| 00:03.45 | Strom_M | instant catsex |
| 00:03.48 | Strom_M | just ad cheese |
| 00:03.52 | Strom_M | s/ad/add/ |
| 00:03.53 | Qwell | ad? |
| 00:04.01 | Strom_M | typo :) |
| 00:06.10 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 00:07.55 | pfn | so... what's overlap dialing? |
| 00:16.05 | *** join/#asterisk `p4r14h (n=j0sh@69.92.145.178) |
| 00:17.42 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 00:17.42 | *** mode/#asterisk [+o mog] by ChanServ |
| 00:27.29 | *** join/#asterisk enkrypted (i=enkrypte@71.4.147.219.ptr.us.xo.net) |
| 00:29.35 | *** join/#asterisk cthorner (n=cthorner@209-234-185-130.static.twtelecom.net) |
| 00:29.39 | *** part/#asterisk cthorner (n=cthorner@209-234-185-130.static.twtelecom.net) |
| 00:30.13 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 00:31.56 | *** join/#asterisk nybble (n=nybble@about/apple/performa/nybble) |
| 00:32.46 | *** join/#asterisk nybble (n=nybble@about/apple/performa/nybble) |
| 00:33.12 | Igbothom_3rd | oh, just a little byte |
| 00:34.28 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-011c0c0e4b3df2b9) |
| 00:41.48 | nybble | lol Igbothom_3rd |
| 00:42.05 | Igbothom_3rd | sad, eh? |
| 00:42.25 | nybble | tis |
| 00:45.49 | nybble | *idles* |
| 00:47.13 | *** join/#asterisk DrukenLPY (n=jdumais@CPE000854ddcdb1-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 00:50.15 | DrukenLPY | happy easter everyone!!! |
| 00:50.19 | DrukenLPY | anyone around? |
| 00:50.40 | Qwell | haha, I just thought of something funny |
| 00:50.57 | Qwell | I put in code a few weeks back to display Mark's birthday... |
| 00:51.06 | Qwell | today is his birthday, and it just so happens to be easter |
| 00:51.41 | Qwell | therefore, my code is a literal easter-egg |
| 00:51.41 | Igbothom_3rd | that's sadder than my comment :) |
| 00:52.33 | nybble | lol |
| 00:53.24 | Qwell | I wonder if anybody anywhere actually enabled that code |
| 00:53.30 | Qwell | ...besides me |
| 00:53.30 | DrukenLPY | hey qwell, have you played with chillispot at all ? |
| 00:53.36 | Qwell | never heard of it |
| 00:53.48 | DrukenLPY | k |
| 00:55.22 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 00:58.03 | brian | is there no way to disable call parking? |
| 00:58.44 | brian | it may be a crazy concept, but I only want to use modules and features that I actually _need_ |
| 00:58.56 | brian | not use, that was a bad word, more like load. |
| 01:01.17 | *** join/#asterisk progcaribu (n=arturo@izones70.izones.net) |
| 01:02.47 | [TK]D-Fender | brian : if you don't include "parkedcalls" they can't park.... |
| 01:03.49 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 01:05.29 | pfn | hmm, is there any way to force a particular sip provider to prefix all extens? |
| 01:05.59 | pfn | e.g. if I do Dial(SIP/provider/12345) is there any way I can make the call to SIP/provider result in 999666#12345 |
| 01:06.39 | ManxPower | pfn: Dial(SIP/provider/999333#12345) |
| 01:06.48 | pfn | ManxPower, without having to do that explicitly in the dialplan |
| 01:06.50 | ManxPower | brian many of the core modules like res_features cannot be unloaded |
| 01:07.19 | ManxPower | pfn: um, just exactly where do expect Dial to get the 999666# from? |
| 01:07.35 | pfn | manxpower like in the sip.conf, perhaps |
| 01:08.16 | pfn | like dialprefix as a peer option |
| 01:08.16 | DrukenLPY | ManxPower: magic, pixie dust.... |
| 01:08.23 | pfn | 'dialprefix' |
| 01:08.33 | pfn | but I guess the answer to that is no |
| 01:08.58 | *** join/#asterisk progcaribu (n=arturo@izones70.izones.net) |
| 01:09.06 | ManxPower | pfn: I'm sure you can find some sort of work around |
| 01:09.30 | pfn | well, it'd be trivial to patch chan_sip to add support for that--but yeah, I'm lazy, I'll just hack it in the dialplan |
| 01:09.46 | pfn | I'd just like to be able to switch providers around transparently... |
| 01:10.56 | pfn | (more lazy on submitting a patch and hoping that it gets integrated) |
| 01:10.56 | [TK]D-Fender | pfn, how often do you change providers? |
| 01:11.04 | ManxPower | pfn: that is easy if each provider expects the same digits |
| 01:11.13 | pfn | ManxPower, it doesn't |
| 01:11.22 | ManxPower | pfn: it sucks to be you |
| 01:11.26 | pfn | [TK]D-Fender, depends |
| 01:11.38 | pfn | ManxPower, it's silly because the provider system needs the prefix as a billing code |
| 01:11.40 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 01:12.21 | ManxPower | why not just set PREFIX=999666# in [global] and then Dial(SIP/provider/${PREFIX}1234) |
| 01:12.29 | [TK]D-Fender | pfn, And don't you have to change all of your Dial statements anyways to change providers? |
| 01:12.33 | pfn | and it's ugly if it's in the dialplan because that shows up in the CDR |
| 01:12.33 | ManxPower | if you do not set the variable it will be empty and not do anything |
| 01:12.44 | pfn | [TK]D-Fender, not if I follow the sample dialplan |
| 01:12.47 | [TK]D-Fender | pfn, : and I was about to suggest exactly what ManxPower just did |
| 01:12.51 | pfn | and use Dial(${TRUNK}/${EXTEN} |
| 01:12.52 | pfn | ) |
| 01:12.57 | ManxPower | pfn: NOBODY follows the sample dialplan |
| 01:13.14 | pfn | well, some practices may be handy |
| 01:13.17 | [TK]D-Fender | pfn, what "sample" dialplan? You mean that psychotic BS that comes when you do "make samples"? |
| 01:13.25 | pfn | anyway, again, the big deal is the CDR in this case |
| 01:13.31 | pfn | I don't want that prefix in the CDR--what to do about that? |
| 01:13.44 | pfn | post-process? yuck |
| 01:13.46 | [TK]D-Fender | pfn, TFB |
| 01:13.57 | pfn | ? |
| 01:14.02 | [TK]D-Fender | pfn, the point of CDR is to show what you DID. |
| 01:14.11 | brian | Which Asterisk module parses extensions.conf? |
| 01:14.17 | pfn | pbx_config |
| 01:14.20 | [TK]D-Fender | pfn, not so you can change everything behind the systems back. |
| 01:14.43 | ManxPower | pfn: I can think of at least 1 way to do what you want |
| 01:14.53 | *** join/#asterisk DrukenHME (n=jdumais@CPE000854ddcdb1-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 01:14.59 | pfn | [TK]D-Fender, it's to show details of the call... this provider specific prefix is unnecessary |
| 01:15.01 | ManxPower | but it is much more complex that just using a prefix |
| 01:15.12 | pfn | ManxPower, details? |
| 01:15.40 | pfn | I can probably hack out dialprefix as a peer option in chan_sip.c in like 5 mins :p the pain is just remembering to keep my patch around when I update, etc. |
| 01:17.03 | [TK]D-Fender | Sounds a lot healthier to post-process..... |
| 01:18.03 | ManxPower | exten => _XXXX,1,Macro(prefix-fixup) |
| 01:18.10 | *** part/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-011c0c0e4b3df2b9) |
| 01:18.32 | ManxPower | exten => _XXXX,n,Dial(SIP/provider/${PREFIX}${EXTEN}) |
| 01:18.45 | ManxPower | bah, that is not right. |
| 01:18.57 | ManxPower | pfn if you want me to write your dialplan then send me money |
| 01:19.07 | pfn | heh, it's ok, thanks anyway |
| 01:19.25 | pfn | I'll just hack up chan_sip.c, it's a cleaner solution |
| 01:19.42 | ManxPower | yeah, much cleaner than setting 1 global variable |
| 01:20.02 | pfn | it results in setting 1 sip.conf config |
| 01:20.02 | wunderkin | probably going to have the same outcome |
| 01:20.05 | pfn | much cleaner |
| 01:20.13 | pfn | rather than having to have a convoluted dialplan |
| 01:20.13 | bulle | and MUCH easier to maintain |
| 01:20.30 | pfn | or a CDR record that is not indicative of whom I called |
| 01:22.09 | ManxPower | uh, your cdrs will be correct with my way |
| 01:22.11 | pfn | [TK]D-Fender, to address what the call "did", my call is to say, 18005551212 on, say, SIP/my_provider, I want the CDR to reflect that, not that I dialed 999666#18005551212 |
| 01:22.31 | pfn | the fact that it's SIP/my_provider already tells me that there was a 999666# in there |
| 01:23.48 | [TK]D-Fender | pfn, So go ahead and patch chan_sip and submit it to get merged. |
| 01:25.23 | ManxPower | pfn: exten => _XXXX,1,Dial(SIP/${TRUNK}/${PREFIX}/${EXTEN}) |
| 01:25.41 | ManxPower | ..er. |
| 01:25.47 | [TK]D-Fender | ManxPower, ;) |
| 01:25.49 | ManxPower | exten => _XXXX,1,Dial(SIP/${TRUNK}/${PREFIX}${EXTEN}) |
| 01:26.24 | [TK]D-Fender | ManxPower, However... that is exactly what he does not what to see.... strange "needs" indeed |
| 01:26.28 | ManxPower | you are ALREADY defining TRUNK= |
| 01:27.30 | ManxPower | [TK]D-Fender: You can either accept Asterisk oddities and live a happy and fulfilled life, or fight Asterisk's oddities and live a miserable and pointless life. |
| 01:27.38 | ManxPower | looks like pfn is in the 2nd catagory |
| 01:28.26 | pfn | ManxPower, not "fight its oddities" as you so say, but write, improve, enhance |
| 01:28.33 | pfn | which is how stuff evolves |
| 01:28.47 | ManxPower | you don't enhance Asterisk if your changes are not merged into the main code |
| 01:28.49 | pfn | now, if I just threw up my hands and said it's garbage, then it's a miserable, pointless life |
| 01:28.51 | *** join/#asterisk Mavvie (n=edwin@ppp250-213.lns3.syd6.internode.on.net) |
| 01:28.55 | pfn | ManxPower, not my fault :p |
| 01:29.16 | [TK]D-Fender | ManxPower, "I" accept them just fine :) |
| 01:29.50 | [TK]D-Fender | pfn, so... you submitting your patch then? |
| 01:30.44 | ManxPower | I think the real solution to the problem is to use a provider that does not suck |
| 01:30.55 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 01:31.38 | ManxPower | what kind of provider can't figure out who to bill without a prefix |
| 01:31.58 | pfn | ManxPower, a lot, apparently (and this is for bulk gateways, not a single b2c) |
| 01:32.08 | [TK]D-Fender | ManxPower, I'd sooner find the time to figure out someone ELSES prefix so I don't get billed ;) |
| 01:32.18 | pfn | stuff like mera softswitches, etc. |
| 01:32.20 | ManxPower | exactly |
| 01:32.46 | ManxPower | I can't even think of a ITSP that requires billing prefixes. |
| 01:33.02 | pfn | again, talking about switching software |
| 01:33.05 | pfn | not b2c itsp |
| 01:33.14 | pfn | b2b, bulk providers |
| 01:33.18 | bulle | sounds pretty insane to rely on prefix for billing, its as you said, a big mess just waiting to happen when people start to missuse it |
| 01:33.20 | ManxPower | you mean like Level3 |
| 01:34.09 | ManxPower | bulle: looks to me like a massive security hole if the provider can't figure out who to bill based on the SIP account IDs |
| 01:34.21 | bulle | ManxPower: yes |
| 01:34.22 | pfn | and yes, it's happened |
| 01:34.34 | pfn | ManxPower, well, the providers don't route your call if you don't have the prefix |
| 01:34.48 | ManxPower | heck, even the TELCOs can figure out what customer to bill without prefixes |
| 01:35.06 | DrukenHME | pfn: what ManxPower is saying is, what is stopping you from using someone else's prefix |
| 01:35.17 | pfn | DrukenHME, the fact that you don't know someone else's prefix |
| 01:35.27 | pfn | if you use any random prefix, then your call won't get routed |
| 01:35.29 | DrukenHME | can't be all that damn hard |
| 01:35.43 | pfn | DrukenHME, and again, this is for bulk routing, not individual b2c routing |
| 01:36.02 | pfn | for bulk routing, it quickly becomes apparent when you have calls that don't conform |
| 01:36.27 | pfn | especially when they're dropped |
| 01:36.43 | ManxPower | pfn: I have never personally seen someone with a request like yours. And I've been using Asterisk and been on this channel for a long time |
| 01:37.28 | DrukenHME | i'd like to know exactly what his defination of BULK is... |
| 01:37.38 | DrukenHME | 10 million mins a day ? |
| 01:38.43 | ManxPower | pfn: which provider is this so I can be sure to avoid them? |
| 01:39.36 | DrukenHME | oh come on ManxPower, you mean you wouldn't get an account and share your prefix with us, then claim the calls aren't yours? |
| 01:39.52 | ManxPower | DrukenHME: you have found me out! |
| 01:40.06 | ManxPower | DrukenHME: Fortunatly I don't bill for calls. |
| 01:40.13 | *** join/#asterisk _lanlv_ (n=chatzill@58.186.174.44) |
| 01:40.37 | DrukenHME | hehehe wouldn't the realtors be pissed over that... hehehe |
| 01:40.56 | ManxPower | DrukenHME: they are always pissed, that would be nothing new |
| 01:41.06 | DrukenHME | touche |
| 01:42.06 | ManxPower | There was a significant drop in complaints about network restrictions when one of the had thier identity stolen |
| 01:42.29 | DrukenHME | hehe |
| 01:42.45 | DrukenHME | who keeps their personal information like that on the computer anyways... |
| 01:43.01 | ManxPower | DrukenHME: Oh, I'm sure they fell for a phishing scheme |
| 01:43.01 | DrukenHME | most i keep on the pc is my name and birthday |
| 01:43.17 | DrukenHME | prolly |
| 01:44.10 | ManxPower | And we had very little fallout when we put in significant web filtering, the CEO noticed a significant increase in network preformance, and he was told why it was faster |
| 01:44.37 | DrukenHME | less porn... |
| 01:44.41 | DrukenHME | heehe |
| 01:44.55 | ManxPower | I think the next step is to just send the CEO a weekly report of the most requested sites |
| 01:45.28 | DrukenHME | how much you wana bet google is #1 every week ? |
| 01:45.35 | ManxPower | I would remove officially sanctioned sites like google |
| 01:46.03 | ManxPower | the various real estate sites, etc |
| 01:46.07 | DrukenHME | that and i dunno if the us has something simillar to mls.ca |
| 01:46.58 | ManxPower | don't get me started on the incompetent evil spawn of ActiveX monopolist creatures that is the MLS |
| 01:47.34 | DrukenHME | hehe |
| 01:48.12 | jnc | I think Qwell wandered off for a bit, are there others here perhaps that know of how to use Asterisk-gui? I have it set up, my question is more asterisk related |
| 01:49.06 | ManxPower | DrukenHME: the MLS is the main reason we cannot move the public workstations to Linux |
| 01:50.57 | pfn | ManxPower, it's a mera/mvts softswitch |
| 01:50.59 | pfn | ManxPower, not a provider |
| 01:52.23 | ManxPower | I shall make sure to stay away from them and any provider that uses them |
| 01:53.27 | brian | Which asterisk modules are responsible for playing sound files? |
| 01:53.44 | brian | I'm getting errors that files aren't there when they really are and all permissions are correct |
| 01:54.40 | DrukenHME | ManxPower: why does mls stop you from using linux? cause of the active x ? |
| 01:54.43 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 01:55.14 | ManxPower | DrukenHME: yes, because of ActiveX and the Crossover product still caused the curser and field selection |
| 01:56.27 | brian | I loaded the module app_playback.so but that's apparently not enough. |
| 01:56.41 | ManxPower | huh? |
| 01:56.50 | wunderkin | brian, your nubb self is probably including the file extension :D |
| 01:57.16 | J4k3 | aww man, don't rub his nub in. |
| 01:57.26 | masked | heh |
| 01:57.26 | ManxPower | Brian: if you are trying to make asterisk small footprint you might be better off looking at the asterisk distros for embeded devices |
| 01:58.38 | brian | wunderkin: no |
| 01:58.44 | wunderkin | liar |
| 01:58.46 | brian | ManxPower: I almost have it running fine. |
| 01:58.56 | brian | ManxPower: I think it's because I'm not loading any "format_" files. |
| 01:59.02 | [TK]D-Fender | brian, please pastebin all of these problems.... |
| 01:59.04 | wunderkin | what is it saying then |
| 01:59.22 | [TK]D-Fender | ~pb |
| 01:59.31 | jbot | somebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 02:01.16 | ManxPower | Brian: that would be one reason I'm sure |
| 02:01.25 | ManxPower | you might also need the codec_ modules |
| 02:03.06 | aptura | ManxPower did you ever get that dmark hum issue resolved a few days ago? |
| 02:03.13 | ManxPower | aptura: nope. |
| 02:03.32 | aptura | so the telco is going to deal with it I guess. |
| 02:03.36 | ManxPower | I'll retest the lines late tonight, write it up and then submit it to the telco repair |
| 02:04.07 | ManxPower | My lines are -13dbm so that could be part of the problem too |
| 02:04.15 | *** join/#asterisk voiper1 (n=luke@60.231.86.58) |
| 02:04.31 | ManxPower | -13dbm telco->cpe and -5dbm for cpe->telco |
| 02:05.08 | ManxPower | I don't know how low it has to be before the telco cares |
| 02:05.18 | Qwell | cares? ha |
| 02:05.31 | Qwell | "will do something about it", on the other hand... |
| 02:05.42 | Qwell | but they certainly won't "care" |
| 02:05.48 | ManxPower | aptura: one of the lines is mostly just 60Hz hum, which asterisk should be filtering out. |
| 02:05.56 | J4k3 | heh, my telco decided to have a T3 card die this morning... I think some tech wanted to make some triple overtime. |
| 02:05.59 | J4k3 | :| |
| 02:06.05 | ManxPower | PTSN is spec'd for 300Hz as the low end |
| 02:06.18 | *** join/#asterisk wubba (n=kmurrey@cable-76-215.sssnet.com) |
| 02:06.27 | aptura | Manx or the cards highpass filters if thay have them. |
| 02:06.43 | ManxPower | aptura: I use Adtran TA750 |
| 02:07.17 | paavum | hello |
| 02:07.30 | ManxPower | aptura: I've had zero luck with the Digium HPEC, but I'll be sending them more info on monday |
| 02:07.34 | paavum | I'm running asterisk 1.2.17 |
| 02:07.40 | paavum | with spandsp 0.02 |
| 02:08.01 | ManxPower | paavum: you really should upgrade your spandsp |
| 02:08.02 | paavum | when I try to do an rxfax |
| 02:08.20 | paavum | ManxPower ... 0.03 wont let me compile rxfax |
| 02:08.23 | *** join/#asterisk coppice (n=chatzill@90.203.17.210.dyn.pacific.net.hk) |
| 02:08.39 | aptura | ManxPower was this line hum a new issue that croped up? |
| 02:08.45 | paavum | it says something about a type conflict |
| 02:08.54 | ManxPower | aptura: I don't THINK so. |
| 02:09.25 | ManxPower | I do not currently use the system (I've not had a chance to run conduit to my cabin) and my users are....less than technical |
| 02:09.47 | paavum | Anyway... |
| 02:09.57 | paavum | when I try to do an rxfax... |
| 02:10.03 | paavum | I get segmentation fault |
| 02:10.20 | paavum | and asterisk breaks |
| 02:10.55 | paavum | -- Executing RxFAX("Zap/37-1", "/var/spool/asterisk/fax/12133377.tif") in new stack |
| 02:10.55 | paavum | Segmentation fault |
| 02:11.10 | ManxPower | paavum: I don't have any suggestions except for searching the mailing list archives |
| 02:11.48 | paavum | Already did and no luck :( |
| 02:12.02 | paavum | could it be my system? |
| 02:12.03 | ManxPower | I think it is time for a cup of coffee |
| 02:12.34 | ManxPower | I have alot of line testing to do at 10pm |
| 02:13.31 | [TK]D-Fender | paavum, no, I've had all the problems you have described over multiple versions of * recently as well |
| 02:15.00 | voiper1 | hi, whats the trick to getting outbound callerid going with te110p cards |
| 02:15.15 | ManxPower | voiper1: Using PRI? |
| 02:15.36 | voiper1 | yes pri, te110p inbound callerid works just not out. |
| 02:15.56 | [TK]D-Fender | voiper1, could be your telco is not permitting it |
| 02:16.02 | ManxPower | voiper1: if you are sending the correct callerid then you must get your telco to allow you to send it |
| 02:16.11 | ManxPower | just remmeber you can ONLY send the CLID number, not name |
| 02:16.34 | voiper1 | thanks i'll try that |
| 02:16.39 | ManxPower | well, you can send CLID name, but pretty much all telcos will ignore it and look it up based on the number |
| 02:16.45 | ManxPower | voiper1: what country are you in? |
| 02:17.07 | voiper1 | australia |
| 02:17.32 | ManxPower | ah. well do not send the leading zeros and do not include the country code when sending the CLID |
| 02:17.44 | ManxPower | Just your area/city code and number |
| 02:19.37 | *** join/#asterisk zimdog (n=zimdog@c-24-9-69-162.hsd1.co.comcast.net) |
| 02:21.04 | ManxPower | LOL! The lab that was trying to recreate the conditions of the birth of the universe had an explosion and the scientist that was being interviewed called the explosion "a fairly big bang" |
| 02:21.49 | Qwell | awful |
| 02:21.50 | coppice | it was sabotage by the US :-) |
| 02:22.06 | ManxPower | they had a magnet fail or something |
| 02:22.18 | coppice | a US built magnet |
| 02:22.22 | ManxPower | Qwell:the wording did not sound deliberate |
| 02:22.30 | Qwell | regardless |
| 02:22.39 | zimdog | I have just installed asterisk. created a sip.conf file with just one entry. When I try to connect i get wrong password. I have checked over the password on the softphone and the conf file and they match. I was wondering if there is some setting to tell which type of authorization to use |
| 02:23.08 | [TK]D-Fender | zimdog, pastebin what you've done. |
| 02:23.09 | [TK]D-Fender | ~pb |
| 02:23.14 | jbot | somebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 02:25.42 | zimdog | [TK]D-Fender: http://channels.debian.net/paste/5995 |
| 02:26.11 | zimdog | I had secret=4520 |
| 02:26.21 | Qwell | hmm, what's a good ipv6 provider these days? |
| 02:26.54 | Qwell | polerin: does comcast give out v6 addresses? :P |
| 02:27.00 | [TK]D-Fender | zimdog, guess you're just filling in the wrong user & pass in your client. Perhaps misinterpreting their prompts. |
| 02:27.35 | *** join/#asterisk sharp (n=sharp@dsl092-234-217.phl1.dsl.speakeasy.net) |
| 02:28.52 | zimdog | [TK]D-Fender: after reinstalling asterisk it would not start correctly. I had to remove the dundi module to get it to work. This would not have anything to do with the sip auth would it |
| 02:29.30 | polerin | Qwell: not yet |
| 02:29.31 | jnc | Qwell: 6bone is like the ultimate. good luck trying to get that running from a comcast endpoint though |
| 02:29.39 | Qwell | polerin: plans to do so? |
| 02:30.02 | jnc | Qwell: hurricane net has ipv6, also wing6 has a pop at ORD |
| 02:30.04 | polerin | Qwell: all of our CMTS's have been upgraded so I'd say yes |
| 02:30.09 | jnc | i'm in Chicago so yeah |
| 02:30.15 | Qwell | neat, I wonder if there's a beta or anything |
| 02:30.19 | polerin | but I think it's going to be q4 at least |
| 02:30.41 | [TK]D-Fender | zimdog, no. |
| 02:30.41 | Qwell | I'd be thrilled to be part of an ipv6 beta for comcast :D |
| 02:30.48 | [TK]D-Fender | zimdog, chek you client |
| 02:30.50 | jnc | right |
| 02:31.04 | jnc | hey Qwell any suggestions for me about asterisk-gui and making contact from SIP? |
| 02:31.04 | [TK]D-Fender | zimdog, when in doubt enable sip debug to see what * is receiving |
| 02:31.05 | polerin | Qwell: I'd be very supprised |
| 02:31.15 | Qwell | oh well, a guy can dream |
| 02:31.23 | jnc | I don't understand how to do this or if it's possible from asterisk-gui as-is |
| 02:31.50 | jnc | if it's not possible then I won't have to mess around, just not knowlegable enough to know what question to ask |
| 02:31.53 | jnc | :/ |
| 02:32.18 | jnc | you'd mentioned a bit about the context |
| 02:32.46 | jnc | it's from-sip-external if I'm reading this correctly, which is asterisk / asterisk-gui default settings |
| 02:32.51 | polerin | Qwell: it's going to be in an area not a group of people |
| 02:32.53 | polerin | that's my guess |
| 02:32.59 | Qwell | yeah |
| 02:33.11 | *** join/#asterisk deoptima (n=deoptima@c-71-228-222-87.hsd1.tn.comcast.net) |
| 02:33.38 | jnc | comcast + ipv6 = routing tables that explode randomly |
| 02:33.53 | Qwell | comcast + ipv4 = routing tables that explode randomly |
| 02:34.35 | zimdog | [TK]D-Fender: I have rechecked all the client settings. Pretty sure it is not on that end. I have sip debug enabled also I just see 401 unathurized. I think something is wrong with the asterisk install. |
| 02:35.19 | [TK]D-Fender | zimdog, No. * does not lie about this stuff. make sure you have applied your changes with a reload, and double check everything, and then SHARE the SIP debug info for your phone |
| 02:35.25 | jnc | I take it my inquiry about asterisk-gui is going ignored :/ |
| 02:35.37 | Qwell | jnc: I don't know |
| 02:35.45 | jnc | oh, sorry to bother you about it then |
| 02:35.59 | Qwell | I just code |
| 02:36.49 | *** join/#asterisk znoG (n=gs@OL132-95.fibertel.com.ar) |
| 02:36.57 | [TK]D-Fender | jnc, I have been unable to actually discern a QUESTION from everything you have written recently. Whats the problem? |
| 02:37.38 | jnc | [TK]D-Fender: using only asterisk-gui, is it possible to configure asterisk to accept a SIP call and route to a user extension (say arbitrary extension 601) |
| 02:38.16 | Qwell | crap, crap, crap |
| 02:38.17 | [TK]D-Fender | jnc, ok... dunno.... have you tried? |
| 02:38.17 | jnc | my question for asterisk-gui people is "why does Incoming Call Route require a trunk" |
| 02:38.22 | Qwell | stupid busybox |
| 02:39.03 | Qwell | we can get busybox grep to segfault with either `echo "" | grep .` or `echo -e "\\n" | grep .`...and both segfault in different places |
| 02:39.03 | jnc | [TK]D-Fender: yeah I get as far as enabling guest SIP calls using asterisk-gui, my Ekiga softsip client can make the call and I hear asterisk "the extension ... is not in service" |
| 02:39.22 | zimdog | [TK]D-Fender: http://channels.debian.net/paste/5996 |
| 02:39.26 | jnc | but how to make that route to an extension? asterisk-gui is coded for a workflow of traditional pstn trunks |
| 02:40.05 | zimdog | I see the Digest algorithm=MD5 maybe I am not loading this? |
| 02:40.15 | *** join/#asterisk tetsuzan (i=fox@201.2.206.6) |
| 02:40.36 | jnc | my desired goal is random SIP soft client -> asterisk -> asterisk extension 601 voicemail |
| 02:40.47 | jnc | not for production, just as a test |
| 02:41.10 | [TK]D-Fender | jnc, from what I've heardApr 8 20:38:06 NOTICE[8573]: chan_sip.c:11228 handle_request_register: Registration from '<sip:4520@69.13.208.123>' failed for '24.9.69.162' - Wrong password |
| 02:41.16 | [TK]D-Fender | zimdog, rather |
| 02:41.34 | [TK]D-Fender | zimdog, you seem to have put an IP where it is expecting a username |
| 02:42.18 | [TK]D-Fender | jnc, You may actually have to code this in * itself. |
| 02:42.29 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 02:42.42 | [TK]D-Fender | jnc, GUI's can't account for every idea in your head and I think this falls sqaurely in that category. |
| 02:43.08 | jnc | [TK]D-Fender: the GUI should account for this |
| 02:43.11 | jnc | IMO anyways |
| 02:43.28 | jnc | what is it I am missing? a service provider definition? |
| 02:43.46 | zimdog | [TK]D-Fender: strange I am using ekiga and set up the account just like I have to the other asterisk servers I register to. |
| 02:43.48 | [TK]D-Fender | jnc, again I don't hink there is a place in the GUI at ALL for this |
| 02:44.04 | russellb | bkruse_home: i'm kind of a big deal, ok |
| 02:44.13 | bkruse_home | lol |
| 02:44.19 | bkruse_home | see redbull |
| 02:44.22 | jnc | heya bkruse |
| 02:44.59 | [TK]D-Fender | jnc, if you are looking for answers on where the GUI puts stuff I'd be guessing you should be asking in #asterisk-gui |
| 02:45.05 | *** join/#asterisk techie (n=gus@voip.routedsystems.com) |
| 02:45.08 | bkruse_home | jnc: wuts up |
| 02:45.54 | jnc | bkruse_home: now that asterisk-gui is operational ;) I'm attempting feign at configuring a softsip phone to dial in locally |
| 02:46.13 | jnc | doesn't appear at first a means to do this from the GUI |
| 02:46.40 | bkruse_home | are you serious....... |
| 02:46.43 | bkruse_home | users, first tab. |
| 02:46.48 | bkruse_home | i think its a user error |
| 02:46.52 | jnc | must be |
| 02:46.54 | bkruse_home | :P |
| 02:47.02 | Qwell | he means anon calls |
| 02:47.35 | Qwell | like, what to do when it hits the guest entry |
| 02:47.42 | zimdog | [TK]D-Fender: I appreciate the help. I think I am going to call it quits for the night. I guess I may still have some junk left over from the freepbx that was installed before. I thought I had deleted all the important stuff /etc/asterisk /usr/lib/asterisk. |
| 02:47.50 | bkruse_home | oh, gotchja |
| 02:47.52 | bkruse_home | gotcha* |
| 02:48.03 | *** join/#asterisk Knight_DKN (n=knight_d@c58-107-190-112.thoms2.vic.optusnet.com.au) |
| 02:48.08 | bkruse_home | zimdog: freepbx != good |
| 02:48.19 | [TK]D-Fender | zimdog, time to completely flush extensions.conf & sip.conf |
| 02:48.51 | zimdog | i did a stop now and restarted it would that not clear everything |
| 02:49.46 | jnc | does it make sense to allow configuration from asterisk-gui of my test case? |
| 02:50.43 | zimdog | I have also noticed that asterisk seems to not respond correctly. when I type reload for instance I did not see it do it anything. |
| 02:51.08 | jnc | zimdog: 'reload now' ? |
| 02:51.15 | *** join/#asterisk etfonhomey (n=etfonhom@74-140-213-69.dhcp.insightbb.com) |
| 02:51.30 | [TK]D-Fender | "reload" |
| 02:51.35 | [TK]D-Fender | IS ALL YOU SHOULD NEED TODO |
| 02:52.58 | zimdog | yea somthing is wrong with my system. I type reload it did nothing. I then typed set verbose 6 and it did nothing |
| 02:53.14 | jnc | bkruse_home: in freepbx I'd do this with a hack, setting up an incoming call rule matching Any CID / Any DID |
| 02:53.28 | zimdog | no commands seem to be doing anything right now. I think It is waiting for a reload to happen. |
| 02:53.56 | zimdog | There must be more stuff I need to get rid of from the freepbx install. |
| 02:54.01 | jnc | not possible w/ asterisk-gui AFAICT |
| 02:54.20 | zimdog | I installed asterisk with svn 1.2 branch |
| 02:54.54 | *** part/#asterisk Cybertoy (n=cybertoy@dsl254-123-112.nyc1.dsl.speakeasy.net) |
| 02:56.00 | zimdog | I appreciate all the help. Tomorrow I will try again to remove all traces of freepbx and see if I can get a clean asterisk install |
| 02:57.19 | zimdog | nite [TK]D-Fender thanks again for the help |
| 03:02.06 | [TK]D-Fender | russellb, You still lurking? |
| 03:02.28 | russellb | yes, but *extremely* tired |
| 03:02.36 | russellb | so my usefulness is near zero |
| 03:03.22 | *** join/#asterisk marcan (n=marcanso@160.10.7.121) |
| 03:03.49 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:04.44 | [TK]D-Fender | russellb, Realtively easy question : I'm about to upgrade to 1.4.2 at home. To get your devstate stuff I need to download SVN current, correct? |
| 03:05.03 | russellb | svn trunk, yeah |
| 03:05.17 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 03:05.20 | [TK]D-Fender | russellb, trunk = latest, right? |
| 03:05.32 | Corydon76-home | Latest what? |
| 03:05.36 | [TK]D-Fender | russellb, As opposed to FTP release |
| 03:05.56 | Qwell | latest, but not the latest 1.4 |
| 03:05.56 | bkruse_home | latest working? maybe....... |
| 03:05.58 | Corydon76-home | You should never be running trunk in production |
| 03:06.03 | bkruse_home | nope |
| 03:06.09 | russellb | R=http://svn.digium.com/svn/asterisk ; svn co $R/branches/1.4 asterisk-1.4 ; svn co $R/trunk asterisk-trunk ; cp asterisk-trunk/funcs/func_devstate.c asterisk-1.4/funcs |
| 03:06.10 | Corydon76-home | SVN 1.2 or SVN 1.4 |
| 03:06.13 | *** join/#asterisk Strom_C (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 03:06.21 | Qwell | russellb: cheater! |
| 03:07.02 | [TK]D-Fender | russellb, So I should be able to simply merge it into "stable" releases? |
| 03:07.03 | Qwell | I'd just svn cat it :D |
| 03:07.19 | russellb | Qwell: nice. |
| 03:07.48 | *** join/#asterisk newsmafia (n=newsmafi@wsip-68-224-174-204.sd.sd.cox.net) |
| 03:07.52 | Corydon76-home | "stable" is a Debianism. We prefer the term "release" |
| 03:07.54 | Qwell | svn cat $R/trunk/funcs/func_devstate.c > asterisk/branches/1.4/funcs/func_devstate.c |
| 03:08.12 | Qwell | save on bandwidth |
| 03:08.24 | russellb | very nice |
| 03:09.39 | *** join/#asterisk HeyItsMe (i=user@adsl-074-167-204-062.sip.hsv.bellsouth.net) |
| 03:10.13 | Corydon76-home | Nooooooo |
| 03:10.23 | russellb | Corydon76-home: you are no longer maintained, sorry |
| 03:10.35 | Qwell | You guys missed me easter-egg comment earlier, heh |
| 03:10.44 | Qwell | my* |
| 03:10.50 | Qwell | Corydon76-home: great, you just broke svnmerge |
| 03:10.53 | Qwell | way to go |
| 03:11.05 | Qwell | 1.0 > 1.2 > 1.4 :P |
| 03:11.05 | HeyItsMe | Hi all. |
| 03:11.12 | Corydon76-home | Oh, right |
| 03:11.20 | Corydon76-home | Oops. :-P |
| 03:11.31 | russellb | great, now automerge blew up on all of my brances |
| 03:11.40 | russellb | Corydon76-home: you clogged the svn tubes |
| 03:12.35 | Corydon76-home | Not exactly what I want to joke about just after having committed a bunch of stuff... |
| 03:12.48 | Qwell | Corydon76-home: russellb: see msgs :p |
| 03:13.01 | russellb | Corydon76-home: sorrrrrrrry |
| 03:13.05 | russellb | nothing is broken :) |
| 03:13.08 | Qwell | heh |
| 03:13.12 | Corydon76-home | phew |
| 03:14.02 | brian | Is there a specific asterisk module that "listens" for DTMF events? |
| 03:14.31 | HeyItsMe | I have a desparate need for help on Trixbox with a te205p. I can't seem to find clear help on exactly how to set this up to use the channels properly. Can anyone help? I hope I am not interupting. |
| 03:14.38 | *** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com) |
| 03:15.43 | *** part/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 03:16.03 | HeyItsMe | ? |
| 03:16.21 | Qwell | HeyItsMe: we could help if you weren't using trixbox |
| 03:16.36 | HeyItsMe | Oh, sorry |
| 03:16.45 | HeyItsMe | What could I do different? |
| 03:16.47 | jnc | there is #trixbox |
| 03:17.04 | Qwell | HeyItsMe: install asterisk by itself on a distro you install |
| 03:17.16 | Corydon76-home | ~trixbox |
| 03:17.30 | jbot | Trixbox is a full linux distro that includes *, FreePBX, and other 3rd party add-ons. It is these things on top of * which make it seriously painful to support and hence you will find little help here for it. Try asking in #freepbx , or their forums at http://www.trixbox.org/modules/newbb/ |
| 03:17.30 | HeyItsMe | thanks, I just was impressed with all the combined features and gui that it offered. |
| 03:18.04 | HeyItsMe | Thank you everybody I really appreciate thet help. |
| 03:18.42 | Corydon76-home | I love canned responses |
| 03:18.49 | HeyItsMe | Is there a way to get a comparable build with a straight build? |
| 03:18.58 | [TK]D-Fender | Corydon76-home, sealed for freshness ;) |
| 03:20.07 | *** part/#asterisk HeyItsMe (i=user@adsl-074-167-204-062.sip.hsv.bellsouth.net) |
| 03:20.12 | *** join/#asterisk techie (n=gus@voip.routedsystems.com) |
| 03:27.34 | brian | on sip to sip calls DTMF is being ignored |
| 03:28.24 | *** join/#asterisk EzWay (n=ez@c66.110.149-45.clta.globetrotter.net) |
| 03:28.28 | ManxPower | Brian: so when you call from one sip phone to another sip phone you can't hear the DTMF when the other side presses numbers? |
| 03:29.10 | brian | nope |
| 03:29.21 | brian | i don't know if I can hear them or not |
| 03:29.30 | brian | The buttons are being pressed to Asterisk. |
| 03:29.35 | ManxPower | uh, well try it |
| 03:29.45 | brian | I looked at the SIP debug though |
| 03:29.48 | omm | I have autologoff set for my agents but its not making static agents pause automatically when they miss a call. What can I do to automatically pause the member in the queues? |
| 03:29.58 | brian | when I press the button I don't see any new SIP packets |
| 03:30.00 | brian | so i assume no |
| 03:30.02 | ManxPower | omm: I don't know if you can |
| 03:30.12 | Qwell | brian: check your dtmfmoded |
| 03:30.14 | Qwell | dtmfmode |
| 03:30.25 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 03:30.30 | [TK]D-Fender | omm, I think you would be doing this in dialplan with chan_local |
| 03:30.41 | ManxPower | Brian: never assume. DTMF problems in SIP Phone to SIP Phone (no provider) is almost always a mismatch between the DTMF mode set on the phone and the DTMF mode set for that user in sip.conf |
| 03:30.49 | brian | Qwell: inband |
| 03:30.55 | Qwell | ugh |
| 03:31.05 | brian | whaaat |
| 03:31.10 | [TK]D-Fender | omm, So you'd add a static member like "member=Local/1234@agents", and set the ring time low eough that you can do "ause" if dial times out. |
| 03:31.12 | brian | is it something i said |
| 03:31.20 | ManxPower | Brian: inband dtmf only works with ulaw and alaw and you will NOT see DTMF packets if you use inband |
| 03:31.36 | brian | i'm using ulaw |
| 03:31.43 | Qwell | what is your phone using? |
| 03:31.48 | Qwell | rfc2833 no doubt |
| 03:31.52 | ManxPower | since with inband the dtmf is just audio |
| 03:31.53 | brian | inband too |
| 03:31.56 | [TK]D-Fender | brian and you will STILL not soo DTMF packets because it is not SIGNALED |
| 03:31.59 | omm | [TK]D-Fender,I've looked all over for an example... got any references I can look at |
| 03:32.16 | [TK]D-Fender | omm, I jsut gave you a small sample, go try. |
| 03:32.28 | brian | [TK]D-Fender: It still has to send the audio |
| 03:32.36 | omm | ok thanks |
| 03:32.44 | [TK]D-Fender | omm, make it dial the aget for LESS than the timeout period, then pause him upon timeout. |
| 03:32.47 | ManxPower | Brian: audio is sent using RTP, not SIP. |
| 03:32.53 | brian | oh |
| 03:32.54 | brian | oops |
| 03:32.58 | tengulre | anybody using shorewall with asterisk in here? |
| 03:33.02 | [TK]D-Fender | brian, its jsut dumb audio at that point. |
| 03:33.08 | ManxPower | SIP, MGCP, SCCP, and H323 are all signalling protocols, they all use RTP for audio |
| 03:33.37 | Qwell | for media ;) |
| 03:33.37 | brian | well why can't asterisk get the DTMF then |
| 03:33.41 | [TK]D-Fender | tengulre, Shorewall is just a front end to IPTABLES. whats the real questio here? |
| 03:33.53 | tengulre | how to using iptables to open port 10000-20000 in rules |
| 03:34.06 | brian | I'm thinking it could be because the module that listens for DTMF isn't loaded |
| 03:34.16 | brian | But I'd think chan_sip did that |
| 03:34.26 | [TK]D-Fender | brian, Why is it you think NO modules are loaded. What the hell kind of installationa re you running? |
| 03:34.29 | ManxPower | Brian: asterisk is not listening for DTMF |
| 03:34.36 | ManxPower | you are doing sip to sip |
| 03:34.53 | brian | so what does that mean |
| 03:35.21 | brian | what is the other option? rfc2883? |
| 03:35.21 | [TK]D-Fender | brian, first you think you have no codecs, no formats, no parser for this, or THAT.... what the heck! |
| 03:35.21 | ManxPower | now if you are having problems with SIP calls into asterisk that would be totally differennt |
| 03:35.34 | ManxPower | [TK]D-Fender: he's trying to find the hardest way to get asterisk setup. |
| 03:35.42 | brian | [TK]D-Fender: I'm keeping the Asterisk setup minimal |
| 03:35.43 | *** join/#asterisk k-man (n=jason@unaffiliated/k-man) |
| 03:36.02 | Qwell | how about getting it working first? |
| 03:36.08 | ManxPower | Brian: perhaps you should get it working before you try to minimilize it. |
| 03:36.12 | *** join/#asterisk asama_hideki (n=marreka@CPE000c416ef188-CM00195edd2378.cpe.net.cable.rogers.com) |
| 03:36.17 | ManxPower | Qwell: are you my long lost twin? |
| 03:36.19 | brian | Yeah i should probably stop prematurely optimizing |
| 03:36.27 | brian | It's bad |
| 03:36.27 | ManxPower | (that I didn't know I had) |
| 03:36.31 | Qwell | ManxPower: must some weird proximity thing |
| 03:36.32 | [TK]D-Fender | brian, Go see if KY is on special.. this is going to HURT. |
| 03:36.41 | *** part/#asterisk asama_hideki (n=marreka@CPE000c416ef188-CM00195edd2378.cpe.net.cable.rogers.com) |
| 03:36.43 | omm | heh |
| 03:36.44 | brian | [TK]D-Fender: You say that about everything. |
| 03:36.47 | ManxPower | Brian: expect to run asterisk for several months in production before you are ready to minimilize it |
| 03:36.54 | tengulre | [TK]D-Fender: do you know? how to open ports 10000-20000 in iptables(shorewall)? |
| 03:37.02 | Qwell | minimilize? |
| 03:37.12 | [TK]D-Fender | brian, No, you seem to be systematically breaking EVERYTHING in your madd attepmts at using *. |
| 03:37.20 | ManxPower | tengulre: I would suggest you find some shorewall support |
| 03:37.31 | brian | [TK]D-Fender: I didn't break anything |
| 03:37.38 | *** join/#asterisk asama_hideki (n=marreka@CPE000c416ef188-CM00195edd2378.cpe.net.cable.rogers.com) |
| 03:37.54 | tengulre | ManxPower: if in iptables how to do? |
| 03:38.08 | ManxPower | tengulre: I have no idea. since iptables IS NOT PART OF ASTERISK |
| 03:38.10 | asama_hideki | Hey. I was wondering if anyone has any experience compiling libpri on Mac OSX PPC. If so, can I ask you for some help? |
| 03:38.42 | Qwell | asama_hideki: I'm not sure what good it'll do without zaptel |
| 03:38.58 | asama_hideki | I'm planning on compiling zaptel as well, but I'm trying to get libpri to compile first |
| 03:39.09 | Qwell | zaptel isn't gonna work on osx |
| 03:39.21 | asama_hideki | oh, really? good to know. hah |
| 03:39.37 | Qwell | it's linux kernel drivers |
| 03:39.50 | Qwell | mostly, anyhow |
| 03:40.13 | Corydon76-home | If you wanted to port it to OS X, you're more than welcome to |
| 03:40.17 | tengulre | ManxPower: do you have firewall on front of your asterisk box ? |
| 03:40.26 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 03:40.29 | Qwell | Corydon76-home: it'll take like an hour, tops |
| 03:40.33 | ManxPower | tengulre: none of my asterisk boxes ever connect to the internet |
| 03:40.37 | asama_hideki | hmm. sounds like unnecessary pain right now. haha ;) |
| 03:40.45 | ManxPower | but really, your question is not asterisk specific |
| 03:40.50 | Corydon76-home | Qwell: heh, yeah, right |
| 03:40.54 | Qwell | asama_hideki: it'll take a VERY significant amount of effort |
| 03:40.55 | [TK]D-Fender | asama_hideki, You've come to the right place... you'll fit right in! |
| 03:41.06 | tengulre | ManxPower: thanks! |
| 03:41.23 | [TK]D-Fender | tengulre, jsut ask in #linux , you'll have an answer in no time at all... |
| 03:41.30 | asama_hideki | hehe. I'm just procrastinating studying for finals by playing around with asterisk. I never realized how cool it could be :) |
| 03:41.37 | [TK]D-Fender | tengulre, I've asked it myself before |
| 03:41.45 | ManxPower | I'm procrastinating doing line tests |
| 03:41.55 | brian | I think the codec for telephone-event isn't loaded. |
| 03:42.12 | [TK]D-Fender | brian, of course not... nothing is loaded :) |
| 03:42.18 | Corydon76-home | Heh, the codec for telephone-event? |
| 03:42.24 | brian | I don't know. |
| 03:42.30 | omm | [TK]D-Fender, I've looked at my queue.conf file... |
| 03:42.31 | [TK]D-Fender | brian, and inband = not an event when * isn't looking for it |
| 03:42.33 | tengulre | [TK]D-Fender: ;) |
| 03:42.33 | ManxPower | Corydon76-home: You know, for inband DTMF! |
| 03:42.33 | EzWay | there is a way to play a audio file , over current channel used t speak... ? |
| 03:42.43 | brian | I'm using rfc2833 |
| 03:42.48 | omm | I dont see a ause |
| 03:43.05 | brian | Because inband wasn't working |
| 03:43.18 | brian | But rfc2833 isn't working either |
| 03:43.22 | [TK]D-Fender | omm, not a feature there, it should be put in the DIALPLAN when you ring the agent |
| 03:43.27 | danp | when was asterisknow beta 4 released? |
| 03:43.33 | omm | ah |
| 03:43.40 | Qwell | danp: beta5 was released |
| 03:43.49 | danp | not according to the website |
| 03:43.54 | Qwell | oh? |
| 03:43.59 | danp | http://www.asterisknow.org/ |
| 03:44.20 | asama_hideki | anyway. Thanks for your help! Good night everyone |
| 03:44.21 | danp | yes, and unfortunately going to the download directory's URL doesn't give a file list |
| 03:44.32 | tengulre | I have other question, how to create a custom linux iso, like asterisknow? |
| 03:44.47 | Qwell | tengulre: WAY out of scope of this channel |
| 03:45.07 | *** join/#asterisk `p4r14h (n=j0sh@69.92.145.178) |
| 03:45.28 | [TK]D-Fender | tengulre, .... #linux <- |
| 03:45.49 | tengulre | Thanks :) |
| 03:45.52 | brian | ugh agh |
| 03:46.04 | [TK]D-Fender | tengulre, or maybe #homerenovations , or #autobodydetailing |
| 03:46.05 | JunK-Y | Qwell: so marko's party's over now? |
| 03:46.11 | Qwell | yeah, for about 2 days :p |
| 03:46.19 | ManxPower | *sigh* We help people with ASTERISK. Not Asterisk GUIs, management web interfaces, linux, building distros, packet filtering, networking, routing, or anything else that is not ASTERISK |
| 03:46.29 | brian | Hi. |
| 03:46.43 | brian | So what module listens to the telephone events? |
| 03:47.05 | ManxPower | And even though there is an H323 channel driver in the asterisk source code, we ill point and laugh at you if you want help with it. |
| 03:47.29 | danp | Qwell: i can't seem to just change beta4 in the ISO url to beta5...is it available somewhere? |
| 03:47.37 | Qwell | danp: I could've sworn it was |
| 03:47.41 | brian | ugh |
| 03:47.42 | aptura | ManxPower yea goes to show you there are still lots of people in this world that need a asterisk consultant even for simple things :) |
| 03:47.49 | brian | Damn you Asterisk!!! |
| 03:48.12 | brian | This is really not working out as planned. |
| 03:48.42 | omm | brian, I feel yoru pain |
| 03:48.55 | aptura | haha should have been here four or so years ago when there was almost no asterisk documentation online :) |
| 03:49.00 | danp | i don't really like GUI/management stuff but i'm working on a 12 FXO, ~55 FXS system that the end user needs to be able to maintain for the most part |
| 03:49.08 | Qwell | aptura: pfft, there was the code |
| 03:49.19 | ManxPower | To effectively manage an IP/TDM PBX, you need to know the software, the OS, Networking, VoIP protocols, routing, and telecom |
| 03:49.26 | *** join/#asterisk mrdigital (n=ASFASDF@pool-72-92-103-188.phlapa.east.verizon.net) |
| 03:49.30 | ManxPower | this is not easy. |
| 03:49.34 | aptura | QWell, I have yet to know enough c to make use of it without getting borred. |
| 03:49.41 | mrdigital | anyone heard of the Zoom 5801 ATA Adapter? |
| 03:49.42 | ManxPower | if you want easy get a couple of analog lines from the telco |
| 03:49.50 | omm | :D |
| 03:50.24 | [TK]D-Fender | mrdigital, Avoid.... low on features, attracts touble. their "FXO" is typically only a failover potr with is near useless |
| 03:50.34 | aptura | Manx would you agree that most lack telcom skills of all the above you mentioned? |
| 03:50.59 | brian | Right so can anyone tell me if there is a asterisk module that listens for telephone events |
| 03:51.11 | ManxPower | aptura: I don't know, I do know that most last some of those required skills |
| 03:51.23 | [TK]D-Fender | brian, telephone events are channel specific. If you have chan_sip, then thats it. |
| 03:51.28 | aptura | yea :) |
| 03:51.34 | ManxPower | I used to manage tech ops for a small ISP so I had a headstart in T-1s and PRIs |
| 03:52.23 | ManxPower | I currently do consulting on WAN, LAN, Linux, packet filtering, e-mail, spam filtering, virus filtering, and QoS, so I had a head start in most of the other aspects required. |
| 03:52.25 | omm | Will FreePBX modify teh dialplan if I make changes to it? |
| 03:52.25 | aptura | btw happy easter :) |
| 03:53.02 | *** join/#asterisk bmg505 (n=leon@196.209.180.232) |
| 03:53.16 | aptura | ;) |
| 03:53.37 | omm | Yeah I know :( |
| 03:53.58 | [TK]D-Fender | omm, you have sold your soul to FreePBX, prepare to burn..... |
| 03:54.00 | danp | ManxPower: pretty much the same for me |
| 03:54.01 | ManxPower | Maybe the people that help most on #asterisk should just put people on /ignore that ask questions that are not on topic |
| 03:54.24 | ManxPower | danp: only a small part of my consulting practice is telecom related |
| 03:54.28 | mrdigital | [TK]D-Fender: do you recommend any $20-100 FXO/FXS Cards or adapters? |
| 03:54.32 | [TK]D-Fender | ManxPower, I haven't yet reached your level of BOFH-ness yet.... ;) |
| 03:54.39 | danp | ManxPower: again, same :P |
| 03:54.48 | danp | i'd like to do more, though |
| 03:54.52 | [TK]D-Fender | mrdigital, depends how many ports, and if thats a cost/port figure. |
| 03:54.54 | danp | i find it pretty fascinating |
| 03:54.58 | aptura | Manx get any kind of residual income from it? |
| 03:55.02 | mrdigital | 2 lines |
| 03:55.24 | [TK]D-Fender | mrdigital, I typically suggest Linksys for small stuff, and Mediatrix/audioCodes for the bigger implementations |
| 03:55.33 | mrdigital | linksys |
| 03:55.34 | mrdigital | hmmm |
| 03:55.38 | mrdigital | which linksys ata you recommend? |
| 03:55.40 | ManxPower | aptura: I have two regular clients, they always want some sort of changes that are more than their onsite person can do |
| 03:55.50 | [TK]D-Fender | mrdigital, As always it depends on your NEEDS. |
| 03:56.14 | ManxPower | aptura: I've been a consultant for my largest customer for almost 11 years |
| 03:56.19 | danp | i've been pretty happy with the pap2 but i'm just using it for testing at the moment |
| 03:56.33 | mrdigital | well i need a cheap way to get 2 PSTN lines into asterisk |
| 03:56.48 | [TK]D-Fender | ~ygwypf |
| 03:56.58 | jbot | ygwypf is probably You Get What You Pay For. If the sole factor in your decision to purchase a product or service is that it's cheaper than everything else out there, don't be surprised if it's also worse in every other respect than everything else out there. |
| 03:56.59 | ManxPower | I stick with T-1 cards, Adtran TS750 channel banks, Tellabs EC, Polycom phones, and PRI lines |
| 03:56.59 | aptura | Manx thats really good track record |
| 03:57.21 | mrdigital | ?? |
| 03:57.37 | danp | ManxPower: is that adtran model number right? i don't get much from google |
| 03:57.55 | ManxPower | aptura: I've been with my 2nd client for something between 3 and 5 years, I don't recall exactly. |
| 03:58.01 | brian | Okay so why isn't * listening to telephone events from my SIP to SIP call :( |
| 03:58.14 | ManxPower | the other 2 or 3 occasional customers I've been doing consulting for for at least 13 years |
| 03:58.37 | brian | I'm just trying to test using SIP to SIP and it's not working :( |
| 03:58.54 | ManxPower | I'm open to accepting new clients, if their needs are a good match for my skills |
| 03:59.10 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 03:59.28 | danp | TA750 maybe |
| 03:59.50 | ManxPower | Total Access 750 |
| 03:59.55 | danp | yeah, got it. thanks |
| 04:00.01 | brian | ManxPower: why would DTMF not work on a sip to sip call if both clients are using the same dtmf mode? |
| 04:00.02 | Qwell | Nugget: $80k |
| 04:00.04 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 04:00.05 | Qwell | Nugget: I'll do it |
| 04:00.09 | ManxPower | Nugget: I'm sorry, but you have failed the customer interview. |
| 04:00.09 | danp | i'm using rhino cb24's and they seem pretty cool |
| 04:00.11 | [TK]D-Fender | brian, what DHOUL it be listening for? typically # just steps out of the way on a normal bridged call |
| 04:00.15 | [TK]D-Fender | SHOULD* |
| 04:00.15 | jnc | clone x100p cards = tears of sadness |
| 04:00.20 | Nugget | and we don't need to buy a server. I've got a bunch of old eMachines celeron boxes in the back room. |
| 04:00.44 | [TK]D-Fender | Nugget, go back to using your telnet! |
| 04:00.44 | Nugget | telnet is eeeeeeevil! |
| 04:00.45 | brian | [TK]D-Fender: It's not a bridged call |
| 04:00.51 | ManxPower | I have to deal with analog lines tonight. *soib* |
| 04:00.55 | Nugget | oof |
| 04:00.59 | brian | [TK]D-Fender: Well, I guess it is bridged. |
| 04:01.02 | [TK]D-Fender | brian, and what do you think * is doing in the middle? |
| 04:01.18 | aptura | convince me to not buy a snagoma if the digium cards have no more issues ;) |
| 04:01.22 | brian | I'm using Gizmo Project to test. |
| 04:02.02 | [TK]D-Fender | brian : "Your failure is complete." - Emperor Palpatine. |
| 04:02.20 | brian | [TK]D-Fender: your mean |
| 04:03.44 | [TK]D-Fender | brian, I prefer the term "realist", thank-you. You are picking every single point of failure and realizing its maximum potential. What you have to ask yourself is WHY. Why are you doing everything the HARD way? WHY is ti you can't jsut follow normal methods and advice? |
| 04:04.34 | [TK]D-Fender | brian, While you're at it, you should try reviving chan_modem and bridge it with H323! there's something I haven't heard you propose yet! |
| 04:04.57 | mrdigital | [TK]D-Fender: spa-3000 ? |
| 04:05.09 | [TK]D-Fender | mrdigital, What is the point of these analog lines? |
| 04:05.10 | mrdigital | nm im better off with the 3102 lol |
| 04:05.13 | brian | [TK]D-Fender: Everything works except for DTMF and I think it's a problem with Gizmo Project. |
| 04:05.16 | mrdigital | i dont know |
| 04:05.26 | [TK]D-Fender | mrdigital, NO COMMENT |
| 04:05.44 | mrdigital | what voip provider do you recommend? |
| 04:05.50 | ManxPower | I wish I understood more about the tests this device does |
| 04:06.16 | [TK]D-Fender | ManxPower, which? |
| 04:06.20 | ManxPower | aptura: standardize on one vendor. Pick one, stay with it. |
| 04:06.31 | danp | vitelity outbound still isn't working for me...something strange is afoot |
| 04:06.43 | ManxPower | [TK]D-Fender: Sage Instruments 930A Communications Test Set |
| 04:06.43 | [TK]D-Fender | danp, Amputate! |
| 04:06.52 | [TK]D-Fender | ManxPower, Still working on that hum? |
| 04:07.14 | ManxPower | Tests: C-MSG, C-NOTCH, 3K FLT, and S/N |
| 04:07.53 | ManxPower | [TK]D-Fender: I've been busy getting the small cabin into a state where I could move into it, i've not had time to deal with the hum |
| 04:08.43 | [TK]D-Fender | ManxPower, downsizing and moving? |
| 04:08.53 | brian | [TK]D-Fender: ugh |
| 04:09.01 | brian | This is so stupid. |
| 04:09.17 | brian | There is no reason why DTMF shouldn't work. |
| 04:09.49 | [TK]D-Fender | brian, get a real SIP client, do a REAL test, install * NORMALLY. |
| 04:10.02 | brian | I'm trying X-Lite for the sip client now |
| 04:10.23 | brian | It is installed "normally" |
| 04:10.46 | brian | I just don't have all the modules loaded, I only have the ones I need loaded. |
| 04:10.50 | ManxPower | [TK]D-Fender: part of the overall plan for building out the cabins |
| 04:10.55 | [TK]D-Fender | do a dialplan test. and by normal I mean not trying to load 3 modules that you think are the only ones absolutele necessary. |
| 04:11.37 | [TK]D-Fender | ManxPower, for you to live in or jsut a client? |
| 04:11.37 | brian | [TK]D-Fender: It's more like 9 modules. |
| 04:11.59 | ManxPower | [TK]D-Fender: for me. |
| 04:12.00 | [TK]D-Fender | brian, start with a better test using the dialplan. client-client is not a good test. |
| 04:12.03 | brian | Actually 13. |
| 04:12.19 | brian | [TK]D-Fender: client-client? |
| 04:12.41 | [TK]D-Fender | brian, what exactly are you doing to test this? |
| 04:12.44 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 04:12.56 | brian | [TK]D-Fender: Calling myself on SIP |
| 04:13.16 | [TK]D-Fender | brian, from what exactly, to what? |
| 04:13.25 | brian | Asterisk to Free World Dialup |
| 04:13.26 | aptura | Well have a good night. I cannot do anything with my bsd box as its compiling. |
| 04:13.32 | ManxPower | [TK]D-Fender: I figure if I just write down the tests the telco guy, who seems like he's been with the telco for YEARS will understand them |
| 04:14.05 | *** join/#asterisk BugKhaM (n=LAMER@ppp-58.8.1.86.revip2.asianet.co.th) |
| 04:14.08 | [TK]D-Fender | ManxPower, can't get him out there yourself huh? |
| 04:14.33 | brian | [TK]D-Fender: Asterisk to Free World Dialup |
| 04:14.44 | [TK]D-Fender | brian, please show a COMPLETE path for your test... |
| 04:15.00 | brian | [TK]D-Fender: Free World Dialup is registered with X-Lite... |
| 04:15.01 | ManxPower | [TK]D-Fender: I'm using a CLEC, but they have been pretty good with problem reports |
| 04:15.02 | coppice | ManxPower: far more people have these test sets than understand the results :-) |
| 04:15.26 | ManxPower | coppice: that is why I'm trying to understand the results |
| 04:16.11 | [TK]D-Fender | brian, that is not a very comprehensive description of your test or call-path |
| 04:16.47 | brian | [TK]D-Fender: ulaw codec |
| 04:17.11 | ManxPower | Brian: I don't see asterisk anywhere in the description |
| 04:17.29 | [TK]D-Fender | brian, I'd say getting warmer, but the fraction is SAD... |
| 04:17.34 | brian | Asterisk originates the call to FWD via SIP |
| 04:17.49 | [TK]D-Fender | brian, how is * originating a call? |
| 04:18.00 | ManxPower | coppice: I have the results for tests on an FXS calling an asterisk extension that just answers and waits, so I have tests for quietline on a known good connection at least |
| 04:18.06 | brian | manager originate command |
| 04:18.21 | [TK]D-Fender | Oh God.... |
| 04:18.22 | pfn | hmm, can I specify a different bindaddr per provider? |
| 04:18.26 | [TK]D-Fender | asfasdasd;akl;sjdkl;asudiou454uiop90-btrewbwret |
| 04:18.26 | Qwell | brian: and where is the other end of the call going? |
| 04:18.45 | brian | free world dialup's sip proxy |
| 04:18.52 | Qwell | the OTHER end |
| 04:19.01 | brian | x-lite |
| 04:19.12 | pfn | rather, a different bindaddr per sip.conf entry |
| 04:19.21 | Qwell | How? You just said xlite wasn't connected to asterisk |
| 04:19.28 | brian | It's not. |
| 04:19.33 | brian | It's connected to Free World Dialup. |
| 04:19.34 | [TK]D-Fender | pfn, not a bindaddr. "port=1234" |
| 04:19.38 | Qwell | so then how is one side going to FWD, and the other going to xlite? |
| 04:20.08 | brian | Because the Asterisk server is dialing a SIP URL with my FWD username? |
| 04:20.13 | [TK]D-Fender | qwell : AMI Originate -> * -> FWD -> X-Lite (reg'd to FWD) |
| 04:20.23 | Qwell | yes, but where is the *OTHER* side? |
| 04:20.29 | brian | What other side? |
| 04:20.33 | [TK]D-Fender | qwell, I have to fgriggen reverse ngineer 15 minutes of Q&A to reassemble that! |
| 04:20.45 | Qwell | What, is the call one-sided? |
| 04:20.45 | ManxPower | Brian: all calls have 2 legs |
| 04:20.52 | Qwell | xlite is talking to nothing? |
| 04:21.05 | brian | The other side of the call is at the FastAGI. |
| 04:21.09 | [TK]D-Fender | qwell ---> <[TK]D-Fender> qwell : AMI Originate -> * -> FWD -> X-Lite (reg'd to FWD) |
| 04:21.51 | Qwell | Do I *really* have to ask the next question? |
| 04:21.53 | ManxPower | Brian: We expect the minimum setup required to reproduce the problem. Now if you insist on making everything as complicated as you can then you will very quickly find that NOBODY wants to help you. |
| 04:21.58 | Qwell | it should be pretty obvious at this point |
| 04:22.11 | brian | Qwell: what |
| 04:22.18 | Qwell | and, with the expected answer, there would be about 6 followup questions |
| 04:22.22 | *** join/#asterisk AzianFlu (n=Yamabush@arcane/supporter/yamabushi) |
| 04:22.34 | brian | Qwell: The FastAGI does wait for digits |
| 04:22.45 | Qwell | okay, now answer the next question |
| 04:22.47 | brian | Qwell: but it doesn't work because * isn't hearing my DTMF |
| 04:22.49 | Qwell | (s) |
| 04:22.58 | brian | Qwell: not sure why |
| 04:23.11 | ManxPower | I've already decided that you are not worth helping. Soon Qwell and [TK]D-Fender will decide that too and then you'll be screwed. |
| 04:23.14 | Qwell | How is it waiting for digits? |
| 04:23.21 | Qwell | ManxPower: I decided that this morning |
| 04:23.23 | brian | the wait for digits AGI command |
| 04:23.37 | *** join/#asterisk mtaht4 (n=m@c-71-198-215-76.hsd1.ca.comcast.net) |
| 04:23.48 | ManxPower | Qwell: then stop helping him and save yourself pain |
| 04:23.51 | Qwell | and after it waits, what does it...do with them? |
| 04:23.56 | *** join/#asterisk [hC] (n=hardcore@adsl-63-200-45-107.dsl.snfc21.pacbell.net) |
| 04:24.31 | brian | Qwell: It doesn't receive any digit. That's the problem! |
| 04:24.41 | pfn | hmm, how can I treat multiple lines as one? like I used to do with, like, Zap/g2 |
| 04:24.42 | brian | Qwell: So it just times out. |
| 04:24.43 | Strom_M | I have the solution |
| 04:24.45 | Strom_M | you need..... |
| 04:24.47 | Strom_M | CATSEX |
| 04:24.51 | Qwell | and dogballs |
| 04:24.53 | pfn | how can I use a single Dial command to reference SIP/provider1 and SIP/provider2 |
| 04:24.59 | Qwell | pfn: & |
| 04:25.05 | [TK]D-Fender | pfn, you DON"T. |
| 04:25.06 | Qwell | SIP/foo&SIP/bar |
| 04:25.12 | pfn | Qwell, not what I want |
| 04:25.14 | pfn | that dials *both* |
| 04:25.19 | Qwell | oh, then...what? |
| 04:25.20 | pfn | I want it to try to grab one of them only |
| 04:25.26 | Qwell | You don't |
| 04:25.26 | pfn | akin to Zap/g1 would do |
| 04:25.28 | [TK]D-Fender | pfn, dial 1, then dial the next on the next priority |
| 04:25.28 | pfn | for group 1 |
| 04:25.29 | brian | fine I guess i'm on my own :( |
| 04:25.45 | Qwell | brian: unless you can simplify the hell out of your setup - yes |
| 04:25.45 | [TK]D-Fender | pfn, there ARE no groups except for Zap channels |
| 04:25.56 | [TK]D-Fender | pfn, , because * has no way to know what is SUABLE. |
| 04:25.59 | [TK]D-Fender | USABLE |
| 04:26.07 | Qwell | all ITSPs are sueable ;) |
| 04:26.07 | brian | Qwell: It's not even complicated. |
| 04:26.16 | [TK]D-Fender | brian, get your sorry ass out of AGI and prove the BASICS work. |
| 04:26.21 | pfn | [TK]D-Fender, that's why I want it to call one after the other |
| 04:26.29 | pfn | instead of having to do it in my dialplan everytime I want such functionality |
| 04:26.34 | brian | [TK]D-Fender: The basics don't work. |
| 04:26.35 | [TK]D-Fender | pfn, : god then I've already ahnded you the solutions |
| 04:26.42 | [TK]D-Fender | brian : SHOW US |
| 04:26.58 | pfn | [TK]D-Fender, I don't really consider that you handing me the solution--I already know that :p |
| 04:27.01 | brian | [TK]D-Fender: Are you one of those weirdos that swears by writing all logic in the diaplan |
| 04:27.30 | Qwell | brian: anything (literally) can be done in dialplan |
| 04:27.52 | [TK]D-Fender | brian, Every single one of my installations work. the FIRST time. I consult in it. |
| 04:27.59 | brian | AGI does the exact same thing the dial plan does it just lets you use your favorite programming language to do it. |
| 04:28.05 | [TK]D-Fender | brian, You come here and NOTHING works for you. Do the math |
| 04:28.08 | Qwell | No it doesn't |
| 04:28.20 | Qwell | AGI adds at least TWO layers of abstraction |
| 04:28.21 | [TK]D-Fender | brian, It has to work YOUR way because you WANT it to. |
| 04:28.47 | Qwell | manager is a third |
| 04:28.48 | [TK]D-Fender | brian, no more excuses! You have had us running in circles over this an we all tire of it. |
| 04:28.51 | Qwell | and FWD is a fourth |
| 04:28.54 | brian | Just forget it I'll figure it out myself like I always do |
| 04:29.14 | [TK]D-Fender | brian, We cannot teach you. Your head is too full. |
| 04:29.32 | pfn | what's brian trying to do... |
| 04:29.42 | brian | The problem is that you people are always asking questions that are irrelevant |
| 04:29.53 | [TK]D-Fender | pfn, spare yourself, and us the reiteration of hours of pointlessness |
| 04:30.00 | ChkDigit | Wow. I just about thought I couldn't do something in Asterisk. |
| 04:30.11 | brian | I'm done. |
| 04:30.18 | [TK]D-Fender | brian, we who have been using * for years, and Qwell who WORKS FOR DIGIUM AND CODES FOR * beg to differ! |
| 04:30.37 | brian | You haven't told me how to debug this at all. |
| 04:30.39 | Qwell | and I've been right at least once today |
| 04:30.42 | brian | You keep asking me retarded questions. |
| 04:30.55 | Qwell | after you refused for nearly an hour to upgrade |
| 04:31.08 | ManxPower | [TK]D-Fender: I assume Qwell is not here in his Digium capacity at the moment |
| 04:31.18 | brian | I did upgrade |
| 04:31.24 | pfn | Qwell, only right once? |
| 04:31.25 | pfn | heh |
| 04:31.27 | Qwell | and it fixed the problem you were having |
| 04:31.35 | [TK]D-Fender | ManxPower, I am merely attesting to your collective experience. |
| 04:32.08 | brian | Qwell: You guys keep asking me the same questions over and over again. |
| 04:32.22 | [TK]D-Fender | brian, Again, your head is too full. You have all the answers. Our advice can do you no good and you will not attempt it. We can do nothing for you because you will not listen and follow. |
| 04:32.42 | brian | You never suggested anything |
| 04:33.00 | brian | And lots of time people in here will tell you to do all sorts of nonsense that doesn't have anything to do with anything. |
| 04:33.24 | etfonhomey | I just want to take this opportunity to thank "brian" for this entertainment. |
| 04:33.30 | [TK]D-Fender | brian. Go into extensions.conf and make an IVR and use the Read command. Show us CLI output of a failed call at high-verbose and SIP debug enabled. have you done any of this and shown us ANYTHING? NON. You have sat here WHINING. |
| 04:33.44 | ManxPower | I give up. |
| 04:33.50 | brian | I already told you what is happening |
| 04:33.58 | brian | You guys just refuse to read what I wrote |
| 04:33.58 | *** part/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com) |
| 04:34.24 | [TK]D-Fender | brian. I Don't trust your setup for shit. You are not listening. Do our test or simply go away. we want OUTPUT. |
| 04:34.57 | [TK]D-Fender | "I am using FastAGI and DTMF isn'g working, why?" doesn't say ANYTHING. |
| 04:35.15 | brian | Okay so I give you output and it's going to be for no reason. |
| 04:35.28 | brian | Because I know how to read debug messages, I'm not fucking stupid. |
| 04:35.56 | etfonhomey | [TK]D-Fender: Just think, with all the effort you've put into brian, you could have helped me to get SLA working!! |
| 04:35.59 | [TK]D-Fender | brian, Do you point a finger at your car an d ask the mechanic why it isn't working without letting him SEE it? |
| 04:36.10 | brian | And I know that Asterisk isn't showing that a key is pressed |
| 04:36.24 | [TK]D-Fender | etfonhomey, Actually... i've been reading over the docs and they are confusing like heck! |
| 04:36.34 | [TK]D-Fender | brian SHOW US |
| 04:36.55 | brian | I'm beginning to think the more I'm in this channel that not even you guys in here understand how asterisk works |
| 04:36.56 | [TK]D-Fender | brian, why haven't we seen your suip.conf settings either? |
| 04:37.07 | *** mode/#asterisk [+b %brian!*@*] by Qwell |
| 04:37.10 | etfonhomey | [TK]D-Fender: You can say that again. I want to do it using a SIP trunk and it's basically leaving that out. |
| 04:37.25 | [TK]D-Fender | <[TK]D-Fender> brian, Do you point a finger at your car an d ask the mechanic why it isn't working without letting him SEE it? |
| 04:37.29 | [TK]D-Fender | ^^^ ANSWER |
| 04:37.35 | Qwell | I'm done. |
| 04:37.39 | Qwell | so is he |
| 04:37.46 | etfonhomey | Did you really ban him? |
| 04:37.54 | Qwell | +q <3 |
| 04:38.47 | etfonhomey | I gotta get a transcript of this for when I have a bad day. I can read it and cheer myself up. |
| 04:40.05 | pfn | hmm, what's the 'a' extension? sample std-exten macro has a catch a |
| 04:40.18 | Qwell | pfn: voicemail exit with * (or 0?) |
| 04:40.20 | etfonhomey | In honor of "The Apprentice" tonight, you should have said, "brian, you're fired!" and then banned him. |
| 04:40.37 | Qwell | I always forget which |
| 04:40.38 | pfn | Qwell, thanks |
| 04:40.55 | Qwell | I think o is 0, and a is * |
| 04:40.55 | *** join/#asterisk annonimous (n=annonimo@189.136.64.167) |
| 04:40.56 | Strom_M | * is a, 0 is o |
| 04:41.18 | Strom_M | today's episode is brought to you by the letter +q |
| 04:41.35 | annonimous | hello hello |
| 04:41.45 | coppice | anyone here use xen with centos? |
| 04:42.34 | tzafrir_laptop | coppice, 4 or 5? |
| 04:42.55 | pfn | can a macro force an exten to "return"? e.g. 1234 => { Command1(); Command2(); &continueOrStop(); Command3(); } |
| 04:43.07 | pfn | can the macro force Command3 to stop? |
| 04:43.11 | pfn | er, not occur |
| 04:43.52 | tzafrir_laptop | pfn, which macro? running where? |
| 04:44.08 | pfn | tzafrir_laptop, in my example 1234 exten above |
| 04:44.15 | coppice | tzafrir_laptop: I'm not sure, but it is running kernel 2.6.16.26-xen, and twice in the last few weeks it has stopped talking to the ethernet port, while the other virtual servers on the same box (Debian) have had no trouble |
| 04:45.24 | tzafrir_laptop | sorry, can't help here... |
| 04:46.21 | annonimous | hi i have a doubt where can i found a list of the commands or variables of asterisk? cause i need to setup a sip trunk and need to setup a little register expires.... where can i found a manual cause google tellme but of spa-3000 not asterisk directly |
| 04:48.14 | AzianFlu | annonimous, i found a good beginners manual. google for Asterisk : The Future of Telephony . I believe it is still availiable as a free E-Book from O'Reilly. |
| 04:48.20 | Strom_M | ~book |
| 04:48.25 | jbot | from memory, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 04:48.39 | AzianFlu | or you can look there! :) |
| 04:48.47 | *** join/#asterisk BBHoss (n=hoss@65.4.28.144) |
| 04:49.25 | annonimous | AzianFlu, ok ill check it cause i want to ask when i got a difficult scenario not for begginner stuff (blushes) |
| 04:49.56 | *** join/#asterisk rrrobert (n=rrobert@58-65-160-140.nayatel.pk) |
| 04:50.00 | *** join/#asterisk axisys (n=axisys@c-69-143-190-152.hsd1.va.comcast.net) |
| 04:50.47 | *** join/#asterisk Mavvie (n=edwin@ppp30-29.lns1.syd6.internode.on.net) |
| 04:51.37 | annonimous | by the way anybody of mexico or anybody who knows the famous r2 modified of mexico? (a kind of e1/t1 but some ppl tells that i need the unicall library....) |
| 04:52.08 | pfn | where's a good definition of the DIALSTATUS? |
| 04:52.09 | [TK]D-Fender | annonimous, that sounds like MRH/R2 (I beliece its called). |
| 04:52.20 | Qwell | pfn: core show application dial |
| 04:52.33 | pfn | Qwell, sorry, what each DIALSTATUS means |
| 04:52.42 | pfn | TORTURE, DIALCANCEL, etc. |
| 04:52.43 | coppice | annonimous: yes, you need the unicall libraries |
| 04:52.45 | Qwell | it says |
| 04:52.51 | Qwell | or, I thought it did |
| 04:52.56 | annonimous | [TK]D-Fender, yep thats it |
| 04:53.25 | annonimous | coppice, and in isdn format? or its the same thing cause some cases come in coaxial cable... |
| 04:53.35 | pfn | Qwell, it kinda does, I have no idea what the 'torture' or 'go away' scripts are |
| 04:54.32 | coppice | you don't need unicall for ISDN. you need it for MFC/R2. An E1 can be of either type, but MFC/R2 is much more common in mexico. if ISDN is available, use that |
| 04:55.17 | annonimous | coppice, ah i see , im trying to get relaxed, the isdn its not equal to an r2 modified and with isdn i wont need unicall only my te right?? |
| 04:56.04 | coppice | ISDN and MFC/R2 are two totally different signalling scheme, which both work over E1 lines |
| 04:56.09 | *** join/#asterisk mrichmanM (n=richmanm@c-24-20-124-243.hsd1.mn.comcast.net) |
| 04:56.36 | coppice | ISDN is the better choice. most people in south american can only get MFC/R2, though |
| 04:56.46 | *** join/#asterisk brimstone (n=brimston@pdpc/sponsor/digium/brimstone) |
| 04:56.56 | Qwell | brimstone: !!! |
| 04:57.08 | annonimous | coppice, ah i see , no no my provider asked for the format, rj45, g711 (i dont remmeber that format) or r2 |
| 04:59.58 | *** join/#asterisk mtaht4 (n=m@c-71-198-215-76.hsd1.ca.comcast.net) |
| 05:01.15 | *** part/#asterisk brimstone (n=brimston@pdpc/sponsor/digium/brimstone) |
| 05:06.15 | J4k3 | hehe |
| 05:14.47 | BBHoss | any way to cheaply integrate a SpectraLink system with *? |
| 05:15.31 | *** mode/#asterisk [-b %brian!*@*] by Qwell |
| 05:15.33 | *** mode/#asterisk [+b brian!*@*] by Qwell |
| 05:15.38 | *** kick/#asterisk [brian!n=north@pdpc/sponsor/digium/Qwell] by Qwell (Qwell) |
| 05:19.17 | annonimous | well see ya later pals |
| 05:20.00 | Qwell | BBHoss: the SIP+DECT phones? If so, it's just SIP and should work fine |
| 05:20.34 | BBHoss | well, wifi is a bigger hassle than normal wireless |
| 05:20.48 | Qwell | You must mean different phones then |
| 05:20.58 | Qwell | the new ones are pretty sweet :D |
| 05:21.12 | BBHoss | are they wifi? |
| 05:21.17 | Qwell | wired SIP base station, with DECT handsets |
| 05:21.25 | BBHoss | hmm |
| 05:21.32 | BBHoss | link? |
| 05:21.37 | *** join/#asterisk deoptima (n=deoptima@c-69-243-207-18.hsd1.tn.comcast.net) |
| 05:21.40 | BBHoss | KIRK DECT? |
| 05:21.47 | Qwell | should be able to see the press release on google news |
| 05:21.58 | Qwell | should be the first hit for spectralink+digium |
| 05:22.10 | BBHoss | polycom bought them now, so who knows whats next |
| 05:22.21 | Qwell | oh? |
| 05:22.30 | *** join/#asterisk aptura (n=root@S010600a0c93f6f7e.vs.shawcable.net) |
| 05:22.40 | BBHoss | its on thier site |
| 05:22.44 | Qwell | ahh |
| 05:24.42 | *** join/#asterisk aptura (n=lork@S010600a0c93f6f7e.vs.shawcable.net) |
| 05:28.23 | BBHoss | are there any other solutions out there that do seamless transitioning |
| 05:28.30 | BBHoss | or have repeaters? |
| 05:31.10 | BBHoss | anybody else that does DECT other than ATS |
| 05:31.15 | pfn | <PROTECTED> |
| 05:31.20 | *** join/#asterisk _pepo_ (n=pepo@201.238.188.166) |
| 05:31.32 | _pepo_ | hi friends |
| 05:31.45 | pfn | e.g., I have register => 12345:password@192.168.1.1, I want it to send From: <12345@mydomain.com> |
| 05:32.13 | pfn | s/want/need |
| 05:33.50 | [TK]D-Fender | pfn, tried setting "domain=" in [general]? |
| 05:34.07 | pfn | [TK]D-Fender, I don't want it to affect everything, I just want it to register with that domain for that particular provider |
| 05:35.16 | [TK]D-Fender | pfn, yeah, I really don't see an option for parameters for that. IMO "register" should be a kind of entry just like peers & users to account for the number of parameters that are involved including forcing re-registration frequency, etc... |
| 05:35.46 | pfn | indeed |
| 05:35.54 | pfn | I remember broadvoice having a similar issue... |
| 05:36.50 | [TK]D-Fender | pfn, can't say I've encounted it with the one client of mine who used them... |
| 05:44.20 | *** join/#asterisk zeeesh (i=zeeesh@202.38.55.125) |
| 05:44.21 | zeeesh | hi |
| 05:45.24 | _pepo_ | How do I connect my asterisk with the local PSTN? Is there some way to use just my unique line or I need necessarily an E1 or T1? |
| 05:46.40 | [TK]D-Fender | _pepo_, You can use all sort of different technologies to connect to the PSN, and each with a number of hardware solutions. |
| 05:46.56 | [TK]D-Fender | _pepo_, What do you ahve by way of lines right now? |
| 05:47.24 | *** join/#asterisk fnordus (n=dnall@24.85.128.203) |
| 05:47.45 | _pepo_ | I've configured Asterisk for my LAN, but now I wanna to call to numbers in the local PSTN |
| 05:49.10 | [TK]D-Fender | _pepo_, and the answer to my question would be....? |
| 05:49.43 | _pepo_ | sorry, now I just have one common line |
| 05:49.50 | ChkDigit | Is presence and Polycom phones like the IP 601 still a problem? |
| 05:50.11 | ChkDigit | _pepo_ And, what hardware connects that line to your Asterisk box? |
| 05:51.02 | _pepo_ | nothing, just my modem to access ;) |
| 05:51.07 | [TK]D-Fender | ChkDigit, he doesn't have any yet from the sound of things. |
| 05:51.21 | ChkDigit | _pepo_:You need an ATA. |
| 05:51.23 | [TK]D-Fender | _pepo_, I've asked TWICE : why kind of line(s) do you have NOW? |
| 05:51.48 | [TK]D-Fender | ChkDigit, And ATA by raw definition is for analog PHONES, not LINES. |
| 05:52.02 | [TK]D-Fender | ChkDigit, No, Presence has been jsut fine on Polycom for a long time. |
| 05:52.14 | *** join/#asterisk antlers (n=antlers@ip70-173-90-39.lv.lv.cox.net) |
| 05:53.01 | _pepo_ | an ATA just connect the phones to my network. now i am using softphones. But, how do I make a call to a number in the PSTN |
| 05:53.23 | ChkDigit | You need an FXO ATA like: http://www.digium.com/en/products/hardware/tdm400p.php |
| 05:53.36 | rudholm | or an SPA-3000 |
| 05:53.54 | [TK]D-Fender | ChkDigit, That card is also not at all an ATA. |
| 05:54.05 | ChkDigit | <[TK]D-Fender> Yes, I'm assuming if he hooks a modem to it, it is an analogue line. |
| 05:54.11 | [TK]D-Fender | ChkDigit, you really need to get your terminology right :) |
| 05:54.20 | [TK]D-Fender | ChkDigit, ASSuming is correct :) |
| 05:54.31 | ChkDigit | Wooo. Cheeky! |
| 05:54.31 | [TK]D-Fender | ChkDigit, ASK, don't ASSume :) |
| 05:54.55 | rudholm | you could argue that a card in a computer is an ATA |
| 05:54.58 | [TK]D-Fender | _pepo_, So what kind of line(s) EXACTLY do you have? |
| 05:55.10 | ChkDigit | [TK]D-Fender: If he can't answer what kind of lines he has, he probably has an analogue line. |
| 05:55.19 | [TK]D-Fender | rudholm, some people like arguing to no real end :) |
| 05:55.33 | rudholm | you among them, clearly :-P |
| 05:55.44 | [TK]D-Fender | rudholm, Oh... you saw that earlier did you? ;) |
| 05:56.11 | rudholm | I've seen you do it multiple times :-P |
| 05:56.11 | [TK]D-Fender | rudholm, I prefer to think of it as "hold out a hope" |
| 05:56.12 | [TK]D-Fender | rudholm, so often disappointed I am |
| 05:56.38 | rudholm | [TK]D-Fender: btw, I can tell you how to make a silent ring cadence now :-P |
| 05:57.08 | [TK]D-Fender | rudholm, Ok... I think you have elevated your "cool factor" on that alone :) Do post it ont he WIKI |
| 05:57.08 | _pepo_ | Ok, first my defense :) I've configured all the telephony enviroment in my LAN using TRIXBOX and the extensions can make call among they. |
| 05:57.15 | _pepo_ | But now I wanna send calls to the PSTN using my NORMAL-ANALOG telephone line |
| 05:57.51 | [TK]D-Fender | _pepo_, aside fromt he fact we should be assassinating you on your use of Trixbox, how about you answer the question I've asked THREE TIMES now? :) |
| 05:58.06 | [TK]D-Fender | OMG, a normal analog phone line! |
| 05:58.19 | [TK]D-Fender | hallelujah! |
| 05:58.35 | _pepo_ | sorry [TK]D-Fender, my engish is not so good :) |
| 05:59.26 | [TK]D-Fender | _pepo_, Ok, on the cheap side you can get an X100P or compatabile card for about $20 on e-bay, a Sipura-Linksys SPA-3XXXX series ATA/Gateway for about $90, , ot something bigger if you require more lines. |
| 05:59.26 | tzafrir_laptop | _pepo_, what adapter do you use to connect Asterisk to that line? |
| 05:59.51 | [TK]D-Fender | tzafrir_laptop, he has none. He's inquiring about what hardware he can USE to do so |
| 06:00.31 | _pepo_ | I dont have adapters now... that is my next step |
| 06:01.20 | [TK]D-Fender | _pepo_, I have just given you 2 inexpensive solutions, netx on the list is the Digium TDM400P or Sangoma A200d if you are thinking "serious" |
| 06:01.49 | _pepo_ | When I obtaing that hardware, the use is transparent or i have to configure something with my telephone company ? |
| 06:02.21 | _pepo_ | I am thinking about 10-15 terminals |
| 06:02.29 | [TK]D-Fender | _pepo_, Typically you only need to plug it in and configure on your end. |
| 06:02.57 | [TK]D-Fender | _pepo_, the SPA-based solution will take longer to figure out. The PCI based solutions will be quicker. |
| 06:03.53 | [TK]D-Fender | rudholm, Get WIKI-ing! |
| 06:04.25 | [TK]D-Fender | rudholm, yours was a very unque question when you started that... this is something very much worth sharing... |
| 06:04.44 | ChkDigit | Anyway, if someone can point me in the right direction with presence... |
| 06:05.27 | ChkDigit | I've got hints setup, buddies watched, * Business Ed. running, but the lamps will not change on the IP601 with attendants... |
| 06:05.46 | ChkDigit | .. sorry, attendant modules... |
| 06:07.35 | [TK]D-Fender | "Popular Mechanics has a very cool video and report about test-driving Hybrid Technologies' L1X-75, a battery powered, 600-hp, carbon-fiber roadster that pulls zero-60 in about 3.1 seconds, and tops out at 175 mph.<--- yay instant *gulp* ! |
| 06:08.09 | [TK]D-Fender | ChkDigit, what ver of * exactly, and you need to be running SIP 1.6.6 or higher ont he phone |
| 06:08.56 | _pepo_ | Ok a lot of thanks, now I have real information and knowledge ... something more: Which ATAs I can buy (inexpensive but goods) |
| 06:09.30 | ChkDigit | Asterisk Business Ed. ABE-B.1-3, phones are on SIP 2.0.1.0291 |
| 06:13.54 | [TK]D-Fender | ChkDigit, I believe there was a bug inst specific revision. I'd suggest you upgrade to 2.3B in that family |
| 06:16.38 | *** part/#asterisk aptura (n=lork@S010600a0c93f6f7e.vs.shawcable.net) |
| 06:17.04 | [TK]D-Fender | 2.0.3.B |
| 06:17.14 | *** join/#asterisk metztlixictli (n=metztlix@189.131.28.54) |
| 06:17.49 | metztlixictli | speak spanish? |
| 06:18.12 | metztlixictli | Strom_C, hi |
| 06:18.41 | [TK]D-Fender | ChkDigit, I could have said jump right to 2.1.0 ;) |
| 06:18.59 | [TK]D-Fender | ChkDigit, but I'm looking to maintain your config compatability as much as posible |
| 06:20.55 | ChkDigit | Yeah, Thanks. I have not looked at the config changes from 2.0 to 2.1... |
| 06:27.00 | rudholm | Strom_M: hi |
| 06:27.06 | rudholm | Strom_C: hi |
| 06:27.24 | Strom_M | rudholm: omg hi |
| 06:27.26 | Strom_M | rudholm: omg hi hi |
| 06:28.13 | rudholm | Strom_M: mobile? |
| 06:28.19 | Strom_M | macbook |
| 06:28.23 | rudholm | ah |
| 06:28.30 | rudholm | C? |
| 06:28.43 | Strom_M | desktop |
| 06:29.11 | rudholm | no IRC from the smartphone? |
| 06:30.47 | Strom_M | i can |
| 06:30.49 | Strom_M | just not right now |
| 06:35.22 | zeeesh | i have 2 peers, peer 100 at machine1 and peer 200 at machine2 ,,, how can both peers comunicate without using iax ,,, can sip make a call ???? |
| 06:37.23 | tengulre | anybody using asterisk behind firewall? |
| 06:42.44 | ChkDigit | tengulre: Sure. The question is, are you wanting to connect to things outside the firewall, or have things outside the firewall connect to you? |
| 06:46.03 | [TK]D-Fender | tengulre, you'll get the same answer as before... you the ports you need to allow, you jsut need to figure out iptables to DO it. |
| 06:47.33 | rudholm | it's fairly trivial to do in iptables |
| 06:48.02 | [TK]D-Fender | zeeesh, Sure |
| 06:50.44 | Strom_M | http://www.asciiartfarts.com/20060609.html |
| 06:50.56 | *** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com) |
| 06:51.07 | ManxPower | Is it safe yet? |
| 06:53.28 | Strom_M | sexe de chat |
| 06:57.32 | [TK]D-Fender | strom_M "Sexe du chat" :) |
| 06:58.06 | ManxPower | I think I just heard the on-hook MWI burst on a line |
| 06:59.46 | ManxPower | Strom_M: You are a PSTN guy. Do you think the telco would fix a line that when calling a 1004Hz @ 0dBm gets the tone at -13dBm? |
| 07:00.57 | Strom_M | hmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmm |
| 07:01.00 | ManxPower | Sorry -10 not -13 |
| 07:01.14 | Strom_M | how long is the loop? |
| 07:01.19 | ManxPower | according to Sandman.com they will fix it if it is lower than -8.5 |
| 07:01.20 | rrrobert | hello people |
| 07:01.47 | ManxPower | Strom_M: 11 miles from the CO, but there's a SLIC96 a couple of miles away that the line goes thru |
| 07:01.56 | Strom_M | ManxPower: yeah, id kvetch |
| 07:02.01 | ManxPower | loop voltage seems OK -47v |
| 07:02.02 | rrrobert | I am (still) looking for a (nice) CRM to complement an Asterisk setup (it should be open source and meant for call centers) |
| 07:02.40 | rudholm | I think you could get them to come out and see if the line is within spec. of course calling and saying "I'm seeing -10dBm on my milliwatt test" might confuse them, I'd just say "the volume seems too low, even on a known-good phone" |
| 07:02.55 | ManxPower | Strom_M: I'm hoping that if they pump up their gain I can lower my gain and get a net gain in the S/N ratio with regards to the hum, which I'm still having issues figuring out. |
| 07:03.27 | zeeesh | <PROTECTED> |
| 07:03.28 | [TK]D-Fender | ManxPower, Welcome to "loaded circuit"-ville..... |
| 07:03.59 | [TK]D-Fender | zeeesh, You need a zaptel timing source to use "trunk=yes" for IAX2 |
| 07:05.34 | Strom_M | i think rudholm has more experience with long loops than I do ;) |
| 07:05.34 | ManxPower | FXS -> Asterisk Answer/Wait = no measurable noise. PSTN DEMARC to telco quiet line = 69-111Hz @-59dBm, FXS -> Asterisk -> telco (same lines as measured before) call to telco quiet line is 512 to 616Hz @ -43.7dBm IT does not make any sense |
| 07:05.42 | rrrobert | zeeesh, I heard its very hot in Lahore |
| 07:06.31 | ManxPower | the 69-111Hz doesn't worry me but something weird is going on with the FXS->FXO |
| 07:06.54 | ManxPower | Strom_M: rudholm is the IRC nick? |
| 07:07.13 | rudholm | yes |
| 07:07.44 | Strom_M | dear pacific bell: |
| 07:07.49 | Strom_M | please install my isdn soon |
| 07:07.51 | Strom_M | love, strom |
| 07:07.52 | rudholm | I have a long loop, plugged into a TDM400, but I don't have noise problems, I have echo problems. |
| 07:07.55 | rudholm | heh |
| 07:07.59 | rudholm | yeah, get that working! |
| 07:08.07 | rudholm | I'm gonna switch, even if it means giving up CNAM |
| 07:08.21 | rudholm | my roommate is all "the phone is too echo-y" |
| 07:08.22 | rudholm | and he's right |
| 07:08.23 | ManxPower | rudholm: I'm using an Adtran TA750 for the POTS connections |
| 07:08.35 | rudholm | ah |
| 07:08.41 | ManxPower | rudholm: what about HPEC? |
| 07:08.50 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 07:09.26 | rudholm | I'm just using the software EC |
| 07:09.32 | rudholm | tried adjusting it |
| 07:09.39 | rudholm | getting levels and tail right |
| 07:09.42 | ManxPower | rudholm: the HPEC is free for analog cards under warrenty |
| 07:09.53 | rudholm | got echo down a bit, but not eliminated |
| 07:10.21 | ManxPower | for me the HPEC eliminated ALL echo. Of course it also eliminated ALL AUDIO, but I'm sure they will fix that. |
| 07:10.26 | Strom_M | hahahaaha |
| 07:10.27 | rudholm | hahaha |
| 07:10.34 | rudholm | I haven't experimented with it |
| 07:10.36 | rudholm | perhaps I should |
| 07:10.41 | rudholm | I'm tired of hearing callers |
| 07:10.44 | ManxPower | rudholm: you should. |
| 07:10.54 | rudholm | when I call Strom, connection is perfect |
| 07:11.03 | ManxPower | I also have a Tellabs EC card or 4 around here. |
| 07:11.18 | rudholm | but that's an IAX link direct between Asterisks |
| 07:12.01 | ManxPower | I should set up an IAX2 test number on my server, the one that is on satellite |
| 07:13.54 | rudholm | Strom_M: sometimes I dial into Phil's step switch just to hear it :) |
| 07:14.23 | rudholm | love that fog-horn busy signal :) |
| 07:14.27 | Strom_M | :) |
| 07:14.38 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 07:17.41 | ManxPower | Strom_M: I don't suppose you know off the top of your head the expected voltages for on hook and off hook top to grnd, ring to grnd and tip to ring? |
| 07:18.13 | Strom_M | tip to ring should be 48vDC, IIRC |
| 07:18.21 | Strom_M | dunno about ground |
| 07:18.41 | rudholm | tip and ground are close to the same potential, but not quite |
| 07:19.11 | rudholm | in fact, iirc, there's more potential between ring and ground than there is between tip and ring |
| 07:19.13 | *** join/#asterisk darken_darken (n=marco@240.129.77.83.cust.bluewin.ch) |
| 07:19.21 | rudholm | I just measured all the voltages on the other day |
| 07:19.28 | rudholm | er, other week, I guess |
| 07:19.57 | rudholm | lemme dig it up |
| 07:20.07 | ManxPower | ring to tip I get 48VDC, ring to grnd I get -46VDC, and tip to grnd I get 0VDC |
| 07:20.34 | rudholm | that sounds reasonable |
| 07:20.35 | ManxPower | I want to make sure my test equipment is not out of whack |
| 07:20.42 | rudholm | what kind of equipment? |
| 07:20.55 | ManxPower | Sage Instruments 930A |
| 07:21.25 | ManxPower | Got it off eBay about a month before Katrina. This is the first time I've needed it AND it was not in storage |
| 07:21.32 | rudholm | ok, here's what I found on my analog line |
| 07:22.19 | rudholm | these are all WRT earth |
| 07:22.22 | rudholm | On-Hook: Tip: +0.34 Ring: -51.8 (so my T-R potential is over 52V) |
| 07:22.39 | rudholm | Off-Hook: Tip: -21.7 Ring: -29.75 (for about 8V T-R potential) |
| 07:23.11 | ManxPower | close enough to what I get for both |
| 07:23.16 | Strom_M | http://www.asciiartfarts.com/20020805.html |
| 07:23.18 | rudholm | the on-hook potentials were what I expected, but I didn't realize that they both dip so much below earth when off-hook |
| 07:23.40 | ManxPower | BTW, I get different reports as to if POTS is 600 Ohm or 900 Ohm |
| 07:24.02 | rudholm | FWIW: Coin Return is dead-on -130VDC and Coin Collect is dead-on +130VDC (again, WRT earth) |
| 07:24.20 | rudholm | that surprised me, given that my loop length is like 26,000 feet |
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| 07:24.35 | rudholm | this is off a DMS-100 |
| 07:25.17 | rudholm | this is a coin line, I didn't test my other analog line, but I would assume it's basically the same, despite the different originating equipment for the coin line. |
| 07:25.19 | ManxPower | I guess my loop length is demarc to the SLIC96? |
| 07:25.25 | rudholm | yeah |
| 07:25.37 | ManxPower | that is only a couple of miles. |
| 07:26.04 | ManxPower | but more than you would expect since they routed my loop half away around the damn mountian |
| 07:26.13 | pfn | hmm, what's the point of ael hints? |
| 07:26.41 | rudholm | you live in some rural area? |
| 07:27.09 | ManxPower | According to the phone guy my loop comes from the SLIC96 a couple of miles away, PAST my driveway to several farms down the road, then back to me |
| 07:27.14 | ManxPower | rudholm: something like that |
| 07:27.42 | rudholm | well, I live 5 miles from downtown Los Angeles, and my loop is > 26,000 feet. |
| 07:27.49 | rudholm | Thanks, Pacific Telephone. |
| 07:28.05 | pfn | 5 miles ~ 25000ft.... |
| 07:28.13 | rudholm | I live across the street from an area code boundary |
| 07:28.26 | rudholm | if COs were cell towers, I'd be in a dead spot. |
| 07:28.32 | ManxPower | any idea what the cap kick would be expected per 1,000 ft of loop? |
| 07:28.58 | rudholm | no. I imagine it varies depending on cable type |
| 07:28.59 | *** join/#asterisk yidiyuehan (n=yidiyueh@58.185.253.70) |
| 07:29.05 | yidiyuehan | hi everyone |
| 07:29.31 | ManxPower | Nod. |
| 07:29.48 | ManxPower | didn't know if there was a standard for the cable type |
| 07:30.08 | rudholm | there may be. I don't know any rules of thumb for regular 22 and 24 AWG wire, though. |
| 07:30.28 | yidiyuehan | my zap line could not hang up on some cases, like i call in and one of the extension's follow me instead of the extensions, and then it rings, before it answers i hangup then it will continue to execute the folloing follow me list. anyone knows the problem? |
| 07:30.34 | rudholm | insulation makes a difference too, of course |
| 07:32.55 | ManxPower | Ah, I didn't know that |
| 07:33.51 | ManxPower | Well I'm pretty sure the telco will fix a 245 - 405Hz hum at the demarc, that is a start at least |
| 07:34.05 | rudholm | yeah |
| 07:34.07 | yidiyuehan | But if the line directs to zap extension instead of the follow me, then i can hang up properly |
| 07:34.36 | ManxPower | yidiyuehan: asterisk is not detecting when the telco hangs up the line. This could be cause by many things |
| 07:35.43 | yidiyuehan | yes, well, there isn't a zap card channel i guess.. the probelm is that i only could not hang up under some cases. |
| 07:37.05 | ManxPower | yidiyuehan: this is usually only a problem with pstn calls into voicemail, but I can see how it could be a problem with follow me type of things |
| 07:38.03 | yidiyuehan | yes, so i just reported here to check it could be a possible bug maybe. |
| 07:38.09 | ManxPower | rudholm: do you know anything about C-NOTCH tests and what an expect S/N would be? |
| 07:38.40 | ManxPower | I would like to state again just how much I dislike analog lines |
| 07:40.59 | rudholm | I don't know the C-Message noise standards for POTS lines, no. |
| 07:41.28 | rudholm | I'm sure they're in the Bell System Practices, though :) |
| 07:42.10 | rudholm | yes, I hate analog lines, too |
| 07:43.53 | rudholm | if you have a good filter, you could test your line, though |
| 07:44.01 | rudholm | would be interesting, at least |
| 07:44.28 | rudholm | you could fake a filter, if you have a decent sound card and can get your milliwatt tone into your computer |
| 07:45.29 | rudholm | record it for 10 seconds or so, then synthesize 10 seconds of 1004Hz tone with software, offset it 180 degrees from the recorded tone, sum it in, and you have a C-Notch test sample |
| 07:46.09 | rudholm | unless you have DSP ninja skillz and can write your own filter :) |
| 07:49.21 | *** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com) |
| 07:49.38 | ManxPower | my test set has a c-notch test |
| 07:49.46 | rudholm | oh sweet |
| 07:49.56 | rudholm | what did you pay for that thing? |
| 07:50.05 | ManxPower | I assume I need to call a 1004 test line to do the test. |
| 07:50.09 | rudholm | yes |
| 07:50.15 | ManxPower | rudholm: I don't remember but I know it was under $100 from ebay |
| 07:50.22 | rudholm | holy crapola |
| 07:50.24 | ManxPower | how about a S/N test. |
| 07:50.27 | rudholm | I want one of those! |
| 07:50.39 | rudholm | try the C-NOTCH test |
| 07:50.45 | rudholm | call your milliwatt test line |
| 07:50.56 | rudholm | it should subtract out the 1004Hz and see what's left |
| 07:50.59 | ManxPower | it does 2 and 4 wire analog and PCM (T-1 I suspect) |
| 07:51.12 | ManxPower | let me plug it back into the dmarc. |
| 07:54.37 | *** join/#asterisk Mahmoud (n=fake@unaffiliated/mahmoud) |
| 07:55.07 | ManxPower | OK, it says "38 dBrnC C-NOTCH TR BAL |
| 07:56.28 | ManxPower | rudholm: http://search.ebay.com/search/search.dll?from=R40&satitle=sage+instruments&category0= |
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| 07:57.20 | ManxPower | rudholm: there are FIVE of the same model (maybe different installed options) as I have |
| 08:00.32 | ManxPower | I assume S/N test also requires 1004Hz test tone? |
| 08:00.45 | rudholm | yeah |
| 08:02.18 | ManxPower | I have C-MSG, C- |
| 08:02.33 | rudholm | C-Message |
| 08:02.37 | ManxPower | I have C-MSG, C-NOTCH, S/N, and 3J FLT tests |
| 08:03.02 | ManxPower | C-MSG is about 80 |
| 08:03.38 | rudholm | it reads C-Message in dBrn, right? |
| 08:03.57 | ManxPower | C-MSG is about 80, C-NOTCH is 38, S/N is 41-42, 3K FLT is 80 |
| 08:04.04 | ManxPower | dBrnC |
| 08:04.04 | rudholm | hmm |
| 08:04.27 | ManxPower | 3K FLT is dBrn |
| 08:04.34 | rudholm | those numbers sound ballpark |
| 08:04.47 | rudholm | but I'm not sure, perhaps the unit came with a manual? |
| 08:04.57 | ManxPower | HAHAHAH! No manual. |
| 08:05.06 | rudholm | oh well, I had to ask :) |
| 08:05.28 | ManxPower | What I found on Sage's web site for a similar box assumed you knew what the numbers should be. |
| 08:05.33 | ManxPower | you say my eBay URL? |
| 08:06.07 | *** join/#asterisk zapp-branigan (n=zapp-bra@81.202.214.78.dyn.user.ono.com) |
| 08:06.13 | rudholm | wait, why'd you think I'd know the standard specs? |
| 08:06.33 | rudholm | who is Sage? |
| 08:06.51 | ManxPower | sage is the company that makes the test set. |
| 08:06.51 | rudholm | oh, you mean the mfgr :) |
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| 08:07.07 | rudholm | I have a friend named Sage, and was thinking, "wait, did he say something??" |
| 08:07.16 | ManxPower | 8-) |
| 08:07.41 | zapp-branigan | hi, i have installed the new version of asterisk and when i enter in asterisk -r i not see than the olderst version, call incoming and colors, who can see the old aspect and see when somebody connect to my asterisk ? |
| 08:09.28 | rudholm | ManxPower: I'm merely an EE, so while I understand info theory and electric telecom, things like standard specs are not in my repertoire :) |
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| 08:09.38 | ManxPower | that is interesting, the noise on the line totally changes between onhook and offhook |
| 08:09.53 | rudholm | hmm |
| 08:10.10 | rudholm | well, some originating equipment is really noisy during on-hook |
| 08:10.23 | rudholm | or is noise going up when you go off-hook? |
| 08:11.56 | ManxPower | on hook 138Hz - 234Hz or so, off hook 268Hz to 418Hz |
| 08:12.14 | rudholm | hmm |
| 08:12.23 | rudholm | that's very bizarre |
| 08:12.30 | ManxPower | I must not be understanding something important. 8-) |
| 08:14.11 | ManxPower | PSTN is 300Hz - 3.4Khz, so I'm not worried about noise below 300Hz or so |
| 08:14.50 | rudholm | yeah, although in some cases, it can carry signals well below 300 |
| 08:15.12 | rudholm | and if they're hitting your ear, then they're reducing the S:N ratio for you |
| 08:16.05 | rudholm | these Sage units are so nice, but I'm already too much of a reclusive junk collector :) |
| 08:16.09 | ManxPower | rudholm: I would hope that a good quality channel bank like my Adtran TA750 would filter out anything below 300Hz |
| 08:16.39 | rudholm | yeah, it should |
| 08:17.01 | ManxPower | i would not care, but a couple of people here are half deaf and turn the volume on their phone WAY up. |
| 08:17.10 | rudholm | ah |
| 08:17.12 | *** join/#asterisk friedrich| (n=friedric@e177253229.adsl.alicedsl.de) |
| 08:17.13 | rudholm | where is "here" btw? |
| 08:17.29 | ManxPower | long story. "campground" is the short answer. |
| 08:17.36 | rudholm | sounds fun |
| 08:17.42 | rudholm | you maintain outside plant? |
| 08:17.44 | ManxPower | about 1 hr NE of Birmingham, AL |
| 08:17.54 | rudholm | oh, you're in Digium Country |
| 08:18.15 | ManxPower | rudholm: it it has to do with technology, but not with Windows or web sites, I maintain it here. |
| 08:18.21 | ManxPower | I do the PBX, the WiFi, the CATV |
| 08:18.24 | rudholm | I've always wanted to build/maintain outside plant |
| 08:18.48 | ManxPower | By "CATV" I really mean the Big Ass Antenna and all the coax and amplifiers and channel processors |
| 08:19.05 | rudholm | yeah, I figured you really did mean Community Antenna |
| 08:19.36 | ManxPower | I have some locally originated content but that is limited by the modulators I have available |
| 08:19.51 | rudholm | ACME Space Modulator |
| 08:20.17 | ManxPower | Blonder Tongue actually, mostly from cable co sales on ebay |
| 08:20.26 | rudholm | (sorry, I had a Chuck Jones Martian ray-gun moment, there) |
| 08:21.00 | ManxPower | I've been up to Digium's HQ once, BTW. |
| 08:21.08 | rudholm | I haven't :( |
| 08:21.27 | rudholm | I visit them vicariously through Strom when he goes... |
| 08:21.42 | ManxPower | I should get pictures of my Noc/CO/Headend |
| 08:21.50 | rudholm | oooh |
| 08:22.19 | ManxPower | Where "CO" means a 66-block, a channel bank, and a PC running Asterisk |
| 08:22.30 | rudholm | hehe |
| 08:22.52 | rudholm | Strom and I went to visit someone I know up in Oregon who is a telephone collector extraordinaire |
| 08:22.55 | rudholm | has his own CO |
| 08:23.05 | ManxPower | nifty! |
| 08:23.11 | rudholm | with about a half-dozen mechanical switches |
| 08:23.16 | rudholm | all working |
| 08:23.36 | rudholm | his centerpeice is a WECO 355A step switch |
| 08:23.41 | rudholm | it's beautiful |
| 08:23.49 | rudholm | and *meticulously* maintained |
| 08:23.57 | rudholm | everything is to Bell System spec |
| 08:24.03 | rudholm | even the wire dressing |
| 08:24.59 | ManxPower | Somehow I think he has no other hobbies. 8-) |
| 08:25.22 | rudholm | I'm amazed that he has time for his wife! |
| 08:25.29 | rudholm | his CO is as big as his house |
| 08:25.57 | rudholm | Strom and I filled the trunk of my car with payphones before we left. |
| 08:26.15 | rudholm | proper Western Electric Bell System payphones |
| 08:26.19 | rudholm | all complete |
| 08:26.58 | AzianFlu | lol |
| 08:27.00 | ManxPower | I want a payphone |
| 08:27.01 | rudholm | heh |
| 08:27.06 | rudholm | yeah? |
| 08:27.10 | rudholm | what kind of payphone do you want? |
| 08:27.22 | ManxPower | they are expensive on eBay |
| 08:27.29 | ManxPower | one that works? |
| 08:27.32 | rudholm | yeah, and a lot of the ones on ebay suck |
| 08:27.37 | rudholm | "modified for home use" |
| 08:27.42 | rudholm | fcuk that shit |
| 08:27.50 | rudholm | it's gotta be complete and original |
| 08:27.56 | ManxPower | I'd have to modify one for Asterisk/channelbank use. |
| 08:27.57 | rudholm | with the totalizer and coin relay, etc |
| 08:28.03 | *** join/#asterisk saftsack (n=oliver@p54a7c28c.dip0.t-ipconnect.de) |
| 08:28.11 | ManxPower | no need to require actual money |
| 08:28.11 | rudholm | hehe, funny you should mention that |
| 08:28.30 | rudholm | well, if you just hook it up to a regular analog line, any Dialtone First payphone will work fine |
| 08:28.42 | ManxPower | right now we have a $5.99 (really) walmart phone on the wall for public use. |
| 08:28.51 | rudholm | (i.e. not a ground-start phone, but those haven't been in use for over 20 years) |
| 08:28.58 | rudholm | oh |
| 08:29.01 | ManxPower | I can do groundstart |
| 08:29.28 | ManxPower | I use Dial to limit calls to 10 mins (in OR out). |
| 08:29.34 | rudholm | well, then you could use a groundstart loop and set the initial rate in the phone to whatever |
| 08:29.55 | rudholm | but you wouldn't be able to have different prices for different calls |
| 08:30.02 | rudholm | you'd have to have flat-rate calls |
| 08:30.09 | ManxPower | I have unlimited LD on the lines anyway |
| 08:30.10 | rudholm | but that might work fo ryou |
| 08:30.21 | *** join/#asterisk nasls_lsa (n=chatzill@athedsl-219545.home.otenet.gr) |
| 08:30.28 | rudholm | I've been working on getting Asterisk to provide coin service support |
| 08:30.44 | nasls_lsa | rudholm cewwwwlll |
| 08:30.49 | ManxPower | I just have the 10 min time limit to keep people from yacking all day on the phone and to make them sign up for service with me. |
| 08:30.52 | rudholm | the only thing I haven't done is the modification to zaptel to get it to recognize the coin tones |
| 08:31.00 | rudholm | I see |
| 08:31.15 | ManxPower | the ONLY cell company that works here is verizon, so there is a lot of need for other phone options |
| 08:31.26 | rudholm | what about WiFi phones? |
| 08:31.35 | rudholm | you do have full 802.11 coverage, right? |
| 08:31.56 | nasls_lsa | rudholm: why don't you use the interface that asterisk gives |
| 08:32.01 | ManxPower | Well when we don't have an AP down. I have to climb on my telephone pole to find out what happened to one of our APs. |
| 08:32.06 | nasls_lsa | and you can connect it with other applications |
| 08:32.09 | rudholm | nasls_lsa: you mean EAGI? |
| 08:32.17 | ManxPower | I have no interest in supporting anything except for POTS service |
| 08:32.18 | nasls_lsa | AGI , yep |
| 08:32.49 | nasls_lsa | the "get-coin" part probably gives a conectivity to the pc |
| 08:32.55 | ManxPower | rudholm: we have WiFi coverage where most people will be camping, not thru the entire 50 acers |
| 08:32.57 | nasls_lsa | and from there you get the signal .. |
| 08:33.16 | rudholm | nasls_lsa: because decoding dual-tones in an external app is a bit hack-ish for my taste. |
| 08:33.40 | ManxPower | rudholm: so I should hit Strom up for a payphone? |
| 08:33.43 | rudholm | nasls_lsa: my TDM400 can decode DTMF, it should be able to decode coin tone |
| 08:33.47 | nasls_lsa | whatever you say ... |
| 08:34.09 | ManxPower | nasls_lsa: zaptel really is the correct place to do it |
| 08:34.22 | rudholm | nasls_lsa: other people have tried it useing EAGI, and they concur that it is very sub-optimal |
| 08:34.53 | rudholm | ManxPower: well, do you want a surface-mount payphone, or a flush-mount? |
| 08:35.08 | rudholm | I know someone who has a lot of both :) |
| 08:35.35 | *** join/#asterisk kavit (n=kavit@ppp167-236-231.static.internode.on.net) |
| 08:35.54 | rudholm | for coin service, ideally, you'd want to modify an ATA, like an IAXy to control the phone directly (think of an IAXy embedded in the payphone) |
| 08:36.29 | rudholm | that way you don't have to deal with coin tone decoding or coin relay voltage generation at all, which also has the side-effect of eliminating red box fraud. |
| 08:37.23 | ManxPower | Ok, there is WiFi in MOST of the areas people camp. I have to move an AP around so I can get WiFi in MY cabin. |
| 08:37.32 | rudholm | hehe |
| 08:37.42 | ManxPower | I really need to dig the trench to get conduit run to my cabin. |
| 08:37.50 | rudholm | this sounds like my kind of camp :) |
| 08:37.58 | ManxPower | I don't have WiFi, phone, CATV, or satellite |
| 08:38.01 | kavit | hehe camping with internet? |
| 08:38.07 | ManxPower | rudholm: Ask Strom if is your kind of camp |
| 08:38.14 | ManxPower | kavit: if you want it. |
| 08:38.48 | rudholm | well, he seems to have gone to bed |
| 08:38.48 | kavit | ManxPower: i'd rather not.... knowing my luck i'd kist get bothered by customers |
| 08:39.10 | ManxPower | rudholm: ask him tomorrow. the campground is an interesting place. |
| 08:39.16 | rudholm | but if it's his kind of camp, it's most likely my kind of camp, too. he and I share many things in common. |
| 08:39.20 | ManxPower | kavit: you don't have to use it 8-) |
| 08:39.47 | kavit | ManxPower: wha? have the internet and not use it.... thats crazy talk :P |
| 08:39.50 | ManxPower | It is a gay, clothing optional, campground on a mountian. |
| 08:39.57 | rudholm | heh |
| 08:40.01 | rudholm | nice |
| 08:40.12 | kavit | ManxPower: brokeback? |
| 08:40.19 | kavit | heh never gets old |
| 08:40.24 | ManxPower | Chandler Mountian, actually |
| 08:40.26 | rudholm | I'm a bit modest for running around naked, but otherwise that sounds perfectly amusing :) |
| 08:40.45 | ManxPower | nakedness is NOT required. |
| 08:41.03 | rudholm | yeah, you did say "optional" |
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| 08:41.29 | kavit | rudholm: its a matter of alcohol |
| 08:41.34 | rudholm | hahaha |
| 08:41.34 | ManxPower | need to get to bed shortly |
| 08:41.37 | *** join/#asterisk ppyy (n=ppyy@222.185.7.64) |
| 08:41.46 | rudholm | yeah, you're on EDT, no? |
| 08:41.55 | ManxPower | CDT actually |
| 08:42.05 | rudholm | still pretty late |
| 08:42.40 | ManxPower | yeah |
| 08:43.00 | ManxPower | I should e-mail my clients telling them I'll be in "Mean Mode" if they call me early |
| 08:43.08 | rudholm | heh |
| 08:43.36 | rudholm | chmod u+m ManxPower |
| 08:44.17 | kavit | rudholm: i'd prefer chattr +mean :P |
| 08:44.24 | rudholm | hehe |
| 08:44.31 | rudholm | maybe he doesn't support extended attributes :) |
| 08:45.22 | kavit | rudholm: he has friggen wifi in a camp in a mountain....ofcourse he supports extended attributes :P |
| 08:46.18 | rudholm | haha |
| 08:47.12 | kavit | anyway i am about to take the plunge and move to 1.4.X |
| 08:47.26 | rudholm | I haven't yet (shhh, don't tell anyone) |
| 08:47.29 | kavit | i dont even like messing with test boxes |
| 08:47.49 | kavit | rudholm: i have to.... we have a custom interface around some features |
| 08:47.50 | rudholm | I need to make sure my syntax is going to work with 1.4 |
| 08:47.55 | rudholm | ah |
| 08:48.03 | kavit | rudholm: they are making me update all the code and test it :/ |
| 08:48.06 | rudholm | yeah, I have no compelling reason other than bug fixes and keeping current |
| 08:48.10 | rudholm | nice |
| 08:48.12 | *** join/#asterisk darkskiez (n=mbryars@195-11-205-216.suip.mezzonet.net) |
| 08:48.13 | kavit | rudholm: i am the fricken owner of the company |
| 08:48.15 | kavit | :( |
| 08:48.33 | rudholm | so, you're making yourself do it? |
| 08:48.53 | kavit | rudholm: everyone voted and I ended up with it |
| 08:49.06 | rudholm | ah |
| 08:49.28 | kavit | rudholm: i should buy out my other partners and have like a mini patriot act for my company |
| 08:49.41 | kavit | FEMA@HOME |
| 08:50.10 | rudholm | what does the company do? |
| 08:50.55 | kavit | rudholm: ITSP, solution design, software development, hosted solutions....blah blah blah... its a list about a page long |
| 08:50.59 | rudholm | are you a good ITSP? |
| 08:51.00 | rudholm | :) |
| 08:51.08 | kavit | rudholm: main focus on telephony and solutions |
| 08:51.25 | kavit | rudholm: by australian standards, yes, premium.... |
| 08:51.37 | kavit | rudholm: business only, fully SLAed |
| 08:51.45 | rudholm | oh, better than Telstra, then? :) |
| 08:51.57 | rudholm | (as if that's saying anything :) ) |
| 08:51.59 | kavit | most of our customers are actually people who buy solutions off us |
| 08:52.11 | kavit | rudholm: hahah much better support |
| 08:52.12 | rudholm | I see |
| 08:52.34 | kavit | integrator first, ITSP to make life easier for us and customers I guess |
| 08:52.45 | rudholm | yeah |
| 08:52.54 | rudholm | an obvious vertical market I suppose |
| 08:53.09 | kavit | but have all features, like CLIP, number porting from PSTN to VoIP... etc etc etc |
| 08:53.18 | rudholm | nice |
| 08:53.26 | kavit | rudholm: there were over 200 VoIP providers in australia, last I counted |
| 08:54.09 | kavit | population of 20 million |
| 08:54.12 | rudholm | yeah, they're popping up like weeks over here, too |
| 08:54.39 | rudholm | that's like 1 VoIP provider for every 10,000 people |
| 08:55.07 | kavit | yeah rudholm, but hopefully the business sector will be different |
| 08:55.25 | kavit | most VoIP providers here are focusing on cheap calls and nothing else |
| 08:55.25 | rudholm | Vonage seems to have run into some major patent snags |
| 08:55.33 | rudholm | Verizon won a judgement against them |
| 08:55.42 | rudholm | some patent to do with PSTN gatewaying |
| 08:55.42 | kavit | rudholm: that has to hurt |
| 08:55.44 | rudholm | dunno the details |
| 08:55.45 | rudholm | yeah |
| 08:55.54 | rudholm | they were enjoined from signing up new customers |
| 08:55.58 | kavit | Verizon are shit everywhere |
| 08:55.58 | rudholm | but I think they got a stay on that |
| 08:56.00 | kavit | :\ |
| 08:56.04 | rudholm | yeah |
| 08:56.19 | rudholm | I don't like the idea of these companies using their patent portfolios to stifle competition |
| 08:56.41 | kavit | One of our customers has a hosted VPN and MPLS network with horizon where they host everything |
| 08:56.46 | kavit | I HATE IT |
| 08:57.03 | kavit | i think they just go out of their way to make life difficult for people |
| 08:58.07 | kavit | "hey can I get access to the logs from x day to see what happened to the link" |
| 08:58.10 | kavit | "No" |
| 08:58.16 | kavit | "err i need them" |
| 08:58.25 | kavit | "get auth from customer.." |
| 08:59.01 | *** join/#asterisk saftsack (n=oliver@p54a7ed79.dip.t-dialin.net) |
| 08:59.02 | kavit | "Ok... you know this is the 15th time this year I am asking you" |
| 08:59.17 | kavit | "you know I am the nominated contact" |
| 08:59.34 | kavit | "yeah, still need auth" |
| 08:59.39 | kavit | STAB STAB |
| 08:59.46 | kavit | STABBITY STAB |
| 09:00.04 | rudholm | haha |
| 09:00.29 | rudholm | one would think you could be given an "until further notice" nomination |
| 09:00.53 | kavit | rudholm: mainly everytime its a different site |
| 09:01.15 | kavit | i think the guy just hates me for stealing them off Verizon business voice |
| 09:01.31 | rudholm | from spam: *SICK and TIRED of trying to FIGURE OUT WOMEN??* What guy isn't? |
| 09:01.34 | rudholm | um, me. |
| 09:01.36 | rudholm | yeah, probably |
| 09:01.51 | rudholm | ok, I must sleep. I have to be at work in the morrow |
| 09:01.57 | kavit | later dude |
| 09:06.05 | nasls_lsa | tomorrow I'll go to my office and I will need some help trying to instert in my dial-plan an ISDN BeroNet BNS40 card |
| 09:06.36 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
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| 09:18.38 | rrrobert | :) |
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| 09:56.52 | IgorG | Hello, all! Who could answer: is anybody need L2 CoS support in asterisk? |
| 09:57.27 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 10:04.27 | *** join/#asterisk voltagex (n=voltagex@124-254-121-97-dsl.ispone.net.au) |
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| 10:32.18 | tzafrir_laptop | IgorG, why do you ask? (not that I have any idea what L2 CoS is) |
| 10:32.30 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 10:33.47 | IgorG | tzafrir_laptop: This is VLAN based QoS. I ask because i find that exist patch for 1.0.9, but feature not implemented in 1.2 and 1.4 |
| 10:34.58 | tzafrir_laptop | IgorG, do *you* need it? |
| 10:36.02 | IgorG | tzafrir_laptop: yes, I need it. Quation is: i do it for my, or make all needed patches and post to bugtracker |
| 10:37.38 | tzafrir_laptop | If you need it, probably someone else also needs it. So if you do that work anyway, please do submit a patch |
| 10:38.26 | IgorG | Ok, I have also think so... |
| 10:50.33 | voltagex | god I'm bored |
| 10:57.08 | *** join/#asterisk tigmmi (n=admin@adsl-245-184-192-81.adsl2.iam.net.ma) |
| 10:57.12 | tigmmi | hello |
| 11:00.35 | tigmmi | Is there a way to destroy zap channel whenever the client quit without deregistring nor hungup the call? The example I do have is related to ekiga, when it I do kill it with ctrl+c (started from terminal), the zap channel keeps up and HAVE TO DESTROY IT FROM asterisk manager in order to free it for furder call. |
| 11:01.49 | tigmmi | do anybody have an idee? |
| 11:01.58 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
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| 11:17.55 | tzafrir_laptop | tigmmi, "destroy zap channel" what do you mean? 'zap destroy channel' does not just end a call |
| 11:18.00 | lvnatic | hi |
| 11:18.43 | tzafrir_laptop | tigmmi, what zap channel is it? analog trunk? |
| 11:18.48 | tzafrir_laptop | (FXO) |
| 11:19.12 | CrazyTux | Whats the best way to intergrate asterisk voicemail (mysql database), any wiki information? |
| 11:19.21 | CrazyTux | For 1.4.2 * |
| 11:19.21 | tigmmi | The zap channel is stil in use, and I do cannot use it for furder cal till I restart asterisk or send zap destroy channel 1 through asterisk manager |
| 11:19.22 | tzafrir_laptop | tigmmi, 'soft hangup' may be what you're looking for |
| 11:19.22 | *** join/#asterisk progcaribu (n=arturo@izones70.izones.net) |
| 11:20.32 | tzafrir_laptop | tigmmi, does 'soft hangup' work? |
| 11:21.24 | tigmmi | I didn't test it yet |
| 11:21.33 | tigmmi | waite please |
| 11:24.14 | tigmmi | that command is for asterisk manager! I do destroy it anyway using zap destroy channel, but the problem is that I wander if ther is a way the detect such problem automatically and frees the zap channel if the soft phone client quit without hanging up |
| 11:27.32 | tigmmi | tzafrir_laptop, The problem is that if the soft phonre exits without hanging up, the asterisk server keeps the zap channel open and not usable for furder use intil you use "soft hanguo" or "zap destroy channel" to free it |
| 11:28.00 | tzafrir_laptop | what do you mean by "keeps the zap channel open"? |
| 11:28.13 | tzafrir_laptop | 'zap show channels' should always show it |
| 11:28.59 | tzafrir_laptop | if you run 'zap show channel 1' , d oyu see it as offhook rather than onhook? |
| 11:29.06 | mvanbaak | what's the max nicklen here ? |
| 11:29.17 | mvanbaak | <--- tuning his irssi theme |
| 11:29.21 | tigmmi | means that if you cannot Dial any more till you use asterisk manager to hangup |
| 11:29.51 | tigmmi | means that you cannot Dial any more till you use asterisk manager to hangup |
| 11:30.00 | tigmmi | sorry |
| 11:31.35 | mvanbaak | 16 it is ;) |
| 11:32.24 | mvanbaak | yay |
| 11:33.36 | tigmmi | tzafrir_laptop, I need asterisk to detect sutch situation and frees the zap channel authomatically, without using asterisk manager. May be somthing like timeout |
| 11:35.29 | tzafrir_laptop | tigmmi, btw: on 1.4 I believe you have 'zap restart'. I you have just a single channel, it will destroy it and reconfigure it |
| 11:35.58 | *** join/#asterisk skambar (n=skambar@p54840002.dip0.t-ipconnect.de) |
| 11:36.46 | tigmmi | asterisk 1.2.17 |
| 11:39.28 | CrazyTux | Anyone in here have experiencing with 1.4.* and voicemail (database setup) |
| 11:39.44 | CrazyTux | Everything I find seems outdated, ODBC 1.2.* |
| 11:46.23 | *** join/#asterisk progcaribu (n=arturo@izones70.izones.net) |
| 11:47.52 | tigmmi | tzafrir_laptop, I do think that may be solved if asterisk detect that there is no audio passing though (timeout) and hangups authomatically! |
| 11:48.35 | tzafrir_laptop | tigmmi, but the issue here that a hangup has failed |
| 11:48.42 | tzafrir_laptop | this is a bug |
| 11:49.20 | tzafrir_laptop | If you can't hang up the call with 'soft hangup', this is a bug |
| 11:50.14 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 11:50.30 | tigmmi | asterisk should normally ping the client periodically, and hangup after a timeout, or detect that there is no audio passing to one dirction and hangups |
| 11:50.53 | tigmmi | tzafrir_laptop, use zap destroy channel, it works for me |
| 11:51.01 | tenzind | hi .. can i test asterisk in a network of just 2 computers? |
| 11:51.19 | tigmmi | zap 1: zap destroy channel 1 |
| 11:51.23 | tzafrir_laptop | tenzind, sure. Why not |
| 11:51.25 | tzafrir_laptop | ? |
| 11:51.52 | tzafrir_laptop | I test Asterisk on my laptop occasionally |
| 11:52.04 | tigmmi | That is waht I'm doing now tenzind |
| 11:53.04 | tzafrir_laptop | tigmmi, again: before you run 'zap destroy channel 1': do you see a running call in: 'core show channels'? |
| 11:53.24 | tenzind | tzafrir_laptop, tigmmi: where can i find a simple how-to .... till now i have only found how-tos to do complicated things with asterisk |
| 11:53.26 | *** join/#asterisk k-man (n=jason@unaffiliated/k-man) |
| 11:53.41 | tzafrir_laptop | ('zap restart' should save you the need to restart. With or without 'zap destroy channel') |
| 11:54.06 | tzafrir_laptop | tenzind, are you familiar with Linux? |
| 11:54.16 | DrukenLPY | where did all this core shit come from anyways? what was wrong with simple old 'show channels' ? |
| 11:54.36 | tenzind | tzafrir_laptop: yes but not an expert though .. and i am using debian |
| 11:54.47 | tzafrir_laptop | DrukenLPY, from the core asterisk developers? |
| 11:54.51 | tigmmi | no core command, I'm using asterisk 1.2.17 |
| 11:54.57 | tenzind | tzafrir_laptop: can i apt-get install asterisk? |
| 11:55.12 | tzafrir_laptop | Etch has a pretty decent version, yes |
| 11:55.19 | DrukenLPY | tzafrir_laptop: smart ass |
| 11:56.17 | tzafrir_laptop | tenzind, though I would recommend you to add the apt source I maintain (which mostly has Sid backports) |
| 11:56.50 | tzafrir_laptop | deb http://updates.xorcom.com/rapid etch main |
| 11:57.12 | tzafrir_laptop | then try: apt-get install asterisk asterisk-config-simple |
| 11:57.47 | tenzind | tzafrir_laptop: ok |
| 11:58.42 | *** join/#asterisk frigidzephyr (i=frigidze@nat/digium/x-e611d799408259eb) |
| 12:00.25 | tzafrir_laptop | tenzind, also try: apt-get install rapid-scripts; /usr/share/rapid-scripts/console-menus on |
| 12:00.50 | tzafrir_laptop | thi should provide a menu on consoles 1, 5 and 6 |
| 12:03.46 | tenzind | tzafrir_laptop: i am geting a GPG error while apt-get updating from your repo .... is it alright? |
| 12:03.56 | *** join/#asterisk Chicago (n=Chicago@76.16.117.173) |
| 12:04.04 | tzafrir_laptop | right. You need the gpg key. |
| 12:04.56 | tzafrir_laptop | apt-get install xorcom-archive-keyring; apt-get update |
| 12:05.21 | tzafrir_laptop | the key can also be found at http://updates.xorcom.com/xorcom.gpg.asc |
| 12:05.28 | Chicago | Good Morning. I defined a SIP client which is integrated into MythTV... Do I also have to add an h323 channel for it to work properly? I'm seeing messages like chan_h323.c: Cannot kill myself in my logs. and in the myth log I see SIP: Unknown parameter in -Authenticate; algorithm |
| 12:05.38 | tenzind | tzafrir_laptop: ok ... thanks |
| 12:05.41 | *** join/#asterisk sysreq (n=sysreq@dsl-10-143-33.b2b2c.ca) |
| 12:05.50 | tigmmi | tzafrir_laptop, The problem is solved by adding busydetect=yes in zapata.conf |
| 12:05.56 | tigmmi | heh, lol |
| 12:06.16 | tzafrir_laptop | tigmmi, this is why I asked you if there's a running call |
| 12:06.53 | tzafrir_laptop | btw: in what coutry are you? doesn't the telco provide better disconnect supervision? |
| 12:06.57 | tzafrir_laptop | e.g: ks |
| 12:07.13 | tigmmi | yes |
| 12:07.19 | tigmmi | it's fine |
| 12:07.58 | tzafrir_laptop | busydetect is a bad hack that should only be used if you have no better alternatives |
| 12:08.30 | tigmmi | What alternative? http://www.voip-info.org/wiki/index.php?page=Asterisk+Disconnect+Supervision |
| 12:09.37 | tigmmi | I'm using A HARDWARE PBX to make zap call to other rooms |
| 12:10.01 | tigmmi | Actually I'm not usnig the TELCO LINES |
| 12:10.07 | DrukenLPY | tzafrir: i didn't know you sold asterisk hardware.... |
| 12:10.11 | tigmmi | only local PBX |
| 12:10.54 | tigmmi | bbl, Many thanks tzafrir_laptop |
| 12:11.48 | *** join/#asterisk af_ (n=getsmart@81-174-45-50.f5.ngi.it) |
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| 12:20.24 | wulfy814 | app_dial.c:1059 dial_exec_full: Unable to create channel of type 'Zap' |
| 12:20.41 | wulfy814 | is what I'm getting after upgrading from 1.2.9 to 1.2.17 |
| 12:20.49 | wulfy814 | incoming calls are fine, it's only on outgoing |
| 12:21.16 | wulfy814 | any ideas what I'm going wrong? |
| 12:24.21 | DrukenLPY | anyone know if it's a holliday in canada today? |
| 12:24.38 | DrukenLPY | does business resume today or tomorrow? |
| 12:24.43 | nasls_lsa | don't know for canada , but in Greece sure ;) |
| 12:26.23 | GreyFoxx | DrukenLPY: IT is for government employees |
| 12:26.33 | GreyFoxx | and it varies from company to company for everyone else |
| 12:26.49 | mvanbaak | DrukenLPY: no idea about canada neither, but in .nl it's a holiday today |
| 12:27.39 | *** join/#asterisk Strom_M (n=strom@netblock-66-159-243-60.dslextreme.com) |
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| 12:32.15 | *** join/#asterisk Knight_DKN (n=knight_d@c58-107-190-112.thoms2.vic.optusnet.com.au) |
| 12:33.12 | Knight_DKN | Evening all |
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| 12:38.21 | Chicago | How do I troubleshoot "Yuck! Error in buffer handling...: Connection reset by peer" when issuing a "stop now" or "stop gracefully" at the console? |
| 12:38.29 | *** join/#asterisk lvnatic (i=lvnatic@84.236.21.135) |
| 12:38.33 | Knight_DKN | Okay boys and girls, I've been asked to setup a VoIP server at work and I've picked Asterisk as the PBX server, if I want a stable production box which stream of Asterisk should I use? 1.2 or 1.4? |
| 12:38.49 | lvnatic | re |
| 12:39.17 | *** join/#asterisk sysreq (n=sysreq@dsl-10-143-33.b2b2c.ca) |
| 12:42.21 | lvnatic | I want to make an originate action via asterisk-java. Everythings OK, except the "channel" parameter. I can't figure it out what to pass into this param, if I want to connect two extension for eg. |
| 12:42.40 | *** join/#asterisk mvanbaak (n=mafkees@vanbaak.xs4all.nl) |
| 12:42.43 | *** join/#asterisk tehmaze (i=wijnand@unaffiliated/tehmaze) |
| 12:42.52 | tehmaze | hello |
| 12:43.07 | Knight_DKN | Hello |
| 12:43.37 | Chicago | Knight_DKN: The 1.2 releases generally include only minor bug fixes... the 1.4 releases usually have major updates. |
| 12:44.48 | Knight_DKN | So 1.2 would be the choice for a production server? |
| 12:44.50 | file | not exactly true, 1.2 and 1.4 are bug fix only branches... it's just 1.4 is the newest major release |
| 12:44.54 | tehmaze | I am trying to call out a pre-recorded message, but I keep getting errors about not being able to find the extension; http://paste-it.net/e7f6a59 |
| 12:45.07 | tehmaze | can somebody give me a clue how to fix this? |
| 12:45.25 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
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| 12:45.42 | *** mode/#asterisk [+o anthm] by ChanServ |
| 12:49.06 | tehmaze | <PROTECTED> |
| 12:49.24 | tehmaze | I do however have an exten => s,1 defined in my [nagios] section |
| 12:50.16 | darkskiez | ok, i just upgraded to asterisk 1.4 and now calls that go out my pri dont actually ring, they appear to be ringing but the number you phone doesnt get called |
| 12:51.30 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-185-4.buckeyecom.net) |
| 12:53.05 | tzafrir_laptop | tehmaze, do you? check: show dialplan nagios |
| 12:57.42 | *** join/#asterisk gr1ncheux (n=devine@unaffiliated/gr1ncheux) |
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| 13:15.46 | tehmaze | tzafrir_laptop: ah I was mixing extensions.conf and .ael |
| 13:16.06 | mvanbaak | gheh |
| 13:16.21 | mvanbaak | that's why I moved everything to extensions.ael |
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| 13:21.57 | *** mode/#asterisk [+o mog] by ChanServ |
| 13:22.02 | *** join/#asterisk shaun33 (i=PPCUzer@cslnk01.ispdr.net.au) |
| 13:22.38 | shaun33 | Hey: Looking for assistance with AGI and asterisk |
| 13:29.28 | *** part/#asterisk shaun33 (i=PPCUzer@cslnk01.ispdr.net.au) |
| 13:38.01 | *** join/#asterisk coppice (n=chatzill@90.203.17.210.dyn.pacific.net.hk) |
| 13:43.37 | darkskiez | anyone any ideas how to progress with debugging my 1.4 upgrade.... my pri line seems dead, it comes up, and i can send calls to it but it seems dead |
| 13:45.50 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
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| 13:57.36 | l2cache | has anyone had any experience with the auto join acd function on polycom phones? |
| 14:00.03 | *** join/#asterisk Farz (n=J3aidan@91.140.139.170) |
| 14:00.38 | Farz | hi |
| 14:01.06 | *** join/#asterisk progcaribu (n=arturo@izones70.izones.net) |
| 14:01.30 | drfreeze | Anyone know if a TDM880 board will fit in a 2U machine? |
| 14:02.15 | Farz | anyone can recommend a card with 2 x FXO with Echo cancellation plz? |
| 14:02.49 | *** join/#asterisk explisit (n=explisit@213.240.243.141) |
| 14:04.18 | Farz | anyone? :) |
| 14:05.05 | Farz | ok, anyone having *any* kind of distro containing Asterisk in a LIVECD in production? |
| 14:06.44 | *** join/#asterisk irule (n=irule@189.164.43.19) |
| 14:06.47 | irule | hi |
| 14:07.16 | drfreeze | Farz: I do |
| 14:07.20 | irule | what is the mos recommended method of assisted call transfer for my diaplan? |
| 14:07.30 | drfreeze | I use a TDM40B card |
| 14:08.22 | drfreeze | Farz: I chose the Live CD because of the ease of installation - os and all |
| 14:08.37 | drfreeze | then blew away /etc/asterisk and put my own configs in |
| 14:11.37 | *** join/#asterisk eald (n=eald@189.157.162.200) |
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| 14:24.22 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 14:26.45 | Mercestes | drfreeze: Hm, that'd be pretty cool, a working asterisk config liveCD. |
| 14:26.50 | Mercestes | I should look into how to make liveCDs. |
| 14:27.44 | *** join/#asterisk Strom_M (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 14:33.11 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 14:33.27 | drfreeze | Mercestes: worked well when I was a newby |
| 14:33.40 | drfreeze | Probably won't do it again |
| 14:34.23 | Mercestes | yea. I was thinking mroe "make my job more efficient" but it could be something I could hand to a newby. |
| 14:36.23 | Mercestes | I got an ISO image of a working embedded system with *. ...I need to get Soentoo going now. |
| 14:36.31 | Mercestes | Problem is the damn thing is too big to fit on a 512 card. =/ |
| 14:37.14 | Mercestes | maybe if I did my astlinux install first...then did the keydisk...and then did the soentoo install...and then made an image of that. |
| 14:37.16 | Mercestes | ...could work. |
| 14:39.55 | polerin | Mercestes is all about making things easy so us stupid support personell can handle it |
| 14:40.11 | Mercestes | lol. Mainly because I'm the stupid support personell. |
| 14:40.42 | *** join/#asterisk `p4r14h`work (n=josh@24-119-48-78.cpe.cableone.net) |
| 14:40.54 | polerin | mmhmm. |
| 14:40.57 | polerin | ack! |
| 14:41.04 | polerin | what who what huh?! |
| 14:41.08 | Mercestes | >.> |
| 14:41.31 | Mercestes | you ruined my "techs are stupid" joke btw. |
| 14:42.04 | polerin | why's that? |
| 14:42.17 | Mercestes | with the "omg, I'm a tech" response. : P |
| 14:42.27 | polerin | heh |
| 14:42.56 | polerin | trust me, I'm perfectly aware there are stupid techs |
| 14:42.59 | polerin | I talk to them every day |
| 14:43.15 | Mercestes | lol |
| 14:43.19 | polerin | Someone literally asked me how you spell "CMD" |
| 14:43.43 | Mercestes | lol |
| 14:43.56 | polerin | it's not like I asked them "Put in Cummmmmmddday" no . I said "C-M-D" |
| 14:44.03 | *** join/#asterisk etfonhomey (n=etfonhom@74-140-213-69.dhcp.insightbb.com) |
| 14:45.15 | etfonhomey | russellb, you around? |
| 14:45.22 | *** join/#asterisk MrWup (n=neil@80-46-38-124.static.dsl.as9105.com) |
| 14:47.03 | *** join/#asterisk navigo (n=navigo@adsl-074-239-053-186.sip.gnv.bellsouth.net) |
| 14:47.54 | navigo | Can anyone here give me an idea how to determine when a remote party answers a call that I have Dial() ed? |
| 14:48.27 | *** join/#asterisk progcaribu (n=arturo@izones70.izones.net) |
| 14:49.11 | *** join/#asterisk cygar (n=cygar@201.216.200.33) |
| 14:49.24 | Farz | what LIVECD you used, drfreeze? |
| 14:49.34 | cygar | Hello |
| 14:49.35 | Farz | is there any TrixBox LIVECD? |
| 14:49.42 | Farz | hi cygar |
| 14:49.44 | Mercestes | Farz: #trixbox |
| 14:50.12 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:50.26 | Farz | ty |
| 14:51.16 | navigo | I am trying to bridge an inbound call to an outbound call, and I need the time the remote party answers the channel.... |
| 14:51.22 | cygar | I need to write a script that shows the 'State' [ ringing, up ,etc ] from a channel... is there any command connecting through the Manager to get that? Not to use 'show channels' and parse the Channel and State column? |
| 14:52.09 | *** join/#asterisk darken_darken (n=marco@240.129.77.83.cust.bluewin.ch) |
| 14:54.10 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 14:54.56 | drfreeze | Farz: the old a@home, but now Trixbox |
| 14:58.58 | drfreeze | Trying to decide between TDM880B and A20400D - Any recommendations out there? |
| 14:58.58 | drfreeze | The Sangoma board is about $200 more expensive |
| 14:59.08 | Mercestes | I've had great results from Sangoma in the past, but...I've two digium boards here too that seem to be working ok too |
| 14:59.55 | Mercestes | but I keep hearing sangoma > digium but i also hear rumor that digium is closing that gap |
| 15:00.13 | Mercestes | and if you buy digium you support asterisk. |
| 15:00.23 | drfreeze | Mercestes: the TDM880B looks like a nice board |
| 15:00.27 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 15:00.51 | drfreeze | but I think the Sangoma has hardware echo cancelation and the TDM doesn't |
| 15:03.21 | irule | what is the industry-wide default method of call transfer? |
| 15:03.45 | Mercestes | Well I start with the transfer button.....and then I dial the number I want, and then I hit the transfer button again. |
| 15:03.53 | *** join/#asterisk menfin (n=cray@LNeuilly-152-21-127-49.w193-253.abo.wanadoo.fr) |
| 15:04.14 | Mercestes | are you asking for like RFC level protocol explanation or ? dial plan methods of directing a call? Or some programatic method of doing an actual transfer? |
| 15:06.12 | Farz | Echo cancellation is a MUST feature? |
| 15:06.21 | Mercestes | no. |
| 15:06.37 | Farz | im trying to setup a home pbx |
| 15:06.57 | Mercestes | echo cancellation is a as needed feature. |
| 15:07.02 | *** join/#asterisk phillipk (n=pkey@216.248.143.77) |
| 15:07.07 | Farz | if i use IP phone devices, they come with Echo Cancellation? |
| 15:07.09 | Mercestes | Rule #1, never echo cancel a fax line. Fax + echo cancel is bad. |
| 15:07.19 | *** part/#asterisk l2cache (n=ghansen@64.128.254.98) |
| 15:07.26 | Mercestes | Rule #2: T1's shouldn ever have echo. Echo cancellers are a coverup for the real problem. |
| 15:07.31 | drfreeze | Mercestes: So, maybe I should be considering the TE120P card since my line are coming in via a T1 |
| 15:07.37 | Mercestes | s/shouldn ever/should never/ |
| 15:08.29 | Mercestes | drfreeze, hardware echo cancellers are trez chic' so it could very well be the way to go |
| 15:09.52 | Farz | will you fix EC with IP Phones? |
| 15:10.13 | *** join/#asterisk marv[work] (n=timr@24.214.206.254) |
| 15:11.09 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 15:11.39 | Mercestes | Rule #3: SIP doesn't echo |
| 15:11.41 | *** join/#asterisk etfonhomey (n=etfonhom@74-140-213-69.dhcp.insightbb.com) |
| 15:11.47 | Farz | oh perfect! :) |
| 15:11.54 | tehmaze | ok wicked, nagios is calling me now using text2wave (festival) to tell me what is wrong |
| 15:12.21 | etfonhomey | [TK]D-Fender, so how did things end up with brian last night? |
| 15:13.02 | [TK]D-Fender | etfonhomey, ... you you really wanna see? |
| 15:13.28 | Mercestes | tehmaze: Sweet. :) |
| 15:13.35 | etfonhomey | [TK]D-Fender: Did you save the conversation? |
| 15:13.49 | tehmaze | asterisk++ |
| 15:14.06 | Farz | whats the BEST LIVECD distro out there? |
| 15:14.11 | Farz | for production |
| 15:14.33 | Mercestes | Sabayon, Kororaa, Gentoo. |
| 15:14.49 | Farz | no a Linux distro |
| 15:14.56 | Mercestes | ... |
| 15:15.03 | Farz | a whole web based Asterisk system |
| 15:15.16 | Mercestes | uh. |
| 15:15.18 | Mercestes | Astlinux? |
| 15:15.24 | Mercestes | We don't do GUIs here |
| 15:16.38 | Farz | i do have a fully running Asterisk on a Fedora doing DID business |
| 15:16.54 | Farz | im trying to pinpoint a LIVECD that would make * more popular |
| 15:16.59 | [TK]D-Fender | Farz, I have never heard of a "live" distro + * CD before. Only ones that INSTALL everything on your HD wiping out whatever ws there before. |
| 15:17.21 | Farz | ok try this plz |
| 15:17.30 | Farz | www.adminparadise.com |
| 15:17.43 | [TK]D-Fender | Farz, we already have Trixbox... don't you think ENOUGH damage has been done to * already? :) |
| 15:17.49 | Mercestes | LMAO |
| 15:17.50 | polerin | Mercestes: Lissa was having fun with the echo test saturday. She was giggling into it and making faces ;P |
| 15:17.56 | Farz | Lol |
| 15:17.59 | Farz | oh come on :) |
| 15:18.05 | Mercestes | Nice. |
| 15:18.09 | Mercestes | Lissa is "Melissa"? |
| 15:18.14 | polerin | yeah |
| 15:18.18 | polerin | my wife |
| 15:18.20 | Mercestes | yea, you said melissa once. |
| 15:18.24 | Mercestes | my wife's name is Melissa too |
| 15:18.25 | Farz | i did try AsteriskNOW LIVECD |
| 15:18.25 | Mercestes | :D |
| 15:18.27 | Farz | its nice |
| 15:18.44 | Mercestes | see, that's *two* things we have in common. |
| 15:18.44 | polerin | fie on you, wife namestealer! |
| 15:18.49 | irule | I try 'blindxfer => ##' and it works perfectly, I then change it to 'blindxfer => #2' and it does not work, what gives? |
| 15:18.55 | Mercestes | lol. Aye, shame on me. |
| 15:19.01 | Farz | Medosa?? |
| 15:19.56 | Mercestes | irule: you do not parse. can you pastebin the extensions.conf, sip show peers, and cli output of working and not working examples? |
| 15:19.58 | Mercestes | ~pastebin |
| 15:20.07 | jbot | pastebin is, like, a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or http://rafb.net/paste/, or http://pastebin.com is usually painfully too slow and unresponsive to use, use one of the other pastebin sites, or dpaste.com is a very nice pastebin as well |
| 15:20.07 | Mercestes | grr. |
| 15:20.08 | Mercestes | Damn you jbot. |
| 15:20.13 | Farz | heh |
| 15:20.14 | Mercestes | ~botsnack |
| 15:20.14 | jbot | Mercestes: thanks |
| 15:20.37 | Farz | anyone can recommend a nice US based online Digium reseller? |
| 15:20.47 | Mercestes | Digium. |
| 15:20.53 | navigo | anyone have any pointers on how to determine when the remote party answers a Zap channel? |
| 15:20.57 | Farz | they have links to distributors and stuff |
| 15:21.06 | *** join/#asterisk Flusher (i=flusher@filer.euroserv.com) |
| 15:21.08 | Mercestes | what are you trying to buy? |
| 15:21.10 | Flusher | hi |
| 15:21.20 | Farz | TDM408P |
| 15:21.23 | Farz | heya Flusher |
| 15:21.51 | Farz | actually i need any 2 x FXO decent card |
| 15:22.19 | Mercestes | Farz: voipstore maybe. |
| 15:22.30 | Farz | k ty |
| 15:22.36 | *** join/#asterisk revcane (i=dng@cpe-76-187-77-70.tx.res.rr.com) |
| 15:22.42 | Mercestes | telephonydepot |
| 15:22.43 | irule | can I transfer a call with flash hook? |
| 15:23.02 | irule | here goes Mercestes |
| 15:23.04 | Sweeper | best way to delete voicemail after N days? |
| 15:23.23 | [TK]D-Fender | irule, depends on the phone & interface its on |
| 15:24.17 | Flusher | I'd like to play a music that contains a short message "welcome to blah" (about 20 secs). My idea is to answer(), then setmusiconhold(answeringmusic), then waitmusiconhold(20), then dial my SIP friends. The problem is that the musiconhold feature doesnt rewind the mp3 if I dial several times. There is a sort of buffer. I tried with a custom player, it's the same thing. What would you advise to me please ? |
| 15:24.50 | irule | [TK]D-Fender analog phone on an ata sipura 2000 |
| 15:25.15 | [TK]D-Fender | irule, yes you can transfer using hook-flash. go download the MANUAL. |
| 15:25.17 | Mercestes | irule: I've programed something like that before, intercepting an incoming call then doing a Flash() and then a senddtmf() to do an analog transfer. |
| 15:25.29 | Mercestes | Sweeper: cronjob |
| 15:25.43 | Sweeper | Mercestes: figures :P |
| 15:25.45 | Mercestes | Flusher: set announce= |
| 15:25.56 | Mercestes | Flusher: periodic_announce I think. |
| 15:26.01 | [TK]D-Fender | Mercestes, he's talking FXS hook-flash for * call transfer. |
| 15:26.31 | *** join/#asterisk CVirus (n=GoD@196.205.192.172) |
| 15:26.31 | irule | http://www.usuarios.lycos.es/thechompons/uploader/telefono.jpg thats my phone |
| 15:26.37 | Mercestes | [TK]D-Fender, so am I. i think. >.> someone gave me a howto on it. You did an Answer, then a flash, then a SendDTMF or Dial or something. |
| 15:26.44 | [TK]D-Fender | Flusher, Soulds like you are just playing a sound file period and dumping them somewhere else once its done. that is called PLAYBAACK |
| 15:27.22 | [TK]D-Fender | Mercestes, what you are describing is getting * to to a transfer on an FXO interface (to the telco). this is the reverse. |
| 15:27.35 | Mercestes | [TK]D-Fender, oh. you are correct. |
| 15:28.40 | *** join/#asterisk wunderkin (n=kev@dslstat-ppp-95.fastq.com) |
| 15:29.12 | [TK]D-Fender | irule, well you've got your answer.. go do it... HOWEVER.... I don't think I've seen pulse dial support anywhere in Sipura's list of features... |
| 15:29.55 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 15:31.16 | Flusher | [TK]D-Fender: but is there a way to playback() a sound file like I wish. I mean : play the sound for 20 secs *then* dial() the SIP friends ? |
| 15:31.59 | [TK]D-Fender | Flusher, how about you just load your recording into your favourite audio editor and CLIP it at 20s? |
| 15:34.43 | Flusher | [TK]D-Fender: okay i'll split the file into an announce and a music, it's easier ;) |
| 15:35.12 | Flusher | then i'll just have to call mp3player() i guess, then musiconhold |
| 15:36.27 | drfreeze | Anyone know how to query an * box to tell what type of card is installed, e.g. TDM card |
| 15:37.01 | Mercestes | drfreeze, lspci |
| 15:37.19 | drfreeze | Mercestes: thx |
| 15:37.25 | Mercestes | np |
| 15:38.00 | deoptima | see |
| 15:38.15 | polerin | smartass |
| 15:39.20 | drfreeze | Mercestes: do you see a TDM card in here: http://pastie.textmate.org/52559 |
| 15:39.21 | [TK]D-Fender | Flusher, no, not mp3player .... PLAYBACK. |
| 15:39.44 | revcane | whats the easiest os to install asterisk on? seems like fedora core 4,5 and 6 have issues. do you recommend FC 3 over any other OS? |
| 15:39.46 | [TK]D-Fender | drfreeze, 2nd to last |
| 15:40.05 | [TK]D-Fender | revcane, CentOS |
| 15:40.19 | polerin | revcane: Try out Etch and see how it is ;) |
| 15:40.32 | *** join/#asterisk deeperror (n=deeperro@mail.banctel.com) |
| 15:40.36 | *** join/#asterisk Modcuts (n=Moducts@88-109-98-148.dynamic.dsl.as9105.com) |
| 15:40.49 | drfreeze | Mercestes: dmesg found: Found a Wildcard TDM: Wildcard TDM400P REV I (4 modules) |
| 15:40.51 | *** join/#asterisk andrebarbosa (n=andrebar@83.240.150.150) |
| 15:40.55 | revcane | i dont want to do the "try out" stuff, i want something quick, easy and stable |
| 15:41.37 | [TK]D-Fender | revcane, again, CentOS |
| 15:41.37 | revcane | cool, i'll check it out. |
| 15:41.37 | andrebarbosa | anyone knows why in recently zaptel packages we need to load wcte4xxp before wctdm? |
| 15:41.46 | andrebarbosa | on zaptel1.2.6 i load wctdm first and i dont have problems |
| 15:41.47 | andrebarbosa | :s |
| 15:42.45 | darkskiez | i just had to rollback a 1.4 upgrade back to 1.2, my pri line just didnt work, with no obvious reason |
| 15:43.19 | Modcuts | Does anybody know if you can map two tables to one RealTime family in the extconfig? |
| 15:44.55 | anonymouz666 | can I loop a read() var with fieldqty() using a blank delimiter? |
| 15:45.04 | anonymouz666 | ,, |
| 15:45.54 | anonymouz666 | the read var is 123123 format, and I need to play each digit |
| 15:46.11 | [TK]D-Fender | anonymouz666, Sure |
| 15:46.26 | Corydon-w | I don't think FIELDQTY does what you think it does |
| 15:47.03 | Corydon-w | Modcuts: yes, you can |
| 15:47.09 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 15:47.23 | Corydon-w | sippeers and sipusers, for example, are usually configured to the same table |
| 15:48.00 | *** join/#asterisk mvanbaak (n=mafkees@vanbaak.xs4all.nl) |
| 15:50.00 | *** join/#asterisk gosht (n=user@cmodem-237-235.tricom.net) |
| 15:50.05 | gosht | hello all |
| 15:50.30 | gosht | anyone have experience connecting an Asterisk to a Nortel Succession 1000? |
| 15:50.38 | *** join/#asterisk Defraz (n=t0tal@fw.fuzecore.com) |
| 15:50.46 | anonymouz666 | I just need to know how much digits the read() var has to loop N times and use playback() to each digit |
| 15:51.19 | anonymouz666 | Corydon-w: i can't use filedqty() because I don't have a pattern |
| 15:51.26 | anonymouz666 | value1,value2 |
| 15:51.28 | anonymouz666 | etc |
| 15:51.44 | Modcuts | Corydon-w: i mean mapping sip_users to two tables how do you seperate the two ","? |
| 15:52.44 | GreyFoxx | Modcuts: you'd really need to use MySQL Views to create a "single" table that asterisk would look at |
| 15:53.00 | GreyFoxx | the view can be defined to pull in the proper fields from the 2 other tables |
| 15:54.24 | Modcuts | GreyFoxx: ah ok thats makes sense, i'll give that a go. |
| 15:54.33 | Modcuts | GreyFoxx: cheers |
| 15:54.38 | Mercestes | gosht: I see pain in your future, padawan. |
| 15:55.49 | gosht | Mercestes: it's a Succession 1000s...how much pain are we talking about? |
| 15:56.42 | Mercestes | gosht: It's a Nortel. |
| 15:57.03 | Mercestes | gosht: Has this nortel *ever* had a problem in any way, shape or fashion since it's inception? |
| 15:57.16 | *** join/#asterisk Deeewayne (i=dwayne@nat/digium/x-e9485fa67a0df090) |
| 15:57.37 | blitzrage | anonymouz666: ${FIELDQTY(${MY_VALUES},\,)} doesn't work? |
| 15:57.38 | gosht | Mercestes: not that i'm aware...it's brand new |
| 15:58.37 | blitzrage | anonymouz666: what does the string look like? I bet you can use a While(${CUT(...)}) combination |
| 16:00.47 | Mercestes | gosht: why would you buy a nortel switch and then plug * into it? |
| 16:00.54 | *** join/#asterisk mrichmanM (n=richmanm@70.89.184.1) |
| 16:01.33 | *** join/#asterisk HeyItsMe (n=HeyItsMe@69.18.124.82) |
| 16:01.50 | gosht | Mercestes: not us...a customer of ours...down here it's very common to buy Nortel only to discover they wasted their money...the telco here has a monopoly |
| 16:02.10 | Sweeper | gosht: what kind of IO do you get off of that? PRI? |
| 16:02.23 | [TK]D-Fender | gosht, a monopoly on what... the hardware market? Do they ban imports too? |
| 16:02.35 | gosht | Sweeper: no, T1 |
| 16:02.44 | Sweeper | gosht: eh, that works too |
| 16:03.00 | Sweeper | not gonna be hard, so long as you have access to the Nortel's conig |
| 16:03.12 | Sweeper | *config |
| 16:03.23 | Sweeper | I've done stuff with Option11's |
| 16:03.33 | gosht | [TK]D-Fender: they don't ban imports, it's just that no one else is in the business of bringing telecom equipment |
| 16:03.34 | Mercestes | gosht: My point is, yo ushouldn't have bought the nortel to begin with. |
| 16:03.37 | Sweeper | only hitch was when the pbx guy didn't program the port correctly |
| 16:04.07 | *** join/#asterisk ucfMethod (n=ucfmetho@office.eyestreet.com) |
| 16:04.11 | gosht | Mercestes: again...this was the customer, not us...we are simply a tech consulting firm |
| 16:04.36 | gosht | Mercestes: we've done probably half the * implementations down here and that's why they came to us |
| 16:04.45 | [TK]D-Fender | gosht, well then go buy foreign and ship it in! |
| 16:04.58 | Sweeper | way to ignore the guy |
| 16:05.03 | Sweeper | the nortel is already THERE |
| 16:05.09 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 16:05.36 | gosht | [TK]D-fender: we prefer * and Digium ;) |
| 16:06.41 | gosht | so anyone got some tutorials or articles they can point me to? we have a lot of experience with the Meridian small PBXs but not with a Sucession 1000s |
| 16:06.50 | Mercestes | Sweeper: Sorry? I was ignoring you. |
| 16:07.00 | Mercestes | gosht: I would use FXO/FXS or a T1 interface to plug into the Nortel then. |
| 16:07.05 | HeyItsMe | Hi all |
| 16:07.15 | Sweeper | gosht: that'll be mostly nortel docs, only thing asterisk will see is the T1 |
| 16:07.21 | Mercestes | Gosht: try your best to convince NOrtel that your not there |
| 16:09.07 | Sweeper | gosht: and you really will want to get PRI's going over the T1's, it'll make life eleventy billion times easier for you, integration-wis |
| 16:09.07 | gosht | Sweeper: with access to the Nortel config and a T1 interface...what would we have to do? |
| 16:09.19 | [TK]D-Fender | hey, how do you go "back" in "make menuconfig" for * 1.4? |
| 16:10.00 | Sweeper | gosht: uh, depends. if you're terminating through asterisk, amke the nortel send all calls outside the internal callrange out over the PRI |
| 16:10.22 | Sweeper | and tell asterisk about the nortel phones.... |
| 16:10.35 | Sweeper | in the dialplan |
| 16:11.06 | gosht | Sweeper: it's going to be the other way around...the Nortel will be terminating |
| 16:11.45 | Sweeper | gosht: ah |
| 16:12.00 | Sweeper | well, I'm not a nortel programmer... |
| 16:12.18 | Sweeper | but I imagine you'd tell it there were N extensions |
| 16:12.20 | gosht | Sweeper: nobody is...it's a dying breed :D |
| 16:12.21 | Sweeper | off that PRI |
| 16:12.36 | Sweeper | set it up as a tie line |
| 16:13.15 | Sweeper | and then just tell asterisk to send all non-local calls over the PRI |
| 16:13.18 | gosht | Sweeper: first thing's first...how do we get PRI over T1? we've done T1 ties before on Meridians |
| 16:13.29 | Sweeper | gosht: no fucking clue, on the nortel end :D |
| 16:13.43 | *** join/#asterisk viler (n=mv@134.42.128.66.PPPoECali.dynamic.telesat.net.co) |
| 16:13.50 | gosht | Sweeper: ok |
| 16:13.53 | Sweeper | I mean, it's doable with just T1, I'd assume |
| 16:14.00 | Sweeper | but I think it'd be a much larger PITA |
| 16:14.12 | Sweeper | what with signalling and such |
| 16:14.25 | Sweeper | PRI makes things much simpler, as far as asterisk goes |
| 16:14.37 | *** join/#asterisk Rhab (n=Rhab@p3EE0F08D.dip.t-dialin.net) |
| 16:14.46 | Rhab | Hi |
| 16:14.48 | gosht | Sweeper: ok...so we need some Nortel guy to help us then... |
| 16:15.01 | Mercestes | gosht: not if you just PRI it over to the Nortel |
| 16:15.02 | *** join/#asterisk simonkern (n=simonker@p54aa96d4.dip0.t-ipconnect.de) |
| 16:15.06 | [TK]D-Fender | gosht, Well you're in here asking for help on a Nortel system... DUH! |
| 16:15.07 | simonkern | hi |
| 16:15.09 | Rhab | Some1 here who can help me with hylafax? |
| 16:15.13 | Mercestes | gosht: From Nortel's perspective you'll just be the "out bound interface." |
| 16:15.31 | Mercestes | Rhab: OF COURSE!! This is the #hylafax channel after all. |
| 16:15.42 | *** part/#asterisk HeyItsMe (n=HeyItsMe@69.18.124.82) |
| 16:15.42 | Mercestes | Rhab: Did you try the mailing list? |
| 16:15.45 | Rhab | My email attached faxes are broken, while the tif files in recvq are all fine |
| 16:15.48 | Rhab | no |
| 16:16.04 | Rhab | there is no hylafax channel afaik |
| 16:16.11 | Mercestes | Rhab: seriously? I would go through Ifax or the mailing list, both Darren and "faxguy" are stellar. |
| 16:16.44 | simonkern | i installed asterisk + misdn today on my debian etch machine. but when i want to start asterisk, chan_misdn shows an error (http://paste.debian.net/25269). |
| 16:17.05 | Rhab | no hylafax channel |
| 16:17.25 | Mercestes | Rhab: go on over to hylafax.org and join the mailing list, and you can probably find the solution in the archives, but feel free to mail the list anyways for free support, they are fairly interactive. |
| 16:17.34 | simonkern | then i patched the kernel with the std2kern program, but nothing changed |
| 16:17.41 | Rhab | mailing list would be the next, but i think its only little problem |
| 16:17.42 | gosht | Sweeper: thnks...we'll try PRI over T1 and if not just go with T1 |
| 16:17.48 | Mercestes | Rhab: Ifax offers very good support as well but you have to A: Buy their stuff or B: buy a contract |
| 16:18.11 | Rhab | im only home user so lfax isnt a option |
| 16:18.14 | gosht | [TK]D-fender: sorry for the Nortel questions but it's the first step in replacing a Nortel for Asterisk |
| 16:18.25 | Rhab | since its really just a little issue |
| 16:18.29 | gosht | Mercestes: thnks too |
| 16:18.35 | Mercestes | gosht: NP, good luck. |
| 16:18.46 | *** join/#asterisk ppyy (i=ppyy@218.93.69.219) |
| 16:18.59 | Mercestes | Rhab: broken tif attachments in your email is not a little issue and may very well be a ghostwriter issue. |
| 16:19.10 | gosht | bye all...off to tear down Nortels! |
| 16:19.22 | Mercestes | Go, padawan, destroy Nortels in my name. |
| 16:19.31 | polerin | Rhab: just out of curiosity, the tif's are ok on the server, but broken when you email. have you run a diff on the files? |
| 16:19.43 | Rhab | however i have the faxes on my server, just the emailing doenst work |
| 16:19.58 | Rhab | run a what? |
| 16:20.18 | Rhab | polerin what u mean by diff? |
| 16:20.28 | Rhab | it has |
| 16:20.41 | Rhab | so i insalled metamail recently |
| 16:20.45 | polerin | Rhab: possible you are sending the file in the wrong way then? |
| 16:21.29 | Mercestes | Usually in Hylafax distros one uses ghostwriter/ghostscript to convert the tif into a pdf and then email it as an attachment. |
| 16:21.30 | Rhab | which wrong way? the emails reach my imap server and i access them from my windows client |
| 16:21.43 | Mercestes | so AFIAK he's not emailing a .tif |
| 16:21.47 | Mercestes | but he could be. |
| 16:22.02 | polerin | Mercestes: nods |
| 16:22.07 | Rhab | im tried emailing tif and pdf |
| 16:22.15 | Mercestes | and the tif itself is broken? |
| 16:23.04 | Rhab | the tif in the email, yes |
| 16:23.13 | Mercestes | it's time for a reality check then. Go to your server and find a working tiff, and then manually sendmail it over frmo the command line. |
| 16:23.14 | Rhab | the one on the server is allright |
| 16:24.00 | Rhab | Mercestes: ok, i will try this, but im kinda newb with sending emails on debian |
| 16:24.36 | Mercestes | if you dig around enough you'll probably find a script that is calling sendmail and then catting the contents of your tif finto the email body after making the proper header statements. |
| 16:25.02 | polerin | yey for improper headers :D |
| 16:25.03 | Mercestes | and basically you want to run that script manually. |
| 16:25.21 | Mercestes | polerin: ain't nothing worse than improper headers huh? |
| 16:25.36 | Mercestes | polerin: makes you just wanna slap somebody. >.> |
| 16:25.41 | polerin | hehe |
| 16:25.53 | Rhab | hm ill try that sendmail |
| 16:26.33 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 16:27.52 | Mercestes | lol |
| 16:28.04 | Rhab | Mercestes: could u maybe post me a simple command to send an email with an attached fax0000001.tif to the local user "xam"? All this email stuff is confusing me |
| 16:28.11 | Mercestes | see what happens when I get +b in other channels!?! *stares at no one in particular.* |
| 16:28.48 | *** join/#asterisk EzWay (n=ez@c66.110.149-45.clta.globetrotter.net) |
| 16:28.54 | Mercestes | Rhab: Do you have an existing .eml file sitting on yoru server that i can use for reference? I don't happen to have all that memorized at the moment. |
| 16:29.22 | Rhab | I have exim4 and mutt installed |
| 16:29.28 | Mercestes | Rhab: I would just take everything above the strange looking "nonsense", edit as necessary, and then cat the .tif file into the end of the email msg |
| 16:30.22 | Rhab | im familiar with man pages but email stuff is kinda different |
| 16:31.25 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 16:31.26 | Rhab | I think there might be a 7 / 8-bit rproblem |
| 16:31.28 | Mercestes | rhab: Well good news. :) You'll be learnign something new today |
| 16:32.02 | Rhab | That really sucks dick |
| 16:33.12 | Mercestes | learnign something new? |
| 16:33.37 | Rhab | no that email shite |
| 16:33.44 | Mercestes | ah |
| 16:33.47 | *** join/#asterisk nasls_lsa (n=chatzill@athedsl-219545.home.otenet.gr) |
| 16:34.12 | Mercestes | I really think the mailing list is going to be your best resource. |
| 16:34.19 | Mercestes | and they suck alot less dick. I think. |
| 16:35.06 | *** part/#asterisk ucfMethod (n=ucfmetho@office.eyestreet.com) |
| 16:35.11 | Mercestes | you can expect a oneliner from faxguy telling you what broken thing to look at tho but atleast it will be in the right direction |
| 16:35.34 | Rhab | whos that faxguy |
| 16:35.42 | Rhab | where can i find him |
| 16:35.50 | Mercestes | on the MAILING LIST. |
| 16:35.54 | *** join/#asterisk shinux__ (n=shinux@208.70.5.150) |
| 16:36.04 | *** join/#asterisk sergee (i=kvirc@195.94.224.197) |
| 16:36.30 | sergee | any cisco experts here? :) i have an issue with 5350... |
| 16:37.23 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 16:37.33 | *** join/#asterisk ucfMethod (n=ucfmetho@office.eyestreet.com) |
| 16:38.17 | Mercestes | sergee: I'm not an *expert* but I can tell a router from a birdfeeder. What' sthe question?? |
| 16:38.22 | ucfMethod | the 1.2.17 update is probably security fixes right. I am running 1.2.15, is it worth the update potential headache? |
| 16:38.37 | Mercestes | ucfMethod, From what I hear, yes. |
| 16:38.43 | Mercestes | ucfMethod, But I don't know first hand. |
| 16:38.47 | *** join/#asterisk bblack (i=bblack@38-118-35-20.rapiddsl.net) |
| 16:39.11 | sergee | Mercestes: i have a 5350 ... it is connected to PSTN via E1 and to Asterisk via SIP |
| 16:39.46 | sergee | Mercestes: when i'm placing a call from sipura to pstn via asterisk - everything ok |
| 16:40.46 | sergee | Mercestes: but when i'm trying to place a call from asterisk to pstn via callfile - i have something strange between 5350 and pstn, and 5350 returns "500 internal error" to asterisk... |
| 16:40.58 | *** join/#asterisk gerphimum (n=trekkie@207.190.62.44) |
| 16:41.06 | ucfMethod | Mercestes: Thanks man |
| 16:41.20 | Mercestes | ucfMethod, np. Good luck |
| 16:41.33 | Mercestes | sergee, 5350 console give anything interesting back? |
| 16:41.57 | Mercestes | sergee, And I'm going to blindly guess tha tthe sipura success and the call file failure is generated on the same number, right? |
| 16:41.58 | sergee | Mercestes: yes! debug isdn q931 -> http://pastebin.ca/431365 |
| 16:42.27 | sergee | Mercestes: line 24 - i've never seen it before... |
| 16:42.58 | sergee | Mercestes: right, i'm trying to call to the same number |
| 16:43.23 | Mercestes | sergee: Can you "catch" call file made by the Sipura? |
| 16:43.28 | sergee | Mercestes: line 29 also seems unfamiliar to me :) |
| 16:43.48 | sergee | yes |
| 16:44.16 | errr | Im having some trouble sending faxes. I setup iaxmodem and hylafax and incoming works like a champ, but when you try to send the fax you get an error: SEND FAILED: JOB 3 DEST 3662407 ERR No local dialtone any idea what I need to look at to get this working? |
| 16:44.23 | *** join/#asterisk grandy (n=chatzill@c-69-181-153-165.hsd1.ca.comcast.net) |
| 16:44.47 | sergee | Mercestes: http://pastebin.ca/431370 <- isdn debug q931 (from sipura's call) |
| 16:45.01 | Mercestes | your getting a NAK response back from outside the Cisco |
| 16:45.10 | Mercestes | which is calling a disconnect |
| 16:45.19 | Mercestes | circult/channel unavailable. |
| 16:45.26 | grandy | hello... i am using voipstreet and iax for a small business phone system... after a few minutes of crystal clear conversation the audio is suddenly cutting out.... anyone have any idea how I can debug this? |
| 16:45.28 | Mercestes | are you providing callerID in the callfile? |
| 16:45.35 | sergee | Mercestes: what is a "NAK"? |
| 16:45.51 | Mercestes | grandy, jitter/network. |
| 16:45.59 | Mercestes | sergee: =/ uh, you google it. :P |
| 16:46.10 | Mercestes | and then tell me. |
| 16:46.12 | grandy | Mercestes: hmmm... any particular settings you'd recommend for jitter buffer? |
| 16:46.25 | Mercestes | grandy: A better network. |
| 16:46.26 | sergee | Mercestes: i tryed to provide, also i tryed to comment CallerID from callfile - same results... |
| 16:46.26 | grandy | Mercestes: oddly there is no jitter or anything before the audio goes away |
| 16:46.36 | grandy | Mercestes: we have 25 ms ping times to voipstreet |
| 16:46.49 | Mercestes | grandy: I think I recall "400" somewhere in that mix but honsetly you have to tinker with it and make lots of test calls. |
| 16:47.02 | Mercestes | grandy: ping under 500 means nothing to me. |
| 16:47.13 | grandy | Mercestes: hmm... ok... |
| 16:47.32 | Mercestes | grandy: I can run a good call at 500-750 ping if it *stays* at the exact same ping time and never deviates. |
| 16:47.47 | grandy | Mercestes: ahh... interesting... |
| 16:48.02 | Mercestes | grandy: you get issues when it starts bouncing around between 22ms and 750ms. Try an mtr to voipstreet and watch for standard deviation and set it to display jitter settings. |
| 16:48.53 | Mercestes | sergee: Are you passing callerid in th ecall file? yoru carrier coudl be rejecting you for an invalid source number. |
| 16:49.18 | grandy | Mercestes: hmm... what is an mtr to voipstreet? |
| 16:49.23 | Mercestes | sergee: see if you can "catch" a call file made from your sipura device, and then duplicate it and mv it into your outbound directory and see if ti works that way. |
| 16:49.39 | Mercestes | sergee, if it does work then try to use that call file as a template, if ti does not work then there is a different issue. |
| 16:49.55 | Mercestes | grandy: mtr is "my traceroute" and it's alot more informative than ping. |
| 16:49.59 | sergee | Mercestes: thanks! will try it now |
| 16:50.34 | Mercestes | sergee, and make sure you *mv* your call fileinto outbound and not cp, otherwise * mayjump the gun and mulch it up before it's completed |
| 16:50.54 | [TK]D-Fender | I have run into a snap on * 1.4.2 I could use hand with : I flushed my 1.2 setup (backing up configs), did a full compile & install of Base + Addons. I then did "make samples" and copied back my extensions.conf & sip.conf (which were 1.4 compliant). I have sincene lost the ability to dial #'s starting with "*" from my Polycom SIP phones. I used to use "pedantic=yes" to allow this to operate and now it fails to work with it enabled or otherwise |
| 16:51.41 | sergee | Mercestes: same: NAK Response Received - command 0xF201, result code 0x800E, msg id 0xF2FF, session id 0x6D, msg tag 0x0, slot/port 1/1 |
| 16:52.04 | Mercestes | Hrm. |
| 16:52.07 | Mercestes | that *should* work. |
| 16:52.08 | sergee | Mercestes: do you have any vocabulary for command 0xF201, result code 0x800E ? |
| 16:53.24 | grandy | Mercestes: thanks much for that tip... going to try it |
| 16:53.29 | [TK]D-Fender | continued : in SIP debug I do see the "*" clearly in the "To:", but am getting a 404 on it for some strange reason. |
| 16:54.28 | Mercestes | sergee, "NAK" seems to be an ambiguous term referring to "negative acknolwedgement." |
| 16:54.36 | Mercestes | yoru call is being rejected by your telco it seems. |
| 16:54.55 | sergee | Mercestes: yeah i found NAK definition too :) |
| 16:55.24 | [TK]D-Fender | OMG... nevermaind.... I forgot an included config file... |
| 16:55.26 | [TK]D-Fender | ashdkjshdal |
| 16:56.06 | *** join/#asterisk CrashHD (n=crashhd@c-67-166-155-233.hsd1.ca.comcast.net) |
| 16:56.37 | [TK]D-Fender | AHHH there we go :) |
| 16:56.41 | *** part/#asterisk bblack (i=bblack@38-118-35-20.rapiddsl.net) |
| 16:56.45 | [TK]D-Fender | another <5 min problem solved :) |
| 16:57.38 | [TK]D-Fender | ok, am now on to the fun stuff :) |
| 16:59.13 | grandy | Mercestes: ok running mtr on chi-reg.voipstreet.com |
| 16:59.13 | grandy | Mercestes: worst jmax is 237 |
| 16:59.13 | sergee | Mercestes: what is the best way to "capture" callfile from sipura? |
| 17:00.18 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
| 17:02.11 | anonymouz666 | blitzrage: sorry for delay to answer. the read() var looks like "123456789" I need to loop how many digits the var has to playback() each digit. how to use while(cut(..)) if I don't have delimiter? |
| 17:02.20 | grandy | Mercestes: here is the output of mtr... in case you can offer an interpretation of it: http://pastebin.ca/431395 |
| 17:02.55 | *** join/#asterisk PupenoR (n=pupeno@200.123.183.91) |
| 17:03.05 | *** join/#asterisk gr1ncheux (n=devine@unaffiliated/gr1ncheux) |
| 17:03.14 | Rhab | Mercestes: its working, but emails sending still sucks dick |
| 17:03.44 | Rhab | Was some issue with base64_encode |
| 17:04.02 | blitzrage | anonymouz666: I think you want to use LEN() then |
| 17:04.14 | blitzrage | that'll tell you how many characters you have in a string |
| 17:04.22 | Rhab | nn |
| 17:04.24 | Rhab | bb |
| 17:04.27 | *** part/#asterisk Rhab (n=Rhab@p3EE0F08D.dip.t-dialin.net) |
| 17:04.50 | blitzrage | While($[${X} < ${LEN(${MY_VAR})}]) |
| 17:05.06 | blitzrage | Set(X=$[${X} + 1]) |
| 17:05.09 | blitzrage | EndWhile() |
| 17:08.15 | *** join/#asterisk CrashHD (n=crashhd@c-67-166-155-233.hsd1.ca.comcast.net) |
| 17:10.32 | *** join/#asterisk Rhab (n=Rhab@p3EE0F08D.dip.t-dialin.net) |
| 17:11.12 | *** join/#asterisk Pr0t3la (n=plstofje@209.114.137.202) |
| 17:12.20 | Rhab | Hi |
| 17:12.21 | anonymouz666 | blitzrage: thanks! |
| 17:12.31 | Rhab | Again a question concerning hylafax: |
| 17:12.40 | Rhab | it mails me 2 emails |
| 17:12.55 | Rhab | one with no attachment and one with it |
| 17:13.04 | Rhab | for each received fax |
| 17:13.23 | Rhab | I just wanna get one email with the attachment |
| 17:13.26 | Pr0t3la | I want to get a phone number in the US so that when my customers call i can play a recorded tape like "welcome to blah support center...bla bla bla" |
| 17:13.29 | Pr0t3la | any hints? |
| 17:13.55 | Strom_M | a recorded tape, eh? |
| 17:13.56 | Strom_M | is it still 1981 or did I not get the memo |
| 17:14.02 | Pr0t3la | i mean |
| 17:14.04 | Pr0t3la | a recorded voice |
| 17:14.08 | Pr0t3la | (not english) |
| 17:14.12 | Strom_M | ah ok :) |
| 17:14.13 | Pr0t3la | im spanish sorry |
| 17:14.16 | Pr0t3la | you got it anyway |
| 17:14.43 | anonymouz666 | blitzrage: do I need to iniatialize(x=0) the ${X} before using it? |
| 17:15.00 | blitzrage | of course |
| 17:16.06 | *** join/#asterisk sysreq (n=sysreq@219.47-ppp.3menatwork.com) |
| 17:17.16 | *** join/#asterisk SGN (n=a@adsl-66-122-88-147.dsl.renocs.nvbell.net) |
| 17:17.39 | SGN | Does anyone know if digium cards support duplex phone lines? Or do I have to split the duplex's before the cards. |
| 17:17.42 | *** join/#asterisk shinux__ (n=shinux@208.70.5.150) |
| 17:20.11 | Strom_M | SGN: each port supports only a single pair |
| 17:20.31 | SGN | Strom_M: Ok, That explains that then - thanks. |
| 17:21.38 | deeperror | anyone aware of any fxs type hardware for projects or embedded systems where the jacks can be soldered on manually for a project? |
| 17:23.27 | Sweeper | deeperror: intel modem on minipci maybe? |
| 17:23.56 | *** join/#asterisk cspot (i=cspot@ip68-109-8-207.pn.at.cox.net) |
| 17:24.12 | deeperror | sweeper: not sure will look into this |
| 17:26.35 | [TK]D-Fender | Sweeper, Intel modem would be FXO, not FXS.... |
| 17:27.17 | deeperror | yea looking for more of a solder type solution where i have a main board that hooks into an fxs board with pinouts |
| 17:27.22 | deeperror | some low level hardware |
| 17:28.48 | Sweeper | ah |
| 17:29.11 | Sweeper | good luck :) |
| 17:29.38 | Corydon-w | deeperror: generally not, because FXS hardware generates heat when the power is not utilized |
| 17:30.09 | deeperror | i'm needing about 6 ports |
| 17:30.33 | deeperror | all i can really find is 4port cards and module pci type cards but nothing to grab an extra 2 ports and maintain a small size |
| 17:30.41 | cpm | deeperror, this might be what you are looking for |
| 17:30.43 | cpm | http://www.rowetel.com/ucasterisk/index.html#quickstart |
| 17:31.01 | deeperror | yea i was actually just there haha |
| 17:31.17 | deeperror | i do like what they are using with regards to main board though |
| 17:31.24 | deeperror | will look into that for sure |
| 17:33.13 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 17:33.25 | cpm | his magic box will hold 8 of his magic ports, |
| 17:33.37 | cpm | he's a neat guy too, |
| 17:37.43 | *** join/#asterisk Qwell[] (i=qwell@pdpc/sponsor/digium/Qwell) |
| 17:37.43 | *** mode/#asterisk [+o Qwell[]] by ChanServ |
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| 17:42.28 | *** join/#asterisk chefrs (n=joe@c-24-8-226-145.hsd1.co.comcast.net) |
| 17:42.44 | chefrs | Any idea why when I dial long-distance outbound I get a message saying "You have not dialed enough digits to place your call" ? |
| 17:43.33 | *** join/#asterisk ManxPower (n=manxpowe@99.sub-70-223-44.myvzw.com) |
| 17:46.43 | [TK]D-Fender | chefrs, pastebin the CLI output of your failed call. |
| 17:46.43 | [TK]D-Fender | ~pb |
| 17:46.47 | jbot | i guess pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 17:46.59 | chefrs | [TK]D-Fender: It looks just like a normal, succesful call. |
| 17:47.07 | [TK]D-Fender | chefrs, SHOW please... |
| 17:47.11 | chefrs | You're the boss. |
| 17:47.12 | Sweeper | ~penisbin |
| 17:47.14 | jbot | Please leave all e-peens at the door, we have conveniently provided a bin! |
| 17:48.48 | chefrs | [TK]D-Fender: http://channels.debian.net/paste/6009 |
| 17:49.56 | [TK]D-Fender | chefrs, Sounds like its the telco telling you that. |
| 17:50.06 | chefrs | It is. |
| 17:50.18 | chefrs | Earlier I was getting "You must first dial a 1" so I added a wait, all was fine. All of a sudden I get this today. |
| 17:50.37 | [TK]D-Fender | chefrs, maybe you need more "w"'s in front of your dial to wait for dialtone... I'm betting they aren't getting the first "1" |
| 17:50.43 | chefrs | I've had up to 4. |
| 17:50.54 | chefrs | I just changed it down to 3, 2, and now 1 to try out different things. Still getting the message. |
| 17:51.14 | [TK]D-Fender | chefrs, plug an analog phone in parallel with your * and test side by side to see what you're hearing |
| 17:51.29 | chefrs | Mmkay. |
| 17:52.32 | ManxPower | chefrs: there is a tone length option in /etc/asterisk/zapata.conf you might want to set it to 150 or 200. perhaps the telco is not detecting the short DTMF tone lengths Asterisk is sending |
| 17:53.21 | [TK]D-Fender | chefrs, Also very valid... try that first. |
| 17:53.33 | chefrs | It happened on my analog phone. Weird. |
| 17:53.47 | chefrs | ManxPower: I'll try that though. |
| 17:54.10 | [TK]D-Fender | chefrs, if it fails on an analog phone I don't think its * or your phone, I'm thinking telco here... |
| 17:54.22 | chefrs | Yeah but... I wonder what could cause that. It's so weird. |
| 17:54.33 | *** join/#asterisk darken_darken (n=marco@10.154.77.83.cust.bluewin.ch) |
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| 17:54.40 | *** part/#asterisk Rhab (n=Rhab@p3EE0F08D.dip.t-dialin.net) |
| 17:55.01 | *** join/#asterisk shinux__ (n=shinux@208.70.5.150) |
| 17:55.03 | chefrs | ManxPower: Do you know the name of the variable to be set? |
| 17:55.33 | [TK]D-Fender | chefrs, you are using FreePBX which is not supported here. |
| 17:55.53 | [TK]D-Fender | chefrs, You should be asking in #freepbx if you need help setting it up to do waht you want |
| 17:56.44 | chefrs | [TK]D-Fender: Yeah but I can vi the .conf's just fine and bypass FreePBX. Easier in some instances. But yeah, you're right. |
| 18:01.25 | *** join/#asterisk yenno (i=yunien@84-72-188-194.dclient.hispeed.ch) |
| 18:02.28 | yenno | hi, is it normal that asterisk replaces the session description and puts its own ip and codecs in there? |
| 18:02.49 | blitzrage | yes, it's a b2bua |
| 18:04.44 | yenno | so is my asterisk server also a proxy? |
| 18:06.23 | [TK]D-Fender | yenno, No. * is NOT a proxy. It is a B2BUA, PERIOD |
| 18:06.41 | file | [TK]D-Fender: you forgot a period at the end of your sentence |
| 18:06.54 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 18:07.15 | [TK]D-Fender | file, what are you talking abot? I spelled it out! ;) |
| 18:07.22 | syzygyBSD | how do I enter the tftp server into a cisco 7960? |
| 18:07.25 | yenno | ok, and that means that all rtp sessions go over the sip server? |
| 18:07.27 | [TK]D-Fender | file, oh.. and PONG for the call earlier ;) |
| 18:07.42 | file | [TK]D-Fender: I think that was my room mate typing in the wrong extension |
| 18:07.49 | [TK]D-Fender | yenno, that is a DIFFERENT amtter. that is dependent on your use of SIP reinvites |
| 18:07.56 | [TK]D-Fender | file, :O |
| 18:08.45 | yenno | ok, thanks |
| 18:18.12 | [TK]D-Fender | What was it I have to do in * 1.4.2 to get Polycom Presence functional again? I tried "type=friend" and "type=peer" as well with some vague recollection about tha to no avail. googling is going slow. Any care to lend a hand? |
| 18:18.28 | polerin | curry + duck + pinapple == awesomenss |
| 18:18.51 | rudholm | [TK]D-Fender: yeah, try Yahoo! :) |
| 18:21.13 | [TK]D-Fender | rudholm, yeah.. umm.. thanks... *sough* |
| 18:21.50 | polerin | MSN! **hides** |
| 18:22.53 | chefrs | [TK]D-Fender: Figured it out. |
| 18:22.55 | jebba | since upgrading to 1.4.2 I've had a number of people reporting DTMF issues.... any clues? :) |
| 18:23.14 | [TK]D-Fender | chefrs, And the verdict is? |
| 18:23.34 | chefrs | Boss put PAT codes on our phone lines. I had no idea wth they were till just now. |
| 18:24.00 | *** join/#asterisk SmaxTheFrog_ (n=M_A_X@i577A4C5F.versanet.de) |
| 18:26.20 | SmaxTheFrog_ | what would be the syntax for gotoIfTime to jump to position x if matches? |
| 18:28.20 | *** join/#asterisk thoughtpolice (n=austin@c75-111-145-138.plaicmtc01.tx.dh.suddenlink.net) |
| 18:32.21 | danp | i discussed this some last night but wanted to see if anyone had any more advice...i'm trying to make TalkSwitch analog phones work with asterisk and i'm not sure how or if i can make intercom/paging work. it was suggested that a second pair is probably used for that signalling. any other tips for things to check out? any other "standard" way analog phones implement intercom/paging on the same pair the voice line is on? |
| 18:33.14 | ManxPower | danp: NO standard analog phone supports paging. |
| 18:33.39 | ManxPower | If the phone supports intercom then it is not a standard analog phone |
| 18:33.41 | danp | that's what i thought but i just wanted to make sure |
| 18:35.31 | danp | thanks! |
| 18:35.49 | sb_mx | SmaxTheFrog_, this could help http://www.asteriskguru.com/tutorials/gotoiftime.html |
| 18:39.02 | *** join/#asterisk froguz (n=alvaro@pc-69-217-46-190.cm.vtr.net) |
| 18:39.21 | Mercestes | polerin: sounds ....interesting. |
| 18:44.25 | *** join/#asterisk Greek-Boy (n=Greek-Bo@196.45.144.42) |
| 18:45.53 | *** join/#asterisk Waverly360 (n=irc@209.12.249.243) |
| 18:46.31 | Waverly360 | Hey guys, what would cause asterisk to give me a dial tone when I try to Dial out through a PRI. |
| 18:46.31 | Waverly360 | ? |
| 18:46.39 | cygar | I need to write a script that shows the 'State' [ ringing, up ,etc ] from a channel... is there any command connecting through the Manager to get that? Not to use 'show channels' and parse the Channel and State column? |
| 18:47.14 | polerin | Mercestes: it is the bestes. It was gone inside of 5 minutes of hitting my desk ;) |
| 18:47.35 | Mercestes | hehe. *takes notes* |
| 18:47.41 | Mercestes | asterisk, linux and duck recipes |
| 18:49.11 | [TK]D-Fender | cygar, "show channels concise" |
| 18:49.57 | [TK]D-Fender | cygar, minor parsing still required, but delimited vs colum width |
| 18:50.40 | *** join/#asterisk darkskiez (i=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 18:51.38 | *** join/#asterisk menfin (n=cray@LNeuilly-152-21-127-49.w193-253.abo.wanadoo.fr) |
| 18:53.12 | SmaxTheFrog_ | is it possible to use * for time and weekday to match at all times in gotoiftime ? |
| 18:53.49 | Mercestes | SmaxTheFrog_, basically, yes. |
| 18:55.17 | cygar | [TK]D-Fender: Thanks a lot ! I didn't know concise :) |
| 18:56.29 | [TK]D-Fender | cygar, glad to help |
| 18:56.46 | Farz | Whats Intercom exactly? |
| 18:56.51 | Farz | like door bell intercom? |
| 18:56.55 | SmaxTheFrog_ | thx |
| 18:56.57 | Mercestes | Farz: exactly like that. |
| 18:57.23 | Mercestes | ok, not exactly. |
| 18:57.25 | Farz | it can be linked to FXS? |
| 18:57.32 | Mercestes | Intercom == management screaming interface. |
| 18:57.38 | [TK]D-Fender | Mercestes, BETTER :) |
| 18:57.41 | Mercestes | :D |
| 18:57.44 | Mercestes | Farz: sometimes. |
| 18:57.47 | *** join/#asterisk etfonhomey (n=etfonhom@74-140-213-69.dhcp.insightbb.com) |
| 18:58.04 | etfonhomey | [TK]D-Fender, any luck with SLA? |
| 18:58.24 | [TK]D-Fender | etfonhomey, Nope, and I'm stuck getting basic PRESENCE to function again. |
| 18:58.41 | Farz | hmmm i better put Intercom in my plan :) |
| 18:58.44 | darkskiez | I'm so depressed, upgraded 1.2 to 1.4 today and my E1 Pri stopped working without error :( it just stopped receiving pri commands from it. |
| 18:58.44 | [TK]D-Fender | etfonhomey, Would have been easier had I updated earlier when these topice were more common around here |
| 19:00.42 | Farz | Mercestes, thanks for the online shops you gave me. :) |
| 19:00.46 | etfonhomey | [TK]D-Fender, yeah, when I asked about PRESENCE on here over the weekend when you weren't on, I got a "I would help you, but it's been asked on here so many times and it's well document." |
| 19:02.47 | ManxPower | etfonhomey: I would suggest the mailing list archives and the Wiki. That is how *I* figured out how to do presence |
| 19:02.49 | Mercestes | Farz: Np. :) |
| 19:03.33 | ManxPower | darkskiez: what in the world made you want to upgrade to 1.4? |
| 19:04.25 | [TK]D-Fender | ManxPower, we're talking 1.4 which introduced some men hiccups into the process. the normal 1.2 way does not seem to work as-is |
| 19:04.37 | [TK]D-Fender | mean* |
| 19:04.37 | etfonhomey | Well, I got it most of the way working. |
| 19:04.42 | Farz | anyone tried Grandstream IP phones? |
| 19:04.54 | [TK]D-Fender | Farz, .... |
| 19:04.55 | [TK]D-Fender | ~gs |
| 19:04.56 | jbot | i guess gs is South Georgia and the South Sandwich islands, or ghostscript. GrandSuck phones are cheap junk which should be avoided with extreme prejudice |
| 19:05.07 | ManxPower | I'll stick to 1.2 thankyouverymuch |
| 19:05.20 | etfonhomey | An Xlite registration was the one I was monitoring. With Xlite running, it said I was "Idle" and when I closed out Xlite, it said I was "Unavailable" |
| 19:05.29 | [TK]D-Fender | ManxPower, understandable. I just took the plunge today and am suffering from oxygen deprivation.... |
| 19:05.31 | Farz | Gulp! |
| 19:05.33 | darkskiez | ManxPower: I wanted to use gtalk integration, and was hoping the new rtp code would work better on our dodgy internet connection. |
| 19:05.54 | etfonhomey | I could never get it to tell me any other states. |
| 19:06.34 | Farz | ok, whats the recommended and affordable IP phones? |
| 19:06.44 | etfonhomey | Polycom |
| 19:07.07 | Farz | k ty :) |
| 19:08.25 | ManxPower | Farz: that would depend on how you define "affordable". If you compare the price of an IP phone to a cheap walmart phone then there are NO affordable IP phones that work well. If you compare the price of an IP phone to what a traditional PBX phone would cost then pretty much every IP phone out there is "affordable" |
| 19:08.35 | *** join/#asterisk bawb2 (n=bawb2@ip51051.estcmp.ku.edu) |
| 19:08.42 | Qwell[] | ManxPower: I am so quoting you on that. |
| 19:08.51 | ManxPower | Qwell[]: 8-) |
| 19:09.13 | Farz | was wondering what makes Grandstream so sucky |
| 19:09.48 | *** join/#asterisk Corydon76-home (i=five@pdpc/supporter/sustaining/Corydon76-home) |
| 19:09.48 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 19:10.01 | ManxPower | Farz: I don't think they have released a stable firmware for their hardware since the company started. If you use GS products you basically have to just keep trying different firmware verisons until you find one that works in your enviroment. |
| 19:11.01 | [TK]D-Fender | Farz, low qaulity crap witha cheap feel, low usability, poor LCD usage, flakey firmware, etc. |
| 19:11.10 | Farz | firmware issues are truely scary |
| 19:11.17 | [TK]D-Fender | Farz, ... |
| 19:11.19 | [TK]D-Fender | ~phones |
| 19:11.20 | jbot | it has been said that phones is http://bani.anime.net/phones/. SIP Hardphones in order of quality/suggestability : Polycom (any), Aastra 480i, Cisco 7940(+), Linksys SPA-9XX, Snom |
| 19:11.25 | polerin | just out of curiousity, how bad propritary are most of the avaya phones? (I have to deal with them at work) |
| 19:12.00 | *** join/#asterisk eltech (i=G00Ds@ool-457c94a3.dyn.optonline.net) |
| 19:12.02 | Farz | GS it not even listed Lol |
| 19:12.32 | wunderkin | polycom -ip430 :P |
| 19:12.46 | Qwell[] | jbot_: no, phones is http://bani.anime.net/phones/. SIP Hardphones in order of quality/suggestability : Polycom (any), Aastra 480i, Cisco 7940(+), Linksys SPA-9XX, Snom, everything else on the market, Grandstream |
| 19:12.48 | *** join/#asterisk sevard (n=sev@c-76-102-2-4.hsd1.ca.comcast.net) |
| 19:12.49 | Farz | ty gonna look it up |
| 19:14.00 | darkskiez | Qwell: lol, harsh |
| 19:14.30 | *** join/#asterisk flujan_ (n=Marketin@200-160-115-020.static.spo.ctbc.com.br) |
| 19:14.38 | flujan_ | [TK]D-Fender, hi!!!! |
| 19:14.44 | flujan_ | hi all... |
| 19:14.53 | Mercestes | lmao |
| 19:14.57 | flujan_ | guys, I need to put asterisk to work with a sip provider |
| 19:15.10 | flujan_ | My users connects in asterisk and asterisk dials to the proxy |
| 19:16.04 | [TK]D-Fender | qwell : lol.... ACCURATE ;) |
| 19:17.23 | Mercestes | D-Fender: Your one of those "loud 'n proud" types aren't ya? |
| 19:17.38 | flujan_ | here goes the information of sip.conf extensions.conf and the error |
| 19:17.39 | syzygyBSD | hmm, so polycom > cisco I just got? |
| 19:17.39 | flujan_ | http://pastie.caboo.se/52608 |
| 19:17.55 | syzygyBSD | I have both on my desk, can't configure the cisco... :( |
| 19:17.55 | Mercestes | syzygyBSD, in a word. yes. |
| 19:18.10 | Farz | what about WI-FI SIP mobile phones? |
| 19:18.12 | [TK]D-Fender | syzygyBSD, Yes, as you'd likely hear every time asked in here. |
| 19:18.13 | flujan_ | having problems with the registry... |
| 19:18.21 | *** join/#asterisk ToyMan (n=Stuart@74-32-55-210.dsl1.mdl.ny.frontiernet.net) |
| 19:18.32 | flujan_ | could you help me ? |
| 19:18.33 | syzygyBSD | hmm, I will have to compare the two |
| 19:18.38 | [TK]D-Fender | Farz, Wifi phones suck. All of them. HARD. Some only slightly less than others. |
| 19:18.40 | Mercestes | register teste@200.170.136.196 There's your first problem. Should be plural.\ |
| 19:19.03 | syzygyBSD | lol Mercestes maybe he is french |
| 19:19.10 | Farz | bad voice quality? |
| 19:19.14 | Mercestes | heh, that explains the singularity. :D |
| 19:19.16 | flujan_ | Mercestes, teste is the password the sip proxy provide me |
| 19:19.23 | GreyFoxx | Farz: Avoid the utstarcom, the linksys WIP300, and the dlink one based on the same ahrdware as the WIP300 ... all suck |
| 19:19.28 | Mercestes | flujan_, n/m. hehe. |
| 19:19.34 | flujan_ | I am from Brazil. :D |
| 19:19.41 | Farz | so maybe Hitachi? |
| 19:20.06 | GreyFoxx | bad software, poor interface, shoddy construction, poor range, and with the WIP300's, the mics pick up your voice out of the speaker causing an echo |
| 19:20.26 | Farz | crap! Lol |
| 19:20.27 | [TK]D-Fender | farz, itachi is slightly better than the rest, but I highly recomment ATA + standard cordless for a single site phone. |
| 19:20.40 | GreyFoxx | yeah, definately |
| 19:20.43 | Farz | yeah i agree with that |
| 19:20.47 | [TK]D-Fender | GreyFoxx, add to that most cant transfer, conference or do much else of any use |
| 19:20.53 | GreyFoxx | yup |
| 19:21.01 | GreyFoxx | and way to easy to lockup the interface |
| 19:21.16 | Farz | so what the best ATA quality-wise? |
| 19:21.21 | [TK]D-Fender | Farz, its official.. you are systematically shopping down the aisle of DOOM. |
| 19:21.25 | Farz | Cisco 181? |
| 19:21.35 | Farz | i know Lol |
| 19:21.36 | etfonhomey | Farz, I had some wifi phones and returned them after a week. (UTStarcomm F3000) |
| 19:21.43 | flujan_ | Mercestes, any idea? |
| 19:21.43 | GreyFoxx | I'd like to get my hands on the Nokia N93 cellphone, which apparently has a SIP stack and can be gotten to work with asterisk when in wifi range |
| 19:21.47 | Mercestes | flujan_, Try qualify=1000 just for kicks. that error seems counter intutive tho. not sure what it means by interval too brief. |
| 19:21.54 | Mercestes | flujan_, Did you google that error? |
| 19:22.01 | flujan_ | yeap |
| 19:22.16 | [TK]D-Fender | Farz, Linksys ATA's are fine for the most-part, Mediatrix is rather complicated but VERY powerful. Priced to match... then there's the Meditrix & AudioCodes mass gateways if you're looking for a lot of posrt. |
| 19:22.19 | [TK]D-Fender | ports* |
| 19:22.28 | Farz | GreyFoxx, is it better than the rest? |
| 19:22.31 | Mercestes | flujan_, any hints? |
| 19:22.39 | syzygyBSD | can anyone tell me what the hell I have to do to configure this cisco 7960 I just got? |
| 19:22.51 | GreyFoxx | Farz: I've been in love with the N93 since I first saw it. But I've never used it personally |
| 19:22.56 | [TK]D-Fender | Farz, avoid Cisco for pretty much anything VoIP with regards to *. |
| 19:23.05 | Mercestes | flujan_, I'd try a qualify=1000 and a dial(${exten}|5000) or something for good measure just to try and forcefeed it long intervals just to try and make the error go away. if not, something else is wrong. |
| 19:23.17 | Mercestes | syzygyBSD, configure it. |
| 19:23.31 | syzygyBSD | lol, through which method |
| 19:23.35 | Farz | then what should i get for a simple 1 or 2 FX? ATA? |
| 19:23.38 | Farz | PAP2 maybe? |
| 19:23.46 | [TK]D-Fender | Farz, Linksys SPA-2102 |
| 19:23.54 | Farz | which is PAP2? |
| 19:23.56 | flujan_ | Mercestes, http://bugs.digium.com/view.php?id=7254 |
| 19:23.58 | Mercestes | syzygyBSD, tftp |
| 19:23.58 | [TK]D-Fender | PAP2 is a cheap lower quality version of the SPA series. |
| 19:24.16 | syzygyBSD | ya, cant get the phone to recognize the correct one :( |
| 19:24.22 | syzygyBSD | oh well |
| 19:24.23 | Farz | how many ports with SPA-2102? |
| 19:25.57 | Mercestes | flujan_, Is this the same carrier? Looks like you need this patch |
| 19:26.11 | brad_mssw | I'm running asterisk 1.2 with my dialplan in AEL, and looking to move to 1.4 ... anyone know of a AEL->AEL2 migration FAQ of any sort? I see a statement which says "It's upward compatible with the original AEL, but adds several features, and removes some limititations.", hoping that's entirely true |
| 19:26.13 | flujan_ | Mercestes, aparently I am not registered in the sip proxy |
| 19:26.47 | Mercestes | flujan_, That would be what this message means. |
| 19:27.11 | etfonhomey | Anyone know an xml editor for linux that's like XML Notepad 2007? (for editing Polycom config files) |
| 19:27.40 | flujan_ | Mercestes, X-Lite just works with this configuration... |
| 19:27.57 | flujan_ | I want asterisk to work like a sip phone... |
| 19:28.02 | flujan_ | is my configuration correct? |
| 19:28.15 | Mercestes | flujan_, Did you try the changes I suggested? |
| 19:28.26 | flujan_ | qualify-1000 |
| 19:28.33 | flujan_ | does not change nothing... :( |
| 19:28.38 | Mercestes | Try 9000 |
| 19:28.47 | Mercestes | wait. |
| 19:28.51 | Mercestes | that's probably not helping. |
| 19:29.01 | codefreeze | brad_mssw: One way to tell for sure: compile a 1.4, and use its aelparse to check your dialplan... you don't even have to install 1.4 to see what the issues will/might be. |
| 19:29.20 | Mercestes | flujan_, Is the provider in question the same as the provider in this bugreport? |
| 19:29.38 | flujan_ | no |
| 19:29.44 | brad_mssw | codefreeze: aelparse is a separate application |
| 19:29.46 | brad_mssw | ? |
| 19:29.48 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 19:29.49 | Mercestes | flujan_, It' slikely you still need this patch. |
| 19:30.13 | Mercestes | flujan_, You can alter your min_expirey to be a much higher number and alot closer to max_expirey to try and work around. |
| 19:30.30 | *** join/#asterisk squirrellb (n=squirrel@200.56.251.199) |
| 19:30.45 | squirrellb | hi everyone |
| 19:31.01 | squirrellb | does anyone know Rene Mendoza??? |
| 19:31.06 | Mercestes | heyas, Squirrell. you any relation to russellb? |
| 19:31.12 | squirrellb | mmm |
| 19:31.13 | squirrellb | nope |
| 19:31.21 | Mercestes | oh, you have the same last name. |
| 19:31.30 | squirrellb | it´s probably |
| 19:32.09 | squirrellb | hey mercestes, did you know Rene |
| 19:32.10 | squirrellb | ?? |
| 19:32.21 | *** join/#asterisk shinux__ (n=shinux@208.70.5.150) |
| 19:32.21 | Mercestes | Rene Mendoza the programmer? |
| 19:32.27 | squirrellb | yes |
| 19:32.33 | Mercestes | No, don't know her. |
| 19:32.44 | squirrellb | mmm |
| 19:32.47 | squirrellb | to bad |
| 19:32.52 | squirrellb | hehehehe |
| 19:32.58 | Mercestes | I have an email address. |
| 19:33.08 | Mercestes | why do you ask? |
| 19:33.34 | squirrellb | he was the programmer in a project inthe company i work |
| 19:33.48 | squirrellb | then i lost all contact with him |
| 19:33.58 | squirrellb | now i got a project in mexico |
| 19:33.58 | anonymouz666 | can you help to think a logic about reading dynamic values and building extensions in another context at the same time based on this values |
| 19:34.14 | squirrellb | and need someone wgood like him |
| 19:34.21 | squirrellb | in asterisk |
| 19:34.48 | Mercestes | rene@NO-SPAM.publicidadefectiva.com |
| 19:35.03 | [TK]D-Fender | anonymouz666, Sorry, that dialect of gibberish will not get you far in here..... |
| 19:35.05 | Mercestes | I'm pretty good with asterisk too tho |
| 19:35.27 | Mercestes | anonymouz666, for only $250 an hour. yes |
| 19:35.58 | squirrellb | jejeje |
| 19:36.01 | squirrellb | in mexico??? |
| 19:36.05 | Mercestes | I work remotely. |
| 19:36.11 | Mercestes | the great thing about ssh and the Internet. |
| 19:36.19 | Mercestes | but if you pay me enough I'll be in Mexico |
| 19:36.25 | Mercestes | I hear Monterrey is nice this time of year. |
| 19:36.39 | squirrellb | that´s the problem i need someone who can talk with the technical personel of my client |
| 19:36.41 | *** join/#asterisk dc3aes (n=matt@S01060001023fe8ca.no.shawcable.net) |
| 19:36.53 | squirrellb | a litle hot to me |
| 19:36.59 | squirrellb | but yes it´s nice |
| 19:36.59 | Mercestes | That's the great thing about telephones. |
| 19:37.07 | squirrellb | lol |
| 19:37.45 | Mercestes | but I'm guessing from this convo that this doesn't pay me enough to go to Mexico. |
| 19:37.51 | squirrellb | and do you speak spanish??? |
| 19:38.02 | Mercestes | Do I speak spainish? |
| 19:38.03 | anonymouz666 | lol. calm down. I didn't ask for a complete solution. Just a orientation to put me in the right way. |
| 19:38.55 | Mercestes | <PROTECTED> |
| 19:39.09 | [TK]D-Fender | anonymouz666, and I'm saying your question was completly incomprhensible. Try rephrasing it. |
| 19:39.09 | squirrellb | de donde eres? |
| 19:40.06 | anonymouz666 | [TK]D-Fender: I have a dynamic loop. this can go from 1 to 100 or from 1 to 1000. If its from 1 to 100 I need to build 100 extensions in another context |
| 19:40.17 | Mercestes | Mi gente construyó los templos sobre las tierras santas que construyes sobre y caminas sobre, nosotros adoraba el sol y las estaciones antes de que nos oprimieras y soy uno del pocos los descendientes que viven de los aztecs. Por supuesto la lengüeta de mi paÃs de la madre. |
| 19:40.54 | drfreeze | squirrellb: http://media.ebaumsworld.com/retro.wmv |
| 19:40.56 | squirrellb | ok |
| 19:41.03 | squirrellb | disculpa tengo que salir un rato |
| 19:41.08 | squirrellb | mi esposa me espera para comer |
| 19:41.15 | squirrellb | espero estes aqui cuando regrese |
| 19:41.31 | anonymouz666 | [TK]D-Fender: I am thinking something using a global val |
| 19:42.02 | squirrellb | pd. no me contestaste de donde eres, tu texto dice mucho, y al mismo tiempo no dice nada |
| 19:42.03 | squirrellb | saludos |
| 19:42.19 | [TK]D-Fender | anonymouz666, I fail to understaand how you mar MAKING extensions within your dialplan... |
| 19:42.50 | Mercestes | Soy de Carolina del Norte. |
| 19:43.00 | Mercestes | saludos. |
| 19:43.46 | anonymouz666 | [TK]D-Fender: I still don't make the extensions |
| 19:43.53 | ManxPower | [TK]D-Fender: Each extension must be lovingly crafted from bits harvested from rain forests, of course. |
| 19:44.06 | *** join/#asterisk Vec (n=Vec@dsl-244-208-173.telkomadsl.co.za) |
| 19:44.11 | [TK]D-Fender | shdfasfdkllhfaklsdhfaskldf |
| 19:44.14 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 19:44.16 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 19:44.42 | anonymouz666 | I am reading values through CURL... |
| 19:44.57 | anonymouz666 | so I need to building extensions in another context |
| 19:45.48 | blitzrage | ManxPower: lol |
| 19:46.16 | anonymouz666 | because If I read from 1 to 100 if user types 101 I need to report it as wrong extension |
| 19:46.45 | xheliox | Greetings all. :) Anyone using SIP clents as members of queues using call-limit, and ringinuse = no? I'm using 1.4.2 with that same setup, and it seems that occasionally/randomly Asterisk will stop sending that client calls and "show queues" says that channel is in use, when it's not. |
| 19:47.20 | xheliox | sip show channels shows a constant INVITE for both fo the clients are being a problem (at the moment), I'm not sure that has anything to do with anything, however. |
| 19:48.26 | anonymouz666 | [TK]D-Fender: got it? |
| 19:48.36 | [TK]D-Fender | anonymouz666, No, I don't |
| 19:48.38 | anonymouz666 | or it is still obscure |
| 19:49.29 | anonymouz666 | I get a dynamic list through CURL and save into a var. Set(var=Value1,Value2,Value3) |
| 19:50.02 | anonymouz666 | then I have to make a loop using fieldqty(). |
| 19:50.11 | Mercestes | anonymouz666, exten => i,1,Playback(tt-monkeys) |
| 19:50.13 | codefreeze | brad_mssw: yep, itsa standalone. It will read from /etc/asterisk/extensions.ael, and tell you if you got any errors/warnings. |
| 19:50.26 | anonymouz666 | after that I need to do something 1 for value1, 2 for value2, 3 for value3 |
| 19:50.31 | anonymouz666 | this is ok |
| 19:50.40 | anonymouz666 | my problem is, to handle the invalid options |
| 19:50.45 | Mercestes | exten => i,2,Goto(i,1) |
| 19:51.11 | anonymouz666 | ohh it's dynamic |
| 19:51.13 | [TK]D-Fender | 10 PRINT "I AM GOING INSANE!" 20 GOTO 10 |
| 19:51.23 | Mercestes | exten => i,1,System(sudo dd if =.....ahh nevermind ManxPower already yelled at me about this =/) |
| 19:51.26 | brad_mssw | codefreeze: cool, thanks, i'll give that a shot |
| 19:51.41 | [TK]D-Fender | Mercestes, now THATS funny! |
| 19:51.45 | Mercestes | lol |
| 19:51.46 | anonymouz666 | Mercestes: if 4 is invalid now, in the next list can be a valid value! |
| 19:51.56 | Mercestes | anonymouz666, yes. |
| 19:52.22 | Mercestes | [TK]D-Fender, peek 254,0 |
| 19:52.43 | anonymouz666 | Mercestes: yes and extension i does not work in this case |
| 19:53.10 | [TK]D-Fender | anonymouz666, your thought, instead of coming out like a ripe tomato are instead coming out as CHUNKY SALSA. Pastbin whatever psychotic dialplan you have created and let us figure it out by deciphering your code. I'm wagering better odd that your description :| |
| 19:53.29 | *** join/#asterisk Mahmoud (n=fake@unaffiliated/mahmoud) |
| 19:53.48 | jnc | when making a SIP call between extensions using Ekiga softphone SIP client, the DTMF isn't working and also the first second or two of the audio gets clipped on new calls |
| 19:53.57 | jnc | is that a client issue or something I need to tweak in asterisk? |
| 19:54.05 | *** join/#asterisk darken_darken (n=marco@10.154.77.83.cust.bluewin.ch) |
| 19:54.23 | [TK]D-Fender | jnc, describe what part is "not working" in there.... |
| 19:55.03 | etfonhomey | uh, oh, is this brian under a new name? |
| 19:55.24 | jnc | [TK]D-Fender: i'm dialing my extension to check voicemail as a test, it's working a lot better than yesterday. I get audio and the voicemail prompts now, though I notice "Mailbox" audio prompt gets clipped off initially |
| 19:55.55 | jnc | using Ekiga to dial DTMF, no joy. I think that much at least is a client issue |
| 19:56.09 | anonymouz666 | [TK]D-Fender: anyway thanks for trying to help. I think it will take more time to you to understand the whole dialplan |
| 19:56.22 | ManxPower | AHA! I have found the Bellsouth standard test numbers! |
| 19:56.33 | ManxPower | I shall beat them with their own tariffs! |
| 19:56.35 | Mercestes | Yes! publish them! |
| 19:56.58 | ucfMethod | test numbers? dtmf test numbers? sweet pass em out please |
| 19:58.01 | J4k3 | haha ManxPower sounds as excited as I was when I discovered the test #'s at the co. |
| 19:58.15 | J4k3 | ... with that I also found the phone that rings inside the CO. heh... wanna get something fixed? call the CO! |
| 19:58.24 | ManxPower | ucfMethod: no, test numbers as in the lowest allowed S/N, the max allowed lost, etc |
| 19:59.18 | etfonhomey | ManxPower: that's what I was hoping for, too. |
| 19:59.18 | [TK]D-Fender | jnc, you mean you can't enter a mailbox #? |
| 19:59.26 | etfonhomey | ManxPower: Are you in Nashville? |
| 19:59.35 | jnc | [TK]D-Fender: correct :P |
| 19:59.36 | etfonhomey | I remember somone on here being from there. |
| 19:59.58 | jnc | [TK]D-Fender: there's a bug or something apparently in opal / ekiga in conjunction with asterisk 1.4 |
| 20:00.04 | jnc | updating now to see if that is a fix |
| 20:00.30 | pfn | woot, got my block of 100 408 DIDs :) |
| 20:00.42 | [TK]D-Fender | jnc, news to me.... |
| 20:00.52 | *** join/#asterisk adam1 (n=adasdfas@d150-208-34.home.cgocable.net) |
| 20:01.22 | [TK]D-Fender | pfn, World domination! |
| 20:01.34 | adam1 | Help! I installed Trixbox 2.0 onto one of my boxes and it's missing asterisk and zaptel from the install how do I get them to install? I tried yum install asterisk but no go |
| 20:02.47 | etfonhomey | Type "/join #trixbox" |
| 20:02.56 | jnc | etfonhomey: #freepbx actually |
| 20:03.07 | pfn | [TK]D-Fender, that thought had crossed my mind |
| 20:03.39 | etfonhomey | jnc, doh! I'm just posing like I know something anyway. |
| 20:03.48 | *** join/#asterisk shinux__ (n=shinux@208.70.5.150) |
| 20:03.52 | jnc | I made the same mistake the other day |
| 20:07.11 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 20:07.31 | *** join/#asterisk henryv2 (n=henry_vo@82-39-113-10.cable.ubr03.newy.blueyonder.co.uk) |
| 20:09.35 | henryv2 | hi, I'm hoping for a bit of advice on how to setup IRQs on a CentOS linux system. I have installed an x100p in my Athlon MP system and disabled the serial and parallel ports to ensure there are spare IRQs. |
| 20:10.11 | henryv2 | I manually configured the IRQs for the ethernet card and the x100p to 10 and 11 which were free, but when linux loads and i do cat /proc/interrupts i get the following results: |
| 20:10.19 | henryv2 | 169: 2462 2502 IO-APIC-level ide2, ide3 |
| 20:10.19 | henryv2 | 177: 1956 13 IO-APIC-level eth0 |
| 20:10.19 | henryv2 | 185: 489737 488904 IO-APIC-level wcfxo |
| 20:10.42 | henryv2 | do interrupts of > 16 mean that the same IRQ is in use? |
| 20:10.51 | [TK]D-Fender | henryv2, that looks fine |
| 20:11.59 | jnc | henryv2: interrupt sharing is usually okay |
| 20:12.01 | henryv2 | [TK]D-Fender, it does work and I can run zttest -v which results in a consistant score of 99.97%. I was hoping to get this up to 99.98 or 100% if the lower score was being caused by sharing an IRQ. |
| 20:13.13 | ManxPower | henryv2: wcfxo is not sharing an irq |
| 20:13.42 | ManxPower | henryv2: manually setting the IRQ only works in XT-PIC mode. in APIC mode, the system picks the IRQs |
| 20:13.47 | henryv2 | what do the irqs of 169, 177 and 185 mean? |
| 20:14.20 | Hmmhesays | how long does canned soup stay edible? |
| 20:14.46 | rudholm | practially forever |
| 20:14.46 | henryv2 | Hmmhesays: I'd be suspicious after 1 year :-D |
| 20:14.46 | rudholm | as long as the seal isn't breached |
| 20:15.15 | Hmmhesays | I think I would probably go with rudholms answer |
| 20:15.19 | jnc | if the can is bulging, toss it in the garbage |
| 20:15.39 | Hmmhesays | there if there are no outside contaminates there should be nothing able to grow in there |
| 20:15.40 | jnc | sometimes bacteria can form in products that aren't 100% pasteurized |
| 20:16.03 | jnc | not all canned products stay forever though |
| 20:16.14 | Hmmhesays | soup I'm thinking will last a long long time |
| 20:16.31 | jnc | true, you do mention soup |
| 20:16.44 | Hmmhesays | especially since you sterilize it when you cook again |
| 20:16.56 | [TK]D-Fender | Hmmhesays, they have BBF dates on them. Go read! |
| 20:17.04 | jnc | say what is " == Connect attempt from '127.0.0.1' unable to authenticate" repeatedly doing when i set verbosity to 3 ? |
| 20:17.06 | Hmmhesays | yes I know that |
| 20:17.16 | jnc | is that the odbc stuff? |
| 20:17.27 | jnc | (I don't have any cdr database stuff set up that I know of) |
| 20:17.49 | [TK]D-Fender | Hmmhesays, across 2 types of Cambells I have here, apparently about 2 years |
| 20:17.57 | jnc | there's a follow up line too, "[Apr 9 15:30:11] NOTICE[15182]: manager.c:1002 authenticate: 127.0.0.1 tried to authenticate with nonexistent user 'user'" |
| 20:18.26 | [TK]D-Fender | jnc, says "stop using a stupid GUI, or learn how to auth it right!" |
| 20:18.36 | jnc | heh |
| 20:18.46 | jnc | welll yes there is no user 'user' |
| 20:18.50 | pfn | henryv2, wow, I get 99.975% when I use ztdummy :p |
| 20:18.53 | rudholm | "best by" and "expiration" dates are the same thing, fwiw |
| 20:18.58 | jnc | what is asking for it to auth with user 'user' though? |
| 20:19.01 | jnc | the web browser? |
| 20:19.09 | henryv2 | pfn: same here, do you find it works well? any voice quality problems? |
| 20:19.31 | henryv2 | pfn: what do you get when you use a hardware timer? |
| 20:19.33 | pfn | it only matters for meetme... and no, I haven't tried using meetme enough to tell |
| 20:19.38 | pfn | I get 99.987% |
| 20:19.48 | henryv2 | my application (vicidial) uses meetme heavily :-D |
| 20:19.52 | pfn | Opened pseudo zap interface, measuring accuracy... |
| 20:19.52 | pfn | 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 100.000000% 99.987793% |
| 20:20.09 | henryv2 | which card is that? |
| 20:20.16 | pfn | henryv2, just test it, I don't think 99.975% is a big deal... |
| 20:20.21 | henryv2 | it could be that this x100p is pants |
| 20:20.59 | *** join/#asterisk Defraz (n=t0tal@fw.fuzecore.com) |
| 20:21.00 | rudholm | Hmmhesays: http://www.foodreference.com/html/tcannedfoodshelflife.html |
| 20:21.12 | THX2000 | anyone have experience running asterisk on the WRAP? |
| 20:22.31 | [TK]D-Fender | henryv2, besides jsut being a number, what does this zttest score mean to you in the REAL WORLD? |
| 20:22.46 | Hmmhesays | yeah that says my soup is ok to eat |
| 20:22.52 | *** join/#asterisk cocomp (n=jeremy@82-43-235-140.cable.ubr02.pres.blueyonder.co.uk) |
| 20:23.10 | rudholm | Hmmhesays: yes, just like I said :-) |
| 20:23.18 | bulle | <PROTECTED> |
| 20:23.21 | bulle | damn |
| 20:23.32 | rudholm | I'm me! |
| 20:23.44 | rudholm | what would you like to know? :-) |
| 20:23.48 | bulle | ye, i was thinking your name might imply you are scandinavian |
| 20:23.58 | rudholm | it does imply that |
| 20:24.07 | bulle | and if you were, i would suggest that you should send a can of surströmming to Hmmhesays |
| 20:24.14 | pfn | hmm, how useful is asterisk-java? |
| 20:24.19 | pfn | it's something I might want to use... |
| 20:24.23 | rudholm | well, I live in Los Angeles, not sure I can find it. |
| 20:24.41 | bulle | rudholm: he, i think it would be an experience for Hmmhesays, a new type of canned food |
| 20:24.44 | *** join/#asterisk Exhar (n=Roy@213-73-139-87.cable.quicknet.nl) |
| 20:24.53 | Mercestes | LA, zomg, I was there not long ago |
| 20:25.22 | henryv2 | i have setup a few vicidial systems on different hardware (with and without the x100p, on normal hardware and on vmware) and have variable voice quality and variable zttest scores. I was just looking for some anecdotal feedback on how it might work out in production systems using vicidial where the timing is quite important because of the high load and dependency on meetme |
| 20:25.24 | rudholm | but yes, I'm a citizen of The Kingdom of Sweden, fwiw. |
| 20:25.32 | bulle | rudholm: ye, me to fwiw |
| 20:25.38 | Mercestes | I forgive you. |
| 20:25.50 | bulle | Mercestes: you wouldnt, if you had ever tasted surströmming |
| 20:26.21 | [TK]D-Fender | henryv2, In short, the X100P is a flaming pile of [expleteve deleted]. You want quality, be prepared to pay for it. |
| 20:26.58 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
| 20:28.23 | pfn | it's just a simple 8khz crystal generating interrupts.... |
| 20:28.25 | Mercestes | bulle, wouildn' tmind trying it once. |
| 20:29.29 | Mercestes | I ate pho' once. I lived through that. |
| 20:29.29 | henryv2 | [TK]D-Fender: i was also wondering if people had experience of running zttest (i know i keep harping on about the test, but it's the only objective assessment i know of) on an x100p and also on a higher quality card in the same machine? any differences in the timing ability of the cards? (even if the rest of the facilities are vastly different) |
| 20:29.30 | florz | 8 kHz crystal, sure ... =:-) |
| 20:29.32 | *** join/#asterisk Strom_M (n=strom@70.141.71.195) |
| 20:29.42 | pfn | henryv2, my understanding in the past is that irq sharing is a bad idea for the timing device |
| 20:29.45 | pfn | don't know about now |
| 20:29.54 | [TK]D-Fender | henryv2, all the "objective" tests in the world don't mean much whn you can sum up in 2 seconds "thi soulds like [expletive deleted]" |
| 20:30.00 | Mercestes | that's changed with the latest zaptel drivers (supposedly) |
| 20:30.05 | [TK]D-Fender | henryv2, Let go! |
| 20:30.13 | henryv2 | I'm not actually using the x100p for calling, just timing since the calls will occur on voip! |
| 20:30.27 | Mercestes | then why not use ztdummy? |
| 20:30.40 | Hmmhesays | theres a wild idea |
| 20:30.54 | henryv2 | ztdummy doesn't work very well on vicidial because it uses meetme heavily (apparently?) |
| 20:31.10 | pfn | henryv2, you haven't controlled your environment, nor specified which environments have poor meetme quality |
| 20:31.21 | pfn | you've mentioned you also use VMs |
| 20:31.41 | henryv2 | yeah, i have tried a few different environments for testing. the VMs were totally rubbish! |
| 20:31.53 | Hmmhesays | rubbish, you a brit? |
| 20:31.56 | henryv2 | yep |
| 20:31.56 | Hmmhesays | A BLOODY BRIT! |
| 20:31.59 | Hmmhesays | lol |
| 20:32.02 | jnc | couple of things you want to avoid sharing IRQs with, the video adapter, and USB |
| 20:32.05 | henryv2 | :-) |
| 20:32.06 | jnc | avoid like the plague |
| 20:32.08 | Hmmhesays | tell my why voipjet doesn't terminate 077 numbers |
| 20:32.11 | pfn | as well, in the past, running an X11 server also caused problems |
| 20:36.22 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 20:36.23 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- Asterisk 1.4.2 (Mar. 19, 2007), Asterisk 1.2.17 (Mar. 19, 2007), Zaptel 1.2.16 (Mar. 19, 2007), Zaptel 1.4.1 (Mar. 23, 2007) -=- Other fun channels: #asterisk-gui, #asterisknow, #asterisk-commits -=- Join #freepbx for freepbx/trixbox support. |
| 20:36.30 | Mercestes | I guess I'm a "red hat hobbist" then because I have to use the GUI to do anything in Redhat |
| 20:36.33 | Hmmhesays | caio1982: it actually does work fine if you want to add a few skype routes to a pbx |
| 20:36.36 | caio1982 | Hmmhesays: i dont know if we're talkijng about the same chanskype.com... a brazilian telco uses that and I'm quite sure that would be too stupid for them in order to use the chan_skype module |
| 20:36.37 | [TK]D-Fender | Ok, its been a few hours since I last asked and some useful people are active : I jsut upgraded to * 1.4.2 and I am failing to get Presence function on my 2 Polycom phones. can someone point me in an immediately useful direct as to what it'll take to get this back and running like normal? |
| 20:36.58 | Mercestes | [TK]D-Fender, What did you upgrade from? |
| 20:36.59 | Hmmhesays | caio1982: it works fine |
| 20:37.02 | Farz | whats the max number of modules you can install on a TDM400P? |
| 20:37.10 | pfn | Farz, 4 |
| 20:37.10 | [TK]D-Fender | Mercestes, 1.2.17 |
| 20:37.15 | Hmmhesays | just keep the X server running it away from your main pbx |
| 20:37.16 | Farz | ty |
| 20:37.21 | *** join/#asterisk RazaMetaL (n=razameta@200.93.220.27) |
| 20:37.24 | RazaMetaL | hi guys |
| 20:37.27 | danp | [TK]D-Fender: i had that problem a while back when i first tried 1.4...i think i read somewhere that setting call-limit would fix it |
| 20:37.31 | [TK]D-Fender | Mercestes, Phones are on SIP 2.1.0 and were perfectly function brfore under * 1.2 |
| 20:37.33 | Farz | hi Raza |
| 20:37.42 | [TK]D-Fender | danp, to what? |
| 20:37.47 | danp | just something |
| 20:38.03 | RazaMetaL | it's possible to change the ttl on asterisk only for one peer ? |
| 20:38.03 | Hmmhesays | caio1982: you install the module locally, don't use their servers |
| 20:38.22 | caio1982 | Hmmhesays: that's terribly funny to hear, if I can say that :) |
| 20:38.22 | Farz | if you want 2 FXO ports and 8 FXS - whats the best card to choose in order to be cost effective? |
| 20:38.22 | caio1982 | you needing to run multiple X to use the chan_skype.so |
| 20:38.22 | danp | probably whatever you have callsPerLine set to would be a good start |
| 20:38.22 | caio1982 | hehe |
| 20:38.23 | Farz | Digium that is |
| 20:38.31 | pfn | Farz, the 24 port fxo/fxs thinger |
| 20:38.36 | pfn | or a tdm400p and the 24port fxs |
| 20:38.41 | Hmmhesays | caio1982: actually it is pretty ingenious of them, not having the source, only the shitty skype api to work with |
| 20:38.41 | Farz | the 2400 model? |
| 20:39.04 | Farz | TDM2400? |
| 20:39.06 | caio1982 | Hmmhesays: well, yeah |
| 20:39.06 | Hmmhesays | i mean they are using dbus for god sakes |
| 20:39.11 | Farz | whats a 24port? like gateway? |
| 20:39.17 | pfn | tdm2400p |
| 20:39.25 | Farz | ah ok |
| 20:39.26 | Farz | ty |
| 20:39.26 | Strom_M | hey guys - which sftp daemon would you recommend for debian? vstfpd? |
| 20:39.30 | Hmmhesays | but... like I said it works suprisingly well if you know what you are doing |
| 20:39.52 | Farz | Strom WinSCP |
| 20:39.54 | Mercestes | [TK]D-Fender, I would've stuck with 1.2.17. Dunno, nothign should've changed with presence... |
| 20:39.55 | Hmmhesays | if not don't bother you'll probably crash your phone system, start the server on fire, and piss off your boss |
| 20:40.00 | Strom_M | I said debian, Farz |
| 20:40.01 | Strom_M | and daemon |
| 20:40.04 | Strom_M | not windows client |
| 20:40.09 | e-milio | Hello all |
| 20:40.13 | danp | i definitely had that problem when i tried 1.4 |
| 20:40.22 | [TK]D-Fender | danp, yup, I set it to 10, and that seems to have solved it. Thanks |
| 20:40.22 | Farz | normal sftp client then :) |
| 20:40.43 | Strom_M | DAEMON |
| 20:40.44 | Strom_M | NOT CLIENT |
| 20:40.44 | Strom_M | moron |
| 20:40.44 | [TK]D-Fender | Mercestes, It was a hot topic as users started migrating. |
| 20:40.44 | danp | http://bugs.digium.com/view.php?id=8405 |
| 20:41.12 | danp | that's what i ran into...says it's closed but i haven't used 1.4 since then |
| 20:41.24 | Farz | sshd airhead |
| 20:41.37 | Farz | with root-jail |
| 20:41.51 | Farz | if you have a friggin clue what that means |
| 20:43.00 | [TK]D-Fender | strom_M : I use vsftpd for normal FTP, not SFTP, but it seems pretty functional and well respected. |
| 20:43.04 | Strom_M | thanks, [TK]D-Fender |
| 20:43.05 | *** join/#asterisk ghosto (n=user@cmodem-237-235.tricom.net) |
| 20:43.10 | Mercestes | Farz: I would definately suggest pissing off Strom_M as much as humanly possible. |
| 20:43.49 | Mercestes | Strom_M: I use sshd too tho. |
| 20:44.04 | danp | [TK]D-Fender: cool that it works. sucks that it's another option that has to be specificed for each phone, though (i assume). a hassle for realtime SIP config |
| 20:44.15 | Farz | i try ;) |
| 20:44.16 | [TK]D-Fender | Strom_M, Good for chroot-ing, limiting users, etc. I like it. CentOS has taught me about a number of different tools I might not have used otherwise. Coming from Slackware I'm used to ProFTPD which was good as well. I'm just kinda following the "corporate install-base" |
| 20:44.30 | froguz | sometimes, when i call to certain places using the FXO channel, there's a rare tone sequence (like a singing bird) when the call begins. at the 3rd sequence loop, Asterisk thinks the far end is busy and hangup. why? |
| 20:44.49 | [TK]D-Fender | danp, Yeah I suppose, but I can ENTIRELY live with these things as long as they're documented. |
| 20:44.51 | Mercestes | Strom_M: emerge -av sshd |
| 20:45.01 | Strom_M | i'm running debian and i already have sshd ;) |
| 20:45.04 | danp | [TK]D-Fender: yeah, that's true |
| 20:45.06 | [TK]D-Fender | danp, I sometimes wich I could apply for a job to completely renovate the WIKI :) |
| 20:45.16 | froguz | this is the CLI debug for my Zap channel: [ TYPE: Control (4) SUBCLASS: Busy (5) ] [Zap/4-1] |
| 20:45.20 | *** join/#asterisk Mavvie (n=edwin@ppp29-32.lns1.syd6.internode.on.net) |
| 20:45.21 | danp | yes, me too |
| 20:45.26 | *** join/#asterisk darken_darken (n=marco@10.154.77.83.cust.bluewin.ch) |
| 20:45.31 | e-milio | hello all |
| 20:45.53 | e-milio | where is a good place to look for a DID delivered via sip or iax ? |
| 20:46.02 | ghosto | hello all |
| 20:46.09 | rudholm | Strom_M: WinSCP |
| 20:46.13 | [TK]D-Fender | froguz, set "callprogress=no" for your zaptel channel |
| 20:46.37 | Strom_M | rudholm: thanks, but I think i'm going to use the DOS FTP client instead |
| 20:46.48 | rudholm | oh yeah, that works even better |
| 20:46.54 | rudholm | cuz it's command-line |
| 20:46.57 | [TK]D-Fender | Strom_M, I would personally recommend FileZilla as a client.... |
| 20:47.02 | froguz | [TK]D-Fender, do you know if that tone sequence belongs to an R2 line? |
| 20:47.16 | [TK]D-Fender | froguz, no clue. |
| 20:47.23 | froguz | thanks |
| 20:47.38 | ghosto | ok...earlier i was asking how to connect a Nortel Succession 1000s to Asterisk (i was under the nick gosht)...we went to our client to inspect the hardware and found no open T1 slots for us to do a T1 tie with our Asterisk |
| 20:47.52 | ghosto | so we were thinking of doing IP trunking...any ideas? |
| 20:48.47 | *** join/#asterisk CuriosCat (i=stian@dynamic172.noc.host.net) |
| 20:49.18 | rudholm | Strom_M: you don't even know what a root-jail is! :-P |
| 20:49.30 | Strom_M | true |
| 20:49.33 | rudholm | Strom_M: I pnz0r you! |
| 20:49.37 | Strom_M | my linux-fu is not fu enough |
| 20:49.38 | CuriosCat | Howdy all. I got my FXO card and I'm compiling the zaptel driver; my ultimate goal is to be able to have my system call me and read specific emails (outage alerts) back to me using festival text-to-speech. |
| 20:50.00 | pfn | that's not a question |
| 20:50.02 | CuriosCat | I'd appreciate some pointers on hw to set that up |
| 20:50.17 | pfn | that's still not a question |
| 20:50.51 | CuriosCat | "how do I output digitized audio to a phone line connected to an FXO card?" |
| 20:50.55 | rudholm | well, it's not a "linux" thing per se, it's common to most unix and unix-like OSes |
| 20:51.03 | [TK]D-Fender | Strom_M, most of the time FTP logins default to a *nix user's home dir. This does NOT prevent them from drilling down with their client to see stuff you probably don't WANT them to see. |
| 20:51.27 | [TK]D-Fender | Strom_M, chroot is where you basically set a folder as being the "rott" of what they can see. |
| 20:51.45 | [TK]D-Fender | Strom_M, Good for accounts used for provisioning phones, etc |
| 20:51.50 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 20:51.55 | rudholm | Strom_M: and this can be done either at the OS level (safer) or the application level (less safe) |
| 20:52.02 | Strom_M | ah, ok |
| 20:52.03 | pfn | eh? you don't provision phones outta tftp? |
| 20:52.07 | Strom_M | that makes sense |
| 20:52.11 | [TK]D-Fender | pfn, tftp=ass |
| 20:52.26 | [TK]D-Fender | pfn, I like leading a NON trivial life :) |
| 20:52.27 | rudholm | Strom_M: you can actually run the ftp(s) daemon in a "chroot"ed environment, so even it can't escape |
| 20:53.08 | [TK]D-Fender | rudholm, and you'd chroot it to what, /home basically? |
| 20:53.08 | rudholm | Strom_M: some ftp(s) daemons support creating a false "root" for the purposes of serving documents |
| 20:53.09 | ghosto | anyone know how to connect a Nortel to an Asterisk via IP trunking? |
| 20:53.50 | pfn | [TK]D-Fender, well, what phones support stuff other than tftp? cisco is still tftp-only, isn't it? |
| 20:54.11 | [TK]D-Fender | ghosto, your questions are Nortel specific, you should go read some Nortel docs or ask a tech |
| 20:54.17 | rudholm | well, you often have to set up the chroot environment if you're going to do it with the OS --often apps that run in it will need access to some system files (that are not in /home) |
| 20:54.24 | danp | polycom does FTP and HTTP |
| 20:54.28 | pfn | speaking of which, how much do the 7960g go for these days, I have a spare I need to sell |
| 20:54.33 | [TK]D-Fender | pfn, and nowhere do I ever suggest Cisco. Almost every other make can provision through FTP, SFTP, HTTP, HTTPS, and all sorts of other nicer ways |
| 20:54.35 | danp | rudholm: depends on the FTP server; if it handles "ls" and things internally there's no setup required |
| 20:54.37 | [TK]D-Fender | danp, and far mnore |
| 20:54.37 | rudholm | danp: yes |
| 20:54.40 | danp | i personally use lukemftpd |
| 20:54.42 | pfn | [TK]D-Fender, indeed |
| 20:55.22 | pfn | damn, ciscos have depreciated quickly |
| 20:55.29 | pfn | should sell my broken 7960 while it can still make some money |
| 20:55.37 | flujan_ | hey [TK]D-Fender ... :) |
| 20:55.44 | flujan_ | could you help me with this error? |
| 20:55.45 | flujan_ | http://pastie.caboo.se/52608 |
| 20:55.54 | Mercestes | pfn: I'll give you $20 |
| 20:56.06 | bkruse | i see that bet and raise you 2 dollars |
| 20:56.07 | bkruse | ill give you 22 |
| 20:56.14 | bkruse | jbot: [TK]D-Fender++ |
| 20:56.16 | bkruse | for helping :] |
| 20:57.08 | ghosto | [TK]D-Fender: true...there is Nortel stuff there but i can find no articles or anything on the subject...and besides, i'm trying to integrate with an Asterisk so i thought someone here might have experience...about talking to a tech? out of the question, the local telecom here is tight on giving info |
| 20:57.09 | anonymouz666 | how to insert a value into MySQL table using ODBC with ast 1.2? |
| 20:57.34 | anonymouz666 | does I need to install the func_odbc? |
| 20:57.34 | Mercestes | jbot: mercestes-- for being an insatiable troll |
| 20:57.39 | anonymouz666 | do |
| 20:57.44 | Mercestes | anonymouz666, asterisk-addosn |
| 20:57.44 | Mercestes | anonymouz666, asterisk-addons |
| 20:57.49 | anonymouz666 | do you wanna mean app_mysql ? |
| 20:57.49 | Mercestes | ~mercestes |
| 20:57.51 | jbot | it has been said that mercestes is almost a total nub |
| 20:57.57 | pfn | ~pfn |
| 20:58.00 | bkruse | ~karma Mercestes |
| 20:58.00 | jbot | mercestes has neutral karma |
| 20:58.07 | Mercestes | :) |
| 20:58.10 | bkruse | you cannot give yourself karma.... |
| 20:58.11 | bkruse | lol |
| 20:58.13 | bkruse | gt |
| 20:58.39 | [TK]D-Fender | ~[TK]D-Fender |
| 20:58.41 | jbot | well, [tk]d-fender is rockin' the casbah !!! |
| 20:58.50 | [TK]D-Fender | ~karma [TK]D-Fender |
| 20:58.50 | jbot | [tk]d-fender has karma of 6 |
| 20:58.51 | polerin | polerin-- |
| 20:58.53 | cocomp | Hi all, quick question - just re-compiled asterisk (& zaptel * libpri) 1.2 (updated from svn) and now getting: |
| 20:58.53 | cocomp | Executing Wait("Zap/3-1", "1") in new stack |
| 20:58.53 | cocomp | Spawn extension (incoming, s, 1) exited non-zero on 'Zap/3-1' |
| 20:58.53 | cocomp | previously all working fine - any ideas? |
| 20:58.57 | [TK]D-Fender | only? *sigh* |
| 20:59.16 | [TK]D-Fender | cocomp, pastebin your dialplan. |
| 20:59.17 | [TK]D-Fender | ~pb |
| 20:59.21 | jbot | pb is, like, a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 21:00.58 | pfn | ~karma pfn |
| 21:00.58 | jbot | pfn has karma of 2 |
| 21:01.01 | pfn | :) |
| 21:01.59 | RazaMetaL | hi guys :) |
| 21:02.28 | RazaMetaL | I've a weird trouble .. |
| 21:02.37 | cocomp | http://pastebin.ca/431699 |
| 21:02.37 | RazaMetaL | I've one provider for a-z but all the calls to USA are being dropped after 12~35 seconds |
| 21:02.56 | RazaMetaL | any one here are using five9snetwork? |
| 21:04.57 | cocomp | [TK]D-Fender: dialplan as in 'show dialplan' from console? |
| 21:05.36 | bkruse | ~karma bkruse |
| 21:05.36 | jbot | bkruse has neutral karma |
| 21:05.38 | bkruse | :( |
| 21:07.46 | [TK]D-Fender | cocomp, maybe its jsut me, but I don't see and [incoming] context in there anywhere... |
| 21:08.08 | [TK]D-Fender | bkruse++ |
| 21:08.08 | cocomp | I was simplifying for in here |
| 21:08.13 | bkruse | [TK]D-Fender: yay |
| 21:08.14 | cocomp | it's sp_incoming |
| 21:08.35 | [TK]D-Fender | cocomp, simplifying is also known as "lying" and "confusing" :) |
| 21:09.00 | cocomp | :P |
| 21:09.03 | cocomp | :-P |
| 21:09.32 | [TK]D-Fender | cocomp, pastebin some real CLI output please.... |
| 21:09.37 | *** join/#asterisk CuriosWrk (i=stian@dynamic172.noc.host.net) |
| 21:09.50 | CuriosWrk | Hrm, fun. |
| 21:10.51 | cocomp | [TK]D-Fender: http://pastebin.ca/431718 |
| 21:11.35 | [TK]D-Fender | cocomp, got "immediate=yes" in your zapata.conf definition? |
| 21:12.09 | cocomp | quite possibly - will check |
| 21:13.29 | cocomp | no - immediate=no |
| 21:14.47 | [TK]D-Fender | good |
| 21:15.16 | squirrellb | i´m back |
| 21:15.19 | [TK]D-Fender | cocomp, extens your wait to 10s and see how long till it bails. then remove the wait altogether. |
| 21:15.35 | CrashHD | ERROR[14508]: channel.c:1543 queue_frame_to_spies: Translation to slin failed, dropping frame for spies with mixmonitor being used |
| 21:15.38 | CrashHD | anyone have any ideas? |
| 21:15.39 | ManxPower | did anyone want the list of all the test lines for BellSouth? |
| 21:15.45 | CrashHD | ulaw is the call type |
| 21:15.50 | CuriosWrk | manxpower: If you have it, sure |
| 21:16.14 | ManxPower | CuriosWrk: I'll upload it. |
| 21:16.22 | RazaMetaL | [TK]D-Fender, what is the use of immediate=yes ? |
| 21:16.25 | ManxPower | it will take a while |
| 21:16.53 | ManxPower | RazaMetaL: immediate=yes would be for e Bat Phone type of setup. Pick up the phone and the system dials a predefined number |
| 21:17.12 | [TK]D-Fender | RazaMetaL, causes analog FXS to dial immediately upon going off-hook, and tends to kill FXO's outright |
| 21:17.45 | [TK]D-Fender | RazaMetaL, where "predefined number" = "s" |
| 21:17.56 | ManxPower | So unless you are hot for a guy in a cape, I suggest you don't set it |
| 21:18.01 | CuriosWrk | How can I use zaptel to take a phone line off hook? |
| 21:18.04 | RazaMetaL | thnkz ... |
| 21:18.11 | CuriosWrk | e.g. akin to ATH1 on a regular modem? |
| 21:18.16 | ManxPower | CuriosWrk: your question makes no sense |
| 21:18.49 | ManxPower | Pick up a phone on the system, dial 555 and have an exten => 555,1,Dial/Zap/1/) |
| 21:19.00 | CuriosWrk | ManxPower: I have a phone line going into an FXO card |
| 21:19.09 | CuriosWrk | (regular phone line, not voip phone) |
| 21:19.15 | CuriosWrk | trying to figure out how to, well, do anything with it |
| 21:19.25 | *** join/#asterisk Corydon76-home (i=brown@pdpc/supporter/sustaining/Corydon76-home) |
| 21:19.25 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 21:19.27 | *** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca) |
| 21:19.31 | CuriosWrk | and if the questions make no sense, I'll try to ask better ones. I'm a newbie to this stuff. |
| 21:19.31 | ManxPower | CuriosWrk: you can't do anything with it. You need something to act like a phone |
| 21:20.38 | SmaxTheFrog_ | when dialing to outside, may i just user dial(PROT/exten) as if i would dial to an internal number ? |
| 21:20.42 | CuriosWrk | ManxPower: That something could be a piece of software, though |
| 21:20.47 | ManxPower | CuriosWrk: Asterisk is a PBX. If your question makes sense for a PBX then chances are it will make sense for Asterisk |
| 21:20.59 | CuriosWrk | and that software, like asterisk, would need to be able to control the phone line somehow |
| 21:21.06 | ManxPower | SmaxTheFrog_: not really |
| 21:21.18 | cocomp | [TK]D-Fender: Removing the Wait worked |
| 21:21.34 | cocomp | Is that deprecated or something? |
| 21:21.38 | ManxPower | If your FXO port is Zap/1 then you would Dial(Zap/1/${EXTEN} or something similar |
| 21:21.51 | SmaxTheFrog_ | ok thx man |
| 21:22.12 | [TK]D-Fender | cocomp, nope, and I can't really explain it. I just gave you a sane methodology to start removing doubts about the origin of the problem... |
| 21:22.24 | cocomp | hmm |
| 21:22.31 | cocomp | It's worked fine for months |
| 21:22.42 | [TK]D-Fender | cocomp, Chalk it up to 1.4 crazyness! |
| 21:22.50 | cocomp | then doesn't work anymore :-S |
| 21:23.08 | cocomp | but it's 1.2 branch :-P |
| 21:23.15 | [TK]D-Fender | cocomp, I upgraded today as well... Presence flaked out on me just the same... thats life I guess |
| 21:23.50 | CuriosWrk | manx: Well, zttool seems to at least detect the card, although it wants me to configure it |
| 21:24.04 | cocomp | On another note, I'm tempted to try 1.4 - is it likely to be straight forward upgrading from 1.2 config files? |
| 21:24.34 | ManxPower | CuriosWrk: http://www.fnords.org/~eric/Testline.TestlineReport.xls |
| 21:25.16 | e-milio | where is a good place to look for a DID delivered via sip or iax ? |
| 21:25.44 | *** join/#asterisk bawb2 (n=bawb2@ip51051.estcmp.ku.edu) |
| 21:28.03 | [TK]D-Fender | e-milio, depends where obviously. |
| 21:28.32 | cocomp | WARNING[7492]: chan_zap.c:6780 do_monitor: Read failed with 0: Resource temporarily unavailable |
| 21:28.42 | cocomp | just noticed after a config 'reload' |
| 21:28.51 | *** part/#asterisk ghosto (n=user@cmodem-237-235.tricom.net) |
| 21:28.53 | [TK]D-Fender | cocomp, depends. there are things that were deprecated in 1.2 that were removed entirely from 1.4. Other quirks to be accounted for as well in different scenarios |
| 21:29.08 | cocomp | k |
| 21:29.39 | cocomp | I wonder if that other message has anything to do with the borked Wait() call |
| 21:29.51 | CuriosWrk | thanks manx |
| 21:30.06 | e-milio | [TK]D-Fender: 305 |
| 21:30.16 | e-milio | [TK]D-Fender: 305 - FL US |
| 21:30.30 | [TK]D-Fender | e-milio, describe your needs. |
| 21:30.52 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
| 21:30.57 | e-milio | [TK]D-Fender: One DID 10 channels, inbound devlivered via IAX (pref) or SIP |
| 21:31.41 | *** join/#asterisk goozbach (n=goozbach@brooks.netradius.com) |
| 21:32.02 | goozbach | could someone explain to me why disa is needed on an asterisk system? |
| 21:32.27 | goozbach | isn't it just for system access? running commands from the shell, etc? |
| 21:33.14 | [TK]D-Fender | goozbach, who said you NEEDED it? |
| 21:33.47 | [TK]D-Fender | goozbach, And it has nothing to do with the shell |
| 21:33.49 | *** join/#asterisk JunK-Y (n=junky@modemcable140.185-70-69.mc.videotron.ca) |
| 21:34.09 | [TK]D-Fender | mmmmm BBQ salmon goodness... |
| 21:34.55 | rudholm | bbq salmon? |
| 21:35.21 | goozbach | I misunderstood what DISA is |
| 21:35.28 | [TK]D-Fender | e-milio, take a look here : http://connect.voicepulse.com/ |
| 21:35.34 | [TK]D-Fender | rudholm, INDEED |
| 21:35.52 | goozbach | it allows internal extension dialing? |
| 21:35.58 | rudholm | Jar-Jar say DISA phones ring! |
| 21:36.05 | *** join/#asterisk MACscr (n=MACscr@adsl-75-23-80-143.dsl.peoril.sbcglobal.net) |
| 21:36.31 | [TK]D-Fender | goozbach, it gives you dialtone in the context of yuor choice. Good for letting outside callers act like inside phones |
| 21:36.46 | rudholm | [TK]D-Fender: how does one bbq salmon? throw it on the grill more-or-less whole? |
| 21:37.00 | [TK]D-Fender | goozbach, Honestly the only point to it is that it gives you dialtone as a basic IVR wouldn't |
| 21:37.02 | goozbach | rudholm: look into cedar planks |
| 21:37.11 | Strom_M | rudholm: I should share my recipe for cilantro sauce for salmon :) |
| 21:37.18 | rudholm | I see |
| 21:37.21 | MACscr | i just moved * box from outside nat to inside nat. My box is completely sip. I do have DID's, but through my voip provider and they work great. I cant do any sip to sip anymore though. What changes do i need to do besides fwding ports in my firewall? |
| 21:37.24 | rudholm | so you get some cedar smoke |
| 21:37.26 | [TK]D-Fender | rudholm, Seasoned & baked. but on an actual BBQ typically you'd wrap it in tin-foil |
| 21:37.29 | goozbach | [TK]D-Fender: thanks, I was mis-understanding what it did |
| 21:37.45 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 21:38.39 | cocomp | pants |
| 21:38.48 | CuriosWrk | So I cannot, simply for testing, cause an FXO card to dial out on the phone line attached to it? |
| 21:38.55 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com) |
| 21:38.59 | cocomp | calledid has stopped being picked up with this version now as well |
| 21:39.25 | Strom_M | hey, here's a dumb question: can you hint an FXS port? |
| 21:39.35 | rudholm | [TK]D-Fender: ok, like one might bbq anything too small for the grill |
| 21:40.00 | pfn | what's the point of hinting? |
| 21:40.11 | [TK]D-Fender | rudholm, its to help keep the juices in. otherwise it'd dry out too fast and disintegrate |
| 21:40.39 | pfn | ael.txt doesn't explain the point of hint() very well |
| 21:40.59 | pfn | MACscr, externip |
| 21:41.12 | [TK]D-Fender | pfn, PRESENCE. Go read about it on the WIKI |
| 21:41.19 | goozbach | worse case senario with DISA enabled is that someone will be able to dial in, then dial out in some other context? |
| 21:41.29 | MACscr | pfn: where does that need set? |
| 21:41.34 | pfn | MACscr, sip.conf |
| 21:41.38 | MACscr | thanks |
| 21:42.14 | pfn | what's presence all about anyway? being able to determine if a user is on the line or not? |
| 21:42.33 | MACscr | pfn: can i just set a hostname since im using a dynamic ip? |
| 21:42.42 | lesouvage | I'm struggling for hours. I echo this "exten => 0592xxxxxx/064xxxxxxx,1,SET(KLANT=xxxxxx)" into a file and this "exten => 0592711296/0642507558|1|SET(KLANT=0642507558) " ends up in the file. Problem is that with coma's * registers the extension on reload but with | it doesn't. Any sggestion? |
| 21:42.55 | pfn | MACscr, I don't recall, but you'll have to restart asterisk everytime your IP changes |
| 21:43.06 | MACscr | oh, thats shitty. hmm |
| 21:43.10 | [TK]D-Fender | goozbach, You can do that with an IVR that includes contexts you'd best reconsider giving access to. Again, the only difference is DISA gives them DIALTONE while awaying input |
| 21:43.24 | *** join/#asterisk uski (n=uski@bre01-1-88-162-0-210.fbx.proxad.net) |
| 21:43.38 | CuriosWrk | Ah. There's an "originate" command. |
| 21:44.11 | MACscr | [TK]D-Fender : any advice on modifying my sip.conf for working with dynamic ip's? my asterisk server is now NATed |
| 21:44.30 | MACscr | i had to do it in order to setup QOS, grr |
| 21:44.33 | [TK]D-Fender | ~sipnat |
| 21:44.39 | jbot | sipnat is, like, for for more information about configurtion of Asterisk with SIP behind NAT, see http://voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 21:44.39 | uski | hi; do you know if, by any chance, this kind of device has a chance to work with asterisk ? http://minilien.com/?9RfN2xmOcs |
| 21:45.02 | pfn | dynamic IP is teh suck |
| 21:45.16 | pfn | I don't see why cable providers and cheapass dsl providers insist on dynamic IP |
| 21:45.28 | pfn | dhcp is fine, but why not a fixed-lease.... |
| 21:45.30 | MACscr | because they can charge more for static |
| 21:45.32 | CuriosWrk | pfn: Because ARIN got stingy with IPs. |
| 21:45.36 | MACscr | and most people dont need static |
| 21:45.48 | pfn | CuriosWrk, doing dhcp doesn't fix the shortage of IPs |
| 21:45.56 | pfn | rather, dynamic IPs doesn't fix it |
| 21:46.03 | *** join/#asterisk AMUG (n=junky@modemcable140.185-70-69.mc.videotron.ca) |
| 21:46.11 | pfn | as soon as more customers come on than you have IPs, you're fubar |
| 21:47.56 | CuriosWrk | pfn: It does when you're oversubscribed: more customers than IPs |
| 21:48.27 | DrukenLPY | pfn is saying when you have more access points than ip's... |
| 21:48.36 | *** part/#asterisk goozbach (n=goozbach@brooks.netradius.com) |
| 21:48.41 | Sweeper | yea, doesn't fix it |
| 21:48.50 | Sweeper | lots of places just do NAT |
| 21:48.54 | DrukenLPY | then you get dickheads like me... who have a block and don't REALLY need one.... |
| 21:49.07 | cocomp | hmm |
| 21:49.21 | DrukenLPY | pfn: i have a 32 block |
| 21:49.22 | DrukenLPY | hehe |
| 21:49.27 | cocomp | went back to zaptel 1.2.8 and callerid works fine |
| 21:49.28 | lesouvage | Is asterisk supposed to register a line like exten => 0592xxxxxxxx/06xxxxxxxxx|1|SET(KLANT=06xxxxxxxx) with "|" instead of " ," |
| 21:49.29 | pfn | 6mbps/768kbps dsl, reasonable enough for stuff |
| 21:49.31 | *** join/#asterisk zogulus (n=zogulus@58.98.adsl.brightview.com) |
| 21:49.32 | [TK]D-Fender | I could get a /24 on mine if I asked my ISP, but I don;'t really care.... 1 is good enough. |
| 21:49.35 | pfn | DrukenLPY, but like you said, you're a dickhead :p |
| 21:49.46 | *** join/#asterisk jcmoore (n=Curt@unaffiliated/tgrman) |
| 21:49.51 | DrukenLPY | roflmao |
| 21:50.14 | Sweeper | so, I have a block the size of the entire IPv4 namespace |
| 21:50.18 | Sweeper | AWSOME |
| 21:50.25 | uski | (and useless) |
| 21:50.29 | uski | (sorry ;)) |
| 21:50.35 | Sweeper | not really :D |
| 21:50.40 | pfn | as soon as everyone routes ipv6, then it might become useful... |
| 21:50.41 | uski | yea you can have nice IP addy |
| 21:50.42 | uski | like |
| 21:50.45 | Sweeper | lots of people have IPv6 thesedays |
| 21:50.46 | uski | xxx:dead:beef |
| 21:50.51 | [TK]D-Fender | lesouvage, * still parses ","'s to "|" internally. |
| 21:51.11 | MACscr | ah, i guess i can just do externip = home.mydomain.com, that will work fine |
| 21:51.16 | uski | or xxx:bad:c0f:fee maybe |
| 21:51.31 | uski | :D |
| 21:51.47 | uski | (anyway, does anyone knows a cheap usb FXS adapter that works with asterisk ?) |
| 21:51.53 | CuriosWrk | sweeper: I have a block equal to that squared. |
| 21:52.03 | CuriosWrk | and I'm using all of about 30 IPs in it. |
| 21:54.15 | lesouvage | D-Fender: The problem is that with | * doesn't register while that is what I get when I use the echo command. When I change the | manually it works perfectly. It's for a automatic sign in routine for a telephone service so I don't have to edit .conf files while getting new costumers. |
| 21:54.31 | MACscr | does anyone have a moment to make a test sip call to me? |
| 21:55.59 | *** join/#asterisk sahafeez (n=sahafeez@ip68-6-223-156.sd.sd.cox.net) |
| 21:56.20 | lesouvage | D-fender: the only thing not in place is the |. It's driving me kind of crazy. |
| 21:57.28 | lesouvage | Is there any shell command to change the | into a , |
| 21:58.14 | *** join/#asterisk mmartinn (n=martin@adsl-065-005-200-225.sip.gnv.bellsouth.net) |
| 21:58.15 | *** join/#asterisk MrTelephone (n=test@bas13-toronto63-1242371209.dsl.bell.ca) |
| 21:58.26 | MrTelephone | does anyone have an asterisk failover setup using ucarp? |
| 21:58.42 | e-milio | [TK]D-Fender: thanks |
| 21:59.10 | [TK]D-Fender | lesouvage, no idea |
| 22:00.03 | uski | what are my options to have one FXS port with asterisk ? |
| 22:00.25 | uski | apart from the TDM400P |
| 22:00.36 | [TK]D-Fender | uski, Best option = Linksys SPA-2102 |
| 22:01.17 | MrTelephone | does anyone want to buy some digium cards from me? |
| 22:01.37 | uski | [TK]D-Fender, ty, i guess it's the kind of box that i'll have to unlock |
| 22:01.38 | uski | ? |
| 22:01.48 | [TK]D-Fender | uski, No, its not locked |
| 22:02.11 | MrTelephone | i have a tdm400p 4 fxo card, 1 tdm2400p 8 fxo+echo cancel, and 1 tdm2400p 4 fxo 4fxs no echo ccancel cards |
| 22:02.23 | uski | [TK]D-Fender: ok |
| 22:03.48 | uski | [TK]D-Fender: i'm about to order one, are you sure it will work with asterisk ? ;) ;) |
| 22:04.19 | [TK]D-Fender | uski, Yes |
| 22:04.22 | uski | assuming i'm clever enough to install it properly |
| 22:04.23 | uski | thanks |
| 22:04.38 | [TK]D-Fender | uski, Thats often the first mistake people make... |
| 22:04.57 | uski | yea |
| 22:05.00 | *** join/#asterisk Dovid (n=Dovid@l192-117-114-1.broadband.actcom.net.il) |
| 22:05.15 | uski | and it has two fxs, it's cheap and great |
| 22:05.40 | *** join/#asterisk Stridernzl (n=neville@125-239-171-252.jetstream.xtra.co.nz) |
| 22:15.34 | uski | [TK]D-Fender: last question: will the SPA3102 (one FXO, one FXS) work with asterisk ? |
| 22:16.34 | uski | (it seems so, but i'd like to be sure) |
| 22:16.48 | Dovid | uski: i never used it but it should. |
| 22:16.56 | uski | ok, thanks |
| 22:17.12 | uski | well, i'll think of all this while sleeping. thanks a lot ! good night |
| 22:17.21 | uski | (to those who are in the same TZ) |
| 22:17.51 | Dovid | night |
| 22:17.53 | uski | ty |
| 22:20.37 | *** join/#asterisk justchris (n=jc@dsl-217-155-34-164.zen.co.uk) |
| 22:20.57 | justchris | what do ppl think is the best sip softphone for linux and windows? |
| 22:21.22 | justchris | ive tried a few, xlite is good on windows, but isnt OSS, and on linux im struggling |
| 22:21.40 | justchris | ekiga is flaky, twinkle is more than flaky |
| 22:21.49 | Strom_M | oh no! it's not oss! it's merely free as in shut the fuck up :D |
| 22:22.10 | justchris | lol |
| 22:22.19 | justchris | if i felt like that i wouldnt use it ;) |
| 22:22.34 | Strom_M | i'm just being a smartass - ive spent all day yelling at AT&T and need to let off some steam |
| 22:22.52 | justchris | so... back on topic - good softphone for linux anyone? |
| 22:22.52 | syzygyBSD | unless you want to modify it, which I can't really see since you want something that just works... |
| 22:23.16 | syzygyBSD | hah, you asking them why they store all your call records? |
| 22:23.29 | Strom_M | syzygyBSD: no, i'm ordering ISDN service |
| 22:23.32 | Strom_M | and they lost the order |
| 22:23.38 | Strom_M | like...literally, they have no record of it |
| 22:23.40 | syzygyBSD | lol, even better |
| 22:24.00 | Strom_M | hey, do any of you know if you can hint zaptel channels? |
| 22:24.01 | syzygyBSD | "I thought you kept all those records, can't you check with the government and the copy you sent them?" |
| 22:24.51 | Strom_M | this is AT&T California, not ATTIX, silly |
| 22:26.02 | syzygyBSD | IX? |
| 22:26.11 | Strom_M | interexchange |
| 22:26.55 | ManxPower | Strom_M: You have not ordered ISDN before I assume. BRI? |
| 22:27.02 | syzygyBSD | I just think it would be fun to ask the poor support person that |
| 22:27.56 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 22:27.59 | *** part/#asterisk frigidzephyr (i=frigidze@nat/digium/x-e611d799408259eb) |
| 22:28.16 | syzygyBSD | just like when I call up I am "Mr Blood", it lets them know to give me what I want with as little shit as possible.... I think... |
| 22:28.47 | ManxPower | I am SO tempted to switch my lines to BRI. Then the telco will HAVE to condition the line |
| 22:29.55 | file | Analog: Just say no. |
| 22:30.48 | ManxPower | file: ISDN BRI, of course. I don't expect to have more than 4 lines in the near future |
| 22:31.33 | file | I looked up the rate for a BRI (Microlink to my telco)... ~$100/mth for 2B+D |
| 22:31.40 | file | CAD. |
| 22:31.40 | ManxPower | I would just run the ISDN into an Ascend BRI box and break out the voice lines. 8-) |
| 22:32.06 | ManxPower | file: $100/month is about average for much of the USA |
| 22:32.49 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com) |
| 22:34.23 | [TK]D-Fender | Yay, I have just screwed GLIBC on my * server and am about to FLUSH! |
| 22:34.23 | [TK]D-Fender | linux = toast! |
| 22:34.23 | cocomp | nasty |
| 22:34.37 | [TK]D-Fender | yup.... |
| 22:35.23 | [TK]D-Fender | And since my * box is also my router, my primary PC also goes dark during the reinstall/upgrade |
| 22:35.35 | [TK]D-Fender | if wanpipe failes, i=stranded |
| 22:36.14 | blitzrage | ewww |
| 22:36.14 | cocomp | ouch |
| 22:36.53 | Qwell[] | [TK]D-Fender: That'll teach you :p |
| 22:37.32 | [TK]D-Fender | qwell, Yeah... tried a SWARET upgrade because I was missing something and nuked everything... |
| 22:38.21 | [TK]D-Fender | qwell, on the upside I'll be moving from Slack 10.2 to CentOS 4.4 and getting Zaptel support ;) |
| 22:38.39 | Qwell[] | but on the down side, it's centos |
| 22:39.04 | blitzrage | centos is good stuff! |
| 22:39.04 | cocomp | CentOS rocks |
| 22:39.04 | blitzrage | :) |
| 22:39.04 | blitzrage | Qwell[]: you probably use Debian or something.... ewww |
| 22:39.11 | Qwell[] | blitzrage: gentoo :P |
| 22:39.22 | blitzrage | even worse! |
| 22:39.41 | *** join/#asterisk fluffyfluffy (n=fluffyfl@h69-130-215-2.69-130.unk.tds.net) |
| 22:39.52 | [TK]D-Fender | blitzrage, I suggest, prefer CentOS myself now, its jsut that when its your home server you something just don't want to touch it at all :) |
| 22:40.10 | blitzrage | [TK]D-Fender: agreed! |
| 22:40.27 | *** part/#asterisk viler (n=mv@134.42.128.66.PPPoECali.dynamic.telesat.net.co) |
| 22:40.54 | *** join/#asterisk mivck (n=mv@134.42.128.66.PPPoECali.dynamic.telesat.net.co) |
| 22:42.32 | fluffyfluffy | Anyone have any hands on experience with the aastra 480i ct. I'm still trying to find cordless|wifi sip phones for home and I'm getting that feeling that they all suck one way or another. |
| 22:42.34 | Strom_M | [TK]D-Fender: can I configiure presence on the built-in line keys on this 601, or must I use a sidecar? |
| 22:42.49 | [TK]D-Fender | Strom_M, Either |
| 22:43.09 | Strom_M | ok |
| 22:43.09 | *** join/#asterisk crweb (n=tom@12.206.114.162) |
| 22:44.30 | [TK]D-Fender | Strom_M, If you only use a single reg I typically assign a 6XX series 3 linekeys @ 1 call each and leave 3 for speedial/presence |
| 22:44.55 | Strom_M | yeah, thats what im dicking around with now |
| 22:45.35 | [TK]D-Fender | Strom_M, Excellent, I hope to be back up and running within a few hours and cal follow up with you then. |
| 22:46.40 | Strom_M | ok? |
| 22:47.36 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 22:49.10 | [TK]D-Fender | can* |
| 22:49.25 | Strom_M | ah |
| 22:52.28 | *** join/#asterisk ToyMan (n=Stuart@74-32-50-149.dsl1.mdl.ny.frontiernet.net) |
| 22:53.34 | cocomp | anyone any ideas what changed with callerid detection between Zaptel 1.2.8 and current svn (1.2 branch)? |
| 22:54.04 | blitzrage | you could check the commits mailing list? |
| 22:54.19 | cocomp | hmm |
| 22:54.42 | cocomp | location? |
| 22:55.09 | *** join/#asterisk SGN (n=a@adsl-66-122-88-147.dsl.renocs.nvbell.net) |
| 22:55.36 | blitzrage | lists.digium.com |
| 22:55.57 | blitzrage | if you're going to be using SVN, reading the commit logs is a requirement |
| 22:56.03 | cocomp | thx just got there |
| 22:56.29 | cocomp | yeah it's just broke callerid |
| 22:57.52 | SGN | Hey guys, I got a problem. Two asterisk boxes, nearly identical. From a SIP Phone, the MP3Player works fine on *1.2. On *1.4 the MP3Player plays for about 2 seconds then starts to become extremly staticy and die. Any thoughts? |
| 23:01.13 | jnc | is the purpose of AEL to extend capabilities afforded by extensions.conf, or to serve as a potential replacement for extensions.conf? |
| 23:02.16 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 23:02.27 | ManxPower | SGN: I hate to tell you this, but 1.2 and 1.4 are not "nearly identical" |
| 23:02.45 | SGN | ManxPower: I ment interms of everything else _but_ asterisk. |
| 23:02.55 | blitzrage | jnc: AEL is simply another way of writing dialplan logic -- AEL is actually converted into dialplan |
| 23:03.09 | ManxPower | jnc: AEL is converted at run time into a standard extensions.conf, so I guess you could call it a "more programming-like way of building an extensions.conf" |
| 23:03.18 | jnc | oh okay |
| 23:03.20 | *** join/#asterisk Corydon76-home (i=white@pdpc/supporter/sustaining/Corydon76-home) |
| 23:03.20 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 23:03.53 | jnc | my objective I guess is to find a way to use a GUI interface to build asterisk for SoHo |
| 23:04.12 | *** part/#asterisk MrTelephone (n=test@bas13-toronto63-1242371209.dsl.bell.ca) |
| 23:04.13 | blitzrage | jnc: yet another one? |
| 23:04.19 | jnc | smaller companies with 5-10 users that would not want to depend on having an * hacker around |
| 23:04.21 | *** join/#asterisk syzygyBSD_ (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 23:04.22 | jnc | yeah |
| 23:04.25 | jnc | :P |
| 23:04.39 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:04.55 | jnc | it ties in mostly with businesses my friend and I already do work for |
| 23:05.39 | ManxPower | smaller companies of 5-10 users are screwed because you can't have a novice run a PBX and they can't afford to hire someone to manage the PBX |
| 23:05.57 | ManxPower | PBXs are complicated systems |
| 23:06.10 | blitzrage | that's where vPBXs make more sense |
| 23:07.53 | Strom_M | grmph. does vsftpd support ftps, or am I barking up the wrong tree? |
| 23:08.03 | ManxPower | blitzrage: then you just trade the requirement for a PBX admin for the requirement of an expert in networking so your connection to the vPBX vendor works 8-) |
| 23:08.22 | ManxPower | Strom_M: "ftps" is not really FTP. |
| 23:08.29 | blitzrage | ManxPower: having a working network is easier than administering a PBX |
| 23:08.47 | Dovid | i just use sftp |
| 23:08.54 | ManxPower | Open SSH server would normally include the sftp server |
| 23:09.07 | Dovid | u dont need anything on the box for it |
| 23:09.12 | Dovid | it works over ssh |
| 23:09.21 | blitzrage | ok, ball game is on... GO JAYS GO! |
| 23:09.30 | ManxPower | blitzrage: Yes, as long as you don't expect it to be as reliable as a local PBX with phone lines. |
| 23:09.42 | *** join/#asterisk syzygyBSD_ (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 23:10.02 | Strom_M | ManxPower: i'm deploying some polycom sets |
| 23:10.17 | Strom_M | and the polycom set supports "FTPS" but not "SFTP" |
| 23:11.13 | [TK]D-Fender | ok, about to begin the big upgrade/salvage. Pray for me... BB ... eventually.... |
| 23:11.22 | Strom_M | good luck, [TK]D-Fender |
| 23:12.05 | ManxPower | Strom_M: Oh! I've always used FTP |
| 23:12.51 | ManxPower | Strom_M: I'll bet that if you wrap an FTP daemon in an SSH session it will become "FTPS". |
| 23:13.51 | ManxPower | IIRC the same method used for SMTPS before SMTP servers supported it |
| 23:13.59 | Strom_M | eh, it's behind a firewall. FTP is good enough |
| 23:14.37 | Strom_M | i have better things to worry about, like "how do I provision speed dial buttons?" |
| 23:15.26 | Strom_M | oh, that was easy :) |
| 23:15.30 | Mercestes | lol |
| 23:15.53 | ManxPower | Strom_M: Speed Dial buttons are basically line appearances that have no provisioning info (userid/secret, etc) associated with them, then the polycom will take items from the polycom directoy and stuff them on the line appearances as speed dials |
| 23:16.02 | Strom_M | now....how to get the hints working |
| 23:16.11 | Strom_M | ManxPower: yeah, i just discovered that thirty seconds ago :D |
| 23:16.40 | ManxPower | Strom_M: you want to enable "buddy watch" for each of the speed dials you want to monitor the hint for |
| 23:16.56 | Strom_M | yeah, I have it enabled in the XML file |
| 23:17.15 | ManxPower | So, how much did you say you would sell me a payphone for? |
| 23:17.34 | Strom_M | i havent offered yet ;) |
| 23:18.09 | Mushroom | But I'm a fun guy! |
| 23:18.24 | Strom_M | i dont get the reference :/ |
| 23:18.50 | ManxPower | fun guy == fungi |
| 23:18.52 | Qwell[] | Strom_C: say it out loud... |
| 23:19.15 | Strom_M | oh, durh :D |
| 23:20.00 | ManxPower | I called in the trouble ticket on the line today. We'll see how it goes tomorrow when Bell shows up |
| 23:21.47 | Strom_M | anyway, I have buddywatching on... |
| 23:21.58 | ManxPower | http://www.fnords.org/~eric/polycom-config-examples/ |
| 23:26.43 | Strom_M | but the phone isnt telling asterisk "watch this hint" |
| 23:27.23 | ManxPower | you need to enable <bw> for each of the speed dials. |
| 23:27.30 | ManxPower | I can give you a sample if you want |
| 23:27.35 | Strom_M | yeah, I did |
| 23:27.41 | Strom_M | I have <bw>1</bw> |
| 23:27.50 | ManxPower | standby |
| 23:27.52 | pfn | so what is preferred for building dialplans these days? AEL or using the old exten syntax? |
| 23:28.04 | pfn | AEL sure seems nicer, at least syntactically |
| 23:30.02 | ManxPower | Strom_M: So you have feature.1.name="presence" feature.1.enabled="1" in either the mac-phone.cfg or in the phone1.cfg or sip.cfg ? |
| 23:30.08 | ManxPower | So -- Do |
| 23:30.11 | *** part/#asterisk crweb (n=tom@12.206.114.162) |
| 23:30.30 | Strom_M | ah...heh, now I do ;) |
| 23:30.52 | ManxPower | 8-) |
| 23:31.11 | *** join/#asterisk jebba (n=jebba@220-179-89-200.fibertel.com.ar) |
| 23:31.20 | ManxPower | I have it in the mac-phone.cfg since most of the phones won't use it. I don't recall where it would be put globally |
| 23:31.35 | Strom_M | it was in sip.cfg |
| 23:31.50 | *** join/#asterisk jebba (n=jebba@220-179-89-200.fibertel.com.ar) |
| 23:35.14 | *** join/#asterisk techie (n=gus@voip.routedsystems.com) |
| 23:35.26 | *** join/#asterisk CuriosCat (i=stian@truck-71.dynamic.bigrig.org) |
| 23:37.22 | Strom_M | hahahahaahaha, that is kickass |
| 23:37.32 | Strom_M | i have hints working on FXS ports |
| 23:37.52 | cocomp | hints? what's that all about? |
| 23:38.10 | CuriosCat | I have an FXO card whose status is "OK" |
| 23:38.14 | CuriosCat | I suppose that's a start :P |
| 23:39.57 | *** join/#asterisk HeyItsMe (n=HeyItsMe@69.18.124.82) |
| 23:40.08 | HeyItsMe | Hi all |
| 23:41.21 | HeyItsMe | I am trying to get my box to talk through this te205p to an Adtran TA905. Does anybody know about the te205p? |
| 23:41.34 | Strom_M | yep |
| 23:41.46 | ManxPower | HeyItsMe: for one thing you need a crossover T-1 cable |
| 23:42.03 | ManxPower | which is NOT the same as a crossover ethernet cable. The wiki has wiring diagrams |
| 23:42.29 | HeyItsMe | OOOOOOoooh, I can't just plug it into the adtran 905 |
| 23:42.43 | Strom_M | nope |
| 23:42.47 | ManxPower | HeyItsMe: sure you can. But it won't work. |
| 23:42.58 | Strom_M | damn you, how do you get the polycom to reload the directory XML file? |
| 23:43.01 | Strom_M | a reboot doesnt force it |
| 23:43.17 | HeyItsMe | ty al |
| 23:43.21 | HeyItsMe | all |
| 23:43.22 | ManxPower | Strom_M: try touching the file the power cycling the phone. Other than that I have no idea |
| 23:43.29 | mcab | Strom_M: reboot should pick it up |
| 23:43.37 | Strom_M | tried it |
| 23:43.52 | Strom_M | HeyItsMe: and be sure to use a T1 crossover, not an ethernet crossover |
| 23:44.09 | mcab | Strom_M: are you changing the 00000000000-directory.xml file or the <mac>-directory.xml file? |
| 23:44.16 | Strom_M | 0000000000 |
| 23:44.41 | HeyItsMe | Like this? http://www.voip-info.org/wiki/view/T1+Crossover+Cable |
| 23:44.50 | cocomp | any online docs for "hints"? |
| 23:44.51 | mcab | Strom_M: it'll only pick that up if it doesn't have an existing directory file in it's flash already |
| 23:45.04 | mcab | Strom_M: otherwise it will only check for the <mac>-directory.xml file |
| 23:45.26 | Strom_M | catsex. |
| 23:45.53 | Strom_M | can't it pick up the damn file on reboot? |
| 23:45.56 | mcab | Cat6? I just use 5e myself... |
| 23:46.06 | Strom_M | no no, catSEX, not cat six |
| 23:46.32 | rudholm | sexe du chat |
| 23:46.43 | Strom_M | sexo del gato |
| 23:46.50 | Strom_M | kazegeschlecht |
| 23:46.56 | *** join/#asterisk infernix (i=nix@unaffiliated/infernix) |
| 23:47.05 | mcab | Strom_M: not the 000000000000 (unless you've just toasted the filesystem) 'cp 000000000000-directory.xml 0004f2xxxxxx-directory.xml' and see if that works |
| 23:47.23 | Strom_M | yeah, but im trying to replicate it across four sets |
| 23:47.26 | Strom_M | ill just symlink |
| 23:47.33 | mcab | catsex? sounds noisy... and furry... |
| 23:48.50 | jnc | must translate to english something like "son of a bitch!" |
| 23:49.14 | Strom_M | rudholm: how would you say "catsex" in German, anyway |
| 23:50.52 | rudholm | basically what you wrote |
| 23:51.13 | rudholm | but there's a t in katze |
| 23:51.38 | Strom_M | typo |
| 23:52.04 | Strom_M | woot, this presence thing rocks. i'm kinda bummed the client hasn't ordered the sidecars yet :) |
| 23:54.15 | blitzrage | Strom_M: hrmmm... you got it working eh? |
| 23:54.23 | blitzrage | I never got mine working on this polycom... |
| 23:54.30 | blitzrage | which reminds me I gotta figure out how to get this MWI stuff working again... |
| 23:55.44 | Strom_M | yeah, i got it working |
| 23:55.48 | Strom_M | with FXS ports, no less |
| 23:59.28 | *** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
