| 00:01.08 | [TK]D-Fender | Nope, thought that was a phone for a moment there. |
| 00:01.15 | [TK]D-Fender | Guess we ARE still on the Polycom side of things |
| 00:01.25 | [TK]D-Fender | For which you don't need DHCP for any of this |
| 00:01.34 | [TK]D-Fender | specify the server right on the phone |
| 00:04.30 | *** join/#asterisk crazed1 (themrrober@unaffiliated/themrrobert) |
| 00:06.42 | Ice_Strike | If I remember correctly Press Menu -> Advance Setting -> Enter Password 456 -> Network Config -> Server Menu |
| 00:10.56 | crazed1 | he's already tried that, he said a non-default boot image was loaded from previous server with a non-default pw |
| 00:12.13 | Ice_Strike | <PROTECTED> |
| 00:12.15 | [TK]D-Fender | Trash the bootrom entirely then |
| 00:12.49 | Ice_Strike | [TK]D-Fender Do you use TFTPD? |
| 00:13.06 | [TK]D-Fender | Ice_Strike: Not unless I have to, and for Polycom I don't |
| 00:13.46 | Ice_Strike | Ahhh, what do you use then? |
| 00:13.53 | [TK]D-Fender | Ice_Strike: Only one I ever did for for some decript Uniden UIP-200's I got when I first deployed my office. I took them out of service entirely a while back. |
| 00:14.01 | [TK]D-Fender | Ice_Strike: FTP across the board |
| 00:14.44 | Ice_Strike | FTP to download bootrom from a server? |
| 00:14.48 | Ice_Strike | Cool |
| 00:15.14 | Ice_Strike | I use TFTPD at work and bootrom stored on /tftptboot on linux |
| 00:15.29 | Ice_Strike | Work quite well |
| 00:15.42 | [TK]D-Fender | I use VSFTPD and work and it's stored in an equally nice folder on linux |
| 00:15.55 | *** join/#asterisk Phoebus (~Phoebus@pdpc/supporter/active/phoebus) |
| 00:16.14 | Ice_Strike | <PROTECTED> |
| 00:16.37 | hackish | Do those devices typically accept an unsigned firmware image? |
| 00:16.50 | [TK]D-Fender | ftp is the easiest to test right off of just about an OS. Even windows has a command-line FTP client |
| 00:18.51 | hackish | Kind of scary if you think about it. What is to stop an intruder from literally "bugging" your phones via a firmware update? |
| 00:21.28 | Ice_Strike | [TK]D-Fender I use /usr/sbin/in.tftpd |
| 00:22.05 | Ice_Strike | I will check out VSFTPD tommrow and see how good it is. |
| 00:22.16 | Ice_Strike | I will also try FTP |
| 00:22.29 | Ice_Strike | I like to check which is most reliable. |
| 00:24.28 | hackish | vsftpd has had a lot of security problems in the past. |
| 00:26.34 | *** join/#asterisk taylorbyte2013 (~cyberninj@2001:470:1f05:12cc:1e6f:65ff:fec2:bec7) |
| 00:33.09 | [TK]D-Fender | So has everything in the past |
| 00:33.13 | [TK]D-Fender | Welcome to the present |
| 00:33.49 | hackish | Some things have really bad track records because the architecture is poor or the coding isn't very good. |
| 00:34.26 | [TK]D-Fender | And we're still talking about the past |
| 00:36.35 | *** join/#asterisk bn-7bc (~bjarne-im@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 00:36.47 | hackish | well reviewed stuff can still have holes, but sometimes history continues into the present and future. |
| 00:38.28 | hackish | NOTICE[27190] chan_aes.c Registration key missing. Not loading secchannels. |
| 00:38.40 | hackish | Any ideas? Google shows nothing about it. |
| 00:38.59 | [TK]D-Fender | Sometimes generic talking-point based conversations about "security" should be chopped off at the knee-caps since continuing them makes one look like a FUD-flinging tin-foil hat wearing paranoid schizophrenic :p |
| 00:41.59 | unicron | paranoia makes a good servant, but a poor master |
| 00:42.38 | hackish | one can make a good living working in that area. |
| 00:42.39 | beardy | FTP must die. |
| 00:43.19 | unicron | +1 beardy |
| 00:45.20 | [TK]D-Fender | This following talk of TFTP |
| 00:45.21 | hackish | I would argue that ftp has already died. lots of years ago. |
| 00:46.05 | hackish | the only thing I've ever seen tftp used for is to transfer firmware images. |
| 00:46.50 | hackish | do you care how secure tftp is so long as the image is signed? |
| 00:51.16 | crazed1 | not everyone requires signed images- the fatal flaw |
| 00:51.41 | [TK]D-Fender | Even if the image itself is signed, how much damage can just the CONFIGS used by the image do? |
| 00:51.56 | [TK]D-Fender | Stop assuming that the image itself is the only threat |
| 00:52.13 | [TK]D-Fender | That is so much harder to do. |
| 00:52.44 | [TK]D-Fender | Also... you're running on the assumption of it being unsigned and unprotected against being dummied-up |
| 00:53.16 | hackish | if you use tftp to pick up a config image that's not signed, same principle applies. The difference is, with a compromised firmware you could have a device do things it would otherwise not be permitted to do, even one that pretends to take firmware upgrades. |
| 00:53.36 | crazed1 | [TK]D-Fender - they have protections against dummy-ups now? |
| 00:53.54 | crazed1 | didnt know that was possible |
| 00:54.32 | crazed1 | anyone is capable of screwing up even the simplest task |
| 00:54.38 | crazed1 | or maybe that sjust me XD |
| 00:54.50 | hackish | sure, build something idiot proof they just build a better idiot |
| 00:55.12 | crazed1 | Yeah, totally y^ |
| 00:55.21 | [TK]D-Fender | No, that's when you discover you're working with MORONS, from which there is no protection |
| 00:56.30 | file | hackish, chan_aes is not an in-tree module so you will be lucky if anyone here has experience with it |
| 00:57.38 | hackish | file, I think I figured it out.. I didn't realize it was part of a package of modules I installed. |
| 00:58.17 | hackish | found it in the /usr/local/lib/asterisk/modules |
| 00:58.28 | crazed1 | Ahh, classic. brb |
| 01:00.20 | *** part/#asterisk coreyf1513 (~cfarrell@ws2.cfware.com) |
| 01:13.45 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 01:19.56 | *** join/#asterisk deo (~deo@122.53.72.218) |
| 01:20.44 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 01:32.18 | *** join/#asterisk themrrobert (themrrober@unaffiliated/themrrobert) |
| 01:32.49 | DoYouKnow | anyone know of an ANAC number that reads back caller id? |
| 01:33.11 | DoYouKnow | I've been having trouble finding one that reads back the full caller id |
| 01:35.43 | themrrobert | 888-624-9266, press 2 at prompt. not true anac tho |
| 01:40.57 | themrrobert | 800-444-4444 - this one is easier to remember |
| 01:42.12 | DoYouKnow | those don't read the caller id |
| 01:42.29 | DoYouKnow | not as far as I can tell, anyway |
| 01:42.33 | DoYouKnow | I just get a number |
| 01:42.35 | themrrobert | you mean cnam? |
| 01:42.41 | themrrobert | callerid /name |
| 01:42.44 | DoYouKnow | yeah |
| 01:43.05 | themrrobert | you're sol in that department with me, sry |
| 01:48.22 | hesco | [TK]D-Fender and lorsungcu : I'm back from laundromat now, ready for work this week, and have reviewed the conversation in my absence. [TK]D-Fender: you advise "Trash the bootrom entirely then", which was my original question, how is it I do that. This phone for the past nearly four years has registered to two pbx servers, one belonging to a client and one which I administer. I used to administer the client's pbx as well, but have not |
| 01:48.47 | hesco | I'm trying to sort out how to break in and reset the phone so I can configure it from scratch. |
| 01:49.15 | [TK]D-Fender | hesco: upgrade it from what it's running now |
| 01:49.21 | hesco | how? |
| 01:49.28 | lorsungcu | do exactly what i said earlier |
| 01:49.33 | [TK]D-Fender | by providing omething NEWER |
| 01:49.45 | hesco | w/o the password? |
| 01:50.28 | [TK]D-Fender | no need |
| 01:53.10 | hesco | I do not understand. Reviewing the channel's buffer gives me no clue as to what exactly lorsungcu said earlier. |
| 01:54.12 | lorsungcu | have you factory reset the phone? |
| 01:54.32 | hesco | you started to talking about manipulating the dhcp options, but that conversation seems to have trailed off as I was running out the door to do laundry. I also reported that my consumer applianc router did not seem to let me configure the Option 66 and 150 settings. |
| 01:54.41 | lorsungcu | well |
| 01:54.48 | lorsungcu | go download tftpd32 |
| 01:54.48 | hesco | that is what I keep asking, HOW IS IT I FACTORY RESET THE PHONE? |
| 01:54.57 | lorsungcu | that sounds like an awesome google query |
| 01:55.03 | lorsungcu | why don't you try it |
| 01:55.09 | lorsungcu | and come back when you have an idea |
| 01:55.44 | hesco | As reported in this channel earlier, I did try that that and the best I came up with was 4-6-8-*, which failed to wipe the non-default password. |
| 01:56.01 | lorsungcu | alright |
| 01:56.03 | hesco | Another article advised using the mac address as a password, but that did not work either. |
| 01:56.20 | lorsungcu | i assume you tried other common passwords |
| 01:56.42 | hesco | I tried 4-5-6 |
| 01:56.53 | hesco | unaware of 'other common passwords' for this phone |
| 01:57.33 | lorsungcu | 1111 |
| 01:57.34 | lorsungcu | 2222 |
| 01:57.34 | lorsungcu | 3333 |
| 01:57.35 | lorsungcu | 4444 |
| 01:57.37 | lorsungcu | so on |
| 01:57.39 | lorsungcu | 1234 |
| 01:57.40 | lorsungcu | 12345 |
| 01:57.44 | lorsungcu | use your imagination |
| 01:57.54 | lorsungcu | if you do these |
| 01:57.56 | lorsungcu | and none of them work |
| 01:58.07 | lorsungcu | then you need to download tftpd |
| 01:58.11 | [TK]D-Fender | hesco: I asked you WHEN you tried the factory reset. |
| 01:58.43 | [TK]D-Fender | hesco: I have since also suggested providing it a provisioning source and flushing the bootroom by UPGRADING it |
| 02:01.50 | hesco | I'm trying that again. |
| 02:01.57 | hesco | it rejected 4-5-6 |
| 02:02.04 | hesco | trying the mac address |
| 02:02.07 | lorsungcu | holy fuck |
| 02:02.14 | lorsungcu | hesco |
| 02:02.19 | lorsungcu | have you met xmovies |
| 02:02.20 | hesco | yes |
| 02:02.26 | hesco | no |
| 02:02.28 | lorsungcu | hmm |
| 02:03.16 | hesco | 2500+ porn movies? |
| 02:03.24 | lorsungcu | what? no |
| 02:04.02 | hesco | first page of google results to a search on that string are all about porn |
| 02:04.29 | lorsungcu | yeah not what i was referring to |
| 02:04.31 | hesco | sip501 now loading application, not sure from where though. |
| 02:04.41 | lorsungcu | oh |
| 02:04.56 | hesco | I fear from the same IP to my client's pbx ftp server |
| 02:05.07 | hesco | we shall see |
| 02:05.09 | lorsungcu | SoundPoint IP 501: Press and hold the Up and Down Arrow keys, Delete key, and Menu key in combination. |
| 02:05.13 | lorsungcu | have you tried that |
| 02:05.38 | hesco | no, first I've heard of that. will try it if this does not work |
| 02:05.50 | lorsungcu | is the phone an IP501? |
| 02:05.53 | hesco | now running sip.ld |
| 02:06.02 | lorsungcu | hesco its just booting |
| 02:06.07 | lorsungcu | don't need a play by play |
| 02:06.22 | hesco | a'right |
| 02:06.38 | [TK]D-Fender | hescosip501 now loading application, not sure from where though. <- from the NVRAM on the phone |
| 02:07.08 | hesco | thanks. |
| 02:19.09 | hesco | it came back with the old configuration again. |
| 02:19.26 | hesco | next I try the arrow-del-menu trick |
| 02:30.38 | hesco | using the arrow-del-menu trick again brought up the config I'm trying to shake |
| 02:31.46 | hesco | still locked out by non-default password |
| 02:33.04 | *** join/#asterisk Sprocks (~Sprocks@bas1-brampton37-2925499447.dsl.bell.ca) |
| 02:33.12 | lorsungcu | hesco: |
| 02:33.21 | lorsungcu | first result in google for 'flashing firmware polycom phone' |
| 02:33.22 | lorsungcu | http://kb.nexmatrix.net/index.php/article/printer/flashing-your-polycom-phones |
| 02:34.06 | hesco | thanks, reading that next |
| 02:54.48 | *** join/#asterisk nanoha-sama (~nanoha-sa@2001:470:e97f:1003:215:5dff:fe07:4806) |
| 03:00.00 | *** join/#asterisk cchhat01 (~cchhat01@207-237-28-173.c3-0.elm-ubr2.qens-elm.ny.cable.rcn.com) |
| 03:02.31 | BludSuckingFiend | Hey, has anyone been able to get the res_calendar_ews module to work? |
| 03:15.46 | *** join/#asterisk deo (~deo@222.127.13.226) |
| 03:26.04 | *** join/#asterisk dpilon (~dpilon@c-50-138-178-238.hsd1.ct.comcast.net) |
| 03:28.22 | WIMPy | At least the guy who wrote it, I guess. |
| 03:28.50 | *** join/#asterisk mchou (~quassel@unaffiliated/mchou) |
| 03:38.37 | mjordan | ccchat01: you can ask in here, there's an excellent chance someone else in here will know the answer when I don't :-) |
| 03:39.56 | mjordan | BludSuckingFiend: what version and what problem are you having? |
| 03:46.27 | *** join/#asterisk FireAndIce (~FireAndIc@123.201.2.132) |
| 03:52.29 | WIMPy | mjordan: You are looking in to making CDRs more usefull? |
| 03:55.36 | BludSuckingFiend | It's asterisk 11.2.1 and the included module |
| 03:56.42 | BludSuckingFiend | It isres_calendar_ews.c:548 send_ews_request_and_parse: Unable to communicate with Exchange Web Service at 'https://nothing.to.see/ews/Exchange.asmx': Could not read status line: connection timed out |
| 03:56.50 | BludSuckingFiend | That's the error, with my domain removed |
| 03:57.10 | BludSuckingFiend | a packet capture reveals that it is trying to send unencrypted HTTP to my HTTPS port |
| 03:57.20 | BludSuckingFiend | despite that the URL I gave it has https |
| 03:58.03 | mjordan | ah. Yeah. I would not be surprised that it doesn't support https |
| 03:58.04 | BludSuckingFiend | after adding a couple of extra logging lines to it, I am thinking it may be neon screwing up |
| 03:58.20 | BludSuckingFiend | the example included is https... |
| 03:58.25 | mjordan | hm |
| 03:58.36 | *** join/#asterisk serafie (~erin@24.214.158.242) |
| 03:59.04 | mjordan | that particular module came from a community member, so I can't verify a lot of it other than some of the XML parsing. I do know some folks have gotten it working, if for no other reason than they've tested some patches |
| 03:59.07 | mjordan | looks up a jira issue |
| 03:59.21 | BludSuckingFiend | and it says it supports https... it even passes the uri.scheme as https to the neon library |
| 04:00.03 | mjordan | hm. ASTERISK-19738 is the only issue I've worked with someone who was testing it |
| 04:00.17 | mjordan | we don't have an Exchange server unfortunately, so we've had to rely on people who have them to test |
| 04:00.40 | BludSuckingFiend | ASTERISK-16881 |
| 04:00.50 | BludSuckingFiend | I found that... segfault that happens to have the same error in it |
| 04:00.53 | WIMPy | You think not having an exchange server is unfortunate? ;-) |
| 04:00.59 | BludSuckingFiend | mine isn't segfaulting though |
| 04:01.13 | BludSuckingFiend | you can bang against my exchange server if you want |
| 04:01.28 | BludSuckingFiend | I've unfortunately got 2 of them -_- |
| 04:01.37 | BludSuckingFiend | a 2007 and a 2010 |
| 04:01.55 | mjordan | what's the version of neon you're using? |
| 04:02.21 | BludSuckingFiend | 0.25.5-10.el5_4.1 |
| 04:02.27 | mjordan | WIMPy: having had the dubious honor of administering one once in my life (and I've completely blocked that knowledge out), no, it's not unfortunate |
| 04:02.39 | BludSuckingFiend | CentOS freshest |
| 04:03.06 | BludSuckingFiend | at least CentOS5 |
| 04:03.24 | mjordan | just reading through their change log on http://www.webdav.org/neon/ |
| 04:03.44 | BludSuckingFiend | Yeah, I looked over that and didn't note any fixes that looked obvious |
| 04:04.34 | WIMPy | should go to sleep, but I'm running on damned good chilli. |
| 04:04.49 | mjordan | is running on beer and popcorn right now |
| 04:04.55 | BludSuckingFiend | I could manually download and install 0.25.6, but who knows what that might break on CentOS. |
| 04:05.16 | mjordan | yeah :-\ I wonder if it's actually making the request. You could try wiresharking it |
| 04:05.29 | BludSuckingFiend | Actually that's how I know it's doing HTTP |
| 04:05.29 | WIMPy | Beer is more likely to help falling asleep. |
| 04:05.32 | mjordan | ah |
| 04:05.39 | BludSuckingFiend | It's sending plaintext HTTP to the SSL socket |
| 04:05.46 | mjordan | boggle |
| 04:05.57 | BludSuckingFiend | and then it waits for a couple of minutes and times out |
| 04:06.09 | mjordan | silly question: openssl is installed? |
| 04:06.43 | BludSuckingFiend | I'll verify, but I think so |
| 04:07.47 | BludSuckingFiend | it seems like this would affect the ical one via ssl too, as they both use neon the same way |
| 04:08.41 | mjordan | other random guess: neon library might not have been built with ssl support |
| 04:09.44 | BludSuckingFiend | also good to check. I'd be surprised being a vanilla CentOS5 install |
| 04:09.50 | WIMPy | Althoug none of them would be a reason to make a http request when https is requested. |
| 04:10.25 | mjordan | definitely not, but then again, when would it be reasonable to do that? |
| 04:10.29 | BludSuckingFiend | I'd find that out... but I think someone locked up my asterisk box -_- |
| 04:10.35 | mjordan | eek |
| 04:10.43 | WIMPy | never |
| 04:18.20 | BludSuckingFiend | lol it won't die |
| 04:18.31 | BludSuckingFiend | even survives kill -9 |
| 04:19.58 | WIMPy | Are you sure it didn't come back? |
| 04:22.29 | edong23 | yeah, if it "survived" kill -9 it would be defunt |
| 04:23.14 | BludSuckingFiend | Yeah, I am probably going to have to drive to work to reboot it |
| 04:23.15 | WIMPy | That does happen, but it hasn't happen to me on Asterisk so far. |
| 04:23.16 | *** join/#asterisk radic (~radic@dslb-094-216-254-067.pools.arcor-ip.net) |
| 04:34.43 | *** join/#asterisk nightrid3r (~kvirc@ip-81-11-201-96.dsl.scarlet.be) |
| 04:35.18 | BludSuckingFiend | the lockup thing is my fault |
| 04:35.28 | BludSuckingFiend | an infinite loop is possible in the dialplan |
| 04:36.46 | BludSuckingFiend | openssl is 0.9.8e-7.el5 |
| 04:41.20 | BludSuckingFiend | I'm lucky a remote reboot worked. |
| 04:44.24 | WIMPy | So you're saying an infinite loop in the dialplan makes Asterisk survive a kill -9? |
| 04:44.36 | BludSuckingFiend | seems to be what happened |
| 04:45.03 | WIMPy | Interesting "feature". |
| 04:45.11 | BludSuckingFiend | The person that admin'd this box before me set it to continuously retry failed dialouts on our ISDN trunks |
| 04:45.27 | BludSuckingFiend | I haven't ever gone back and made it more sane using a counter |
| 04:45.48 | BludSuckingFiend | I just upgraded to 11 from 1.8 so g729 codec failed to load |
| 04:45.48 | WIMPy | Why would you retry anyway? |
| 04:46.15 | BludSuckingFiend | and someone with their phone set for g729 tried to make a call.... and round and round it went |
| 04:46.33 | BludSuckingFiend | we sometimes had channel sticking |
| 04:46.45 | BludSuckingFiend | so one B channel would fail and the next would succeed |
| 04:47.10 | WIMPy | You try to assign cahnnels manually? |
| 04:47.14 | BludSuckingFiend | I think fixes made in 1.8.13 fixed those issues |
| 04:47.28 | BludSuckingFiend | nope... but trying to dial again makes it hit the next channel |
| 04:47.48 | WIMPy | Interesting "feature" again. |
| 04:49.02 | BludSuckingFiend | I do wish DAHDI handled PRIs more like mISDN |
| 04:49.22 | WIMPy | prefers to use mISDN as well. |
| 04:49.41 | BludSuckingFiend | it was pretty tough to get a good picture of which T1 channels were in use by what with DAHDI |
| 04:49.41 | themrrobert | I have 2 pbx's with PRI cards in them, they all receive calls fine, but the ones on pbx2 don't make outbound calls |
| 04:50.00 | themrrobert | it just fails, the configs look the same idk why it wouldn't work |
| 04:50.06 | BludSuckingFiend | once "pri show channels" was added that got a little better |
| 04:50.15 | themrrobert | asterisk 1.8 |
| 04:50.41 | BludSuckingFiend | My PRI provider is finicky about callerID |
| 04:50.48 | WIMPy | Well, no T1 in mISDN as far as I know. |
| 04:50.54 | BludSuckingFiend | if we don't set it to a proper DID we own, the call out fails |
| 04:51.07 | themrrobert | hmm that sounds like it might be the issue |
| 04:51.15 | WIMPy | themrrobert: use pri debug to see what's happening. |
| 04:51.17 | BludSuckingFiend | It looks like you can set up PRIs with mISDN... and it allows directly accessing the T1 channels |
| 04:51.54 | BludSuckingFiend | I was considering moving to mISDN... but decided too great a chance of breaking my production box |
| 04:51.58 | WIMPy | You certainly can use PRIs with mISDN. I have done so, but mISDN is DSS1 only as far as I know. |
| 04:52.16 | BludSuckingFiend | DS1 = T1 |
| 04:52.23 | BludSuckingFiend | or E1 |
| 04:52.36 | WIMPy | The signalling. |
| 04:52.59 | BludSuckingFiend | not familiar with DSS1 |
| 04:53.06 | BludSuckingFiend | I know DMS, ESS, and NI/NI2 |
| 04:53.30 | BludSuckingFiend | those are just protocol variations of ISDN though |
| 04:53.36 | WIMPy | And AFAIK none of them are supported. |
| 04:53.49 | BludSuckingFiend | hmmm |
| 04:54.07 | BludSuckingFiend | Yeah, I'll stick with DAHDI I think. |
| 04:54.46 | WIMPy | I guess you have to. |
| 04:55.22 | WIMPy | Unless the difference doesn't matter. |
| 04:55.32 | BludSuckingFiend | I have this asterisk CTI phone I put together for allowing someone to telecommute and use their home phone for bearer |
| 04:55.36 | WIMPy | Have you used mISDN then? |
| 04:55.42 | BludSuckingFiend | Uses the asterisk management interface.... |
| 04:56.02 | BludSuckingFiend | Unfortunately I don't seem to see any way of figuring out what channel events are associated with MY originate command |
| 04:56.05 | BludSuckingFiend | never |
| 04:56.06 | BludSuckingFiend | I was looking at it |
| 04:56.17 | BludSuckingFiend | Because DAHDI was being a pain. We had B channels "sticking" |
| 04:56.28 | BludSuckingFiend | but it was almost impossible to see which PRI channels were in use |
| 04:56.53 | WIMPy | I like the realtime usage display :-) |
| 04:57.07 | BludSuckingFiend | mISDN? |
| 04:57.22 | WIMPy | LCR to be precise. |
| 04:58.54 | BludSuckingFiend | I ended up making some bash/perl scripts that did a DAHDI show channel X for each channel |
| 04:59.07 | BludSuckingFiend | then did a core show channel on it to get more detail |
| 04:59.31 | BludSuckingFiend | then builds a nice HTML table of all the T1 channels and what they are connected to |
| 04:59.41 | WIMPy | Might be easier to listen for events on AMI. |
| 04:59.43 | BludSuckingFiend | It puts a lot of load on asterisk though |
| 05:00.07 | BludSuckingFiend | well on the version I upgraded from none of the AMI events had the DAHDI span or channel info |
| 05:00.10 | BludSuckingFiend | just the channel name |
| 05:00.31 | BludSuckingFiend | does now though, so I may end up using that for some things |
| 05:00.46 | BludSuckingFiend | I really wish Asterisk had a CSTA-style interface |
| 05:01.07 | WIMPy | Do you get to see the channel at all? |
| 05:01.46 | WIMPy | Other than using the CHANNEL function that is. |
| 05:01.46 | BludSuckingFiend | this was in 1.8.1... it just showed the DAHDI/CLIDNUM-id name |
| 05:01.50 | themrrobert | WIMPy - it is because of caller id. we have 2 different providers that come into the box, 1 doesn't require a valid CID, the other requires it to be one of their numbers |
| 05:02.02 | themrrobert | is there a way to change the order of PRI's used? |
| 05:02.05 | themrrobert | for outbound |
| 05:02.26 | BludSuckingFiend | You could probably put them into different groups |
| 05:02.34 | BludSuckingFiend | and try group x first |
| 05:02.55 | WIMPy | DAHDI/in/callerid |
| 05:03.22 | WIMPy | Yes, it's all about groups and the type of group. |
| 05:03.36 | BludSuckingFiend | and yeah.. I suppose I could see the span # |
| 05:03.38 | BludSuckingFiend | but no channel |
| 05:03.53 | WIMPy | in chan_dahdi.conf you set up which channels belong to a group and in the dialplan you set the strategy in which order they are used. |
| 05:04.32 | *** join/#asterisk Natureshadow (nik@shore.naturalnet.de) |
| 05:11.43 | themrrobert | i'm looking at this file and it says autogenerated by wancfg_dahdi, i'm ok with playing with it, but it looks like it assigned group=0 to all the PRI's but group=1 underneath [channels] |
| 05:12.39 | BludSuckingFiend | one thing you can do is assing them to 2 groups... for instance... |
| 05:12.47 | BludSuckingFiend | the first one to 0,11 |
| 05:12.54 | BludSuckingFiend | second to 0,12 |
| 05:13.05 | BludSuckingFiend | that way you can access them individually via 11 or 12 |
| 05:13.10 | BludSuckingFiend | or together via 0 |
| 05:13.17 | themrrobert | clever |
| 05:13.22 | WIMPy | Note tha chan_dahdi.conf is a bit reverse in that all options apply to the following channels. |
| 05:13.25 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-srjlqpzruipibagi) |
| 05:13.50 | themrrobert | right, like group=x, then channels=1-23 |
| 05:15.56 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-dvpqmbiombuuhhbm) |
| 05:21.36 | themrrobert | sweet i was able to make it work! thanks guys :) |
| 05:41.31 | *** join/#asterisk deo (~deo@58.71.19.178) |
| 05:51.42 | *** join/#asterisk bogrd__ (~asteriskp@115.119.115.26) |
| 06:01.58 | hackish | Did anyone here ever buy the asterisk definitive guide 4th edition (early release) |
| 06:02.17 | hackish | Seems kinda odd that o'reilly is selling access to a book they describe as not finished yet. |
| 06:05.56 | BludSuckingFiend | Nope. Never really had a need for books |
| 06:06.00 | BludSuckingFiend | not sense the Internet.... |
| 06:06.05 | BludSuckingFiend | since* |
| 06:06.24 | hackish | the book is a pdf |
| 06:07.08 | BludSuckingFiend | Have you read it? |
| 06:07.36 | hackish | I read the first half until they started skipping pages. |
| 06:07.58 | BludSuckingFiend | if you have to pay for the early release, I should hope they will give you the finished product too |
| 06:08.29 | hackish | they do. I just wonder how complete it is. |
| 06:09.28 | hackish | suppose I could probably just read the free HTML version. Just doesn't seem right to go read someone's book and not pay for their work. |
| 06:09.54 | BludSuckingFiend | I wonder how much original content they really have |
| 06:10.44 | BludSuckingFiend | if you already paid for it, I don't see the harm in reading a different format if it is more complete |
| 06:10.45 | hackish | yeah, that's possible that they're just revising/updating. |
| 06:11.23 | BludSuckingFiend | There's just so much free content online, I can't imagine getting a book |
| 06:11.30 | hackish | no, I haven't purchased yet, was just pondering which version to buy. |
| 06:12.17 | hackish | somehow the authors need to get paid. |
| 06:12.31 | BludSuckingFiend | I mean free as in open |
| 06:12.43 | BludSuckingFiend | not paid material someone uploads for free... |
| 06:12.47 | BludSuckingFiend | like wiki.asterisk.org |
| 06:12.53 | BludSuckingFiend | voip-info.org et |
| 06:12.55 | BludSuckingFiend | etc* |
| 06:13.39 | BludSuckingFiend | give me a reference manual and sample configs over someone's interpretation any day though... |
| 06:15.23 | hackish | it doesn't really seem like there is such a thing as simple configs. It's like editing an old school sendmail.mc file |
| 06:15.50 | BludSuckingFiend | sample* |
| 06:16.08 | hackish | haha... misparsed that one. |
| 06:16.28 | BludSuckingFiend | The sendmail config is pretty bad lol; |
| 06:16.54 | hackish | there is a reason I've been running postfix since before it was called postfix |
| 06:17.02 | hackish | vmailer baby |
| 06:17.08 | BludSuckingFiend | I prefer exim |
| 06:18.07 | hackish | that's one I've never tried. ran a big free webmail system for years. 250k users. you don't want to be running sendmail on that. |
| 06:18.56 | BludSuckingFiend | Yeah. I like how exim is so configurable. Nearly everything is scriptable |
| 06:20.31 | BludSuckingFiend | Pairing it with courier-imap works well |
| 06:20.59 | *** join/#asterisk Invader (~Invader@unaffiliated/invader) |
| 06:21.02 | hackish | wrote my own imap server. What a mistake. |
| 06:21.16 | hackish | much harder than you'd think |
| 06:21.17 | BludSuckingFiend | whoah... that has to be a pain |
| 06:21.28 | hackish | the protocol looks pretty simple. |
| 06:21.31 | BludSuckingFiend | I know exactly how hard that would be... IMAP is definitely not simple |
| 06:21.42 | hackish | connect 50,000 users to it and it ain't that simple anymore |
| 06:21.54 | BludSuckingFiend | For awhile I was running and testing hMailServer |
| 06:21.57 | hackish | most of the problem is doing it efficiently |
| 06:22.02 | BludSuckingFiend | a windows-based open source IMAP server |
| 06:22.23 | BludSuckingFiend | I saw all the code that went into properly parsing the headers, particularly with different character sets |
| 06:22.58 | hackish | haha. yep that's ugly and nobody follows the RFC. Worst of all everyone gets pissed if you don't process non-compliant emails. |
| 06:23.37 | BludSuckingFiend | I needed to integrate an IMAP server with active directory |
| 06:23.50 | hackish | hell half of the smtp servers out there don't even follow the rfc |
| 06:23.51 | BludSuckingFiend | I never even considered writing one from scratch |
| 06:23.53 | *** join/#asterisk bogrd__ (~asteriskp@115.119.115.26) |
| 06:24.03 | BludSuckingFiend | I wrote a perl script to replace courier-authd instead |
| 06:24.44 | BludSuckingFiend | the SMTP RFC is kind of outdated... a lot of the RFC breakage is intentional to prevent spam |
| 06:24.58 | hackish | honestly don't know too much about windows servers. back in the NT days we made all our money tossing out windows NT machines and replacing them with unix boxes. |
| 06:25.20 | *** join/#asterisk seanbright (~sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 06:25.24 | BludSuckingFiend | tarpitting I avoid windows for servers when I can. |
| 06:25.49 | hackish | not really. a lot of it was just people not reading the rfc. You'd get tons of messages with stupid crap like the date field in an arbitrary format etc. |
| 06:25.52 | BludSuckingFiend | unfortunately I have to use Exchange at work |
| 06:25.56 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 06:26.14 | hackish | or for example the other end of an SMTP server is suposed to say HELO <postmaster@foobar.com> or whatever. |
| 06:26.19 | BludSuckingFiend | the executives demand their MS Outlook with calendar sharing |
| 06:26.22 | hackish | lots of them just say HELO <> |
| 06:26.26 | hackish | or nothing at all. |
| 06:26.52 | BludSuckingFiend | HELO followed by the hostname for clients |
| 06:26.54 | hackish | only because someone never bothered to enter the server name in the config |
| 06:26.56 | BludSuckingFiend | EHLO for servers generally |
| 06:27.21 | BludSuckingFiend | exim is very strict on the RFC |
| 06:27.29 | hackish | ja, couldn't remember if it was hostname or postmaster address. |
| 06:27.48 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 06:27.48 | BludSuckingFiend | for instance... all E-mail servers are supposed to wait for the banner from the server BEFORE sending EHLO or HELO |
| 06:27.53 | BludSuckingFiend | per the RFC |
| 06:27.57 | hackish | in any event at one point fewer than 60% of servers were compliant |
| 06:28.02 | BludSuckingFiend | many spam clients just connect and immediately send to save time |
| 06:28.21 | hackish | yep that's true. |
| 06:28.24 | BludSuckingFiend | Exim will intentionally tarpit for about 10 seconds before sending the banner |
| 06:28.39 | BludSuckingFiend | and will disconnect the client with an error if it tries to send anything beore the banner |
| 06:28.55 | hackish | if we waited 10 seconds on every connect the server would probably overload :) |
| 06:29.17 | hackish | we did it via grey listing. |
| 06:29.20 | *** join/#asterisk jmls (~somefake@77.107.171.82) |
| 06:29.29 | hackish | told the unknown hosts to try again later. |
| 06:29.46 | hackish | however a good percent of email comes from the same big machines. google, hotmail, yahoo etc. |
| 06:29.52 | BludSuckingFiend | It's a configurable wait. I usually change it. Though it also will skip it if it gets proper reverse DNS and it check's out |
| 06:30.05 | hackish | you run spf and all that? |
| 06:30.21 | hackish | that too I find useful |
| 06:30.22 | BludSuckingFiend | Yeah, SPF, a coule of DNSBLs |
| 06:30.27 | *** join/#asterisk bogrd__ (~asteriskp@115.119.115.26) |
| 06:30.41 | hackish | yep the rbls are useful. |
| 06:31.04 | BludSuckingFiend | not so useful lately... Sudden spike in spam getting through |
| 06:31.06 | hackish | when you've got a big mail system you get hundreds of thousands of the same spam from the same host in a single day. |
| 06:31.11 | BludSuckingFiend | it's a never-ending battle |
| 06:31.34 | hackish | It's pretty easy to recognise it. |
| 06:31.35 | BludSuckingFiend | Yeah, my biggest exchange server only handles about 150 accounts |
| 06:31.44 | hackish | Most spam comes from about the same 1000 IPs anyway. |
| 06:32.30 | hackish | surprisingly 150 accounts is probably harder to keep spam free than 50,000 |
| 06:32.40 | BludSuckingFiend | yeah. Blocking mail from DHCP ranges can help a lot |
| 06:33.15 | hackish | yes but there is still a lot of legit email that can come from those and emails with no reverse dns |
| 06:33.26 | BludSuckingFiend | my server probably rejects about 50-75% of the messages it gets as spam |
| 06:33.33 | BludSuckingFiend | accurately |
| 06:33.36 | hackish | for a while I tried blocking all that but there were just too many complaints |
| 06:33.58 | hackish | if you're not already try greylisting. It's surprisingly effective. |
| 06:34.12 | BludSuckingFiend | my company had issues with an ISP that was one of their customers... they were using Postini and it was blocking our SMTP |
| 06:34.15 | *** join/#asterisk mihamina (~mihamina@ip-41-190-237-66.orange.mg) |
| 06:34.17 | hackish | that alone cut spam by 55-60% |
| 06:34.28 | BludSuckingFiend | They refused to even talk to us about getting us whitelisted |
| 06:34.59 | BludSuckingFiend | they insisted we were misconfigured even when shown the results of manually telnetting to send a message. The response was "telnet isn't a mail program" |
| 06:35.00 | Nugget | telnet is eeeeeeevil! |
| 06:35.18 | BludSuckingFiend | I feel sorry for that ISPs customers |
| 06:35.39 | hackish | we had a problem like that once. Sorry your servers can't SMTP out. They will have to relay through ours... Um, sir are you absolutely positively 100% sure you want what you're asking us to do??? |
| 06:35.57 | hackish | we were unblocked within 45 minutes. |
| 06:36.05 | BludSuckingFiend | that's what most consumer (residential) ISPs do |
| 06:36.19 | BludSuckingFiend | I can't imagine a business class ISP blocking outbound SMTP |
| 06:36.30 | hackish | I think they spent the next 3 days waiting for their spool directories to clear |
| 06:36.33 | BludSuckingFiend | pretty ridiculous |
| 06:36.45 | hackish | there is a retard in every fishbowl |
| 06:36.57 | BludSuckingFiend | doing a lot of E-mail marketing eh? |
| 06:37.20 | hackish | no, 250k accounts generate a _LOT_ of traffic |
| 06:37.26 | BludSuckingFiend | we were sending too many messages to the SMS gateways of Sprint |
| 06:37.43 | BludSuckingFiend | we ended up having to sign up for their enterprise gateways |
| 06:37.50 | hackish | don't know the stats but somewhere around 1,000,000 messages a day. |
| 06:38.18 | BludSuckingFiend | yeah, that's quite a bit |
| 06:38.35 | hackish | every time there is a new phishing scheme I know about it first. |
| 06:38.57 | BludSuckingFiend | honeypot accounts are handy |
| 06:39.04 | hackish | yep. |
| 06:39.22 | hackish | I've always wanted to setup a few honeypot linux machines on a VM somewhere. |
| 06:40.03 | BludSuckingFiend | yeah, set the disks to differencing |
| 06:40.04 | hackish | it would be fun just sending random error messages to the kiddies |
| 06:40.13 | BludSuckingFiend | on VMWare anyway... so if they break it, just reboot and you're good |
| 06:40.18 | *** part/#asterisk DoYouKnow (~michael@75-28-180-206.lightspeed.bnsvil.sbcglobal.net) |
| 06:41.08 | hackish | you could do fun stuff like pretend it's a mega machine returning someting crazy like 32 cpus but peg it's load like it's really doing hard work and make it pailfully slow for them to use. |
| 06:41.52 | BludSuckingFiend | if you ever got a live person |
| 06:42.09 | BludSuckingFiend | most of the hits would be script kiddies running a script against hundreds of IPs |
| 06:42.13 | BludSuckingFiend | or worms |
| 06:42.28 | hackish | I assume they probably run their skripts to get in then log in live to play. |
| 06:42.34 | BludSuckingFiend | You know those web-to-email forms? |
| 06:42.47 | BludSuckingFiend | some sites have where just submitting them sends an email to someone |
| 06:42.57 | hackish | true. You have no idea how effective those things are. |
| 06:43.00 | BludSuckingFiend | most people use captcha's on them now to stop the scripts |
| 06:43.22 | BludSuckingFiend | One site I did awhile back I did a little differently |
| 06:43.23 | hackish | I've busted webmail accounts that had over 100,000 credit cards or banking usernames/passwords collected in them. |
| 06:43.44 | BludSuckingFiend | hidden field on the form that defaulted to a certain value that would get it rejected |
| 06:43.56 | hackish | oh sorry was thinking of the websites they stick up on a hacked site to look like a bank. |
| 06:44.05 | BludSuckingFiend | javascript hook on the text area, so if you actually pressed keys in the box it toggled the hidden field to allow... |
| 06:44.14 | BludSuckingFiend | that stopped all scripts and just let people actually typing messages through |
| 06:44.19 | hackish | that's a neat idea. |
| 06:44.37 | BludSuckingFiend | it would not work if you're big enough that a person targets you |
| 06:44.47 | BludSuckingFiend | but it stops the automated scripts when you're fairly low visibility |
| 06:44.55 | hackish | we used to put limits on how many bounces an account can get. |
| 06:45.28 | hackish | new users got an auto - 15 minute queue after 5 messages were sent. If they got a bounce it stuck them on an hour queue. |
| 06:45.51 | BludSuckingFiend | people where I work type bad addresses so much |
| 06:46.02 | hackish | So you'd get 10 spam bounces and they'd waste the next hour sending email that just landed in a deleteme queue. |
| 06:46.22 | hackish | another thing to look at is the ratio of successful to non-successful |
| 06:46.36 | hackish | In the first hour you know if it's a spammer or not. |
| 06:47.09 | *** join/#asterisk v0lZy (~Thunderbi@mail.silk-group.net) |
| 06:47.12 | BludSuckingFiend | ah, for newly created accounts? |
| 06:47.58 | BludSuckingFiend | SMTP is a pretty terrible protocol these days. The people that developed the standard obviously didn't anticipate spam |
| 06:48.40 | hackish | of course. I still remember the first spam I ever got. was like in the very early 1990's. I was like who the heck is this guy??? |
| 06:48.48 | *** join/#asterisk v0lZy (~Thunderbi@mail.silk-group.net) |
| 06:49.23 | BludSuckingFiend | for awhile I was making up aliases when signing up for stuff |
| 06:49.32 | BludSuckingFiend | an alias that was basically an MD5 of the domain I was on |
| 06:49.39 | BludSuckingFiend | so I could go back and determine who sold my info |
| 06:50.02 | hackish | I used to use spam@domain.com. Every spammer used to filter out the spam@ addresses. |
| 06:50.17 | drmessano | ha |
| 06:50.27 | drmessano | I never thought of that |
| 06:50.30 | BludSuckingFiend | clever |
| 06:50.48 | hackish | unfortunately as time went on humans assumed the same too. |
| 06:51.23 | hackish | something funny is to google a really old email address you used to have. Amazing some of the stuff you will dig up. |
| 06:51.36 | hackish | I found usenet posts from the '90s |
| 06:51.50 | BludSuckingFiend | yeah, amazing that stuff is still around |
| 06:52.06 | *** join/#asterisk edong23 (~quassel@mptc-dhcp-50-220.mptelco.com) |
| 06:52.27 | hackish | I pity the kids of today. People born in the '95 or so... Their whole life history will be on FB and all that. |
| 06:53.09 | hackish | hell I remember when Netscape 1.0 came out. It was so much better than NCSA mosaic |
| 06:58.24 | hackish | you don't really need qualify=yes for an ATA with a static IP on the asterisk side of the firewall correct? |
| 06:59.02 | BludSuckingFiend | probably not |
| 06:59.15 | BludSuckingFiend | makes more sense over a link that is likely to have variable latency |
| 06:59.21 | BludSuckingFiend | or subjet to going up and down |
| 07:00.20 | hackish | ja, just trying to slim a few things down here, get rid of unnecessary stuff. |
| 07:00.52 | *** join/#asterisk bogrd__ (~asteriskp@115.119.115.26) |
| 07:01.25 | hackish | asterisk 66729 0.0 1.1 690744 46108 ?? Is 12:41AM |
| 07:01.46 | hackish | 690m of RAM ouch |
| 07:08.31 | *** join/#asterisk corretico (~luis@190.211.93.38) |
| 07:13.22 | *** join/#asterisk ThomasLocke (~ThomasLoc@pdpc/supporter/active/thomaslocke) |
| 07:13.34 | themrrobert | someone was mentioning that versions of asterisk newewr than 11.0.3 (or possibly > 11.1.x) had some serious issues with sRTP or some other important but obscure protocol. is there any validity to this, or do the newer versions work just fine? |
| 07:15.57 | phix | *shrugs* I still use 1.8.x |
| 07:16.02 | phix | any reason to use 11? |
| 07:16.05 | *** join/#asterisk din3sh (2988fac5@gateway/web/freenode/ip.41.136.250.197) |
| 07:16.22 | edong23 | i still use 1.6 |
| 07:16.35 | edong23 | unless a feature comes out i need, ill likely remain on 1.6 |
| 07:17.28 | hackish | I've been trying to setup 1.8. The only thing I haven't found it able to do is the dishes but there is probably a module for that somewhere |
| 07:18.26 | drmessano | If you're interested at all in Google Voice, 11 is the only way to go. Also, conferencing |
| 07:19.16 | phix | What's wrong with MeetMe() ? |
| 07:19.22 | drmessano | lol |
| 07:19.25 | drmessano | MeetMe sucks |
| 07:19.32 | phix | It is conferencing :) |
| 07:19.37 | drmessano | ConfBridge is 10x better |
| 07:19.37 | phix | Conferencing sucks |
| 07:19.39 | themrrobert | i like 11 because of database query |
| 07:19.48 | phix | you can do that in 1.4 themrrobert |
| 07:19.55 | themrrobert | not in 1.8 |
| 07:20.01 | phix | Any *REAL* advantages of using 11? |
| 07:20.06 | themrrobert | at least not on the version at my office |
| 07:20.15 | drmessano | I just provided 2 |
| 07:20.16 | themrrobert | i think it uses berkley db instead of sqlite |
| 07:20.17 | phix | themrrobert: AGI is supported since way back |
| 07:20.19 | drmessano | How are they not REAL? |
| 07:20.49 | phix | if an earlier version had the same feature then you can't use that as a reason to upgrade to 11 :) |
| 07:20.56 | themrrobert | phix what does agi have to do with that |
| 07:20.59 | drmessano | I think he's talking about querying astdb directly now that it's sqlite |
| 07:21.03 | themrrobert | ^^^^ |
| 07:22.02 | drmessano | phix, No earlier version had ConfBridge rewritten, with the same featureset as MeetMe, and no earlier version had chan_motif for GoogleVoice/Jingle |
| 07:22.06 | drmessano | So um yeah |
| 07:22.57 | *** join/#asterisk stev3n (~Stiv@sever-kirov.ru) |
| 07:22.57 | drmessano | The fact that ASTDB is sqlite also means its less likely to corrupt |
| 07:23.01 | hackish | there seem to be many database queries/modules. Are these mostly just used for businesses with tons of lines? |
| 07:23.06 | drmessano | As a matter of fact, I have yet to have it corrupt |
| 07:23.48 | themrrobert | me either, but i havent had corruption issues with 1.8 bkdb ... .. yet it seems |
| 07:24.34 | edong23 | i sparked quite the engagement |
| 07:24.35 | edong23 | lol |
| 07:24.49 | stev3n | hi, I update my asterisk and dahdi (=2.6.1-r1), but in CLI it's show me: |
| 07:24.50 | stev3n | asterisk*CLI> dahdi show version |
| 07:24.50 | stev3n | DAHDI Version: 2.5.0.2 Echo Canceller: HWEC |
| 07:24.59 | drmessano | Reboot |
| 07:25.25 | edong23 | all of those are probably good points, but i have no need for google voice with asterisk, and conferencing |
| 07:25.37 | hackish | haha /sbin/shutdown -r now |
| 07:25.44 | stev3n | no |
| 07:25.56 | stev3n | I just reboot |
| 07:26.03 | hackish | oh dear |
| 07:26.09 | drmessano | If you're making SIP calls and using 5% of the Asterisk featureset, theres really no need to go past 1.2. |
| 07:26.37 | stev3n | I rebooted server |
| 07:26.40 | drmessano | I guess that's why the whole "why should I upgrade" ends up being as stupid as a distro war. If you don't want any of the new stuff, don't ask |
| 07:26.47 | hackish | even a linux box probably doesn't ever need to be rebooted. |
| 07:27.04 | hackish | it's not windows you know. |
| 07:27.19 | drmessano | That's the only way to load the new DAHDI modules cleanly after an upgrade |
| 07:27.24 | phix | I see |
| 07:27.32 | v0lZy | linux doesnt really benefit from reboots much |
| 07:27.33 | hackish | seriously? |
| 07:27.36 | stev3n | hi, I update my asterisk and dahdi (=2.6.1-r1), and then reboot my linux mashine |
| 07:27.42 | v0lZy | but install an nvidia driver... and you'll have to reboot :D |
| 07:27.42 | stev3n | :) |
| 07:27.45 | phix | stev3n: hai |
| 07:27.51 | phix | v0lZy: incorrect |
| 07:27.53 | hackish | it's been years since I ran a linux box but I didn't think they were that bad |
| 07:28.00 | drmessano | DAHDI involves kernel modules |
| 07:28.06 | phix | hackish: what do you use? |
| 07:28.13 | drmessano | It's not just an app |
| 07:28.15 | hackish | bsd |
| 07:28.20 | phix | drmessano: corrupt, modprobe loads kernel modules |
| 07:28.22 | edong23 | its also nice to check to be sure the system comes up full working after power failure or system crash |
| 07:28.24 | phix | corrupt = correct |
| 07:28.28 | edong23 | so yeah, you might want to reboot |
| 07:28.29 | hackish | don't you have loadable kernel modules? |
| 07:28.30 | v0lZy | phix: You dont anymore? |
| 07:28.31 | stev3n | also I run module-rebuild |
| 07:28.35 | phix | nice |
| 07:28.38 | stev3n | system: gentoo |
| 07:28.51 | *** join/#asterisk gerhard7 (~gerhard7@82-169-24-72.ip.telfort.nl) |
| 07:28.59 | phix | hackish: yes dahdi uses loadable kernel modules |
| 07:29.04 | phix | you dont need to reboot |
| 07:29.09 | phix | same with the NVIDIA Modules |
| 07:29.10 | hackish | I was hoping not. |
| 07:29.27 | hackish | last time I ran linux the disto was called slackware |
| 07:29.37 | hackish | so I've been out of that game too long |
| 07:29.38 | phix | hackish: so what you run now? |
| 07:29.44 | phix | FreeBSD |
| 07:29.45 | phix | ? |
| 07:29.45 | edong23 | slackware is still a distro |
| 07:29.54 | edong23 | a rock solid distro |
| 07:29.55 | hackish | FreeBSD and OpenBSD |
| 07:30.01 | phix | yeah slackware still exists, most people just use debian or ubuntu though |
| 07:30.11 | phix | hackish: shibby |
| 07:31.25 | hackish | unfortunately I have to also run slowaris |
| 07:32.09 | hackish | don't care much, unix is unix... annoying sometimes when people put things in different places but it's like riding a bike |
| 07:32.19 | v0lZy | phix: What do you do to get the nvidia modules to run without a reboot? modprobe doesnt work for me. |
| 07:34.11 | stev3n | maybe re-emerge asterisk will help |
| 07:35.39 | hackish | nvidia is for CUDA? |
| 07:36.43 | stev3n | no, re-emerge asterisk not helped, still DAHDI Version: 2.5.0.2 |
| 07:36.50 | themrrobert | i remember when red hat was selling RHEL for $9.99 at fry's electronics, that was my first linux, i was so excited, like 12 years old |
| 07:37.08 | themrrobert | actually i think i was 13 or 14 |
| 07:37.20 | themrrobert | man linux has come a long way since then |
| 07:37.54 | hackish | it used to be that you had to apply kernel patches every other week just to keep up with all the security holes. |
| 07:39.11 | *** join/#asterisk areski (~areski@80.174.255.57.dyn.user.ono.com) |
| 07:41.54 | edong23 | heh, hackish sounds like someone i know... |
| 07:42.05 | edong23 | but its true, linux has had a bumpy ride |
| 07:42.10 | drmessano | I remember when video was a pain in the ass on a linux desktop... and umm, wait |
| 07:42.11 | hackish | everyone praised linus as being a god but I worked a lot with his old stuff. It was OK code but in no way comparable to the old AT&T labs stuff. Writen by a competent coder but not a visionary and not a software achitect. |
| 07:42.20 | edong23 | but mostly with local exploits, so i guess it would depend on what your systems did |
| 07:42.40 | edong23 | lol |
| 07:42.51 | edong23 | hackish: hows that AT&T labs software doing these days? |
| 07:43.22 | hackish | edong, some of their original code remained in the kernels for longer than linux has existed so I'd say their stuff lasted pretty good. |
| 07:43.43 | hackish | seems that the linux network stack has been rewritten how many times? |
| 07:44.03 | edong23 | and improved |
| 07:44.06 | edong23 | lol |
| 07:44.28 | edong23 | is your arguement that making drastic (needed) changes is a bad thing? |
| 07:44.32 | *** join/#asterisk studybot_ (~studybot_@gateway/tor-sasl/studybot/x-68286794) |
| 07:44.41 | drmessano | I know a guy that's been using Linux for 25 years (yes, yes, I am aware of the math) |
| 07:45.43 | hackish | sure, after 15 rewrites they probably did get the issues ironed out but that really supports what I'm saying. the guy did a lot of cool stuff and was a good coder but if you want to separate what I consider to be a true software god you have to look at someone who came up with a good design then coded it, not someone who hacked something up a zillion times and eventually made it work right. |
| 07:46.05 | drmessano | There is only one god.. Stallman |
| 07:46.35 | hackish | I agree. I still have a bunch of his books. |
| 07:46.52 | hackish | I swear he probably had the entire posix spec in memory |
| 07:47.05 | edong23 | hackish: i think... i see the issue here |
| 07:47.11 | edong23 | there is an educational gap |
| 07:47.14 | hackish | as for linux I struggle to remember what year I first ran it. I'd like to say it was early 1990's |
| 07:47.27 | edong23 | you should look into the actual history of WHO coded this so called "linux" |
| 07:47.35 | edong23 | it was all written by torvolds |
| 07:47.37 | edong23 | at all |
| 07:47.39 | edong23 | even close |
| 07:48.03 | hackish | considering back then he actually ftp'd me his source code I'd like to say at the time it was all written by him |
| 07:48.54 | hackish | I haven't followed it in a lot of years but I know back when I ran it the kernel was a pretty ugly mess under the hood. |
| 07:49.13 | edong23 | as most code is when you first fire it up |
| 07:49.17 | edong23 | then you modify it... |
| 07:49.31 | edong23 | then you write code drunk |
| 07:49.33 | edong23 | then yo modify it |
| 07:50.07 | hackish | sure. like I said good on him for what he accomplished. Just understand it's not quite the same as what came out of AT&T. |
| 07:52.42 | edong23 | well, given that is a totally different beast, its liek me saying a chevy pickup just isnt quite the same as a BMW |
| 07:53.08 | hackish | haha that's actually a good analogy. I was going to say field medic versus brain surgeon |
| 07:53.55 | drmessano | I have a family friend who is an old Unix guy and he says Linux is almost as good as WIndows Vista |
| 07:54.03 | drmessano | :( |
| 07:54.03 | edong23 | an ancient monopoly building software isnt the same as someone writing it for free in his basement |
| 07:54.05 | edong23 | so yeah |
| 07:54.06 | drmessano | lol |
| 07:54.32 | hackish | At the end of the day they were initially setup for the same basic thing. Ignore the fact linux appears to have moved far toward the desktop environment now... |
| 07:54.57 | edong23 | its not a move |
| 07:55.01 | edong23 | its an expansion |
| 07:56.01 | hackish | Of course. However, those ancient relics with their berkely and harvard PhD's brought a certain level of "engineering" to the plate. |
| 07:56.04 | edong23 | i like the linux motion for desktop though. it gets people involved in understanding the workings of a computer, even the buntus |
| 07:56.25 | edong23 | like i said, it isnt the same thing |
| 07:56.45 | edong23 | it is, for instance, like you putting 3 asterisk systems together and comparing it to broadsoft |
| 07:57.10 | hackish | not the same skills but being used to build the same thing |
| 07:57.10 | drmessano | The Linux desktop is good for sucking in those that don't want to blow up a machine to stare at a blinking cursor. |
| 07:58.54 | hackish | I was actually thinking of installing it for my mother in law. I get tired of going over and trying to remove a zillion gizmos. At one point she at about 2" of web browser visible because she had every single searchbar and addon available. |
| 07:58.57 | edong23 | hackish: no, you keep (trollingly) jumping over the point. i get it. but stay on point here. The "asterisk comparison" you could possibly put together would require that you write some code to make it compariable to broadsoft. but you likely wont, broadsoft is oh.. give or take $100k for the platform |
| 07:59.21 | edong23 | thus, that corperation will pay software engineers and push out releases (with massive bugs i will add) |
| 07:59.45 | hackish | edong, I can't comment about that point because I don't know enough about the communications stuff. |
| 08:00.19 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 08:00.21 | edong23 | well, if we were talking about you building a car vs a corperation building it... |
| 08:00.24 | edong23 | is what im gettign at |
| 08:00.38 | *** join/#asterisk vlad_starkov (~vlad_star@178.177.7.151) |
| 08:00.46 | edong23 | im not saying whether its a good thing or bad thing |
| 08:00.59 | edong23 | only that there is a different work flow. especially in the open source community |
| 08:01.08 | edong23 | we shit out vomit |
| 08:01.08 | hackish | sure but the point I'm getting at is from the consumer's point of view it is often times better to buy the factory built car. |
| 08:01.19 | edong23 | and other people take it to the cleaners quite often |
| 08:01.49 | edong23 | until you build one that is ugly as shit, but runs on hydrogen cells or something cookey |
| 08:01.50 | edong23 | lol |
| 08:02.11 | hackish | don't get me started on hydrogen powered cars... |
| 08:02.17 | edong23 | get started |
| 08:02.34 | hackish | it's the stupidest idea ever |
| 08:02.36 | edong23 | a friend of mine has one |
| 08:02.37 | edong23 | lol |
| 08:02.41 | hackish | h2 doesn't store well |
| 08:02.44 | edong23 | its cool as shit |
| 08:02.47 | hackish | haha |
| 08:03.08 | edong23 | you mean as in dangerous storage? |
| 08:03.09 | hackish | that's a good pun too.... given that you need to supercool hydrogen to get it into liquid form. |
| 08:03.20 | hackish | actually it's not that bad in storage. |
| 08:03.27 | *** join/#asterisk linocisco (c186f20c@gateway/web/freenode/ip.193.134.242.12) |
| 08:03.29 | *** join/#asterisk deo (~deo@122.53.72.218) |
| 08:03.40 | edong23 | hackish: thats not how its stored |
| 08:03.43 | edong23 | it isnt liquid |
| 08:03.43 | hackish | some will use a chemical to store it. can't think of the term off the top of my head. |
| 08:04.34 | edong23 | he bought this kit a few years ago (might be 5 now) |
| 08:04.49 | hackish | check it out, they supercool it to get it into liquid or they absorb it into anhyrdous form or something like that... only trouble is it's poisionous when you rupture the tank in an accident. |
| 08:04.50 | edong23 | it was outrageously expensive, but he has bank like ill never have |
| 08:04.53 | *** join/#asterisk deo (~deo@203.177.214.78) |
| 08:05.49 | edong23 | well, i know for a fact anhydrous isnt used in his kit its a metal its stored in |
| 08:06.00 | linocisco | hi all, I am using nortel IP phone 1140E with asterisk using http://blog.michaelfmcnamara.com/2011/01/avaya-ip-1100-series-ip-phone-upgrade-to-sip. but I found licnese is evalution though I turned off advanced features |
| 08:06.04 | edong23 | its bonded to the inside of this tank in mesh |
| 08:06.14 | edong23 | and cannot be released without passing electricy through it |
| 08:06.23 | edong23 | or possibly, heating to a certain temp |
| 08:06.26 | edong23 | i dont remember |
| 08:06.28 | linocisco | how can I make that sip licence permanent or no expiration |
| 08:06.43 | hackish | metal hydride. had to look it up. |
| 08:06.58 | edong23 | yeah anyway, its a cool kit |
| 08:07.01 | edong23 | but really expensive |
| 08:07.09 | hackish | how far can he go? |
| 08:07.11 | edong23 | his kit cost more than both my vehicles |
| 08:07.34 | hackish | I'm keen on the PEVs now. Despite my dislike for nissan I might actually buy a Leaf |
| 08:07.35 | edong23 | he has 4 of these tanks, he said it gets him to <insert nearby city) and back without an issue |
| 08:07.44 | edong23 | but, whats nice is he can pull a lever and be on gas anyway |
| 08:08.02 | hackish | ah so he's running an H2 ICE |
| 08:08.30 | hackish | there was a company called ballard that made fuel cells. They actually worked pretty well. |
| 08:08.46 | hackish | I think they put most in buses though. |
| 08:08.46 | edong23 | he sent me the link to the stuff a long time ago |
| 08:09.01 | hackish | In any event, I firmly believe batteries are where it's at. |
| 08:09.02 | edong23 | but something happened, tehy quit selling it, or whatever, it didnt matter cause i didnt have the money |
| 08:09.13 | edong23 | hackish: i agree... but alternatives are nice |
| 08:09.19 | edong23 | batteries require a new card |
| 08:09.21 | hackish | Not that poly batteries are any better when ruptured! |
| 08:09.27 | edong23 | hydrogen doesnt |
| 08:09.29 | edong23 | but will cost more |
| 08:09.30 | edong23 | lol |
| 08:09.52 | hackish | sure. but what do you spend your $$$ on with your car? |
| 08:10.05 | hackish | fuel, oil changes, exhaust system, brakes |
| 08:10.37 | hackish | so pev has regen brakes and aside from suspension wear components most of that junk is gone. |
| 08:10.53 | hackish | you still need replacement batteries eventually... |
| 08:11.20 | hackish | But I'll still take that for 0 emissions |
| 08:11.21 | *** join/#asterisk x1user (~x1user@212.36.13.6) |
| 08:11.34 | *** join/#asterisk willryder (~tedryder@nc-184-3-102-84.dhcp.embarqhsd.net) |
| 08:12.53 | hackish | as you can probably guess I used to work in that industry |
| 08:13.40 | bogrd__ | Hello, I have the asterisk 11.2.1 running with configurations for SIP softphones on Ubuntu 12.04. I need to try out outbound calls across the telephone network. The region being India,Asia. Can anyone please suggest the hardware to be used to be able to do so ? |
| 08:14.54 | x1user | [Feb 25 10:13:46] ERROR[13296][C-000046bc]: utils.c:1189 ast_carefulwrite: write() returned error: Broken pipe |
| 08:15.11 | x1user | I see this all the time,thinks like some agi is producing it;> |
| 08:15.30 | edong23 | bogrd__: fxo or pri or some form of isdn? |
| 08:15.51 | edong23 | or you can do it without hardware by finding a sip gateway provider |
| 08:16.44 | bogrd__ | edong23: I am trying to get a PRI connection from the service provider |
| 08:16.52 | din3sh | bogrd__: sip provider or E1/T1 |
| 08:16.57 | *** join/#asterisk LiuYan (~LiuYan@211.154.128.171) |
| 08:18.10 | bogrd__ | din3sh: How do know more about that ? Not sure about the specifications exactly .. |
| 08:18.15 | edong23 | bogrd__: then grab a digium card, or sangoma, or.. that other company |
| 08:18.30 | edong23 | i cant remember |
| 08:19.09 | din3sh | PRI --> E1? |
| 08:19.28 | din3sh | grab a single span card from digium |
| 08:19.53 | bogrd__ | edong23: I see a lot of these .. http://sangoma.com/products/telecom_boards.html .. How do I find out what suits me the best ? |
| 08:20.30 | edong23 | well, um... |
| 08:20.34 | edong23 | that depends on what you need |
| 08:20.52 | edong23 | as din3sh asked... are we talking about a single e1? |
| 08:21.06 | edong23 | you cna get the a101 |
| 08:21.15 | edong23 | then what about echo cancellation? |
| 08:21.22 | din3sh | suits yourself best? |
| 08:21.23 | edong23 | try it in software? or do you want hardware? |
| 08:21.44 | din3sh | you need to understand your own requirements first for that |
| 08:22.39 | din3sh | try to understand your problem before trying to find a solution, you cant find a solutionfirst then fit it to your need |
| 08:23.13 | din3sh | is it T1 or E1? how many T1 or E1s? etc.... |
| 08:23.35 | linocisco | hi all, I am using nortel IP phone 1140E with asterisk using http://blog.michaelfmcnamara.com/2011/01/avaya-ip-1100-series-ip-phone-upgrade-to-sip. but I found licnese is evalution though I turned off advanced features |
| 08:23.56 | bogrd__ | din3sh: I have set up the asterisk confs to work well with SIP softphones . NOw I need to extend the same over a phone the telephone network .. |
| 08:24.15 | bogrd__ | din3sh: What specific requirements should I be concentrating on ? |
| 08:24.23 | din3sh | i jst told you |
| 08:25.01 | din3sh | are you new to how a pbx works? |
| 08:25.22 | bogrd__ | din3sh: edong23: Thanks for the help .. I research more on the E1/T1 requirements and get back to you asap .. |
| 08:25.31 | bogrd__ | din3sh: yes |
| 08:25.42 | din3sh | am pretty sure, you have all the details in asterisk wiki |
| 08:25.50 | din3sh | how to connect to pstn ect |
| 08:26.30 | bogrd__ | din3sh: thanks il spend some time on that .. |
| 08:27.11 | edong23 | bogrd__: ask your local telco what they can offer |
| 08:27.13 | edong23 | then come back |
| 08:27.34 | edong23 | look into a pri |
| 08:27.44 | edong23 | i assume you call it that even in india |
| 08:27.47 | edong23 | or something like that |
| 08:28.05 | din3sh | india i think it might be E1 |
| 08:29.36 | edong23 | well, here its a t1 there its and e1, but that only describes the channel numbers and size |
| 08:29.54 | edong23 | but i think they still have somethign that describes a dchannel and i think 30 b channels |
| 08:30.15 | din3sh | yep, where i am , there's only E1 |
| 08:30.22 | edong23 | a pri, or some india/euro anology |
| 08:30.57 | edong23 | but sangoma doesnt care anyway |
| 08:31.01 | edong23 | you can do all of that |
| 08:31.07 | edong23 | i have limited digium experience |
| 08:31.56 | din3sh | doesnt matter T1 or E1, jst a matter of config as you say |
| 08:32.21 | din3sh | i have dozens of prod boxes using digium E1 |
| 08:33.24 | edong23 | cool |
| 08:33.31 | edong23 | i have been thinking, i shoudl support digium |
| 08:33.38 | edong23 | since they are behind asterisk |
| 08:33.49 | edong23 | but i probably wont be buying any more hardware for a while |
| 08:34.05 | edong23 | i have several 4 port cards and an oct |
| 08:35.34 | *** join/#asterisk deo__ (~deo@58.71.19.178) |
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| 09:14.25 | alami | hello i have an asterisk on gentoo and a new one on debian, can i copy all config file from one to the other one? |
| 09:16.42 | v0lZy | guys, asterisk question regarding dialplan and a goto loop... how do i repeat it 3 times only? |
| 09:16.54 | v0lZy | (without this breaking for other calls) |
| 09:19.48 | *** join/#asterisk ghost75 (~trechber@dslb-178-010-045-219.pools.arcor-ip.net) |
| 09:23.51 | kleszcz | alami: ofc. |
| 09:25.01 | kleszcz | v0lZy: GotoIf($[${L} > 3]?end) |
| 09:25.05 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:7cd9:9293:83df:16fe) |
| 09:25.31 | v0lZy | kleszcz: does that effect new calls that fall into that context |
| 09:25.41 | v0lZy | im doing an IVR |
| 09:25.50 | v0lZy | and i want multiple people to call it |
| 09:25.56 | v0lZy | and each person to have 3 tries |
| 09:26.01 | v0lZy | after 3 tries, i want it to hangup |
| 09:26.34 | kleszcz | v0lZy: http://pastebin.pl/74366 |
| 09:28.24 | v0lZy | kleszcz: does the variable ${L} get reset for each call? |
| 09:28.32 | v0lZy | i mean, its independant on each channel? |
| 09:28.53 | v0lZy | or is it global... so if it inrements to 2 on 1 call, and another person calls, its set to 3? |
| 09:29.00 | v0lZy | or does it start over for each new call? |
| 09:30.23 | rahulr92 | Hi. I am having running asterisk (1.8.20) server on a public IP. I am able to register and initiate calls from my remote twinkle sip client. But no audio is heard though the CLI shows 'Playing 'demo-echotest.gsm' for the demo exten 600. |
| 09:30.32 | alami | kleszcz: what's ofc? |
| 09:30.38 | kleszcz | for example, somebody call to you, and do nothing (digit) 3xtimes then have hangup |
| 09:30.58 | kleszcz | alami: you can copy scp -r /etc/asterisk/ host:/etc/ |
| 09:31.05 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 09:31.14 | alami | kleszcz: and it will work? |
| 09:31.18 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 09:31.22 | kleszcz | yeap |
| 09:31.23 | alami | i don't need any other confige? |
| 09:31.28 | alami | :d waw |
| 09:31.31 | alami | that's cool |
| 09:31.34 | kleszcz | 1ist you must install ;) |
| 09:31.46 | alami | so only on the hard phone the ip of the new server and that's all |
| 09:31.51 | kleszcz | and then you can copy files |
| 09:31.59 | alami | waw cool |
| 09:32.05 | alami | that's awesome :D |
| 09:32.09 | kleszcz | its just config files |
| 09:32.25 | alami | yes but on this way you confige one asterisk on your life |
| 09:32.28 | kleszcz | thats rigjt |
| 09:32.42 | alami | for the other just copy paste and a little changes ;) |
| 09:32.46 | alami | you know what i mean? |
| 09:33.03 | kleszcz | yes |
| 09:33.26 | alami | cool thanks a lot |
| 09:33.32 | rahulr92 | I've tried all the NAT related solutions available online.. involving modification of sip.conf. No use so far. Can anyone please help? |
| 09:34.03 | kleszcz | when i have many call centers, and i must install newone, i just use scp and simple install of asterisk |
| 09:34.33 | kleszcz | rahulr92: hmm codec? |
| 09:34.42 | kleszcz | and NAT i think |
| 09:35.45 | rahulr92 | kleszcz: I am able to head the audio while connecting to a local installation of asterisk.. |
| 09:36.32 | rahulr92 | Its only when I connect to the server that I'm facing the issue. Can this have anything to do with sever settings on UDP? |
| 09:36.46 | linocisco | hi all, I am using nortel IP phone 1140E with asterisk using http://blog.michaelfmcnamara.com/2011/01/avaya-ip-1100-series-ip-phone-upgrade-to-sip. but I found licnese is evalution though I turned off advanced features |
| 09:38.43 | v0lZy | kleszcz: thanks for the example, but im still not clear about the ${L} variable. Is the variable growing for each call independantly? |
| 09:38.47 | v0lZy | or for all calls together? |
| 09:40.13 | *** join/#asterisk ulogic (421e6b4f@gateway/web/freenode/ip.66.30.107.79) |
| 09:43.58 | *** join/#asterisk thornz (~thorn@109.232.228.27) |
| 09:44.54 | thornz | Hello, is it possible in any of the asterisk versions to specify the source ip that Asterisk will use for engaging outgoing SIP & RTP streams per sip peer? |
| 09:46.18 | kleszcz | for incoming calls, you have IVR, somebody call to you, and ist of the dialplan you have exten => s,1,Set(L=1} the start count' |
| 09:46.24 | kleszcz | i show you exmaple, one min. |
| 09:48.00 | kleszcz | v0lZy: http://pastebin.pl/74367 |
| 09:48.35 | kleszcz | i use this for timeout |
| 09:49.16 | v0lZy | thanks kleszcz |
| 09:49.44 | linocisco | hi all, I am using nortel IP phone 1140E with asterisk using http://blog.michaelfmcnamara.com/2011/01/avaya-ip-1100-series-ip-phone-upgrade-to-sip. but I found licnese is evalution though I turned off advanced features |
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| 10:28.56 | Wimdev | does anyone know how i can avoid a (soft)phone with video support from sending video if the other end isn't a video phone? Right now i notice they always send video, even if the other end only supports voice |
| 10:29.53 | Wimdev | i found an old issue about this in 1.8, and it still seems to be the same in the latest v11 https://issues.asterisk.org/jira/browse/ASTERISK-17470 |
| 10:30.41 | ectospasm | if you're not including the video codec in the allow= line in sip.conf, then it's the fault of the phone |
| 10:31.00 | ectospasm | or at least, if you have an explicit disallow= before the allow= line(s) |
| 10:31.26 | Wimdev | it is included, because its a video phone. But it should not send a video stream if calling a voice only phone |
| 10:31.48 | ectospasm | the phone has no way of knowing that |
| 10:32.00 | Wimdev | i know, but asterisk should know |
| 10:32.33 | Wimdev | so asterisk should send a sdp without the video codec when connecting the call to a sip without video support imho |
| 10:32.45 | Wimdev | *to a sip peer |
| 10:32.58 | ectospasm | Asterisk will set up the call leg between the calling phone first. |
| 10:33.39 | ectospasm | ...if that peer has the video codec allowed, it will advertise video in the SDP. The call to the receiving audio-only phone is independent. |
| 10:34.04 | ectospasm | Unless you expect Asterisk to reinvite to the video phone once an audio-only phone is bridged. |
| 10:34.14 | Wimdev | that makes sence, but asterisk is able to renegotiate codecs after the calls is up |
| 10:34.40 | ectospasm | Yes, but I don't know if Asterisk will renegotiate codecs in this instance. |
| 10:35.07 | Wimdev | well, by default it doesn't it seems, so thats why im asking if it can be configured |
| 10:36.50 | Wimdev | because it seems like a bug, to let a peer send video to the asterisk when it's a voice only call |
| 10:37.50 | ectospasm | why is that a bug? If you don't want the video phone to send video, you need to disallow the video codec. |
| 10:38.11 | ectospasm | anyway, Asterisk will pass the video data to the bit bucket |
| 10:38.44 | Wimdev | its a bug because it should not say the phone can send video if the other end cant receive it |
| 10:39.24 | Wimdev | and disallowing video isn't an option, because it's a video phone, when it calls another video phone it should send video :-) |
| 10:39.25 | ectospasm | ...but as I said, Asterisk doesn't know that until it contacts the receiving phone. And I don't see what the problem is here. |
| 10:40.00 | ectospasm | why do you want it not to send video? Is your network that bandwidth constrained? |
| 10:40.04 | Wimdev | the problem is it generates waste bandwith, it can be a problem when you have an asterisk not in the same lan |
| 10:40.31 | ectospasm | so file a new bug, if you think it's a bug. Although I think this is a feature request. |
| 10:43.10 | Greenlight | I would have thought that during codec negotiation that video would be disallowed |
| 10:43.20 | Greenlight | Although I've not done much work with video |
| 10:44.06 | Greenlight | Have you looked at a SIP trace, to see what side the problem is at, Asterisk or the phone ? |
| 10:45.30 | Wimdev | i have, the problem is asterisk |
| 10:45.47 | Wimdev | the phone will send video if asterisk tells it to |
| 10:45.52 | ectospasm | I think he expects Asterisk to renegotiate the originating leg of the call when the terminating end negotiates only audio. |
| 10:46.00 | Wimdev | exactly |
| 10:46.06 | Greenlight | Hmm yea I see |
| 10:46.16 | Wimdev | and sure, on a lan its not really a problem |
| 10:46.24 | Greenlight | It would only do that if directmedia=yes |
| 10:48.02 | *** join/#asterisk Ice_Strike (~Ice_Black@host213-120-117-212.in-addr.btopenworld.com) |
| 10:48.08 | Ice_Strike | Hi |
| 10:48.14 | Greenlight | Hiya |
| 10:48.50 | *** join/#asterisk vlad_starkov (~vlad_star@178.176.82.29) |
| 10:53.09 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 10:55.07 | linocisco | hi all, I am using nortel IP phone 1140E with asterisk using http://blog.michaelfmcnamara.com/2011/01/avaya-ip-1100-series-ip-phone-upgrade-to-sip. but I found licnese is evalution though I turned off advanced features |
| 10:57.16 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:7cd9:9293:83df:16fe) |
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| 11:39.16 | *** join/#asterisk pietro (~pietro@78-134-103-172.v4.ngi.it) |
| 11:39.19 | pietro | hello |
| 11:40.08 | pietro | I have a phone that can play music on hold by himself |
| 11:40.29 | pietro | can I use this feature with asterisk ? |
| 11:40.59 | pietro | (I don't wont to use the asterisk internal music on hold) |
| 11:41.24 | kleszcz | yes |
| 11:44.54 | pietro | kleszcz: how ? |
| 11:50.53 | *** join/#asterisk FireAndIce (~FireAndIc@219.91.142.228) |
| 11:55.28 | *** join/#asterisk salz212 (~chatzilla@203.215.176.22) |
| 11:56.05 | kleszcz | sorry wrong terminal pietro. I dont know, you can use file, ore streaming from network, but from phone? i dont know |
| 11:56.48 | sekil | I think you can disable internal moh |
| 11:56.49 | sekil | somehow |
| 11:57.44 | kleszcz | yes, module unload res_musiconhold.so ;) |
| 11:58.05 | Greenlight | If the phone is generating it's own MOH, then why does asterisk need to be told of anything ? |
| 11:58.27 | Greenlight | Surley the phone will just send MOH instead of the microphone... |
| 12:00.27 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 12:10.11 | salz212 | what is the best way of keeping spool directory centralized (e.g. for voicemail directory) for multiple asterisk servers. I want to use voicemail application on each server and just dump/retrieve the files from just one. |
| 12:17.18 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 12:17.42 | sekil | Greenlight: because it's generating RTP by default |
| 12:17.51 | sekil | Greenlight: and stays in the path |
| 12:17.56 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 12:17.56 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 12:18.47 | Greenlight | I thought that's what was wanted, the phone to generate the MOH ? |
| 12:19.24 | sekil | Greenlight: yes but asterisk will send it's own |
| 12:19.36 | Greenlight | Only if the phone tells it to |
| 12:19.52 | sekil | Greenlight: huh |
| 12:19.54 | sekil | ? |
| 12:20.24 | Greenlight | If the phone is going to generate it's own MOH then I can't see why it would need to tell Asterisk anything about a call being on hold |
| 12:21.04 | sekil | Greenlight: ok.. |
| 12:22.13 | sekil | Greenlight: phone A calls B...B press hold button and emits whatever... on hold asterisk will change rtp to 0.0.0.0 to it's moh ..so A will hear asterisk moh |
| 12:22.49 | sekil | Greenlight: if the asterisk is not in rtp path (canreinvite=yes) A should hear the RTP flowing from the B |
| 12:22.50 | Greenlight | And you want phone B to play it's own MOH ? |
| 12:23.09 | sekil | Greenlight: I don't want anything...I'm explaining what OP wanted |
| 12:23.41 | Greenlight | I'm not 100% certain, but would imagine if the phone is doing the MOH itself, then it shouldn't tell Asterisk anything. |
| 12:23.59 | Greenlight | If the phone tells asterisk to put a call on hold, then that's what Asterisk is going to do... |
| 12:25.12 | sekil | Greenlight: phone doesn't "tell" asterisk anything at all |
| 12:25.24 | sekil | it just send sreinvite |
| 12:27.09 | Greenlight | If the phone doesn't tell asterisk anything, then how does Asterisk know to put the call on hold... |
| 12:27.31 | sekil | because of reinvite conforming to rfc |
| 12:27.34 | Greenlight | https://www.dialogic.com/webhelp/IMG1010/10.5.3/WebHelp/description/sip/sip_re-invite_(hold).htm |
| 12:27.35 | Greenlight | Like that |
| 12:28.11 | Greenlight | The RFC dictates that the gateway should place the call on hold. This is what is happening. If the phone is to play it's own MOH, then this cannot happen. |
| 12:28.13 | sekil | it doesn't tell him "play this" |
| 12:28.18 | sekil | it's just sends call hold reinvite |
| 12:28.36 | Greenlight | Yea, exactly |
| 12:29.15 | sekil | but I tend to use on my pbxes for the internal calls not to send rtp to server...so on reinvite my linksys/cisco phones generate moh |
| 12:29.18 | sekil | i.e. |
| 12:35.17 | sekil | so he was asking how to disable asterisk's moh.. |
| 12:35.22 | sekil | hence all my writing.. |
| 12:36.04 | Greenlight | But I think you're missing the point, if the phone is telling Asterisk to place a call on hold, then that's where the issue is |
| 12:36.53 | Greenlight | Does the RFC not define a sip hold as a reinvite to 0.0.0.0 ? |
| 12:37.50 | *** join/#asterisk pietro (~pietro@78-134-103-172.v4.ngi.it) |
| 12:38.33 | pietro | Greenlight: the phone sends a re-INVITE with a sendonly sdp |
| 12:39.36 | sekil | I think that was old style rfc...nowadays I think there should be sendonly |
| 12:39.48 | sekil | rfc 2543 mandated 0.0.0.0 |
| 12:44.12 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-jausllpyxolmomvb) |
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| 13:39.10 | reuf | i have this astgeriks card: http://www.stocksy.co.uk/articles/Networks/review_x100p_wildcard_fxo_card_from_x100p_com - what kind of software can i use to run asteriks server on i |
| 13:41.33 | R1ck | well, you'll need an OS (Linux), Asterisk itself and drivers for the card (Zaptel) |
| 13:43.31 | [TK]D-Fender | DAHDI ..... Zaptel hasn't existed for many years |
| 13:45.38 | leifmadsen | s/Zaptel/DAHDI |
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| 14:01.20 | ketas | DoYouKnow: what, i have asterisk install, what about it? |
| 14:01.58 | DoYouKnow | ketas: oh, I thought you had some kind of electronics chat |
| 14:02.06 | DoYouKnow | on your asterisk box |
| 14:02.09 | DoYouKnow | frogs said you did |
| 14:04.18 | ketas | what does it even mean |
| 14:06.09 | leifmadsen | double rainbow! omg! |
| 14:11.05 | DoYouKnow | ketas: a conference room |
| 14:11.58 | ketas | DoYouKnow: oh, well no... does i even have some use?! |
| 14:12.45 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 14:13.41 | ketas | s/i/it/ |
| 14:14.02 | ketas | yea, i also find myself useless at times |
| 14:14.46 | DoYouKnow | ketas: not sure, telephreak has a PBX where you can discuss phone phreaking and other technical matters |
| 14:16.19 | ketas | hmm |
| 14:18.32 | ketas | no, i don't have such service |
| 14:23.09 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
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| 14:48.15 | ketas | my mobile isp offers voip now |
| 14:48.57 | ketas | strange pricing however |
| 14:49.46 | ketas | how come it's "business" service still |
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| 14:54.09 | *** join/#asterisk ZeKoU (~amer@servisi.bih.net.ba) |
| 14:54.27 | ZeKoU | Hi |
| 14:54.35 | *** join/#asterisk biomorph (~biomorph@91.85.204.16) |
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| 14:54.52 | ZeKoU | Could someone help me with VoIP scenario that I have? |
| 14:54.59 | leifmadsen | ~ask |
| 14:54.59 | infobot | 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. |
| 14:55.41 | ZeKoU | I have workstation in the office with X100P FXO card, and I have POTS line that company pays as much as I spend... |
| 14:55.42 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-yaqtbvbznikmaojx) |
| 14:55.42 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:56.13 | ZeKoU | I also have Mobile phone plan with 2GB of 3G included.... |
| 14:56.43 | ZeKoU | I want to route all my calls from cell phone (3G) to my workstation and then to POTS line.... |
| 14:57.09 | ketas | this isn't hard |
| 14:57.17 | Greenlight | I'm waiting for the question... |
| 14:57.35 | ketas | ZeKoU: can you replace pots with sip? |
| 14:57.49 | ZeKoU | I know I will have to install Asterisk or some flavor of it on the workstation, but how to setup public network call between cellphone and workstation which is in NAT and behind few other firewalls |
| 14:58.23 | ZeKoU | I want to use POTS as this line will be then charged for any landline or GSM call I make... |
| 14:58.34 | ZeKoU | and company pays whatever I spend... |
| 14:59.03 | Greenlight | Ok, firstly, call your mobile provider and check they don't block SIP, if you've not already. Some do, and it will save you hassle finding out later. |
| 14:59.26 | Greenlight | I seem to recall it was T-mobile here in the UK who we discovered blocked SIP |
| 14:59.28 | ketas | they do dpi? |
| 14:59.41 | ketas | some mobile operator does dpi for sip? |
| 14:59.42 | ketas | :P |
| 14:59.55 | Greenlight | Greedy mofo's these mobile operators |
| 15:00.29 | Greenlight | That, and they realise people will try to do exactly what he's trying :P) |
| 15:00.36 | ketas | mine OFFERS sip |
| 15:00.42 | ketas | rather than blocking it |
| 15:01.03 | igcewieling | ketas: that is a compelling reason to block SIP other than SIP to/from their own servers |
| 15:01.10 | Greenlight | Indeed |
| 15:01.39 | Greenlight | that's what we found, they block SIP other than their own, unless you purchsed some overpriced "addon" package |
| 15:03.14 | ketas | i wonder how they exactly block sip |
| 15:03.33 | igcewieling | *grumble* My new ISP charges as much for a 3Mbps connection (no TV) as my former ISP charged for a 25Mbps connection with basic Cable TV |
| 15:03.38 | ketas | sounds like very hard task |
| 15:03.53 | igcewieling | ketas: usually they just block port 5060 |
| 15:03.59 | Greenlight | Indeed |
| 15:04.03 | ketas | what if i use other port? |
| 15:04.21 | igcewieling | ketas: they count on most users being too stupid to do that. |
| 15:04.23 | *** join/#asterisk navaismo (~navaismo@189.241.118.172) |
| 15:04.43 | Greenlight | Or most users being unable to change the port at the other sidef |
| 15:05.00 | igcewieling | or that 8-) |
| 15:05.01 | Greenlight | You can also VPN around it |
| 15:05.06 | ketas | data traffic in mobile networks is rising anyway... they lost the game anyway |
| 15:05.49 | ZeKoU | Ok I think they do not block SIP...(third world country here - no better plans to offer)... |
| 15:05.50 | Greenlight | Personally I find the latency and reliability too poor for general SIP calls, but maybe I'm picky |
| 15:06.03 | igcewieling | in the USA it doesn't matter. Prices and caps are the same. |
| 15:06.05 | ketas | i haven't tried over 3g much |
| 15:06.17 | ketas | sip |
| 15:06.25 | ketas | it's crap i know |
| 15:06.26 | Ice_Strike | How many channels is possible with 1Mbit upload speed? |
| 15:06.30 | igcewieling | ZeKoU: I found G729 is OK over 3G, but not ulaw or HD voice. |
| 15:06.31 | Ice_Strike | Ulaw |
| 15:06.35 | Ice_Strike | via SIP |
| 15:06.42 | Greenlight | Personally I'd say 8 |
| 15:06.47 | Greenlight | Give yourself headroom |
| 15:06.59 | ketas | i opened up my asterisk for trolling autocallers |
| 15:07.07 | ketas | i put them for waiting forever |
| 15:07.12 | ketas | i trolled my link full instead |
| 15:07.16 | Greenlight | Ice_Strike: Although I think with overheads, each call will be about 80kbps |
| 15:07.25 | Greenlight | So, *in theory* a few more |
| 15:07.31 | igcewieling | Ice_Strike: an ulaw w/rtp call is 80Kbps |
| 15:07.34 | ketas | 15 channels and it was fully filled |
| 15:07.57 | Greenlight | Also, 1mb sounds like ADSL, which is why I'm saying 8 channels |
| 15:08.44 | ketas | your 1mbit isn't 1 too |
| 15:08.46 | igcewieling | 1Mbps DSL usually has 256Kbps upload, for calls with 80Kbps |
| 15:08.50 | Ice_Strike | Greenlight Well with Virgin IPS here with 1Mbit upload speed.. currently 9 concurrents calls without issue. |
| 15:08.56 | Ice_Strike | I was supprise. |
| 15:09.03 | Greenlight | Virgin Media? |
| 15:09.09 | ketas | virgins |
| 15:09.21 | Greenlight | ADSL or Cable ? |
| 15:09.26 | Ice_Strike | Cable |
| 15:09.27 | ketas | isp by virgins for virgins |
| 15:09.31 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 15:09.54 | ZeKoU | Ok guys, let's assume I have great connection (I work at ISP) at work office.. |
| 15:10.28 | ketas | ZeKoU: can't get machine out of fw range? |
| 15:10.35 | ketas | well out of nat |
| 15:10.39 | ketas | fw will be fine |
| 15:11.07 | ZeKoU | http://www.speedtest.net/result/2533499759.png |
| 15:11.10 | Greenlight | Yea, ideally you want a public IP specifically for that machine |
| 15:11.20 | ZeKoU | nope - my boss is strict with pfSense firewalls... |
| 15:11.29 | Greenlight | Then you're SOL |
| 15:11.53 | ZeKoU | and I am not employee for a long time...so I have no map of network and LANs yet... |
| 15:12.04 | Ice_Strike | I am getting Leased Line soon - with 6Mbit Upload and 6Mbit Download |
| 15:12.24 | Ice_Strike | £300 per month! |
| 15:12.27 | Greenlight | Ice_Strike: That'll give you much more capacity. And with a leased line you can guarentee the speed |
| 15:12.34 | Ice_Strike | Yep |
| 15:12.39 | Greenlight | Ice_Strike: Sounds about right price |
| 15:12.50 | igcewieling | Ice_Strike: Thank you! You've made me feel better about how much I pay for internet service |
| 15:12.58 | Ice_Strike | Aprox how many agents? |
| 15:13.02 | Ice_Strike | LOL igcewieling |
| 15:13.18 | Greenlight | Ice_Strike: 50 ish |
| 15:13.23 | igcewieling | Ice_Strike: you already have the numbers to figure that out. |
| 15:13.24 | Ice_Strike | 50 ish?! |
| 15:13.27 | Greenlight | Depends what else you need commection for |
| 15:13.47 | Greenlight | Perhaps more, but that's utilising more that I'd feel happy with |
| 15:13.52 | Ice_Strike | Greenlight Strictly for Asterisk only. |
| 15:13.54 | Ice_Strike | Not for browsing. |
| 15:13.58 | Greenlight | Ok |
| 15:14.06 | Greenlight | 70 at a push |
| 15:14.11 | Ice_Strike | Yep |
| 15:14.19 | igcewieling | Ice_Strike: why not use G729? |
| 15:14.45 | Ice_Strike | I could do. |
| 15:14.49 | Greenlight | G729 you'll get SHITLOADS |
| 15:15.03 | Ice_Strike | Hmm |
| 15:15.09 | Ice_Strike | I'll will look into that |
| 15:15.14 | ZeKoU | so no solution whatsoever for my scenario...? |
| 15:15.21 | Lars_G | did you require a license for g729? |
| 15:15.34 | Greenlight | Lars_G: Yes, you do need one |
| 15:15.36 | Greenlight | £10/channel |
| 15:15.57 | Ice_Strike | Greenlight Asterisk is connected to voip provider (SIP) using ulaw |
| 15:15.57 | ketas | bleh |
| 15:16.05 | Ice_Strike | Not sure if G729 gonna work. |
| 15:16.14 | Greenlight | Ice_Strike: Most ITSP allow g729 |
| 15:16.27 | Greenlight | If they don't, warning bells |
| 15:16.27 | Ice_Strike | £10/channel per month? |
| 15:16.30 | ketas | why does one need that? |
| 15:16.32 | Greenlight | Ice_Strike: Naa, one off |
| 15:16.37 | Greenlight | ketas: ? |
| 15:16.40 | Greenlight | ketas: Need what ? |
| 15:16.44 | igcewieling | Greenlight: maybe maybe not. If the phones are set to G729 and the ISP support g729 than most of the time most calls will not be transcoding, there WILL still be some transcoding, voicemail, IVRs, ringback, etc. so you may not actually need one license per call |
| 15:16.45 | ketas | a benefits of that codec |
| 15:16.47 | Lars_G | Greenlight: for g711 too? |
| 15:16.50 | Ice_Strike | Oh great |
| 15:17.05 | Greenlight | igcewieling: He;s a call centre; i presumed call recording |
| 15:17.21 | Greenlight | Lars_G: No, G711a/u is royalty free |
| 15:17.21 | igcewieling | Greenlight: if they only record ing f729 they will be OK. |
| 15:17.24 | Ice_Strike | each call is approx 20Kbps speed? |
| 15:17.32 | Greenlight | Ice_Strike: 30 with overheads |
| 15:17.38 | Greenlight | Wel, that's the maths I use :) |
| 15:17.43 | Greenlight | The codec is 8kbs! |
| 15:17.55 | Greenlight | If you IAX2 trunk it can get like 100 channels per mb! |
| 15:17.58 | Lars_G | thanks Greenlight |
| 15:18.07 | ketas | how come codecs end up with usage costs anyway |
| 15:18.09 | Lars_G | Greenlight: Is 729 that much better? |
| 15:18.24 | Ice_Strike | What g729 sound quality like? |
| 15:18.32 | igcewieling | if you transcode more than about 20 calls at a time I recommend a hardware transcoder card. g729 is CPU intensive |
| 15:18.33 | Greenlight | Lars_G: Depends what you mean by "better". Quality is worse, but not *that* noticable. |
| 15:18.34 | Ice_Strike | I usally convert to MP3 at midnight |
| 15:18.59 | Greenlight | Lars_G: The main reason to use g729 is the lower bandwidth requirements |
| 15:19.31 | Greenlight | And yea, for larger setups, the transcoding overhead cannot be sniffed at, either use a card, and a dedicated server(s) to transcode |
| 15:19.45 | Greenlight | s/and/or |
| 15:20.36 | reuf | how come there is no solution for ZeKoU |
| 15:20.48 | reuf | ZeKoU: only the sky is the limit |
| 15:20.55 | Greenlight | refu: Cause he has no control over firewall(s) |
| 15:21.08 | Kobaz | anyone know if you can put two ftp/tftp servers in option 66 for polycom phones |
| 15:21.10 | reuf | what if he had control over firewalls? |
| 15:21.18 | igcewieling | reuf: then everything changes |
| 15:21.25 | Greenlight | reuf: Then he'd be sorted |
| 15:21.31 | Kobaz | and will it round-robin the ftp servers |
| 15:21.47 | igcewieling | Kobaz: unlikely, but you might be able to point it to a hostname with multiple IP addresses |
| 15:21.48 | Ice_Strike | Which hardware transcoder card? |
| 15:21.59 | Greenlight | digium transcoder card g729 |
| 15:22.08 | igcewieling | Ice_Strike: Sangoma has one, as does Digirum |
| 15:22.13 | Ice_Strike | wow |
| 15:22.14 | Ice_Strike | £796.61 |
| 15:22.21 | Greenlight | Ice_Strike: If you can, stick with G711, imo |
| 15:22.27 | igcewieling | Ice_Strike: I never said it was cheap. |
| 15:22.30 | Ice_Strike | LOL |
| 15:22.31 | eirirs | Greenlight: that's alaw/ulaw? |
| 15:22.33 | Greenlight | Bandwidth is not that expensive in this country |
| 15:22.39 | Greenlight | eirirs: What ? |
| 15:22.51 | Ice_Strike | well the Server is i7 CPU |
| 15:22.53 | Greenlight | eirirs: Oh, G711 is yes |
| 15:22.58 | eirirs | ok |
| 15:23.34 | Lars_G | Greenlight: should g711 be usseable at 64k? |
| 15:23.52 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 15:24.03 | igcewieling | Lars_G: RTP overhead |
| 15:24.03 | Greenlight | Lars_G: That's the raw codec. With overheads it'll go up to 80Kbs. I usually allow 100Kbs for my calculations (am cautious!) |
| 15:24.13 | Ice_Strike | We have BT PBX here for 3 years with over 60 lines... soon we gonna move to SIP provider and use Internet connectin |
| 15:24.16 | Ice_Strike | It work out cheaper. |
| 15:24.39 | Greenlight | Ice_Strike: Indeed, SIP generally will be cheaper |
| 15:24.57 | Greenlight | What sort of call volumes you doing ? |
| 15:25.18 | *** join/#asterisk fireman_biff (~biff@65.48.133.102) |
| 15:25.53 | Ice_Strike | 8 hours calls from approx 50 agents |
| 15:26.06 | Greenlight | What mix between mobiles/landlines ? |
| 15:26.11 | Greenlight | This is outbound mostly, yes ? |
| 15:26.11 | Lars_G | ok |
| 15:26.19 | Ice_Strike | Yes outbound to mobile phones |
| 15:26.28 | Greenlight | Ice_Strike: All to mobiles ? |
| 15:26.31 | Lars_G | Greenlight: I've used the asterisk online calculator. What worries me at 64k is quality |
| 15:26.36 | Ice_Strike | Yep. |
| 15:26.38 | Lars_G | I wonder when g729 expires. |
| 15:27.12 | Ice_Strike | Greenlight Just 07 mobiles numbers |
| 15:27.34 | [TK]D-Fender | <Lars_G> Greenlight: I've used the asterisk online calculator. What worries me at 64k is quality <- doesn't make much sense |
| 15:27.48 | Greenlight | Ice_Strike: You paying more than 3ppm ? |
| 15:27.56 | fireman_biff | My "Goto(${CALLERID(dnid)},1)" is being run as "Goto("DAHDI/12-1", ",1")" according to the CLI. What's the best way to make sure that the DNID is set and is a number? |
| 15:28.52 | *** join/#asterisk gerhard7 (~gerhard7@82-169-24-72.ip.telfort.nl) |
| 15:29.31 | [TK]D-Fender | fireman_biff, "core show application gotoif" , "core show function IF" <- |
| 15:29.47 | Ice_Strike | Greenlight I am not sure, my boss pay the bill |
| 15:30.53 | Greenlight | Ice_Strike: I push a lot of call traffic at present. If you find out we should talk :) |
| 15:32.04 | Greenlight | Ice_Strike: Also, if you're moving to SIP with call centre type traffic make sure you get *at least* 3 decimal place billing; makes a MASSIVE difference for that type of traffic, and SIP sales people try to hide it |
| 15:32.40 | Ice_Strike | Greenlight Currently 9 agents on the calls that connected to SIP provider for testings.. other agents are connected to BT PBX |
| 15:33.07 | Ice_Strike | The one that are connected to SIP provider - £0.028 |
| 15:33.29 | Ice_Strike | :) |
| 15:33.39 | Kobaz | if you did option 150, which is ip address |
| 15:33.47 | Kobaz | i wonder if you can push the phone two servers |
| 15:34.00 | Kobaz | apparently cisco switches can push out two addresses on option 150 |
| 15:34.31 | Greenlight | Ice_Strike: If they give CDR's check that you get billed to 3 decimal places or more. Eg, so that a call to an answer machine costs £0.001 and NOT £0.01 (You can imagine the difference it makes) |
| 15:35.26 | Ice_Strike | Yes it is deffo billed to 3 decimal places from a sip provider. |
| 15:35.34 | Ice_Strike | I can see it from the billing page |
| 15:36.01 | Greenlight | That's good - and £0.028 to mobiles is a decent price for that volume of traffic. |
| 15:36.07 | Ice_Strike | Yep. |
| 15:36.22 | Ice_Strike | We are moving out from BT PBX soon :) |
| 15:36.22 | fireman_biff | [TK]D-Fender: thanks, would it be enough to do "GotoIf($["${CALLERID(dnid)}" != ""]?${CALLERID(dnid)},1)" or should I actually confirm that the dnid is numeric? |
| 15:36.26 | Greenlight | I beleive our resale rate card is a little higher at that volume at present |
| 15:37.07 | Lars_G | [TK]D-Fender: Why doesn't it make sense? |
| 15:37.26 | Ice_Strike | I need maximum 70 concurrents calls - I hope 6mbit download and 6mbit upload would be enough |
| 15:37.37 | Greenlight | Ice_Strike: How many agents ? |
| 15:37.45 | Ice_Strike | 70 |
| 15:37.52 | Greenlight | Ice_Strike: You're not using a DIaller ? |
| 15:38.00 | [TK]D-Fender | Lars_G, How is getting more BW for your call supposed to result in LOWER quality? |
| 15:38.01 | Ice_Strike | Yes using dailler |
| 15:38.22 | Greenlight | Predictive dialler's will dial more calls than there are agents. Unless this is a hosted system is it ? |
| 15:38.40 | [TK]D-Fender | fireman_biff, I might confirm it first... your call... |
| 15:38.59 | Lars_G | [TK]D-Fender: WHo said anything about that? |
| 15:39.27 | Ice_Strike | Greenlight It is our own Predictive dialler |
| 15:39.31 | Greenlight | Lars_G: The g711 a/u codecs are not variable bandwidth. They are fixed at 64kbs. This is the same quality as a mobile phone call. |
| 15:39.41 | fireman_biff | [TK]D-Fender: cool, thanks a lot |
| 15:39.50 | Lars_G | Greenlight: Ok, I kinda remembered 711 was used in GSM but was unsure |
| 15:40.08 | [TK]D-Fender | Lars_G, You seem to be right there.... |
| 15:40.10 | Lars_G | Greenlight: So 729's trick is it's vbr with a top at xKb... |
| 15:40.11 | Greenlight | Ice_Strike: But, by defenition, a predictive dialler will dial more calls than there are agents. For example at times I see 150 agents and at start of shift we can be dialling 300+ calls |
| 15:40.35 | Greenlight | Lars_G: g729 is NOT variable bandwidth either, it just uses less |
| 15:40.58 | Lars_G | [TK]D-Fender: Nope I'm not. right there I said I used the asterisk calculator, to calculate transfer with overhead. I do say I'm worried about quality but I say nothing about more (or less) bandwidth, there or anywhere. |
| 15:41.03 | Lars_G | Greenlight: Ok. |
| 15:41.12 | Lars_G | Greenlight: I'd guess vbr is not too desirable in voip. |
| 15:41.19 | Lars_G | As it'd make network calculations impossible |
| 15:41.45 | Greenlight | Lars_G: G711 at 64Kbs is quite fine for telephone quality calls. |
| 15:41.58 | Ice_Strike | Greenlight Hmmmmm that is not the way it work here. When customer hang up the call - it will automatically dail to next number and agent have to wait. |
| 15:42.08 | Lars_G | Greenlight: What I need to find now is a rough estimate on T38 bw, but I think all of this is manageable. |
| 15:42.29 | Greenlight | Ice_Strike: Ahh, that;'s not a predictive dialler. That's a progressive dialler. We should speak - my company does SIP and DIaller systems. |
| 15:42.33 | [TK]D-Fender | Lars_G, "I do say I'm worried about quality" <- you say this about 64kbps... that is what ULAW / ALAW uses on the PSTN. There is nothing bigger in the real world. How do you expect LESS quality for using that? |
| 15:42.39 | [TK]D-Fender | Lars_G, G.729 is LESS quality |
| 15:42.44 | [TK]D-Fender | Lars_G, as is GSM |
| 15:42.57 | [TK]D-Fender | Lars_G, Your sense of direction seems reversed |
| 15:43.15 | Lars_G | [TK]D-Fender: No, I DO NOT say i'm worried about that around 64kbps |
| 15:43.50 | Greenlight | Lars_G: The way to best imagine it is that g711 at 64Kbs is the "base" telephone quality. |
| 15:44.07 | Ice_Strike | Greenligh I let you know ;) What billing rate you can do? ;) |
| 15:44.17 | Lars_G | [TK]D-Fender: Btw, since I've clarified that is not the case, why do you keep arguing my sense of direction is backwards? as I've established that was a missunderstanding on your/our side? |
| 15:44.44 | Lars_G | [TK]D-Fender: Or you mean 729 has less quality than 711? |
| 15:44.59 | [TK]D-Fender | Lars_G, G.729 < G.711 |
| 15:45.28 | Lars_G | [TK]D-Fender: Yes I reread and it seems I missread Greenlight there |
| 15:46.17 | Lars_G | Any ideas when g729 expires? or, is there an option for free 729 for asterisk for countries where there are no software patents? or is 729 outside of the software realm of patents? |
| 15:46.22 | Greenlight | Ice_Strike: Well at 60 agents, I can say that we'd increase "talk time" by around 50-100% based on past setups, using the Dialler/CRM system. At that level I could beat your call rates as well. |
| 15:47.02 | Greenlight | Lars_G: There's an educational version "out there". |
| 15:47.15 | Greenlight | Lars_G: Expires - no time soon :) |
| 15:47.15 | Lars_G | ok |
| 15:47.28 | Lars_G | ok |
| 15:47.57 | Greenlight | Lars_G: £10/channel one of cost is hardly expensive though if you do end up using it |
| 15:48.11 | drmessano | I find some of these "newer" codecs more interesting.. Like Speex and SILK |
| 15:48.12 | Greenlight | Lars_G: Also, the Digium verison is better than the "educational" one |
| 15:48.32 | Greenlight | Speex is used quite a lot in other voip applications |
| 15:48.42 | Lars_G | Greenlight: a) when I say my country doesn't has software patents, it's not an excuse to pirate stuff. b) I cannot purchase anything in foreign currency |
| 15:48.44 | drmessano | The "educational" g729 is a bit buggy... |
| 15:48.46 | Greenlight | It's variable bandwidth I think |
| 15:49.03 | Lars_G | drmessano: First time I see speex called newer :) |
| 15:49.37 | drmessano | Newer than G729 and G711 |
| 15:49.46 | igcewieling | the thing about the illegal g729 codec is that NOBODY will help you. In fact, it is the only topic I am aware of which is censored on the Digium mailing lists. |
| 15:49.53 | Greenlight | Lars_G: I'm not sure of the export regulations or patent laws, sorry, but Digium don't give it away to such countries ;/ |
| 15:50.02 | Lars_G | Greenlight: I mean, maybe I could, but I'd have to make a VERY strong case, up to the country's vice president, and it'd take about two years I'd expect |
| 15:50.13 | Greenlight | Lars_G: You need a new country |
| 15:50.24 | Lars_G | Really? |
| 15:50.25 | Lars_G | Wow |
| 15:50.26 | igcewieling | Lars_G: then your only option is to buy a G729 transcoding card locally. |
| 15:50.27 | Lars_G | thanks. |
| 15:50.28 | Greenlight | lol |
| 15:50.36 | [TK]D-Fender | Lars_G, Then find someone local to resell it to you in yours |
| 15:50.37 | Greenlight | Sorry only joking with you |
| 15:51.08 | igcewieling | Greenlight: if their country's policies are such that he cannot spend $10/channel of a G729 codec then he needs to find a better country. |
| 15:51.44 | Lars_G | Is 729 really SO vital? |
| 15:51.45 | Greenlight | Some countires are under import/export embargos and the such |
| 15:52.05 | igcewieling | Lars_G: only if you need to save bandwidth or if your carrier requires it. |
| 15:52.05 | [TK]D-Fender | igcewieling, Parallels they irony of "zero tolerance" policies being interchangeable with "zero intelligence" policies. |
| 15:52.07 | Greenlight | Lars_G: I don't know, is it ? What are you trying to do ? |
| 15:52.15 | [TK]D-Fender | igcewieling, Some people love shooting themselves in the foot |
| 15:52.36 | Lars_G | igcewieling: And if your analysis capability is such that you jump to such a conclusion from a single line of information from a single user in irc, you need better analysis capabilities :D |
| 15:52.42 | igcewieling | for example our wholesale carriers ONLY send calls TO us as G729, they will accept calls FROM us using ulaw or g729 |
| 15:52.47 | drmessano | Those countries where everyone wants to use G729 over VPN over GSM over carrier pigeon to get around the local telco seem to have restrictions on G729. Shocking |
| 15:52.59 | Lars_G | Greenlight: Actually, what I'm trying to do, is to be finally able to rip this Panasonic PBX out of the wall, and chuck it out the window |
| 15:53.39 | Greenlight | Lars_G: For that you merely need a crowbar, not g729 :) |
| 15:53.51 | Greenlight | A hammer can make it more fun as well |
| 15:53.53 | igcewieling | [TK]D-Fender: like hunting deer without your contacts/glasses -- it will not end well. |
| 15:54.07 | Greenlight | has had "experiences" with Panasonic PBX's |
| 15:54.27 | Greenlight | Lars_G: Joking aside, you're ISDN or SIP at present ? |
| 15:55.40 | drmessano | I replaced 3 NEC PBX's when we went Asterisk. They wanted to save them "Just in case". Somehow, much like the lamp in "A Christmas Story", they ended up with broken controller cards |
| 15:55.40 | Lars_G | Greenlight: None. ISDN never reached here. I will use SIP internally. I'll feed this from a few FXO gateways into asterisk. |
| 15:55.51 | drmessano | :( <--- |
| 15:56.13 | Lars_G | Greenlight: I have around 14 analog COs comming in from the street into the FXB, I'm too small to ask for an E1 either. |
| 15:56.51 | Lars_G | Greenlight: With that I'll feed 86 internal extensions (agents?), plus open two channels over internet (vpn) to our ministry. |
| 15:56.57 | *** join/#asterisk Leddy (leddy@krypton.evosurge.com) |
| 15:57.08 | Greenlight | Lars_G: So you're wanting to move over to just SIP? Or keep the existing connections ? |
| 15:57.49 | Lars_G | Internally I will move completely to only SIP. Not keep current wiring hot. |
| 15:58.00 | Greenlight | And, externally ? |
| 15:58.05 | Lars_G | And externally, I'll use an FXO interface at the FXB for lines. |
| 15:58.15 | Lars_G | analog lines only. |
| 15:58.24 | Greenlight | So, no need to worry about G729 at all |
| 15:58.46 | Greenlight | You'll have fun with all those analogue lines, but am sure you're used to that already :) |
| 15:58.47 | Lars_G | Well, I might want to use 729 for our two channels to the external ministry, to save bw, but that is in the long future |
| 15:58.52 | Lars_G | Yeah... |
| 15:58.54 | Lars_G | lot's of fun. |
| 15:59.19 | Lars_G | Starting with the fact we have canadian fxbs. (Bix), just getting a punch tool here was hell on earth |
| 15:59.20 | Greenlight | That's where the hammer and the Panasonic PBX help... |
| 15:59.29 | Lars_G | Not to mention the pbx is ailing, and slowly ports on the cards in it die out |
| 15:59.50 | Lars_G | yuppers |
| 15:59.52 | Lars_G | thanks for the feedback |
| 15:59.57 | Greenlight | So, you use G711a/u internally, and you're sorted. No need to worry about g729 |
| 15:59.58 | Lars_G | I hope I get none on the lines. |
| 15:59.59 | Greenlight | Or licenses |
| 16:00.06 | Lars_G | right |
| 16:00.25 | Lars_G | And since the ministry uses asterisk too, I could agree to any sort of codec with them for the exchange |
| 16:00.30 | Lars_G | we're not limited by that either |
| 16:00.47 | Greenlight | When it comes to connecting that many analogue lines up I'm not sure what the best plan is; I've only used digital lines - sure someone can chip in and advise tho |
| 16:00.59 | Lars_G | Ok. |
| 16:01.16 | Katty | HI LADS. |
| 16:01.18 | Lars_G | I was thinkin on using 3x FXOx8 gateways (for expansibility int he future) |
| 16:01.26 | igcewieling | Lars_G: BTW, you would likely actually use alaw, since that is what the PSTN in your coutnry likely uses, but alaw and ulaw or so similar they are not different in any meaningful way |
| 16:01.33 | Lars_G | Since my receiving point is 4 floors bellow my server room |
| 16:01.51 | Greenlight | Yea, that's what I was thinking |
| 16:02.02 | Greenlight | 4 floors |
| 16:02.04 | Greenlight | Hmm |
| 16:02.12 | Lars_G | igcewieling: I give you an idea, until at least one year ago, my country used r2 for E1 signaling, which I understand is archaic |
| 16:02.34 | Lars_G | Greenlight: And the FXB is a small box, behind reception in the hall, so I can't healthily house a full pc there |
| 16:02.41 | igcewieling | Where are you, the Pitcairn islands? |
| 16:03.47 | Lars_G | Greenlight: Are you a telco guy? |
| 16:04.20 | Greenlight | Lars_G: Not particularly - just what I've picked up from installing and playing with Asterisk I guess |
| 16:04.45 | Lars_G | Ok. |
| 16:04.51 | Greenlight | Lars_G: We've had plenty of customers with weird and wonderful setups, so get to see a lot :) |
| 16:05.26 | Lars_G | I was asking in case you understood what a pain it is to try and find unbroken pairs in a cable loop installed 60 years ago on a 6 story, two wings building. |
| 16:05.45 | Greenlight | Oh, that I know! |
| 16:06.25 | Lars_G | I've been dreaming of asterisk for two years, it seems this year it'll come to fruition. Wish me luck |
| 16:06.47 | Greenlight | Least with moving to SIP, you can manage things a bit better - just 2 pairs and you've an "uplink" for the switch |
| 16:06.56 | igcewieling | Lars_G: that is why we usually outsource cable stuff. |
| 16:07.51 | *** join/#asterisk ujjain (ujjain@unaffiliated/ujjain) |
| 16:08.44 | Ice_Strike | Greenlight Cool stuff, I let you know but it has to be without Dialler system. Just sip. |
| 16:11.41 | Lars_G | What are Dialer systems? |
| 16:11.48 | Lars_G | Termination services? |
| 16:12.40 | navaismo | Lars_G, those system that call you every morinig offering services, or delivering messages, annoying robots |
| 16:12.48 | igcewieling | Lars_G: dialer systems are used by telemarketers and collection agencies |
| 16:13.16 | igcewieling | When you use one you forfeit your soul. |
| 16:13.16 | Lars_G | Ah yes |
| 16:14.33 | *** join/#asterisk Katty (~Katty@97-91-121-70.static.stls.mo.charter.com) |
| 16:20.15 | Greenlight | yea, generally people seem to dislike telemarketing, and therefore diallers. It has, and I suspect always will be, the cheapest form of marketing, and therefore companies using it can usually deliver products cheaper, so it's not all evil |
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| 16:20.40 | igcewieling | It is evil. |
| 16:20.48 | Greenlight | :) |
| 16:21.22 | igcewieling | Dialers can be used for good, but they are usually used for evil. |
| 16:21.31 | Greenlight | Like guns I guess |
| 16:21.46 | Greenlight | But you're right, there's a LOT of miuse and mismanagement |
| 16:22.06 | igcewieling | Greenlight: guns are usually used for good, it is the occasions where they are used for evil which make the news. |
| 16:22.20 | igcewieling | but the ratio is not even close to that of dialer software |
| 16:22.25 | Greenlight | I get hugely frustrated to get an automated recorded message call, since I know that is illegal |
| 16:23.00 | Greenlight | At least in the UK |
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| 16:34.54 | Lars_G | igcewieling: Like us |
| 16:35.05 | Lars_G | igcewieling: We have huge mailing lists, for press anoucements. |
| 16:35.17 | Lars_G | igcewieling: But every single address in that list, is someone who subscribed themselves |
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| 16:35.55 | Lars_G | igcewieling: Of course we have the eventual user who subscribes. Then when we mail them, reports us as spam to their ISP. then, they stop receiving stuff, and they try resubscribing again |
| 16:35.58 | igcewieling | Lars_G: you call them all to tell them you have a new press release? Otherwise it is not comparable |
| 16:36.08 | Lars_G | igcewieling: One of these days I'm gonna meet one of the morons. |
| 16:36.20 | navaismo | XD |
| 16:36.23 | Lars_G | igcewieling: Yeah I know. At least in my country dialers have not reached here yet. I'm safe |
| 16:36.45 | WIMPy | Lars_G: Where are you? |
| 16:36.46 | Lars_G | igcewieling: I've heard both about your telemarketers, and your political campaign robot dialers too |
| 16:36.50 | Lars_G | WIMPy: Venezuela |
| 16:37.18 | igcewieling | some non-evil uses for dialers include notification of school closings or other school related stuff, reminder of doctor's appointments. |
| 16:37.26 | Lars_G | Yeah |
| 16:37.33 | Lars_G | Speaking of which |
| 16:37.47 | Lars_G | part of what I want with this is the possibility of an IVR. |
| 16:38.16 | Lars_G | Originally I wanted to "cast" a gurl in the company to record the message snippets... but I worry how normal it is to be makign tweaks and changes constantly. |
| 16:38.27 | Lars_G | Another option would be text to speech, but that sounds hideous |
| 16:38.48 | Lars_G | In your exprience does an IVR tree tend to be pretty stable? or do you have to change it a lot? |
| 16:40.13 | navaismo | depend on your application |
| 16:40.26 | Greenlight | has just got of a phone call about changing IVR's |
| 16:40.50 | Greenlight | They also "auditioned" a girl to be the voice of the company |
| 16:40.56 | navaismo | some iVR never changes other are very dynamic and need to change everyday |
| 16:41.16 | Greenlight | As navaismo says, it really depends on your application |
| 16:41.22 | Lars_G | Ok |
| 16:41.29 | igcewieling | Hire Allison Smith |
| 16:41.44 | Greenlight | That ^^ |
| 16:42.08 | paulc | The nice thing with Allison is that she "gets" IVR - in terms of tone, inflection, call flow etc... more than just a pleasant voice.. |
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| 16:49.34 | BorjaGVO_ | Hi everyone, |
| 16:50.08 | Greenlight | Hi BorjaGVO_ |
| 16:50.49 | BorjaGVO_ | AGENTLOGIN and AGENTLOGOFF events are not in my queue_log. I don't know if there is anything I can do so that these events are enabled. I thought that this was enabled by default (I don't have any agents being assigned dinamically) |
| 16:51.15 | BorjaGVO_ | Hi Greenlight |
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| 17:23.23 | tzanger | hm, this is something I haven't runa cross yet... is there a definitive "this is the license for asterisk/dahdi" page anywhere on asterisk's site? |
| 17:23.35 | tzanger | I see it in the code but something I could point out to a suit |
| 17:24.30 | leifmadsen | tzanger: I'd try emailing Malcolm about that maybe? |
| 17:26.54 | tzanger | what's his email address? |
| 17:27.19 | tzanger | (I don't know his last name or I'd search myself) |
| 17:30.28 | [TK]D-Fender | tzafrir_laptop, I seem to recall GPL across theboard |
| 17:31.12 | tzafrir_laptop | All the files generally refer to the top-level file LICENSE file, |
| 17:31.32 | tzafrir_laptop | which is (for Asterisk) GPL (v2) with extra two exceptions |
| 17:31.53 | leifmadsen | tzanger: good point by tzafrir_laptop, just link to svn.asterisk.org/svnview/ |
| 17:32.37 | *** join/#asterisk ujjain (~ujjain@unaffiliated/ujjain) |
| 17:33.03 | tzanger | heh that errors out |
| 17:33.32 | tzanger | tehre we go |
| 17:33.40 | tzanger | svnview.digium.com/svn/ |
| 17:43.49 | tzanger | so if I understand the two exceptions are a) explicit permission to link with openssl/h323/uw-imap and to state that the agi/ami protocol use does NOT create a "derivative work" ? |
| 17:57.36 | tzanger | hm, so if I make res_tzanger.so and I do not negotiate a different license with Digium, does res_tzanger.so have to be GPLv2-compatible? The Asterisk license file makes specific exception for a few libraries but doesn't explicitly say for custom modules |
| 17:57.55 | tzanger | for DAHDI I believe a custom driver must be GPLv2-compatible |
| 17:57.59 | tzanger | (to use DAHDI APIs) |
| 18:03.21 | Lars_G | Btw, what is your favorite/best/etc asterisk web ui? |
| 18:04.25 | Greenlight | Lars_G: The most common one is likely FreePBX |
| 18:04.43 | pigpen | Lars_G, yeah, honestly, I haven't seen one I like. |
| 18:05.33 | WIMPy | The best one is the one you will write for yourself. |
| 18:05.43 | pigpen | ^^^ exactly what I am doing |
| 18:06.00 | pigpen | I had a few guys that worked for me write a few different vers, but never liked them. |
| 18:06.44 | pigpen | so, I am writing a bash script ver that sticks it all in a DB, then later will come back and write a Web UI. At least this way, two different ways to access, both SSH and Web if necessary. |
| 18:07.08 | pigpen | Honestly, if in a db, nothing more is needed than knowledge of dialplan and a good editor like navicat. |
| 18:07.14 | WIMPy | bash. wow. |
| 18:07.38 | pigpen | yeah, well, I don't know php, python, C or such. |
| 18:07.47 | pigpen | but I can do pretty good with a simple script |
| 18:07.59 | pigpen | and it is nothing more than a Q & A and a bit of DB shuffling. |
| 18:08.05 | pigpen | why get elaborate. |
| 18:08.45 | WIMPy | Quite a lot of work to do it in bash. |
| 18:09.03 | Ice_Strike | Why would you want to write in bash? |
| 18:09.06 | Ice_Strike | Bash is slow. |
| 18:09.37 | Ice_Strike | You would write less code in PHP, Perl or Python than in bash |
| 18:09.45 | Ice_Strike | and do same thing |
| 18:10.19 | pigpen | Ice_Strike, What vehicle to you have? |
| 18:11.40 | Ice_Strike | BMW and Golf |
| 18:11.42 | Ice_Strike | Why? |
| 18:12.32 | pigpen | BMW Why? Too much maintenance, costs to much, insurance is high, low ground clearance. You should drive a Dodge 3500 Dually, or a Chevy Silverado 2500. |
| 18:12.43 | pigpen | Oh, but that is your choice right? Just like mine. '-) |
| 18:13.33 | pigpen | But I am sure, just like your Beemer, it gets you from Point A to B. Just like my scripts. They work. Every time. (unlike your beemer) |
| 18:14.20 | WIMPy | Some will take you a lot faster from A to B. |
| 18:15.12 | Lars_G | Greenlight: Thanks |
| 18:15.20 | pigpen | heh, but think about it. updating 3 entries into a psql db. How much time are we talking about here? You can type and hit enter just so fast. |
| 18:15.24 | Lars_G | WIMPy: yeah I am affraid of that, but it might end up with it. |
| 18:15.52 | Lars_G | I wonder if I could integrate this into my ldap tree eventually, and write a FusionDirectory plugin |
| 18:15.54 | pigpen | splitting hairs in my opinion. And if I had time, yeah, I would prefer a higher level language. But just don't have the time at the time. |
| 18:16.28 | WIMPy | It obviousely depends on how much your script does. |
| 18:16.37 | [TK]D-Fender | hears "Gentoo" in the wind.... |
| 18:16.39 | Ice_Strike | pigpen That is silly comparision. Bash is made for different purpose. |
| 18:16.55 | WIMPy | If you only want to go nerxt house, you will be faster by foot than with any car. |
| 18:18.31 | pigpen | Ice_Strike, Bash scripts can be written for many purposes. But tell you what. You come work for free, do my job, make me money and I'll learn another language. Just like your beemer. You have a right to buy it and like it. It is your choice. |
| 18:18.48 | pigpen | Oh, and it is a perfect comparison, you just don't like it when the tables are turned. Man up. |
| 18:20.26 | Ice_Strike | LOL! |
| 18:20.34 | Ice_Strike | pigpen Take it easy dude |
| 18:25.59 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 18:26.58 | pigpen | WIMPy, Very true. The scripts do very little. Short and sweet. The last thing I need is to give customers too many options. Now with Gentoo (haha) |
| 18:27.17 | pigpen | [TK]D-Fender, Good memory bty. |
| 18:35.37 | *** join/#asterisk blizzow (~jburns@75-171-154-172.hlrn.qwest.net) |
| 18:40.08 | edong23 | pigpen: ever heard of bash on bawls? |
| 18:40.31 | edong23 | bash on balls |
| 18:40.32 | edong23 | sorry |
| 18:40.39 | *** join/#asterisk vlad_starkov (~vlad_star@178.177.197.139) |
| 18:40.46 | edong23 | its a web development suite built on bash |
| 18:40.47 | edong23 | lol |
| 18:42.15 | *** join/#asterisk mattchis (~mattchis@75-145-122-77-Colorado.hfc.comcastbusiness.net) |
| 18:52.06 | *** join/#asterisk bn-7bc (~bjarne-im@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 18:52.26 | pigpen | too funny. |
| 18:55.33 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 18:57.27 | *** join/#asterisk garymc (~chatzilla@host86-174-143-125.range86-174.btcentralplus.com) |
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| 19:12.59 | edong23 | pigpen: i played with it. I didnt get far, but it was kinda cool |
| 19:13.00 | edong23 | lol |
| 19:13.11 | edong23 | i dont think it would ever compete with pylons or something |
| 19:20.58 | *** join/#asterisk miltux (~miltux@62.1.139.246.dsl.dyn.forthnet.gr) |
| 19:30.54 | *** join/#asterisk CunningPike (~CunningPi@d28-23-24-84.dim.wideopenwest.com) |
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| 19:43.39 | *** join/#asterisk themrrobert_ (cf4322ee@gateway/web/freenode/ip.207.67.34.238) |
| 19:45.19 | themrrobert_ | having some serious issues with AMI timing out. we've got software connecting to asterisk via dedicated nic card, point-2-point (no switches) and it still times out |
| 19:45.58 | themrrobert_ | not always, just like 1/20 or 1/30 commands, and it just times out, the server was changed to use threading, but then it gets to the point where 20 threads are waiting on responses from the server |
| 19:46.02 | WIMPy | time out in what way? |
| 19:46.12 | themrrobert_ | no response is received from the server |
| 19:46.47 | WIMPy | And what is "the server"? |
| 19:47.03 | themrrobert_ | in this case i meant from asterisk sorry |
| 19:47.40 | themrrobert_ | the asterisk server is only using 3% - 5% of capacity, so i don't understand what would delay it so much |
| 19:48.11 | WIMPy | So it's your application timing out, waiting for a response from Asterisk? |
| 19:48.57 | themrrobert_ | yes, after 30 seconds i believe, should ami take that long? example i could dial in the queue manually and log in in less time |
| 19:49.28 | WIMPy | That surely depends on what you're doing. |
| 19:49.30 | Qwell | Have you looked at a trace of the connection, to make sure things are being sent properly? |
| 19:50.18 | WIMPy | ... and what Asterisk is actually sending. Maybe it's sending somethign unexpected? |
| 19:50.20 | themrrobert_ | ooh thats a good idea. i'll have to do that. but most likely they are as it processes multiple similar actions |
| 19:50.26 | themrrobert_ | and most go thru without problem |
| 19:51.02 | themrrobert_ | its checking for any response including the action id |
| 19:51.13 | [TK]D-Fender | themrrobert, On what version exactly? |
| 19:51.21 | themrrobert_ | 11.0.1 |
| 19:51.45 | [TK]D-Fender | 2 behind... |
| 19:58.24 | themrrobert_ | i'll compile today, upgrade tonight, test tomorrow, and return with results |
| 19:58.48 | themrrobert_ | do you anticipate i will need to add any debugging compiler options or is optimize ok |
| 20:00.02 | *** join/#asterisk Alagar (~helpdesk@vsusg1.vernalissystems.com) |
| 20:03.08 | *** join/#asterisk ed207 (~ed207@ip24-253-68-188.lv.lv.cox.net) |
| 20:03.10 | ed207 | hello |
| 20:03.47 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 20:05.19 | *** join/#asterisk kresp0 (~screspo@178.Red-80-33-64.staticIP.rima-tde.net) |
| 20:05.39 | ed207 | Figures :) |
| 20:05.52 | themrrobert_ | hey ed |
| 20:06.02 | ed207 | Hi |
| 20:06.09 | ed207 | I'm about to go insane here :) |
| 20:06.17 | themrrobert_ | ditto ;) what's your poison |
| 20:06.22 | ed207 | Using 1.6.1.12 |
| 20:06.28 | ed207 | Have Realtime Queues |
| 20:06.36 | ed207 | Everything is working fine on one machine |
| 20:06.37 | WIMPy | Don't do that. |
| 20:06.48 | ed207 | It's what I got right now |
| 20:06.59 | themrrobert_ | Ya go somewhere else to go insane :p hehe |
| 20:07.06 | ed207 | My problem is that the app timeout to go to another queue in the dialplan is not working |
| 20:07.07 | [TK]D-Fender | Get something else |
| 20:07.19 | ed207 | It will work in normally with the DID |
| 20:07.20 | WIMPy | ~upgrade asterisk |
| 20:07.21 | infobot | Before requesting assistance, you should be running the latest version of a supported release branch. See the channel topic for the latest versions available in currently supported branches. |
| 20:07.37 | ed207 | This is in production right now with callers |
| 20:07.50 | ed207 | I need to know why the app timeout is being ignored |
| 20:08.15 | leifmadsen | show examples and outputs |
| 20:08.15 | ed207 | It's not ignored in one place in the dialplan, but it is in another |
| 20:08.20 | ed207 | Makes no sense |
| 20:08.36 | ed207 | exten => 1987192837415091863440916235976,n,Queue(ESTR1111_CUSTSERVICE,,,,90) |
| 20:08.58 | ed207 | That's just a passthru from one asterisk machine to antoher with an IAX2 tunnel |
| 20:09.02 | ed207 | It works |
| 20:09.11 | ed207 | After 90 seconds it drops into another queue |
| 20:09.31 | [TK]D-Fender | ~pb |
| 20:09.31 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 20:09.32 | [TK]D-Fender | ^^^^ |
| 20:09.38 | ed207 | exten => 1987192837415091863440916235976,n,Queue(ESTR1111_GENERAL,,,,90) |
| 20:09.46 | [TK]D-Fender | show the complete trace of the call that DOESN'T work as expected |
| 20:10.26 | ed207 | Trace? |
| 20:10.28 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 20:10.44 | ed207 | You want all the logs? |
| 20:11.14 | *** join/#asterisk rgsteele (~rgsteele@12.150.6.65) |
| 20:11.19 | ed207 | Just to be clear, your not asking for a SIP trace are you? |
| 20:11.31 | ed207 | You just want the message logs from Asterisk? |
| 20:14.29 | [TK]D-Fender | * CLI |
| 20:14.31 | [TK]D-Fender | verbose 10 |
| 20:17.48 | ed207 | ok, I must be SOS today |
| 20:18.06 | ed207 | I do -r with 10 v's and I get verbosity of 63 |
| 20:18.13 | ed207 | What is the CLI command to change verbosity |
| 20:18.16 | ed207 | I can't find it |
| 20:18.40 | WIMPy | core set verbose |
| 20:19.13 | ed207 | ty |
| 20:22.05 | [TK]D-Fender | 63 is every bit as good as 10 |
| 20:25.21 | ed207 | Never used Pastebin before |
| 20:25.25 | ed207 | I think this is it |
| 20:25.27 | ed207 | http://pastebin.com/S1rgXBpY |
| 20:25.35 | *** join/#asterisk lorsungcu (~anonymous@75-146-164-137-Minnesota.hfc.comcastbusiness.net) |
| 20:25.40 | ed207 | This is the dialplan portion being executed |
| 20:25.43 | ed207 | exten => 2,1,Playback(estrella/internalmsg_transfercall) |
| 20:25.43 | ed207 | exten => 2,n,Queue(ESTR1111_CUSTSERVICE,Rtwkx,,,,90) |
| 20:25.43 | ed207 | exten => 2,n,Queue(ESTR1111_GENERAL,Rtwkx,,,,,90) |
| 20:26.26 | ed207 | After 90 seconds it does not go to the next line in the dialplan |
| 20:26.35 | ed207 | After 180 seconds it just drops the call |
| 20:26.40 | leifmadsen | suspects error in queues.conf |
| 20:26.47 | [TK]D-Fender | ed207, PASTEBIN <- |
| 20:26.59 | [TK]D-Fender | ed207, Show us the CALL and the dialplan |
| 20:26.59 | leifmadsen | well, not error, but rather, a misconfiguration for what you're expecting to happen |
| 20:27.21 | ed207 | queues.conf is empty |
| 20:27.27 | ed207 | It's realtime |
| 20:27.37 | [TK]D-Fender | ed207, I also already SEE that you don't have the same number of commas right there between the options and the TIMEOUT <- |
| 20:27.39 | ed207 | Timeout for retying agents is 15 seoncds |
| 20:28.00 | [TK]D-Fender | <ed207> exten => 2,n,Queue(ESTR1111_CUSTSERVICE,Rtwkx,,,,90) <- 4 |
| 20:28.07 | [TK]D-Fender | <ed207> exten => 2,n,Queue(ESTR1111_GENERAL,Rtwkx,,,,,90) <- 5 |
| 20:28.16 | [TK]D-Fender | Clearly not in the same place |
| 20:28.34 | ed207 | But it is correct for the CUSTSERVICE? |
| 20:28.43 | ed207 | According to documentation I can find |
| 20:28.54 | ed207 | Next line is a typo, I see that |
| 20:29.07 | [TK]D-Fender | ed207, "core show application queue" <- if you're looking anywhere other than there, you're looking in the wrong place |
| 20:29.36 | [TK]D-Fender | ed207, We'll have more to say when we see real configs and real calls |
| 20:31.09 | ed207 | Well queues.conf is empty |
| 20:31.19 | ed207 | Or I should say, commented out |
| 20:31.38 | [TK]D-Fender | ed207, You are finding ever more ways of not sowing us the calls or your setup..... |
| 20:31.46 | leifmadsen | well when I said queues.conf, I meant your queue configuration, whether it is a flatfile or the realtime config |
| 20:32.11 | *** part/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:32.19 | ed207 | I'll put a screenshot of the SQL dump for the queue |
| 20:32.51 | *** join/#asterisk elico (~Thunderbi@bzq-79-181-219-40.red.bezeqint.net) |
| 20:33.45 | *** join/#asterisk garymc (~chatzilla@host86-174-143-125.range86-174.btcentralplus.com) |
| 20:34.24 | ed207 | http://pastebin.com/6pEB14Xq |
| 20:35.12 | ed207 | queus.conf http://pastebin.com/ebHqdCRh |
| 20:35.55 | ed207 | After reading the core show application queue this is what I think the syntax should be: |
| 20:35.56 | ed207 | exten => 2,n,Queue(ESTR1111_CUSTSERVICE,Rtwkx,options,url,announceoveride,90) |
| 20:36.08 | ed207 | So the 90 is in the correct place? |
| 20:36.36 | ed207 | sorry |
| 20:36.46 | ed207 | exten => 2,n,Queue(ESTR1111_CUSTSERVICE,Rtwkx,url,announceoveride,90) |
| 20:37.57 | [TK]D-Fender | What do the instructions say? |
| 20:37.58 | hackish | First real test here... so I've got a ciso ATA properly registering its 2 lines via SIP. I want to setup an extension so I can dial 200 to call the other line. In sip I have redphone and bluephone. Then in extensions I have [redphone] with 200,1,Dial(sip/?????)... I'm not clear on what to put in dial, the SIP name for the other phone or the context it's associated with? |
| 20:38.32 | [TK]D-Fender | hackish, the device name in sip.conf |
| 20:39.19 | ed207 | Instructions say : Queue(queuename[,options[,URL][,announceoverride][,timeout][,AGI][,macro][,gosub][,rule]): |
| 20:39.30 | ed207 | I believe my syntax is correct? |
| 20:41.50 | ed207 | I guess not |
| 20:41.56 | hackish | TK, by device name you mean the contents of [redphone] so Dial(SIP/redphone) |
| 20:43.05 | [TK]D-Fender | hackish, if that is your peer entry name in sip.conf, then yes |
| 20:43.21 | hackish | ok, must be something else wrong then. |
| 20:44.09 | ed207 | Some days you just should not get out of bed man :) |
| 20:44.11 | [TK]D-Fender | " Then in extensions I have [redphone] with 200,1,Dial(sip/?????)" <-- we have no idea if your devices are pointed to this dialplan context BTW |
| 20:44.20 | ed207 | Thanks for helping me find my own stupidity :) |
| 20:44.25 | ed207 | Peace out guys. Thanks |
| 20:48.44 | hackish | I think they are... in sip.conf context=redphone |
| 20:50.24 | [TK]D-Fender | hackish, show > think |
| 20:50.31 | kresp0 | Hi all, |
| 20:50.31 | kresp0 | I already configured sip.conf with language=es, but I need to Set(CHANNEL(language)=es) because asterisk seems ignore that language=es option on sip.conf :/. What could be wrong? |
| 20:50.40 | [TK]D-Fender | hackish, Same goes for configs as for CLI trace of the call |
| 20:50.51 | [TK]D-Fender | kresp0, Show us |
| 20:51.07 | hackish | I've been trying to figure out what I need to do to get the trace to show up in CLI. |
| 20:51.23 | [TK]D-Fender | hackish, "sip set debug on" , "core set verbose 10" |
| 20:51.23 | edong23 | hackish: is it registered? |
| 20:51.28 | edong23 | or, are they registered? |
| 20:51.43 | edong23 | sip show peers |
| 20:51.47 | hackish | sip show peers lists them |
| 20:52.09 | [TK]D-Fender | hackish, "sip set debug on" , "core set verbose 10" <-- |
| 20:52.10 | edong23 | then you "should" be able to send calls to them, as [TK]D-Fender said, lets see some consolage |
| 20:52.16 | kresp0 | [TK]D-Fender: show what? |
| 20:52.34 | [TK]D-Fender | kresp0, How about the calls where you set this and it does noting so we can see? |
| 20:53.01 | [TK]D-Fender | kresp0, And the configs |
| 20:53.07 | hackish | curiously it all of a sudden started working. |
| 20:53.18 | edong23 | hackish: that means it will work forever! |
| 20:53.19 | [TK]D-Fender | \o/ |
| 20:53.19 | edong23 | lol |
| 20:53.27 | [TK]D-Fender | Or until it stops! |
| 20:53.28 | hackish | there isn't normally a delay when you issue a dialplan reload is there? |
| 20:53.35 | edong23 | no... |
| 20:53.44 | edong23 | hm... |
| 20:54.09 | edong23 | unless asterisk is running on a pentium pro |
| 20:54.37 | hackish | no it's a decent machine... q8400 I think. |
| 20:55.00 | hackish | I probably do need to go through the module list because astrisk is taking up 500mb of ram. |
| 20:56.17 | kresp0 | [TK]D-Fender: If I do Set(CHANNEL(language)=es), it works. But I want it to work without having to do that on every call. |
| 20:56.42 | *** join/#asterisk vlad_starkov (~vlad_star@178.177.197.139) |
| 20:56.52 | edong23 | man... i dont even know what channel language is |
| 20:56.53 | edong23 | ... |
| 20:56.54 | [TK]D-Fender | kresp0, Show us the setup is a NON working state |
| 20:58.58 | *** join/#asterisk garymc (~chatzilla@host86-174-143-125.range86-174.btcentralplus.com) |
| 21:03.04 | kresp0 | sip.conf -> http://pastebin.com/raw.php?i=5WQCTtLV |
| 21:03.04 | kresp0 | extensions.conf -> http://pastebin.com/raw.php?i=D5YD7K0J |
| 21:03.04 | kresp0 | errors.conf -> http://pastebin.com/raw.php?i=D2fwetGN |
| 21:03.04 | kresp0 | macros.conf -> http://pastebin.com/raw.php?i=j1gPqNN8 |
| 21:05.05 | [TK]D-Fender | kresp0, Where's the call? |
| 21:05.49 | kresp0 | [TK]D-Fender: you mean the CLI output? |
| 21:06.06 | edong23 | yes, preferably only the call and with some verbosity set |
| 21:06.29 | [TK]D-Fender | kresp0, Yes... the evidence of actual FAILURE |
| 21:08.34 | kresp0 | here: http://pastebin.com/raw.php?i=uL0faqBi |
| 21:08.58 | kresp0 | note that "(language 'en')" when it should be "(language 'es') |
| 21:09.15 | [TK]D-Fender | kresp0, and I see nothign showing it should be anything else... |
| 21:09.56 | [TK]D-Fender | kresp0, Nothing you've shown tells me it should be 'es' |
| 21:10.06 | kresp0 | [TK]D-Fender, on sip.conf I've language=es |
| 21:10.12 | [TK]D-Fender | kresp0, so? |
| 21:10.37 | kresp0 | I think that this language=es sets the sip channel language variable |
| 21:10.45 | [TK]D-Fender | kresp0, What SIP channel? |
| 21:10.53 | [TK]D-Fender | kresp0, that is a DAHDI CHANNEL <------------- |
| 21:10.59 | kresp0 | :S |
| 21:11.05 | [TK]D-Fender | kresp0, You call is from DAHDI. NOT SIP |
| 21:11.14 | [TK]D-Fender | kresp0, Your SIP config means nothing |
| 21:11.31 | [TK]D-Fender | kresp0, You did not set the language in chan_dahdi.conf properly |
| 21:11.36 | kresp0 | right |
| 21:11.46 | kresp0 | thank you [TK]D-Fender |
| 21:12.10 | kresp0 | but I've language=es on chan_dahdi.conf! |
| 21:12.43 | [TK]D-Fender | kresp0, Ask me how myuch I trust what I don't see.... |
| 21:12.52 | kresp0 | chan_dahdi -> http://pastebin.com/raw.php?i=1i1EZabR |
| 21:13.00 | kresp0 | :) |
| 21:13.09 | [TK]D-Fender | kresp0, that config is broken |
| 21:13.20 | edong23 | whare at the spans? |
| 21:13.32 | [TK]D-Fender | kresp0, NOTHING after that #include gets used at all ever |
| 21:13.46 | [TK]D-Fender | edong23, In the #include |
| 21:14.04 | kresp0 | dahdi_channels -> http://pastebin.com/raw.php?i=6VW4hQ75 |
| 21:14.16 | [TK]D-Fender | kresp0, parameters must be set BEFORE the "channel =>" lines you want them to apply to. |
| 21:14.21 | kresp0 | i mean dahdi-channels.conf |
| 21:14.46 | [TK]D-Fender | kresp0, go fix it |
| 21:15.55 | kresp0 | ok, I'm on it |
| 21:16.18 | edong23 | [TK]D-Fender: yeah, thats what i meant, there are not spans in what he posted |
| 21:16.32 | edong23 | err... |
| 21:16.38 | edong23 | i guess thats not a span anyway, |
| 21:16.44 | edong23 | i remember those wildcards |
| 21:16.57 | kresp0 | \o/ |
| 21:17.02 | [TK]D-Fender | edong23, i see his expectation and don't even care if there was at least a bare channel line in there at all or not. |
| 21:17.07 | kresp0 | thank you again [TK]D-Fender |
| 21:17.11 | kresp0 | now it works! |
| 21:17.13 | [TK]D-Fender | edong23, He expected the setting at the bottom to mean something. |
| 21:17.30 | [TK]D-Fender | kresp0, You're welcome. |
| 21:17.35 | kresp0 | :) |
| 21:18.15 | edong23 | yeah |
| 21:18.48 | [TK]D-Fender | kresp0, there are several other things you should do to fix your dilapln. Like splitting the IVR menu option you are Goto()ing to into another context and INCLUDING it instead of manual goto's. Also not splitting up "s" in there like you are. That is a VERY bad practice and can lead to breakage |
| 21:23.14 | *** join/#asterisk phiscribe (~david@98.157.5.116) |
| 21:23.57 | *** join/#asterisk vlad_starkov (~vlad_star@178.177.197.139) |
| 21:25.25 | *** join/#asterisk areski (~areski@80.174.255.57.dyn.user.ono.com) |
| 21:25.29 | *** join/#asterisk UnixDev (~UnixDev@unaffiliated/unixdev) |
| 21:25.31 | kresp0 | [TK]D-Fender, so, I should remove all these Gotos and replace them with include => to-internal ? |
| 21:25.51 | kresp0 | I mean, should I... |
| 21:26.12 | UnixDev | I can read local_jitter and remote_jitter from a channel… but how can I get the local and remote ip? |
| 21:28.45 | [TK]D-Fender | kresp0, Clearly not as-is... there is more in there than just what you need. Which is why you should split it up. |
| 21:29.10 | [TK]D-Fender | checkout time, BBIAB |
| 21:44.36 | *** part/#asterisk scottyob (~scottyob@27-33-131-215.static.tpgi.com.au) |
| 22:14.44 | hackish | Is there a good way to figure out what modules you need to accomplish some task? I'm trying to get rid of all the stuff I'll never use |
| 22:19.12 | *** join/#asterisk t (tom@freenode/staff/tomaw) |
| 22:23.43 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 22:27.28 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 22:29.36 | pabelanger | ~book |
| 22:29.36 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 22:29.40 | pabelanger | hackish, ^ |
| 22:29.55 | phix | Is that book even up to date? |
| 22:30.06 | phix | It is based on 1.2 isn't it? |
| 22:30.09 | pabelanger | phix, yes |
| 22:30.36 | phix | ok 1.8 |
| 22:30.47 | phix | fairly up to date, I Might grab it too :P |
| 22:31.05 | drmessano | I have a signed copy |
| 22:31.08 | drmessano | :) |
| 22:31.10 | *** join/#asterisk fprior (be0bbc86@gateway/web/freenode/ip.190.11.188.134) |
| 22:31.22 | phix | drmessano: You signed it? |
| 22:31.23 | phix | :) |
| 22:31.26 | drmessano | Nope |
| 22:32.19 | drmessano | Russell was cleaning up before he moved to Charleston, and I got 3 copies off him |
| 22:32.23 | phix | CTCP DCC phix AsteriskTheDefinitiveGuide2rdEd.pdf :) |
| 22:32.48 | phix | Russel hey |
| 22:32.54 | themrrobert_ | 2rd ed? |
| 22:32.59 | drmessano | lol |
| 22:33.06 | phix | typo |
| 22:33.06 | drmessano | 2rd egzachary |
| 22:33.12 | phix | you get the drift :) |
| 22:33.15 | themrrobert_ | :) |
| 22:33.34 | *** part/#asterisk themrrobert_ (cf4322ee@gateway/web/freenode/ip.207.67.34.238) |
| 22:33.42 | *** join/#asterisk crazed1 (cf4322ee@gateway/web/freenode/ip.207.67.34.238) |
| 22:33.43 | phix | s/2/3/ :) |
| 22:33.57 | pabelanger | 4th |
| 22:33.58 | *** part/#asterisk crazed1 (cf4322ee@gateway/web/freenode/ip.207.67.34.238) |
| 22:34.15 | drmessano | Is the latest based on 10 or 11? |
| 22:34.23 | pabelanger | 12 |
| 22:34.29 | phix | there is a 4th? |
| 22:34.33 | pabelanger | yes |
| 22:34.37 | drmessano | huh |
| 22:34.46 | drmessano | Interesting |
| 22:34.51 | pabelanger | ~book |
| 22:34.52 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 22:34.56 | pabelanger | read the url |
| 22:36.19 | *** join/#asterisk leedm777 (~leedm777@nat/digium/x-cdupulgwhosluvbp) |
| 22:36.51 | fprior | Hi all, can I share with you a trouble with DTMF detection ? |
| 22:37.58 | drmessano | http://shop.oreilly.com/product/0636920025894.do <-- 11, actually |
| 22:39.04 | edong23 | fprior: share |
| 22:39.10 | *** join/#asterisk crazed1 (~robertmos@unaffiliated/themrrobert) |
| 22:42.27 | fprior | Here is: http://pastebin.com/RPk5vjCs |
| 22:42.54 | hackish | the list of modules in the book is only somewhat complete. I managed to figure out which module I was missing but didn't really find a good way to determine which ones were not needed. |
| 22:43.41 | phix | so back to my question I asked last night, why should I care about 11 if 8 is doing everything I need? Is 8 getting dropped any time soon? |
| 22:43.59 | fprior | rtp log show dtmf arriving but I've no dialplan detection |
| 22:44.07 | mjordan | phix: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
| 22:44.18 | mjordan | phix: https://wiki.asterisk.org/wiki/display/AST/New+in+10 |
| 22:44.24 | phix | mjordan: Cheers |
| 22:44.25 | mjordan | phix: https://wiki.asterisk.org/wiki/display/AST/New+in+11 |
| 22:45.58 | igcewieling | and UPGRADE*.txt files in the asterisk source code |
| 22:46.22 | phix | hmmm, 1.6 is stilled used in Debian Squeeze |
| 22:46.49 | *** join/#asterisk Minotaur01 (~minotaur0@S01060018e7f9c7df.hm.shawcable.net) |
| 22:46.59 | drmessano | The 11 link needs a "Brief" section |
| 22:47.28 | phix | Luckily 1.8 is in backports, however I am using the one from the digium repo |
| 22:47.52 | mjordan | drmessano: a toc? |
| 22:48.02 | phix | Or should I use the one from Backports now it is in there? |
| 22:48.07 | phix | What you say mjordan ? |
| 22:48.09 | drmessano | Check the 10 link.. There is a "Brief" paragraph |
| 22:48.17 | drmessano | mjordan ^ |
| 22:48.22 | mjordan | drmessano: ah. Hm. |
| 22:48.26 | mjordan | I can write that up. |
| 22:48.28 | drmessano | Sorry, In Brief |
| 22:48.52 | drmessano | Thats would be the answer to "Whats so great about 11" |
| 22:48.59 | drmessano | Just as the "In Brief" is for 10 |
| 22:49.18 | phix | support for skype :/ is that really a good thing though? |
| 22:49.36 | drmessano | Thats no longer valid :) |
| 22:49.52 | fprior | edong23, any idea ? |
| 22:49.56 | phix | (just reading the what's new in 10) |
| 22:49.59 | drmessano | Oh |
| 22:50.07 | drmessano | No, thats Skypes SILK codec |
| 22:50.11 | drmessano | Not "Skype" |
| 22:50.43 | drmessano | SILK has been implemented by quite a few clients |
| 22:51.01 | phix | ah |
| 22:52.06 | phix | "Silk is a natural protein fibre, some forms of which can be woven into textiles" hmmm I don't think that is what I am after |
| 22:52.06 | igcewieling | I'm scared to upgrade to Asterisk 11. |
| 22:52.16 | drmessano | We already tried Skype integration. That didn't last long |
| 22:52.28 | drmessano | They turned out to be dicks |
| 22:52.31 | phix | drmessano: I would imagine Skype sent out a few lawsuits? |
| 22:52.36 | drmessano | No |
| 22:52.40 | drmessano | It was a partnership |
| 22:52.47 | phix | oh |
| 22:53.13 | navaismo | why igcewieling ? |
| 22:53.15 | phix | Partnership with a competitor |
| 22:53.23 | drmessano | Competitor? |
| 22:53.26 | drmessano | Not hardly.. |
| 22:53.36 | phix | They basically offer the same thing as asterisk, just only as a service |
| 22:54.11 | igcewieling | there were multiple ways to get Skype, some windows/skype hack and the skype/diigium licensed solution, I'm sure there was at least one other attempt. Eventually you must realize they (Sykpe) does not want you and move on to someone who does. |
| 22:54.25 | phix | IT Admins like SoaS these days as it takes the blame off them when something goes wrong, Managers like that too |
| 22:54.28 | phix | <PROTECTED> |
| 22:54.57 | drmessano | Skype doesn't offer a PBX replacment.. thats false |
| 22:55.05 | artyx | free unprotected sip server at 66.27.48.254:5060 |
| 22:55.09 | artyx | hyuks |
| 22:55.32 | igcewieling | navaismo: Not sure. Maybe because has, historically, only been mature enough for our use about half way through the official lifecycle? |
| 22:56.08 | phix | igcewieling: Yeah, frankly I stay away from Skype, but alot of clients want to contact me via it :\ I guess I should just sell them an ansterisk system and setup a trunk :P |
| 22:56.21 | igcewieling | though with the weird issues we've been having maybe I should give it a try |
| 22:56.27 | phix | artyx: :/ |
| 22:56.28 | mjordan | drmessano: added |
| 22:56.52 | artyx | phix: Why the long face? |
| 22:57.07 | drmessano | mjordan: Sweet |
| 22:57.08 | igcewieling | artyx: you don't have to announce it, leave it open for a day and you'll get traffic. 8-| |
| 22:57.17 | phix | Pasting unprotected sip servers hey |
| 22:57.20 | artyx | Lol. i thought i'd speed it up... Trying to test f2b |
| 22:58.00 | phix | ah |
| 22:58.01 | phix | hehe\ |
| 22:58.23 | drmessano | Unprotected SIP server at 192.168.3.10 .. Go get em, guys |
| 22:58.24 | igcewieling | one of our customers was hit by international fraud this weekend. Their PBX (not asterisk, not managed by us) was hacked. |
| 22:58.29 | artyx | laugh |
| 22:58.41 | artyx | Yeh, thats no joke |
| 22:58.51 | artyx | we have had somethign like 10,000 attempts this month (feb) |
| 22:58.59 | igcewieling | they are all POTS |
| 22:59.07 | igcewieling | this customer at least. |
| 22:59.09 | artyx | we have 4 pots lines we do all our dialing on |
| 22:59.22 | artyx | all the did for users is on sip, then the main line is the same 4 pots |
| 22:59.58 | igcewieling | more than 99% of our users are on-net, so we block all off-net access and put in a few exceptions for the ones which are off-net. |
| 23:01.31 | *** join/#asterisk [TK]D-Fender (~chatzilla@bas1-montreal08-1279584386.dsl.bell.ca) |
| 23:04.16 | hackish | So I pick up my extension and dial the other extension... 6 or so seconds of silence and it starts to ring. The other extension starts ringing as soon as that happens and I also see the logs starting to roll. Does this sound like the ATA adapter is twiddling its thumbs before making a connection? not really sure how to diagnose this problem. |
| 23:05.11 | navaismo | sip debug |
| 23:05.25 | drmessano | What kind of ATA? |
| 23:05.32 | hackish | cisco spa122 |
| 23:05.40 | hackish | that's with sip debug rolling. |
| 23:05.51 | hackish | this is why I suspect it's network related somehow. |
| 23:05.57 | drmessano | Are you using a dialplan that doesnt have a pattern match for the number youre dialing? |
| 23:06.11 | drmessano | If so, its going to wait out the interdigit timeout |
| 23:06.24 | hackish | they are both on the same subnet, both have dns and reverse dns etc. |
| 23:06.25 | drmessano | Dialplan on the ATA, that is |
| 23:06.31 | eirirs | I have . as pattern match in the outbound route hehe |
| 23:06.35 | hackish | hmmm. that's a good point. |
| 23:06.46 | drmessano | Linksys ATA's are set for 5 seconds |
| 23:06.46 | *** join/#asterisk dpilon (~dpilon@c-50-138-178-238.hsd1.ct.comcast.net) |
| 23:07.00 | drmessano | You CAN lower it, but setting the dialplan is better |
| 23:07.06 | *** join/#asterisk robertmo1s (~robertmos@unaffiliated/themrrobert) |
| 23:07.21 | hackish | I suppose I really don't want the ATA to act like it's got a brain at all. |
| 23:07.40 | drmessano | You have to... |
| 23:07.42 | hackish | the asterisk should probably do all the thinking no? |
| 23:08.11 | drmessano | If you don't tell the ATA what patterns to match, its not going to think.. and you're waiting out timers |
| 23:08.19 | eirirs | where do u set it? |
| 23:08.20 | [TK]D-Fender | hackish: There is no play-by-play. The ATA sends the number complete once it thinks it's done |
| 23:08.29 | eirirs | on the phone or on the asterisk? |
| 23:08.39 | igcewieling | hackish: That is how I think it should be. This is not how it works with SIP. This is how it worked with MGCP, but that protocol lost out to SIP. 8-( |
| 23:08.49 | hackish | ah, OK I thought the ATA sent numbers as it got them. |
| 23:08.54 | drmessano | Nope |
| 23:09.14 | igcewieling | hackish: the dialplan on the ATA is what determines when numbers are sent |
| 23:09.25 | hackish | (*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.) |
| 23:09.31 | drmessano | Thats the problem.. The ATA buffers.. waiting for a match.. when it doesnt get one, and no digits have been entered for 5 seconds, it sends what you entered |
| 23:09.33 | hackish | that's probably entirely default |
| 23:09.46 | [TK]D-Fender | hackish: Looks lik |
| 23:09.48 | [TK]D-Fender | e |
| 23:10.07 | hackish | I setup 200 as a local extension to the other phone. |
| 23:10.12 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 23:10.22 | drmessano | So you need xxx in there |
| 23:10.26 | drmessano | Between 2 pipes |
| 23:10.39 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 23:10.39 | hackish | The thing I really didn't understand is how you'd differentiate an external area code if there is a 200-xxx-xxxx |
| 23:10.42 | drmessano | (xxx|*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.) |
| 23:10.56 | drmessano | Thats what the timeout is for |
| 23:11.09 | drmessano | There is a long timeout and a short timeout |
| 23:11.35 | drmessano | If you hit a match, and you dont continue entering digits past the SHORT timeout time, it will send |
| 23:11.38 | hackish | so I want to call joe blow in that area code I dial 200-123-1234... but the first 200 will match the internal extension and start rolling it. |
| 23:11.46 | drmessano | So NO match is like 5 seconds, the SHORT timeout is like 2 |
| 23:11.52 | drmessano | ^^^ |
| 23:11.52 | igcewieling | To avoid a timeout you can require all external calls be dialed with a leading 1 or 011 and not use extensions starting with a 1 or 0. You could also use 9 or other digit for an outside line. |
| 23:11.54 | drmessano | Im telling you |
| 23:12.29 | drmessano | If you have XXX in there as I showed you, and dial 200 -- Waits 2 seconds for more digits, then sends |
| 23:12.42 | hackish | gotcha drmessano, just crossed your answer with my second part of the question. |
| 23:12.44 | drmessano | If you DONT have XXX in there as I showed you, and dial 200 -- Waits 5 seconds for more digits, then sends |
| 23:13.45 | igcewieling | something like *xx|911|1[2-9]xx|011xxxxxx.|1[2-9]xx[2-9]xxxxxx|[5-6]xx (where 5xx and 6xx are your extensions |
| 23:13.47 | hackish | in practise is there any downside to using a 2 digit or a 1 digit extension? |
| 23:13.48 | drmessano | I think those are the long interdigit timeout and short interdigit timeout in the config. Tweaking them is useless because then you'll be rushed or misdial during a pause. Match your patterns and it will all work |
| 23:14.26 | hackish | maybe something like #1 work work? |
| 23:14.29 | igcewieling | never use one digit extensions, try very hard to use 3 digit or more extensions |
| 23:14.55 | hackish | There will only be 2 extensions here anyway |
| 23:14.57 | drmessano | I agree with igcewieling.. If all your extensions are 2XX, you could put 2xx in there and that will give you a litttle more of a catch from accidentally passing a call before you've dialed completely, and still match your extensions |
| 23:15.11 | drmessano | So if you dial 803, pause, it wont send 803 |
| 23:15.24 | hackish | yes, I like that idea. |
| 23:15.38 | igcewieling | *nod* for example there is no area code 200 |
| 23:16.01 | hackish | is there a good reason why they don't use # combinations? |
| 23:16.18 | igcewieling | you mean ending the number with a # when dialing? |
| 23:16.31 | hackish | nope, start the number with # or * |
| 23:16.45 | drmessano | * and # are often associated with feature codes |
| 23:16.46 | hackish | no real phone number should ever have a # or * in it right? |
| 23:17.09 | igcewieling | # is generally considered an end of number digit to shortcircuit the timeout on many systems |
| 23:17.21 | drmessano | If you're using a system like FreePBX that is loaded with, AND/OR if your ATA/Phone is loaded with feature codes, you wont start changing things |
| 23:17.40 | hackish | no freepbx here. |
| 23:17.51 | drmessano | Your Cisco ATA is loaded with them |
| 23:17.54 | drmessano | hence the AND/OR |
| 23:18.13 | hackish | ah.. OK. |
| 23:18.37 | drmessano | Removing them is a good idea.. so is avoiding them |
| 23:18.56 | hackish | as you type that I'm looking at how to disable some of the stuff I'll never use. |
| 23:19.13 | drmessano | Tab backspace, tab backspace, tab, backspace |
| 23:19.29 | hackish | fax huh. Local musieum has a fax machine. |
| 23:19.59 | phix | drmessano: ^T ^H? |
| 23:21.32 | hackish | g711u is a decent default codec for a decent pipe? 2 lines with 2mbps up, 20 down. |
| 23:22.07 | hackish | poor quality sound drives me nuts. I hate my cellphone just for that. |
| 23:22.10 | carrar | Oh I don't know, g.711u uses about 88k per call |
| 23:22.20 | carrar | might break that 2meg link :) |
| 23:22.31 | *** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/x-isgdjnnoighiatbj) |
| 23:23.19 | hackish | it gives me g711u, g711a, g726-32 and g729 |
| 23:25.04 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 23:31.16 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 23:31.19 | hackish | This phone can only call the other phone... (200|*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.) Delay is much improved but still a bit long. |
| 23:31.40 | hackish | since it should understand the 200 would you not expect it to make the call immediately? |
| 23:32.27 | [TK]D-Fender | [2-9]xxxxxx <------ |
| 23:32.36 | [TK]D-Fender | it could be 200XXXX |
| 23:32.52 | [TK]D-Fender | it is not UNIQUE until you FAIL to match other patterns |
| 23:33.13 | hackish | ah ok |
| 23:35.27 | hackish | these things look a bit like regex. |
| 23:36.03 | igcewieling | you still have [2-9]xxxxxx which will partially match 200 |
| 23:36.57 | hackish | I was thinking [2-9][1-9][1-9]xxxxx |
| 23:37.08 | hackish | make 200 the extension that just means the other phone. |
| 23:37.42 | hackish | but that doesn't work for 204-123-1234 |
| 23:43.02 | hackish | I assume this timeout is called T2? |
| 23:44.30 | *** join/#asterisk jsjc (~Adium@163.Red-83-38-209.dynamicIP.rima-tde.net) |
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| 23:52.09 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-yaqtbvbznikmaojx) |
| 23:54.32 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
