| 00:01.12 | *** join/#asterisk retentiveboy (n=pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
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| 00:06.40 | pscottdv | I have a new asterisk system set up with the ATRPMS for Fedora 11. I am getting the following error in my logs: chan_dahdi.c: Unable to enable echo cancellation on channel 2 (No such device) and am having terrible echo. This problem occurs with any call between my linksys SPA921's and an outside line. Outside lines are connected via four POTS lines connected to a TDM400p with four FXO cards. I have enabled echocancellation and echo |
| 00:07.01 | pscottdv | echocancel and echotraining in chan_dahdi.conf |
| 00:07.19 | pscottdv | Here is my chan_dahdi.conf file: |
| 00:07.47 | pscottdv | [trunkgroups] |
| 00:07.47 | pscottdv | ; used for NFAS and GR-303 connections |
| 00:07.47 | pscottdv | [channels] |
| 00:07.47 | pscottdv | ; hardware channels |
| 00:07.47 | pscottdv | ; default for channels inherited to all channels |
| 00:07.48 | pscottdv | ;usecallerid=yes |
| 00:07.50 | pscottdv | hidecallerid=no |
| 00:07.52 | pscottdv | callwaiting=no |
| 00:07.54 | pscottdv | threewaycalling=yes |
| 00:07.56 | pscottdv | transfer=yes |
| 00:07.58 | pscottdv | echocancel=yes |
| 00:08.00 | pscottdv | echotraining=yes |
| 00:08.02 | pscottdv | echocancelwhenbridged=yes |
| 00:08.04 | pscottdv | rxgain=0.0 |
| 00:08.06 | pscottdv | txgain=0.0 |
| 00:08.08 | pscottdv | signaling=fxs_ks ; FXO channels use FXS signalling |
| 00:08.10 | pscottdv | ;define channels |
| 00:08.12 | pscottdv | context=incoming_ofc ; standard office line (inherited to lines 1, 2 & 3) |
| 00:08.14 | pscottdv | channel => 1 |
| 00:08.16 | pscottdv | channel => 2 |
| 00:08.18 | pscottdv | channel => 3 |
| 00:08.20 | pscottdv | context=incoming_mkt ; marketing line |
| 00:08.22 | pscottdv | channel => 4 |
| 00:08.24 | pscottdv | What am I missing? |
| 00:08.35 | *** join/#asterisk manxpower (n=EWieling@24.42.221.26) |
| 00:08.41 | manxpower | ~answers |
| 00:08.42 | infobot | methinks answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 00:08.46 | thx2000 | pastebin? |
| 00:09.01 | retentiveboy | -pb |
| 00:10.21 | pscottdv | The echo is heard on the SPA921, BTW, not on the other end. |
| 00:10.36 | manxpower | pscottdv: you never hear echo on the analog side |
| 00:11.04 | pscottdv | Never. Only on my SPA921. |
| 00:11.30 | pscottdv | But never between them--only when the other side is via POTS lines. |
| 00:12.11 | manxpower | correct |
| 00:13.14 | manxpower | all calls with a far end anlog look have echo. But in a non-VoIP world, the echo happens SO fast you can't hear it. Like your voice bouncing off the walls of a small room (echo comes back so fast you can't hear it) .vs. your voice bouncing off the walls of a very very large room (you hear the echo because it happens slow enough) |
| 00:13.49 | pscottdv | Sure, I read about that. But why the error message? Why no echo cancellation? |
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| 00:14.51 | manxpower | pscottdv: I've not seen your erro message |
| 00:15.10 | pscottdv | chan_dahdi.c: Unable to enable echo cancellation on channel 2 (No such device) |
| 00:15.19 | manxpower | that means you do not have a channel 2 |
| 00:15.46 | manxpower | What you should do is call the maker of the card and give them that errpr |
| 00:15.52 | manxpower | I assume your call is actually going out on channel 2? |
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| 00:16.08 | pscottdv | yes, call actually goes out on channel 2 |
| 00:16.30 | pscottdv | Card is digium TDM400P |
| 00:16.31 | manxpower | sounds to me like a hardware problem |
| 00:16.44 | pscottdv | With four FXO cards |
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| 00:17.54 | pscottdv | It feels like a software problem to me. The calls connect, what could the card do to prevent asterisk from enabling echo cancellation? |
| 00:18.37 | manxpower | Do you know what "no such device" means? |
| 00:18.43 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
| 00:18.47 | manxpower | It means "You don't have a channel 2". To me that is a hardware issue. |
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| 00:19.22 | pscottdv | I can call in or out on every channel. |
| 00:19.30 | manxpower | exactly. |
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| 00:20.15 | pscottdv | If there really is "no such device", then how are the calls getting through? |
| 00:20.52 | manxpower | It doesn't make sense. I know that. |
| 00:21.06 | manxpower | In any case, *I* made my suggestion. |
| 00:21.16 | pscottdv | Thank you. |
| 00:22.02 | pscottdv | I have posted the problem on digium's forums. |
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| 00:54.58 | retentiveboy | Is there a dialplan variable to get the IP address of the local machine? |
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| 01:35.38 | ZX81 | NoOp(127.0.0.1) :D |
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| 02:28.01 | HAhmadi | Hi |
| 02:28.09 | HAhmadi | I have a SIP client |
| 02:28.46 | HAhmadi | And an AC(Access Server) |
| 02:29.02 | HAhmadi | My client can not register for incomming calls |
| 02:29.25 | HAhmadi | I want use Asterisk, But im not sure it can do that for me or not? |
| 02:29.30 | HAhmadi | Can anyone help me? |
| 02:30.49 | HAhmadi | My AC is cisco 5350, And don't support registeration in its hardware, i must use a SIP Server with it, But im not success to found any free SIP Server software that support registaration. |
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| 02:36.04 | m477au | anyone have experience with B410P and asterisknow? |
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| 02:37.03 | HAhmadi | no idea? |
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| 02:43.10 | manxpower | HAhmadi: Your SIP clients can register to Asterisk |
| 02:43.46 | HAhmadi | Can Asterisk wirk with my AC? |
| 02:43.53 | HAhmadi | wirk->work |
| 02:43.57 | HAhmadi | manxpower |
| 02:46.36 | *** join/#asterisk Schmee (n=zaphod@ppp100-124.static.internode.on.net) |
| 02:48.08 | HAhmadi | manxpower, I mean that Is Asterisk support incomming call from direct gateway and distribute them on my SIP Clients? |
| 02:49.12 | manxpower | HAhmadi: yes |
| 02:49.26 | manxpower | HAhmadi: Read The Asterisk Book |
| 02:49.28 | manxpower | ~book |
| 02:49.28 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 02:49.48 | Schmee | hi all. I'm in need of some advice and my google-fu is failing me today. I have an asterisk system (PIAF) at one location with a single handset (cisco 7940) and 3 remote locations each with a cisco 7940 which I'd like to treat as local handsets. So far this is working via SIP, however I was thinking about firmware updates, and so forth and I'm loath to enable tftp on an internet facing system. Can anyone point me in the right dir |
| 02:49.48 | Schmee | ection for a reasonable way to update these handsets without needing a) a net facinf TFTP server or b) a local tftp server at each location |
| 02:50.00 | HAhmadi | ~download |
| 02:50.01 | infobot | You can Download the Asterisk PBX from: http://www.asterisk.org/index.php?menu=download or from ftp://ftp.asterisk.org/pub/asterisk - you can also do ~voip-info for information on the wiki about prebuilt packages of Asterisk |
| 02:52.22 | chuckf | Schmee: vpn? |
| 02:53.31 | Schmee | chuckf: not a viable option as the remote locations are residences, no server, just a DSL router |
| 02:55.05 | HAhmadi | manxpower, Thank you for your help, Im installing it |
| 02:55.07 | Schmee | good suggestion though |
| 02:56.46 | chuckf | turn on the tftp server when you need to update, turn it off when you're done |
| 02:57.43 | Schmee | chuckf: I'm beginning to think that's my best option. I'd love to be able to restrict to set IPs via firewall, but the remote extensions are on dynamic IPs |
| 02:59.07 | chuckf | can you trust something like dyndns? |
| 02:59.38 | chuckf | also with the on/off tftp it is not like you'll be updating the remote phones on a regular basis |
| 02:59.50 | Schmee | probably, but firewall rules tend to be IP based rather than name based, also I've had issues with dynamic names and resolving/caching |
| 03:00.48 | Schmee | very true, it's only when there's a reason for the updates. I see one issue though and that's the TFTP settings in the phones themselves, I expect timeouts, etc when the phones are reset for whatever reason |
| 03:01.09 | chuckf | Schmee: well I was thinking of using dyndns, pinging the host every hour or so then modifying the rules if the ip address changes with a script |
| 03:01.19 | Schmee | also the SIP<mac address>.cnf files are retrived fro mthe tftp server as well for small config changes |
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| 03:11.00 | p3nguin | schmee: Why can't you configure a tftpd at each location? |
| 03:11.07 | BeeBuu | is there any command can let me know which agent is free now? |
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| 03:11.13 | thehar | show agents |
| 03:11.22 | m477au | Schmee: can you just remote control their pcs? |
| 03:11.25 | thehar | or somethin like that |
| 03:11.25 | p3nguin | schmee: I'm sure you have at least one computer at each place. |
| 03:11.27 | m477au | use something like logmein ? |
| 03:11.38 | Schmee | p3nguin: becasue there's no server at each location, and it's not garuanteed that there will even be a PC at each location, only the phone |
| 03:11.41 | BeeBuu | thehar: thanks |
| 03:12.06 | thehar | np |
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| 03:31.36 | TJNII | Schmee: What would be really slick would be to write a script that gets the phones' IPs from * and writes firewall rules based off of that. You could do that with a cron job, and if the script is smart and the target IPs don't change often your firewall rules would be pretty constant. |
| 03:32.51 | [TK]D-Fender | tnjand what would you be opening & closing with this? |
| 03:32.58 | Schmee | TJNII: not a bad option. I am mainly worried about the possibility of someone being able to retrieve data from the TFTP server (extension login details for example) |
| 03:33.16 | TJNII | Though, personally, I do the opposite. My provisioning files are out there on a http server and I restrict access to *. I do this to keep out the SIP hackers. |
| 03:33.33 | TJNII | Schmee: True. |
| 03:33.46 | Schmee | TJNII: the cisco 7940 doesn't config via http or ftp, if they did then my question wouldn't be relevant |
| 03:34.03 | Schmee | at least, to my knowledge they don't config via anything but tftp |
| 03:34.12 | TJNII | I don't consider it to be that big a risk as the phones I have in the wild generally can't make outside calls, but I see your point. |
| 03:34.26 | TJNII | Yea, I wouldn't want tftp pointed at the internet, either. |
| 03:34.56 | TJNII | I don't know anything about Cicso, I'm a low budget user so I have Grandstreams. |
| 03:35.07 | TJNII | puts on asbestos suit for the ensuing flames |
| 03:35.23 | Schmee | I picked up the ciscos cheap, so I have no budget for any extra hardware (vpn routers, etc) |
| 03:36.01 | TJNII | You can lock the tftp port down to only the IPs the phones are on, that should solve 95% of the issue. |
| 03:36.13 | TJNII | How to do it s the challenge. :) |
| 03:36.24 | [TK]D-Fender | fills the INSIDE of TJNII's suit was gasoline... then throws in a lit match |
| 03:36.31 | drmessano | tftp now allows directory listing? |
| 03:36.41 | Schmee | personally I hear good things about aastra 57i, but no reseller in Australia |
| 03:36.48 | drmessano | or is guessing MAC addresses of phones some new sploit? |
| 03:37.36 | Schmee | drmessano: considering that ciso phones are probably the most common ones, I'd be suspecting that they have a dedicated range of MAC addresses which limits the possibilities |
| 03:37.46 | drmessano | lol |
| 03:38.03 | drmessano | Do the math |
| 03:38.19 | drmessano | It's pretty insane |
| 03:39.03 | florz | hu? how are 24 million packets insane? |
| 03:39.19 | Schmee | maybe I'm being paranoid, but I still would prefer a slightly dafer option |
| 03:39.25 | Schmee | err, safer option |
| 03:41.00 | drmessano | If I let someone run a brute force attack of that magnitude on my system unnoticed, I should be shot |
| 03:41.00 | florz | erm, sorry, 16 million, of course, 24 bits ... ;-) |
| 03:41.10 | florz | or rather 8 million average for finding a single phone |
| 03:41.19 | Schmee | agreed, but that doesn't preclude the said attacker from being lucky and grabbing one very early on |
| 03:41.30 | box2 | prepares a double barrel of buckshot fun |
| 03:41.56 | drmessano | Schmee: That doesnt prevent anyone from guessing the SIP credentials of a remote endpoint on the first try either |
| 03:42.04 | drmessano | Schmee: or winning the lottery twice |
| 03:42.10 | Schmee | true |
| 03:42.10 | drmessano | Schmee: or being hit by a meteor |
| 03:42.30 | Schmee | I'd just rather not give them the opportunity to try |
| 03:42.42 | drmessano | Turn off all outside access then |
| 03:42.45 | [TK]D-Fender | or being struck twice by lightning, indoors weating a wetsuit on a clear day. |
| 03:42.50 | florz | drmessano: erm ... why? there isn't really much you can so against it anyhow ... |
| 03:43.10 | florz | that's a gigabyte or so of traffic |
| 03:43.20 | florz | really not that much these days |
| 03:44.10 | drmessano | florz: Really? I guess you've never seen even basic brute force protection? |
| 03:44.29 | florz | drmessano: erm, no, what would that look like? |
| 03:45.19 | Schmee | florz: blocking fter a certain number of connection attempts is normal |
| 03:45.40 | florz | Schmee: hu? artificial DoS vulnerabilities are "normal"? IC ... |
| 03:46.23 | drmessano | florz: Dude, there's simple shit perl scripts out there that look for failed authentication attemps or multiple connection attempts that are indicative of malicious activity and throttle the connection... Even Windows 2000 starts bitching when you enter an incorrect password the 4th time |
| 03:46.32 | Schmee | only if not set up correctly, I block ssh attempts on servers in conjunction with a whitelist of always allowed IPs and ssh keys (no passwords) |
| 03:46.48 | drmessano | No intelligence needed |
| 03:47.02 | florz | drmessano: well, yeah, that's an artificial DoS vuln (in the given scenario in particular) |
| 03:47.30 | florz | Schmee: erm, now, if you _do_ have a whitelist, why not just use that for filtering exclusively in this scenario? |
| 03:47.37 | drmessano | Basic math.. If they start getting throttled after n attempts, eventually they will move on, unless you're a choice target |
| 03:47.38 | Schmee | sorry, I'm not trying to start an argument or a flame war of any kind. I'm really just looking for an alternative to get configs to my cisco 7940 phones at remote locations without needing dedicated PCs/servers/specialised routers at each termination point |
| 03:47.58 | florz | Schmee: drmessano's argument basically was that you don't need any of that as you can't guess MACs anyhow |
| 03:48.17 | Schmee | florz: the whitelist isn't much good for dynamic IPs, I was using it as an example |
| 03:48.20 | drmessano | florz, I missed your basic trolling |
| 03:49.05 | florz | drmessano: well, yeah, I don't argue that this does not help against guessing MACs - but you get an easy to abuse DoS vulnerability instead, so what's the use? |
| 03:49.41 | drmessano | How is that an easy to abuse DoS vuln? |
| 03:50.11 | florz | drmessano: you simply send a bunch of wrong requests from the ip address of actual telephones? |
| 03:50.28 | drmessano | lol |
| 03:50.44 | drmessano | Do you really THINK that someone is going to go through this much trouble? |
| 03:51.10 | florz | which "much trouble"? and it obviously depends on your goals ... |
| 03:51.17 | Nugget | if the clamp-down is triggered on failed login attempts you can't do that, since your forged-source DoS will not be able to negotiate the connection and attempt the login |
| 03:51.44 | drmessano | We're talking about basic "beware of big dog" sign sort of prevention here.. Versus leaving soemthing wide open.. If you're looking to sploit someone, you go for the easy targets |
| 03:51.55 | florz | also, you should consider that that's the easiest way to get rid of the "brute force protection", as admins will probably then disable it in order to make the telephones working again |
| 03:52.31 | drmessano | If add basic, simple throttling, someone will get bored and move on.. I've seen this sort of thing in logs for years |
| 03:52.33 | florz | Nugget: you are aware that we are talking about tftp? |
| 03:52.49 | Nugget | oh, no, I thought you were discussing ssh |
| 03:53.28 | florz | that would be quite a bit less problematic, yeah |
| 03:54.19 | florz | drmessano: not if they are interested in _you_, obviously ... so, yeah, it depends on the attacker, of course |
| 03:54.23 | drmessano | florz: Rather than your usual gimmick of pointing out the obvious that nothing is foolproof, do you actually have a suggestion? |
| 03:54.39 | drmessano | florz: Otherwise, this is boring |
| 03:54.40 | Schmee | ssh is a lot easier to secure, just don't use password auth and block the IP after a desired number of incorreect attempts |
| 03:55.23 | Schmee | the other posibility is that the cisco 7940 can be configured another way? maybe some kind of hack to the firmware or something |
| 03:55.45 | florz | and no, I don't have a clue of cisco phones |
| 03:56.24 | Schmee | wished he'd been able to get some decent phones for the same price as the ciscos |
| 03:56.26 | drmessano | Schmee: You can do the same with tftp, SIP, <insert here> |
| 03:56.26 | florz | Schmee: the access routers in place don't possibly have some VPN capability? |
| 03:56.53 | drmessano | Schmee: Add basic connection throttling like BFD or something common and simple, and move on |
| 03:57.00 | Schmee | florz: nope, the routers are generic home dsl ones. no VPN, except I think one has vpn passthrough |
| 03:57.12 | drmessano | Schmee: Practical vs Ad nauseum mathematics |
| 03:57.38 | florz | Schmee: well, some of those can be turned into some Open/Free/DD-WRT ... |
| 03:57.53 | florz | Schmee: or do support installation of openvpn, inofficially |
| 04:00.07 | Schmee | nope, these are low end billion routers for the most part, and one linksys |
| 04:01.45 | florz | Schmee: the telephones don't keep passwords that don't get reprovisioned? |
| 04:02.08 | florz | Schmee: then you could just drop passwords from the config after the initial configuration, maybe? |
| 04:02.49 | Schmee | florz I was actually hoping to use the configs to update passwords at regular intervals |
| 04:03.03 | drmessano | and use psychic.pl to detemine when the phones may go offline or get rebooted and need to pull the config? |
| 04:03.43 | Schmee | this would have been so much simpler if I could use http or ftp for this |
| 04:04.00 | drmessano | Too much overthinking |
| 04:05.35 | drmessano | If someone wants to target your system, you're fucked.. Period.. Everything and anything you do is at risk to be exploited.. Block and random attacks with basic brute force throttling that pushes them on to the next random target, and have another cup of imported black tea on me. |
| 04:05.46 | drmessano | any* |
| 04:06.13 | Schmee | I can get around it by installing a PC at each location, have it hooked into the network permanently, running dhcp + tftp just to handle a single IP phone, but that isn't exactly an economical solution |
| 04:07.45 | drmessano | Here is what you do |
| 04:08.46 | drmessano | Get some WRT54GLs, install DD-WRT.. Setup TFTP and DHCP.. Run a shell script that pulls new configs every N minutes over SCP from the main location |
| 04:09.00 | drmessano | $50 per location, done |
| 04:11.24 | drmessano | Matter of fact, these dont need to be your branch office router if you already have something better.. throw them in as an extra wireless AP or just leave everything off but the TFTP |
| 04:12.37 | Schmee | nice in theory. If nothing else comes up as a serious possibility, then I may have to consider it. |
| 04:13.01 | drmessano | I think thats overkill anyway |
| 04:13.19 | Schmee | realistically, I don't want to be installing anythign else at the remote locations, they are residences (not offices) and they tend to have things set up how they like them |
| 04:13.33 | *** join/#asterisk Jenna (n=JJ@unaffiliated/jenna) |
| 04:14.04 | drmessano | As I said, simple brute force detection for the tftp server is all you practically need. If someone really wants you, you're talking about taking measures that are well outside your price range |
| 04:14.15 | Schmee | nods |
| 04:14.28 | Schmee | very true, I was just hoping that there was another way that I overlooked |
| 04:15.26 | Schmee | if it can be handled in software that's a bonus, I have an existing budget of $0 to finish this project up |
| 04:15.54 | *** join/#asterisk felipe_ (n=felipe@my.nada.kth.se) |
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| 04:19.23 | drmessano | Security at the near $0 level is best accomplished by taking simple steps to make yourself a less desirable target than the other guy that does nothing |
| 04:19.28 | manxpower | Many phones support provisioning via some form of SSL. |
| 04:37.15 | Gokee2 | Is anyone else having problems with asterisk.org? |
| 04:41.53 | Jenna | Gokee2, it seems its down |
| 04:42.29 | Jenna | Gokee2, nope its alive. but way too slown |
| 04:42.31 | Jenna | slown* |
| 04:42.33 | Jenna | slow* |
| 04:44.10 | Gokee2 | Jenna, Hmm, I am getting sql bugs while trying to get a account setup |
| 04:44.27 | Gokee2 | O hey it finished at last! |
| 04:44.44 | Gokee2 | Member for 22 min 33 sec! Wow that took a long time |
| 04:48.42 | Jenna | btw anyone has an idea how can I check my voice mailbox. ? |
| 04:48.53 | Jenna | I have setup ekiga as a sip softphone |
| 04:49.04 | p3nguin | call it? |
| 04:49.56 | p3nguin | You'll probably have to create an extension for the voicemail system if there isn't one already made. |
| 04:50.36 | Jenna | I did create one .e.g. my sip # ext is 444 & my mailbox ext is 888 . |
| 04:51.03 | p3nguin | So dial 888 and check your voice mail. |
| 04:51.19 | Jenna | ekiga does display that I have 5 voice mails. but how do I go & listen to them |
| 04:51.57 | p3nguin | I use exten 9000 to take me into VoicemailMain. |
| 04:52.08 | p3nguin | From there, it asks for mailbox and password. |
| 04:52.16 | p3nguin | I enter those, then I can check my voicemail. |
| 04:52.16 | Jenna | http://www.asteriskguru.com/tutorials/voicemailmain.html this tutorial is not very clear |
| 04:52.46 | p3nguin | I would recommend creating an extension like I did for voicemailmain. |
| 04:52.57 | p3nguin | exten => 9000,1,VoicemailMain(@default) |
| 04:53.24 | *** join/#asterisk nighty^ (n=nighty@210.188.173.245) |
| 04:53.29 | Jenna | is this ,VoicemailMain(@default) a special directive or something ? |
| 04:53.46 | p3nguin | It's the voicemail main menu command. |
| 04:54.21 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip98-186-180-143.ks.ks.cox.net) |
| 04:54.32 | Jenna | hmm. okay let me give it a go |
| 04:54.36 | Jenna | thanx |
| 04:56.08 | p3nguin | http://www.voip-info.org/wiki/view/Asterisk+cmd+VoiceMailMain |
| 04:58.21 | p3nguin | There isn't much to it, really. |
| 04:58.51 | *** join/#asterisk maour (n=gnu@unaffiliated/maour) |
| 04:59.01 | p3nguin | You just use it like any other command via "exten =>" in your dialplan. |
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| 05:00.42 | Jenna | hmm.IC |
| 05:01.19 | Jenna | exten => 401,2,VoicemailMain(777@mb_internal) |
| 05:03.10 | p3nguin | I think I would leave out the 777 part of it. If you just specify the context, it will ask for the ext. and passcode. |
| 05:03.57 | p3nguin | Have one single ext. like I have for the main menu rather than specifying one for everyone. |
| 05:04.21 | Jenna | hmm. okay. cuz now its saying loging incorrect blah blah.. |
| 05:05.51 | p3nguin | If you would match the ext. with the voice mail box, you could do something as simple as: exten => _NXX,1,VoicemailMain(${EXTEN}@mb_internal) |
| 05:06.23 | p3nguin | That would mean that you dial 401 and you end up at the voicemail main menu for 401. |
| 05:07.55 | Jenna | p3nguin, I get this in the console http://pastebin.ca/1566433 |
| 05:09.14 | Jenna | here is my extentions.conf http://pastebin.ca/1566437 |
| 05:09.23 | p3nguin | You've entered mailbox 401@default. Is that where you wanted to be? |
| 05:09.57 | p3nguin | Your extensions are not good. |
| 05:11.09 | Jenna | yeah my extentioni s 401 & my mailbox is 777 |
| 05:12.19 | Jenna | what do u suggest ? btw is there a good practices doc which I can rtfm |
| 05:12.32 | p3nguin | http://pastebin.ca/1566441 |
| 05:13.12 | p3nguin | That's how I would make an extension.conf for your system. I would also NOT have different mailbox numbers from the extension numbers. |
| 05:13.44 | Jenna | p3nguin, I few other extentions as well 402, 403. would this new setting conflict with them |
| 05:13.46 | Jenna | ? |
| 05:14.19 | p3nguin | _4XX matches all of your extensions between 400-499. |
| 05:14.48 | p3nguin | and ${EXTEN} matches whatever extension you dialed. |
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| 05:16.25 | Jenna | hmm. so I suppose I should remove other lines as well (...exten => 403,2,VoiceMail(888@mb_internal) cuz this new entry would take automatch their respetive mailboxes as well. right ? |
| 05:18.29 | p3nguin | Is there some reason that you use arbitrary mailbox numbers instead of matching it to the phone's extension? |
| 05:19.57 | Jenna | no. no reason. just checking out the mailbox feature. neva done it before. yeah using the same exten for the mailbox does make sense. but I hesitated so as it wont crash/go ape on me |
| 05:20.37 | p3nguin | There are reasons to not have them the same, but if you don't have one of those reasons already, just match them up. |
| 05:20.50 | p3nguin | It's a lot easier for the dialplan that way. |
| 05:21.53 | p3nguin | That paste I gave you takes care of all of your 4xx extensions and the voicemail for all of them too. No additional lines are needed to be able to call anyone in the 400s and get their voicemail if they are unavailable. |
| 05:23.29 | Jenna | hmm. [mb_internal] |
| 05:23.29 | Jenna | 777 => 401,401,401@localhost |
| 05:23.29 | Jenna | 888 => 403,403,403@localhost |
| 05:23.49 | p3nguin | What a mess. |
| 05:23.50 | Jenna | this is my voicemail.conf entry |
| 05:23.52 | p3nguin | Fix up the voicemail.conf file to match up user's voicemail numbers, and reload. That should be it. |
| 05:23.59 | Jenna | :( |
| 05:24.39 | Jenna | okay let me enter a corresponding entry for the mb of ext. |
| 05:25.18 | p3nguin | Try something more like: 401 => 0000,Jenna,jenna1234@yahoo.com |
| 05:26.00 | Jenna | for the starters I have set it like this 401 => 401,401,401@localhost |
| 05:26.04 | p3nguin | 401 is your corresponding phone extension, 0000 is your passcode for voicemail, Jenna is the name on the voicemail box, the last is your email address. |
| 05:26.22 | Jenna | yeah I gather that |
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| 05:27.27 | p3nguin | And these need to be under a context of [mb_internal] based on the dialplan we already created. |
| 05:28.57 | Jenna | yup they are |
| 05:29.37 | p3nguin | If you're fine with what you have changed, reload the thing and see what happens when you dial 401. |
| 05:32.15 | Jenna | yeah I did that. http://pastebin.ca/1566441 this is the only entry at the bottom of my extentions.conf. I have asked a person to call me on my ext 401. I hope he would have an exten automatically assigned to him |
| 05:32.24 | Jenna | btw what about sip.conf |
| 05:33.10 | p3nguin | You can't get an extension automatically assigned. You must configure peers in sip.conf and then register the phone with the user and secret. |
| 05:33.27 | Jenna | I have an entry in the ¨internal¨ context in the extentions stanza . something like. mailbox=777@mb_internal |
| 05:34.05 | p3nguin | You need much more than that. |
| 05:34.14 | p3nguin | But you left, so I'm essentially talking to myself. |
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| 05:37.37 | makafre | hey guys, the issue must have been discussed before, but what's the deal with requirecalltoken, I set it to no or auto but it wont change anything, zoiper doesnt connect.. |
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| 09:35.32 | *** join/#asterisk GNU\colossus (i=colo@truschnigg.info) |
| 09:35.36 | GNU\colossus | hi all |
| 09:35.39 | *** join/#asterisk Anakin (i=anakin@unaffiliated/anakin) |
| 09:37.01 | GNU\colossus | we're having problems with an asterisk server of ours. incoming speech is hard to understand, while outgoing voice traffic is working just fine apparently. any ideas what could be wrong? |
| 09:37.06 | GNU\colossus | could it be a codec issue? |
| 09:39.11 | *** join/#asterisk garymc (n=garymc@host81-134-0-102.in-addr.btopenworld.com) |
| 09:39.20 | tzafrir_laptop | what codec do you use? |
| 09:39.57 | garymc | Yo... anyone know how i turn my apple Airport into a switch? |
| 09:40.13 | garymc | so the bt router dishes out the ips and not the Apple airport? |
| 09:40.33 | garymc | Trying to sort this port forwarding problems |
| 09:40.51 | GNU\colossus | tzafrir_laptop: if i recall correctly, it's "G711 alaw" |
| 09:41.06 | *** join/#asterisk mort_gib (n=mjensen@215.210.244.195.dsl.static.gibconnect.com) |
| 09:42.38 | *** join/#asterisk mort_gib (n=mjensen@215.210.244.195.dsl.static.gibconnect.com) |
| 09:43.02 | garymc | Nevermind, apple and mac have a channel too |
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| 09:48.57 | *** join/#asterisk tech_adrian (n=alexandr@88.158.159.126) |
| 09:48.58 | GNU\colossus | what do I need to in order to be able to use the speex codec with my asterisk server? is that dependant on the clients involved? |
| 09:53.40 | *** join/#asterisk wathek (n=wathek@41.224.148.68) |
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| 09:59.39 | *** join/#asterisk PanicMan (i=panicman@122.102.33.67) |
| 10:00.06 | PanicMan | hello, I'm new in asterisk, Trying to use portech MV-370 |
| 10:00.16 | PanicMan | getting extension error |
| 10:01.30 | PanicMan | Call from '198' to extension '88017131111111' rejected because extension not found |
| 10:01.35 | PanicMan | any helper :( |
| 10:01.56 | PanicMan | [outgoing] |
| 10:01.56 | PanicMan | exten => _880,1,Dial(SIP/103,60,r) |
| 10:01.56 | PanicMan | exten => _880,2,Hangup() |
| 10:01.56 | PanicMan | this is the config |
| 10:02.02 | PanicMan | hello |
| 10:03.11 | *** join/#asterisk tm1985 (n=tm@082-146-101-077.stat.adsl.xs4all.be) |
| 10:03.19 | PanicMan | any helper ? |
| 10:03.33 | tm1985 | hello need some asterisk help |
| 10:03.52 | tm1985 | can someone help me please? |
| 10:06.32 | *** part/#asterisk ming_zym (n=ming_zym@124.127.101.0) |
| 10:06.56 | *** part/#asterisk BeeBuu (n=beebuu@121.9.232.26) |
| 10:08.07 | *** join/#asterisk AL-Hadi (n=User@unaffiliated/al-hadi) |
| 10:08.15 | tm1985 | Our provider uses *21*telphonenumber# to redirect to for example a mobile phone. I need to know how I activate this in asterisk ? |
| 10:08.45 | tm1985 | we us Asterisk 1.6.1.6 with mISDNv2 and chan_lcr for the moment |
| 10:09.42 | tm1985 | our provider is a ISDN provider and no VOIP provider |
| 10:12.22 | tm1985 | exten => 600,2,Dial(LCR/outgoing/*21*TELNUM#) : this is we got it now? |
| 10:14.04 | dwery | tm1985: you mean "transfer" for "redirect" ? |
| 10:16.31 | *** join/#asterisk Moz (n=me@81.179.238.144) |
| 10:18.19 | tm1985 | When we redirect this call, This call doesn't come on the asterisk, But on the provider which sends it to this number |
| 10:18.43 | dwery | so it's a permanent call deflection? |
| 10:18.52 | dwery | (CFU) |
| 10:19.08 | tm1985 | what is a call deflection? |
| 10:19.22 | dwery | quite probably, the thing you described :D |
| 10:19.38 | tm1985 | then probably yes |
| 10:20.17 | dwery | if LCR is sending the DTMF digits correctly it should be doable using the so called "call files" |
| 10:22.06 | wathek | any one would help me please to test my Asterisk configuration ? address : wathek.homelinux.org username : 102 password:guest |
| 10:22.17 | wathek | would you please call 101 which is me |
| 10:22.51 | wathek | thank you |
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| 10:28.08 | tm1985 | dwery, http://forums.digium.com/viewtopic.php?t=71417, in this thread are my errors with LCR |
| 10:28.20 | tm1985 | It's is the last post |
| 10:29.54 | dwery | tm1985: mmm.. I'd drop LCR and use DAHDI directly |
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| 10:31.57 | tm1985 | Our card doesn't dahdi, it has to go through isdn |
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| 10:32.07 | tm1985 | *support |
| 10:32.09 | dwery | tm1985: which card do you have? |
| 10:32.41 | tm1985 | Cologne Chip Designs GmbH ISDN network controller [HFC-PCI] |
| 10:32.49 | dwery | tm1985: it compatible with dahdi |
| 10:32.52 | dwery | it's* |
| 10:32.59 | dwery | driver zaphfc |
| 10:33.09 | garymc | Hey anyone wanna test an extension for me, i think ive fixed the POrt mapping problem? |
| 10:33.39 | garymc | using zoiper? |
| 10:34.48 | tm1985 | We gone try that thx for the info |
| 10:42.12 | tech_adrian | Hello guys ,Is there any implementation of Sip Method Message in Asterisk 1.6.x ? Maybe phpagi .... ? |
| 10:43.41 | *** join/#asterisk DonAlex (n=DonAlex@80.177.97.15) |
| 10:43.57 | DonAlex | Morning all.. |
| 10:44.06 | tech_adrian | Any ideeas how to implement chat using eyebeam and asterisk ? |
| 10:46.13 | Tim_Toady | tech_adrian asterisk does not support sip messaging |
| 10:47.13 | Tim_Toady | if u really need it u can setup a ser/openser proxy infront of asterisk |
| 10:49.00 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
| 10:50.04 | tech_adrian | Many thanks . I will see about openser . |
| 10:56.43 | DonAlex | *groans* Dunno what I did to break this .. but why am I getting WARNING[381]: chan_sip.c:12099 handle_response_invite: Received response: "Forbidden" from messages dialling out on a SIP trunk now..? |
| 10:59.01 | *** join/#asterisk renzoe (n=clc@213.132.40.2) |
| 10:59.12 | renzoe | hi guys. care to help me on a digium product? |
| 11:00.39 | renzoe | im planning to buy VPMADT032 echo cancellation. but i dont know how this thing works. it says it supports 32 channels and that means it can echo cancel 32 simultaneous calls or i need to assign it per phone to use this? |
| 11:19.00 | *** join/#asterisk chandoo (n=chandoo@ool-4353b978.dyn.optonline.net) |
| 11:20.21 | *** join/#asterisk maxvit (n=max@217.24.125.83) |
| 11:25.30 | *** join/#asterisk garymc (n=garymc@host81-134-0-102.in-addr.btopenworld.com) |
| 11:25.56 | garymc | Hi, I got the ports opened UDP 5060 , 10001-20000 |
| 11:26.10 | garymc | for sip device zioper to act as an extension |
| 11:26.59 | *** join/#asterisk magronez (n=eusei@unaffiliated/magrao/x-2903) |
| 11:27.04 | garymc | When the office calls my extension 202 and im at a remote location the call works great. Now when i remotley call extension 201 at the office, they hear me but i dont hear them. Im also breaking up alot. |
| 11:27.28 | garymc | What could be the cause of this? |
| 11:28.31 | *** join/#asterisk Chek (n=choog@91.192.150.1) |
| 11:29.54 | *** part/#asterisk maxvit (n=max@217.24.125.83) |
| 11:33.25 | Chek | guys, i need some help. i have a call-center with cisco 7940 as agents phones. agents log in via AgentCallbackLogin function. if call to agents phone directly and then dial queue phone #, call to queue ended with COMPLETEAGENT|0|0|1 |
| 11:33.48 | Chek | sorry for my language skills |
| 11:35.00 | Chek | any suggestions? |
| 11:39.47 | GNU\colossus | does the VoIP-encoding of POTS/ISDN-Calls happen at the asterisk server or at the asterisk server's VoIP-gateway (our VoIP-provider)? |
| 11:39.55 | tm1985 | Can anyone tell me of zaphfc works with mISDNv2 |
| 11:43.14 | Chek | GNU\colossus, all calls terminating by the asterisk server. sip only |
| 11:43.53 | Chek | for 7940 phones call-limit set to 1 |
| 11:44.19 | GNU\colossus | Chek: well, we're using a SIP provider, and there are inbound calls originating from non-voip phones. so the conversion to SIp takes place at our SIP-provider, is that correct? |
| 11:46.21 | Chek | GNU\colossus, our or your? i've been think that you did answer on my question :) |
| 11:47.53 | GNU\colossus | Chek: oh, sorry, we got something mixed up here. I was asking a very basic question myself, actually ;) |
| 11:48.00 | GNU\colossus | sorry |
| 11:48.30 | *** join/#asterisk coppice (n=chatzill@68.166.17.210.dyn.pacific.net.hk) |
| 11:48.51 | tm1985 | Can anyone tell me of zaphfc works with mISDNv2 |
| 11:50.01 | wathek | is there any web client to use a SIP account ? |
| 11:51.23 | tm1985 | Can anyone tell me of zaphfc works with mISDNv2 |
| 11:51.36 | Chek | GNU\colossus, but in your case, answer is "yes". sip provider myst convert traffic from pots/isdn to sip with some codecs (usually g711alaw/ulaw sometimes g729) communicate with sip provider |
| 11:51.48 | Chek | must* |
| 11:53.37 | GNU\colossus | Chek: thanks! does asterisk automatically determine which "incoming" codec is being used? |
| 11:54.46 | Chek | GNU\colossus, if asterisk know him, then 50/50 :) |
| 11:55.00 | garymc | anyone know if i need to reload anything after altering the RTP.conf file? |
| 11:55.32 | garymc | wathek : Zoiper ? |
| 11:55.46 | wathek | garymc, ok thank you |
| 11:56.08 | Chek | GNU\colossus, u can get list of codecs via "show codecs" command in asterisc console |
| 11:56.25 | garymc | wathek : zoiper is just a softphone. Peice of software you install on your machine. Is that what you mean? or Do you mean a website interface? |
| 11:57.06 | wathek | garymc, a website interfacez |
| 11:57.17 | *** join/#asterisk Freman (n=twitsrus@ppp178-75.static.internode.on.net) |
| 11:57.41 | GNU\colossus | Chek: thanks. I see a list of codecs; now how can I determine which one our SIP provider is using? |
| 11:57.48 | garymc | wathek : right i dont know of one, im sure one will be about that uses php or something. Not sure would be good to know though |
| 11:57.49 | GNU\colossus | (we have bad voice quality on incoming calls) |
| 11:58.07 | Freman | ok, so I've got my asterisk set up and running, I even sorted out the LUA extensions *grin*, I can make and recieve calls thruogh my vsp but I'm trying to place a call to a specific extension on another asterisk box on another network well beyond my controll and all I get is silence... |
| 11:58.10 | Freman | me or them? :) |
| 11:58.18 | wathek | garymc, could you please help me to try my asterisk configuration ? |
| 11:58.30 | garymc | i can if you want |
| 11:58.38 | wathek | garymc, ok thank you |
| 11:59.00 | Freman | dial(sip/sphinxtest@home.scribblej.com) |
| 11:59.27 | Chek | GNU\colossus, sip set debug ip host |
| 11:59.46 | Chek | GNU\colossus, or ask your provider |
| 12:01.37 | GNU\colossus | Chek: thanks very much for helping me out! :) |
| 12:01.49 | Chek | GNU\colossus, u r welcome :) |
| 12:03.22 | Chek | GNU\colossus, you can edit sip.conf and list codecs for providers gate, that you want |
| 12:06.16 | *** join/#asterisk italorossi (n=kvirc@189.23.15.3) |
| 12:06.27 | *** join/#asterisk Faustov (i=user@gentoo/user/faustov) |
| 12:08.06 | *** join/#asterisk gabri-shatana (n=shatana@95.235.120.253) |
| 12:08.07 | gabri-shatana | hi |
| 12:09.00 | *** join/#asterisk tcseke (n=chatzill@217.20.134.239) |
| 12:09.39 | gabri-shatana | what's the default password for asterisk user? |
| 12:10.35 | gabri-shatana | [Sep 15 14:10:21] NOTICE[2440]: chan_sip.c:14035 handle_request_invite: Call from '' to extension '055398**14' rejected because extension not found. |
| 12:10.56 | Chek | gabri-shatana, default on what? |
| 12:11.18 | Chek | extension not found |
| 12:11.27 | Chek | edit your dialplan |
| 12:11.33 | gabri-shatana | just a moment |
| 12:11.46 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 12:13.25 | gabri-shatana | !pastebin |
| 12:13.45 | Chek | gabri-shatana, also u may need to edit dialplan on your phone |
| 12:14.00 | gabri-shatana | phone? |
| 12:14.04 | gabri-shatana | i0'havent it |
| 12:14.26 | gabri-shatana | [055398**14] |
| 12:14.27 | gabri-shatana | type=user |
| 12:14.27 | gabri-shatana | host=dynamic |
| 12:14.27 | gabri-shatana | secret=******** |
| 12:14.27 | gabri-shatana | context=prova |
| 12:14.32 | Chek | Call from '' to extension '055398**14' -- asterisks in phone #? |
| 12:14.34 | gabri-shatana | this is into sip.conf |
| 12:14.45 | gabri-shatana | no |
| 12:14.54 | gabri-shatana | i have a system() |
| 12:15.17 | gabri-shatana | [prova] |
| 12:15.17 | gabri-shatana | exten => 2,1,System(Halt) |
| 12:15.26 | gabri-shatana | this is into extension.conf |
| 12:15.33 | gabri-shatana | *extensions |
| 12:17.04 | gabri-shatana | register => "call" the [05539****] who call [prova] |
| 12:18.31 | gabri-shatana | http://pastebin.com/d61c31437 |
| 12:19.27 | Chek | gabri-shatana, if u want to call on sip phone, u must use something like Dial(SIP/ext@host) |
| 12:19.34 | Chek | in your dialplan |
| 12:21.13 | Freman | can anyone tell me why I can hear sound when I call through (either way) my vsp, but when I dial(sip/sphinxtest@home.scribblej.com) I get nothing? |
| 12:22.36 | Chek | Freman, did home.scribblej.com in your sip.conf? |
| 12:24.35 | Freman | no, it's just a random one off call |
| 12:24.53 | tzafrir_laptop | infobot, tell gabri-shatana about pb |
| 12:25.00 | Chek | Freman, you get answer :) |
| 12:25.04 | Freman | if it helps diagnose, I have a sip client outside of my network and it has no sound when calling in |
| 12:25.07 | Chek | got* |
| 12:25.39 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:27.19 | garymc | hi [TK]D-Fender |
| 12:27.28 | gabri-shatana | i don't want call form sip phone |
| 12:27.35 | gabri-shatana | i want call the sip phone by a normal phone |
| 12:27.39 | garymc | I got the ports open UDP 5060 , 10001-20000 |
| 12:27.54 | gabri-shatana | and pressing some number i want happens something |
| 12:28.04 | gabri-shatana | but asterisk has that error |
| 12:28.18 | [TK]D-Fender | gabri-shatana: Pastebin the call attempt |
| 12:28.24 | gabri-shatana | NOTICE[2440]: chan_sip.c:14035 handle_request_invite: Call from '' to extension '055398**14' rejected because extension not found. |
| 12:28.38 | garymc | [TK]D-Fender : Now when the office calls my extension and im at a remote location it works great. But if I dial the office remotley from zoiper i cant hear anything. They hear me but its all broken sound |
| 12:29.20 | [TK]D-Fender | gabri-shatana: look at the SIP DEBUG to see what PEER it is matching and what CONTEXT it is looking for that extension in. there clearly isn't a match for it there |
| 12:29.39 | [TK]D-Fender | garymc: Try another client |
| 12:29.57 | garymc | like what |
| 12:30.07 | gabri-shatana | [TK]D-Fender, <gabri-shatana> [Sep 15 14:10:21] NOTICE[2440]: chan_sip.c:14035 handle_request_invite: Call from '' to extension '055398**14' rejected because extension not found. |
| 12:30.09 | gabri-shatana | sorry |
| 12:30.15 | gabri-shatana | [TK]D-Fender, http://pastebin.com/d61c31437 |
| 12:30.30 | [TK]D-Fender | ~sofphone |
| 12:30.44 | garymc | [TK]D-Fender any recomendations? |
| 12:30.59 | [TK]D-Fender | ~softphone |
| 12:30.59 | infobot | [~softphone] A soft-phone is a program that lets your computer use a compatible audio interface (usually a speaker&mic, or headset), and optionally a webcam, to place VoIP calls with a given protocol (SIP/IAX/H.323/MGCP/etc). For links enter ~zoiper , ~xlite , ~twinkle , ~bria , ~eyebeam , ~ekiga |
| 12:31.02 | [TK]D-Fender | garymc: WhATEVER |
| 12:31.03 | Chek | garymc, i'm useing sjphone for tests |
| 12:31.17 | garymc | is it working ok |
| 12:31.20 | Chek | works perfect |
| 12:31.33 | garymc | Im using Zoiper, but it not working when i call the office remotley |
| 12:31.35 | [TK]D-Fender | gabri-shatana: And just like the error says, the call is looking for that # at the end of your register statement and you have NO MATCH. So go make one |
| 12:31.46 | gabri-shatana | ... |
| 12:31.50 | gabri-shatana | i have no peers |
| 12:31.56 | garymc | Chek : Would you loginto my asterisk box and test for me? |
| 12:31.57 | Chek | [TK]D-Fender, i told him :) |
| 12:32.00 | gabri-shatana | i want only use System() |
| 12:32.25 | Chek | garymc, one moment |
| 12:32.46 | Chek | gabri-shatana, but you trying to call on phone |
| 12:32.59 | gabri-shatana | i call by my mobile |
| 12:33.03 | gabri-shatana | to the voip number |
| 12:33.06 | *** join/#asterisk casnik (n=Nick@fw1-e0-2.dth.xiocom.net) |
| 12:33.21 | gabri-shatana | *from |
| 12:33.36 | gabri-shatana | * from my mobile to my voip number |
| 12:33.45 | [TK]D-Fender | gabri-shatana: http://pastebin.com/m16db20f8 |
| 12:35.19 | gabri-shatana | http://pastebin.com/d428e589f |
| 12:36.53 | Chek | garymc, give me connection settings |
| 12:36.56 | [TK]D-Fender | gabri-shatana: NO. read my pastebin AGAIN |
| 12:38.21 | dandre | hello, |
| 12:39.40 | [TK]D-Fender | gabri-shatana: [whatever[ <- sections like this in sip.conf are cgenerally called "peers" here |
| 12:39.59 | gabri-shatana | ok.. |
| 12:40.27 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 12:40.35 | dandre | if I use extensionstate manager command to know wether an extension is in use or not, I get a status of 8 if ringin that is correct. But if my phone issue a call, the status is still 0. This behaviour is for a sip phone |
| 12:40.52 | dandre | btw the status is correct for a zap phone |
| 12:41.39 | [TK]D-Fender | dandre: Ringing is what the phone is doing, not what it is hearing. |
| 12:42.43 | dandre | ok but when I place a call the sttus shouldn't be 0 = Not in use |
| 12:42.57 | *** join/#asterisk Skeeter- (n=wil_c_wi@c216.218.2-65.clta.globetrotter.net) |
| 12:43.16 | Skeeter- | morning guys |
| 12:44.08 | *** join/#asterisk oej (n=olle@132.177.253.250) |
| 12:44.09 | dandre | I expected the status to be 1 = In Use |
| 12:44.40 | Skeeter- | i got 1 servers in each facilities(got 2 facilities) each server is trunking a voip provider. i need to trunking everything. calling server 1, dialing server 2 ext. calling server 1 voip dialing server 2 ext. etc... |
| 12:44.41 | garymc | afternoon ;P |
| 12:44.54 | [TK]D-Fender | dandre: pastebin <- |
| 12:45.15 | [TK]D-Fender | dandre: because I strongly suspect that its fine and your understanding is skewed |
| 12:45.45 | [TK]D-Fender | Skeeter-: #freepbx <----- |
| 12:45.58 | dandre | do you the manager trace? |
| 12:46.14 | [TK]D-Fender | dandre: that too. |
| 12:47.20 | *** join/#asterisk xpot (n=james@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 12:48.02 | dwery | [TK]D-Fender: got a little patch for you http://pastebin.com/f25e93ed8 |
| 12:48.12 | dwery | [TK]D-Fender: solves the double sip: in some uris |
| 12:48.36 | Skeeter- | aight |
| 12:48.57 | [TK]D-Fender | dwery: What would I do with this? |
| 12:49.12 | dwery | [TK]D-Fender: thought you were involved in the development |
| 12:49.21 | [TK]D-Fender | dwery: Nope |
| 12:49.32 | dwery | [TK]D-Fender: ouch, sorry ;) |
| 12:51.54 | Chek | [TK]D-Fender, maybe you may halp me. . i have a call-center with cisco 7940 as agents phones. agents log in via AgentCallbackLogin function. if call to agents phone directly and then dial queue phone #, call to queue ended with COMPLETEAGENT|0|0|1 |
| 12:54.16 | *** join/#asterisk friehmaen (i=freeman@xers.de) |
| 12:55.01 | dandre | [TK]D-Fender: manager trace for a call from sip/43 to ZAP/2, at the end the extenstate command: http://pastebin.fr/5554 |
| 12:56.55 | [TK]D-Fender | dandre: Where do I see that it shoudl say otherwise? |
| 12:57.30 | dandre | sip/43 is placing a call to zap/2 |
| 12:57.42 | dandre | so it is in use |
| 12:57.47 | [TK]D-Fender | dandre: .... and? |
| 12:58.45 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:58.45 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:59.04 | [TK]D-Fender | dandre: SIP/43 is a ***DEVICE*** not an EXtensioN |
| 12:59.16 | [TK]D-Fender | dandre: You don't really seem to understand the difference |
| 13:01.18 | leifmadsen | a device called [43] |
| 13:01.35 | leifmadsen | you should really abstract extensions from users, and users from devices. |
| 13:01.57 | leifmadsen | using the MAC address is better practise when naming devices |
| 13:05.21 | [TK]D-Fender | dandre: exten => 43,1,Dial(SIP/fred) . If a device authing calls as [mary] from sip.conf dialed that exten, then EXTENSION 43 is "ringing" |
| 13:05.39 | dandre | ok but if I place a call from zap/2 (which is extension 50@from-internal) to sip/43 every seems ok: http://pastebin.fr/5555 |
| 13:06.00 | [TK]D-Fender | leifmadsen: I like making sure when I want focus on a number, that nothing else resembles it :) |
| 13:06.21 | *** join/#asterisk spck (n=spck@unioncab.com) |
| 13:06.35 | [TK]D-Fender | dandre: because the EXTENSION in the dialplan happens to share the same name as the device. |
| 13:06.44 | [TK]D-Fender | dandre: exten => 43,1,Dial(SIP/fred) . If a device authing calls as [mary] from sip.conf dialed that exten, then EXTENSION 43 is "ringing" <----------- |
| 13:06.59 | [TK]D-Fender | dandre: otherwise Wtf would it need to knwo the CONTEXT for? |
| 13:07.08 | [TK]D-Fender | dandre: extensionstate looks at DIALPLAN. |
| 13:07.45 | dandre | ok but I have a hint => in from-internal context |
| 13:07.55 | [TK]D-Fender | dandre: IRRELEVANT |
| 13:08.39 | [TK]D-Fender | dandre: hints a re for presence and is completely separate from extensionstate |
| 13:08.43 | *** join/#asterisk mutante (i=mutante@wiktionary/Mutante) |
| 13:08.53 | dandre | <PROTECTED> |
| 13:09.20 | [TK]D-Fender | dandre: hints a re for presence and is completely separate from extensionstate <------ |
| 13:09.43 | dandre | ok so how can I know the stae of a device from the manager |
| 13:09.59 | [TK]D-Fender | dandre: extensiosnstate in no way has anything to do with the status of a device |
| 13:10.08 | [TK]D-Fender | dandre: Go look at the other AMI commands |
| 13:10.56 | *** join/#asterisk coppice (n=chatzill@68.166.17.210.dyn.pacific.net.hk) |
| 13:11.43 | mutante | hi, i find many examples of how to play a .gsm file with Playback() when Asterisk is being called. But i would like that Asterisk is the client that calls me, and after i pickup it should play the message. I can let my phone ring by now, but it does not play a message, but hangs up. I tried this extension,, in order: _X.,1,SetCallerId,SIPID _X.,2,Dial(SIP/${EXTEN}@sipgate-out,30,trg) _X.,3,Playback(alarm-foobar) _X.,4,Hangup |
| 13:11.59 | mutante | is that the right way to go? SetCallerId, Dial, Playback, Hangup? |
| 13:12.13 | leifmadsen | [TK]D-Fender: using a persons name is no better an abstraction than a number |
| 13:12.30 | kaldemar | mutante: no, use a Dial option that executes something on the called channel when it answers |
| 13:12.44 | [TK]D-Fender | leifmadsen: it is when I don't want them mixing up fred vs 42 :) |
| 13:13.37 | leifmadsen | if you're going to teach someone the proper abstractions, you should just do it right the first time, and not just substitute one poor naming convention for another |
| 13:13.47 | mutante | kaldemar: i may not get it, but the phone that i call is not connected to Asterisk. Asterisk is the client to a SIP provider, and from SIP provider it calls the POTS phone on my desk |
| 13:14.42 | kaldemar | mutante: it doesn't matter whether the phone is directly connected to your asterisk box or not |
| 13:14.51 | mutante | ok |
| 13:15.18 | kaldemar | mutante: "core show application Dial" will give you more than one possible option to do that. |
| 13:15.31 | [TK]D-Fender | leifmadsen: Oh I fdid that too... some people are considerably thicker than bricks :) |
| 13:15.36 | mutante | kaldemar: thank you |
| 13:15.43 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:16.12 | *** part/#asterisk redax (i=redax@r6.hu) |
| 13:16.27 | mutante | kaldemar: ok, more googling... i need that in a call file.. but ok.. rtfm |
| 13:17.13 | kaldemar | mutante: it's just as doable with a call file |
| 13:17.54 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:18.27 | casnik | Anyone know where to obtain a copy of Asterisk Cookbook from oreilly? (is it even worth it to get now anyway?) |
| 13:18.27 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
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| 13:24.35 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 13:26.49 | Freman | scratches head - DEVICE_STATE, ChanIsAvail, Sippeer - none of them can tell me if a device has an open line? |
| 13:27.15 | casnik | leifmadsen, know where to obtain a copy of your Asterisk Cookbook from oreilly? (is it even worth it to get now anyway?) |
| 13:27.27 | garymc | [TK}D-Fender : I look at sip debug in asterisk cli I recieve call from sip user and it shows them on port 5061 why is that? |
| 13:27.30 | leifmadsen | casnik: the cookbook was never written |
| 13:27.43 | casnik | oh? I just saw it out of print |
| 13:27.44 | *** join/#asterisk superbeef (n=superbee@74.84.194.4) |
| 13:27.47 | casnik | booo! |
| 13:27.54 | garymc | [TK]D-Fender : ^^ |
| 13:28.05 | dwery | Freman: probably not :D But if you find a way please drop me a note ;) |
| 13:28.14 | leifmadsen | casnik: www.asteriskcookbook.com might have some useful info though |
| 13:28.16 | *** join/#asterisk flohack (n=fhackenb@lancelot.acoveo.com) |
| 13:28.32 | leifmadsen | casnik: note that was intended to be the working version of the cookbook, but it never happened |
| 13:28.40 | leifmadsen | although it seems other people have added stuff |
| 13:28.44 | casnik | leifmadsen, yeah I looked at that some , wasn't sure how out of date it was considering I am still pretty noob |
| 13:28.45 | dwery | Freman: btw DAHDI has no support for devicestate, according to "core show channeltypes" |
| 13:28.58 | kaldemar | casnik: http://etel.wiki.oreilly.com/wiki/index.php/Main_Page |
| 13:29.02 | leifmadsen | casnik: ya, I would suggest that to be your primary resource |
| 13:29.10 | *** join/#asterisk Naikrovek (n=jjohnson@63-252-251-77.ip.mcleodusa.net) |
| 13:29.14 | leifmadsen | kaldemar: that is the old link -- to the site I just provided above |
| 13:29.21 | Freman | dwery looking for sip |
| 13:29.42 | Freman | http://fremnet.net/article/194/channel-check used to work... perhaps I need to revive it |
| 13:29.48 | kaldemar | leifmadsen: gah. mut human eyes didn't recognize it without http:// :P |
| 13:29.53 | leifmadsen | :) |
| 13:29.58 | casnik | leifmadsen, thanks a ton I thought I was missing out on some literal gem that mysteriously went oout of print |
| 13:30.00 | ross` | hey, im trying to get some ip handsets that work on my wireless network |
| 13:30.21 | ross` | i basically want normal wireless phones that wont scare my mom |
| 13:32.11 | garymc | anyone know why a remote caller is using port 5061? |
| 13:32.28 | leifmadsen | garymc: because they wanted to? |
| 13:32.35 | casnik | doesn't that mean tls? |
| 13:32.40 | leifmadsen | garymc: (typically because something else on their network was registered with 5060 already) |
| 13:33.13 | kaldemar | casnik: no, it only means they use port 5061, nothing more. :P |
| 13:33.47 | leifmadsen | I set my 2 polycoms to register as 5501 and 5430 so I know which phones I'm looking at on the screen |
| 13:33.48 | leifmadsen | :) |
| 13:33.53 | casnik | right , yeah ... totally off on that ... I recalled something about that in the opensips default stuff (totally wrong disregard) |
| 13:34.49 | garymc | but what if those ports arnt open in the NAT? |
| 13:34.58 | leifmadsen | they don't need to be typically |
| 13:35.10 | leifmadsen | on the PBX side, the phone is registering TO 5060, but FROM 5061 |
| 13:35.52 | *** join/#asterisk asif (n=chatzill@122.166.40.72) |
| 13:36.02 | asif | hello all! |
| 13:36.03 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 13:36.22 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 13:36.42 | asif | I'm having trouble getting adaptive ODBC up with MySQL |
| 13:39.56 | garymc | ok |
| 13:40.30 | asif | please see this: http://www.pastebin.ca/1566993 |
| 13:40.30 | garymc | so why do calls to a remote extension only work when I call the remote extension from the office. |
| 13:40.56 | garymc | When the remote extension call the office, the remote caller gets no sound, but the office hears the sound? |
| 13:41.09 | Naikrovek | garymc: NAT problems |
| 13:41.34 | garymc | well im only using one router now with all ports opened |
| 13:41.52 | Naikrovek | you are, office may be using something different |
| 13:41.57 | Naikrovek | and don't announce that out loud |
| 13:42.04 | Naikrovek | your IP address is not hidden |
| 13:42.06 | garymc | no not all ports |
| 13:42.13 | asif | any idea what could be wrong? |
| 13:42.13 | garymc | just the udp 5060 etc |
| 13:42.41 | garymc | im in the office now |
| 13:42.45 | *** join/#asterisk Whitor (n=Whitor@24.97.4.146) |
| 13:42.51 | garymc | so i opened all ports |
| 13:42.56 | Naikrovek | asif: "data source name not found" there's your problem |
| 13:43.17 | garymc | Naikrovek : Dont suppose you use a decent softphone and you could do a quick test with me? |
| 13:43.28 | *** join/#asterisk youngproguru (n=youngpro@smtp.deltasoniccarwash.com) |
| 13:43.29 | Naikrovek | garymc: i don't, sorry |
| 13:43.36 | garymc | ok |
| 13:43.43 | Naikrovek | asif: http://forums.digium.com/viewtopic.php?p=30324&sid=e7bfb1de7759e55564a6c6cb16138b11 |
| 13:43.54 | garymc | Anyone else have a softphone that works other than Zoiper they could do a quick test with? |
| 13:44.03 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:46.11 | SuPrSluG | garymc: if you're getting one way audio it is a nat issue. |
| 13:46.14 | *** join/#asterisk retentiveboy (n=pdugas@atl.pra-corp.com) |
| 13:46.26 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 13:46.28 | *** part/#asterisk italorossi (n=kvirc@189.23.15.3) |
| 13:46.28 | garymc | when i call from office to remote its two way audio |
| 13:46.37 | garymc | just remote to office is one way |
| 13:46.54 | SuPrSluG | remote to office is nat issue |
| 13:47.04 | garymc | right.... hmmm |
| 13:47.12 | garymc | so what should i do? |
| 13:47.33 | garymc | is it ports 10000-20000 or ports 10001-20000 |
| 13:47.51 | garymc | i think i done ports 10001-20001 |
| 13:48.29 | dandre | [TK]D-Fender: I had to put call-limit and limitonpeers according to this issue: https://issues.asterisk.org/view.php?id=8800 |
| 13:48.38 | dandre | and everything work |
| 13:48.55 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 13:49.15 | garymc | Also do i have to reload anything after altering rtp.conf? |
| 13:49.17 | [TK]D-Fender | dandre: Yes, that is required for hints, but still has absolutely nothing to do with ExtensionState |
| 13:50.16 | superbeef | are there configure optiosni need to pass to get asterisk 1.4.26 to use pri, zaptel? I built pri, then zaptel before asterisk but I seem to lack the pri show command when I run asterisk |
| 13:50.26 | dandre | but extensionstate works now |
| 13:51.26 | asif | thanks for that link, Naikrovek |
| 13:51.37 | kaldemar | garymc: it is what you define in rtp.conf. by default, 10000-20000. |
| 13:51.48 | garymc | ok |
| 13:51.56 | asif | though i'm getting an "invalid object" error with odbcinst |
| 13:52.05 | dandre | superbeef: as far as I know, zaptel isn't supported for version newer than 1.4.2 |
| 13:52.05 | garymc | if i set NAT in router to 10001-20000 would that cause an issue? |
| 13:52.08 | dandre | 1.4.21 |
| 13:52.51 | superbeef | hmm |
| 13:53.20 | superbeef | zaptel is supposed to still be available in 1.4 |
| 13:53.33 | superbeef | Asterisk 1.4 releases later than 1.4.21, and all releases of Asterisk 1.6, will automatically use |
| 13:53.34 | superbeef | DAHDI in preference to Zaptel, even if Zaptel is still installed on the system. |
| 13:53.49 | *** join/#asterisk propellerhead (n=yogurt2u@host251.200-82-124.telecom.net.ar) |
| 13:53.52 | garymc | anyone know what these options in my router mean. 1.stealth mode - 2. Block Ping -- 3.strict UDP session control |
| 13:54.00 | garymc | The only one ticked is number 1 |
| 13:54.03 | superbeef | i guess I could suck it up and try to get my sangoma card to play with DAHDI |
| 13:54.29 | kaldemar | garymc: office end hears the sound, you said earlier. stop playing with the office side router then and try to find the real problem. |
| 13:54.41 | _trine | garymc, you need to get your computer locked down ,, Finance Facility Customer Detail View |
| 13:54.53 | garymc | yeah thanks for that |
| 13:55.06 | garymc | im just testing with that page |
| 13:55.11 | garymc | enter something in if you like |
| 13:55.18 | garymc | :) |
| 13:55.55 | _trine | well i hope your customer ID's are only test as well |
| 13:56.18 | garymc | kaldemar: Thats my problem what do i need to do then? |
| 13:56.24 | *** join/#asterisk coppice (n=chatzill@68.166.17.210.dyn.pacific.net.hk) |
| 13:56.28 | garymc | they all arfe _trine |
| 13:56.46 | kaldemar | garymc: make RTP packets from asterisk reach the remote client. |
| 13:56.52 | _trine | ok |
| 13:56.55 | *** join/#asterisk TJNII (n=TJNII@207.189.199.58) |
| 13:57.01 | _trine | just thought I would let you know |
| 13:57.05 | garymc | kaldemar: how? |
| 13:57.30 | kaldemar | garymc: depends on the problem. |
| 13:57.35 | garymc | _trine : you wont beable to look now i dont think |
| 13:57.59 | kaldemar | garymc: have you ever take a sip debug of a failed call and shown it here? |
| 13:58.06 | garymc | yes |
| 13:58.10 | garymc | to no avail |
| 13:59.08 | kaldemar | keep showing actual information of your setup instead of falling back to the same general questions over and over again. |
| 13:59.37 | garymc | right so you want a paste of actuall call where i get audio but remote gets none? |
| 14:00.22 | kaldemar | yes. with sip.conf and a description of the current network setup. |
| 14:01.35 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 14:01.50 | garymc | right |
| 14:01.56 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 14:01.56 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:06.26 | *** join/#asterisk dajhorn (n=dajhorn@transmisor.vanadac.com) |
| 14:07.16 | garymc | right ok i had my firewall ports on router set to 10001-20000 |
| 14:07.33 | garymc | I changed the setting to 10000-20000 and now it works GREAT! |
| 14:08.56 | *** join/#asterisk creativx (n=creadure@197.82-134-19.bkkb.no) |
| 14:11.05 | *** join/#asterisk voipmonk (n=voipmonk@67.212.7.67) |
| 14:11.35 | *** join/#asterisk Grof (n=dule@89.201.165.226) |
| 14:11.43 | Grof | hey guys |
| 14:12.32 | Grof | anyone up for answering some ConfBridge questions? |
| 14:12.52 | leifmadsen | ~ask |
| 14:12.53 | 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. |
| 14:13.17 | voipmonk | corrects his robes |
| 14:13.42 | coppice | beta questions frequently yield alpha answers |
| 14:13.47 | casnik | straightens his wizard hat |
| 14:13.58 | Grof | since now i've used MeetMe and his BACKGROUND_AGI functionality |
| 14:14.23 | seanbright | MeetMe is actually a she |
| 14:14.27 | seanbright | and she's a whore |
| 14:14.31 | Grof | is there a way to get the same (or similiar) behaviour with ConfBridge |
| 14:14.32 | Grof | ? |
| 14:15.02 | *** join/#asterisk Moz (n=me@81.179.238.144) |
| 14:15.22 | *** join/#asterisk oej_ (n=olle@132.177.254.166) |
| 14:15.37 | seanbright | Grof: let me take a look at the source. moment. |
| 14:16.08 | *** join/#asterisk oej__ (n=olle@132.177.253.250) |
| 14:16.24 | seanbright | no. nothing built-in. |
| 14:17.00 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 14:17.35 | *** join/#asterisk moy (n=moy@74.12.131.104) |
| 14:18.01 | Grof | the problem is that i want to have control over channel after I put it in the conference room |
| 14:18.23 | ccesario | hello.... |
| 14:18.28 | Grof | what if I used ast_bridge_impart instead of ast_bridge_join? |
| 14:18.31 | ccesario | somebody have any idea about this ? |
| 14:18.33 | ccesario | http://pastebin.com/m6b1c4a6a |
| 14:18.57 | Grof | (through FastAGI) |
| 14:19.21 | leifmadsen | ccesario: I've not used directed pickup, but it looks like your format might be wrong? |
| 14:19.32 | leifmadsen | should it be SIP/3602@something ? |
| 14:19.53 | leifmadsen | given, I didn't look at the 'core show application' output |
| 14:20.52 | ccesario | leifmadsen, hmmm let me try this |
| 14:23.54 | *** join/#asterisk UQlev (n=yuriy@87.228.199.125) |
| 14:24.19 | Grof | noone' |
| 14:24.20 | Grof | ? |
| 14:24.24 | Grof | :( |
| 14:24.53 | Grof | is ConfBridge developer visiting this channel |
| 14:24.54 | Grof | ? |
| 14:25.04 | Grof | (Joshua Colp) |
| 14:25.23 | leifmadsen | file is currently at SIPit doing mad amounts of testing on chan_sip |
| 14:25.35 | kaldemar | Grof: there is also #asterisk-dev for discussing the code |
| 14:25.43 | Grof | y? ok, tnx |
| 14:25.49 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 14:25.50 | leifmadsen | Grof: and you'll probably get more response from asterisk-dev mailing list |
| 14:26.14 | Grof | tnx leifmadsen |
| 14:26.24 | leifmadsen | Grof> y? ok, tnx <-- this is what is wrong with Americas spelling today :) |
| 14:26.54 | *** join/#asterisk garymc (n=garymc@host81-134-0-102.in-addr.btopenworld.com) |
| 14:26.54 | ccesario | leifmadsen, as http://www.voip-info.org/wiki/view/Asterisk+cmd+Pickup |
| 14:27.02 | ccesario | Im using this http://pastebin.com/m2e958dd3 |
| 14:27.03 | Grof | leifmadsen: sorry :( i'm from croatia |
| 14:27.24 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 14:27.32 | Grof | leifmadsen: i do not know your ways XD |
| 14:27.37 | casnik | yeah luckily not all of us Americans spell like that |
| 14:27.39 | leifmadsen | you mean like... full words? :) |
| 14:27.47 | Grof | leifmadsen: :P |
| 14:28.11 | Grof | y, never heard of "thanks", only "tnx", "10x", "gr8" |
| 14:28.12 | Grof | :D |
| 14:28.24 | leifmadsen | <G> |
| 14:28.43 | leifmadsen | ccesario: interesting... I've never used it, but it looks like the right syntax unless it has changed in 1.6 or something |
| 14:29.11 | casnik | outside of American chat culture people use "y" to say "yes" whereas inside we use "y" to say " why" |
| 14:29.20 | [TK]D-Fender | O U 8 1 2? |
| 14:29.57 | leifmadsen | casnik: I just type 'yes' and 'why' because saving 2 characters doesn't make up for the lack of useful communication |
| 14:30.08 | Nugget | I like when stupid chat laziness has a language collision. Like saying "N8 M8" |
| 14:30.09 | [TK]D-Fender | I O U 1 U C..... |
| 14:30.10 | ccesario | leifmadsen, in asterisk-1.6.1 this work ... but now in 1.6.2 dont work :/ |
| 14:30.29 | leifmadsen | ccesario: then in that case you make wish to file an issue |
| 14:30.33 | casnik | leifmadsen, me to except I spell them out because I communicate with a lot of people outside the US |
| 14:31.41 | *** join/#asterisk wathek (n=wathek@41.224.202.206) |
| 14:33.43 | seanbright | i hope that any features that are added to ConfBridge are well thought out |
| 14:33.51 | *** join/#asterisk lordmortis (n=lordmort@203-59-207-20.dyn.iinet.net.au) |
| 14:33.58 | *** join/#asterisk andres833 (n=andres83@190.144.75.22) |
| 14:33.59 | *** join/#asterisk Faithful (n=Faithful@124.217.119.133) |
| 14:34.08 | seanbright | instead of the typical "here is a variable or flag for every thing anyone could ever do" pattern |
| 14:34.50 | *** join/#asterisk lordmortis (n=lordmort@203-59-207-20.dyn.iinet.net.au) |
| 14:35.02 | Chek | still need some help. i have a call-center with cisco 7940 as agents phones. agents log in via AgentCallbackLogin function. if call to agents phone directly and then dial queue phone #, call to queue ended with COMPLETEAGENT|0|0|1 |
| 14:38.55 | *** join/#asterisk Dovid (n=annon@67.85.226.151) |
| 14:42.50 | Guest70048 | is there a way to setup a call routing table in mysql or something |
| 14:43.00 | Guest70048 | like if someone dials one it goes to 101 |
| 14:43.28 | afink | thanks leifmadsen |
| 14:43.38 | raden | or a way to make it so via web interface I can change where calls are routing to for the day ? |
| 14:43.43 | leifmadsen | afink: lol nice :) |
| 14:43.57 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 14:43.57 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:44.12 | leifmadsen | raden: sure -- sounds like you want something like func_odbc |
| 14:44.21 | leifmadsen | putnopvut: the master of queues! |
| 14:44.48 | angryuser | Good day, i am searching stable DID providers from Spain and Usa, can someone reccomend me ? Thank you. |
| 14:45.22 | leifmadsen | infobot: tell angryuser about itsp-us |
| 14:45.34 | raden | i want something when we come in, in the morning that i can goto a browser set option 1 going to 101 option 2 going to 103 etc... |
| 14:45.37 | leifmadsen | hmmm.... I thought that was the tag |
| 14:45.42 | raden | cause when people are out things become a pITA |
| 14:45.44 | *** join/#asterisk retentiveboy (n=pdugas@atl.pra-corp.com) |
| 14:46.07 | leifmadsen | raden: sure, just use func_odbc in your dialplan to lookup what to dial from the database |
| 14:46.17 | leifmadsen | it'll get looked up each time someone hits your IVR |
| 14:46.22 | angryuser | ~providers |
| 14:46.23 | infobot | [providers] http://www.voipreview.org/service.all2.aspx?Country=1&Area_Code=0&CallingArea=0&provider=0&serviceType=1&Adv=1&Features=43 |
| 14:46.28 | leifmadsen | ~itsp-list |
| 14:46.28 | infobot | extra, extra, read all about it, itsp-list is Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 14:46.53 | leifmadsen | I personally have had pretty good luck with bandwidth.com |
| 14:46.58 | raden | leifmadsen, so i can use mysql then |
| 14:47.05 | leifmadsen | raden: yes |
| 14:47.13 | angryuser | leifmadsen, thanks |
| 14:47.15 | leifmadsen | raden: see the database chapter of TFoT |
| 14:47.15 | raden | sweet as long as mysql dont fail LOL |
| 14:47.20 | raden | Tfot ? |
| 14:47.20 | leifmadsen | raden: right |
| 14:47.28 | leifmadsen | infobot: tell raden about thebook |
| 14:47.36 | raden | oreilly book i have it |
| 14:47.46 | leifmadsen | TFoT == shorthand for The Future of Telephony |
| 14:47.54 | leifmadsen | ~tfot |
| 14:47.55 | infobot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 14:48.01 | leifmadsen | aha, someone added that already |
| 14:48.02 | leifmadsen | nice |
| 14:48.14 | leifmadsen | although that is old data... |
| 14:48.45 | Grof | :D |
| 14:48.54 | leifmadsen | and now I've updated it |
| 14:49.05 | leifmadsen | infobot: thanks! |
| 14:49.05 | infobot | leifmadsen: gern geschehen |
| 14:50.52 | raden | whats the proper way to start and stop asterisk outside the asterisk CLI ? |
| 14:50.57 | *** join/#asterisk retentiveboy (n=pdugas@atl.pra-corp.com) |
| 14:51.42 | Katty | GOOD MORNING |
| 14:51.52 | *** join/#asterisk wcselby (n=wcselby@216.110.88.254) |
| 14:52.28 | Katty | it's a beautiful day in the neighborhood! |
| 14:52.33 | Katty | a beautiful day for a neighbor! |
| 14:52.40 | [TK]D-Fender | Katty: Would you be min? Could you be min? |
| 14:52.44 | wcselby | won't you be my neighbor |
| 14:52.45 | [TK]D-Fender | mine* |
| 14:53.47 | kaldemar | raden: an init script usually |
| 14:53.56 | *** join/#asterisk Faithful (n=Faithful@124.217.119.166) |
| 14:53.58 | Katty | let's make the most of this beautiful day! |
| 14:55.30 | raden | anyone think of a way to pickup another extension that is rining lets say im at 103 and i hear 101 running id like to be able to hit like #101 or something to pickup that extension is it possible |
| 14:56.15 | *** join/#asterisk retentiveboy (n=pdugas@atl.pra-corp.com) |
| 14:56.18 | Pan3D | writes a mrmcfeely context |
| 14:56.58 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:57.10 | *** join/#asterisk wathek (n=wathek@41.224.199.142) |
| 14:58.35 | *** join/#asterisk Moz (n=me@81.179.238.144) |
| 14:59.05 | *** join/#asterisk retentiveboy (n=pdugas@atl.pra-corp.com) |
| 14:59.42 | Katty | i have a lot of respect for him. |
| 15:00.00 | Katty | or his memory, i guess. |
| 15:00.49 | Moz | has anyone got any experience using chan_mobile? |
| 15:01.30 | *** join/#asterisk wonderworld (n=w@ip-62-143-22-226.unitymediagroup.de) |
| 15:02.29 | wonderworld | hey, is there a way to log the cli outut into a text-file? |
| 15:03.04 | leifmadsen | ccesario: mnicholson just mentioned he is working on a directed pickup bug, but not sure if it is related to your issue or not |
| 15:03.17 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 15:03.29 | leifmadsen | wonderworld: asterisk -rvvvv | tee /tmp/myfile.txt |
| 15:03.56 | wcselby | you could also play with the logger.conf settings to get a lot of the * info into a log file |
| 15:04.46 | ccesario | leifmadsen, hmmmm cool news... when he fix the bug I can test ... |
| 15:05.25 | leifmadsen | ccesario: coolio, not sure what the bug number is. I'll try and find out. |
| 15:05.40 | leifmadsen | mnicholson: are you paying attention here? |
| 15:08.48 | ccesario | leifmadsen, yea... but the mnicholson patch, can solve this tooo :D |
| 15:08.56 | leifmadsen | ok great :) |
| 15:10.51 | Chek | hehe, found it :) it's error in source |
| 15:12.29 | *** join/#asterisk jeroen_h (n=jeroen@082-146-101-077.stat.adsl.xs4all.be) |
| 15:12.31 | ccesario | leifmadsen, ;) |
| 15:18.14 | *** join/#asterisk Mango (n=Mango@96.49.69.137) |
| 15:18.28 | Mango | Can anyone tell me how many gateways a SPA3102 supports? |
| 15:20.32 | *** join/#asterisk asterwiki (n=asterwik@69.77.169.14) |
| 15:20.44 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-94-246.no.no.cox.net) |
| 15:23.01 | wcselby | Mango - http://www.cisco.com/en/US/products/ps10027/ |
| 15:29.53 | *** join/#asterisk hugorebelo (n=hugorebe@200-171-132-124.completo.com.br) |
| 15:30.05 | *** join/#asterisk nfi|ermes (n=ErMeS@217.220.121.62) |
| 15:31.09 | *** join/#asterisk ttl- (n=patrick@94-224-78-192.access.telenet.be) |
| 15:36.52 | *** join/#asterisk voipmonk_ (n=voipmonk@dsl-67-212-15-216.acanac.net) |
| 15:37.35 | *** join/#asterisk proute (n=AnthonyC@LMontsouris-152-63-16-150.w217-128.abo.wanadoo.fr) |
| 15:37.38 | proute | hello |
| 15:37.46 | wcselby | howdy |
| 15:38.19 | proute | I use * 1.4.25.1 use misdn 1.1.9-2. today on my log i have: mISDN_rdata: rport queue overflow (after 500ms!) 256/256 [addr:52010401 prim:120282 dinfo:ffffffff] |
| 15:38.20 | *** join/#asterisk p3nguin (i=BuhPjX1J@asterisk-klv5.a2infotech.com) |
| 15:38.36 | proute | Does anybody have any idea about this log? |
| 15:38.49 | proute | Moreover this log flood syslog |
| 15:38.57 | proute | thanks for your help |
| 15:44.25 | wcselby | sorry, don't use mISDN, can't help. someone here might be able to though |
| 15:45.26 | proute | ok thank :) |
| 15:48.37 | *** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com) |
| 15:49.04 | wcselby | other than THE BOOK, what are some good asterisk books? I've got THE BOOK and also the Asterisk 1.4 the Professional Guide from PACKT publishing (by Colman Carpenter, David Duffett, Nik Middelton and Ian Plain). I'm looking to simply expand my knowledge on the subject...any suggestions? |
| 15:49.39 | mnicholson | leifmadsen, eh? |
| 15:50.29 | mnicholson | Moz, i have some experience using chan_mobile |
| 15:50.44 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
| 15:51.41 | mnicholson | leifmadsen, ccesario, that directed pickup bug was issue 15100 |
| 15:51.54 | leifmadsen | mnicholson: thanks |
| 15:52.40 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:53.45 | voipmonk | wcselby: get your hands dirty and dive in. |
| 15:53.56 | wcselby | voipmonk - believe I have. :) |
| 15:54.06 | ccesario | leifmadsen, thanks.... let me read |
| 15:54.10 | wcselby | always looking to learn new things is all. |
| 15:55.06 | [TK]D-Fender | wcselby: read all the docs in the tarball, "core show fuinctions" "core show applications", and you've got most of it |
| 15:55.25 | ccesario | mnicholson, thanks |
| 15:55.46 | ccesario | mnicholson, I can apply this patch in SVN-branch-1.6.2-r218364 ? |
| 15:56.36 | mnicholson | ccesario, that bug is closed. The patch has already been applied to all supported branches. |
| 15:57.05 | mnicholson | ccesario, i am not sure what you are trying to do, I don't know if it is related to that issue or not |
| 15:57.22 | bmoraca | what are the merits for using r or R instead of g or G when dialing out a zap group? we had an issue today where our provider had an issue on channel 1, and because we were using Zap/g0, no one could make outbound calls...changing to Zap/r0 would have masked the issue, so I wouldn't find it as quickly... |
| 15:57.43 | *** join/#asterisk Tim_Toady (n=moi@adsl297-103.kln.forthnet.gr) |
| 15:57.51 | *** join/#asterisk oej (n=olle@132.177.253.250) |
| 15:57.57 | ccesario | mnicholson, ooops... yes... alread applied |
| 15:58.03 | bmoraca | is there any compelling reason to use one over the other? |
| 15:58.08 | ccesario | i'm having the same problem... (I think) :P |
| 15:58.18 | ccesario | http://pastebin.com/m6b1c4a6a |
| 15:58.33 | Qwell | bmoraca: If your provider sends calls 1,2,3, you'll want to send backwards to avoid glare. Providers do it differently. |
| 15:59.00 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 15:59.01 | bmoraca | they round-robin from the highest order channel |
| 15:59.24 | Qwell | my answer is for r vs R, of course |
| 15:59.29 | Qwell | (or g vs G) |
| 15:59.30 | bmoraca | right |
| 15:59.45 | bmoraca | so, since they round robin, I should also be doing round robin? |
| 16:00.01 | Qwell | not necessarily |
| 16:00.02 | bmoraca | in the reverse order, of course |
| 16:00.33 | bmoraca | i'm just trying to figure out how to keep this issue from happening again, while still being able to be alerted that it happened |
| 16:01.12 | bmoraca | s/happening/being catastrophic/ |
| 16:07.43 | Naikrovek | omfg i'm going to quit this damn near perfect job because of one thing |
| 16:07.52 | Naikrovek | rational clearcase |
| 16:07.59 | Naikrovek | THE worst bit of software EVER |
| 16:08.05 | Naikrovek | grumbles |
| 16:08.09 | casnik | Swingline Stapler go missing again? |
| 16:08.11 | casnik | lol |
| 16:08.23 | Naikrovek | i believe you have my sanity? |
| 16:08.51 | Naikrovek | i'm going to go emo or some shit over this POS clearcase |
| 16:09.04 | casnik | what is that software for? |
| 16:09.36 | Naikrovek | it's IBMs flagship source control management software |
| 16:09.38 | Naikrovek | and i have to admin it |
| 16:09.43 | casnik | eyew |
| 16:09.45 | mutante | Qlikview is also pretty annoying |
| 16:09.53 | Naikrovek | cvs, svn, ..., clearcase |
| 16:10.23 | Naikrovek | mutante: it's certainly spelled annoyingly |
| 16:10.25 | casnik | whats wrong with just plain ol subversion |
| 16:10.47 | Naikrovek | well we do a LOT of work for caterpillar, so we have to integrate with their systems, and those guys use ClearCase |
| 16:10.53 | Naikrovek | so, we have to use clearcase |
| 16:10.59 | casnik | ah lame |
| 16:11.14 | casnik | old guys sitting in rooms drinking whiskey.... make bad calls |
| 16:11.14 | Naikrovek | it's like having a girlfriend to loves to go to the opera. Now you have to go to the opera |
| 16:11.37 | Naikrovek | you can NOT go to the opera, but the relationship will be severely damaged, if not destroyed |
| 16:12.22 | casnik | it's like when I did engineering drawing , you had to make them all in CAD then create raster images out of them for word docs |
| 16:12.41 | Naikrovek | ugh. |
| 16:12.47 | Naikrovek | well education maybe could have solved that |
| 16:12.58 | Naikrovek | education can't solve clearcase; if it could, clearcase would be dead |
| 16:13.19 | casnik | yeah , but the government contrats required it |
| 16:13.31 | Naikrovek | ah so same situation |
| 16:13.34 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 16:13.34 | casnik | your dealing with private .... so there is a chance for you lol |
| 16:14.07 | Naikrovek | these retards also store EVERY BUILD in clearcase. so they have /maybe/ 1G of source code, but fucking 750GB of files that have to be synchronized. |
| 16:14.26 | Naikrovek | and guess what? we're on a T-1 |
| 16:14.31 | casnik | hahaha |
| 16:14.39 | Naikrovek | i swear if I ever leave this job it'll be over clearcase |
| 16:14.40 | Qwell | What's wrong with storing builds in SCM? |
| 16:14.47 | casnik | 128k upload probably |
| 16:15.09 | Naikrovek | Qwell: nothing, if you didn't have one-time test version in there that have been there for years, and that aren't used in production |
| 16:15.25 | Naikrovek | Qwell: and the problem is that we're all on slow-as-hell links synching 750GB of stuff |
| 16:15.31 | Naikrovek | casnik: full t1 at least |
| 16:15.34 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip98-186-180-143.ks.ks.cox.net) |
| 16:15.55 | Qwell | Naikrovek: Don't do that then |
| 16:16.06 | *** join/#asterisk docelmo (n=chatzill@65.114.160.138) |
| 16:16.08 | Naikrovek | oh i wish it were that easy |
| 16:16.13 | casnik | still not that fast for that kind of work |
| 16:16.23 | Naikrovek | there are 80 some developers who don't understand english that I have to tell this to |
| 16:16.26 | casnik | it's like streaming pron with a 9600 baud modem |
| 16:16.33 | dwery | anyone has a contact at Thomson? I'd like to discuss a few bugs of their ST2030 |
| 16:17.36 | Naikrovek | i'm a developer. i have a career of software development behind me, but i have the added benefit of having a lot of networking experience, a lot of system admin experience, and all of that |
| 16:17.43 | Naikrovek | most developers have nothing but development experience |
| 16:17.52 | Qwell | most developers don't even have that |
| 16:18.04 | Naikrovek | they do not understand that when they put binaries in a source code management system that it causes fucking issues |
| 16:18.15 | Naikrovek | causes non-fucking issues too |
| 16:18.17 | Qwell | It doesn't cause issues. Your crappy network does. |
| 16:18.19 | *** join/#asterisk p3nguin (i=BuhPjX1J@asterisk-klv5.a2infotech.com) |
| 16:18.27 | Naikrovek | Qwell: you do not understand clearcase |
| 16:18.31 | casnik | <---- ISP Network engineering ... breaking into system admin/Development |
| 16:18.33 | Qwell | I understand SCM. |
| 16:18.42 | Naikrovek | Qwell: yes i do too, but clearcase is a POS |
| 16:18.56 | Qwell | You're doing it wrong. :) |
| 16:18.57 | Naikrovek | i've admined plenty of CVS and SVN systems with binaires, no problem |
| 16:19.00 | Naikrovek | clearcase chokes on them |
| 16:19.04 | Naikrovek | also |
| 16:19.05 | Naikrovek | get this |
| 16:19.08 | Qwell | Then don't use clearcase |
| 16:19.16 | Naikrovek | Qwell: no choice |
| 16:19.44 | Naikrovek | get this; if a sync fails between sites, clearcase can't detect it, does nothing to remedy it, and it requires manual intervention to fix |
| 16:19.57 | Qwell | So fix your broken network. :p |
| 16:19.58 | Naikrovek | and if you don't fix it FAST sync packets (not tcp/ip packets) pile up |
| 16:20.00 | coppice | clearcase handles binary files just like SVN. you need to tag them as binary |
| 16:20.17 | Qwell | coppice: well, svn does that automatically for binary files |
| 16:20.23 | Qwell | (by default, anyways) |
| 16:20.35 | coppice | no it doesn't. you must tag binary files with SVN |
| 16:20.47 | Qwell | It should do it on add |
| 16:20.59 | Naikrovek | so i have ONE developer in this office who requires clearcase, and I spend at least 3 hours a day fixing issues for the ONE developer |
| 16:21.02 | coppice | otherwise it tries to do end of line conversions on them |
| 16:21.18 | Naikrovek | caterpillar has come in, IBM has come in, there are no network or system issues. they're clearcase issues |
| 16:21.30 | Qwell | coppice: only if you've got autoprops set, and it matches (like a binary .c file) |
| 16:21.34 | Naikrovek | IBM: "yeah that's a bug, we're working on it." |
| 16:21.37 | Qwell | or, that's how it *should* do it |
| 16:21.42 | *** join/#asterisk RypPn_OuT (i=TuMbL@rosscom.demon.co.uk) |
| 16:22.42 | Chainsaw | Naikrovek: Or you have a failing switch that is causing intermittent packet loss under high load. |
| 16:23.01 | Chainsaw | Naikrovek: I've had that, and people have gone to the brink of insanity trying to find the intermittent transfer bug in Asterisk 1.2 (which isn't there). |
| 16:23.36 | Naikrovek | not a switch problem. network tests show no issues over week-long tests |
| 16:23.45 | Naikrovek | 0 missed packets |
| 16:23.48 | Chainsaw | TCP packets do not disappear on a local network. |
| 16:23.52 | Chainsaw | If they do, you have a problem |
| 16:24.04 | Naikrovek | show me a switch that has a problem that doesn't demonstrate the problem on a local network |
| 16:24.28 | Naikrovek | don't confuse a clearcase packet with a network packet. |
| 16:24.43 | Naikrovek | clearcase packets are lots of megabytes in size, and are basically files sent over the network |
| 16:24.53 | Naikrovek | tcp will take care of most, if not all, network issues. |
| 16:24.58 | Naikrovek | it's not that the packets aren't arriving, i should add |
| 16:25.04 | Naikrovek | its that suddenly they can't be applied |
| 16:25.18 | Chainsaw | casts a long stare at Naikrovek, turns around and walks off |
| 16:26.02 | Chainsaw | (If I was Fender I'd try and yell some sense into you, but I'm not) |
| 16:26.03 | Naikrovek | Chainsaw: no, don't chicken out, tell me what your issue is |
| 16:26.07 | Naikrovek | try |
| 16:26.51 | Naikrovek | i can virtually guarantee that it's not a network issue |
| 16:27.07 | Chainsaw | No, you can't. And you don't understand networking, which is why this is a futile conversation. |
| 16:27.58 | Naikrovek | yes, Chainsaw i can. ibm was in here to test our network, and our bladecenter, where the clearcase server lives |
| 16:28.05 | Naikrovek | they couldn't find anything |
| 16:28.26 | [TK]D-Fender | Looking for problems is more profitable than finding them :) |
| 16:28.33 | Naikrovek | besides, the connection goes straight from the router into the bladecenter |
| 16:29.12 | Naikrovek | i could understand, and i'd agree with you, if any of these issues were caused by a T1 that went down or something |
| 16:29.30 | Naikrovek | and yes, I do understand networking |
| 16:29.35 | *** join/#asterisk jasonwoot (n=some@69.73.89.233) |
| 16:29.55 | Qwell | You're doing this over the public internet? |
| 16:30.00 | Qwell | yeah, don't do that |
| 16:30.15 | Naikrovek | Qwell: that's the only way Caterpillar does it |
| 16:30.36 | Naikrovek | clearcase encrypts the packets before sending, and the receiving server decrypts them |
| 16:30.57 | casnik | how big are the packets? |
| 16:31.11 | casnik | > 1500? |
| 16:31.15 | Naikrovek | do not confuse clearcase packets with networking packets. tcp/ip packet size is 1500 |
| 16:31.32 | Naikrovek | ibm stupidly uses the same word for their sync packets, which can be very very large |
| 16:31.33 | Chainsaw | casnik: See? Lost cause right there. |
| 16:31.52 | Naikrovek | you fucker you tell me what you're not understanding |
| 16:31.55 | Naikrovek | mtu = 1500 |
| 16:32.04 | casnik | I am just simply trying to understand why the packets are so large and transverse the "internet" |
| 16:32.10 | Maliuta | Naikrovek: Caterpillar can't do most things right ... they once developed an excavator where the vibrations of the engine sheered the bolts holding the oil filter to the carbody. This cause the filters to fall off and the machine to go down |
| 16:32.26 | [TK]D-Fender | Naikrovek: Thats what packet fragmentation is for... |
| 16:32.28 | Chainsaw | casnik: Careful, he'll call you names too. |
| 16:32.41 | wcselby | i think Naikrovek is having a bad day.... |
| 16:32.52 | Naikrovek | wcselby: yes, because of clearcase |
| 16:32.55 | Naikrovek | Chainsaw: i can take anything |
| 16:33.08 | wcselby | Naikrovek - i get what you're saying. |
| 16:33.13 | casnik | we aren't trying to make Naikrovek look dumb right ? We are just trying to understand why Clearcase is so wicked |
| 16:33.44 | Naikrovek | casnik: the sync packets are large because that's how clearcase works. a clearcase packet is not a network packet, just as an html page is not a network packet |
| 16:33.49 | casnik | I've never used it personally , but I am a networking guy |
| 16:33.58 | Naikrovek | they're transmitted over tcp/ip, but they're whole files |
| 16:34.05 | wcselby | he already stated that IBM came in and identified a bug in their software (clearcase). why everyone is harping on him I don't understand |
| 16:34.08 | jasonwoot | ~clearcase |
| 16:34.36 | Naikrovek | jasonwoot: clearcase is ibm's source control software. like cvs, svn, git, visual source safe, perforce, etc |
| 16:35.08 | jasonwoot | ah, fucking thank you |
| 16:35.22 | Katty | eats chex for lunch. |
| 16:35.32 | Naikrovek | 60% less fat than potato chips |
| 16:35.40 | wcselby | who is chex? |
| 16:35.43 | wcselby | oh wait....sorry |
| 16:35.46 | Naikrovek | lol |
| 16:35.51 | casnik | win |
| 16:36.02 | Qwell | We called my friend chexmix... because most people couldn't pronounce his last name O.o |
| 16:36.03 | Katty | not amused. |
| 16:36.09 | Qwell | and it sounded like chexmix |
| 16:36.14 | wcselby | sorry katty - Naikrovek needed to laugh |
| 16:36.15 | Qwell | </random> |
| 16:36.20 | Naikrovek | wcselby: me too |
| 16:36.22 | casnik | indeed |
| 16:36.24 | *** join/#asterisk fofware (n=fofware@190.7.25.160) |
| 16:36.28 | Katty | http://farm4.static.flickr.com/3584/3357824071_29d0882929_o.jpg <- lunch. |
| 16:36.49 | Maliuta | waves at Katty |
| 16:36.53 | Katty | hi (= |
| 16:37.19 | *** join/#asterisk wopsy (n=80475@cap31-3-82-227-199-59.fbx.proxad.net) |
| 16:37.23 | Katty | they keep making the side of the boxes smaller. |
| 16:37.28 | Katty | and raising the prices. |
| 16:37.32 | fofware | Hello, I want to know if it's possible make a call from webpage to Asterisk-extension? |
| 16:37.35 | Naikrovek | Katty: so does everyone |
| 16:37.39 | Katty | but keep telling farmers they don't want to pay them more for corn, for fear the cost of food will go up. |
| 16:37.46 | Naikrovek | fofware: you'll need to install a sip phone on the page somehow |
| 16:37.58 | Katty | it irritates me. |
| 16:38.04 | Katty | also, have you seen shampoo and body wash? |
| 16:38.04 | Qwell | fofware: sure |
| 16:38.09 | wcselby | my little sister in law does her smiley faces backwards like katty just did....i never understood that |
| 16:38.14 | Katty | they're also slightly smaller, but the opening the stuff comes out with is LARGER |
| 16:38.17 | [TK]D-Fender | fofware: Go lookup Flash / Java softphones on the WIKI |
| 16:38.19 | fofware | Naikrovek: thanks is there someone free? |
| 16:38.20 | [TK]D-Fender | ~wikis |
| 16:38.21 | infobot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 16:38.23 | [TK]D-Fender | ^^^^^^^^^ |
| 16:38.24 | Katty | s/with/from/ |
| 16:38.32 | Naikrovek | fofware: listen to [TK]D-Fender |
| 16:38.49 | fofware | Naikrovek: thanks verry much |
| 16:38.50 | Katty | forcing you to be careful, less you use more of the product than you really intended. |
| 16:38.59 | Naikrovek | [TK]D-Fender: fofware thanks you via me |
| 16:39.08 | Naikrovek | Katty: you and i notice the same things |
| 16:39.22 | Naikrovek | Katty: my dad would have loved you - he want bananas when squeeze bottles came out |
| 16:39.33 | Katty | all my recipes that call for 16 oz packages. |
| 16:39.35 | fofware | [TK]D-Fender: thanks |
| 16:39.36 | Katty | they're not 14 oz. |
| 16:39.43 | Katty | and all my recipes that call for 8 oz pkgs, are now 6 |
| 16:39.48 | Naikrovek | he would scream that he coudln't get the last 10% out because it woudln't fall into the hole to get squirted out |
| 16:40.07 | *** join/#asterisk hfb (n=hfb@pool-98-112-210-252.lsanca.dsl-w.verizon.net) |
| 16:40.18 | Katty | that is irritating |
| 16:40.23 | Katty | i keep bottles upside down in the fridge. |
| 16:40.31 | Katty | and it's funny...cause it really shouldn't bother me, but it does. |
| 16:40.31 | Naikrovek | yeah |
| 16:40.35 | Katty | like spaghetti sauce, or alfredo |
| 16:40.37 | Katty | that irritates me. |
| 16:40.41 | Naikrovek | he invented all kinds of systems, centrifuges, etc |
| 16:40.59 | Katty | somehow throwing away a container that still has stuff in it is a sin. |
| 16:40.59 | Katty | idk |
| 16:41.00 | *** join/#asterisk bmoraca (n=chatzill@66.242.174.254) |
| 16:41.04 | bmoraca | damn power outages |
| 16:41.06 | Naikrovek | Katty: yes |
| 16:41.17 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 16:41.28 | Katty | not clearing the time off the microwave when you're done...also irritating. |
| 16:41.33 | Naikrovek | hehe |
| 16:41.37 | Naikrovek | that's pure woman right there |
| 16:41.38 | *** join/#asterisk andres833 (n=andres83@190.144.75.22) |
| 16:41.41 | Naikrovek | my wife does that |
| 16:41.43 | Katty | for some reaosn, i can't STAND that >.< |
| 16:41.51 | Naikrovek | complains that there's a 1 on the microwave instead of a 0 |
| 16:41.55 | Katty | ONE little button!!!! |
| 16:41.58 | Katty | STOP/CLEAR |
| 16:42.00 | Naikrovek | lol |
| 16:42.01 | Katty | i wish people would use it! |
| 16:42.05 | Qwell | Katty: I agree. it's annoying when the clock isn't flashing 12:00 |
| 16:42.16 | wcselby | lol Qwell |
| 16:42.20 | [TK]D-Fender | Katty: Worse still.. the idiots who open the door with < 3 seconds left and the settings still active |
| 16:42.21 | Naikrovek | it could just clear automatically if it's less than 5 seconds when you open the door.. |
| 16:42.21 | Katty | Qwell: clock flashing 12:00 is also irritating. |
| 16:42.26 | *** join/#asterisk ebroad (n=EB@72.11.213.195) |
| 16:42.27 | voipmonk | you know you like it when u look at the microwave at 9 am and it reads 1:27 |
| 16:42.31 | Katty | [TK]D-Fender: yes. that too. |
| 16:42.38 | voipmonk | :) |
| 16:42.41 | [TK]D-Fender | 's microwave only requires ONE button, and bypasses this nonsense |
| 16:42.49 | Naikrovek | "popcorn" |
| 16:42.53 | Katty | lol |
| 16:42.56 | Katty | i don't use that button |
| 16:42.56 | wcselby | my microwave doesn't flash 12:00 - it just doesn't display any time. looks like it's not even on |
| 16:43.07 | Katty | i use the 1 min and 2 min button |
| 16:43.13 | Katty | and occasionally the add 30 sec button :P |
| 16:43.25 | Qwell | There's more than the "Minute Plus" button on microwaves? |
| 16:43.26 | KavanS | if you have a flashing microwave...you are in technology, and a) don't care because you are that l33t, or b) don't know, because you are a fool |
| 16:43.27 | [TK]D-Fender | Naikrovek: Nope.... 30s button :) t he "GO" button |
| 16:43.41 | [TK]D-Fender | Naikrovek: Keep pressing for more.. because thats what more MEANS |
| 16:43.43 | KavanS | [TK]D-Fender, haha me too...30secs is the "go" button, just press it |
| 16:43.45 | KavanS | hahahaha |
| 16:43.46 | KavanS | yes. |
| 16:43.48 | Naikrovek | [TK]D-Fender: yeah that one works too |
| 16:43.49 | KavanS | I am the EXACT same |
| 16:43.56 | KavanS | I never use any other button than the 1 button |
| 16:43.58 | [TK]D-Fender | ONLY DIFFErenT! |
| 16:43.59 | KavanS | "add 30 more seconds..." |
| 16:44.12 | [TK]D-Fender | KavanS: Well.. I do use the STOP button to :0 |
| 16:44.15 | Katty | 30 seconds makes all the difference ;) |
| 16:44.24 | [TK]D-Fender | KavanS: but my ideal mirowave wouldn't need it |
| 16:44.30 | KavanS | naw I don't use stop, I just open door |
| 16:44.31 | Katty | also. |
| 16:44.34 | Katty | chrome undocking my windows. |
| 16:44.36 | Katty | also irritating. |
| 16:44.42 | KavanS | haha yes, I agree...the microwave in many aspects has become too complicated |
| 16:44.57 | KavanS | maybe when multi-touch interfaces become cheap, we can customize our own "UI" for the microwave |
| 16:45.08 | KavanS | "go" will be the button I put on there, big and green. |
| 16:45.20 | Katty | watches KavanS poke the go button 6 times. |
| 16:45.38 | Qwell | KavanS: why even have a button? just write a for loop.. if door = closed, start |
| 16:45.40 | Katty | why can't we have voice activation? |
| 16:45.50 | Qwell | Why would you put food in there, if you don't plan on cooking it right this second? |
| 16:45.54 | Katty | Early Gray Tea. Hot |
| 16:46.07 | KavanS | Qwell, ahh true...but then what happens when you need to close the door for appearance sake? |
| 16:46.12 | casnik | because if you told your microwave to "get maked" inadvertently that would be bad |
| 16:46.16 | Katty | KavanS: no weight |
| 16:46.22 | casnik | "get Naked"* |
| 16:46.23 | Katty | if weight + door closed = nukerwave |
| 16:46.38 | Katty | s/=/then/ |
| 16:46.39 | Qwell | also, this is slightly OT :p |
| 16:46.46 | *** join/#asterisk neurosys (n=vinix@173.9.159.182) |
| 16:46.51 | Katty | it won't be the end of the world. |
| 16:46.53 | Katty | yet |
| 16:47.03 | Chainsaw | Qwell: You should frame this, look back on it if you ever need to design a UI again. |
| 16:47.08 | Katty | it can't be. i've not stocked up on enough cans of chef boyardi |
| 16:47.50 | Katty | anyone been watchin Jerico? |
| 16:48.09 | Naikrovek | Katty: i didn't even watch Jeremiah and my name is Jeremiah |
| 16:48.12 | Chainsaw | Katty: That two-season show that got cancelled twice? |
| 16:48.26 | KavanS | heh I see "Katty" once in awhile from infobot or other people...I forgot she's in here....I ignored her sometime back when she was ranting about something that offended me |
| 16:48.27 | Katty | no idea. found it on netflix. |
| 16:48.40 | Katty | KavanS: yeah, sorry about that. |
| 16:48.42 | KavanS | I'm sure I've done the same though to irritate people with my stupid ?'s |
| 16:48.56 | Chainsaw | Katty: Protagonist Jake, the could-be-terrorist has a Toughbook? |
| 16:48.56 | Katty | KavanS: i occasionally snap when people irriate me :P |
| 16:49.21 | Katty | Chainsaw: yeah. that's the one. |
| 16:49.28 | Katty | Chainsaw: i'm only a few episodes in, but i'm really diggin it |
| 16:49.30 | Chainsaw | Katty: It's an awesome series. I liked it, so they cancelled it. |
| 16:49.34 | Chainsaw | Katty: I'm truly sorry. |
| 16:49.37 | Katty | bummer. |
| 16:50.06 | Katty | Chainsaw: kinda makes you wonder if that's what it looked like in some small towns around the Great Depression |
| 16:50.28 | KavanS | [TK]D-Fender, right on....glad to hear I use the M-wave the same way someone else does, I've had this mentioned to me "why do you only use the 30 sec button?" |
| 16:50.44 | *** join/#asterisk generalhan (n=asd@about/windows/staff/generalhan) |
| 16:50.45 | Chainsaw | Katty: Yeah, it does. Postapolyptic stuff really makes you think if it's done right. |
| 16:50.47 | Katty | and someone please tell KavanS to stop ignoring me ;) |
| 16:50.58 | Naikrovek | <Katty> and someone please tell KavanS to stop ignoring me ;) |
| 16:51.06 | KavanS | lol |
| 16:51.07 | Naikrovek | that should do it, Katty |
| 16:51.14 | Katty | excellent. lol. |
| 16:51.30 | KavanS | I wish more women were on irc |
| 16:51.34 | Naikrovek | no joke |
| 16:51.49 | jaytee | Jericho was awesome, wish they'd kept it on longer than 2 seasons and hadn't screwed up the second season with the cancellation crap |
| 16:52.07 | Corydon76-dig | KavanS: clearly, you just need to have a sex change. Then you get your wish. |
| 16:52.24 | Maliuta | I have female DNA |
| 16:52.31 | KavanS | Corydon76-dig, I think counseling people on a "sex change" is a bad career move for you, please stick to SSH ;) |
| 16:52.44 | KavanS | Corydon76-dig, that being said...for all you know, I am female |
| 16:52.52 | KavanS | god bless the internet :P |
| 16:53.11 | Corydon76-dig | KavanS: doesn't matter to me, either way. I'm bi. |
| 16:53.19 | Katty | hides extra cans of chef boyardi at jaytee's house. |
| 16:53.20 | Maliuta | I count as something like 1/3 female (that is one of the three DNA strands is female) |
| 16:53.32 | KavanS | Corydon76-dig, ahh |
| 16:53.33 | jaytee | :-) |
| 16:54.00 | Katty | i really need to take some emergency field courses. |
| 16:54.08 | Maliuta | Katty: cans? did you chop him up or something? |
| 16:54.14 | Katty | Maliuta: :P |
| 16:54.22 | Katty | Maliuta: it's better than plain canned tuna. |
| 16:54.29 | Katty | Maliuta: plus it has veggies :P |
| 16:54.33 | Maliuta | Katty: I have no idea what you're on about |
| 16:54.37 | Katty | oh |
| 16:54.39 | Katty | sec |
| 16:54.42 | Maliuta | Katty: not something we have here in .au |
| 16:55.09 | Katty | is getting a photo |
| 16:55.19 | Naikrovek | is getting a placebo. |
| 16:55.25 | Maliuta | not everyone live in north america [or wants to] |
| 16:55.30 | Katty | Maliuta: i hear ya |
| 16:55.34 | Naikrovek | i lived in sydney for 2 years |
| 16:55.38 | Katty | Maliuta: we're in pretty bad shape right now. |
| 16:55.47 | Maliuta | Naikrovek: get "Sleeping with Ghosts" |
| 16:55.48 | jaytee | Katty, have you seen the viral vid of the cat that sticks it's head under the faucet and drinks? |
| 16:55.52 | Katty | Maliuta: and the next few weeks will determine just how bad a shape we're gonna be in. |
| 16:56.00 | Maliuta | Naikrovek: sydney blow goats |
| 16:56.01 | Katty | jaytee: oh yeah. i watched that twice ;) |
| 16:56.10 | Naikrovek | Maliuta: oh yeah. I don't like you. |
| 16:56.19 | Naikrovek | i almost forgot |
| 16:56.32 | Katty | Maliuta: http://www.bestairmiledeals.com/wp-content/uploads/2007/06/44065_p06_ab-cmyk-73.jpg |
| 16:56.42 | casnik | Purest Awesome cat vid ever: -->http://i11.photobucket.com/albums/a159/seemill/e466f9149c6257f73baee0ebadfd60de.gif |
| 16:56.43 | Katty | Maliuta: they make a lot of different Things. |
| 16:56.52 | Katty | Maliuta: spaghetti and meatballs. |
| 16:56.57 | Maliuta | ahh |
| 16:57.04 | casnik | well , it's a gif not a vid , but still awesome |
| 16:57.11 | Katty | beef and pasta, in spaghetti sauce |
| 16:57.19 | Katty | generally the same stuff made differently |
| 16:57.23 | Katty | pasta, tomato, meat. |
| 16:57.38 | Maliuta | I don't do those things |
| 16:57.50 | Katty | you do if there's no food at the store. |
| 16:57.57 | Maliuta | I'm and amature chef ... I cook from scratch |
| 16:57.57 | Katty | and people are fighting over gas. |
| 16:58.28 | Katty | you don't cook when the grocery store is empty tho :P |
| 16:58.49 | Katty | Maliuta: http://42ndrecipestreet.blogspot.com/ <- i also cook. |
| 16:59.18 | Katty | tho most often i don't make up my own recipes. just alter other ones. |
| 17:00.03 | Maliuta | If I want to cook something I look at like 2 dozen recipes and then do my own that distils the dish to purity |
| 17:00.10 | jaytee | back before they had MRE's when I was in the Air Force they still had C-Rations. I once had a can of spaghetti and meatballs that was canned in 1953. This was in 1976. I was born in 1954. Kinda scary eating something that's been in a can since a year before you were born. |
| 17:00.10 | Maliuta | and tastes nicerer |
| 17:00.43 | Maliuta | jaytee is old |
| 17:00.49 | Katty | jaytee: ryan keeps a lot of MREs around....i don't think he ever got C-Rations |
| 17:00.59 | *** join/#asterisk Grof (n=krash@78.0.254.251) |
| 17:01.05 | Katty | jaytee: they stock pile the MREs up at the conservation department for disaster. |
| 17:01.21 | Katty | jaytee: never tried one tho. |
| 17:01.43 | jaytee | the C-Rations used to come with a little pack of 5 cigarettes and a small pack of Chicklets gum. |
| 17:01.54 | Katty | no chocolate? |
| 17:01.57 | Naikrovek | they're not as bad as you'd think |
| 17:01.57 | Katty | bummer. |
| 17:02.03 | Naikrovek | yes they have like a kitkat or something |
| 17:02.09 | Naikrovek | tiny bottle of tobasco |
| 17:02.14 | Katty | i meant in the crations |
| 17:02.23 | Naikrovek | yes |
| 17:02.27 | Katty | the MREs have a ton of food in them. |
| 17:02.40 | jaytee | sometimes. they had little round cans with 3 round "Nestle Crunch" style chocolate things |
| 17:02.41 | Naikrovek | MRE = C Ration I thought |
| 17:02.57 | Katty | jaytee: hmm, neat. |
| 17:03.16 | jaytee | MRE's are more like a freeze dried meal. C-Rations were canned food |
| 17:03.22 | Maliuta | MRE != C Ration |
| 17:03.29 | Naikrovek | ah ok, C-Rations went away in 1958 in the US |
| 17:03.42 | jaytee | Katty, you should have tried the 20 year old pound cake. it was delicious!!!! |
| 17:03.44 | Maliuta | don't MRE's do the pull the toggle and heat thing? |
| 17:03.49 | Naikrovek | Maliuta: some du |
| 17:03.50 | Katty | jaytee: lol!!!! |
| 17:03.51 | jaytee | Maliuta, yup |
| 17:03.53 | Naikrovek | s/du/do/ |
| 17:04.11 | dwery | you guys made me hungry....gotta go to the supermarket... |
| 17:04.14 | dwery | ;) |
| 17:04.39 | Katty | it's only a matter of time tho, in a state of emergency... |
| 17:04.53 | Katty | stores will get ransacked in a matter of hours. |
| 17:05.05 | Katty | and when the stores run out, then you've got riots |
| 17:05.16 | jaytee | ok, be back in a few. I have to go fix an Office 2007 issue (Damn you, Bill Gates!!!!) |
| 17:05.17 | Katty | and people breaking into other people's houses. |
| 17:05.22 | Katty | jaytee: cy |
| 17:05.24 | Katty | jaytee: cya |
| 17:05.25 | Naikrovek | Katty: read The Road if you've not already |
| 17:05.45 | Katty | McCarthy? |
| 17:05.48 | jaytee | The Road? On the Road? |
| 17:05.55 | Naikrovek | Katty: yes |
| 17:06.03 | Naikrovek | jaytee: see Katty |
| 17:06.17 | Naikrovek | most awesome book i've ever read |
| 17:06.31 | Katty | seems like i remember hearing about it |
| 17:06.33 | Naikrovek | when it's over if you don't want to stockpile food there's something wrong with you |
| 17:06.41 | Katty | some guy and his kid, treking across somewhere |
| 17:06.44 | Katty | and his mom had comitted suicide |
| 17:06.54 | Naikrovek | Katty: read it it's awesome and don't spoil it |
| 17:06.58 | Katty | k |
| 17:07.09 | Katty | makes note to check library |
| 17:09.22 | afink | reminds of the book One Second After about what would happen if an EMP bomb detonated above the us |
| 17:09.29 | afink | ^^ that was a good one too |
| 17:09.52 | Naikrovek | post-apocolyptic anything is usually good |
| 17:10.15 | Katty | what's up with google logo today |
| 17:10.28 | Katty | did we sign a contract with some aliens? |
| 17:10.40 | Naikrovek | they do that sometimes |
| 17:11.02 | Naikrovek | probably once a week or so they'll change the logo to celebrate a birthday or a death or a holiday |
| 17:11.11 | Naikrovek | sometimes it's once a week, sometimes it's less |
| 17:11.45 | Katty | hmmmm.k |
| 17:12.20 | Naikrovek | i don't usually see it because i search from the search bar of my browser |
| 17:13.51 | *** join/#asterisk garymc (n=garymc@host86-173-16-209.range86-173.btcentralplus.com) |
| 17:14.51 | garymc | Hi Had toe SIP office to remote working today. I get home and now its not working again. Heres my sip debug. Anyone take a look? http://pastebin.ca/1567251 |
| 17:15.57 | garymc | 202 is remote Zoiper and 201 is office |
| 17:16.57 | garymc | im wondering if i messed up my NAT settings again. But I was having an issue where the freepbx GUI was showing 2 phone connected when there was only 2? |
| 17:17.22 | [TK]D-Fender | garymc: Contact: <sip:Unknown@192.168.1.68> <--- your NAT settings are incorrect |
| 17:17.35 | garymc | right |
| 17:18.03 | garymc | so i must have messed them up somewhere when i was doing the SSH ports for other server. I might have pressed wrong buttons |
| 17:18.13 | garymc | :~( |
| 17:18.17 | garymc | god damn it |
| 17:18.25 | garymc | so its not my settings at home? |
| 17:18.26 | Naikrovek | you'll get it |
| 17:18.48 | *** join/#asterisk uski (n=uski@nor75-27-88-178-184-116.fbx.proxad.net) |
| 17:20.08 | uski | hi; i'm using Dial to call a mobile phone and I use M to allow the called person to accept or deny the call. I'm also using a timeout value to prevent the call from being transferred to the voicemail of the phone. The problem is that the timeout starts as soon as the call is initiated, and i'd like to timeout to start when the cellphone is actually ringing |
| 17:20.40 | uski | this corresponds to the message " -- SIP/freephonie-out-096b47d0 is ringing" and " -- SIP/freephonie-out-096b47d0 is making progress passing it to ..." in the console |
| 17:20.42 | uski | any idea? |
| 17:21.08 | uski | the problem is that the network takes a random amount of time between the time i Dial and the time the phone rings |
| 17:22.57 | Naikrovek | uski: don't know that you can react based on when the ringing starts |
| 17:23.03 | *** join/#asterisk IBC_jkenney (n=jkenney@65.44.169.66) |
| 17:23.11 | [TK]D-Fender | garymc: "doing ssh ports"? huh? |
| 17:23.18 | IBC_jkenney | Good afternoon i hope everyone is well |
| 17:23.52 | IBC_jkenney | i want to use presence in asterisk 1.6.4 the thing is i have it running out of mysql |
| 17:23.53 | [TK]D-Fender | uski: You can't actually know. |
| 17:24.09 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 17:24.18 | seanbright | 1.6.4 is out?! |
| 17:24.21 | seanbright | rushes to download |
| 17:24.24 | [TK]D-Fender | ibc1.6.4? Holy shit... |
| 17:24.37 | Qwell | 1.6.4 is ancient history |
| 17:24.48 | Naikrovek | i think he means 1.6.0.4 or 1.4 |
| 17:24.56 | kaldemar | garymc: the router is not the issue, asterisk's nat settings are. |
| 17:24.59 | [TK]D-Fender | loads his beta copy of res_fluxcapacitor.so |
| 17:25.31 | Naikrovek | [TK]D-Fender: grab me the linux 3.4 kernel while you're out, yes? |
| 17:25.46 | casnik | 4.8 is way better |
| 17:25.48 | casnik | has AI |
| 17:25.52 | Naikrovek | says you |
| 17:25.56 | garymc | where are asterisk nat settings |
| 17:26.03 | casnik | I've been there , I'm from the future |
| 17:26.08 | Naikrovek | oh |
| 17:26.25 | Naikrovek | give me a call and tell me to quit this job so i can abandon clearcase please |
| 17:26.30 | Naikrovek | or |
| 17:26.36 | Naikrovek | if it turns out good, tell me to stay |
| 17:26.43 | casnik | ok brb |
| 17:26.48 | casnik | ok back |
| 17:26.54 | casnik | number was disconnected |
| 17:27.00 | Naikrovek | ack |
| 17:27.01 | *** join/#asterisk jlnt (n=jlnt@70.255.193.190) |
| 17:27.04 | Naikrovek | guess that answers that |
| 17:27.09 | casnik | lol |
| 17:27.58 | [TK]D-Fender | casnik: Artificial Intelligence is no match for Natural Stupidity |
| 17:28.07 | Naikrovek | how about artificial stupidity |
| 17:28.22 | Naikrovek | one day computers will talk themselves into stupid activities just like humans do |
| 17:28.22 | jlnt | ^_^ |
| 17:28.32 | [TK]D-Fender | Naikrovek: Revlon already produces that shade :p |
| 17:28.32 | casnik | [TK]D-Fender, indeed |
| 17:28.37 | Naikrovek | hah |
| 17:28.45 | casnik | 4.8 has Natural Stupidity built in to |
| 17:29.10 | casnik | but you can only build both into the kernel |
| 17:29.33 | Naikrovek | heh. hopefully SelfAware++ isn't in that version yet |
| 17:30.09 | garymc | sureley 4.8 builds itself? |
| 17:30.14 | IBC_jkenney | yes i mean 1.6.0.4 |
| 17:30.16 | Naikrovek | my computer will hack the neighbor's pc knowing that the counterattack will be aimed at me |
| 17:30.27 | garymc | and doesnt need a server, it works in thin air? |
| 17:30.38 | Naikrovek | garymc: that's 6.0 |
| 17:30.45 | Naikrovek | k this has gotten silly |
| 17:31.49 | casnik | indeed , but good stuff for me to choke on my sammich with |
| 17:31.50 | Naikrovek | i love how this stupid internet connection gets completely killed when someone downloads something |
| 17:32.14 | voipmonk | hahah |
| 17:32.32 | *** part/#asterisk ebroad (n=EB@72.11.213.195) |
| 17:32.34 | voipmonk | someone is making money |
| 17:33.29 | Naikrovek | i need to download windows server iso from MS, and of course call quality hits the crapper. worst part is, upstream provider won't implement QoS, but I have, so everyone can always hear us perfectly but the other end always stutters as packets get dropped |
| 17:33.50 | voipmonk | fire your provider |
| 17:33.53 | voipmonk | find another |
| 17:33.54 | casnik | ^ |
| 17:33.59 | casnik | ^^ |
| 17:34.00 | voipmonk | not in that order |
| 17:34.01 | Naikrovek | yeah i'm looking at alternatives now |
| 17:34.22 | Naikrovek | comcast ethernet = $1800/mo, 20Mbps each direction |
| 17:34.29 | voipmonk | jeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeesus |
| 17:34.30 | casnik | or you could install a second router ...make like a internal ghetto QOS |
| 17:34.40 | casnik | holy crap |
| 17:34.41 | Naikrovek | comcast cable = $100/mo, 20Mbps down, 2 up |
| 17:34.44 | casnik | that's crack prices |
| 17:34.57 | Naikrovek | crack prices? high ? low ? |
| 17:35.01 | Naikrovek | i don't do crack |
| 17:35.06 | casnik | assuming crack is pricey |
| 17:35.06 | Naikrovek | :) |
| 17:35.14 | voipmonk | c'mon Naikrovek , its whats for dinner |
| 17:35.18 | Naikrovek | lol |
| 17:35.20 | casnik | and those pricing crack are sampling the product |
| 17:35.28 | garymc | Kaldemar : which asterisk nat settings? the RTP.conf or sip.conf? |
| 17:36.09 | kaldemar | ~sipnat |
| 17:36.10 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 17:36.32 | Naikrovek | casnik: i think what we'll wind up doing is leaving this T1 for voip since it's under contract. except it's $400/month for 1.5Mbps ... |
| 17:36.41 | kaldemar | garymc: rtp.conf has little to do with nat. |
| 17:36.57 | voipmonk | string together 3 or 4 cable internet connections |
| 17:36.58 | garymc | ok which NAT settings you mean? |
| 17:37.07 | casnik | Naikrovek, and the T-1 would be ideal for VOIP |
| 17:37.11 | Naikrovek | voipmonk: my manager said the same thing. i dunno if comcast will go for that |
| 17:37.18 | Qwell | garymc: Read what was linked to you. |
| 17:37.18 | casnik | Naikrovek, still very high price though |
| 17:37.21 | *** join/#asterisk joelsolanki (i=joelsola@124.125.148.111) |
| 17:37.24 | Naikrovek | casnik: yes but it's expensive to use just for phones |
| 17:37.29 | voipmonk | one for each floor |
| 17:37.36 | voipmonk | and then u bring htem all together |
| 17:37.42 | voipmonk | or one for each room |
| 17:37.44 | voipmonk | :) |
| 17:37.48 | voipmonk | different names |
| 17:37.53 | Naikrovek | voipmonk: well we do have two subsidiaries here, two company names and all |
| 17:37.58 | Naikrovek | may be possible |
| 17:38.00 | voipmonk | excellent |
| 17:38.09 | voipmonk | see if it makes sense on paper first |
| 17:38.12 | casnik | Naikrovek, if anything step it down a notch or find a cheaper alternate... that is just crazy expensive and isn't helping your business |
| 17:38.13 | [TK]D-Fender | [13:34]<casnik>that's crack prices <- I would never do a drug named after a part of my ass... |
| 17:38.20 | casnik | lol |
| 17:38.21 | Naikrovek | casnik: i agree totally |
| 17:38.35 | casnik | what local are you in? |
| 17:38.41 | casnik | locale* |
| 17:38.50 | Naikrovek | casnik: US. Illinois. Peoria |
| 17:39.03 | casnik | rural or urban area? |
| 17:39.07 | Naikrovek | urban |
| 17:39.18 | casnik | man I bet you can find way better then |
| 17:39.24 | Naikrovek | i don't know |
| 17:39.25 | Naikrovek | maybe |
| 17:39.33 | Naikrovek | once i get clearcase out of the way i'll have time to look |
| 17:39.48 | *** join/#asterisk fskrotzki (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 17:39.56 | casnik | are you in a leased workspace , like a floor of a bigger building or a have your own building |
| 17:40.01 | Naikrovek | but i have to do something about this. downsize to a partial T1 with G729, just for phones, comcast cable for everything else |
| 17:40.10 | Naikrovek | casnik: own buildilng. significant size as well |
| 17:40.31 | Naikrovek | warehouse with office |
| 17:40.42 | casnik | yeah , Comcast is relatively poor where I have been nefore , and there is some little guy out there just waiting to snipe their business away |
| 17:40.49 | voipmonk | go to fatasspipeinseconds.com |
| 17:40.53 | joelsolanki | Hi all. i want to compile asterisk with curl support. ./configure --with-curl=/usr/bin/curl is this corect way ? |
| 17:40.55 | voipmonk | oh wait... |
| 17:40.56 | Naikrovek | ooh that sounds promising |
| 17:40.58 | voipmonk | am I awake yet? |
| 17:41.00 | Naikrovek | lol |
| 17:41.12 | Naikrovek | joelsolanki: try it out |
| 17:41.12 | Qwell | joelsolanki: no, just install the curl development packages, and Asterisk will find it on its own with ./configure |
| 17:41.20 | Naikrovek | or listen to Qwell |
| 17:41.25 | voipmonk | one day we will buy bandwidth not cell plans |
| 17:41.45 | Naikrovek | one day bandwidth will be cheap like water or electricity |
| 17:41.54 | voipmonk | 800 megabit to your personal information device...... p.i.d sounds like a disease |
| 17:42.04 | joelsolanki | Qwell: i have already installed curl development packages then i should use just ./configure ?? will it compile with curl support? |
| 17:42.14 | Naikrovek | joelsolanki: yes. |
| 17:42.30 | joelsolanki | oh gr8. |
| 17:42.30 | Naikrovek | joelsolanki: after ./configure is done, scroll back and look to see if it was detected |
| 17:42.48 | joelsolanki | got it |
| 17:43.01 | IBC_jkenney | Naikrovek what part of the world are you in what state |
| 17:43.06 | Naikrovek | us, illinois |
| 17:43.08 | IBC_jkenney | if your in michigan |
| 17:43.08 | Naikrovek | peoria |
| 17:43.17 | IBC_jkenney | oh |
| 17:43.31 | IBC_jkenney | (has a lot of friends in michigan) (carriers) |
| 17:43.35 | Naikrovek | ah |
| 17:43.54 | Naikrovek | when i worked at Verio i knew a lot of people who could help but that was uh... a decade ago |
| 17:44.15 | Naikrovek | then they fired me because i got cancer, 2 days after a perfect annual review |
| 17:44.17 | Naikrovek | assholes |
| 17:44.18 | IBC_jkenney | I know a few clecs out here and also ISP's and datacenters |
| 17:44.23 | wonderworld | i am getting very often ISDN hangup Cause 16 "Notmal call clearing" back from my telco. the call in question is never connected to the other party. the number is actually in use and can be called with asterisk sometimes. ist that an asterisk issue or some problem at telco-side? |
| 17:44.30 | joelsolanki | also i wanted to ask. i am going to install asterisk 1.6.1 i want to have FAX support. T38 will asterisk 1.6.1 fully ? |
| 17:45.07 | Naikrovek | wonderworld: sounds like telco |
| 17:45.23 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 17:45.32 | Naikrovek | joelsolanki: maybe google for t.38 and asterisk. |
| 17:46.13 | joelsolanki | actually i searched and tested a lot. |
| 17:46.20 | joelsolanki | but i have not been sucessful yet :( |
| 17:46.30 | joelsolanki | so atlast i put my query here |
| 17:46.45 | Naikrovek | joelsolanki: ah |
| 17:46.58 | joelsolanki | i really looking for t38 support badly.:( |
| 17:47.01 | Naikrovek | joelsolanki: well i don't think people in here have had much luck with it, but, how many fax lines do you need |
| 17:47.19 | joelsolanki | well not much. maybe 5 to 10 maximum at moment |
| 17:47.44 | wonderworld | i tried to make a fax reliably work with asterisk last week. i gave up after a day and attached it to a small extra hardware pbx |
| 17:47.58 | IBC_jkenney | has anyone in here done asterisk monitoring with cacti |
| 17:48.02 | IBC_jkenney | and do you have any pointers |
| 17:48.04 | Naikrovek | joelsolanki: digium offers some software fax solution, i'm not sure if it uses t.38 or not but they will support it. 1 concurrent fax is free, see here: http://store.digium.com/products.php?category_id=94 |
| 17:48.22 | Naikrovek | IBC_jkenney: what are you looking to monitor |
| 17:48.42 | IBC_jkenney | queue's usage threads the usual |
| 17:48.50 | IBC_jkenney | on asteirsk itself |
| 17:48.52 | IBC_jkenney | not the machine |
| 17:49.12 | Naikrovek | IBC_jkenney: it should be fairly easy if asterisk can expose that stuff via SNMP. cacti does other methods too, but I can't seem to remember them right now |
| 17:49.32 | IBC_jkenney | Yes i thought there was a template |
| 17:49.33 | casnik | you can do the Zabbix thing |
| 17:49.38 | joelsolanki | i see. i will take a look at it. |
| 17:49.52 | Naikrovek | IBC_jkenney: you can write a little program that cacti can call whenever it wants the status, and in that program you can collect whatever you want |
| 17:50.56 | garymc | well im stumped, as ive done all this stuff, had it working great when i was at the office but now it doesnt work ? when i get home. Was talking to a guy from tunisa no problems earlier he was connected to my asterisk box with zoiper |
| 17:51.08 | IBC_jkenney | is looking into it :) |
| 17:51.29 | *** part/#asterisk joelsolanki (i=joelsola@124.125.148.111) |
| 17:52.03 | *** join/#asterisk kannan (n=kann@121.246.242.95) |
| 17:52.04 | Qwell | garymc: Please hire a consultant. You have been given links to fix your problems repeatedly, and you simply do not follow them. |
| 17:52.24 | garymc | Qwell : you sure im not following them |
| 17:52.34 | [TK]D-Fender | garymc: I've told you to your face |
| 17:52.37 | [TK]D-Fender | garymc: REPEATEDLY |
| 17:52.41 | kaldemar | garymc: there are many of us who are sure |
| 17:52.53 | [TK]D-Fender | garymc: your CONTACT IP was wrong. that = sip.conf |
| 17:54.04 | neurosys | Sorry, Google doesnt seem to be very helpful: what exactly is an rj26? |
| 17:54.48 | Katty | sways. |
| 17:55.01 | Naikrovek | neurosys: connector. never heard of the 26 variety |
| 17:55.02 | Qwell | # RJ26X: 50-pin miniature ribbon connector, for multiple data lines, universal |
| 17:55.09 | garymc | [TK]D-Fender : so my sip.conf is incorrect? |
| 17:55.14 | Qwell | garymc: Clearly |
| 17:55.15 | Naikrovek | garymc: sounds like it |
| 17:55.23 | neurosys | Naikrovek: Me neither. Thats why im confused :P |
| 17:55.37 | garymc | Ok freepbx builds my sip.conf should i go there and bother them instead? |
| 17:56.17 | Naikrovek | garymc: you should be asking the config questions in #freepbx if you're using freepbx |
| 17:56.47 | Katty | i need a good song. |
| 17:56.53 | Katty | my current music..is old. |
| 17:56.57 | Katty | and worn out |
| 17:57.01 | garymc | Naikrovec mainly same people in there as ar ein here |
| 17:57.10 | Naikrovek | i remember when music used to get worn out |
| 17:57.23 | Naikrovek | garymc: yes but there are freepbx people there who lurk |
| 17:57.24 | Katty | i might have to listen to lady gaga |
| 17:57.30 | Naikrovek | Katty: eek |
| 17:57.37 | garymc | pa pa pa poker face |
| 17:57.43 | Katty | love game, actually. |
| 17:57.49 | garymc | oh |
| 17:57.54 | garymc | just as bad |
| 17:57.59 | Katty | probably. |
| 17:58.05 | [TK]D-Fender | Lady Gag <- no need for the "a" at the end |
| 17:58.07 | *** join/#asterisk friartuck (n=pmccary@66.162.90.56) |
| 17:58.12 | Katty | [TK]D-Fender: *hee* |
| 17:58.13 | [TK]D-Fender | Good voice, shit material |
| 17:58.13 | Naikrovek | wait! "poker face" ? I though it was poke her face |
| 17:58.22 | Katty | >.< |
| 17:58.28 | Naikrovek | [TK]D-Fender: she's a songwriter first, performer second |
| 17:58.32 | [TK]D-Fender | Naikrovek: I was sure it was a tongue-in-cheek myself ;) |
| 17:58.40 | [TK]D-Fender | Naikrovek: Or some other appendage ;) |
| 17:58.48 | Naikrovek | i don't know |
| 17:59.07 | Naikrovek | i love women and never understood why someone would sing anything that could be misheard as "poke her face" |
| 17:59.38 | casnik | that woman is freaking nuts |
| 18:00.06 | Katty | probably for the same reason someone wrote honkey tonk badonkadonk |
| 18:00.16 | [TK]D-Fender | Naikrovek: I'm wondering if you seek Amy... |
| 18:00.25 | wonderworld | i am getting very often ISDN hangup Cause 16 "Normal call clearing" back from my telco using dahdi. the call in question is never connected to the other party. the number is actually in use and can be called with asterisk sometimes. ist that an asterisk issue or some problem at telco-side? |
| 18:01.52 | wonderworld | i google ISDN Cause 16 and wasn't able to find a lot of information on that error. as i understood, it basicly says, that the far end requested a hangup |
| 18:04.57 | *** join/#asterisk gabri-shatana (n=shatana@95.235.120.253) |
| 18:04.59 | gabri-shatana | hi |
| 18:05.37 | Naikrovek | hello again, gabri-shatana |
| 18:05.42 | gabri-shatana | lol |
| 18:05.53 | casnik | hide your lettuce |
| 18:05.58 | gabri-shatana | this is my home.. |
| 18:06.21 | gabri-shatana | so.. |
| 18:06.36 | gabri-shatana | i want make a submenu in extensions.conf |
| 18:07.35 | gabri-shatana | exten => s,2,Goto(extension, priority) |
| 18:08.25 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 18:08.32 | gabri-shatana | anyone? |
| 18:08.44 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 18:08.54 | Naikrovek | gabri-shatana: you'll have to create a second IVR and route from the first to the second, as I recall |
| 18:09.08 | gabri-shatana | IVR ? |
| 18:09.12 | Naikrovek | menu |
| 18:09.15 | gabri-shatana | mhh |
| 18:09.26 | gabri-shatana | like [submenu] |
| 18:09.28 | gabri-shatana | ? |
| 18:09.44 | Naikrovek | i think so... i use freepbx so i don't know, honestly |
| 18:09.54 | gabri-shatana | lol |
| 18:09.58 | Naikrovek | i haven't had time to rebuild our trixbox into raw asterisk yet |
| 18:10.05 | gabri-shatana | ok |
| 18:10.31 | gabri-shatana | i've found! |
| 18:10.32 | gabri-shatana | http://www.voip-info.org/wiki/view/Asterisk+tips+ivr+menu |
| 18:10.39 | gabri-shatana | <PROTECTED> |
| 18:10.41 | gabri-shatana | yeah |
| 18:10.43 | Naikrovek | there ya go |
| 18:11.14 | IBC_jkenney | hello |
| 18:11.16 | [TK]D-Fender | gabri-shatana: Goto <- |
| 18:11.28 | [TK]D-Fender | gabri-shatana: "you'll want to jump into another CONTEXT |
| 18:11.35 | gabri-shatana | yes |
| 18:11.39 | gabri-shatana | i know |
| 18:11.50 | gabri-shatana | a context is like [subenu] ? |
| 18:11.52 | Naikrovek | gabri-shatana isn't stupid, just bad with English. |
| 18:11.55 | Naikrovek | gabri-shatana: yes |
| 18:12.28 | gabri-shatana | http://pastebin.com/d180ece67 |
| 18:12.33 | gabri-shatana | where's the error? |
| 18:12.52 | *** join/#asterisk errotan (n=errotan@5403E46D.catv.pool.telekom.hu) |
| 18:13.03 | [TK]D-Fender | gabri-shatana: Not specifying the EXtensION |
| 18:13.14 | [TK]D-Fender | gabri-shatana: "core show application goto" |
| 18:13.17 | Naikrovek | Goto(Submenu1,1) -- there is no 1 extension |
| 18:13.29 | [TK]D-Fender | has shift-retension issues lately... |
| 18:13.34 | Naikrovek | yeah |
| 18:13.35 | Naikrovek | sorry |
| 18:13.39 | Naikrovek | i must be lagging |
| 18:13.43 | Naikrovek | i'm typing these as you are |
| 18:13.53 | casnik | lol |
| 18:13.55 | KyleK | omglag |
| 18:14.06 | *** join/#asterisk saisoma (n=irchon@166.137.5.152) |
| 18:14.08 | gabri-shatana | Submenu1 is the extensions and "1" is the prioprity |
| 18:14.12 | [TK]D-Fender | Naikrovek: beat me to one yesterday :) |
| 18:14.14 | gabri-shatana | *priority |
| 18:14.25 | [TK]D-Fender | gabri-shatana: no, submenu1 is a CONTEXT |
| 18:14.26 | Naikrovek | i wanna hear gilbert gottfried scream ACK LAG like he says AFLAC |
| 18:14.41 | gabri-shatana | sorry |
| 18:14.48 | Naikrovek | gabri-shatana: no probs |
| 18:15.02 | gabri-shatana | what i have to do to make it work? |
| 18:15.43 | Naikrovek | [TK]D-Fender: would this work for him/her: http://pastebin.com/m3e2f792e |
| 18:16.00 | [TK]D-Fender | gabri-shatana: http://pastebin.com/m676872d9 |
| 18:16.24 | Naikrovek | ah |
| 18:16.28 | [TK]D-Fender | Naikrovek: No |
| 18:16.38 | gabri-shatana | ok |
| 18:16.39 | Naikrovek | yeah i see what i did there |
| 18:16.40 | gabri-shatana | but |
| 18:16.47 | [TK]D-Fender | [14:13]<[TK]D-Fender>gabri-shatana: "core show application goto" <------- read the instructions |
| 18:17.02 | gabri-shatana | i want to have a password |
| 18:17.08 | gabri-shatana | "199419942" |
| 18:17.27 | Naikrovek | gabri-shatana: get it working without one first, then add password? |
| 18:17.29 | [TK]D-Fender | gabri-shatana: "core show application read" , "core show application authenticate" |
| 18:17.40 | Naikrovek | password is additional logic to add once you know you'can get there okay |
| 18:17.49 | gabri-shatana | yes |
| 18:18.58 | Naikrovek | gabri-shatana: why don't you just use different extensions for your functions, rather than different contexts |
| 18:20.13 | gabri-shatana | [TK]D-Fender, http://tinyurl.com/nptsu8 |
| 18:20.15 | gabri-shatana | -.-" |
| 18:20.20 | gabri-shatana | only the log |
| 18:20.44 | [TK]D-Fender | gabri-shatana: go type that in * CLI |
| 18:20.55 | [TK]D-Fender | gabri-shatana: And let me know when you've found a clue... |
| 18:21.06 | gabri-shatana | <PROTECTED> |
| 18:21.10 | casnik | is laughing so hard at the LMGTFY woosh |
| 18:21.23 | Naikrovek | gabri-shatana: asterisk -r |
| 18:21.26 | Naikrovek | gabri-shatana: run that |
| 18:21.28 | Naikrovek | then type |
| 18:21.31 | [TK]D-Fender | casnik: Laugh harder when you realize that the links quote ME <- |
| 18:21.38 | Naikrovek | core show application goto |
| 18:21.49 | casnik | [TK]D-Fender, didn't get that far but that is hilarious |
| 18:22.24 | [TK]D-Fender | casnik: http://www.google.com/search?btnG=1&pws=0&q=%22core+show+application+authenticate%22 |
| 18:22.26 | dustybin | if somebody leaves a voicemail, is there a way for my polycom to indicate it? |
| 18:22.26 | casnik | goes back to nubbery reading tfot |
| 18:22.46 | [TK]D-Fender | dustybin: "mailbox=" in your sip peer |
| 18:22.54 | dustybin | ace :-) |
| 18:23.26 | Naikrovek | dustybin: mine have a red light that is crazy bright when i have a voicemail |
| 18:23.29 | casnik | [TK]D-Fender, lol at the irc logging |
| 18:23.55 | Naikrovek | lol |
| 18:24.11 | Naikrovek | that's what half the people in here are, i bet. loggers |
| 18:24.29 | [TK]D-Fender | Naikrovek: No, most are idlers or trolls :) |
| 18:24.47 | Naikrovek | reading reading reading but never asking |
| 18:24.55 | casnik | I'll be honest , I am a complete noob that just picked up Asterisk for the first time last week. |
| 18:25.10 | casnik | reading a book and trying to get a voip system up on a BSD server |
| 18:25.11 | Naikrovek | i've been using it for a couple months, but i'm still very much a newbie |
| 18:25.33 | casnik | only reason I am here is to look and read other peoples issues to try and learn |
| 18:25.43 | [TK]D-Fender | casnik: Yes, but n00b != idiot. Not mutually inclusive. Some people just singed up for the package deal ;) |
| 18:25.46 | Naikrovek | casnik: if you want a quick solution try asterisknow, you'll have plenty of things to fiddle with there, but you'll get started |
| 18:26.09 | Naikrovek | then you can do what i want to do, rebuild from scratch, vanilla asterisk, but while keeping all the neat features of freepbx |
| 18:26.12 | [TK]D-Fender | casnik: GUI's are where you end up when you don't have a choice and you know what you are doing. |
| 18:26.20 | casnik | [TK]D-Fender, agreed ... I'm better than a noob then I guess |
| 18:26.41 | [TK]D-Fender | Naikrovek: And its remarkably hard to keep anything FreePBX does while "leaving it" |
| 18:26.59 | Naikrovek | no i mean keep all the features of freepbx but recreating them via vanilla asterisk |
| 18:27.02 | casnik | I thought about trying the ast-now thing , but I was gonna just go for the full learn the hard way method lol |
| 18:27.49 | [TK]D-Fender | casnik: Good, keep on that way |
| 18:28.07 | casnik | Read the book , read the wiki , try to fudge a in house Asterisk box that can connect a few phones (no POTS or PBX) just internal softphone to polycom or another softphone |
| 18:28.17 | Naikrovek | casnik: i inherited this system so it's cool, i'd be lost if i'd started at vanilla asterisk and my boss said "I want to be able to turn on call forwarding whenever i want, to whenever i want, today" |
| 18:28.42 | dustybin | this polycom KICKS ASS |
| 18:28.45 | Naikrovek | yes |
| 18:28.47 | Naikrovek | yes it does |
| 18:28.51 | Naikrovek | loves his polycoms |
| 18:28.53 | casnik | Naikrovek, yeah I don't have anything besides a setup being done in the Dominican that I can log into |
| 18:28.55 | dustybin | quality |
| 18:29.03 | [TK]D-Fender | Naikrovek: Yes, there goes a dozen lines of dialplan... |
| 18:29.09 | dustybin | i have 1 new message !! |
| 18:29.10 | casnik | Naikrovek, BUT that installation / Implementation is freakishly advanced |
| 18:29.14 | dustybin | checks message |
| 18:29.17 | Naikrovek | [TK]D-Fender: i know, but i'm still new at this stuff |
| 18:29.22 | dustybin | its me :( |
| 18:29.27 | Naikrovek | dustybin: lol |
| 18:29.42 | casnik | like I went "pkg-add -r asterisk" |
| 18:29.45 | casnik | and there is where I am |
| 18:29.54 | Naikrovek | fair enough |
| 18:29.58 | Naikrovek | love freebsd |
| 18:30.05 | casnik | noob at that to lol |
| 18:30.09 | Naikrovek | *bsd > linux |
| 18:30.16 | casnik | I've been a Debian / Gentoo person for 4 years |
| 18:30.29 | Naikrovek | ubuntu > other linuxs i've tried |
| 18:30.44 | Naikrovek | except debian maybe |
| 18:30.49 | casnik | I can't agree for the reason that Ubuntu is TOO easy |
| 18:31.15 | casnik | It takes out some control and tries to add security and ease of use |
| 18:31.57 | *** join/#asterisk dlynes (n=daniel@bas5-hamilton14-1242444747.dsl.bell.ca) |
| 18:32.12 | casnik | if I could just register 2 users on my system and make calls from one to the other right now i'd be happy |
| 18:32.23 | dustybin | the look and feel, and sound quality of this polycom is excellent |
| 18:32.41 | dustybin | thanks for your advise people :D |
| 18:35.44 | *** part/#asterisk ilteris (n=ozzz@ip67-155-145-199.z145-155-67.customer.algx.net) |
| 18:36.32 | *** join/#asterisk bluOxigen (n=xainix@static-host119-73-70-69.link.net.pk) |
| 18:36.40 | bluOxigen | hi |
| 18:37.24 | gabri-shatana | exten => s,2,Goto(Submenu1,1) |
| 18:37.24 | gabri-shatana | ;Submenu1 |
| 18:37.24 | gabri-shatana | exten => 1,1,Authenticate(1234321) |
| 18:37.41 | gabri-shatana | [TK]D-Fender, ? |
| 18:38.36 | [TK]D-Fender | gabri-shatana: Go read the INSTRUCTIONS and go test it |
| 18:38.48 | gabri-shatana | i've read but |
| 18:38.55 | bmoraca | anyone had any luck with SIP passing through an AT&T 2wire? |
| 18:38.59 | gabri-shatana | tell me if this is correct: |
| 18:39.11 | IBC_jkenney | is anyone in here using cacti to monitor asterisk |
| 18:39.19 | bluOxigen | i have deployed asterisk at my client - over all IVR functionality is fine, but I am facing problem in queus, i have three queues CustomerService, Inbound and Outbound, the call seems to transfer in the queus as required, but when it is disconnected either by the PSTN caller or dropped by the agent from the softphone, the queu got stuck and keeps showing the that particular extention as busy |
| 18:39.26 | gabri-shatana | Authenticate(1234321,Goto(Submenu2,1)) |
| 18:39.29 | bluOxigen | for next 10 min |
| 18:39.49 | bluOxigen | what could be the reason |
| 18:39.54 | [TK]D-Fender | gabri-shatana: No, that last one is clearly not correct. |
| 18:40.04 | [TK]D-Fender | gabri-shatana: and yo don't seem to be reading the instructions. |
| 18:40.05 | gabri-shatana | mhh |
| 18:40.11 | gabri-shatana | i'm tryng |
| 18:40.20 | gabri-shatana | but isn't simply |
| 18:40.46 | gabri-shatana | *it isn't |
| 18:41.55 | bluOxigen | ?? |
| 18:43.39 | *** join/#asterisk user4545 (n=sipip@p57B1F101.dip.t-dialin.net) |
| 18:43.55 | user4545 | Hi |
| 18:44.04 | wcselby | bmoraca - I've used a softphone over an AT&T 2wire |
| 18:44.06 | *** join/#asterisk Olobola (i=Olobola@240.sub-75-209-40.myvzw.com) |
| 18:44.08 | bluOxigen | any one plz help |
| 18:44.37 | user4545 | I have one problem with asterisk, "sip show registry" is empty why? |
| 18:44.52 | bmoraca | wcselby: yeah, one phone works fine...but two phones do not |
| 18:45.02 | kaldemar | user4545: why do you expect it not to be? |
| 18:45.17 | wcselby | haven't tried two phones...I can try tonight when I get home and let you know tomorrow.... |
| 18:45.27 | gabri-shatana | - Executing [s@eutelia:1] Answer("SIP/83.211.2.132-0a1d1d38", "") in new stack |
| 18:45.27 | gabri-shatana | <PROTECTED> |
| 18:45.27 | gabri-shatana | <PROTECTED> |
| 18:45.27 | user4545 | i can't bekome any incomming calls |
| 18:45.41 | dustybin | is there a way to upload a numbers list to a polycom phone? would this be the advantage of using ftp? |
| 18:46.00 | gabri-shatana | [TK]D-Fender, ? |
| 18:46.23 | wcselby | dustybin - you can use the MAC-directory.xml file to do that, but it's got to be perfectly formatted |
| 18:46.30 | [TK]D-Fender | gabri-shatana: Sorry, do you have an actual QuEStION to ask? |
| 18:46.31 | dustybin | aye ok |
| 18:46.42 | gabri-shatana | i've posted the error |
| 18:46.57 | [TK]D-Fender | gabri-shatana: You didn't tell * to WAITEXTEN |
| 18:47.02 | *** join/#asterisk [8none1] (n=[8none1]@cerberus.franklinamerican.com) |
| 18:47.03 | wcselby | or you can set the numbers up in the calls directory on your phone, and then the next time it talks to a configured ftp server it will upload the file for you, properly formatted. |
| 18:47.06 | [TK]D-Fender | gabri-shatana: and the WIKI samples are 1.0 ancient crap |
| 18:47.16 | gabri-shatana | lol |
| 18:48.03 | [TK]D-Fender | [14:45]<gabri-shatana> == Auto fallthrough, channel 'SIP/83.211.2.132-0a1d1d38' status is 'UNKNOWN' <--- ran out of things to do so the call ends |
| 18:48.21 | gabri-shatana | ok.. |
| 18:50.18 | *** join/#asterisk dandate2 (n=gtejkgjk@112.202.95.21) |
| 18:50.30 | dandate2 | will having ringall and autofill set destroy a queue |
| 18:51.09 | [TK]D-Fender | dandre: they are conflicting options. Setting them both to yes is just stupid |
| 18:51.28 | [TK]D-Fender | dandre: So... how'd that work out for you? ;) |
| 18:52.00 | *** join/#asterisk blkry (n=chatzill@64.147.222.130) |
| 18:54.26 | [TK]D-Fender | dandate2: rather... |
| 18:54.51 | dandate2 | i been afraid to try |
| 18:54.54 | [TK]D-Fender | hates it when even 4 chars doesn't net himt he autocomplete he wants.. |
| 18:55.03 | [TK]D-Fender | dandate2: well DON'T. |
| 18:55.17 | dandate2 | i was wondering if mabye the ring all would confuse asterisk in a positive way in that multiple people on hold will ring to seperate agents simultaneously |
| 18:55.44 | dandate2 | instead of waiting for each to get answered, i'm in a real pickle because my agents do not use DND so if i use round robin it wastes time ringing them |
| 18:55.50 | [TK]D-Fender | dandate2: everyton getting multiple calls at one... |
| 18:56.08 | [TK]D-Fender | dandate2: gic your agents |
| 18:56.12 | [TK]D-Fender | fix* |
| 18:56.30 | dandate2 | in ringall i think that the person who has been waiting on hold the longest rings to everyone , and when that is answered then the next ring cycle is only the next person on hold |
| 18:56.58 | dandate2 | haha fix the agents, they are across the world theres no getting through to them |
| 18:57.04 | dandate2 | i don't allow them to take credit cards anymore the QA is so bad |
| 18:57.05 | wcselby | dandate2 - unless you've got announcements, which can screw with the orders of people waiting in your queues |
| 18:57.27 | dandate2 | oh no just using the periodic you are next in line |
| 19:03.34 | bmoraca | dandate2: you can't fix business policies by trying to rewrite the applications. your policies should conform to the limits of your apps, not the other way around. if employees don't want to follow the business policies, then you should get new employees |
| 19:05.13 | *** join/#asterisk rossand (n=rossand@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 19:05.33 | *** join/#asterisk xbmodder_ (n=Sargun@atarack/Staff/Sargun) |
| 19:07.31 | *** join/#asterisk j_kroon (n=jkroon@dsl-240-185-181.telkomadsl.co.za) |
| 19:07.33 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 19:07.50 | dandate2 | is there any way to tell from the cli whose not doing it? |
| 19:07.55 | dandate2 | 'setting dnd' |
| 19:10.42 | [TK]D-Fender | dandate2: Go look at the CLI output. |
| 19:14.53 | dandate2 | i stare at it all my life lol |
| 19:16.13 | bmoraca | your queue stats should show which agents are and are not answering the calls |
| 19:16.22 | gabri-shatana | exten => 1,2 Goto(eutelia,2,1) |
| 19:16.22 | gabri-shatana | ;Menu |
| 19:16.22 | gabri-shatana | exten => 2,1,Background(sai-welcome) |
| 19:16.32 | gabri-shatana | it's correct? |
| 19:16.55 | *** join/#asterisk bluOxigen (n=xainix@static-host119-73-70-30.link.net.pk) |
| 19:17.28 | wcselby | no one has told gabri-shatana about pastebin yet? |
| 19:17.29 | dandate2 | queue stats where do i find that |
| 19:17.30 | wcselby | ~pb |
| 19:17.30 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 19:17.42 | bmoraca | your queue log file... |
| 19:17.52 | gabri-shatana | <PROTECTED> |
| 19:18.01 | gabri-shatana | i've posted 3 lines |
| 19:18.39 | bmoraca | yes, but what you've pasted isn't enough to help you with anything...need more 'context' |
| 19:18.43 | gabri-shatana | ok |
| 19:18.58 | gabri-shatana | http://pastebin.com/d25d4fd8b |
| 19:19.07 | *** join/#asterisk wcselby (n=wcselby@216.110.88.254) |
| 19:19.24 | gabri-shatana | http://pastebin.com/d25d4fd8b |
| 19:19.33 | wcselby | wow, somehow fat fingered a "close" window hot-key combo |
| 19:19.35 | bmoraca | and does it work? |
| 19:20.04 | wcselby | what I was going to say was you were taking away useful info in order to get under three lines |
| 19:20.09 | wcselby | which wasn't helping us help you |
| 19:20.16 | gabri-shatana | i've posted the extensions.conf |
| 19:20.18 | gabri-shatana | http://pastebin.com/d25d4fd8b |
| 19:20.52 | wcselby | I mean, what you've posted will technically work, but you've used far too much logic to do it |
| 19:21.09 | wcselby | you could put all of that into the 's' extension without any of the GoTo's |
| 19:21.26 | gabri-shatana | ok |
| 19:21.53 | *** join/#asterisk propellerhead (n=yogurt2u@host26.190-137-6.telecom.net.ar) |
| 19:22.01 | wcselby | http://pastebin.com/d55a45221 |
| 19:22.06 | wcselby | gabri-shatana ^^ |
| 19:22.55 | bluOxigen | my queues are getting stuck - |
| 19:23.04 | bluOxigen | after receiving a call for at least 10 min |
| 19:23.28 | bluOxigen | and the agent can not receive any call during that time |
| 19:23.29 | *** join/#asterisk The_Boy_Wonder (n=davidvos@asterisk/batman-developer/dvossel) |
| 19:23.29 | bmoraca | wcselby: should be more obfuscated than that, imo :P |
| 19:24.04 | gabri-shatana | where are located the file |
| 19:24.13 | gabri-shatana | for Background() ? |
| 19:24.37 | dandate2 | if an extention has call waiting disabeld but the queue is set to ring busy agents will they get stacked with ringing while on a call? |
| 19:25.30 | bmoraca | gabri-shatana: /var/lib/asterisk/sounds |
| 19:26.02 | bmoraca | dandate2: are you using freepbx? |
| 19:26.13 | gabri-shatana | bmoraca, mhhh |
| 19:26.14 | *** join/#asterisk retentiveboy (n=pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
| 19:26.26 | gabri-shatana | there is a directory wich is empty |
| 19:27.51 | gabri-shatana | ... |
| 19:28.58 | [TK]D-Fender | bmoraca: Yes |
| 19:29.20 | gabri-shatana | [TK]D-Fender, where are located the sounds file? |
| 19:29.49 | [TK]D-Fender | gabri-shatana: under the lib folder section listed in asterisk.conf |
| 19:29.57 | gabri-shatana | ty |
| 19:30.14 | wcselby | gabri-shatana - /var/lib/asterisk/sounds |
| 19:30.32 | gabri-shatana | empty |
| 19:30.59 | wcselby | did you load any sound files there? |
| 19:31.10 | gabri-shatana | /usr/share/asterisk/sounds |
| 19:31.12 | wcselby | check in asterisk.conf |
| 19:31.38 | gabri-shatana | ivve found it |
| 19:31.40 | gabri-shatana | *i've |
| 19:32.53 | gabri-shatana | how can i play those? |
| 19:34.15 | [TK]D-Fender | gabri-shatana: With an audio player. |
| 19:34.20 | gabri-shatana | mhh |
| 19:34.37 | gabri-shatana | Totem didn't work |
| 19:34.46 | [TK]D-Fender | VLC <- |
| 19:35.10 | wcselby | gabri-shatana - download the file to your computer, then play it there to test it? |
| 19:35.19 | bmoraca | Playback()? |
| 19:35.54 | [TK]D-Fender | bmOh sure, try the SMRT thing... |
| 19:36.09 | Naikrovek | ooh- 48port poe switch $330 |
| 19:36.09 | gabri-shatana | wcselby, nothing |
| 19:36.14 | gabri-shatana | [TK]D-Fender, nothing |
| 19:36.21 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
| 19:36.22 | bmoraca | Naikrovek: what brand? |
| 19:36.27 | [TK]D-Fender | gabri-shatana: Works for me... |
| 19:36.30 | Naikrovek | bmoraca: cisco |
| 19:36.37 | bmoraca | Naikrovek: 3550? |
| 19:36.39 | wcselby | Naikrovek - link? |
| 19:36.40 | gabri-shatana | .gsm file? |
| 19:36.40 | Naikrovek | bmoraca: cisco for small business |
| 19:36.50 | bmoraca | Naikrovek: oh, you mean Linksys then |
| 19:36.50 | Naikrovek | http://www.ipphone-warehouse.com/Cisco-SLM224P-Smart-Switch-48-ports-SLM224P-p/cisco-slm224p.htm |
| 19:36.55 | Naikrovek | bmoraca: yes |
| 19:37.25 | gabri-shatana | [TK]D-Fender, what kind of audio can i put there? |
| 19:37.27 | gabri-shatana | mpr? |
| 19:37.29 | gabri-shatana | mp3? |
| 19:37.31 | gabri-shatana | wav? |
| 19:37.33 | bmoraca | Naikrovek: watch your PoE budget on those...it's usually about 1/4 what you'd need to power all ports...and remember that they won't work with pre-standard Cisco gear. |
| 19:37.46 | [TK]D-Fender | gabri-shatana: Go read up on the formats * can support. |
| 19:37.50 | Naikrovek | bmoraca: yeah i don't have any pre-standard stuff |
| 19:37.56 | [TK]D-Fender | gabri-shatana: WIKi is good for that at least.. |
| 19:38.03 | Naikrovek | who runs that wiki |
| 19:38.13 | Naikrovek | can regular people even edit it? |
| 19:38.14 | [TK]D-Fender | ~wikis |
| 19:38.15 | infobot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 19:38.18 | [TK]D-Fender | Naikrovek: yes |
| 19:38.20 | bmoraca | commpartners...worst wholesaler ever |
| 19:38.31 | Naikrovek | worst looking wiki ever |
| 19:38.31 | wcselby | Naikrovek - I've modifed a few pages before |
| 19:38.32 | Naikrovek | as well |
| 19:38.36 | Naikrovek | wcselby: ah |
| 19:39.05 | *** join/#asterisk wonderworld (n=w@ip-62-143-22-226.unitymediagroup.de) |
| 19:39.28 | *** join/#asterisk feder (n=feder@host.190.15.192.172.static.itcsa.net) |
| 19:40.23 | bmoraca | Naikrovek: that site's fubared it's description...the SLM224P is the partnumber for the 24-port model, not the 48 port model |
| 19:40.31 | Naikrovek | bmoraca: yes i just noticed that as well |
| 19:40.56 | Naikrovek | they got the model number wrong throughout |
| 19:41.02 | Naikrovek | but they describe the 48 port version |
| 19:41.32 | bmoraca | doesn't say the PoE budget either |
| 19:41.37 | Naikrovek | nope |
| 19:41.46 | bmoraca | cisco's data sheet doesn't even say it |
| 19:41.56 | bmoraca | i wouldn't trust that switch as far as i could punt it without a foot |
| 19:42.38 | Naikrovek | really |
| 19:42.38 | Naikrovek | ah poe only on 24 ports as well |
| 19:43.12 | Naikrovek | bmoraca: "enough to support 7.5W for 24 PoE |
| 19:43.12 | Naikrovek | ports or 15.4W on 11 ports" |
| 19:43.20 | Naikrovek | ooh nasty hidden newline |
| 19:45.09 | *** join/#asterisk louben (n=lou@212-70-216-131.ath.static.tee.gr) |
| 19:48.40 | *** join/#asterisk s14ck (n=jtorres@ccscliente156.ifxnetworks.net.ve) |
| 19:49.08 | bluOxigen | is this asterisk support channel ???????? |
| 19:49.10 | *** join/#asterisk hakr (n=Bryan@pdpc/supporter/active/hakr) |
| 19:49.50 | wcselby | bluOxigen - support is sometimes provided here free of charge when people have time and knowledge of your particular issue |
| 19:49.56 | wcselby | what exactly is your particular issue? |
| 19:50.36 | bluOxigen | thx for reply - well i have issues with queus |
| 19:50.55 | bluOxigen | basically i have three queues customer service, inbound and outbound |
| 19:51.13 | bmoraca | Naikrovek: that's not much... |
| 19:51.36 | Naikrovek | bmoraca: enough for my polycom phones |
| 19:51.45 | bluOxigen | when an agent lets say 5100 recevies the call |
| 19:51.45 | bmoraca | Naikrovek: how many do you have? |
| 19:51.46 | Naikrovek | bmoraca: as long as they don't all turn on at once heh |
| 19:51.59 | Naikrovek | bmoraca: 12 hooked into that server room |
| 19:52.04 | Naikrovek | 4500w each |
| 19:52.09 | bluOxigen | the queu become busy and is only released after 10 minutes |
| 19:52.10 | Naikrovek | uh |
| 19:52.14 | Naikrovek | 4.5W each |
| 19:52.32 | bmoraca | Naikrovek: also, take into account attenuation in the length of the cabling. 550s are 6 watts over a 3ft cable...could be 50% higher over a 300ft cable |
| 19:52.33 | bluOxigen | wcselby: u with me ???? |
| 19:52.39 | wcselby | bluOxigen - you need to post relevant information to a pastebin |
| 19:52.59 | Naikrovek | bmoraca: yes. these are all relatively close |
| 19:53.15 | bluOxigen | wcselby: hmmm ok |
| 19:53.17 | bmoraca | might be ok, then...330s are relatively lightweight |
| 19:53.39 | Naikrovek | bmoraca: within 50ft. however, i have some that are MUCH further; they'd get their own local PoE switch probably |
| 19:55.55 | bmoraca | Naikrovek: used 3550s on ebay are about that...and probably a hell of a lot more stable :p |
| 19:57.16 | gabri-shatana | mhh |
| 19:57.18 | gabri-shatana | i have a problem |
| 19:57.54 | gabri-shatana | http://pastebin.com/d2cc6143c |
| 19:58.01 | gabri-shatana | i have no time to press 1 or 2 |
| 19:58.44 | bmoraca | check your timeouts, etc |
| 19:58.53 | gabri-shatana | where i set the timeout? |
| 19:59.00 | bmoraca | in the dialplan |
| 19:59.07 | bmoraca | examples are in the wiki |
| 19:59.08 | gabri-shatana | ,ResponseTimeout,10? |
| 19:59.09 | wcselby | you need a WaitExten(xx) at the s,4 |
| 19:59.15 | bmoraca | and probably the book, too |
| 19:59.26 | gabri-shatana | wcselby, ok |
| 20:01.08 | gabri-shatana | yeah |
| 20:01.10 | gabri-shatana | now it work |
| 20:01.15 | gabri-shatana | tanks you |
| 20:05.24 | *** join/#asterisk brezular (n=brezular@adsl-dyn245.78-99-16.t-com.sk) |
| 20:06.09 | *** join/#asterisk nny (n=scott@64.203.237.47) |
| 20:09.26 | *** join/#asterisk voipmonk (n=voipmonk@dsl-67-212-15-216.acanac.net) |
| 20:10.19 | voipmonk | anyone have a server for sale in toronto? |
| 20:10.20 | *** join/#asterisk Carlos_PHX (n=carlos@68.108.193.174) |
| 20:10.29 | *** part/#asterisk Carlos_PHX (n=carlos@68.108.193.174) |
| 20:14.35 | *** part/#asterisk asterwiki (n=asterwik@69.77.169.14) |
| 20:14.47 | wcselby | i love how that guy bluOxigen bitched that no one was helping, and when I tried to help, he disappeared |
| 20:15.44 | Qwell | wcselby: Welcome to IRC. |
| 20:15.46 | *** join/#asterisk DerkKo (n=afernand@75-149-178-131-Miami.hfc.comcastbusiness.net) |
| 20:16.02 | wcselby | Qwell - lol I know |
| 20:16.05 | wcselby | still |
| 20:16.53 | DerkKo | Hey Guys,,,, I have a small question.. I want to be able to start and stop a recording on an active sip call on asterisk on demand... (When i say on demand i mean trough agi or manager interface) without interupting the active call. |
| 20:17.19 | [TK]D-Fender | DerkKo: AMI <- |
| 20:17.29 | [TK]D-Fender | DerkKo: Go read the docs on this |
| 20:17.45 | DerkKo | I been reading trough docs... Let me read about AMI |
| 20:17.50 | wcselby | actually, he hasn't asked a question yet |
| 20:18.07 | DerkKo | Well i was going that way :-P |
| 20:18.11 | wcselby | i know :P |
| 20:19.01 | DerkKo | Anyways the actions on the AMI none are about recording the call. |
| 20:19.13 | nny | setting up a "cloud" (meh) hosted server with asterisk. They are using xen as the underpinnings afaik, any precautions or advice in regards to configs? No hardware dahdi on it, just have dahdi there for the dummy interface and I assume timing |
| 20:19.43 | nny | although I assume with no dahdi hardware timing is handled by the system itself |
| 20:20.50 | DerkKo | I see StopMonitor but never anything like start monitor |
| 20:21.20 | *** join/#asterisk timeshell (n=chatzill@142.46.193.194) |
| 20:21.32 | wcselby | DerkKo - what about Monitor (I don't know, just throwing it out there) |
| 20:21.43 | wcselby | <--- hasn't read the AMI docs |
| 20:21.50 | Katty | GASP |
| 20:22.13 | wcselby | and Katty is alive |
| 20:22.16 | timeshell | ~AMI |
| 20:22.17 | infobot | rumour has it, ami is the Asterisk Manager Interface, a way to control an Asterisk server via a TCP/IP socket. See http://voip-info.org/wiki/view/Asterisk+manager+API |
| 20:22.48 | timeshell | ~beef |
| 20:22.49 | infobot | i guess beef is what's for dinner. dead cow, or mad, or tasty |
| 20:23.24 | wcselby | http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Monitor |
| 20:23.30 | wcselby | DerkKo ^^ |
| 20:23.55 | DerkKo | Cool I see |
| 20:24.13 | DerkKo | Thanks :-) |
| 20:26.40 | wcselby | np |
| 20:26.56 | wcselby | thanks timeshell :) |
| 20:27.02 | *** join/#asterisk gardo (n=gardo@121.97.178.41) |
| 20:27.43 | dustybin | a strange thing happens to my queue system, at first it rings, then a voice says i am 1st in the queue, after that it goes silent |
| 20:27.59 | wcselby | dustybin - do you have any musiconhold defined? |
| 20:28.01 | timeshell | What are you expecting? |
| 20:28.04 | timeshell | lol |
| 20:28.17 | dustybin | exten => s,1,Queue(queue,rtwh,,,60) |
| 20:28.36 | dustybin | i would rather it rings than play MOH |
| 20:28.50 | *** join/#asterisk moy (n=moy@mail.e-contact.cl) |
| 20:28.56 | timeshell | Then play a ring tone for your MOH |
| 20:29.02 | timeshell | Or use a ring group. |
| 20:29.21 | wcselby | but it should be ringing since he's using the 'r' option |
| 20:29.25 | dustybin | i thought the 'r' option is enough |
| 20:29.32 | timeshell | Oh, yah |
| 20:29.40 | timeshell | Good point :p |
| 20:29.50 | dustybin | it rings at first, then the voice, then it goes silent, however, the asterisk log says it should be ringing |
| 20:30.04 | dustybin | maybe the polycom turns the ring off itself |
| 20:30.11 | wcselby | well I'd love to help, but I have no idea how you've got your queue configured :P |
| 20:31.02 | dustybin | http://paste.debian.net/46635/plain/46635 |
| 20:33.25 | wcselby | is it hanging up on you after the position announce? |
| 20:33.36 | wcselby | or are you still sitting in the queue? |
| 20:33.37 | dustybin | not handing up no |
| 20:33.43 | dustybin | still in the queue, then it goes into my menu |
| 20:33.49 | dustybin | it works, just goes silent |
| 20:34.07 | dustybin | let me try with my softphone |
| 20:34.45 | wcselby | i need to see output from the cli |
| 20:34.48 | dustybin | the softphone works, its ringing after the voice |
| 20:35.26 | wcselby | actually, it's time for me to headout |
| 20:35.40 | wcselby | bmoraca - I'll try two phones through my 2wire tonight and let you know tomorrow |
| 20:35.53 | nny | reading up on this, but just pinging the channel. Can you use Queue for calling macros? |
| 20:35.56 | wcselby | dustybin - sorry, just didn't have enough time tonight |
| 20:38.43 | *** join/#asterisk hugorebelo (n=hugorebe@200-171-132-124.completo.com.br) |
| 20:39.28 | *** join/#asterisk Whitor (n=Whitor@24.97.4.146) |
| 20:40.01 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 20:41.54 | ariel_ | Hello folks |
| 20:42.46 | Dovid | hi |
| 20:42.56 | Mango | How can I tell if my provider is proxying audio, or if I'm talking directly to the carrier? |
| 20:43.04 | ariel_ | I would like to see if there is any script out there to force asterisk to send a message to phones that there is voicemail waiting? |
| 20:43.11 | Dovid | Mango: RTP debug |
| 20:43.13 | Dovid | or sip trace |
| 20:43.15 | Mango | Dovid, cheers |
| 20:43.23 | generalhan | ahh, anyone in here ever work on a LG-Nortel 6812? i have this client that refuses to switch out his phones, cause he spent "so much money on them" but i cant figure out how to switch them from MCGP to SIP :( |
| 20:43.35 | Corydon76-dig | Mango: you can't, if they're doing it correctly |
| 20:43.38 | Dovid | generalhan: via tftp |
| 20:44.11 | generalhan | Dovid: i understand how to update the firmware, but i cant seem to find a firmware to switch it to SIP rather then the MGC that its using gnow |
| 20:44.12 | Dovid | Corydon76-dig: why not via rtp debug or looing for the audio IP in a sip trace ? |
| 20:44.36 | Corydon76-dig | Dovid: He asked whether they're proxying the audio |
| 20:44.45 | Dovid | generalhan: That i never did. just updaded them via tftp. i would assume u can get it from their site |
| 20:45.04 | *** join/#asterisk Skeeter- (n=wil_c_wi@c216.218.2-65.clta.globetrotter.net) |
| 20:45.06 | generalhan | Dovid: just got off the phone with nortel... they dont support this phone anymore. |
| 20:45.08 | dustybin | http://news.bbc.co.uk/1/hi/business/8257289.stm |
| 20:45.09 | Dovid | Corydon76-dig: why can't you just see if the rtp is going to them or not. If it's another IP some where else you can "assume" that they arent |
| 20:45.13 | dustybin | whoops wrong chan |
| 20:45.26 | Corydon76-dig | Dovid: it assumes that your provider uses a single IP |
| 20:45.30 | Dovid | generalhan: why whould they ;). checking what I have here |
| 20:45.48 | Skeeter- | http://www.asterisk.org/node/48325, is there anyway to edit that to had voicemail support, buddy add a msn like picture next to the ext. how to modify it to support voicemail |
| 20:45.51 | Corydon76-dig | Dovid: good providers use geographically distinct blocks |
| 20:46.01 | Dovid | Corydon76-dig: even if they use multiple. then again i assuming if they are proxying and they were the (man in the middle0. |
| 20:46.33 | Dovid | generalhan: I have some files. I will PM U |
| 20:49.15 | *** join/#asterisk luca`gervasi (n=ashura@host79-160-dynamic.55-79-r.retail.telecomitalia.it) |
| 20:49.18 | luca`gervasi | hello |
| 20:50.27 | *** join/#asterisk kondela (i=kondela@116.68.103.250) |
| 20:52.01 | *** join/#asterisk Tim_Toady (n=moi@adsl297-103.kln.forthnet.gr) |
| 20:59.00 | kondela | hey |
| 21:00.00 | kondela | i do have interresting issue across all versions of asterisk, can i dicuss it here..? |
| 21:01.06 | Mango | No. We only discuss the rare Albino Rhinocerous here. |
| 21:01.14 | Mango | (Yes, you may.) |
| 21:01.26 | kondela | nice |
| 21:01.48 | *** join/#asterisk Gokee2 (n=gokee2@24-113-159-168.wavecable.com) |
| 21:02.40 | kondela | this is my first time with IRC.. so excuses |
| 21:05.08 | Mango | No problem :) |
| 21:09.42 | ariel_ | every time I see the your name Mango I get hungry for some fruit |
| 21:09.56 | Mango | <grins!> |
| 21:10.11 | Mango | recommends a mango blueberry smoothie. |
| 21:12.22 | ariel_ | hehe, mango yogurt with a granola bar |
| 21:16.07 | kondela | mango can i send you a sample extensions.conf file..? |
| 21:22.14 | kondela | i do have an issue while using the combination of pattern matching (_1XX) and ${EXTEN} together.. anyone there could help? |
| 21:23.21 | Mango | kondela: Pastebin your extensions.conf and hopefully someone here will be able to help. |
| 21:29.18 | Pan3D | send everything to Mango |
| 21:29.21 | Pan3D | via PM |
| 21:29.25 | Pan3D | :) |
| 21:29.35 | ariel_ | kondela: what is the main issue? more info will also be needed as to what is happening and what is expected |
| 21:30.58 | kondela | i have created a text file, which explains the situation.. but i am just trying to figure out how "PASTEBIN".. well.. u know ..its my first time with an IRC |
| 21:31.28 | Corydon76-dig | ~pb |
| 21:31.29 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 21:31.29 | ariel_ | go to pastebin.ca and post it there. |
| 21:34.44 | kondela | i did so. here is the link |
| 21:34.47 | kondela | http://pastebin.com/m3b85c8f4 |
| 21:35.18 | Pan3D | what exactly is the problem? |
| 21:36.05 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 21:36.33 | kondela | ok.. anyone from context 'internal' can dial to 'reception' and vice-versa |
| 21:37.21 | nny | hrrrm |
| 21:37.29 | kondela | though these patterns math in two different contexts and i am not using any 'include' here |
| 21:37.32 | nny | anyone see a problem with this sip output? |
| 21:37.32 | nny | http://pastebin.com/ma648ac2 |
| 21:37.38 | nny | i don't get a ring on the other end |
| 21:37.48 | Pan3D | kondela: pose your issue in the form of a "This is what I want... this is what happens" |
| 21:38.09 | nny | suspect it's a missing setting in sip.conf, like the vitel-outbound host is trying to dial 192.168.100.128:5060 directly |
| 21:38.11 | Pan3D | first, your Dial() is odd |
| 21:39.13 | kondela | why, is there anything wrong with that format.. i mean this method serves a purpose |
| 21:39.28 | Pan3D | what are you trying to do? what happens instead? |
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| 21:39.45 | lirakis | nny, it says circuit is busy - and it never shows you getting a 100 trying or any other msg vrom vitelity |
| 21:40.06 | lirakis | nny, do you have plain sip debug enabled ? |
| 21:40.12 | lirakis | nny, or is it for a peer |
| 21:40.17 | nny | basically I have my server on a public IP and the phone is behing NAT on another network |
| 21:40.18 | nny | peer |
| 21:40.29 | lirakis | nny, but the call never goes through |
| 21:40.43 | nny | yeah but I am calling in on Vitelity too |
| 21:40.46 | nny | er |
| 21:40.54 | nny | Vitelity -> PBX -> desk phone |
| 21:41.01 | kannan | how to watch for any connections on the tftp server? |
| 21:41.04 | lirakis | nny, other way around |
| 21:41.16 | lirakis | nny, desk phone -> pbx -> vitelity -> pstn |
| 21:41.20 | kannan | to see if a cisco is connecting.. i am getting a timed out |
| 21:41.21 | lirakis | i am assuming? |
| 21:41.29 | kondela | here we go.. these are two different departments (reception and internal), they dont t even dial between them, but they share inbound and outbound ports of the same server |
| 21:41.30 | lirakis | nny, since you are calling out to vitelity |
| 21:41.34 | ariel_ | kondela: it looks ok what does it say in your cli when you dial what is your issue? |
| 21:41.46 | lirakis | nny, at any rate... has this ever worked? or did you just sign up with vitelity? |
| 21:41.56 | *** join/#asterisk Grof (n=krash@78-0-254-251.adsl.net.t-com.hr) |
| 21:41.58 | Pan3D | kondela: Ok, but what do you want to happen? and what happens instead? |
| 21:42.09 | Pan3D | I've asked the same question three times, and you seem to not be able to be concise. |
| 21:42.17 | lirakis | nny, you may need to log into your vitelity account mgt and authorize the ip for your server.... b/c it looks like your getting no response from vitelity |
| 21:42.26 | nny | likrakis hmm no |
| 21:42.35 | nny | likrakis I am calling the vitelity number |
| 21:42.41 | nny | and reaching the pbx |
| 21:42.46 | nny | so the connection is working *there* |
| 21:42.56 | nny | but when the pbx tries to pass the call to SIP/190 it fails |
| 21:43.00 | lirakis | nny, oaky so you have a vitelity did ? |
| 21:43.08 | kondela | as per documentation and my digium trainers, exten 101 should not dial 103.. because they are in different contexts.. |
| 21:43.19 | nny | lirakis yeah that's the number I am calling into |
| 21:43.26 | lirakis | nny, ok |
| 21:43.40 | nny | lirakis and I am reaching the PBX associated with that account |
| 21:43.54 | kondela | i dont want these people to communicate.. and i thought doing it in two different context is the best way |
| 21:44.11 | lirakis | nny, yeah i see now ... you had a lot of crap in the beginning of the trace |
| 21:44.31 | nny | lirakis but when the pbx tries to connect the two together it dies I suspect I am not telling asterisk to let the peers connect directly |
| 21:44.35 | kondela | CLI shows a normal SIP dial is taking place |
| 21:44.38 | nny | I thought reinvite=no handled that |
| 21:44.49 | Pan3D | kondela: yes, different contexts is correct. The caller is being dropped into the correct context initially, correct? |
| 21:44.57 | lirakis | nny, reinvite is just a means of changing the path of the rtp stream |
| 21:45.09 | lirakis | nny, it has nothing to do with an actual sip session |
| 21:45.14 | Pan3D | you can test this by putting unique logging or audio in the context, or looking at the console output. |
| 21:45.28 | nny | lirakis is it possible that the vitelity side is trying to dial 192.168.100.XXX:5060 directly after? |
| 21:45.32 | lirakis | nny, are they part of the same context? |
| 21:45.37 | kondela | NO.. thats the point i want to reach |
| 21:45.46 | lirakis | nny, no... your pbx registers to vitelity |
| 21:46.28 | nny | lirakis context is [main_menu] with a include => transfer, [transfer] is 190,s,1,Dial(SIP/190) |
| 21:46.44 | Pan3D | kondela: Ok, but you have to determine if the caller is dropped into the proper context. |
| 21:46.44 | lirakis | nny, and the context for the extension 190 ? |
| 21:46.48 | kondela | if you can still read my pastebin, caller 101, in context internal is happily calling exten 103, which is in complete different context - reception |
| 21:46.48 | lirakis | okay |
| 21:47.20 | kondela | yes the call originating from the right context |
| 21:47.36 | nny | I must have pebkac'd something hard, I have done this a thousand times and it usually just works (tm) |
| 21:47.46 | kondela | but it lands in a completely non-relevant context |
| 21:47.48 | ariel_ | kondela: your not restriting them enough |
| 21:48.23 | kondela | ariel... here we go.. |
| 21:48.29 | nny | lirakis the context for 190 is [transfer] |
| 21:48.47 | kondela | i just want highlight this part |
| 21:48.48 | nny | er do you mean the context in sip.conf? |
| 21:49.22 | kondela | I have tried the same , but just remooving the pattern matches |
| 21:49.35 | lirakis | nny, so you dialed your did, you got a menu, you dialed 199 and got an invalid ext msg, then you dialed 190 |
| 21:49.53 | lirakis | nny, it looks kinda jacked that its dialing vitel out to try and reach 190 |
| 21:49.54 | Pan3D | kondela: you don't need the pattern match in the hangup. |
| 21:50.05 | nny | lirakis er yeah sorry the 199 was a screw up |
| 21:50.26 | nny | lirakis yeah why is that? It shouldn't dial vitel-out |
| 21:50.37 | kondela | Pan3D,... give me a break.. you are not addressing the basic issue here.. |
| 21:50.51 | ariel_ | I have to go, but kondela it's a device on your asterisk so if your dial plan matches it going to dial the device |
| 21:51.22 | nny | lirakis want me to capture a full sip debug without the screwup? |
| 21:51.27 | kondela | the issue is , context overlap while using the the combi of pattern + ${EXTEN} variable |
| 21:51.35 | lirakis | nny, no |
| 21:51.43 | ariel_ | kondela: but your context knows of the dial pattern |
| 21:51.51 | ariel_ | devices are not restricted like that |
| 21:51.59 | Pan3D | ? |
| 21:52.09 | Pan3D | a device dials into a context |
| 21:52.14 | Pan3D | it should be limited ot that context |
| 21:52.18 | ariel_ | _1XX can call all devices like a sip phone |
| 21:52.50 | nny | lirakis just a curiosity but is it normal for the ports to list as something other than 5060 if the peer is behind NAT? |
| 21:52.55 | kondela | i think here we conflict with the the TFOT authors |
| 21:52.56 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 21:53.05 | nny | 190/190 64.203.244.XXX D N 1025 Unmonitored |
| 21:53.25 | Pan3D | kondela: you have a bad attitude. goodbye. |
| 21:53.27 | lirakis | nny, yeah |
| 21:53.29 | Pan3D | & |
| 21:53.33 | kondela | oops |
| 21:53.36 | kondela | why..? |
| 21:53.39 | lirakis | nny, ports can be negotiated for registrations |
| 21:53.50 | nny | lirakis yeah figured, guess that was a straw grasp |
| 21:53.56 | [TK]D-Fender | exorcises Pan3D |
| 21:54.00 | lirakis | nny, is that peer set for qualify=yes? |
| 21:54.03 | ariel_ | kondela: think of it this way. restrck it more like in one context _1X[1-2] other has _1X[3-4] |
| 21:54.17 | ariel_ | for dialing rule |
| 21:54.24 | kondela | ariel... yeah your solution works.. |
| 21:54.46 | kondela | i did that, and endup desired result |
| 21:54.47 | Pan3D | [TK]D-Fender: s/exorcises/excercises |
| 21:54.57 | ariel_ | context is more for inbound then out your devices are able to be dialed if you have the dial patter matching |
| 21:55.04 | [TK]D-Fender | Pan3D: No, EXORCISES |
| 21:55.13 | Pan3D | lol |
| 21:55.44 | nny | lirakis no |
| 21:55.46 | nny | lirakis hmmph |
| 21:55.47 | ariel_ | you seperate dial patters with context then include the ones you need for the devices |
| 21:55.51 | nny | lirakis and then.. it works |
| 21:56.00 | [TK]D-Fender | Pan3D: Your sense of humour and/or dictionary is broken |
| 21:56.05 | kondela | so i should seek other methods to restrict calls within the system..? is that what you are saying |
| 21:56.15 | ariel_ | kondela: yes |
| 21:56.21 | kondela | ok.. |
| 21:56.24 | ariel_ | remember dial patterns and devices |
| 21:56.31 | Pan3D | [TK]D-Fender: sorry, doing 2 things at once -- prepping for radio show |
| 21:56.35 | [TK]D-Fender | kondela: Contexts separate who can dial what |
| 21:56.38 | *** part/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 21:56.50 | nny | lirakis something odd here one sec |
| 21:57.05 | nny | lirakis oh crud |
| 21:57.07 | kondela | TK... thats something new to me |
| 21:57.10 | nny | lirakis realm=oldrealm |
| 21:57.11 | kondela | interesting |
| 21:57.20 | ariel_ | [TK]D-Fender: I have a question for you but I need to go to an hoa meeting are you going to be around is a few hours? |
| 21:57.32 | [TK]D-Fender | kondela: This is the most important part of *. TEH DIALPLAN. |
| 21:57.45 | [TK]D-Fender | ariel_: in & out throughout the evening |
| 21:57.54 | ariel_ | ok fair enough t/y |
| 21:58.06 | ariel_ | bbl off to get yelled at in the HOA meeting |
| 21:58.34 | kondela | i have been playing around with asterisk 1.2 , 1.4 and 1.6 all these years |
| 21:58.48 | Pan3D | billions and billions |
| 21:59.22 | nny | lirakis yeah, pbekac |
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| 21:59.27 | nny | lirakis pebkac too |
| 21:59.40 | kondela | and i really thought i could write a dialplan as i wrote |
| 21:59.53 | nny | kicks self |
| 22:00.13 | kondela | and my belief has been questioned unitl i write that piece of dialplan |
| 22:00.16 | lirakis | nny, working now? |
| 22:00.34 | nny | lirakis hmm kinda one sec |
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| 22:01.25 | kondela | TK... there is some interresting info to share, i found while trouble shooting this issue |
| 22:01.42 | [TK]D-Fender | kondela: Get a bigger gun. |
| 22:02.07 | [TK]D-Fender | has no trouble shooting things |
| 22:03.54 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 22:04.16 | kondela | i moved my dial part to macro, replaced pattern matching with specific match, while keeping ${EXTEN} as agrument for my macro |
| 22:04.43 | nny | lirakis yeah, i dunno lot's of pebkac here. Moved my system from my office to a host offsite. Copied the config over but failed to catch the realm statement in sip.conf. Corrected that, and tried again, and even though I started at the hosted system, when I dialed the phone, the local office system grabbed the call |
| 22:04.48 | kondela | and this worked as exactly in desired result |
| 22:05.10 | nny | lirakis so I disabled asterisk on that box for now, I guess having the PBX at the gateway, + a phone behind it trying to use the remote host was the suorce of the badness |
| 22:07.21 | nny | ffs |
| 22:07.27 | nny | lirakis no still having an issue |
| 22:08.21 | nny | lirakis but mayeb a new one, i'll get it figured out |
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| 22:43.41 | bmoraca | i freakin hate soho gear |
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| 23:59.27 | Sandheaver | bmoraca: why |
