| 00:00.56 | kfife | [TK]D-Fender: YES, see that's exactly my point! I'm going to draw a line in the sand here and say I agree with you. A SONG or even a POEM may, just may, be... 'softer' somehow. |
| 00:01.19 | [8none1] | It's one of three boxes in a separate context. The other boxes in the default context email fine. |
| 00:01.31 | [8none1] | Any suggestions on how I can debug this further? |
| 00:02.59 | [TK]D-Fender | [8none1]: Start with solid proof that the VM recording is fine on the server in each format it records in. verify the box. Ferifyt he address its going to. Show us something of value, because right now we have nothing to advise you on. |
| 00:03.19 | [8none1] | np, i'll send some proof, jas |
| 00:03.41 | *** join/#asterisk emist_ (n=emist@unaffiliated/emist) |
| 00:04.00 | pids | 8nonel, check where the mail is going to as well. I had a provider that was stripping the attached file and left a blank file attached |
| 00:05.09 | kfife | Headin' out. |
| 00:05.15 | kfife | g'nite gentlemen. |
| 00:05.59 | [8none1] | pids: thanks but I receive mail from my personal mailbox from the same asterisk server just fine. |
| 00:07.13 | *** join/#asterisk mindCrime (n=chatzill@cpe-075-177-141-190.nc.res.rr.com) |
| 00:11.46 | [8none1] | http://pastebin.com/m24ebf3c4 |
| 00:12.20 | [8none1] | There's proof of a message with content, the voicemail.conf entry and the email |
| 00:14.42 | pids | the message says its msg0003.wav but I dont see a msg0003 in the directory only a msg0002 |
| 00:15.16 | [TK]D-Fender | [8none1]: does the file size match? |
| 00:15.30 | [TK]D-Fender | indeed |
| 00:15.32 | [TK]D-Fender | MISMATCH |
| 00:15.49 | [TK]D-Fender | sounds like someone went and disturbed the evidence by cleaning up their folder |
| 00:16.16 | [8none1] | What do you mean? What are you matching? |
| 00:17.03 | pids | 8nonel there is no msg0003.wav in your directory |
| 00:17.16 | [8none1] | here's the rest of the relevent bits of the voicemail.conf : http://pastebin.com/m2369c54c |
| 00:17.53 | [8none1] | [TK]D-Fender: what are you checking that matches? |
| 00:18.40 | [TK]D-Fender | [8none1]: the file in your folder VS the files in the email |
| 00:19.00 | [TK]D-Fender | [8none1]: the box has changed since the e-mail was sent |
| 00:19.01 | [8none1] | [TK]D-Fender: That's the point the email is sending an empty file |
| 00:19.27 | [TK]D-Fender | [8none1]: Point right now is we have no proof this is a defect |
| 00:19.35 | [8none1] | [TK]D-Fender: ah no sorry I sent the wrong body of an email |
| 00:19.36 | [TK]D-Fender | [8none1]: maybe it WAS blank. |
| 00:19.53 | [8none1] | [TK]D-Fender: I did two tests one deleted the message and the second didn't |
| 00:20.05 | [8none1] | [TK]D-Fender: jas and I'll patebin the correct one |
| 00:20.53 | [8none1] | http://pastebin.com/m2e28ce2f |
| 00:20.59 | [8none1] | [TK]D-Fender: soory for the mixup |
| 00:21.25 | [TK]D-Fender | [8none1]: ok, says 4 sec... |
| 00:21.48 | [TK]D-Fender | [8none1]: Now what tells me this is not corect? Maybe the caller hung up. |
| 00:21.57 | [8none1] | Yeah and ARI web page will play the message |
| 00:22.11 | [8none1] | No just a short message from myself |
| 00:23.50 | pids | err msg0003 was .05 seconds now its .04 seconds? |
| 00:24.04 | pids | nm |
| 00:24.12 | [8none1] | pids: first email was the wrong one |
| 00:24.20 | [8none1] | <PROTECTED> |
| 00:24.39 | pids | didnt read up |
| 00:25.35 | [TK]D-Fender | ok, I can't concentrate on this right now, hopefully others can. |
| 00:25.42 | [TK]D-Fender | heads off |
| 00:26.08 | [8none1] | This seems to only be a problem for mailboxes in the [custom] context |
| 00:26.32 | [8none1] | [TK]D-Fender: Thanks for your help anyway |
| 00:28.03 | pids | extra comma in the 1127 entry |
| 00:28.11 | *** part/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 00:28.53 | seanbright | that's supposed to be there |
| 00:29.29 | [8none1] | pids: that's the pager section, it should be there |
| 00:30.39 | pids | nm ,screen redraw was slow on pastebin, saw 3 commas :) |
| 00:30.54 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 00:31.04 | [8none1] | forgives pids but just this once |
| 00:32.35 | jblack | demand perfection. Never forgive. Let people know that you have high standards for them. Win! Win! Win! That's the only answer. |
| 00:35.00 | *** join/#asterisk ManxPower (n=manxpowe@215.sub-75-203-58.myvzw.com) |
| 00:35.29 | [8none1] | jblack: I'm that way when I'm not groveling for help. ;) |
| 00:35.36 | jblack | This lack of spam is disconcerting. My mailbox feels like a haunted house. |
| 00:35.53 | ManxPower | jblack: It's the end times for sure! |
| 00:36.27 | jblack | You know you're in a inhospital place when not even the roaches cling to life. |
| 00:36.44 | pids | 8nonel I found an error in box 1116 :) |
| 00:36.52 | [8none1] | pids: ok? |
| 00:36.55 | jblack | hospitipal. hospitital. Screw it. Unfriendly. |
| 00:37.02 | [8none1] | jblack: FYI about spam : http://www.computerweekly.com/Articles/2008/11/14/233375/spam-levels-drop-after-us-botnet-host-closed-down-by.htm |
| 00:37.06 | pids | attach = on it should be attac=yes |
| 00:37.18 | [8none1] | pids: I suck, thanks |
| 00:37.18 | pids | dont think thats the problem though :) |
| 00:37.26 | jblack | [8none1]: Yeah. That's exactly what I'm referring to. I'm getting _nothing_ any more. |
| 00:37.37 | jblack | well, maybe 2 a day. |
| 00:37.38 | [8none1] | pids: I'll try it anyway |
| 00:37.39 | pids | since you have a global attach statement |
| 00:38.33 | *** join/#asterisk MrNaz (n=mrnaz@203-217-81-147.dyn.iinet.net.au) |
| 00:38.35 | [8none1] | pids: it says attach=yes, you suck |
| 00:39.02 | [8none1] | pids: jk, please help |
| 00:39.07 | [8none1] | grovels again |
| 00:39.18 | seanbright | he is talking about box 1116 |
| 00:39.23 | seanbright | not 1127 |
| 00:39.42 | seanbright | which according to your pastebin says "attach=on" |
| 00:39.43 | [8none1] | ah, well 1116 works |
| 00:39.51 | pids | pids> 8nonel I found an error in box 1116 :) |
| 00:40.05 | [8none1] | ok, he is right. but 1127 is the one that's broken |
| 00:40.10 | seanbright | i think with the config stuff 'on' and 'yes' both work |
| 00:40.12 | seanbright | but don't quote me |
| 00:40.32 | [8none1] | Well, attach=on is working for box 1116 |
| 00:40.37 | seanbright | right |
| 00:40.45 | seanbright | so it's moot |
| 00:40.58 | [8none1] | or moo, cause the cow said it |
| 00:45.21 | [8none1] | could it be there's something special about the context [custom] in voicemail.conf? It's just strange that all the mailboxes I have created in the [default] context work. |
| 00:45.36 | [8none1] | But the ones in [custom] don't. |
| 00:46.01 | ManxPower | usually those sorts of things are syntax errors. i.e. one too many or too few commas |
| 00:49.45 | pids | 8nonel do a ls on the mailbox again and pb it. |
| 00:49.55 | pids | ls -l |
| 00:50.34 | [8none1] | ok, just proved it's not a context issue. I moved it to the default context and it didn't fix it. |
| 00:50.40 | [8none1] | Must be something else on the line. |
| 00:52.07 | [8none1] | ok, I think I found the issue. |
| 00:52.10 | [8none1] | volgain |
| 00:54.05 | [8none1] | Why would that only cause an issue on the email? |
| 00:54.23 | [8none1] | http://lists.digium.com/pipermail/asterisk-bugs/2008-July/022063.html |
| 00:54.25 | [8none1] | nm |
| 00:54.27 | pids | is sox installed? |
| 00:54.42 | [8none1] | but fixed in a newer version |
| 00:54.46 | [8none1] | bug* |
| 00:54.50 | pids | see it |
| 00:54.58 | [8none1] | I'm running 1.4.17 |
| 00:55.07 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 00:56.59 | [8none1] | Thanks guys for your help |
| 00:57.13 | [8none1] | Now I can go back to demanding perfection!! |
| 00:59.18 | [8none1] | FYI for all in the notes of the bug it's not a code issue |
| 00:59.53 | [8none1] | on ubuntu after 8.04 the sox package doesn't install any codecs by default |
| 01:00.03 | [8none1] | if you install libsox-fmt-all it will fix it |
| 01:04.57 | *** join/#asterisk erth64net (n=erth64ne@mail.opensourcery.com) |
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| 01:15.38 | jaytee | well, I crashed and burned on the dCAP practical lab today :-( |
| 01:16.58 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 01:17.10 | jaytee | hey [TK]D-Fender |
| 01:17.15 | joat | really bad? |
| 01:17.27 | [TK]D-Fender | y0 |
| 01:17.34 | jaytee | yeah, I ran out of time and only got a 45 out of a 100 |
| 01:17.40 | joat | ouch |
| 01:18.07 | jaytee | don't know about the written portion. I think I passed that and if I retest I only have to retake the lab if I passed the written this time. |
| 01:18.15 | [TK]D-Fender | jaytee: dCAP results? |
| 01:18.18 | jaytee | yeah |
| 01:18.29 | [TK]D-Fender | yeah... ouch |
| 01:19.20 | jaytee | I had only 2 of 3 phones working and ran out of time before I could finish the rest of the requirements. ran into problems with x-lite and couldn't remember what I'd already reviewed from earlier in the week. |
| 01:19.33 | jaytee | oh, well. I'll pass it next time I take it. |
| 01:19.47 | jaytee | still, the class was awesome |
| 01:28.57 | *** join/#asterisk Thorn_ (n=thorn@unaffiliated/thorn) |
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| 01:54.47 | cryptnix | if i want to preset a default extension to be injected into commedian mail ... *99,1,VoiceMailMain(s${CALLERIDNUM}) ... what portion do i edit |
| 01:55.30 | cryptnix | nvm |
| 01:57.14 | cryptnix | hrm, sEXT |
| 01:57.18 | cryptnix | doesn't ask for pass now. hrm. |
| 01:57.22 | cryptnix | reads more |
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| 02:03.43 | shriven | hey guys. I'm getting some issues building 1.6 on ubuntu 8.10. Everything seems fine until the make install, everytime it tries to make a directory it throws a permissions issue. Anyone have any ideas what that is about? |
| 02:04.44 | shriven | here is an example: http://pastebin.com/m2459b845 |
| 02:06.06 | pids | are you doing sudo make install or just make install? |
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| 02:21.32 | shriven | sudo |
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| 03:11.16 | drmessano | So umm |
| 03:11.47 | drmessano | Are t38_updtl and t38pt_udptl two different options or is one not correct? |
| 03:11.54 | *** part/#asterisk Spirits-Sight (n=christop@c-24-218-128-159.hsd1.ma.comcast.net) |
| 03:13.13 | jaytee | I've never seen t38pt_udptl but that doesn't mean it doesn't exist |
| 03:13.32 | jaytee | and after today I wouldn't take my word on squat :-) |
| 03:28.02 | drmessano | ok, so the PT looks flaky |
| 03:29.13 | drmessano | I am going with sip.conf from the 1.6 tarball |
| 03:29.23 | drmessano | Only mentions t38_udptl |
| 03:29.36 | drmessano | Also trying to get Exchange UM working on 1.6 |
| 03:29.41 | drmessano | Cant seem to pass a call thru |
| 03:32.41 | Micc | I'm still getting crashes after taking out the mysql stuff. The last log entry is app_queue.c: Queue members sucessfully reloaded from database. |
| 03:34.31 | Micc | http://pastebin.com/m1b0feb11 |
| 03:34.44 | Micc | You can see the gap in time after the app_queue line. |
| 03:34.53 | Micc | Anyone have any ideas what is causing the crash? |
| 03:35.57 | giovani | Micc: well if you take out your mssql db connection, does it still happen? |
| 03:36.08 | giovani | if so, at least then you've isolated the problem |
| 03:36.47 | Micc | giavani, I can't remove that for testing. This is a production system that requires the odbc connection for voicemail. |
| 03:37.06 | giovani | well if it's crashing, how functional could it be at this point? |
| 03:37.10 | Micc | why would it be trying to reregister extensions? |
| 03:37.30 | Micc | Maybe I have a loop in my dialplan |
| 03:42.17 | *** join/#asterisk mgroman (n=miles@d60-65-93-136.col.wideopenwest.com) |
| 03:42.40 | mgroman | asterisk? |
| 03:42.42 | *** part/#asterisk mgroman (n=miles@d60-65-93-136.col.wideopenwest.com) |
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| 03:55.00 | jblack | heh. There was reiserfs, now there's hammer. "The king is dead! Long live the king!" |
| 04:02.14 | *** join/#asterisk Micc (n=dotirc@c-67-183-169-202.hsd1.wa.comcast.net) |
| 04:02.30 | jaytee | reiserfs has been renamed to sweartogodididntdoitfs |
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| 04:26.27 | ricko73 | jaytee: nice |
| 04:27.07 | [netman] | I particulary prefer "reiskerfs" |
| 04:27.38 | ricko73 | may I also suggest orenthaljamesfs? |
| 04:27.39 | *** part/#asterisk mastag20 (i=mastag16@stfu.k-thx.biz) |
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| 04:30.23 | drmessano | lol |
| 04:30.37 | coppice | is reiserfs still considered a killer app for Linux? |
| 04:30.50 | drmessano | "My box doesnt work, I cant do any testing, help me fix" |
| 04:32.22 | ricko73 | coppice: are you here for the late show too? |
| 04:37.20 | baliktad | what's a good way to make bulk updates to AstDB |
| 04:37.37 | baliktad | for example, importing a bunch of rates from a .csv |
| 04:51.11 | jaytee | that's something that would be better done with ODBC and SQL |
| 04:53.47 | mankash | anybody help me troublshootoiing why my sip phones ar enot able to call each other |
| 04:59.37 | jaytee | are the extensions in the same context? are your sip phones registering to asterisk? |
| 05:00.14 | jaytee | pastebin a failed call |
| 05:00.45 | [netman] | baliktad: bulk updates to AstDB? I would write a small script |
| 05:01.10 | mankash | yes |
| 05:01.54 | mankash | http://pastebin.com/m2d9ec5b8 |
| 05:02.21 | jaytee | [netman] how would you read in the values from the .csv file from the dialplan? |
| 05:03.00 | [netman] | from the .csv to AstDB? |
| 05:03.19 | [netman] | and from AstDB to dialplan, I gues... |
| 05:03.30 | [netman] | guess |
| 05:04.53 | jaytee | [netman] I was asking how you'd "read in" the values in the csv from within the dialplan. I know you can use System() to run a script but it won't return values to Asterisk. How would you read those values from within the dialplan in order to input them into the AstDB? |
| 05:06.08 | baliktad | how indeed, I briefly considered just using asterisk -rx "database put ..." a thousand times in a row, but figured there has to be a better way |
| 05:06.27 | jaytee | mankash, that pastebin just shows manager logins and logouts from AMI. it doesn't show a failed call. do a core set verbose 10 and make a another test call. |
| 05:07.16 | mankash | <PROTECTED> |
| 05:07.16 | mankash | <PROTECTED> |
| 05:07.17 | jaytee | baliktad, you could do that in a while loop until an end of file I guess. |
| 05:07.20 | [netman] | jaytee: use an AGI if you want return values |
| 05:07.24 | mankash | it only show these 2 lines |
| 05:07.35 | mankash | and at the client 404 not found |
| 05:08.01 | jaytee | mankash, what version of *? |
| 05:08.16 | mankash | 1.6 |
| 05:08.47 | jaytee | and did you try increasing the verbosity? |
| 05:08.52 | mankash | yes |
| 05:09.07 | jaytee | if you do a sip show peers does it show the two phones? |
| 05:09.12 | mankash | yes |
| 05:09.22 | jaytee | and they're registered? |
| 05:09.26 | mankash | yes |
| 05:10.37 | [netman] | jaytee: could you write an AGI script which imports the csv file into the AstDB? |
| 05:11.00 | jaytee | [netman] not tonight |
| 05:12.33 | [netman] | jaytee: come on! you can use your favourite language :) |
| 05:13.00 | jaytee | you mean the one with an irish accent and all kinds of colorful "metaphors"? |
| 05:16.16 | [netman] | you can choose any language |
| 05:16.31 | [netman] | you only need some libraries |
| 05:16.51 | jaytee | are you asking me if I want a consulting contract? |
| 05:17.32 | jaytee | because I'm not looking for any side projects at the moment |
| 05:17.43 | ricko73 | mankash: do you have a dial plan that allows the phones to call each other? |
| 05:19.58 | [netman] | jaytee: I'm not asking you that |
| 05:20.42 | mankash | they are in the same context, is there any other settings |
| 05:20.43 | [netman] | I only say if you are programmer in any language, it shouldn't difficult for you to write an AGI |
| 05:22.18 | jaytee | [netman] I'd agree with that statement. it shouldn't be |
| 05:22.42 | ricko73 | mankash: yes, you need to write in the dial plan something like exten => 1XX,1,Dial(SIP/${ARG1}) |
| 05:23.40 | jaytee | mankash, try the book |
| 05:23.44 | jaytee | ~book |
| 05:23.44 | jbot | i heard book is probably 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://tfot.leifmadsen.com or see ~buybook |
| 05:25.48 | jaytee | nite all |
| 05:26.14 | [8none1] | [netman]: you could bulk import with a script to the manager interface |
| 05:27.48 | [netman] | mmmmmmm |
| 05:27.58 | [netman] | I should be |
| 05:28.20 | [netman] | at least, I know I can invoke System() from AMI :P |
| 05:29.49 | [8none1] | in AMI you have DBput, DBget, and DBdel |
| 05:30.10 | [netman] | so, what's the matter? |
| 05:30.17 | [netman] | I only have to iterate |
| 05:30.25 | [8none1] | correct |
| 05:33.44 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 05:35.52 | [TK]D-Fender | Or you could just us a BDB library and load the thing directly. Now who loads AstDB for an *AGI* anyways? Why is the dialplan doing a mass-process like this? |
| 05:36.08 | [netman] | so, do I need System() to iterate in AMI? |
| 05:36.51 | *** join/#asterisk Meaw (n=dino@213.244.81.144) |
| 05:37.09 | [TK]D-Fender | [netman]: First you're talking AGI, then AMI, and now you're talking about System. What on earth ar you trying to DO? |
| 05:37.40 | [netman] | [TK]D-Fender: lol |
| 05:37.49 | [netman] | I was talking about AGI, |
| 05:37.54 | [TK]D-Fender | [netman]: Any program can use AMI, AGI is when you want an app to manipulate your current channel |
| 05:37.55 | [netman] | then [8none1] ask me for AMI |
| 05:38.11 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 05:38.14 | [TK]D-Fender | [netman]: Let me know when you've go your head screwed on straight |
| 05:38.25 | [TK]D-Fender | [netman]: WTF are you trying to DO? |
| 05:38.59 | [netman] | [TK]D-Fender: may be.... I have awaken for 24 h |
| 05:39.18 | [netman] | [TK]D-Fender: I only try to answers questions |
| 05:41.19 | [TK]D-Fender | [netman]: Whose question? |
| 05:41.23 | [8none1] | [netman]: you said you wanted to load a CSV of rates into AstDB |
| 05:41.28 | [TK]D-Fender | [netman]: So far this looks like lot of talk about nothing. |
| 05:41.41 | [netman] | anyway, a bulk import of a csv into AstDB can be made simply from dialplan, and maybe an auxiliar AGI |
| 05:41.56 | *** join/#asterisk kisu (n=kvirc@daniel1117.broker.freenet6.net) |
| 05:41.57 | [TK]D-Fender | Why would you even BOTHER with AstDB? |
| 05:42.13 | [8none1] | [TK]D-Fender: I fixed my empty wav in email problem earlier. |
| 05:42.14 | [TK]D-Fender | If you have this table, just parse it when you NEED it. |
| 05:42.20 | [TK]D-Fender | [8none1]: What did you find> |
| 05:43.02 | [8none1] | It was an ubuntu issue with sox not having codecs. from 8.04 on the sox package dosen't include the codecs |
| 05:43.32 | [8none1] | [TK]D-Fender: you have to install libsox-fmt-all |
| 05:43.39 | [TK]D-Fender | [8none1]: So the file was fine and your player b0rked? |
| 05:44.17 | [netman] | BTW [8none1] why did you tell me if I can do that in AMI instead AGI? |
| 05:44.30 | [TK]D-Fender | [netman]: You can. |
| 05:44.32 | [8none1] | I found a asterisk bug that explained it |
| 05:44.34 | [8none1] | http://bugs.digium.com/view.php?id=12939 |
| 05:45.39 | [8none1] | if you want to bulk load a CSV into AstDB you could write a script to do that through AMI rather than your idea of calling "asterisk -r -x" over and over |
| 05:45.57 | *** join/#asterisk Maliuta (n=scooby@kiev.lusan.id.au) |
| 05:46.39 | [TK]D-Fender | [8none1]: Then again why would anyone waste time loading up AstDB in the first place? Its a stupid thing to maintain for a large list. |
| 05:47.24 | [8none1] | [TK]D-Fender: stupid is as stupid does |
| 05:48.22 | [netman] | [8none1]: but, if I would use AMI, how could I read the csv file, and how could I iterate over AstDB? I don't see the answer |
| 05:48.32 | [TK]D-Fender | [8none1]: Then he should find the abusrdly complex way possible to do it.. I say make the script on another server an have it E-MAIL the command over. then have Cron run hourly and pull the e-mails out 1 at a time to extract the instructions of the next record to add |
| 05:49.00 | [netman] | [TK]D-Fender: what is "large" ? |
| 05:49.10 | [8none1] | [TK]D-Fender: you forgot to use ftp in there somewhere |
| 05:49.29 | [TK]D-Fender | [8none1]: You need to leave room to draft up a 2.0 release! |
| 05:49.53 | [TK]D-Fender | [8none1]: Doing it right the first time is good business... doing ir right after 37 tries is job security L_ |
| 05:50.04 | [8none1] | hacks out a 2mb script to load 30kb of data |
| 05:52.12 | *** join/#asterisk billyjean (n=db@c-67-161-253-24.hsd1.ut.comcast.net) |
| 05:52.33 | [8none1] | [netman]: here's the deal. if you want to read from a csv in the dial plan write an agi. if you want to load the csv into astdb periodically (not the best solution) write a script to use the AMI to import. then you could use the AstDB in the dialplan. |
| 05:52.36 | [TK]D-Fender | You know I could have actually done the whole jobe with EXCEL in less time than we've spent talking about it... |
| 05:54.26 | [netman] | lol |
| 05:55.18 | [netman] | It's a pitty jaytee couldn't read this |
| 05:55.48 | [netman] | I don't know whether he wanted to make periodically updates |
| 05:56.45 | [TK]D-Fender | [netman]: [8none1] and you forgot the easiest way : cut & paste direct into CLI <- |
| 05:57.03 | [netman] | lol |
| 05:57.25 | [TK]D-Fender | that would have been done LONG ago. |
| 05:58.05 | [netman] | [8none1]: now I understood the AGI approach, thank you :) |
| 06:00.15 | *** join/#asterisk pcrane (n=pcrane@202.20.97.154) |
| 06:00.17 | [TK]D-Fender | I presonally thing the Excel + C&P to CLI is as far as this should have gotten |
| 06:00.38 | [TK]D-Fender | I personally think the Excel + C&P to CLI is as far as this should have gotten |
| 06:01.30 | [8none1] | [TK]D-Fender: it depends on how variable the data is. It's not a solution you could automate. |
| 06:01.50 | [TK]D-Fender | [8none1]: Variable??? its CSV <- |
| 06:04.20 | baliktad | to the asterisk command line or system command line? |
| 06:04.54 | [TK]D-Fender | Anyone who wanted this automatic would simply point an ODBC-CSV driver directly at the file and use ODBC in the dialplan instead of AstDB. Updating is as fast as copying the file there |
| 06:05.12 | [TK]D-Fender | baliktad: Either, but I was thinking * CLI |
| 06:05.40 | [8none1] | [TK]D-Fender: but from where? it could be an export that changes. I agree the C&P is an easy one time solution. I'm used to dealing with daily update feeds in CSV type formats. |
| 06:05.58 | [TK]D-Fender | [8none1]: See above |
| 06:06.13 | [8none1] | [TK]D-Fender: agreed good solution |
| 06:06.16 | baliktad | experimenting... system CLI adds ~0.5 - 1 sec overhead to each db put |
| 06:06.24 | baliktad | C&P to * CLI works almost instantly |
| 06:06.39 | [TK]D-Fender | [8none1]: Smart people don't bother with AstDB in the first place. it is a SHIT place to store DB info like this. Get a REAL database |
| 06:07.15 | [TK]D-Fender | baliktad: Yup, and as usual noone was able to commit to an est # of transactions to do. All talk, no info. |
| 06:07.26 | [TK]D-Fender | blahBLAHblahBLAHblahBLAHblahBLAH |
| 06:07.40 | baliktad | 7500 |
| 06:07.45 | [TK]D-Fender | pretty much the norm, All "how" when the "why" is worthless |
| 06:08.15 | [TK]D-Fender | 7500? EW |
| 06:08.22 | baliktad | standard intl rate list |
| 06:08.28 | [TK]D-Fender | yup, this should be in an external DB in the first palce. |
| 06:10.34 | baliktad | I wonder how much you could copy & paste at once |
| 06:11.18 | [TK]D-Fender | baliktad: interesting theory question, but still just highlights the absurdity of what this does in the end. |
| 06:16.52 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 06:21.38 | *** join/#asterisk freakazoid0223 (n=mattc@pool-68-162-78-237.phil.east.verizon.net) |
| 06:28.31 | *** join/#asterisk jameswf (n=james@ip68-3-60-66.ph.ph.cox.net) |
| 06:39.11 | *** join/#asterisk MrNaz (n=mrnaz@210-84-52-246.dyn.iinet.net.au) |
| 06:39.18 | *** join/#asterisk TrentCreek (n=kvirc@adsl-70-254-118-180.dsl.hrlntx.swbell.net) |
| 06:40.10 | TrentCreek | well |
| 06:40.18 | pcrane | I need a hand with some callerID stuff |
| 06:40.20 | pcrane | I've got: |
| 06:40.28 | pcrane | Set(CALLERID(name)=blah) |
| 06:40.45 | pcrane | then Dial(SIP/801&SIP/802&SIP/803,,rtTwWo) |
| 06:40.54 | pcrane | but the caller id doesn't show on the phone's display |
| 06:41.03 | pcrane | just Unknown Unknown |
| 06:41.05 | pcrane | :( |
| 06:41.41 | TrentCreek | you using a SIP device? |
| 06:41.52 | TrentCreek | hardware |
| 06:41.55 | pcrane | no, it's a call from the PSTN |
| 06:42.07 | pcrane | ISDN incoming goes to a context which does the above |
| 06:42.13 | TrentCreek | ohh..then you got me there.>i do not mess with that |
| 06:43.33 | pcrane | the situation is essentially this: call comes in, I know what DID it comes in on, I send the call off to different contexts depending on what DID was called... |
| 06:44.09 | pcrane | so, 801, 802, 803 are all answering for different DIDs, but they all need to know what DID was called |
| 06:44.11 | TrentCreek | try the older context |
| 06:44.19 | pcrane | older context? |
| 06:44.26 | TrentCreek | from v1.4 |
| 06:44.32 | TrentCreek | or 1.2 |
| 06:44.44 | pcrane | the o option? |
| 06:44.52 | pcrane | o: Restore the Asterisk v1.0 CallerId behaviour (send the original caller's ID) in Asterisk v1.2 (default: send this extension's number) |
| 06:44.52 | TrentCreek | no... |
| 06:45.14 | TrentCreek | what are you using? |
| 06:45.26 | pcrane | Dial(SIP/801&SIP/802&SIP/803,,rtTwWo) |
| 06:45.29 | pcrane | ah |
| 06:45.30 | pcrane | sorry |
| 06:45.32 | pcrane | I see |
| 06:45.34 | pcrane | asterisk 1.4 |
| 06:45.37 | pcrane | my bad |
| 06:45.38 | pcrane | :p |
| 06:45.40 | pcrane | is tired |
| 06:45.55 | TrentCreek | oh..you were mentioning "O option" |
| 06:46.02 | TrentCreek | I have no idea what that is |
| 06:46.17 | pcrane | o: Restore the Asterisk v1.0 CallerId behaviour (send the original caller's ID) in Asterisk v1.2 (default: send this extension's number) |
| 06:46.59 | TrentCreek | i dont know where that is |
| 06:47.06 | *** join/#asterisk tripps (n=sean@72.20.150.206) |
| 06:47.11 | TrentCreek | sure you using Asterisk 1.4?? |
| 06:47.19 | pcrane | yes |
| 06:47.36 | pcrane | # asterisk -r |
| 06:47.36 | pcrane | Asterisk SVN-branch-1.4-r147681, Copyright (C) 1999 - 2008 Digium, Inc. and others. |
| 06:47.40 | TrentCreek | you mean at compile time? |
| 06:47.45 | TrentCreek | ohh.. |
| 06:47.48 | TrentCreek | okay |
| 06:47.49 | TrentCreek | no |
| 06:47.50 | TrentCreek | not that |
| 06:47.58 | TrentCreek | in your dial plan |
| 06:47.59 | pcrane | heh |
| 06:48.10 | pcrane | how do I find out? |
| 06:48.18 | TrentCreek | ~book |
| 06:48.19 | jbot | hmm... book is probably 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://tfot.leifmadsen.com or see ~buybook |
| 06:48.22 | TrentCreek | no more jbot |
| 06:48.47 | pcrane | <cheeky>chapter, verse?</cheeky> |
| 06:48.50 | pcrane | looks |
| 06:49.14 | drmessano | "do it for me" ? |
| 06:49.25 | pcrane | lol |
| 06:49.32 | pcrane | hence the cheeky |
| 06:49.41 | TrentCreek | under dialplans |
| 06:49.43 | drmessano | It's not really cheeky |
| 06:49.50 | pcrane | sorry |
| 06:49.51 | drmessano | More "annoying" |
| 06:49.56 | pcrane | It was ment as a joke |
| 06:50.02 | pcrane | I didn't mean to come off that way |
| 06:50.06 | drmessano | Theres no joking in here |
| 06:50.08 | drmessano | Ever |
| 06:50.24 | drmessano | Its in the bylaws |
| 06:50.29 | drmessano | ~joke |
| 06:50.30 | jbot | What's a chicken coupe with 4 doors - a Chicken Sedan! |
| 06:50.43 | pcrane | ... |
| 06:50.43 | drmessano | See, even the bot cant pull it off |
| 06:54.56 | *** join/#asterisk terracon (n=garry@CPE001cf097a750-CM0012254076d6.cpe.net.cable.rogers.com) |
| 07:01.46 | *** join/#asterisk pids (n=pids@dsl081-072-084.sfo1.dsl.speakeasy.net) |
| 07:02.06 | jblack | jbot: knock, knock |
| 07:02.27 | jblack | somehow, I'm just a little disappointed. |
| 07:03.21 | TrentCreek | in what? |
| 07:03.33 | TrentCreek | and stop being so Shallow, Hal |
| 07:09.27 | drmessano | ~sipjoke |
| 07:09.28 | jbot | Knock, knock. Who's there? SIP. SIP who? SIP really authenticates this way. |
| 07:20.33 | *** join/#asterisk fnordus (n=dnall@70.71.225.48) |
| 07:28.54 | *** join/#asterisk MrNaz (n=mrnaz@210-84-52-246.dyn.iinet.net.au) |
| 07:32.21 | pcrane | I'm still not having any luck... |
| 07:34.06 | TrentCreek | why not?I used the older CALLERID and it worked fine |
| 07:34.22 | pcrane | I'm using Set(CALLERID(name)=test) |
| 07:34.31 | pcrane | should I be doing something else? |
| 07:34.37 | TrentCreek | yes |
| 07:34.54 | TrentCreek | use what the 1.2 book has |
| 07:34.58 | pcrane | so like SetCallerID |
| 07:35.05 | pcrane | or SetCIDName |
| 07:35.53 | TrentCreek | yes..it worked fine |
| 07:36.06 | TrentCreek | I even would change the CID on people a lot..buwhahahaah |
| 07:36.09 | pcrane | WARNING[27348]: pbx.c:1832 pbx_extension_helper: No application 'SetCIDName' for extension |
| 07:36.12 | pcrane | can't |
| 07:36.13 | pcrane | :( |
| 07:36.32 | TrentCreek | well I used what the book for 1.2 had in it |
| 07:36.38 | pcrane | ok |
| 07:37.34 | TrentCreek | even though I got warning about it would be fased out |
| 07:39.36 | kaldemar | SetCIDName and SetCIDNum were removed for 1.4. |
| 07:40.31 | pcrane | so, I *have* to use Set(CALLERID(name)=test) |
| 07:40.34 | TrentCreek | it worked in 1.4 for me.. |
| 07:41.08 | TrentCreek | Set(CALLERID(name)=test) |
| 07:41.19 | TrentCreek | Set(CALLERID(name)=test) could never get to work either |
| 07:44.27 | pcrane | dialplan snippet: |
| 07:44.27 | pcrane | http://pastebin.com/m529a6e7c |
| 07:44.37 | pcrane | execution snippet |
| 07:44.38 | pcrane | http://pastebin.com/m750e9d4 |
| 07:46.13 | TrentCreek | looking |
| 07:46.57 | pcrane | (the g option's there so I can see the caller ID *after* the call has finished) |
| 07:47.55 | TrentCreek | shouldn't there be a decimal value in there>? |
| 07:48.16 | pcrane | in the Set(CALLERID(name) ? |
| 07:48.53 | TrentCreek | of however you set it up |
| 07:49.27 | pcrane | I don't follow... |
| 07:49.38 | pcrane | the name is just the name you'd like to display |
| 07:50.02 | pcrane | call in to this context, calling two SIP devices should present the caller id name that's set... |
| 07:50.18 | pcrane | I don't understand how a decimal value can fit in there? |
| 07:50.21 | TrentCreek | you trying to set the name or number? |
| 07:50.27 | pcrane | name |
| 07:50.39 | pcrane | I don't really care about the number |
| 07:50.49 | pcrane | (once the name works, I can use the same method to set the number) |
| 07:50.50 | TrentCreek | that is tricky and not guranteed to work |
| 07:51.15 | pcrane | ok |
| 07:51.20 | TrentCreek | number will work, but name does nto always work |
| 07:51.25 | TrentCreek | I have tested it |
| 07:51.29 | pcrane | ok |
| 07:51.36 | pcrane | but on the internal lan? |
| 07:51.42 | TrentCreek | sometimes the carrier will reverse name lookup |
| 07:51.48 | pcrane | this is for calls coming in from the outside... |
| 07:52.32 | TrentCreek | oh |
| 07:53.47 | pcrane | it doesn't work if I set the number either |
| 07:54.57 | TrentCreek | yeah I had problems with the new one so I just used the older |
| 07:55.56 | pcrane | you're using asterisk 1.4? |
| 07:55.58 | pcrane | or 1.2? |
| 07:56.01 | *** join/#asterisk korihor (n=korihor@200-71-161-1.genericrev.telcel.net.ve) |
| 07:56.15 | TrentCreek | i am not sure anymore..I restarted on a whole new server |
| 07:57.36 | TrentCreek | let me see what I got now |
| 07:57.45 | TrentCreek | i think I saved some files |
| 07:59.23 | kaldemar | pcrane: take a SIP debug of a call and pastebin it. |
| 08:00.20 | pcrane | ok |
| 08:04.59 | pcrane | http://pastebin.com/m6a98a90b |
| 08:05.06 | pcrane | 811 answers the phone |
| 08:10.28 | kaldemar | strange. if you're not doing something to the caller id in sip.conf, then your svn version's behaviour is different from 1.4.22. |
| 08:13.01 | pcrane | nope, nothing |
| 08:13.20 | pcrane | nothing apart from setting the caller id of the extensions to be: 801 <801> |
| 08:13.26 | pcrane | (for example) |
| 08:13.42 | pcrane | but I don't see how that relates... |
| 08:14.01 | pcrane | call flow is from the PSTN -> dial(sip/801&sip/802) |
| 08:14.23 | drmessano | He's using 1.4 branch, it's not gonna be different than 1.4 latest |
| 08:14.39 | pcrane | before the dial, I try to set the caller id, which doesn't work |
| 08:14.44 | pcrane | for the record: |
| 08:14.51 | pcrane | # asterisk -r |
| 08:14.52 | pcrane | Asterisk SVN-branch-1.4-r147681, Copyright (C) 1999 - 2008 Digium, Inc. and others. |
| 08:15.04 | drmessano | Thats twice you pasted that |
| 08:15.08 | pcrane | yes |
| 08:15.24 | drmessano | Most of us can scroll, but thanks for the spam |
| 08:19.53 | pcrane | http://bugs.digium.com/view.php?id=13647 |
| 08:19.55 | pcrane | woot |
| 08:20.34 | *** join/#asterisk xnixan (n=xnixan@unaffiliated/xnixan) |
| 08:20.49 | pcrane | thanks for the help TrentCreek, kaldemar, drmessano |
| 08:21.46 | TrentCreek | working? |
| 08:21.48 | pcrane | yep |
| 08:22.00 | pcrane | you can't change the callerid presentation from the dialplan, your best option is the set 'sendrpid = yes' in sip.conf. |
| 08:22.08 | pcrane | set that, and away it goes |
| 08:23.01 | *** join/#asterisk kisu (n=kvirc@daniel1117.broker.freenet6.net) |
| 08:23.26 | TrentCreek | now if I can log into my darned server as SU |
| 08:23.32 | TrentCreek | maybe it was highakced |
| 08:24.03 | drmessano | or you typoed the password... O.o |
| 08:24.15 | TrentCreek | 20 times? |
| 08:24.34 | drmessano | When you made it, duh |
| 08:24.34 | TrentCreek | i just logged in 2 days ago |
| 08:24.39 | drmessano | mr highakced |
| 08:24.55 | TrentCreek | I have had this server for 4 months |
| 08:25.02 | TrentCreek | not changeded SU pass in 2 |
| 08:25.06 | drmessano | Still no sense of humor I see.. |
| 08:25.10 | drmessano | Nevermind |
| 08:25.17 | drmessano | highakced <----- |
| 08:25.45 | TrentCreek | well I am doing 4 confs at the same time |
| 08:27.44 | pcrane | now I'm getting: WARNING[27749]: chan_sip.c:6983 build_rpid: Unsupported callingpres (-1) |
| 08:27.47 | pcrane | grr |
| 08:30.44 | TrentCreek | well good thing there is a CP for the server |
| 08:31.02 | pcrane | ah SetCallerPres.... |
| 08:31.20 | TrentCreek | someome must have hijacked it..I changed the PW |
| 08:50.24 | *** join/#asterisk raasdnil (n=mikel@60-241-138-146.static.tpgi.com.au) |
| 08:52.19 | kerx | anyone know of a good voip provider for wholesale? |
| 08:52.29 | pcrane | where? |
| 08:53.06 | TrentCreek | yes |
| 08:53.23 | raasdnil | ok... i've been searching around for about 24 hours, read every bloody dahdi doc I could find and I am stuck. Anyone have any idea on this? http://www.pastie.org/315481 |
| 08:53.24 | TrentCreek | but not so easy to setup |
| 08:55.11 | TrentCreek | kerx: Carrie Exchange |
| 08:55.34 | kerx | i can't find there website |
| 08:55.37 | kerx | carrier u mean? |
| 09:19.37 | TrentCreek | who uses free PBX? |
| 09:19.40 | *** join/#asterisk Aces1up (n=Now@ip70-173-52-152.lv.lv.cox.net) |
| 09:20.04 | Aces1up | i need to connect my asterisk box to pots lines, so i need a fxo port right? |
| 09:22.05 | TrentCreek | sure |
| 09:22.36 | TrentCreek | why spend lots of $$ on that when DIDs are so much cheaper |
| 09:22.54 | Aces1up | ok, when i installed my sangoma a200 card.. it said i had the following ports.. when doing a ztcfg -vvvvv i got the following: Channel 01: FXS Kewlstart (Default) (Slaves:01) |
| 09:23.02 | Aces1up | is that correct for a fxo port? |
| 09:23.22 | TrentCreek | i dont know..I dont mess with those things..I do 100% internet |
| 09:55.02 | gambler1 | Hi, does anyone here use cdr adaptive odbc module or maybe with curl support? |
| 10:11.30 | *** join/#asterisk cryptnix (n=andrew@216.111.201.3) |
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| 13:23.46 | *** join/#asterisk fromnewark (n=fromnewa@adsl-75-63-3-167.dsl.austtx.sbcglobal.net) |
| 13:25.47 | fromnewark | help? any experts online? |
| 13:28.46 | *** join/#asterisk mankash (n=rom10@bas1-toronto63-1096579767.dsl.bell.ca) |
| 13:31.26 | x86 | fromnewark: no one can help if you don't describe the problem ;) |
| 13:33.08 | Maliuta | x86: problem == (clue < 0) |
| 13:33.17 | postel | fromnewark: http://www.catb.org/~esr/faqs/smart-questions.html |
| 13:35.39 | x86 | Maliuta: hah |
| 13:36.08 | fromnewark | Site A has asterisk appliance, site B has two phones. All traffic is running through the appliance. Can voice traffic just run from phone to phone? |
| 13:36.45 | Maliuta | depends on the configurations |
| 13:37.11 | mvanbaak | ~ask |
| 13:37.12 | jbot | ask is, like, Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 13:37.16 | Maliuta | and by "can" do you mean "is it possible for" or "will"? |
| 13:39.15 | x86 | fromnewark: it is possible using re-invites |
| 13:41.27 | fromnewark | thanks x86. So the fact that SIP voice traffic all flows through the asterisk server is normal operation? For some reason i thought a SIP proxy just setup the call, and wasn't expecting to see that. |
| 13:42.56 | fromnewark | Maiuta, by can I mean that I want voice traffic to go from phone to phone and the asterisk to just setup the call. I need configure in this way. |
| 13:46.04 | *** join/#asterisk UnixDawg (n=UnixDawg@cpe-98-28-149-231.cinci.res.rr.com) |
| 13:52.28 | x86 | fromnewark: you can do it either way |
| 13:54.47 | fromnewark | Which configuration option controls this behavior on the appliance? I'm using an aa50 and a linksys ATA (2fxs) |
| 13:55.38 | x86 | niether of which are supported here |
| 13:55.51 | x86 | aa50 == #switchvox |
| 13:57.42 | fromnewark | ok. thanks. |
| 13:59.07 | *** part/#asterisk fromnewark (n=fromnewa@adsl-75-63-3-167.dsl.austtx.sbcglobal.net) |
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| 14:21.06 | *** join/#asterisk fromnewark (n=fromnewa@adsl-75-63-3-167.dsl.austtx.sbcglobal.net) |
| 14:21.46 | fromnewark | Is there an IRC support channel for the AA50 appliance by digium? |
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| 15:28.55 | neobsd | hi |
| 15:29.10 | neobsd | please how i can configure asterisk as sip provider ? |
| 15:29.12 | neobsd | .. |
| 15:29.16 | neobsd | is possible ? |
| 15:29.28 | *** join/#asterisk Dr-Linux|home (n=Nothing@119.63.130.34) |
| 15:29.50 | Dr-Linux|home | any idea, from where i can get "dial-tone" wav or gsm file? |
| 15:30.36 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 15:33.39 | doug | http://www.telephonetribute.com/audio/dial_tone.wav |
| 15:36.41 | Dr-Linux|home | doug: not working for me |
| 15:40.14 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 15:40.14 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:47.41 | doug | check http://drlinux.con.com |
| 15:47.44 | doug | what's not working about it? |
| 15:52.49 | drmessano | thinks there is a vast global conspiracy surrounding SIP TCP |
| 15:53.26 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 15:56.28 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
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| 16:03.47 | *** part/#asterisk feeds (n=feeds@85-135-225-22.adsl.slovanet.sk) |
| 16:06.22 | drmessano | [TK]D-Fender: How much do you know about qualify? |
| 16:07.49 | [TK]D-Fender | drmessano: yes=2000, and its the response time allowed, not the interval. |
| 16:08.00 | [TK]D-Fender | drmessano: interval is fixed in the code somewhere IIRC |
| 16:08.12 | drmessano | qualifyfreq= |
| 16:08.29 | drmessano | and good.. because seems half the internet doesnt know how that works |
| 16:08.36 | drmessano | I wanted to be sure I did |
| 16:08.49 | [TK]D-Fender | RMod_: is that an actual sip.conf parm? |
| 16:09.01 | [TK]D-Fender | drmessano: ^ |
| 16:09.05 | drmessano | I need to find a default value for qualifyfreq= |
| 16:09.11 | drmessano | Yes, actually.. per sip.conf |
| 16:09.48 | drmessano | Let me see if I can spam without spamming |
| 16:10.27 | drmessano | ;qualifyfreq=60 ; Qualification: How often to check for the host to be up in seconds. Set to low value if you use low timeout for NAT of UDP sessions |
| 16:10.38 | drmessano | But I dont know what the default is |
| 16:12.11 | drmessano | les.net apparently cracked down on garbage sip traffic and went on a crusade to shut everyones qualify off.. I dont think my service with them has worked in the week since.. They asked about limiting traffic to > every 60 seconds, but if that ^^^^^ is correct then thats already being done |
| 16:12.52 | [TK]D-Fender | drmessano: yOU DON'T REALLY NEED IT FOR THEM ANYWAYS... |
| 16:13.18 | [TK]D-Fender | drmessano: you don't need a keep-alive for a provider.. tis jsut a nicety to speed up FAILURE if they go down so you don't wait in dial. |
| 16:13.29 | [TK]D-Fender | drmessano: but thenn, how often is THAT supposed to happ4en? |
| 16:13.47 | [TK]D-Fender | drmessano: Most would think that if they need to check up on their provider that they should get a NEW one. |
| 16:13.48 | drmessano | I wouldnt think so either, but right after I turned it off, I started having problems with audio |
| 16:13.52 | *** join/#asterisk terracon (n=garry@CPE001cf097a750-CM0012254076d6.cpe.net.cable.rogers.com) |
| 16:13.52 | drmessano | lol |
| 16:13.58 | drmessano | I am seriously considering the later |
| 16:14.03 | [TK]D-Fender | drmessano: :) |
| 16:14.15 | [TK]D-Fender | drmessano: Sometimes it really is hidden in the big print :) |
| 16:14.27 | [TK]D-Fender | drmessano: Kinda of an open-mike statement |
| 16:14.41 | drmessano | I think I am gonna turn it back on, just for shits and giggles, and see if my audio problems disappear |
| 16:15.06 | drmessano | If they, do they can go to H-E-Double-Hockey-Sticks |
| 16:15.15 | drmessano | errr |
| 16:15.24 | drmessano | If it does, they can go to H-E-Double-Hockey-Sticks |
| 16:15.33 | [TK]D-Fender | drmessano: What kind of "problems"? |
| 16:15.52 | drmessano | Complete and total lack of audio.. very NAT-like |
| 16:16.28 | drmessano | 100% working config for over a year, only change I made is turning qualify off that they bitched about |
| 16:16.44 | *** join/#asterisk raasdnil (n=mikel@60.241.138.146) |
| 16:17.05 | drmessano | Try the line two days later, no audio.. CLI looks completely fine.. Playing audio on the IVR, etc |
| 16:17.48 | [TK]D-Fender | drmessano: that makes little sense... |
| 16:18.23 | drmessano | I am stumped myself.. What I did should NOT have had this effect |
| 16:18.44 | drmessano | Of course, I tweak.. but it's been a pretty tweakless two weeks here |
| 16:18.47 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 16:19.00 | drmessano | So I can say pretty confidently, I havent changed anything else that I will claim to know about sober |
| 16:20.11 | [TK]D-Fender | drmessano: You dun got mah daughter in a family way! |
| 16:20.16 | [TK]D-Fender | grabs his shotgun |
| 16:20.42 | [TK]D-Fender | y'all there's gonna be a weddin'! |
| 16:20.59 | drmessano | lol |
| 16:21.45 | drmessano | Ive realized very quickly the good experiences I have had with les.net are now being overshadowed by their shitty response this recent issue |
| 16:22.30 | raasdnil | anyone hooked up a legacy PBX to Asterisk via a PRI before? |
| 16:22.39 | raasdnil | is having all sorts of fun with an NEC system |
| 16:22.55 | raasdnil | got inbound calls and inbound DID all working fine, but outbound is just not talking. |
| 16:23.15 | raasdnil | inbound = Telco => E1 => * => E1 => NEC |
| 16:23.57 | raasdnil | I have a thread going on *-users called "PBX -> PRI -> * -> Telco not working" if any of ya have some good ideas .... :/ |
| 16:24.05 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
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| 16:26.10 | [TK]D-Fender | raasdnil: You aren't showing us what you're getting from the PBX... |
| 16:26.27 | [TK]D-Fender | raasdnil: So until you do we can't help you much... |
| 16:26.35 | [TK]D-Fender | raasdnil: PASTEBIN is your friend |
| 16:26.39 | [TK]D-Fender | ~pb |
| 16:26.40 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 16:27.13 | raasdnil | [TK]D-Fender: here is a pastie of a pri debug session on an outbound call from PBX via asterisk: http://www.pastie.org/315597 |
| 16:28.35 | raasdnil | the CLI without debug just reports -- Extension 's' in context 'from-nec' from '' does not exist. Rejecting call on channel 0/31, span 1 ---- which seems like it just isn't getting any numbers from the NEC. Thus the debug |
| 16:28.57 | [TK]D-Fender | raasdnil: then thats all there is. |
| 16:29.15 | [TK]D-Fender | raasdnil: Dialplan error and it is telling you to your face what its looking for. |
| 16:29.32 | [TK]D-Fender | raasdnil: ` Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ] <--- unassigned = * refuses because it doesn't match the requested # |
| 16:30.04 | raasdnil | hmm... so it is after extension 1 ?? |
| 16:30.41 | [TK]D-Fender | raasdnil: Extension 's' in context 'from-nec' from â does not exist <-- |
| 16:30.54 | x86 | mark spencer is rad |
| 16:31.14 | x86 | someone as busy as him taking time out of his schedule to talk to me... that's so cool :) |
| 16:31.22 | raasdnil | [TK]D-Fender: I get that bit. But if I provide a default extension then I'm not going to capture any number to actually dial out right? |
| 16:31.40 | [TK]D-Fender | raasdnil: they AREN'T dialing a number in that call. |
| 16:31.54 | [TK]D-Fender | raasdnil: pastebin your zapata.conf |
| 16:31.59 | *** join/#asterisk sucituanbo (n=fail@c-24-21-121-148.hsd1.wa.comcast.net) |
| 16:32.06 | raasdnil | [TK]D-Fender: no... the problem is I _AM_ dialing a number in that call. That debug was from me dialing out on the NEC |
| 16:32.09 | raasdnil | ok, one sec |
| 16:32.22 | [TK]D-Fender | raasdnil: well the PBX didn't pass it in that debug |
| 16:32.32 | raasdnil | right |
| 16:32.35 | [TK]D-Fender | raasdnil: Perhaps they intend to pass it as DTMF post-connect <- |
| 16:33.46 | raasdnil | using dahdi, here is the dahdi/system.conf http://www.pastie.org/315602 |
| 16:38.03 | *** join/#asterisk Dovid (n=annon@tony09-121-90.inter.net.il) |
| 16:38.10 | [TK]D-Fender | raasdnil: chan_dahdi.conf please... |
| 16:39.39 | raasdnil | http://www.pastie.org/315608 |
| 16:39.46 | raasdnil | [TK]D-Fender: there you go. |
| 16:40.05 | raasdnil | I am reading up on that PRI errors via google... |
| 16:45.10 | *** join/#asterisk korihor (n=korihor@200-71-160-1.genericrev.telcel.net.ve) |
| 16:46.12 | raasdnil | this is dahd-channels.conf http://www.pastie.org/315612 |
| 16:46.16 | *** join/#asterisk sdaniels (n=chatzill@cpe-72-190-15-241.tx.res.rr.com) |
| 16:46.48 | sdaniels | What file do I edit to change the logging to /var/log/asterisk/messages? |
| 16:46.57 | Dovid | logger.conf |
| 16:47.02 | sdaniels | thx |
| 16:53.17 | [TK]D-Fender | raasdnil: Go answer it with an IVR and see if it dials DTMF |
| 16:53.29 | raasdnil | k |
| 16:53.32 | raasdnil | bbs |
| 16:53.38 | [TK]D-Fender | raasdnil: Actually even an open-ended "Read" should do |
| 16:53.55 | raasdnil | [TK]D-Fender: if you can help me get it working, I'll be willing to help empty a bit of an amazon wish list... |
| 16:53.59 | [TK]D-Fender | raasdnil: Enought to prove if that's how it will pass the # |
| 16:54.17 | [TK]D-Fender | raasdnil: Sure |
| 17:00.07 | [TK]D-Fender | raasdnil: And do yourself a favor and rip out all of the commented lines from your configs. it confuses what you hve actually set yourself. |
| 17:00.34 | raasdnil | [TK]D-Fender: I'll do that. |
| 17:00.36 | raasdnil | Read works. |
| 17:00.42 | raasdnil | <PROTECTED> |
| 17:00.43 | raasdnil | <PROTECTED> |
| 17:00.44 | [TK]D-Fender | raasdnil: so its post-dial DTMF? |
| 17:00.53 | [TK]D-Fender | raasdnil: Does it matchw ath you dialed? |
| 17:01.02 | [TK]D-Fender | raasdnil: Does it match what you dialed? |
| 17:01.15 | raasdnil | yes |
| 17:01.26 | [TK]D-Fender | raasdnil: ok, there you go |
| 17:01.54 | raasdnil | so now, doing something like exten => s,1,WaitExten() |
| 17:02.04 | raasdnil | then catching the "t" and doing a Dial on the EXTEN variable? |
| 17:02.44 | [TK]D-Fender | raasdnil: Maybe just a pure "read would do the job, but an IVR to collect the digits works as well. |
| 17:03.26 | [TK]D-Fender | raasdnil: an IVR with an exten => _X. should do just fine. |
| 17:03.46 | raasdnil | IVR = Interactive Voice Response? |
| 17:03.53 | [TK]D-Fender | raasdnil: If you hit a timeout it means there wasn't at least 2 digits dialed. I don't imagine that happening, do you? |
| 17:04.05 | raasdnil | no. |
| 17:04.09 | [TK]D-Fender | raasdnil: Yes. IVR = DTMF response menu |
| 17:04.11 | raasdnil | will always be at least 5 |
| 17:04.46 | [TK]D-Fender | raasdnil: so all you should really need is to Answer, and the set a minimal timeout along with a response timeout of say 5, and a digit of 2. |
| 17:04.55 | [TK]D-Fender | raasdnil: so your PBX users don't wait TOO long |
| 17:05.58 | [TK]D-Fender | raasdnil: You can tweak his as you go |
| 17:06.31 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 17:07.12 | raasdnil | [TK]D-Fender: ok... looking how to implement that now |
| 17:09.18 | *** join/#asterisk pluesch0r (n=pluesch0@vie-078-142-129-071.dsl.sil.at) |
| 17:11.49 | raasdnil | [TK]D-Fender: ok... question.. I get the value into the DialedNumber variable... but how do I get it then to dial? http://www.pastie.org/315619 |
| 17:12.31 | raasdnil | oh.. remove _X. and replace with s ? |
| 17:12.31 | [TK]D-Fender | raasdnil: exten => s,n,Dial(DAHDI/g2/${DialedNumber},,T) |
| 17:12.38 | raasdnil | I just figured that out as well :) |
| 17:12.40 | [TK]D-Fender | raasdnil: You MIXED those 2 ideas up together. |
| 17:12.46 | raasdnil | yeah, got it... |
| 17:12.56 | [TK]D-Fender | raasdnil: Do it as a read or as an IVR with waitexten, but not both |
| 17:13.00 | raasdnil | ok, fixed that..... |
| 17:13.03 | raasdnil | now I get : |
| 17:13.03 | raasdnil | [Nov 16 15:12:56] WARNING[7990]: app_dial.c:827 wait_for_answer: Unable to forward voice or dtmf |
| 17:13.34 | [TK]D-Fender | raasdnil: Show a complete call |
| 17:14.33 | raasdnil | http://www.pastie.org/315619 |
| 17:14.50 | raasdnil | 1414 is the dial prefix needed for our telco, the NEC system apends it automatically |
| 17:15.38 | raasdnil | http://www.pastie.org/315619 <== reload to see extensions.conf |
| 17:16.44 | [TK]D-Fender | raasdnil: try dialing out g2 with a SIP phone. |
| 17:16.54 | raasdnil | ok |
| 17:20.55 | *** join/#asterisk interfaithquest (n=interfai@ip67-88-184-130.z184-88-67.customer.algx.net) |
| 17:21.43 | interfaithquest | hello has anyone had any success with p2p calling ? |
| 17:22.18 | interfaithquest | gtalk and chan_iax both seem to have problems with asterisk behind a nat/firewall |
| 17:23.44 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 17:24.19 | raasdnil | [TK]D-Fender: http://www.pastie.org/315622 |
| 17:24.29 | [TK]D-Fender | interfaithquest: p2p? As in? |
| 17:24.32 | raasdnil | interfaithquest: if you mean * to * through a firewall? yes, I do it all day |
| 17:24.37 | interfaithquest | peer to peer |
| 17:25.18 | [TK]D-Fender | interfaithquest: As opposed to? |
| 17:25.19 | raasdnil | interfaithquest: how is the network setup? * -> FW -> Internet -> FW -> * ? |
| 17:25.36 | interfaithquest | well if both units are behind a nat /firewall then it seems some configuration is needed on the nat/firewall |
| 17:25.47 | [TK]D-Fender | interfaithquest: Try describing your actual call in detail and the networking between |
| 17:25.59 | interfaithquest | a 3rd central server |
| 17:26.05 | [TK]D-Fender | interfaithquest: Yes, IAX has to be forwarded on each router |
| 17:26.28 | interfaithquest | one central server has multiple clients all asterisk behind nat/firewalls |
| 17:26.57 | interfaithquest | the goal is to have the clients call each other directly if possible to avoid audio thru the server |
| 17:27.26 | [TK]D-Fender | interfaithquest: this is not an issue. All you need is for *'s side (or the one that is registered against r receives unsolicited calls) to have IAX forwarded to it |
| 17:27.27 | interfaithquest | gtalk seems to fail as well as iax to setup direct p2p media |
| 17:28.17 | interfaithquest | the central server does see /show the ports that the clients expose to iax |
| 17:28.40 | [TK]D-Fender | interfaithquest: Try describing your actual call in detail and the networking between <--- |
| 17:28.56 | interfaithquest | yet when calling from client1 to client2 via central server the audio still routes via the center |
| 17:29.04 | interfaithquest | even though transfer=yes is in the config file |
| 17:29.20 | interfaithquest | ok |
| 17:29.31 | interfaithquest | the main server is exposed no nat /firewall |
| 17:29.33 | raasdnil | [TK]D-Fender: ^^^ my pastie above looks like the outbound pri is rejecting the call now... |
| 17:29.58 | [TK]D-Fender | raasdnil: Ask your telco what they see wrong. The cause code isn't very informative |
| 17:30.19 | interfaithquest | the asterisk clients are each behind a nat/firewall |
| 17:30.30 | interfaithquest | each client registers with the central server |
| 17:30.49 | [TK]D-Fender | interfaithquest: And you effectively want a "reinvite" between the 2 legs? |
| 17:30.54 | interfaithquest | each client is configured with "transfer=yes" |
| 17:31.01 | interfaithquest | yes |
| 17:31.12 | [TK]D-Fender | interfaithquest: pastebin an actual call attempt at verbose 10 |
| 17:31.14 | [TK]D-Fender | ~pb |
| 17:31.14 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 17:31.16 | [TK]D-Fender | ^^^^^^^^^ |
| 17:31.26 | raasdnil | [TK]D-Fender: well.. thanks for giving it your best shot. I really appreciate it. |
| 17:31.38 | [TK]D-Fender | raasdnil: np |
| 17:32.46 | interfaithquest | ok i will paste the debug from a call attempt to paste bin |
| 17:38.37 | interfaithquest | just did a call from client 1 to client 2 http://www.pastebin.ca/1257057 |
| 17:45.36 | *** join/#asterisk maxxim (n=maxxxim@93-96-96-106.zone4.bethere.co.uk) |
| 17:46.02 | maxxim | i'm getting a lot of such messages, what does it means?: [Nov 15 19:43:07] DEBUG[16726]: channel.c:1146 channel_find_locked: Avoiding initial deadlock for channel '0x826e530' |
| 17:46.14 | [TK]D-Fender | interfaithquest: -- Executing [5256@from-sip:2] Dial("IAX2/ready026-6", "IAX2/ready021/5256|20|tTr") in new stack <- remove the tTr |
| 17:46.34 | *** join/#asterisk dotirc (n=dotirc@97-113-1-157.tukw.qwest.net) |
| 17:46.51 | interfaithquest | will try that |
| 17:47.06 | *** join/#asterisk ManxPower (n=manxpowe@243.sub-75-201-33.myvzw.com) |
| 17:47.30 | dotirc | Does anyone know if an aastra 35i or 480i can accept two incoming calls at once? I'm trying it but getting 486 "Busy Here" |
| 17:47.48 | dotirc | first call works fine, but second call gets that busy message. But its a 4 line phone. |
| 17:48.19 | dotirc | Is there some kind of setting I can change in sip.conf to make it work on two lines? |
| 17:49.30 | Micc | woops. forgot to change my name. |
| 17:49.35 | *** join/#asterisk murdock_ut (n=chatzill@c-98-202-154-226.hsd1.ut.comcast.net) |
| 17:49.41 | Micc | Anyone awake in here? |
| 17:49.52 | *** join/#asterisk snapper14 (n=snapper1@cpc3-reig1-0-0-cust301.hers.cable.ntl.com) |
| 17:50.23 | murdock_ut | HI. |
| 17:51.04 | [TK]D-Fender | Micc: Yes, they can. |
| 17:51.17 | [TK]D-Fender | Micc: they are either misconfigured, or on DND |
| 17:52.09 | maxxim | i'm getting a lot of such messages, what does it means?: [Nov 15 19:43:07] DEBUG[16726]: channel.c:1146 channel_find_locked: Avoiding initial deadlock for channel '0x826e530' |
| 17:53.07 | ManxPower | maxxim: What version of Asterisk? |
| 17:53.16 | snapper14 | Hi, can anyone recommend the best solution for one way audio when going through NAT. If I call my asterisk server directly it works fine, but from the SIP phone registered with asterisk I can only here the incoming audio not the outgoing. Thx |
| 17:53.26 | [TK]D-Fender | ~sipnat |
| 17:53.27 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 17:53.29 | [TK]D-Fender | snapper14: ^^^ read |
| 17:53.31 | ManxPower | snapper14: canreinvite=no in sip.conf |
| 17:54.06 | snapper14 | Thanks [TK]D-Fender I did as per your suggestions a couple of days ago but I still haven't got anywhere. |
| 17:54.14 | maxxim | ManxPower> 1.6.01 |
| 17:54.26 | ManxPower | snap then you have NOT read that document carefully enough. |
| 17:54.27 | Micc | TKD-Fender, I'm checking out this page http://www.voip-info.org/wiki/view/Asterisk+and+Aastra+Phones do you know of any other information I can look at to fix this issue? |
| 17:54.31 | [TK]D-Fender | snapper14: And like happens so often you aren't showing us what you;ve done so we can't tell if you actually did it right |
| 17:54.39 | ManxPower | maxxim: so an unreleased development version? |
| 17:54.40 | [TK]D-Fender | Micc: the MANUAL |
| 17:54.57 | [TK]D-Fender | ManxPower: No, thats released |
| 17:55.19 | ManxPower | [TK]D-Fender: > 1.6.0.1 would mean greater than 1.6.0.1 |
| 17:55.19 | maxxim | ManxPower> what version do you recommend? |
| 17:55.25 | snapper14 | sorry, I thought by explaining what I had tested already it would have shown I had at least tried to do some of my own diag but now I am at a loss. |
| 17:55.33 | ManxPower | maxxim: I suggest trying released version. |
| 17:55.34 | Micc | TKD-Fender, I think I recycled the manual already. |
| 17:55.38 | maxxim | ManxPower> Asterisk 1.6.0.1 seems to be released on www.aterisk.org |
| 17:55.53 | maxxim | there is 1.4.22 and 1.6.0.1 |
| 17:56.01 | ManxPower | snapper14: I have never ever ever seen someone not get NAT working after they CAREFULLY read the aocomputing.net link. |
| 17:56.05 | interfaithquest | wireshark still shows the audio going via the central server, some other tweek ? |
| 17:56.30 | ManxPower | maxxim: Ah, I see. Not "ManxPower greater than 1.6.0.1" |
| 17:56.33 | interfaithquest | transfer=yes is set for each client in iax.conf in the central server |
| 17:56.37 | *** join/#asterisk terracon (n=greisky@CPE001cf097a750-CM0012254076d6.cpe.net.cable.rogers.com) |
| 17:56.41 | maxxim | ManxPower> that debug message means that there is a error with asteris? |
| 17:56.58 | ManxPower | There's a reason most of us use : unstead of > |
| 17:57.03 | *** join/#asterisk bird_of_Luck (n=aurlov@77.73.232.232) |
| 17:57.30 | maxxim | :) i see |
| 17:57.48 | ManxPower | maxxim: There is nothing you can do in the config to fix that message. |
| 17:57.56 | interfaithquest | the iax source code is chalk full of transfer code, strange there is little documented on how to get this 'reinvite' to work |
| 17:58.10 | bird_of_Luck | Does anybody know if I can use SIPADDHEADER(Alert-Info: ...) in calls to cisco 7911G ? It works for 7960/7940 |
| 17:58.13 | snapper14 | Manxpower: The only part of the link I haven't tried is the rtp.conf settings but as it works when directly connecting through to the asterisk server which is itself behind NAT I did not believe this would make a differece. |
| 17:58.19 | maxxim | ManxPower: that message means that something is wrong with asterisk? it means it not work properly? |
| 17:58.21 | *** join/#asterisk TOrrIeri (n=torrieri@nelug/crew/torrieri) |
| 17:58.36 | ManxPower | snapper14: stop thinking and follow the directions. |
| 17:58.47 | *** join/#asterisk darkskiez (n=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) |
| 17:58.57 | [TK]D-Fender | snapper14: You still aren't showing us your configs, and we're trusting them less the more you say you're sure you did it right with each passing second |
| 17:59.44 | ManxPower | maxxim: You will have to search the mailing list archives. It means "two or more parts of Asterisk are trying to access the same thing (usually channel)" If asterisk stop working then you have a problem. If it does not stop working then don't worry about it. |
| 17:59.54 | snapper14 | [TK]D-Fender: I know there is error in the configs you don't need to blatently tell me they are wrong, I wouldn't be here asking for help otherwise. |
| 17:59.59 | ManxPower | [TK]D-Fender: We both know what he did wrong. |
| 18:00.02 | interfaithquest | Fender: perhaps one client is behind a NASTY nat , i will try to setup both clients behind simpl nat devices |
| 18:00.06 | [TK]D-Fender | snapper14: so PASTEBIN them |
| 18:00.07 | ManxPower | snapper14: PASTE your CONFIGS on PASTEBIN.CA |
| 18:00.07 | [TK]D-Fender | ~pb |
| 18:00.08 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 18:00.09 | [TK]D-Fender | ^^^^^^^^^^^^^^ |
| 18:00.36 | ManxPower | [TK]D-Fender: My question is what we should do to him if he did not actually follow the directions to the letter. |
| 18:00.40 | [TK]D-Fender | interfaithquest: So far I see nothing. Why am I going to go on a wild goose chase for retarded possibilities when I don't trust the BASICS? |
| 18:00.42 | snapper14 | I was just doing an updated pastebin afer [TK]D-Fender told me about it earlier |
| 18:00.44 | ManxPower | An example needs to be made. |
| 18:01.02 | [TK]D-Fender | ManxPower: Plenty of examples out there... we've got a pike shortage ;) |
| 18:01.23 | ManxPower | We can just scream at him until he breaks down I guess. |
| 18:01.48 | [TK]D-Fender | ManxPower: Oh... and I don't like the .ca ... doesn't send you to the link so you ahve to follow or target click it to get from the address bar :) |
| 18:01.55 | maxxim | ManxPower: mate, that message did not stop asterisk to work, but i'm trying to find out why my ringback is not working when i'm placing a call after playback message. i've post a message on forum with the debug logs, it will be great if you can have a look, and to tell me your opinion about is: http://forums.digium.com/viewtopic.php?p=120110 |
| 18:02.40 | ManxPower | maxxim: make sure you have a /etc/asterisk/indications.conf (the default one should be fine) |
| 18:02.45 | interfaithquest | http://www.pastebin.ca/1257064 |
| 18:03.13 | Micc | what is BLF? |
| 18:03.21 | [TK]D-Fender | ~blf |
| 18:03.22 | jbot | it has been said that blf is Busy Lamp Field, aka little lights next to speed dials that light up when the person is on the phone and blink when that line is ringing. hint extensions are static mapped to SIP or other channels. |
| 18:03.25 | maxxim | ManxPower: mate, the ringingback works fine if i'm not doing playback or answer before calling Dial cmd... so, my indicaitons.conf file is fine... |
| 18:03.49 | [TK]D-Fender | interfaithquest: Still no-go? |
| 18:03.53 | ManxPower | BTW, DataPilot / Susteen basically provide no access to drivers if your one year support that comes with the product expires. That's right, you can't even download updated USB drivers for the product without buying an upgrade. |
| 18:04.11 | interfaithquest | no call transfer |
| 18:04.19 | ManxPower | maxxim: You see, the CLASSIC reason for people not getting ringback is that the line is answered in the dialplan but there is no indications.conf. |
| 18:04.26 | ManxPower | And I mean like over 90% |
| 18:04.27 | [TK]D-Fender | interfaithquest: Ok, I've hit the limit of my experience with this then... |
| 18:04.29 | maxxim | ManxPower: ringback does work when: Wait,Dial(xxx). ringback does not work when: Wait,Answer,playback,Dial(xxx) |
| 18:04.31 | interfaithquest | perhaps one client is behind a NASTY nat device |
| 18:04.37 | interfaithquest | ok |
| 18:04.46 | ManxPower | maxxim: I'm starting to think someone is lying about indications.conf |
| 18:05.05 | snapper14 | Here you go: http://pastebin.com/m3b505ad6 |
| 18:05.05 | maxxim | ManxPower: i will put my indication conf file in pastebin, 1 sec pls |
| 18:05.23 | [TK]D-Fender | interfaithquest: For your scenario, possible I suppose.. IAX2 is over a single port so the mapping should be active. the worse that I can picture is if the router cares about the origin IP as well... |
| 18:05.24 | ManxPower | once a call is answerd in the dialplan then any indications (ringback, busy, etc) are not sent out of band, but are sent inband and inband is controlled by that file. |
| 18:05.29 | [TK]D-Fender | interfaithquest: Which WOULD be a nasty POS |
| 18:05.45 | [TK]D-Fender | interfaithquest: Can you take NAT out of the picture for any of this? |
| 18:05.59 | interfaithquest | yes i can experiment |
| 18:06.38 | [TK]D-Fender | snapper14: What do you have forwarded to *? |
| 18:06.39 | interfaithquest | the goal though is to have a system , phone service where 90% of the time the 're-invite' will work |
| 18:07.06 | maxxim | ManxPower: http://rafb.net/p/FINmWk55.html |
| 18:07.16 | ManxPower | snapper14: you do not have the canreinvite while CLEARLY says it's IMPORTANT in the natdoc we sent. |
| 18:07.28 | [TK]D-Fender | snapper14: And you have missed a CRUCIAL setting which was announced as such in that guide. |
| 18:07.56 | ManxPower | [TK]D-Fender: why do you think people don't read carefully anymore? |
| 18:08.14 | interfaithquest | Fender, thx for the interest btw, has anyone reported success with chan_gtalk for this kind of p2p calling ? |
| 18:08.49 | ManxPower | snapper14: Why do you think that document exists? It exists because getting NAT and SIP working with Asterisk is a multi-step process where each step must be done correctly. |
| 18:08.51 | maxxim | ManxPower: mate, i don't really understand about 'inbound' , 'outband', can you give me plase an example, or how can i make the ringback to work? thanks! |
| 18:09.22 | interfaithquest | Fender , my tests with chan_gtalk (which uses stun) seem to work ok when one end is google gtalk soft client, but not if both are asterisk nat'd clients |
| 18:09.48 | ManxPower | maxxim: try the one that came in the asterisk source code. |
| 18:09.54 | [TK]D-Fender | "IMPORTANT! phones must not be allowed to attempt to directly connect with each other" <-- Seriously. WTF. No really. WTF. Is this not &^%$#ing blatantly screaming in your face? |
| 18:09.59 | [TK]D-Fender | thinks some bold flashing colour is required for that statement. |
| 18:10.07 | interfaithquest | Fender, so i iax /gtalk fail i may develop a state of the art udp hole punching call setup channel |
| 18:10.32 | ManxPower | I give up. snapper14 I cannot and will not help you further. |
| 18:10.36 | [TK]D-Fender | interfaithquest: Well if its a CPE NAT issue you can't save them... without inventing your own protocol anyways... |
| 18:10.47 | [TK]D-Fender | interfaithquest: what is the IAX endoint in this case? |
| 18:10.55 | ManxPower | maxxim: inband .vs. outof band should be talked about in the Asterisk book. |
| 18:10.57 | [TK]D-Fender | ManxPower: I think he got it... |
| 18:10.57 | interfaithquest | asterisk |
| 18:11.06 | ManxPower | [TK]D-Fender: yeah, this time. |
| 18:11.08 | interfaithquest | embedded asterisk in ixp425 board |
| 18:11.17 | interfaithquest | like the 'slug' |
| 18:11.17 | [TK]D-Fender | interfaithquest: if each remote system is *, then each site should be forwarded and NAT should not be an issue |
| 18:11.32 | [TK]D-Fender | (should not) |
| 18:11.35 | interfaithquest | yes.. life should be like that |
| 18:11.36 | interfaithquest | ha ha |
| 18:11.39 | ManxPower | [TK]D-Fender: I'm so tempted to write an application to auto-config asterisk for nat support. |
| 18:11.46 | snapper14 | Okay lesson learnt. Thats what I get for looking at this things at 1am. Thanks |
| 18:11.59 | ManxPower | you CAN discover your localnet and externip programmically. |
| 18:12.13 | interfaithquest | likely one of the NAT is a NASTY nat.. so i will add another 3rd client with a NICE NAT and re test |
| 18:12.30 | [TK]D-Fender | ManxPower: I got a shell script from docelmo I believe that divines the WAN IP... tiny mod to scan for localnets (non-VPN) to add. |
| 18:12.46 | maxxim | ManxPower: i've just took the indications.conf.sample file from 'configs' directory of source package of asterisk 1.6.0.1. It didn't help. the same situation, i can't hear the ringback in answered channel/// what else can i try? thanks |
| 18:13.44 | ManxPower | maxxim: there is nothing else to try. If indications.conf does not fix it, I've never heard of it being fixed. |
| 18:14.05 | maxxim | :[ |
| 18:15.09 | [TK]D-Fender | ManxPower: As it is I'm going to refine that post a bit. |
| 18:16.20 | ManxPower | [TK]D-Fender: number the steps, bold the config options. I think reformatting is needed. |
| 18:16.40 | ManxPower | maxxim: did you restart asterisk after copying the indications.conf? |
| 18:16.42 | [TK]D-Fender | ManxPower: I'll get around to that. |
| 18:18.09 | maxxim | ManxPower: sure, i'm an it engineer. what i've noticed, is that, the ringback could be heared for just first 500ms, if i'm putting the Dial cmd straight after Playback cmd... |
| 18:18.32 | maxxim | ManxPower: i have such impression, that somethins in asterisk stops ringback to by |
| 18:18.37 | maxxim | ManxPower: i have such impression, that somethins in asterisk stops ringback |
| 18:18.43 | maxxim | sorry |
| 18:19.54 | ManxPower | maxxim: Asterisk will ALWAYS provide ringback if it thinks it should provide the ringback. |
| 18:20.19 | maxxim | ManxPower: how can i track what makes ringback to not by heared? |
| 18:21.47 | ManxPower | maxxim: I don't know. |
| 18:21.55 | Micc | doh! I figured it out. I logged in the agent from the same extension and its using ${CALLERID} |
| 18:22.05 | ManxPower | it could be caused by dozens of issues, and I've already given you the only solution I have ever heard of that works. |
| 18:22.13 | Micc | So it registered multiple agents at the same extension. so of course it would be busy on the second line. |
| 18:22.21 | ManxPower | Micc: Well that's silly! Why not use the newer CALLERID variables? |
| 18:22.29 | maxxim | ManxPower: do you know who can help me? any contact details, pls... |
| 18:22.34 | [TK]D-Fender | we tend to call them FUNCTIONS here. |
| 18:22.35 | ManxPower | maxxim: no. |
| 18:22.40 | ManxPower | try asking in the mailing list. |
| 18:23.07 | maxxim | ManxPower> there could take ages to get an proper respose ( |
| 18:23.17 | maxxim | ManxPower: thans for you time! |
| 18:23.26 | ManxPower | maxxim: you are not going to get this solved today or maybe even in a week. |
| 18:23.55 | Micc | ManxPower, really I shouldn't use the calleridnum at all. |
| 18:24.07 | ManxPower | chances are you'll have to dig into the Asterisk source code and the SIP RFC, etc |
| 18:24.17 | ManxPower | Micc: did you read the upgrade.txt files? |
| 18:24.22 | maxxim | ManxPower: i would like to know if guys from mailinglist or forums are happy to fix bugs, or not really? |
| 18:24.30 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 18:24.35 | ManxPower | they tell you everything that is deprecated and what important changes happened. |
| 18:24.46 | Micc | ManxPower, what upgrade.txt file? |
| 18:24.50 | ManxPower | maxxim: you don't even know if it is a bug in asterisk. |
| 18:25.00 | Micc | ManxPower, I've been running 1.2.11 for years. |
| 18:25.01 | ManxPower | Micc: the one that comes in the Asterisk source code. |
| 18:25.21 | Micc | ManxPower, I'm sure I did at one point. |
| 18:25.43 | Micc | ManxPower, the calleridnum dialplan code I copied off of a website. |
| 18:25.53 | Micc | So if its old or wrong, thats why. |
| 18:26.21 | ManxPower | Micc: *nod* That's why they include the upgrade info, so people can realize there is old code out there. |
| 18:26.41 | ManxPower | so why don't you go read it. You can get the 1.6.0.1 code and it lists the upgrade files for 1.6, 1.4, and 1.2 |
| 18:27.03 | ManxPower | ~manxpower |
| 18:27.03 | jbot | it has been said that manxpower is NOT an employee of Digium. He is looking for a training/teaching job in networking and/or Asterisk. Currently doing Asterisk and WAN consulting. Contact: eric@fnords.org http://www.fnords.org/skillslist.html |
| 18:27.14 | maxxim | ManxPower: even musiconhold is no working in this situation. i was thinking to put a musiconhold sound instead of ringback |
| 18:27.35 | maxxim | using Dial(xxxx,y,m) |
| 18:28.44 | *** join/#asterisk seanmh (n=seanmh@70.90.202.94) |
| 18:29.13 | *** join/#asterisk seaq (n=seaq@98.227.60.190.host.ifxnetworks.com) |
| 18:34.49 | *** join/#asterisk jmacz (n=jmacz@190.26.155.231) |
| 18:47.03 | raasdnil | [TK]D-Fender: I GOT IT WORKING! |
| 18:47.12 | raasdnil | dances |
| 18:47.31 | raasdnil | was the asterisk/dahdi_chan.conf |
| 18:47.37 | raasdnil | pridialplan=unknown |
| 18:47.46 | raasdnil | woohoo! |
| 18:47.47 | raasdnil | now... |
| 18:48.14 | raasdnil | The only problem with post connect DTMF is that it doesn't provide a ringing tone.... any ideas how I put that on? |
| 18:53.01 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 18:54.57 | jaytee | add the r option to the Dial app statement |
| 18:55.30 | raasdnil | perfect! |
| 18:55.32 | raasdnil | thanks jaytee |
| 18:56.00 | jaytee | raasdnil, or you can use m for MusicOnHold until it's connected if you prefer |
| 18:56.58 | *** part/#asterisk snapper14 (n=snapper1@cpc3-reig1-0-0-cust301.hers.cable.ntl.com) |
| 18:58.26 | raasdnil | heh... I tthink my phone users would get too confused with that |
| 18:58.58 | raasdnil | hey.... |
| 18:59.35 | raasdnil | how do I change the country ring tone? Using the US tones right now. |
| 18:59.56 | jaytee | raasdnil, loadzone |
| 19:00.05 | jaytee | in chan_dahdi.conf |
| 19:01.10 | jaytee | using the m option is fun, it's nice to give callers a little dose of Abba to spread evil throughout the world. |
| 19:01.30 | *** join/#asterisk pluesch0r (n=pluesch0@vie-086-059-109-158.dsl.sil.at) |
| 19:01.44 | raasdnil | hehe :) |
| 19:01.58 | raasdnil | hmm... loadzone in chan_dahdi looks like timezone loading |
| 19:02.13 | raasdnil | I changed the default in indications.conf to au.... still seems to be using the US ring tones though |
| 19:03.02 | [TK]D-Fender | tonezone, not timezone |
| 19:03.10 | jaytee | change loadzone=au |
| 19:03.11 | *** join/#asterisk elGuille_wugro2 (n=guillerm@190.220.69.22) |
| 19:03.23 | elGuille_wugro2 | hello everyone. |
| 19:03.27 | jaytee | loadzone tells * what tonezone indications to use |
| 19:04.01 | raasdnil | [TK]D-Fender: did you see? Got the darn thing working! Thanks for your help yeah? |
| 19:04.23 | [TK]D-Fender | raasdnil: Glad to hear |
| 19:04.37 | jaytee | [TK]D-Fender, what was his issue with pri? |
| 19:05.01 | raasdnil | jaytee: it ended up being the NEC system needed to use post connect DTMF. |
| 19:05.11 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 19:05.21 | raasdnil | that was mixed in with a problem of needing pridialplan=unknown |
| 19:06.14 | [TK]D-Fender | jaytee: His PBX passes the DID as DTMF post dial so it pulls up immediate and he needed a Read |
| 19:06.38 | jaytee | ah |
| 19:07.52 | [TK]D-Fender | jaytee: I've seen a few things like that in the euroisdn side of things... |
| 19:08.57 | jaytee | [TK]D-Fender, I feel fortunate living in the US using our standards |
| 19:09.23 | jaytee | [TK]D-Fender, I met a guy from Montreal in class named Claude Klimos, ever met him? |
| 19:09.27 | [TK]D-Fender | jaytee: Lowest Common Denomintor FTW! |
| 19:09.36 | [TK]D-Fender | $USD = Lowest Common Denomination ;) |
| 19:09.44 | jaytee | lol |
| 19:09.53 | [TK]D-Fender | jaytee: Yup, he ran (runs) Aheeva |
| 19:09.59 | jaytee | yep |
| 19:10.01 | jaytee | nice guy |
| 19:10.19 | [TK]D-Fender | jaytee: Met him a while back, and met up with him & Jared last year |
| 19:10.45 | jaytee | Jared's a great guy. really good instructor |
| 19:11.30 | [TK]D-Fender | jaytee: I never actually took any course, I just met up after a a "whats new" type meeting at the tail end of a dCAP training wek |
| 19:11.33 | jaytee | even though I blew the practical I learned so much in class my head is swimming with ideas for enhancing my dialplan now. |
| 19:11.57 | [TK]D-Fender | jaytee: Sounds like you should apply at the LHC ;) |
| 19:12.31 | jaytee | do they use *? |
| 19:12.33 | [TK]D-Fender | jaytee: Yeah "practically" blew up too ;) |
| 19:15.02 | raasdnil | goodnight all |
| 19:15.04 | raasdnil | well... |
| 19:15.08 | raasdnil | morning. It's 6:15am here |
| 19:15.13 | raasdnil | but it works.... :) |
| 19:15.17 | raasdnil | is off to bed |
| 19:15.17 | jaytee | "Press 1 if you wish to smash uranium protons, press 2 if you wish to smash lead ions, press 3 if you wish to create a black hole, press 4 to create strangelets or press 0 to speak to an operator." |
| 19:15.39 | jaytee | nite raasdnil |
| 19:15.53 | raasdnil | thanks [TK]D-Fender |
| 19:17.05 | elGuille_wugro2 | well, hello everyone again. I'm newbie using asterisk, and i'd like to know how should i get asterisk working with mysql, users, cdr, extensions and so on.!., is there anything you could tell me to read?, i did google but without nothing complete working. |
| 19:17.16 | [TK]D-Fender | ~book |
| 19:17.16 | jbot | rumour has it, book is probably 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://tfot.leifmadsen.com or see ~buybook |
| 19:17.18 | [TK]D-Fender | ^^^^^^^^^ |
| 19:18.38 | jaytee | elGuille_wugro2, search the WIKI at voip-info.org, there's a howto for using mysql with * for CDR, plus you'll need the asterisk-addons compiled. |
| 19:19.17 | elGuille_wugro2 | i compiled and installed asterisk-addons few minutos ago.!. |
| 19:19.21 | elGuille_wugro2 | jaytee: thanks!. |
| 19:19.37 | jaytee | elGuille_wugro2, yw |
| 19:39.30 | [TK]D-Fender | My next challenge : ML-PPPoE. First on a single link, and then I'm thinking of Bonding 2 links |
| 19:45.52 | *** join/#asterisk zippytech2 (n=chatzill@node73.33.251.72.1dial.com) |
| 19:56.05 | *** join/#asterisk DarkRift (n=dark@65.92.169.191) |
| 19:57.18 | *** join/#asterisk outtolunc (n=me@c-67-188-204-139.hsd1.ca.comcast.net) |
| 20:00.36 | *** join/#asterisk jov4n (n=jovan@87.19.104.34) |
| 20:00.37 | *** join/#asterisk ManxPower (n=manxpowe@70.sub-70-223-194.myvzw.com) |
| 20:00.39 | jov4n | Hi |
| 20:03.31 | *** join/#asterisk kisu (n=kvirc@daniel1117.broker.freenet6.net) |
| 20:03.50 | *** join/#asterisk feeds (n=feeds@85-135-225-22.adsl.slovanet.sk) |
| 20:07.56 | zippytech2 | anyone remember how to enable video with asterisk |
| 20:09.26 | jaytee | videosupport=yes in the general section of sip.conf |
| 20:09.43 | ManxPower | zippytech2: Video with Asterisk is not a common thing yet. You might want to check the mailinglist archives ("/msg jbot ~mailinglist" for more information) or ask on the asterisk-users mailinglist. |
| 20:09.51 | jaytee | and add support for the h263 codec with allow=h264 |
| 20:10.04 | jaytee | oops, allow=h263 |
| 20:10.09 | ManxPower | There should be some docs in the doc directory of the Asterisk source code too. |
| 20:11.48 | zippytech2 | i have a box setup but i don't have access from here to see the changes if i remember it was some what simple |
| 20:12.14 | zippytech2 | like to lines added to sip.conf |
| 20:12.27 | jaytee | zippytech2, all I needed to do to get a webcam and x-lite working with video was what I just mentioned |
| 20:12.34 | jaytee | ^^^^^^^ |
| 20:18.17 | *** join/#asterisk seaq (n=seaq@98.227.60.190.host.ifxnetworks.com) |
| 20:19.18 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 20:21.03 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 20:22.24 | SkramX | ssh phalse |
| 20:22.25 | SkramX | wwops |
| 20:29.55 | *** join/#asterisk javb (n=javb@pool-68-162-58-218.nwrk.east.verizon.net) |
| 20:29.58 | *** join/#asterisk xnixan (n=xnixan@unaffiliated/xnixan) |
| 20:42.08 | *** join/#asterisk root52 (n=root52@ip70-191-120-39.cl.ri.cox.net) |
| 20:44.46 | *** join/#asterisk hi365_m (n=hi365@213.151.36.90) |
| 20:44.55 | *** join/#asterisk Micc (n=dotirc@97-113-1-157.tukw.qwest.net) |
| 20:45.11 | Micc | I'm getting log rotate errors over and over. |
| 20:45.35 | Micc | Rotated Logs Per SIGXFSZ (Exceeded file size limit) |
| 20:46.07 | *** join/#asterisk Cresl1n (n=matt@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 20:46.07 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 20:46.55 | Micc | I have plenty of disk space left. |
| 20:47.43 | Micc | Also, I have a 35i and a 480i aastra with the cordless handset for each. The handset for the 480i doesn't answer when picked up but the 35i handset answers the ringing line when picked up from the charging cradle. |
| 20:48.04 | Micc | Does anyone know if there is a way to turn the auto-answer when picked up off? |
| 20:48.57 | [TK]D-Fender | Micc: You desperately need to read your phone's manual... |
| 20:49.11 | Micc | I don't know where it is. |
| 20:49.19 | [TK]D-Fender | Micc: www.aastra.com <- |
| 20:50.24 | drmessano | hmmm |
| 20:51.34 | Micc | good call. |
| 20:52.04 | *** join/#asterisk Jelle__ (n=jelle@efg.xs4all.nl) |
| 20:54.18 | jaytee | drmessano, PM? |
| 20:55.01 | drmessano | sure |
| 20:55.30 | jaytee | i tried to send a PM request but this version of mIRC is acting strange, it's waiting for your auth |
| 20:55.36 | drmessano | hmmm |
| 20:56.23 | *** join/#asterisk Maliuta (n=scooby@kiev.lusan.id.au) |
| 20:59.22 | Micc | asteirsk keeps crashing now. |
| 20:59.33 | Micc | my log files are recycling . |
| 20:59.41 | Micc | bunch of event_log.# |
| 21:00.35 | [TK]D-Fender | Micc: start * manually ans see what comes up |
| 21:01.19 | *** join/#asterisk SkramX (i=mark@phalse.2600.COM) |
| 21:06.31 | Micc | I'm getting all kinds of errors. |
| 21:06.44 | Micc | what could be causing this state? |
| 21:08.32 | drmessano | app_fukdup |
| 21:08.35 | Micc | chan_zap.so failed loading. |
| 21:08.41 | drmessano | Same reason your box has been crashing |
| 21:08.50 | drmessano | Check for hard disk errors |
| 21:08.58 | [TK]D-Fender | Micc: Maybe you forgot to start Zaptel FIRST |
| 21:09.15 | [TK]D-Fender | Micc: Maybe you should show us the problem before asking WHY |
| 21:09.16 | Micc | its been running all day just fine. |
| 21:09.31 | drmessano | You've been complaining for days that its been crashing as of late |
| 21:09.36 | [TK]D-Fender | Micc>its been running all day just fine. <- high on the list of least informative statements one could provide |
| 21:09.50 | drmessano | "All day" is meaningless unless you *FOUND* and *FIXED* something |
| 21:10.06 | drmessano | But you stated several times the box has been hosed |
| 21:10.38 | drmessano | You're either gonna keep looking for an asterisk problem, living in denial, or start checking the hardware before it dies and you go into "oh shit" mode |
| 21:11.22 | [TK]D-Fender | or... at the very least SHOW US THE PROBLEM. |
| 21:11.33 | drmessano | Your statements have been to the effect of "its been working fine for two years and after I gave my IP address out in here, asterisk crashes all the time" |
| 21:11.49 | Micc | asterisk died with code 1 automatically restarting |
| 21:12.10 | drmessano | Now you're getting random crashes |
| 21:12.22 | interfaithquest | bindport in iax if set to 4570.. fails so show as 4570 when checking via 'iax2 show provisioning' |
| 21:12.28 | [TK]D-Fender | Micc: It can't automatically restart is you started it MANUALLY like I asked |
| 21:12.35 | interfaithquest | <PROTECTED> |
| 21:12.50 | Micc | I did start it manually. /usr/sbin/safe_asterisk |
| 21:13.00 | Micc | I also tried /usr/sbin/asterisk -vvvc |
| 21:13.01 | [TK]D-Fender | Micc: that is to run in DAEMON, not MANULA |
| 21:13.10 | [TK]D-Fender | Micc: Only as "asterisk -gvvvvvvvvvvc |
| 21:13.11 | drmessano | interfaithquest try bindaddr=0.0.0.0:4570 |
| 21:13.20 | interfaithquest | good idea |
| 21:13.21 | interfaithquest | thx |
| 21:13.27 | *** join/#asterisk pluesch0r (n=pluesch0@vie-086-059-083-064.dsl.sil.at) |
| 21:15.57 | *** join/#asterisk root52 (n=root52@ip70-191-120-39.cl.ri.cox.net) |
| 21:16.44 | Micc | http://www.pastebin.ca/1257187 |
| 21:18.00 | *** join/#asterisk ElSonico (n=tav@dondo.ampiainen.net) |
| 21:18.14 | *** join/#asterisk ElSonico (n=tav@dondo.ampiainen.net) |
| 21:20.30 | interfaithquest | that also fails :( |
| 21:20.50 | interfaithquest | it seems iax wont take a different port |
| 21:20.59 | interfaithquest | any other suggestions ? |
| 21:21.11 | [TK]D-Fender | Micc: Yup, looks like zaptel hasn't been started first. |
| 21:21.30 | [TK]D-Fender | Micc: do "ztcfg -vvvv". If that checks out, try starting * again |
| 21:22.01 | interfaithquest | hard code the ip:port ? |
| 21:22.47 | interfaithquest | as it is.. for 1.4.2 it is not taking 4570 when 0.0.0.0:4570 is set |
| 21:23.40 | interfaithquest | Fender, here the idea is to have 2 iax * clients behind a simple NAT to test iax call transfer and p2p calling |
| 21:24.07 | interfaithquest | Fender, strange iax will not take a the bindaddr=0.0.0.0:4570 |
| 21:25.28 | drmessano | 1.4.2? |
| 21:25.31 | drmessano | Oh god |
| 21:25.45 | drmessano | Get something about 2 years newer or so |
| 21:25.47 | drmessano | and then try it |
| 21:26.14 | *** join/#asterisk RobertLaptop (n=rmiddle@m9e0736d0.tmodns.net) |
| 21:26.16 | pids | interfaithquest, the bind address and the port are not set together |
| 21:26.18 | *** join/#asterisk boolean12 (n=random@c-98-199-172-11.hsd1.tx.comcast.net) |
| 21:26.27 | Micc | how can I reload zap? |
| 21:26.32 | pids | bindaddr=0.0.0.0 port=4570 |
| 21:26.43 | drmessano | He tried that |
| 21:26.53 | drmessano | and the suggestion was made based on newer behavior |
| 21:26.54 | Micc | ztcfg -vvv looks normal. |
| 21:27.00 | Micc | 24 channels configured |
| 21:27.02 | [TK]D-Fender | Micc: I just told you something very specific to do. Do it. |
| 21:27.23 | boolean12 | Can anyone verify or disprove that 1.6 does not allow setting a different context with realtime? |
| 21:27.43 | Micc | SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) |
| 21:27.49 | interfaithquest | should be an easy bug to fix.. in chan_iax2.c i guess |
| 21:27.56 | Micc | I think I've run out of memory. |
| 21:28.52 | drmessano | interfaithquest: Why are you using something so old? |
| 21:29.10 | pids | interfaithquest, I have it set to diffrent ports and have no problems. So I doubt its a bug. |
| 21:29.16 | drmessano | You're gonna have more problems with 1.4.2 than not |
| 21:29.54 | drmessano | Especially with IAX2 |
| 21:29.58 | Micc | I have 0 shared buffers free |
| 21:30.10 | interfaithquest | so ? move up to 1.6 ? |
| 21:30.16 | drmessano | .... |
| 21:30.26 | interfaithquest | 1.6 has troubles with linux 2.4 |
| 21:30.38 | drmessano | At the very least get a non-ancient 1.4 |
| 21:30.41 | drmessano | 1.4.2 IS OLD |
| 21:30.42 | [TK]D-Fender | ... |
| 21:30.45 | drmessano | 1.4.22 is current |
| 21:30.52 | drmessano | Why are 1.4.2 and 1.6 your only options |
| 21:30.54 | interfaithquest | oops yes 1.4.22 |
| 21:31.06 | interfaithquest | the client is on the latest 1.4.22 |
| 21:31.20 | jaytee | or a newer version of 1.4, current is 1.4.22 but ya might have issues there with a 2.4 kernel |
| 21:31.37 | interfaithquest | and still fails to take a bind other than 4569 |
| 21:32.06 | jaytee | pastebin your iax.conf file |
| 21:32.09 | *** join/#asterisk harry_v (n=pcsuppor@S0106001d7e52cc78.vs.shawcable.net) |
| 21:32.11 | interfaithquest | hmm calling is ok in general |
| 21:32.12 | interfaithquest | ok |
| 21:32.14 | pids | interfaithquest, your not using a dns srv record are you? |
| 21:32.37 | interfaithquest | not using dns in these tests |
| 21:32.39 | drmessano | Do you have bindaddr specified? |
| 21:32.50 | interfaithquest | just ip |
| 21:32.51 | interfaithquest | no |
| 21:32.55 | drmessano | Do that |
| 21:33.01 | drmessano | bindaddr=0.0.0.0 |
| 21:33.02 | interfaithquest | hard code some ip ? |
| 21:33.03 | drmessano | then your port |
| 21:33.07 | drmessano | No |
| 21:33.09 | interfaithquest | yes i tried that |
| 21:33.17 | interfaithquest | here i will pastbin it |
| 21:33.19 | drmessano | Well, if its going to work, you need that |
| 21:33.24 | jaytee | you should have bindport=4570 and then bindaddr=0.0.0.0 |
| 21:33.40 | drmessano | bindaddr needs to be before bindport too |
| 21:33.42 | jaytee | in that order in [general] in iax.conf |
| 21:33.50 | jaytee | before? |
| 21:34.05 | harry_v | I do not know if there has been some changes in the way vm behaves but getting this message stating that I have no entry in voicemail config file. |
| 21:34.11 | drmessano | Because asterisk reads bindaddr then the bindport to associate with the specified address |
| 21:34.21 | [TK]D-Fender | harry_v: DETAILS man... |
| 21:34.32 | drmessano | you can supply subsequent bindaddrs and bindports that way |
| 21:35.18 | interfaithquest | http://www.pastebin.ca/1257201 |
| 21:35.28 | harry_v | http://pastebin.ca/1257202 |
| 21:35.32 | drmessano | bindaddr FIRST |
| 21:35.43 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
| 21:35.43 | drmessano | and no |
| 21:35.47 | drmessano | bindaddr=0.0.0.0 |
| 21:35.52 | drmessano | bindport=4570 |
| 21:35.54 | drmessano | Like that |
| 21:36.03 | interfaithquest | ok easy to try |
| 21:36.04 | harry_v | some topics in google briefly talk about it and made the changes but no go. |
| 21:36.10 | [TK]D-Fender | harry_v: DETAILS man... |
| 21:36.50 | harry_v | call another extention and it falls though. Do you need to see the voicemailconf file? |
| 21:37.09 | drmessano | Hang on |
| 21:37.18 | [TK]D-Fender | harry_v: Show us the PROBLEM |
| 21:37.23 | pids | fender likes details |
| 21:37.28 | drmessano | This is like a soup sandwich |
| 21:37.43 | [TK]D-Fender | pids: "it doesn't work, HAEEELELPP!?!?! plAESSEZ?! |
| 21:37.45 | drmessano | interfaithquest: Put bindaddr=0.0.0.0=4570 and REMOVE THE BINDPORT |
| 21:37.48 | drmessano | err |
| 21:37.48 | drmessano | shitr |
| 21:37.52 | pids | hahaha |
| 21:37.56 | drmessano | bindaddr=0.0.0.0:4570 |
| 21:38.11 | *** join/#asterisk ManxPower (n=manxpowe@70.sub-70-223-194.myvzw.com) |
| 21:38.40 | drmessano | It shows it right in the sample file |
| 21:38.45 | drmessano | _----> |
| 21:38.54 | drmessano | I think in 1.2 it was addr before port |
| 21:38.56 | *** join/#asterisk kisu (n=kvirc@daniel1117.broker.freenet6.net) |
| 21:38.56 | drmessano | or so I read |
| 21:39.01 | pids | harry_v, pastbin your voicemail.conf without pins |
| 21:39.02 | jaytee | get rid of the :4570 and add bindport=4570 before the bindaddr= statement |
| 21:39.13 | drmessano | jaytee: its in the sample file |
| 21:39.14 | jaytee | port, then address |
| 21:39.16 | [TK]D-Fender | pids: Wrong answer... please try again |
| 21:39.27 | pids | drmessano, the combined format never worked right |
| 21:39.55 | drmessano | well, right now NOTHING is working for him, so if you have a better idea, start typing |
| 21:40.04 | pids | Fender huh? |
| 21:40.22 | drmessano | otherwise, youre just playing armchair quarterback |
| 21:40.41 | drmessano | interfaithquest: try as Jaytee suggested |
| 21:40.53 | drmessano | bindport=4570 then bindaddr=0.0.0.0 |
| 21:40.54 | Micc | what does .... warning, flexibel rate not heavily tested mean? |
| 21:41.04 | pids | drmessano, we have been trying to get him to dump the combined format for ten minutes. |
| 21:41.05 | interfaithquest | no luck |
| 21:41.22 | drmessano | pids, I have been trying to get him working for 20 |
| 21:41.24 | interfaithquest | all combinations fail to take a port other than 4569 |
| 21:41.36 | [TK]D-Fender | pids: he says its now saying "No VM Entry". Well voicemail.conf has never changed the core format for boxes. This is a DIALPLAn error unless he literally typo'd it. Then again you won't even know WHICH ONE unless you see the dialplan anyways |
| 21:41.38 | interfaithquest | using 1.4.22 and new fedora 9 |
| 21:41.47 | file | interfaithquest: are you reloading or restarting Asterisk? |
| 21:41.53 | [TK]D-Fender | pids: So asking for VM config = largely worthless when I KNOW the dialplan app HAS changed |
| 21:41.57 | jaytee | are you doing an IAX2 reload after changing your iax.conf file? |
| 21:41.57 | interfaithquest | will try restart |
| 21:42.04 | file | you can't change it and reload, you have to restart |
| 21:42.11 | file | or unload chan_iax2.so and load it |
| 21:42.23 | Micc | I have too many log files. |
| 21:42.31 | drmessano | oh shit |
| 21:42.34 | interfaithquest | restart also fails |
| 21:42.34 | jaytee | oh, right. only if change user info |
| 21:42.37 | drmessano | oh god |
| 21:42.39 | drmessano | Jaytee |
| 21:42.41 | interfaithquest | nothing can change the iax2 default port |
| 21:42.42 | jaytee | port stuff needs a restart |
| 21:42.44 | pids | fender good ppint |
| 21:42.45 | baliktad | what is the significance of the number * puts in the userfield of the CDR? |
| 21:42.46 | drmessano | JAYTEE |
| 21:42.51 | jaytee | what? |
| 21:43.01 | interfaithquest | just did a restart |
| 21:43.05 | interfaithquest | on a pc |
| 21:43.06 | drmessano | Im such a fucking dumbass.. and I told you.. I told you I was.. I swore up and down "I, am a dumbass" |
| 21:43.17 | ManxPower | I can't imagine why you would want to change the IAX2 port number. |
| 21:43.22 | interfaithquest | hmm |
| 21:43.27 | file | interfaithquest: I just did and it works fine, netstat shows listening on both 4569 and 4570 |
| 21:43.32 | jaytee | drmessano, we're all bozos on this bus :-) |
| 21:43.36 | drmessano | um.. I dont think I restarted asterisk after adding the TCP stuff |
| 21:43.43 | drmessano | NO SERIOUSLY |
| 21:43.44 | interfaithquest | oh ? and iax2 show provisioning ? |
| 21:43.47 | drmessano | I dont think I did |
| 21:43.51 | drmessano | HA |
| 21:43.56 | file | iax2 show provisioning is for provisioning IAXy devices |
| 21:44.11 | jaytee | ZOMG! :-) |
| 21:44.26 | interfaithquest | using two client devices behind ONE dsl device |
| 21:44.42 | drmessano | Just hit me when file mentioned restarting.. Since shit like bindaddr and whatnot hate a simple reload |
| 21:44.52 | file | interfaithquest: iax2 show provisioning has *nothing* to do with the local machine's bindings |
| 21:44.54 | harry_v | http://pastebin.ca/1257206 my voicemail issue. |
| 21:44.57 | drmessano | Im like "Hmm.. did I remember that... doubtful" |
| 21:45.00 | interfaithquest | ok |
| 21:45.17 | pids | interfaithquest, how are you testing the port? |
| 21:45.22 | interfaithquest | so then how can we know that 4570 is working |
| 21:45.28 | interfaithquest | it fails to register |
| 21:45.32 | pids | interfaithquest, telnet to the port |
| 21:45.33 | Nugget | telnet is eeeeeeevil! |
| 21:45.38 | file | telnet won't work, it's UDP |
| 21:45.46 | [TK]D-Fender | pids: Strike 2! |
| 21:45.47 | pids | its iax |
| 21:46.03 | drmessano | there you go.. argue with a dev now |
| 21:46.06 | interfaithquest | the device registers when set to 4569 |
| 21:46.15 | interfaithquest | and fails when set to 4570 |
| 21:46.18 | Micc | where is the code the does the recycling of log files? |
| 21:46.23 | file | interfaithquest: confirm that it is indeed binding to 4570 by looking at the output of chan_iax2 when it is loaded |
| 21:46.27 | ManxPower | You can't telnet to a DUP port |
| 21:46.27 | drmessano | file: What the hell do you know about asterisk anyway? |
| 21:46.39 | [TK]D-Fender | pids: OSI failure... |
| 21:46.42 | interfaithquest | right |
| 21:46.43 | file | interfaithquest: or netstat -a to confirm the UDP port is being listened |
| 21:46.44 | [TK]D-Fender | :p |
| 21:47.12 | jaytee | drmessano, file wrote the lumenvox connector. I suspect he knows alot about * :-) |
| 21:47.19 | file | if it shows that port 4570 is being listened on then do a tcpdump of that port to see if you see the traffic |
| 21:47.25 | Micc | I'm almost positive this is the problem. my log directory has hundreds of files all 40 or 78 bytes. |
| 21:47.46 | ManxPower | Micc: directories do have a max file number limit |
| 21:47.49 | drmessano | jaytee: I dunno, my money is on pids |
| 21:48.06 | *** join/#asterisk subdolus (n=subdolus@subby.afraid.org) |
| 21:48.16 | Micc | but why would it be doing this in the first place? |
| 21:48.28 | ManxPower | Micc: um, logger rotate |
| 21:48.30 | drmessano | jaytee: I have popcorn and 100:1 odds.. Not afraid |
| 21:48.48 | Micc | ManxPower, where is that code? what file is that in? |
| 21:49.00 | interfaithquest | netstat -a | grep 4570 -> nothing |
| 21:49.26 | file | interfaithquest: and you have it configured in iax.conf how? |
| 21:49.33 | ManxPower | Micc: I dunno, it's in the linux kernel and in the filesystem drivers. |
| 21:49.49 | interfaithquest | hmm for a simple NAT.. perhaps two units on 4569 is ok |
| 21:49.59 | Micc | I found it. its in reload_logger in logger.c |
| 21:50.06 | interfaithquest | as each will get a different outside port |
| 21:50.08 | interfaithquest | will try that |
| 21:50.20 | file | interfaithquest: right... |
| 21:50.38 | file | local bindings have no influence over the implementation of the remote device doing the NAT translation... |
| 21:50.56 | ManxPower | interfaithquest: You understand that all connections have a SOURCE port in addition to a destination port, right? |
| 21:52.06 | ManxPower | Proper understanding of NAT requires you understand the source ip/port and dest ip/port of tcp and udp connections. |
| 21:54.46 | ManxPower | ponders a rant about people trying to do VoIP but not understanding basic TCP/IP concepts. |
| 21:55.00 | pids | hmm looks like my sarcasm failed... |
| 21:55.19 | file | offers [TK]D-Fender a drink |
| 21:55.39 | [TK]D-Fender | file: tHANKS, BUT i'VE ALREADY SAVED MYSELF A LOT OF GREIF BY BACKING OFF THIS ONE :) |
| 21:55.44 | [TK]D-Fender | dang caps! |
| 21:55.45 | [TK]D-Fender | :) |
| 21:55.46 | file | [TK]D-Fender: ha |
| 21:56.05 | [TK]D-Fender | file: At work working on budget files \o? |
| 21:56.08 | [TK]D-Fender | file: At work working on budget files \o/ |
| 21:56.13 | [TK]D-Fender | last arm is in question! |
| 21:56.16 | ManxPower | You can lead a whore to culture, but you can't make her think. |
| 21:56.22 | file | [TK]D-Fender: Icanhazmoney? |
| 21:56.40 | [TK]D-Fender | ManxPower: My #^%$ing yogurt has more culture, even if its only BACTERIAL :p |
| 21:56.53 | [TK]D-Fender | file: Right now we're not sure WE have money ;) |
| 21:57.42 | drmessano | I took whore to culture in high school.. I learned the hard was to leaf them alone. |
| 21:57.49 | drmessano | way* |
| 21:58.31 | Micc | ok it is still looping. |
| 21:58.43 | Micc | its restarting logger |
| 21:58.48 | ManxPower | I've spent much of the day and most of the night cursing Novatel Wireless. |
| 21:59.06 | harry_v | some startup company? |
| 21:59.33 | ManxPower | they make a large number of the CDMA USB/PCMCIA cards for Verizon/Sprint/Alltel |
| 21:59.36 | jaytee | I spend half my time cursing Verizon Wireless and the other half cursing AT&T |
| 22:00.35 | ManxPower | they also seem to think an SDK is a few bits of sample code, missing header files, instructions are for a different platform and it's in C++ |
| 22:00.38 | harry_v | my peeve is bell |
| 22:01.05 | Micc | Asterisk Event Logger restarted |
| 22:01.05 | Micc | Asterisk Queue Logger restarted |
| 22:01.05 | Micc | Rotated Logs Per SIGXFSZ (Exceeded file size limit) |
| 22:01.07 | harry_v | gone are the days when cell phones has bigger tranciver modules in them. |
| 22:01.32 | Micc | I keep getting that over and over, I can't understand why it keeps doing it. |
| 22:01.45 | harry_v | more cell sites but gone is the range that one would expect 10-15 years ago. |
| 22:01.48 | ManxPower | Micc: SIGXFSZ is generated by the filesystem, logger is just printing what it got. |
| 22:02.08 | ManxPower | now why don't you just delete a couple of thousand files from that directory and move on. |
| 22:02.26 | Micc | ManxPower, I've deleted every file in that directory a number of times. |
| 22:02.30 | ManxPower | like configuring your system logger to delete the logs after x amount of time. |
| 22:02.37 | ManxPower | Micc: and are they coming back? |
| 22:02.43 | Micc | yes. |
| 22:02.49 | ManxPower | and what do they contain? |
| 22:03.53 | Micc | hmmm.. this is interesting... queue_log contains a ton of CONFIGRELOAD |
| 22:04.05 | Micc | So why its reloading I don't know. |
| 22:04.11 | ManxPower | looks like something could be issuing a zillion reloads. |
| 22:04.22 | Micc | yeah. but what? |
| 22:04.30 | ManxPower | Is this a GUI intall? |
| 22:05.42 | [TK]D-Fender | Micc: You have amillion of those because * was crashing like a 747 due to your zaptel issue, back to back because of running it in safe_asterisk. |
| 22:05.57 | [TK]D-Fender | Micc: When i told you to do it MANUALLY |
| 22:06.07 | [TK]D-Fender | wonders why people don't listen to him... |
| 22:06.31 | [TK]D-Fender | file: I'll take you up on the drink now! |
| 22:06.39 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 22:06.43 | Micc | TKD-Fender, no, it started doing the million restarts before that. |
| 22:06.47 | Micc | And now after that. |
| 22:07.01 | Micc | Its still getting the reloads even while running manually. |
| 22:07.17 | jaytee | doesn't wonder why people don't listen to [TK]D-Fender, he knows it's because most people are either stupid or stubborn or both. |
| 22:08.15 | drmessano | All 3 |
| 22:09.26 | jaytee | bbiab |
| 22:09.40 | [TK]D-Fender | goes back to budget work... |
| 22:10.23 | Micc | I ran out of file descriptors, thats what caused the zap problem. |
| 22:10.37 | Micc | I rebuilt logger.c to not do the recycle. |
| 22:11.21 | pids | Micc I gotta ask, Why?!? |
| 22:11.53 | Micc | pids, because the constant recycling was causing it to eat up file descriptors. As soon as I figure out what the real problem is I'll put it back. |
| 22:15.45 | *** join/#asterisk etfonhomey (n=chatzill@74-143-196-254.static.insightbb.com) |
| 22:15.54 | *** join/#asterisk Strom_C (n=strom@netblock-208-127-61-171.dslextreme.com) |
| 22:16.25 | pids | is convinced that DTMF's are the work of the devil... |
| 22:23.21 | Micc | I found someone else having the same problem online. |
| 22:23.39 | Micc | And it turned out to be Master.csv was too big. I don't have that file anywhere. |
| 22:23.49 | zamba | hehe |
| 22:25.36 | *** join/#asterisk mvanbaak (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 22:27.02 | *** join/#asterisk mvanbaak (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 22:28.18 | Micc | could SIGXFSZ be caused by a bad hard disk? |
| 22:28.56 | tompaw | hi |
| 22:30.06 | tompaw | I have two asterisk boxes: ONE and TWO. On ONE there are two user accounts: 6969 and 6971. On TWO there is one user account - 6971. When I try to make a call from 6971@TWO to 6969@ONE, box ONE says: |
| 22:30.40 | tompaw | Found user '6971' for '6971' |
| 22:30.58 | tompaw | It looks like it's trying to authenticate a remote user basing on the fact that he has the same login. |
| 22:31.14 | tompaw | And it sends back SIP/2.0 401 Unauthorized |
| 22:31.39 | tompaw | Now why would it try to authenticate a REMOTE user from a DIFFERENT domain? |
| 22:31.48 | *** join/#asterisk kisu (n=kisu@daniel1117.broker.freenet6.net) |
| 22:34.26 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 22:34.26 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 22:35.13 | *** join/#asterisk kisu (n=kisu@daniel1117.broker.freenet6.net) |
| 22:40.13 | *** join/#asterisk pluesch0r (n=pluesch0@vie-086-059-124-149.dsl.sil.at) |
| 22:49.09 | *** join/#asterisk Dr-Linux|home (n=Nothing@221.132.117.17) |
| 23:01.15 | *** part/#asterisk elGuille_wugro2 (n=guillerm@190.220.69.22) |
| 23:01.58 | Micc | I think I've isolated it down to the voicemail app |
| 23:05.03 | *** join/#asterisk etfonhomey (n=chatzill@74-143-196-254.static.insightbb.com) |
| 23:06.35 | *** part/#asterisk Cresl1n (n=matt@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 23:09.35 | root52 | Hey all I am blasting about 40 voicemail boxes with the same message. It the inturn send out 40 pages all at once. Is there some asterisk option to stagger thease alerts out a bit. Or is it just going to send them as the message hits the box? |
| 23:11.14 | harry_v | you mean like a broadcast message |
| 23:12.27 | root52 | yes |
| 23:13.26 | root52 | I just have it set up to take one message and send it to 40 mailboxes. ex. (Voicemail(100&1&2&3&4&5.....)) |
| 23:13.34 | *** join/#asterisk snapper14 (n=snapper1@cpc3-reig1-0-0-cust301.hers.cable.ntl.com) |
| 23:14.30 | *** join/#asterisk interfaithquest (n=chatzill@ip67-88-184-130.z184-88-67.customer.algx.net) |
| 23:14.41 | [TK]D-Fender | root52: No, * will send them as quick as you see... not an option |
| 23:15.01 | root52 | ok. Thanks. Just wondering if * could do that or I have to think of something else. Thanks!! |
| 23:15.45 | Micc | Is event_log suppose to be empty? |
| 23:16.08 | interfaithquest | hello 2 * clients call ok via iax, but fail to connect audio peer to peer |
| 23:16.32 | ManxPower | Micc: it should be whatever you configured in logger.conf |
| 23:16.54 | interfaithquest | both iax clients are * units that register to a central server ok via iax |
| 23:16.54 | ManxPower | interfaithquest: any nat involved? |
| 23:17.02 | interfaithquest | yes 1 dsl |
| 23:17.07 | Micc | I didn't configure events, but it keeps saying asterisk event logger restarted and then asterisk queue logger restarted. |
| 23:17.31 | interfaithquest | both clients are on the same dsl nat/firewall one exposed ast 4569 and the other as 1024 |
| 23:17.51 | ManxPower | interfaithquest: unless you portforward in the natted one "peer to peer audio" will not work. |
| 23:18.38 | Micc | When I noload=app_voicemail.so it almost completely stops the SIGXFSZ and reloads. |
| 23:18.56 | interfaithquest | well i can try that.. however if the NAT is a simple NAT.. then audio should connect via the exposed ports |
| 23:19.31 | ManxPower | hmm? |
| 23:19.40 | interfaithquest | so yes i can port forward 1024 and 4569 to test this |
| 23:19.52 | ManxPower | interfaithquest: you don't forward the other ports |
| 23:20.03 | interfaithquest | for a simple NAT there is no need for ANY port forwarding for iax |
| 23:20.04 | ManxPower | the other ports are DYNAMICALLY FORWARDED BY NAT. |
| 23:20.09 | ManxPower | That's the whole point of NAT |
| 23:20.17 | interfaithquest | yes |
| 23:20.17 | boolean12 | Using Asterisk RT in 1.6, Is it true it will ignore a context with the switch => ? |
| 23:20.20 | ManxPower | interfaithquest: there is if you want direct peer audio to work |
| 23:21.02 | ManxPower | also if Asterisk is on a private IP |
| 23:21.02 | interfaithquest | yes..the goal is to have a group of asterisk devices register with a central server, and then call peer to peer |
| 23:21.02 | ManxPower | so forward port 4569 UDP on the NAT box to the internal port 4569 on your IAX2 device. |
| 23:21.04 | interfaithquest | the central server does the call setup ..and hands off the media peer to peer |
| 23:21.09 | ManxPower | interfaithquest: any peer to peer will require port forwarding |
| 23:21.36 | interfaithquest | i can easily TRY that.. in this test both devices are behind the SAME dsl/nat/firewall |
| 23:21.55 | ManxPower | Did you believe that crap about "SIP being so much harder to configure with NAT than IAX2?" |
| 23:22.11 | interfaithquest | one is exposed ast port 1024 and the other as 4569 , shown in the central * iax2 show peers |
| 23:22.22 | interfaithquest | ha ha |
| 23:22.33 | ManxPower | interfaithquest: if they are on the same lan then there is no nat involved. |
| 23:22.48 | interfaithquest | i am considering making a HOLE PUNCH CHANNEL.. to 1st punch thru peer to peer then let the iax take over |
| 23:23.02 | ManxPower | and yet you say "exposed as" so that sounds like nay IS involved. |
| 23:23.08 | interfaithquest | but before reinventing any code.. i want to try what exists |
| 23:23.25 | interfaithquest | the DSL has a NAT/FIREWALL turned ON |
| 23:23.39 | ManxPower | but if they are on the same lan the packets don't even touch the router. |
| 23:24.12 | ManxPower | how about you just try forwarding the ports and see what happens. |
| 23:24.16 | interfaithquest | well i want to have 2 separate dsl , but do not have that convenience |
| 23:24.21 | interfaithquest | yeah i will try that |
| 23:25.22 | ManxPower | interfaithquest: exactly how are you going to "punch thru" a NAT. |
| 23:25.43 | ManxPower | hell things like bittorrent can't even "punch thru nat" |
| 23:26.20 | ManxPower | on Cisco routers you can just see the NAT translation table with 1 command. |
| 23:26.39 | interfaithquest | as skype does |
| 23:27.23 | Maliuta | skype is a major security issue |
| 23:27.29 | interfaithquest | from sources on the net.. one 1st must scan a range of ports to open .. expose one client.. here i will get u the reference |
| 23:27.41 | interfaithquest | here is the latest brainiac on hole punching |
| 23:27.57 | interfaithquest | http://adi.roiban.ro/?p=23 |
| 23:28.03 | Maliuta | it's called my fist in the face of someone I find doing that on my networks |
| 23:28.19 | Maliuta | _that's_ real hole punching |
| 23:28.35 | ManxPower | interfaithquest: "But if the NAT/Firewall is symmetric then a relay node must be present in the direction of the traffic terminating at hat NAT/Firewall, for the duration of the session. This increases the amount of bandwidth consumed by the relay node" from http://www.mocaedu.com/mt/archives/000140.html |
| 23:28.41 | interfaithquest | basically when one client scans a range of ports on the other side..it then opens up itself..for the other side to punch in |
| 23:28.55 | ManxPower | go for it then |
| 23:29.20 | interfaithquest | if i have to.. i was hoping for a temporary solution via iax.. but this has been frustrating so far |
| 23:29.46 | interfaithquest | the plan is to have a 'hole punch |
| 23:30.16 | interfaithquest | <PROTECTED> |
| 23:30.25 | ManxPower | "The basic idea is to have each host behind the NAT contact a third well-known server (usually a STUN server) in the public address space and then, once the NAT devices have established UDP state information, to switch to direct communication hoping that the NAT devices will keep the states despite the fact that packets are coming from a different host." Any firewall that keeps the state despite the fack packets are coming from a differ |
| 23:31.16 | interfaithquest | stun will get about 90% of the nats.. port scanning is NOT done by stun, which fails on a symmetric nat |
| 23:31.44 | [TK]D-Fender | STUN? I'm trying to set my laser printer on **KILL** |
| 23:32.36 | interfaithquest | anyway i was hoping iax would at least get things going.. that is now a faint hope.. as not everyone is keen to fiddle with the dsl router |
| 23:33.53 | interfaithquest | chan_gtalk , i had hopes for that too, but it also failed to go peer to peer |
| 23:34.27 | interfaithquest | i guess via STUN it can ONLY do the simple NAT devices.. that are easily punched thru.. |
| 23:35.26 | Micc | how could I search my file system for any files great than 1GB? |
| 23:35.35 | ManxPower | Micc: "man find" |
| 23:36.02 | drmessano | paypal needs a command line interface |
| 23:36.31 | harry_v | TK, ever been shocked by anything in the kv range before? |
| 23:36.52 | drmessano | kv range is meaningless |
| 23:37.06 | harry_v | I was once hit by 25kv but luckily it was 250 or less ma |
| 23:37.36 | harry_v | never less, I flew backwards about 5 feet and hit the door. |
| 23:37.39 | Micc | what is kcore? |
| 23:37.49 | Micc | and why would it be 2+GB? |
| 23:37.50 | ManxPower | "All I want for xmas is a Tesla coil" |
| 23:38.03 | ManxPower | Micc: kcore is a kernel crash. |
| 23:38.05 | harry_v | I built a tesla coil once. |
| 23:38.06 | harry_v | :) |
| 23:38.17 | Micc | can I delete it? |
| 23:38.44 | harry_v | perfect car protection. Park your car under your 10 foot tesla coil and arm it. |
| 23:39.18 | [TK]D-Fender | harry_v: Been done : Robocop - watch the "ads" |
| 23:40.15 | harry_v | I know |
| 23:40.16 | drmessano | High voltage is overrated. Any 5th grader with a school science program has been shocked with thousand of kv |
| 23:40.18 | harry_v | ;) |
| 23:40.26 | drmessano | thousands |
| 23:40.28 | Micc | aha. I think I found it. |
| 23:40.35 | Micc | sql.log was 2gb |
| 23:40.49 | harry_v | still fun to do some interesting things with it. |
| 23:40.56 | drmessano | Yeah, nothing like deleting a sql log |
| 23:41.06 | drmessano | Those CANT possibly be useful |
| 23:41.12 | drmessano | </sarcasm> |
| 23:41.19 | *** join/#asterisk pluesch0r (n=pluesch0@vie-078-142-128-086.dsl.sil.at) |
| 23:42.02 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 23:42.12 | harry_v | drmessano actually two guys put together a telsa coil musical set. each one made one note and when played by software made some really interesting music. |
| 23:42.50 | harry_v | That would be something to see say at LasVagas. I have to go there again. |
| 23:42.56 | drmessano | I spent too much of my life around high voltage to even want to go near it on purpose |
| 23:43.35 | drmessano | plate voltage is not your friend |
| 23:43.50 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
| 23:44.37 | *** join/#asterisk mchou (n=mchou@unaffiliated/mchou) |
| 23:45.07 | lesouvage | It sounds a bit like the music equipment discribed in the hitchhikers guide |
| 23:45.18 | drmessano | Nothing will make you double-check your paystubs more than 10,000v at a couple Amps arcing near your head |
| 23:45.26 | jaytee | at least it isn't Vogon poetry |
| 23:45.40 | drmessano | oh fredled gruntbuggly |
| 23:45.49 | drmessano | they nicturations are to me |
| 23:46.06 | drmessano | Like something gabbleblotchits on a lurgid bee |
| 23:46.07 | lesouvage | jaytee: it could be. |
| 23:46.15 | drmessano | thy* |
| 23:46.21 | drmessano | Thats about what I remember from the book |
| 23:46.49 | drmessano | and if you were playing the game.... anyone... anyone? |
| 23:46.57 | drmessano | "enjoy poetry" |
| 23:47.20 | mchou | Question re asterisk function PrivacyManager. If call comes in w/no caller ID info, but caller successfully enters his 10 digit number, does a (subsequent) call to the function CALLERID(num) return the number the caller entered? |
| 23:48.21 | drmessano | HA, thats it |
| 23:48.30 | drmessano | Oh freddled gruntbuggly, thy micturations are to me/ As purdled gabbleblotchits on a lurgid bee |
| 23:49.05 | lesouvage | mchou: it depends on how you put into your dialplan. Yo caould do Read(INBOUND_NUMBER,etc, etc.) ad do Set(CallerID(num)=INBOUND_NUMBER), you have to arange something. |
| 23:49.36 | drmessano | In the game you needed to know like the 4th word in the second verse.. only way you got to the second verse was to "enjoy poetry" |
| 23:49.41 | drmessano | Even the game was sick |
| 23:49.47 | mchou | lesouvage: umm, I'm not sure I understand what you men |
| 23:49.57 | mchou | mean* |
| 23:50.46 | mchou | lesouvage: I need to do a read??? |
| 23:51.34 | mchou | lesouvage: If caller enters 10 digit number, how do I "retrieve" that number via asterisk? |
| 23:51.52 | mchou | lesouvage: I mean in the dial plan |
| 23:52.04 | lesouvage | mchou: if callerid is disabled the value of CALLERID(num) ill be empty or a string of zeros (0000000) This can trigger the routing of the call to a context/extensions that ask the caller to enter his/her number. This is done with the Read() application (gogle for Read asterisk cmd). |
| 23:52.46 | lesouvage | sorry, just came out of town, i'm in typo mode. |
| 23:52.54 | mchou | lesouvage: umm, I thought that's what PrivacyManager does |
| 23:53.16 | mchou | lesouvage: the asterisk function, that is |
| 23:53.55 | lesouvage | I have never used PrivacyManger so maybe you are right. |
| 23:54.20 | mchou | lol |
| 23:54.29 | mchou | come on man |
| 23:54.42 | lesouvage | ? |
| 23:54.54 | mchou | if you intend to help please at least know what I'm referring to |
| 23:55.10 | mchou | Dont want blind leading the blind |
| 23:56.27 | lesouvage | mchou: I seem to be your best shot at the moment. Wait a moment,I will do some googling |
| 23:59.42 | lesouvage | mchou: well, think it is kind of nonsense application. Just check if the CALLERID(num) has any value and if not route it to a routine to enter a phonenumber (using Read()). What is relevant is what value CALLERID(num) has when the passing of callerid is disabled on the caller side. |