0.40%) |
| 15:41.34 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 15:45.00 | Qwell | mp3 isn't exactly demanding |
| 15:45.24 | Luyt | What would be demanding then? |
| 15:45.50 | Qwell | video transcoding |
| 15:46.07 | Luyt | Ah, luckily I don't have any video phones ;-) |
| 15:56.08 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 16:04.09 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 16:08.17 | Kobaz | hmm |
| 16:08.24 | Kobaz | why i never noticed this, i don't know |
| 16:08.41 | Kobaz | but it seems you can record a temporary greeting, but there are no voicemail menu options to get rid of it |
| 16:08.46 | Kobaz | unless i'm missing something |
| 16:09.09 | WIMPy | Record nothing? |
| 16:09.31 | Kobaz | seems like a bad way |
| 16:09.40 | WIMPy | Not very user friendly, I guess. |
| 16:09.41 | Kobaz | then you don't get the persons recorded name |
| 16:10.44 | Kobaz | hmm looking at the source there are options |
| 16:10.52 | Kobaz | hacks |
| 16:11.02 | WIMPy | Doesn't it get deleted if it's too short? Or you dump it on review? |
| 16:11.14 | *** join/#asterisk Subdolus (dexterity@203.82.208.13) |
| 16:11.56 | Kobaz | that's for leaving a new voicemail |
| 16:12.02 | Kobaz | minsecs |
| 16:12.21 | Kobaz | ooh, it doesn't update the flag until you log back in |
| 16:12.32 | Kobaz | okay it actually does have tracks about removing it |
| 16:12.39 | Kobaz | but you have to record it, hang up and then dial back |
| 16:12.53 | Kobaz | if you record it, you can't remove it in the same session |
| 16:14.54 | Kobaz | seems like a bug |
| 16:16.12 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 16:24.47 | WIMPy | Interesting. |
| 16:27.35 | bmoraca_work | has anyone here set up SLA on a pure SIP scenario? |
| 16:27.41 | *** join/#asterisk garymc (~chatzilla@host81-139-71-64.in-addr.btopenworld.com) |
| 16:30.16 | StaRetji | WIMPy: sorry for such delay mate |
| 16:30.38 | StaRetji | if some one can give me tip where should I place feature request, that would be nice |
| 16:30.40 | StaRetji | thx |
| 16:32.29 | *** join/#asterisk chasing`Sol (~cS@smtp.master-zone.net) |
| 16:40.28 | WIMPy | StaRetji: https://issues.asterisk.org/jira/ |
| 16:46.24 | tzafrir | http://www.creytiv.com/baresip.html - anybody familiar with it? |
| 16:48.21 | carrar | Welcome to Fresh Meat! |
| 16:53.27 | dwayne | StaRetji, I think your feature request is going to be closed if don't supply a patch with it |
| 16:54.02 | *** join/#asterisk luke-jr (~luke-jr@2001:470:5:265:222:4dff:fe50:4c49) |
| 16:54.22 | luke-jr | I'm trying to register telephony cards with Digium to get support, but their site rejects the serial numbers. Any ideas? |
| 16:54.36 | carrar | call them |
| 16:55.00 | Qwell | where did you get them? |
| 16:56.33 | luke-jr | full system was purchased from Tekniq Data, Corp. in 2008 |
| 16:57.20 | luke-jr | it has a TDM800P and TE205P |
| 16:58.29 | pabelanger | tzafrir: oej method it today on twitter, might want to talk with him |
| 16:58.35 | pabelanger | mentioned* |
| 16:59.39 | tzafrir | I saw it on voip-info. It looks potentially interesting. But they seem to use their own build system |
| 17:00.37 | *** join/#asterisk trelane` (trelane@funtoo/staff/trelane) |
| 17:03.54 | *** join/#asterisk Rholk (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 17:09.06 | coppice | tzafrir: That does indeed look interesting. It build without problems, and seems to run, so its build system passes its first key test |
| 17:18.39 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 17:21.13 | malcolmd | luke-jr: call support, they should be able to help determine the state of the cards |
| 17:31.05 | *** join/#asterisk BuenGenio (Gene@n219076092077.netvigator.com) |
| 17:37.37 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 17:45.54 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 17:50.48 | *** join/#asterisk madwill (~dude@teluq37.teluq.uquebec.ca) |
| 17:51.51 | madwill | anybody has experience with asterisk rtmp channel in order to bridge a web client to a conference room using h323 ? |
| 17:52.17 | wdoekes2 | off-topic: oej (and possibly others): why do you favor kamailio over opensips? |
| 17:52.28 | madwill | I wonder how big of a challenge it is and if i should hire a freelancer to get this going |
| 17:55.14 | wdoekes2 | drat.. asking questions to people who are offline |
| 17:56.06 | madwill | <PROTECTED> |
| 17:56.20 | *** join/#asterisk dlublink (~david@75-119-235-201.dsl.teksavvy.com) |
| 17:57.10 | Luyt | I try to make sense out of a SIP debug log, see http://codepad.org/94RZNDcl . When I call from my ancient home phone (PSTN through oldfashioned telcom) to my voip account telephone number, I get a 'busy' tone and the following log (see the paste). Can you see anything obviously wrong? Do you need more config info? Is it a NAT problem? |
| 17:58.05 | *** join/#asterisk eja (~user@75.110.195.31) |
| 18:00.05 | StaRetji | cyford: ping |
| 18:01.05 | dlublink | I am looking at RTPAUDIOQOS variable. I call into the dialplan using a SIP phone and call out with DIAL to another SIP phone. Which leg is RTPAUDIOQOS reporting on ? |
| 18:02.13 | StaRetji | WIMPy: https://issues.asterisk.org/jira/browse/ASTERISK-18454 |
| 18:12.54 | wdoekes2 | Luyt: do you have an 's' extension in the [default] context? |
| 18:13.31 | Luyt | wdoekes2: Lemme check... |
| 18:14.09 | Luyt | wdoekes2: Yes, http://codepad.org/0GqeXRtN |
| 18:14.24 | wdoekes2 | lol |
| 18:14.28 | wdoekes2 | ok, it works! |
| 18:15.25 | Luyt | wdoekes2: And having a 's' extension in the default context which just hangs up, is that good or bad? |
| 18:15.32 | wdoekes2 | you probably want to (1) set a context= in the sip.conf for your sipnl trunk and (2) either register=>..../exten, or accept calls on 's' |
| 18:15.43 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 18:16.22 | Luyt | wdoekes2: Would it help if I pasted my config files (with passwords obscured) ? |
| 18:16.25 | wdoekes2 | Luyt: the incoming call works.. sipnl is calling you with user-part 's' and all you're doing is Hangup() |
| 18:16.41 | wdoekes2 | what do you want to happen? |
| 18:16.42 | Luyt | wdoekes2: Excellent, there is now hope for me. |
| 18:16.49 | *** join/#asterisk tamiel (~tamiel@ip-183.net-81-220-86.toulouse.rev.numericable.fr) |
| 18:17.07 | Luyt | wdoekes2: I want to ring one of my SIP phones (I have 3 that register to the same asterisk instance) |
| 18:17.33 | wdoekes2 | [incoming] exten => s,1,Dial(SIP/mysipphone) |
| 18:17.47 | wdoekes2 | and set context=incoming on the [sipnl] trunk in sip.conf |
| 18:17.48 | StaRetji | p3nguin: ping |
| 18:17.51 | Luyt | wdoekes2: I'll try it right away |
| 18:17.59 | StaRetji | remember rdnix problem? |
| 18:18.03 | StaRetji | rdnis* |
| 18:18.43 | wdoekes2 | but even better is if you add /0501234567 to your register line.. then you get to do [incoming] exten => 0501234567,1,Dial(...) |
| 18:18.46 | Luyt | wdoekes2: I already have a 'context=incoming' in sip.conf under the [31308******] section. (*** to protect the innocent ;-) |
| 18:19.01 | raden | whats default pass for polycom phones ? |
| 18:19.10 | Naikrovek | Polycom/456 |
| 18:19.14 | Naikrovek | user/pass |
| 18:19.18 | *** join/#asterisk pdtpatrick_ (~pdtpatric@mainstwan.farheap.com) |
| 18:19.26 | wdoekes2 | in that case you're not matching the call to the sipnl trunk |
| 18:19.32 | wdoekes2 | and then it lands in default |
| 18:19.51 | pdtpatrick_ | Question .. how do u setup keys on 1.8 ? In the paste i've built from source and there's usually a folder that allows key creation without paraphrase using -n option |
| 18:20.06 | StaRetji | both operators said they are sending information, so, just wanted to keep you informed, thx |
| 18:20.16 | Luyt | wdoekes2: FYI, this is my dialplan: http://codepad.org/Ft0nuku7 |
| 18:20.49 | wdoekes2 | but the sip.conf? for some reason it's not matching the call => you're not going to the right context |
| 18:20.59 | kaldemar | pdtpatrick_: astgenkey |
| 18:21.05 | Luyt | wdoekes2: Rgr that, posting my sip.conf, moment... |
| 18:23.27 | Luyt | wdoekes2: My sip.conf at http://codepad.org/tBW5ENsd |
| 18:23.32 | Luyt | wdoekes2: Thanks for all the tips! |
| 18:24.11 | *** join/#asterisk CheBuzz (~CheBuzz@76.164.192.124) |
| 18:24.13 | wdoekes2 | peer matching is tricky for that host, as it has multiple A records |
| 18:24.28 | Luyt | four ;-) |
| 18:24.31 | wdoekes2 | you could copy the config 4 times (using config inheritance) and do one for every IP |
| 18:24.35 | *** join/#asterisk billmania (~bill@38.98.130.98) |
| 18:25.05 | CheBuzz | Can anybody tell me how to get debug output in 1.8? "core set debug 9" doesn't seem to do much. I believe it used to be "core set debug on" in 1.6 |
| 18:25.28 | wdoekes2 | or you could do user matching, but that only works if sipnl is able to provide auth |
| 18:26.28 | wdoekes2 | but I figure you can go with the easiest: [3130-tpl](!) .. most config .. [3130-a](3130-tpl) host=91.195.161.2 [3130-b](3130-tpl) host=91.195.161.3 |
| 18:26.41 | Luyt | wdoekes2: by 'copying the config' you mean creating four '[3130*******]' sections with each an other IP? |
| 18:26.48 | Luyt | Aha. |
| 18:27.09 | Luyt | Right. I should dive into config templating too, then ;-) |
| 18:27.35 | wdoekes2 | about 5 cm |
| 18:27.53 | Luyt | But first I'll try putting a Dial() in [default] |
| 18:28.11 | wdoekes2 | yes, but then I can dial you too >;) |
| 18:28.36 | pdtpatrick_ | Question .. i'm trying to load the keys in asterisk 1.8 .. i've done module reload res_crypto.so and then ran keys init ... however keys show - displays no keys. It says 0 known RSA keys |
| 18:28.49 | Luyt | wdoekes2: Dial me? Are you a nice person to speak to? ;-) |
| 18:29.21 | Luyt | So a Dial() in [default] is an invitation to receive all World Spam that exist? |
| 18:29.35 | wdoekes2 | yes, but you can allowguest=no too |
| 18:30.05 | Luyt | Well, I'm eager to get at least *something* working, so I'm trying it now ;-) |
| 18:30.11 | wdoekes2 | I believe there is a SECURITY-IMPORTANT-READ-THIS document in the root of the source somewhere |
| 18:30.25 | Luyt | I'll revert it to the Hangup() after my test. |
| 18:30.32 | *** join/#asterisk Defraz (~Defraz@70.36.76.167) |
| 18:30.34 | wdoekes2 | which should read that you're not supposed to do anything in default |
| 18:30.56 | Luyt | Except hanging up, perhaps? |
| 18:30.59 | wdoekes2 | :) |
| 18:31.13 | Luyt | I can remember reading security warnings, indeed |
| 18:31.30 | Luyt | Call fraud is just a matter of time, if your system is insecure. |
| 18:32.05 | Luyt | Apparently there are call shops out there that like voip accounts to make calls to obscure African countries for 9 euro per minute, all day long. |
| 18:32.39 | _Corey_ | Don't forget North Korea |
| 18:32.46 | Luyt | Waaaa! They too? |
| 18:32.50 | _Corey_ | Oh yeah |
| 18:33.20 | _Corey_ | I had a customer with a crap password who had like 6 channels lit up to Kim Jong Il |
| 18:33.41 | Luyt | I didn't know you could even call to North Korea. |
| 18:34.11 | raden | Naikrovek, u around ? |
| 18:34.12 | _Corey_ | Yeah, they had a cell network installed a few years ago actually |
| 18:34.22 | Naikrovek | raden: yeah |
| 18:34.27 | Naikrovek | wassup |
| 18:34.33 | raden | Naikrovek, have some 321's how do i get these things to config from tftp ? |
| 18:34.41 | raden | I'm missing something here |
| 18:35.08 | Naikrovek | raden: set it up in dhcp, would be the easiest way, but only if you have a linux or windows dhcp server. router dhcp servers won't help |
| 18:35.39 | Naikrovek | or you can set them up on the phones manually |
| 18:35.41 | raden | Naikrovek, ? |
| 18:35.48 | raden | im in the phone web config |
| 18:36.09 | Naikrovek | can't do it from the phone web config. gotta do it on the phone itself or via dhcp |
| 18:36.42 | raden | ok |
| 18:36.45 | Naikrovek | the part of the phone that loads config just bootstraps the part that has the webui in it |
| 18:36.46 | raden | looking in phone config |
| 18:37.24 | Naikrovek | reboot phone, and before it starts booting, there's a setup softkey. push that. enter 456. find the server menu and configure the tftp server location there |
| 18:37.58 | CheBuzz | Anybody? How do I get * 1.8 to show me all the debug information of a session? |
| 18:38.06 | Naikrovek | CheBuzz: sip set debug on |
| 18:38.07 | Naikrovek | ? |
| 18:38.31 | Naikrovek | CheBuzz: what kind of debug information are you looking for |
| 18:39.34 | CheBuzz | Ah, I had thought I tried that :S |
| 18:39.47 | CheBuzz | That's what I was looking for, but I know I had tried it and got nothing. |
| 18:41.47 | raden | Naikrovek, thanks bro !!! |
| 18:41.54 | raden | Naikrovek, also it can be done on cisco |
| 18:41.56 | Naikrovek | yeahbuddy |
| 18:42.04 | raden | Naikrovek, dhcpd option 66 |
| 18:42.08 | Naikrovek | CheBuzz: it'll only spit out data if there's something going on |
| 18:42.11 | Naikrovek | raden: yup |
| 18:42.28 | raden | :D |
| 18:42.34 | Naikrovek | raden: it'll be something like "tftp://192.168.123.123" without the quotes |
| 18:42.42 | Naikrovek | that's what you'll put in option 66 |
| 18:43.04 | Naikrovek | CheBuzz: no worries. learning to read that mess is your next challenge :) |
| 18:43.17 | CheBuzz | Oh, I'm familiar with it, a bit at least. |
| 18:43.22 | raden | where can i get a sample polycom script |
| 18:43.36 | CheBuzz | Need to find out why an ATA is not working like it should. |
| 18:43.44 | pdtpatrick_ | QUESTION .. my asterisk keys created by astgen is located in /var/lib/asterisk/keys .. however .. keys init and keys show .. do not load any of the keys .. can someone please help |
| 18:43.47 | luke-jr | sigh, Digium's solution is to RMA it :/ |
| 18:44.05 | Naikrovek | raden: the polycom SDK has one. |
| 18:44.32 | raden | k |
| 18:44.43 | kaldemar | raden: from polycom. download a firmware package and you'll find one there. also get an admin manual to know what the parameters are. |
| 18:44.53 | Naikrovek | ^^^ what he said |
| 18:44.59 | raden | Sounds good :) |
| 18:45.48 | CheBuzz | And the other odd thing I see is that I have an android phone registered with this PBX, and it sends |
| 18:46.04 | CheBuzz | an OPTIONS every 15 seconds or so. Any idea why? |
| 18:47.07 | Naikrovek | yeah |
| 18:47.19 | *** join/#asterisk m29poff (~m29poff@125.213.206.77.rev.sfr.net) |
| 18:47.33 | Naikrovek | it's basically keeping itself known to the host. or hte host is checking to make sure the phone is still there |
| 18:47.38 | kaldemar | CheBuzz: for keep-alive. |
| 18:47.40 | Naikrovek | useful for keeping NAT holes open |
| 18:47.45 | Naikrovek | kaldemar: well said |
| 18:47.48 | Naikrovek | why did i use so many words |
| 18:48.06 | CheBuzz | Ah, that's what I was wondering. But most keep alives are just tiny packets, right? |
| 18:48.13 | kaldemar | Naikrovek: you're not lazy. :) |
| 18:48.22 | Naikrovek | not enough, i guess |
| 18:48.31 | Naikrovek | though my fat ass would argue with you |
| 18:48.36 | Naikrovek | about the laziness |
| 18:48.50 | kaldemar | CheBuzz: OPTIONS is pretty much as tiny as a SIP message can be. |
| 18:50.02 | CheBuzz | Hm, ok. I guess I have just seen other devices send an empty message. But I could be remembering wrong, it's been a while. |
| 18:50.05 | *** join/#asterisk navaismo (~IceChat77@201.123.65.40) |
| 18:52.27 | wdoekes2 | CheBuzz: some devices send a couple of NULs only |
| 18:53.27 | CheBuzz | So I'm not crazy :) |
| 19:00.06 | Luyt | wdoekes2: Well, I'm finally getting a call from my mobile phone through now by putting 'exten => s,1,Dial(SIP/mac000e08dcdb6e)' in [default]. |
| 19:00.34 | Luyt | But I understand that is not desirable... |
| 19:02.57 | wdoekes2 | Luyt: look for this line "No matching peer for '050*******' from '91.195.161.2:5060' |
| 19:03.24 | wdoekes2 | as long as you see that, you're not matching the sip peer |
| 19:05.14 | Luyt | wdoekes2: I indeed see that message: No matching peer for '0620******' from '91.195.160.3:5060' (when I call with my mobile phone) |
| 19:05.56 | Luyt | But the peer has a 'context=incoming' config statement. |
| 19:06.26 | Luyt | and in my dialplan, [incoming] exten => s,1,Dial(SIP/mac000e08dcdb6e) |
| 19:06.56 | wdoekes2 | do you have 4 peers now? sip show peers |
| 19:07.11 | kaldemar | Luyt: you have no peer that matches 0620******. context setting in sip.conf and dialplan is irrelevant before a peer matches. |
| 19:07.47 | wdoekes2 | kaldemar: he's only supposed to match on IP. that 06 is the CLI |
| 19:08.26 | Luyt | wdoekes2: http://codepad.org/57WED6Gu <-- peers |
| 19:08.37 | Luyt | the 'mac' peers are local sip phones |
| 19:08.52 | wdoekes2 | grmbl.. where is the 160? |
| 19:09.09 | wdoekes2 | have you been a lazy boy and only added 2 of the 4 ips? |
| 19:09.35 | Luyt | wdoekes2: Probably: http://codepad.org/IGT5a8ys |
| 19:09.43 | Luyt | lemme 'dig' sipnl.net |
| 19:09.58 | Luyt | heh yes |
| 19:11.07 | Luyt | Amended: http://codepad.org/ZNrUQQTH |
| 19:12.06 | *** join/#asterisk m_tadeu (~quassel@89.181.12.32) |
| 19:13.03 | Luyt | Seeing more peers now: http://codepad.org/3SkYEz1Y |
| 19:13.07 | Luyt | brb toilet |
| 19:19.41 | Luyt | wdoekes2: Very nice, calling from both my mobile and PSTN phone to my asterisk and voip phone now works. |
| 19:20.10 | wdoekes2 | mooi zo :) |
| 19:20.11 | Luyt | I'm very thankful for all the tips I get on IRC. I think it's a great resource. |
| 19:20.59 | Luyt | nou ;-) Only dialing out from my voip phones (via asterisk) to my mobile (and PSTN) doesn't work yet but that might be an unrelated problem. |
| 19:21.49 | Luyt | [outgoing] exten => _X.,1,Dial(SIP/sipnl-a/${EXTEN}) ; Maybe that should be s,1,Dial(...) |
| 19:22.25 | *** join/#asterisk didnot (~didnot@unaffiliated/didnot) |
| 19:22.29 | Luyt | Ah, and the phones are in the context 'local', and [local] include => outgoing |
| 19:23.11 | Luyt | Hmmmpf, I also should read up about how Asterisk regular-expression-matches phone numbers with patterns like _X |
| 19:23.28 | wdoekes2 | your incoming and outgoing contexts overlap (or at least.. they did here http://codepad.org/Ft0nuku7 ) |
| 19:23.57 | Luyt | wdoekes2: Ah, because of the _X ? |
| 19:25.20 | wdoekes2 | either they should do A or B, not both. your config is telling to Dial your voip phones AND to Dial out.. one of those is going to get lost |
| 19:25.25 | wdoekes2 | try to include only outgoing in local |
| 19:25.36 | Luyt | ok |
| 19:25.50 | wdoekes2 | and add e.g. 200,1,Dial(SIP/mac000e08dcdb6e) |
| 19:25.56 | wdoekes2 | then everything, except 200 will go out |
| 19:27.32 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 19:30.04 | Luyt | allrighty... trying... |
| 19:30.21 | *** join/#asterisk titter (~Justin@c-76-101-15-40.hsd1.fl.comcast.net) |
| 19:31.14 | titter | I am completely pulling a blank today ... if I have two dynamic members in a queue, will leastrecent move on to the next member if the first hits the timeout? It seems to just repeat over and over on the member it tries first. |
| 19:31.32 | *** join/#asterisk nny (~SM@cpe-174-107-223-014.sc.res.rr.com) |
| 19:31.55 | wdoekes2 | this question was raised here earlier this week, titter |
| 19:32.09 | wdoekes2 | I believe leastrecent doesn't move to a next member |
| 19:32.32 | titter | rrmemory didn't seem to either. |
| 19:32.37 | nny | tweaking my default dialplan, looking for a wall or two to bounce ideas off of. I am looking ot generate hints on all registered phones without listing *each* phone,.. any thoughts welcome |
| 19:33.06 | nny | example: http://pastebin.com/h5YHPSmn |
| 19:33.53 | wdoekes2 | titter: I think rrmemory should work though, but that's just based on a quick glance at the source |
| 19:34.08 | treborsux | ivr seems to be working fine but i get some callers telliing me they type an extesion and it read them the extension and says invalid then they try it several more times and it goes through |
| 19:34.17 | *** join/#asterisk trumee (~trumee@cpc2-cmbg7-0-0-cust855.5-4.cable.virginmedia.com) |
| 19:36.20 | nny | right now just thinking I have to define which hints need to exist somewhere, since asterisk just can't dynamically generate hints for registered users. |
| 19:36.36 | nny | ooh dynamic hints in 1.6.1, off to read |
| 19:37.37 | nny | so i can just use a variable, that was easy |
| 19:38.28 | *** join/#asterisk screenn (~screenn@178.151.86.196) |
| 19:40.48 | treborsux | anyone? |
| 20:12.05 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 20:12.05 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.0.0-beta1 (2011/07/22), 1.8.6.0 (2011/08/31), dahdi-linux 2.5.0 (2011/08/08), dahdi-tools 2.5.0 (2011/08/08), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 20:12.46 | p3nguin | It's not what I would do, but not everyone does things in sane ways. |
| 20:12.55 | Kobaz | anyone have a link to a shoutcast stream i can use for testing |
| 20:13.02 | *** join/#asterisk ipc9 (~rob@173-162-245-206-NewEngland.hfc.comcastbusiness.net) |
| 20:13.20 | tilerendering | p3nguin: well I dont have a stationary computer here |
| 20:13.22 | tilerendering | only 2 laptops |
| 20:13.30 | tilerendering | and an iphone |
| 20:13.40 | tilerendering | oh well - maybe install it on the android tablet ;) |
| 20:13.45 | titter | Since I like the way leastrecent works, and it doesn't seem to roll through to the next member ... I kind of worked around it by doing this: http://pastebin.com/k5Bscruw --- however I have 2 issues, a.) the caller ID shows as Anonymous after it rings the first queue member, and goes to the next b.) when it hits the end of the dialplan, it plays back the busy message, but it doesn't hangup the call? |
| 20:14.01 | _Corey_ | Kobaz: http://www.radioparadise.com/musiclinks/rp_64.m3u |
| 20:14.12 | ipc9 | is there a way to make chan_alsa listen on more than one audio device? |
| 20:14.14 | p3nguin | kobaz: http://205.188.215.228:8006 |
| 20:15.09 | Kobaz | nifty |
| 20:15.10 | Kobaz | k |
| 20:16.08 | Kobaz | try just streaming and see if i get rtp |
| 20:18.21 | *** join/#asterisk garymc (~chatzilla@host109-155-155-5.range109-155.btcentralplus.com) |
| 20:25.31 | Kobaz | there's just no rtp |
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| 21:16.58 | lwizardl | hello |
| 21:20.08 | lwizardl | are the Unlocked Linksys Sipura & PAP2T-NA adapters valid for use with asterisk ? |
| 21:20.25 | p3nguin | Absolutely. |
| 21:20.38 | wdoekes2 | Luyt: your cli showed 050**** |
| 21:21.12 | Luyt | wdoekes2: hehe 050 is Groningen ;-) |
| 21:21.31 | wdoekes2 | no shit ;) |
| 21:21.31 | lwizardl | ok then i may just a couple of those |
| 21:21.58 | Luyt | wdoekes2: I do all kinds of stuff, see http://www.michielovertoom.com/ for my website ;-) |
| 21:22.28 | Luyt | Currently working with illustrators and also on a webcam-related side-project |
| 21:26.12 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-ipxvxdgftipeodoq) |
| 21:27.06 | bmoraca_work | if i have a static queue member which is a local channel (Local/line1@sla-inc and Local/line2@sla-inc), and I want them to ring as leastrecent without ringinuse, does the queue application maintain state of local channels such that the first local channel will not be rung if it is in use? |
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| 21:40.08 | Kobaz | queue won't know anything about local channels, since local channels are created on demand |
| 21:42.09 | bmoraca_work | well that's useless |
| 21:42.35 | bmoraca_work | really need devstate to do SLA properly, i guess |
| 21:43.13 | leifmadsen | bmoraca_work: unless you tell it to monitor a specific SIP end point |
| 21:43.36 | leifmadsen | you can tell Asterisk to use the state of a SIP end point as the state for the Local channel |
| 21:43.49 | leifmadsen | (since Local channels aren't real end points, it doesn't make sense to monitor them directly) |
| 21:46.14 | bmoraca_work | well, the problem is that there is no individual endpoint to monitor |
| 21:46.38 | leifmadsen | then what you're trying to do isn't going to work |
| 21:47.01 | bmoraca_work | well, it would work if the SLATrunk command exited the same way every time |
| 21:47.04 | leifmadsen | only SIP end points give "real device state" |
| 21:47.46 | bmoraca_work | right now, i'm using astdb to monitor the state of "line 1" and "line 2", etc...but if the CALLER hangs up, and not the CALLED extension, the dialplan after SLATrunk never executes to clear the state |
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| 21:48.31 | bmoraca_work | if the SLATrunk is picked up via SLAStation, everything works, but dialplan after SLATrunk() is never executed, and thus I cannot clear my state |
| 21:50.20 | bmoraca_work | if the called extension hangs up, or the SLATrunk isn't answered by SLAStation, it does seem to work properly (in that my dialplan after SLATrunk() is executed properly. in this case, i can use simple gotoifs to check astdb and see what lines are available |
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| 21:59.46 | p3nguin | Whoa. The entire Russian hockey team was apparently killed in a plane crash. |
| 22:04.27 | bmoraca_work | ah hah! i think i found a solution |
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| 22:08.58 | bmoraca_work | well, that was easy |
| 22:15.11 | bmoraca_work | a well placed "h" extension and another Hangup() and it's all good! |
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| 22:59.31 | Micc_ | anyone have much success running asterisk in xen para virtual or full virtual mode? |
| 22:59.58 | carrar | works great |
| 23:00.35 | carrar | assuming you don't mix the virtual stuff |
| 23:00.48 | carrar | like a none voip customer on it |
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| 23:34.42 | *** join/#asterisk kaushal (~kaushal@115.118.154.122) |
| 23:34.45 | kaushal | Hi |
| 23:35.07 | kaushal | Does Digium Card support 8 E1 ISDN ports ? |
| 23:35.24 | kaushal | I can see only max 4 ports as per http://store.digium.com/telephony_card_selector.php |
| 23:36.43 | WIMPy | kaushal: AFAIK Sangoma are the only ones doing >4 PRI. |
| 23:36.53 | WIMPy | StaRetji: Patch added. |
| 23:39.01 | kaushal | WIMPy: Thanks |
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