| 00:00.35 | Luke-Jr | the only good sex is that capable of both bonding and procreation |
| 00:00.50 | tsume | Luke-Jr: I guess you've never tried it :) |
| 00:00.51 | ManxPower | The basic problem is consent. Only humans can give consent. |
| 00:01.24 | Luke-Jr | ManxPower: consent is irrelevant to sex, except for the case where both married persons consent to not do it |
| 00:01.44 | Strom_C | ManxPower, don't try arguing with him - he's bonkers |
| 00:01.44 | ManxPower | Luke-Jr, Um, consent is the most important thing. |
| 00:01.55 | justinu | he sounds pretty fanatical |
| 00:01.56 | tsume | Luke-Jr: its great when a strap on enters anally, it hits the prostate. making it more pleasurable |
| 00:02.01 | Luke-Jr | ManxPower: if you're not married, you don't do it, simply |
| 00:02.05 | ManxPower | Anything goes as long as there is 1) consent and 2) no perm damage. |
| 00:02.21 | Luke-Jr | if you're married, you're obligated to do it at least once in a while properly |
| 00:02.38 | Luke-Jr | ManxPower: only under Satan |
| 00:02.41 | ManxPower | Strom_C, who's argueing. I'm right, he's wrong. Pretty simple really. |
| 00:02.42 | tsume | heh |
| 00:02.47 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 00:02.57 | tsume | random sex with friends are perfectly fine |
| 00:03.07 | tsume | sex is a social tool :) |
| 00:03.10 | justinu | if you want to burn in hell!! :P |
| 00:03.12 | tsume | and a reproduction tool |
| 00:03.19 | Luke-Jr | justinu: couldn't have said it better |
| 00:03.22 | justinu | lol |
| 00:03.32 | Strom_C | sex is like perl - there's more than one way to do it |
| 00:03.43 | Strom_C | or if you're the perl camel - "there's more than one way to do me" |
| 00:05.32 | justinu | the real question is: why isn't there a web browser that doesn't completely suck? |
| 00:05.38 | Luke-Jr | I must admit, in today's messed up society, most heterosexuals are just as bad as the homosexuals |
| 00:05.43 | Luke-Jr | justinu: there is |
| 00:06.18 | Luke-Jr | it's called Konqueror |
| 00:09.01 | glm2k | lol |
| 00:09.28 | *** join/#asterisk omal (n=omal@cpe-24-164-111-184.neo.res.rr.com) |
| 00:09.30 | tsume | sex is perfectly fine :) |
| 00:09.58 | tsume | if I had a virus, I'd wipe out the human race and create a race of asexual beans :) |
| 00:10.02 | Luke-Jr | sex is fine and good, provided the people are married and have the right intentions |
| 00:10.08 | tsume | no :) |
| 00:10.15 | tsume | they don't need to be married ;) |
| 00:10.25 | Strom_C | i think Luke-Jr just feels guilty about choking the chicken |
| 00:10.26 | Luke-Jr | yes they do |
| 00:10.35 | tsume | no they dont |
| 00:10.42 | Luke-Jr | you're wrong. bye. |
| 00:10.42 | ManxPower | Strom_C, he sounds very catholic, eh? |
| 00:10.52 | tsume | Luke-Jr: here, I'll screw my pooch while we talk :) |
| 00:10.57 | Strom_C | ManxPower, he is very catholic |
| 00:11.06 | justinu | heh |
| 00:11.10 | tsume | lube up! oh yeah!! |
| 00:11.14 | *** join/#asterisk jhnjwng (n=wj1918@pool-70-21-174-24.nwrk.east.verizon.net) |
| 00:11.30 | glm2k | this chan gives new meaning to "*" |
| 00:11.34 | Strom_C | yes |
| 00:11.37 | Strom_C | yes it does |
| 00:11.42 | tsume | Luke-Jr: beast sex is even more fun :) |
| 00:11.48 | Luke-Jr | tsume: fun can be evil |
| 00:11.58 | Luke-Jr | fun does not make something good |
| 00:11.59 | tsume | dogs have a higher temperature than humans ;) much more warmer |
| 00:12.12 | tsume | Luke-Jr: sure it does, it means getting some relax time. destressing. |
| 00:12.15 | ManxPower | tsume, you are just trying to gross everyone out. |
| 00:12.42 | tsume | ManxPower: I guess you don't ever go to http://www.beastforum.com |
| 00:12.44 | tsume | :D |
| 00:12.49 | file[laptop] | random useless knowledge: a strangely high number of Asterisk developers are gay or bi, coincidence? maybe |
| 00:13.10 | *** join/#asterisk zotz (n=zotz@24.231.32.85) |
| 00:13.10 | glm2k | well, it's knowledge, not a fact. |
| 00:13.26 | ManxPower | file, very smart people frequently are. |
| 00:14.04 | glm2k | is that because smart people supposedly make informed decisions? |
| 00:14.14 | file[laptop] | does that make Asterisk... gay?!? |
| 00:14.20 | Strom_C | Luke-Jr, you should delete asterisk. You've got code written by SINNERS on your machine!!!! |
| 00:14.22 | justinu | yes |
| 00:14.22 | tsume | oh come on. |
| 00:14.35 | tsume | does it matter if they're gay or not, or bi? |
| 00:14.36 | glm2k | Asterisk is everything..hence "*" |
| 00:14.41 | justinu | not to me |
| 00:14.43 | tsume | it just means relationship, not sex. |
| 00:14.44 | Strom_C | asterisk is pansexual |
| 00:14.59 | Strom_C | why, just last week I caught asterisk and apache doing it in /usr/lib |
| 00:15.01 | tsume | sex is always social |
| 00:15.08 | glm2k | Strom_C: lol |
| 00:15.08 | file[laptop] | Strom_C: ooh sounds hot |
| 00:15.19 | Heimidal | hmm... good hold music |
| 00:15.34 | Strom_C | file, yeah, I had fun watching |
| 00:16.15 | tsume | besides, the gay people I know don't act gay. They act normal, but I don't think I'd participate in their extracurriculat activities. |
| 00:16.28 | Heimidal | lol |
| 00:16.34 | ManxPower | come to think of it a major portion of Asterisk is written by people that I know are gay. |
| 00:16.36 | justinu | why not? be social |
| 00:16.46 | tsume | justinu: Its just not my thing :) |
| 00:16.55 | file[laptop] | take a ride on the wild side |
| 00:16.55 | tsume | I can be friends, that is all. |
| 00:17.01 | Strom_C | "tea and cocksucking this afternoon? why, it sounds splendid" |
| 00:17.04 | justinu | hah |
| 00:17.10 | ManxPower | ROFL! |
| 00:17.14 | tsume | Strom_C: is that how the brits say it? |
| 00:17.19 | Strom_C | I have no idea |
| 00:17.20 | glm2k | aye |
| 00:17.35 | Strom_C | I was imagining a stereotypical 1950s suburban American housewife |
| 00:17.36 | Heimidal | lmao |
| 00:17.43 | glm2k | lmao |
| 00:18.00 | ManxPower | .msg Strom_C We call it "crumpets" not "cocksucking" |
| 00:18.03 | ManxPower | oops! |
| 00:18.06 | Strom_C | oh! |
| 00:18.08 | Strom_C | well then |
| 00:18.34 | ManxPower | (yes, really) |
| 00:18.37 | Heimidal | uh |
| 00:18.39 | justinu | is crumpets the current codeword? |
| 00:18.39 | Heimidal | those exist? |
| 00:18.48 | Strom_C | I live near Silver Lake; does that count? |
| 00:18.51 | ManxPower | Heimidal, more than you might think. |
| 00:18.57 | ManxPower | Strom_C, not really. |
| 00:19.01 | Strom_C | damn. |
| 00:19.05 | Heimidal | man, and I thought I had this "gay lifestyle" thing down pretty well. |
| 00:19.18 | justinu | the underground social network gay folks have is impressive |
| 00:19.29 | Heimidal | justinu: you have no idea ;) |
| 00:19.38 | justinu | i'm sure I don't |
| 00:19.55 | ManxPower | Did I mention it's also clothing optional? |
| 00:19.59 | Qwell | ... |
| 00:20.01 | Strom_C | ManxPower, ooh |
| 00:20.05 | Heimidal | ManxPower: sounds exciting |
| 00:20.14 | Heimidal | justinu: heard of Connexion? |
| 00:20.15 | Qwell | I'll...uhh...be back later |
| 00:20.15 | justinu | so that's a private resort, manx? |
| 00:20.24 | Qwell | file[laptop]: look what you've done :P |
| 00:20.24 | ManxPower | Heimidal, *shrug* I had to do something after Katrina |
| 00:20.34 | Heimidal | ManxPower: ah |
| 00:20.44 | ManxPower | justinu, "resort" would give the impression that it's more upscale than it is. |
| 00:20.49 | Strom_C | hey, it's better than that preachy catholic whackjob we had in here earlier :) |
| 00:20.53 | justinu | heimdal: the inflight internet thing? |
| 00:21.02 | justinu | ManxPower: so it's "rustic" |
| 00:21.03 | Heimidal | justinu: no, the gay social networking site :P |
| 00:21.06 | *** join/#asterisk Jon335 (n=jon335@ottawa-hs-209-217-83-12.d-ip.magma.ca) |
| 00:21.08 | Heimidal | it's *huge* |
| 00:21.20 | Heimidal | and it's barely known by anyone outside the "gay culture" |
| 00:21.22 | ManxPower | justinu, First thing I did when I got here was install wifi and link it to the directv internet. |
| 00:21.31 | justinu | bringing in the evils of technology... shame |
| 00:21.33 | justinu | ;) |
| 00:21.44 | ManxPower | justinu, I can't live here if I can't telecommute. |
| 00:21.51 | justinu | that's cool |
| 00:21.58 | Heimidal | ManxPower: what state? |
| 00:22.08 | ManxPower | Granted telecommuting via DirecTV Internet is much like driving a Yougo to work... |
| 00:22.10 | tsume | religion could be declared evil as well |
| 00:22.12 | ManxPower | Heimidal, alabama |
| 00:22.16 | ManxPower | there are three such places in AL |
| 00:22.18 | tsume | it has caused much grief |
| 00:22.21 | Heimidal | ManxPower: holy crap |
| 00:22.22 | justinu | one of my gay friends lives in NC |
| 00:22.32 | tsume | if all humans just believed in living and pushed with science |
| 00:22.49 | Jon335 | How is the best way to stress test asterisk? |
| 00:22.54 | tsume | we have sex with the bears ;) |
| 00:23.14 | ManxPower | Jon335, announcing your SIP url on asterisk-users and offer free calling for a week |
| 00:23.25 | Strom_C | hahaha |
| 00:23.25 | Heimidal | lol |
| 00:23.30 | Jon335 | lol |
| 00:23.35 | ManxPower | other people have done it |
| 00:23.50 | ManxPower | might have been asterisk-biz |
| 00:23.59 | *** part/#asterisk dja_ (n=alden@cpe-24-95-62-160.columbus.res.rr.com) |
| 00:24.32 | ManxPower | Heimidal, I'm sketching out floor plans for my soon-to-be cabin |
| 00:24.51 | Jon335 | is there a program that I can use? |
| 00:24.52 | Heimidal | how do you setup a queue to announce the name of the queue to members when called? |
| 00:25.09 | *** join/#asterisk cyberatom (n=cyberato@2001:5c0:8fff:fffe:0:0:0:4a93) |
| 00:26.13 | ManxPower | Heimidal, no idea. I would just change the callerid of the call before going int othe queue |
| 00:26.37 | Heimidal | hmm |
| 00:27.09 | Heimidal | I want to retain the callerid info, and I'll need to send the callerid info to cell phones :\ |
| 00:30.07 | *** join/#asterisk jeebusroxors (n=jeebusro@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
| 00:30.42 | ManxPower | "# Please leave entries above this comment where they are. Same for those below" |
| 00:31.18 | Strom_C | HAHAHAHA |
| 00:31.19 | Heimidal | lol |
| 00:32.04 | Qwell | Don't let them change the comment either |
| 00:32.34 | Heimidal | is there any way to define an alternate number for an extension (via a variable or somesuch) that I can then reference when dialing? |
| 00:32.40 | *** join/#asterisk ahattar (n=user@ool-43551487.dyn.optonline.net) |
| 00:32.48 | ahattar | hi all |
| 00:34.33 | *** join/#asterisk Jon335 (n=jon335@ottawa-hs-209-217-83-12.d-ip.magma.ca) |
| 00:34.42 | *** join/#asterisk stefan (i=stefan@home.stefan.id.au) |
| 00:34.55 | ahattar | i have a voip phone to my house with DID number from my cable company (cablevision) can I connect it to my asterisk box? |
| 00:34.57 | ManxPower | Heimidal, you have two exten => lines that dial the same device. |
| 00:35.11 | ManxPower | ahattar, does it run SI{? |
| 00:35.17 | Heimidal | ManxPower: right, but I want to store the extension owner's cell phone number |
| 00:35.18 | ManxPower | SIP. that is. |
| 00:35.39 | ahattar | no my cable modem has rj11 analog |
| 00:35.43 | ManxPower | ARGH! I hate C++ |
| 00:35.58 | ManxPower | ahattar, then the answer is "yes, but it won't work well" |
| 00:35.59 | tehdely | don't we all |
| 00:36.16 | ahattar | pluse rj45 for my internet connection only |
| 00:37.47 | Strom_C | going for pizza...back in a few |
| 00:39.10 | ManxPower | ahattar, those devices do not normaly signal when the far end hangs up in a way Asterisk can understand. Also you'll need a TDM400P w/FXO port on it. |
| 00:40.24 | *** join/#asterisk riksta (n=rick@213.121.151.210) |
| 00:40.28 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 00:40.31 | *** join/#asterisk dja_ (n=alden@cpe-24-95-62-160.columbus.res.rr.com) |
| 00:40.56 | ahattar | manxpower, i do not want to connect zap card to convert it back to ip, may be if i will call my cable company they will help me Manxpower: wut do u think? |
| 00:40.58 | *** join/#asterisk loko (n=rbrown@c-67-172-54-135.hsd1.pa.comcast.net) |
| 00:41.23 | ManxPower | ahattar, you can easily see what protocols Asterisk supports. |
| 00:41.46 | ManxPower | ahattar, I have never heard of a cable company allowing users to connect their own devices, they activly try to make it impossible and they are good at stuff like that |
| 00:42.06 | riksta | hi there has anyone got a moment to help me with a TDM30B, i have configured in zaptel.conf fxoks=1-3 and fxsks=4-6 but when I run ztcfg -vv i get the following output http://pastebin.ca/47095 |
| 00:42.16 | riksta | any help appreciated, I'm not sure that I configured the zatel.conf correctly |
| 00:42.21 | *** join/#asterisk loko (n=rbrown@c-67-172-54-135.hsd1.pa.comcast.net) |
| 00:42.42 | ahattar | MANX:wut if i will buy did number from other provider? |
| 00:43.16 | Qwell | ahattar: That would work fine |
| 00:43.25 | ManxPower | ahattar, well you know the protocols Asterisk supports.... |
| 00:43.52 | ahattar | manx: have a look at that http://en.wikipedia.org/wiki/Direct_Inward_Dialing |
| 00:43.58 | ManxPower | riksta, why are you configuring your TDM card with three modules as 6 ports? |
| 00:44.03 | *** join/#asterisk Kumbang (n=kumbang@167.205.24.5) |
| 00:44.29 | riksta | ManxPower: ahh ManxPower I don't know i havent used these cards before, so i have 3fxo and 3fxs ..i only need to configure the fxo ones? |
| 00:44.36 | dja_ | Hi. I'm having trouble with my provider -- I'm not able to pass dtmf through them (to deal with remote voicemail). I have a 2nd provider that's setup exactly the same, and it works fine through them. Suggestions? |
| 00:44.59 | ManxPower | then you have one TDM30B and one TDM03B |
| 00:45.09 | VeNoMouS_ | quit |
| 00:45.14 | riksta | yeah my bad i looked as this wrong, we have just 3fxs |
| 00:45.18 | riksta | err fxo |
| 00:45.21 | riksta | sorry, and thanks |
| 00:45.39 | ManxPower | ~fxsfxo |
| 00:45.41 | jbot | rumour has it, fxsfxo is An FXO port expects to receive dialtone and receive ring voltage. An FXS port expects to provide dialtone and provide ring voltage. |
| 00:45.41 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 00:45.53 | ManxPower | now what do your ports expect? |
| 00:46.02 | VeNoMouS_ | dja_: dtmfmode = rfc2833 |
| 00:46.21 | nettie | Hi guys, anyone know why moh seems to work reading the console verbose messages but no audio is actually played? mpg123 clean;y starts with asterisk and and mp3 file has a bitrate of 128kbit. Any idea please? thanx in advance |
| 00:46.24 | VeNoMouS_ | or hell do inband |
| 00:46.39 | dja_ | VeNoMous_: I'm pretty sure I tried that, but I'll try again. :-) |
| 00:46.49 | VeNoMouS_ | u could try dtmfmode=inband |
| 00:48.10 | ManxPower | inband DTMF will only work if the codec is ulaw or alaw |
| 00:48.11 | dja_ | rfc2833 didn't work, trying inband now |
| 00:48.22 | riksta | ManxPower: so if i have 3x FXS modules...i set the zaptel.conf to fxoks=1-3 right? because it says FXS uses FXO signalling ? |
| 00:48.26 | ManxPower | nettie, what version does mpg123 -v say? |
| 00:48.35 | ManxPower | riksta, correct |
| 00:48.41 | riksta | thanks so much |
| 00:48.54 | dja_ | inband worked (I'm using ulaw to this provider) -- thanks alot everyone (especially VeNoMous_ :) |
| 00:50.09 | VeNoMouS_ | np |
| 00:50.14 | loko | Is there an RPM for asterisk? |
| 00:50.23 | Qwell | loko: There is, but it isn't recommended |
| 00:50.37 | loko | yea normally i compile but I cant get zaptel to compile |
| 00:50.42 | loko | http://rafb.net/paste/results/dXcpfx21.html |
| 00:50.53 | Qwell | loko: What distro? |
| 00:50.59 | cert | what os does everyone use? |
| 00:51.00 | loko | CentOS 4.3 |
| 00:51.03 | Qwell | ~centosbug |
| 00:51.05 | jbot | centosbug is probably a problem with the latest Centos kernel (4.2 and 4.3). To fix it, edit the file /usr/src/kernels/2.6.9-34.EL-i686/include/linux/spinlock.h and change 'rw_lock_t' on line 407 to 'rwlock_t'. This is part of the 'kernel-devel' package. |
| 00:51.09 | Qwell | loko: ^ |
| 00:51.14 | loko | ah |
| 00:51.15 | loko | thank you |
| 00:51.55 | riksta | ManxPower: in zapata.conf when i have FXS modules, what do I use for signalling= is it fxs_ks or fxo_ks ? |
| 00:52.10 | riksta | fxo again? |
| 00:52.18 | Strom_C | fxo |
| 00:52.23 | riksta | tnx |
| 00:53.40 | SplasPood | are there any free softphones that support the URL parameter to Dial() ? |
| 00:55.39 | loko | Is there a bug with SELinux / restorecon as well? |
| 00:55.49 | loko | (I am trying to compile zaptel when I get all these errors) |
| 00:58.11 | justinu | ManxPower: you used to live in New Orleans? |
| 00:59.11 | ManxPower | justinu, for 10 years, then lived in Pensacola FL for 2 years and lived in Waveland MS for about 2 years until Katrina destroyed it. |
| 00:59.42 | justinu | ah, so both your stomping grounds got hammered |
| 01:00.11 | justinu | we had a strong quake here in 94, that was scary... but no significant natural disasters |
| 01:00.23 | ManxPower | justinu, yup. |
| 01:00.33 | ManxPower | now I live on the top of a mountian |
| 01:00.41 | ManxPower | Well, the locals call it a mountian, it's more of a mesa |
| 01:00.45 | justinu | is that gonna keep you safe? |
| 01:00.57 | Strom_C | justinu, bah, northridge was wimpy |
| 01:01.12 | justinu | fucked up our house |
| 01:01.48 | ManxPower | justinu, no idea. 8-) |
| 01:03.05 | justinu | well, good luck with that |
| 01:03.10 | justinu | mesapower! |
| 01:03.27 | russellb | that was bad ... |
| 01:03.50 | MikeJ[Laptop] | hmmm |
| 01:04.46 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 01:05.26 | *** join/#asterisk rene- (n=rene@dsl-201-128-115-34.prod-infinitum.com.mx) |
| 01:07.35 | *** join/#asterisk Darwin_35 (n=Darwin@c-24-9-75-234.hsd1.co.comcast.net) |
| 01:07.47 | *** join/#asterisk simoncion (n=simoncio@68.62.196.15) |
| 01:08.18 | rene- | hello, i have an asterisk box acting as a sip client to an ITSP, i have several lines with them and therefore several register lines to them, i have one [ITSP] section for the outgoing calls but since i have 30 lines that might not be enough if they dont allow sending more than one call tru one sip channel, i could of course use realtime configuration for each of the [lines] and get that working, the [sections] are of type = peer, |
| 01:09.51 | MikeJ[Laptop] | rene-, was there a question there, or were you just sharing/ |
| 01:09.53 | MikeJ[Laptop] | ? |
| 01:10.23 | Strom_C | MikeJ[Laptop], I think rene- is stuck in exposition hell |
| 01:10.44 | MikeJ[Laptop] | I'm stuck trying to load solaris 10 in a virtual machine |
| 01:10.59 | rene- | sorry, :-) my question: how do i get rid of register lines in sip.conf and put them in realtime config |
| 01:11.00 | MikeJ[Laptop] | Iam going to just have to go load it up on the real boxes |
| 01:11.11 | MikeJ[Laptop] | but I havn't wanted to walk downstairs |
| 01:11.16 | *** part/#asterisk dja_ (n=alden@cpe-24-95-62-160.columbus.res.rr.com) |
| 01:11.18 | rene- | what are you using as your host? vmware? |
| 01:11.21 | *** join/#asterisk Eggplant (i=No@6.193.217.216.cascadeaccess.com) |
| 01:11.22 | Heimidal | is there any way to define a variable for each of my devices in sip.conf and access it in an extension? |
| 01:11.27 | MikeJ[Laptop] | virtualpc right now.. |
| 01:11.29 | ManxPower | rene-, you could also leave off the username= in the [ITSP] and Dial(SIP/username@itsp/1234) |
| 01:12.29 | MikeJ[Laptop] | Heimidal, there is stuff in the sample sip.conf about setting vars like that |
| 01:12.32 | rene- | ManxPower: is there something analog to the Zap group for a group of SIP trunks? or do i need to do the load by myself |
| 01:12.55 | MikeJ[Laptop] | rene-, just do a group the fails over... |
| 01:12.59 | Heimidal | oh, I see |
| 01:12.59 | MikeJ[Laptop] | one at a time |
| 01:13.00 | MikeJ[Laptop] | or |
| 01:13.05 | rene- | Heimidal: Dial(SIP/${EXTEN}) |
| 01:13.07 | MikeJ[Laptop] | have a global var |
| 01:13.18 | rene- | if exten equals sip username |
| 01:13.26 | rene- | sip [name] |
| 01:13.26 | MikeJ[Laptop] | or several |
| 01:13.36 | Heimidal | I want to store a device user's cell number and call the number if the phone isn't picked up |
| 01:13.55 | rene- | well your best bet would be DBPut |
| 01:14.01 | rene- | since you want that to persist |
| 01:14.04 | rene- | Heimidal |
| 01:14.19 | Heimidal | well, I'll put it directly in the sip.conf file (there aren't many) |
| 01:14.19 | rene- | asterisk has a database for that sort of stuff |
| 01:14.30 | rene- | it is really easy to use |
| 01:14.33 | Heimidal | I just need to know how to pull it out of the device's info |
| 01:14.54 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 01:14.56 | Heimidal | any links to info on it? |
| 01:15.06 | rene- | mmm you would need to define that as globals |
| 01:15.35 | Heimidal | hmm.. maybe I could just use the database |
| 01:16.44 | rene- | Heimidal : you could have an extension for the user to define its forwarding number, ask the user to dial the number they want to be reached at, store that number under a key named after the user caller id |
| 01:16.54 | rene- | the database is super easy to use |
| 01:17.28 | Heimidal | sounds like a plan |
| 01:17.35 | MikeJ[Laptop] | Heimidal, the vars you set from sip.conf are only for calls FROM that |
| 01:17.47 | MikeJ[Laptop] | you could use astdb= in sip.conf |
| 01:17.53 | MikeJ[Laptop] | off the device name |
| 01:18.04 | rene- | thats an even better solution |
| 01:18.06 | MikeJ[Laptop] | and then do your magic in dialplan to look it out |
| 01:18.26 | MikeJ[Laptop] | that keeps your config in once place, which I like |
| 01:18.40 | Heimidal | astdb? |
| 01:18.41 | rene- | Mike did you meant to do priority based dialing such as priority 1 dial trunk 1 priority 101 dial trunk2 and such? |
| 01:18.51 | MikeJ[Laptop] | ummm |
| 01:18.54 | MikeJ[Laptop] | depends |
| 01:18.56 | rene- | in my case |
| 01:19.00 | rene- | the ITSP |
| 01:19.29 | MikeJ[Laptop] | I was talking about for Heimidal, if you don't know what astdb is... you have some reading to do before you try this out. |
| 01:19.33 | rene- | or can i actually use a group=X inside my [SIP] sections and then do like one does in Zaptel (dial/zap/g1) |
| 01:19.40 | Heimidal | I meant the syntax |
| 01:19.45 | rene- | Heimidal |
| 01:19.48 | Heimidal | astdb is the Asterisk DB, right? |
| 01:19.50 | rene- | show applications |
| 01:20.08 | rene- | show applications dbput and show applications dbget in the cli |
| 01:20.10 | rene- | yes |
| 01:20.42 | Heimidal | alright, so in sip.conf, I would use astdb= lines under [general] |
| 01:20.43 | Heimidal | ? |
| 01:21.27 | rene- | astdb is a dictionary, and inside the dialplan you would use dbget(keyname) in order to get the value out of it, |
| 01:21.53 | rene- | i have never used astdb inside sip.conf but if its possible then sure its cool |
| 01:22.00 | rene- | its even easier on your users |
| 01:22.36 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.236) |
| 01:22.36 | MikeJ[Laptop] | Heimidal, take a look at sip.conf like I suggested... |
| 01:22.43 | MikeJ[Laptop] | then read up some on what astdb is.. |
| 01:23.19 | rene- | MikeJ: if i define the itsp as both a peer and a user do i get to get rid of the register line? |
| 01:23.29 | MikeJ[Laptop] | nope |
| 01:23.38 | Strom_C | rene-, peer + user == friend |
| 01:23.51 | MikeJ[Laptop] | you have a register line there if you want to receive calls from them, so they know how to find you |
| 01:24.03 | MikeJ[Laptop] | nothing to do with sending calls, or beig a peer, or a user |
| 01:24.09 | MikeJ[Laptop] | different thing completely |
| 01:25.55 | rene- | well is not such a big issue, but it would be nice to use mysql for that |
| 01:26.39 | rene- | i was said that in some cases if you were on a fixed ip and used something like SER then you would not need to register |
| 01:26.46 | rene- | but SER is not as fun |
| 01:27.05 | Strom_C | rene-, why do you have thirty individual SIP lines with the ITSP? Surely you can just have one account for outbound and then DIDs on the same account for inbound |
| 01:27.41 | rene- | Strom_C: i guess my itsp has a platform built mostly for use with sipura like appliances |
| 01:27.55 | *** join/#asterisk tuxinator_linux (n=tuxinato@netblock-68-183-112-100.dslextreme.com) |
| 01:28.05 | Strom_C | rene-, and why are you with them again? |
| 01:28.11 | rene- | or they are just not very technically cutting edge |
| 01:28.28 | rene- | well they are a legal source of DIDs and offer good rates |
| 01:29.01 | Strom_C | there are illegal DID providers? o_O |
| 01:29.17 | Qwell | Strom_C: shh, not here |
| 01:29.23 | rene- | gray market |
| 01:29.51 | Strom_M | Qwell, ? |
| 01:30.17 | Qwell | nothing |
| 01:30.41 | tuxinator_linux | who's a good source? |
| 01:30.54 | MikeJ[Laptop] | Ma Bell |
| 01:31.05 | Strom_M | Our Mother of the Bell |
| 01:31.19 | MikeJ[Laptop] | I hear they can give you DID's |
| 01:31.31 | MikeJ[Laptop] | and they have multiple delivery mechinisms |
| 01:31.48 | MikeJ[Laptop] | POTS, centrex, PRI.. almost anything you could think of |
| 01:32.07 | rene- | the project is quite challenging, reliability will be bad, because even if Protel (Mexico) delivers good quailty voice over internet, we are using ADSL 1300/512 links to the internet, and the sales monkey told us that we were going to be able to run 30 voice calls over such a thin pipe |
| 01:32.32 | MikeJ[Laptop] | heh |
| 01:32.36 | MikeJ[Laptop] | good luck with that |
| 01:32.36 | tuxinator_linux | 30, ha, funny |
| 01:32.41 | MikeJ[Laptop] | voice only? |
| 01:32.47 | rene- | im taking an asterisk box to protel labs to see if it is possible to use iax trunking to our advantage |
| 01:32.50 | rene- | mostly |
| 01:32.53 | rene- | ssh |
| 01:33.02 | MikeJ[Laptop] | g729 maybe |
| 01:33.06 | MikeJ[Laptop] | MAYBE |
| 01:33.07 | Strom_M | 30 calls, easy...using lpc10 |
| 01:33.10 | rene- | we are on g729 |
| 01:33.28 | Qwell | 200ms g729? |
| 01:33.28 | MikeJ[Laptop] | if you are actually getting that throughput |
| 01:33.29 | rene- | we will ask them to let us sit an * and do iax trunking to our site |
| 01:33.31 | Jon335 | is there a program to stress test Asterisk? |
| 01:33.41 | rene- | Jon335: sipp |
| 01:33.41 | Qwell | Jon335: asterisk |
| 01:33.48 | MikeJ[Laptop] | heh |
| 01:33.54 | MikeJ[Laptop] | My boot |
| 01:34.26 | rene- | i believed John Todd was able to run over 100 calls using a 1mbit link to the internet |
| 01:35.04 | rene- | using g729, i have half that bandwidth, but i only need to run 30 calls and the ocassional ssh access to the box |
| 01:35.12 | Qwell | 8k/s... |
| 01:35.17 | Qwell | with overhead |
| 01:35.30 | MikeJ[Laptop] | you have 1/3 of that bandwindth.. not half |
| 01:35.31 | Qwell | yeah, you'd have to use an incredibly crappy codec to get 8k with overhead |
| 01:35.58 | rene- | MikeJ i believe 512 is half a megabit |
| 01:36.01 | Strom_M | like i said - lpc10 |
| 01:36.14 | MikeJ[Laptop] | ah.. I read wrong. |
| 01:36.17 | Qwell | Strom_M: unless it's 3k/s... |
| 01:36.30 | rene- | according to most people i have talked to sip + g729 is around 30kbits |
| 01:36.34 | MikeJ[Laptop] | I am telling you over 512 dsl.. it will be tight |
| 01:36.41 | rene- | very |
| 01:36.46 | MikeJ[Laptop] | 30? |
| 01:36.49 | MikeJ[Laptop] | nope |
| 01:36.50 | MikeJ[Laptop] | no way |
| 01:37.12 | X-Rob | 10, 15 maybe. |
| 01:37.19 | X-Rob | with g729 |
| 01:37.35 | MikeJ[Laptop] | let's rumble |
| 01:37.54 | MikeJ[Laptop] | yep... solaris over virtual pc just aint happening |
| 01:38.04 | X-Rob | MikeJ[Laptop], solaris hates you. |
| 01:38.10 | X-Rob | In fact, solaris hates just about everyone. |
| 01:38.15 | rene- | in his tests Todd showed how trunking would make the third and all following calls with g729 go at little over 10kbps |
| 01:38.38 | MikeJ[Laptop] | blah |
| 01:38.38 | rpm | can someone let me know why this overflows my Extension stack: exten => _.,a,VoiceMailMain(100) |
| 01:38.52 | X-Rob | don't use _. |
| 01:38.52 | MikeJ[Laptop] | with iax? |
| 01:38.55 | X-Rob | use _something. |
| 01:39.02 | rene- | yes IAX |
| 01:39.05 | Strom_M | rpm: _X. |
| 01:39.16 | MikeJ[Laptop] | you could do somthing similar w/ sip with larger packet size |
| 01:39.22 | MikeJ[Laptop] | or with h323 w/ trunking too |
| 01:39.34 | MikeJ[Laptop] | 6 of one, 1/2 dozen of..... |
| 01:41.36 | MikeJ[Laptop] | go to 80ms rtp and see how much that saves you |
| 01:41.45 | rene- | it will be interesting to see if it can run, i am not sure how to properly test it, i dont know if there is something like sipp for iax |
| 01:42.02 | rene- | MikeJ i wouldnt know how to change the rtp value of g729 |
| 01:42.04 | MikeJ[Laptop] | rene-, it's called asterisk :P |
| 01:42.14 | MikeJ[Laptop] | rtp value for g729? |
| 01:42.16 | MikeJ[Laptop] | lost me |
| 01:42.47 | rene- | i am lost too, the change you are talking about, does it goes in rtp.conf? |
| 01:42.53 | MikeJ[Laptop] | no |
| 01:42.57 | MikeJ[Laptop] | it's a define in code |
| 01:43.01 | MikeJ[Laptop] | in asterisk |
| 01:43.11 | rene- | in rtp.c or something like that? |
| 01:43.17 | MikeJ[Laptop] | I beleive |
| 01:43.18 | MikeJ[Laptop] | yes |
| 01:43.57 | MikeJ[Laptop] | just saying, there are many ways to sqeeze |
| 01:44.39 | rene- | i could try that, i could use call files to test connectivity to the box and then at some points say at the 10th 20th and 30th try with an actual phone and measure the call quality |
| 01:45.26 | MikeJ[Laptop] | VAD might help you too |
| 01:45.33 | rene- | i would need to be playing audio files in both boxes |
| 01:45.45 | MikeJ[Laptop] | to just bridge the link w/ somthing that supports vad and g729b |
| 01:46.18 | MikeJ[Laptop] | that gets complicated with asterisk |
| 01:46.29 | rene- | MikeJ: asterisk does not support VAD/silence supp right? |
| 01:46.33 | MikeJ[Laptop] | correct |
| 01:46.46 | MikeJ[Laptop] | but you could potentially use somthing to just bridge the link |
| 01:46.56 | rene- | but using another device that does then one could save bandwidth |
| 01:46.59 | MikeJ[Laptop] | but you would need to detect silence and convert |
| 01:47.13 | MikeJ[Laptop] | well.. just supporting vad isnt enough |
| 01:47.16 | MikeJ[Laptop] | that's easy |
| 01:47.53 | MikeJ[Laptop] | it's being able to convert it from a stream using vad to a constant stream of audio, and the other way, to detect silence, and to covert it to use vad |
| 01:48.03 | MikeJ[Laptop] | not sure if there is anything out there that will do that job or not |
| 01:48.10 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 01:48.16 | Ariel_ | hello everyone |
| 01:48.17 | MikeJ[Laptop] | anyone know of anything? |
| 01:48.21 | MikeJ[Laptop] | hello Ariel_ |
| 01:48.26 | rene- | hello Ariel_ |
| 01:48.47 | MikeJ[Laptop] | gotta run ... bbiab |
| 01:48.52 | rene- | see ya |
| 01:48.56 | Ariel_ | hope everything is going well we you all |
| 01:52.00 | *** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) |
| 01:52.32 | *** join/#asterisk oej (n=oej@gateway.digium.com) |
| 01:52.51 | *** join/#asterisk tuxinator_linux (n=tuxinato@netblock-68-183-112-100.dslextreme.com) |
| 01:53.51 | *** join/#asterisk Evanrude (n=david@ip68-107-162-212.lu.dl.cox.net) |
| 01:55.40 | Jon335 | I'm having some problems: http://pastebin.ca/47099 I get this whenever I dial out; running A@H with SPA3k |
| 01:55.59 | *** join/#asterisk brookshire (n=mbrooks@gateway.digium.com) |
| 01:58.22 | *** part/#asterisk rene- (n=rene@dsl-201-128-115-34.prod-infinitum.com.mx) |
| 01:58.43 | omal | hm, looks like asterisk really ahtes NAT |
| 01:58.46 | *** join/#asterisk TUplink (n=sdfgkjm@68-232-82-147.chvlva.adelphia.net) |
| 01:58.46 | ManxPower | Jon335, look at the /topic |
| 01:59.12 | ManxPower | I almost drove up to huntsville today |
| 01:59.24 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 01:59.26 | TUplink | Mar 26 20:59:16 WARNING[41992]: config.c:920 find_engine: Realtime mapping for 'iaxpeers' found to engine 'mysql', but the engine is not available any ideas? |
| 02:00.57 | *** join/#asterisk tehdely (n=delysiid@home.teambarry.org) |
| 02:01.35 | Heimidal | how do I save data from user input? instead of being forwarded to a different extension, I just want the numbers pushed.. |
| 02:01.41 | TUplink | anyone there? |
| 02:02.07 | ManxPower | TUplink, many of us don't run realtime |
| 02:02.23 | ManxPower | Heimidal, you need to read "show applications" |
| 02:02.24 | TUplink | i just messin with it? |
| 02:02.53 | *** part/#asterisk Jon335 (n=jon335@ottawa-hs-209-217-83-12.d-ip.magma.ca) |
| 02:03.45 | TUplink | could it be that i need to install asterisk-devel? |
| 02:05.19 | VeNoMouS_ | anyone tested the new eyebeam with asterisk 1.2.5 with video? |
| 02:05.23 | VeNoMouS_ | as im getting Mar 27 14:02:29 NOTICE[3660]: rtp.c:564 ast_rtp_read: Unknown RTP codec 127 received |
| 02:06.09 | Heimidal | ManxPower: I read throw it.. I still don't see anything that applies |
| 02:06.35 | ManxPower | how about "show application read" and "show application DBPut" |
| 02:06.49 | ManxPower | and of course "show application system" and "show application AGI" |
| 02:07.21 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 02:07.28 | *** join/#asterisk [hC] (n=hardcore@S01060004e21ea953.vc.shawcable.net) |
| 02:07.47 | *** join/#asterisk |Vulture| (n=Vulture@82.115.205.68.cfl.res.rr.com) |
| 02:07.57 | |Vulture| | Anyone here using Dell 850 servers? |
| 02:12.46 | [hC] | I'm using sc420/430's... |
| 02:12.49 | [hC] | and 1850's. |
| 02:13.02 | justinu | ibm 7094, here |
| 02:13.26 | robin_sz | 1850s here |
| 02:13.27 | Heimidal | ManxPower: thank you |
| 02:13.52 | |Vulture| | I use 420s |
| 02:14.10 | |Vulture| | but I was wondering about the 850s cause I would like a 1U |
| 02:14.16 | |Vulture| | and they are dirt cheap |
| 02:14.38 | robin_sz | 1300? |
| 02:14.48 | |Vulture| | ~900 |
| 02:14.50 | jbot | i heard 900 is for all intents and purpose line rate |
| 02:15.31 | |Vulture| | all and wise jbot |
| 02:16.23 | robin_sz | ~gxp2000 |
| 02:16.24 | jbot | [gxp2000] http://www.voip-info.org/wiki/view/GXP-2000 |
| 02:16.35 | robin_sz | coo. it knows too much |
| 02:17.05 | |Vulture| | hmmm xeon for $100 more with the 1850 |
| 02:17.26 | robin_sz | its worse than that ... |
| 02:17.29 | ManxPower | jbot_, grandstream's firmware is about as reliable as Windows98 |
| 02:18.04 | robin_sz | on some of there servers, the cheapest way to get another processor is to buy a second machine, remove the proc, and toss the rest .. |
| 02:18.27 | robin_sz | the Dell pricing has second CPUs more expensive than the whole machine sometimes |
| 02:19.00 | robin_sz | ManxPower: thats an insult ... |
| 02:19.09 | robin_sz | ManxPower: to win98 |
| 02:19.52 | robin_sz | I just wish theyd release something that kep the display working for more than 3 minutes |
| 02:20.24 | justinu | i haven't heard people complain about that |
| 02:20.35 | robin_sz | you are kidding, right? |
| 02:20.45 | robin_sz | new firmware on older phones ... |
| 02:20.55 | robin_sz | display blanks after a few minutes |
| 02:21.09 | justinu | oh, on older phones |
| 02:21.13 | justinu | i don't have any older ones |
| 02:21.13 | robin_sz | yeah |
| 02:21.21 | robin_sz | lucky ewe |
| 02:21.25 | ManxPower | I just can't trust a company that has THIS bad of a reputation for firmware, nor a company that put a numbers only display on an IP phone. |
| 02:21.47 | robin_sz | but ... they are cheap |
| 02:21.55 | justinu | it has a dot matrix display |
| 02:22.21 | robin_sz | mine has a blank display |
| 02:22.32 | |Vulture| | robin_sz: I got my 2850 for like $1300 with dual 3.0 xeons when they ran a deal awhile ago... that was a great deal |
| 02:22.50 | robin_sz | that was a good deal |
| 02:22.57 | |Vulture| | yea second proc free |
| 02:23.01 | robin_sz | coo |
| 02:23.29 | robin_sz | it will come around again |
| 02:23.45 | robin_sz | sigh ... poxy GXP 2000 |
| 02:23.59 | robin_sz | so near, yet so far |
| 02:24.15 | justinu | yeah |
| 02:24.17 | justinu | kinda sad |
| 02:24.49 | *** join/#asterisk jeebusroxors (n=jeebusro@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
| 02:24.56 | robin_sz | if they just hammered the firmware a bit more and had a re-design so it looked nicer, it wouldnt cost apenny more to make |
| 02:25.02 | robin_sz | and would sell |
| 02:25.42 | justinu | it's gotta sound right |
| 02:25.47 | justinu | the sound quality is my big issue |
| 02:25.56 | robin_sz | not been a problem here |
| 02:27.00 | robin_sz | do you think the superglue they use in hospitals is "special" or shall i just use some normal stuff? |
| 02:27.24 | justinu | it's blessed by the gods of the medical association |
| 02:27.27 | justinu | so it costs 100x more |
| 02:27.38 | robin_sz | but apart from that? |
| 02:27.43 | justinu | no idea |
| 02:28.08 | robin_sz | well, gotta be worth trying the normal stuff |
| 02:28.08 | justinu | is it going on your skin? |
| 02:28.31 | robin_sz | into a biggish cut ... |
| 02:28.41 | justinu | i dunno if i'd pour superglue into a cut |
| 02:28.50 | robin_sz | they do in hospitals |
| 02:28.51 | ManxPower | Has anyone seen American Wedding? |
| 02:29.38 | robin_sz | it was originally used in the vietnam debacle to rapid field repairs to soldiers |
| 02:29.39 | ManxPower | i dated a nurse. use good bandage tape, pull it togather, tape over it. |
| 02:29.45 | justinu | there are a lot of different types of glues |
| 02:30.04 | robin_sz | dunno if its exactly the same stuff as regular DIY superglue though |
| 02:30.53 | justinu | hard to say |
| 02:31.13 | justinu | if you could get the hospital product name, and get a list of ingrediants |
| 02:31.15 | justinu | then compare |
| 02:33.14 | justinu | http://en.wikipedia.org/wiki/Superglue |
| 02:34.06 | Strom_M | good old cyanoacrylate |
| 02:34.28 | MacDome | robin_sz: well, superglue was originally designed for medical use... I expect that the despense mechanisms they use in hospitals are better for fixing gashes, but the actual glue is the same |
| 02:34.36 | SwK | the difference between superglue and medical cyanoacrylate is the guarenteed purity of the medical stuff |
| 02:35.04 | MacDome | robin_sz: if you really have a gash, you should probably get it looked at... lest it scar |
| 02:35.09 | robin_sz | right ... some research later ;) |
| 02:35.19 | robin_sz | it turns out the medical stuff is different |
| 02:35.26 | SwK | altho I have used off the shelf superglue to close cuts heh |
| 02:35.40 | robin_sz | regular is metyhy alcohol based |
| 02:35.51 | robin_sz | medical is butyl or octyl based |
| 02:36.25 | robin_sz | but, yeah, people have used regular stuff with success. |
| 02:36.28 | MacDome | nifty, good to know |
| 02:36.34 | robin_sz | sorry .. that was WAY off topic ;) |
| 02:39.22 | loko | has anyone been successful in having asterisk run in vmware |
| 02:47.23 | brookshire | since when is #asterisk ever on topic ;) |
| 02:49.47 | Heimidal | does Playback support mp3 natively? |
| 02:50.25 | loko | do I need to still run the addmailbox command to create voicemail boxes? |
| 02:51.15 | VeNoMouS_ | Heimidal no |
| 02:51.44 | shmaltz | loko, when was the last time you used asterisk? |
| 02:51.51 | loko | long long time ago =) |
| 02:51.52 | VeNoMouS_ | Heimidal |
| 02:51.53 | VeNoMouS_ | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+MP3Player |
| 02:51.56 | VeNoMouS_ | ^^ u want that |
| 02:52.03 | justinu | he could also install format_mp3 |
| 02:52.34 | VeNoMouS_ | well that too |
| 02:52.46 | Heimidal | hmm, I'm trying to use MP3Player and it's not working well.. |
| 02:53.36 | VeNoMouS_ | justinu : whats it like under 1.2.5? |
| 02:53.47 | VeNoMouS_ | i ahvent tried it in awhile but man it sucked last time i tried it |
| 02:54.03 | VeNoMouS_ | to much cpu, and played shit @ wrong speed sometimes |
| 02:54.11 | loko | shmaltz so it just depends on the config now? |
| 02:54.29 | justinu | i haven't played with it lately myself |
| 02:54.39 | shmaltz | yep |
| 02:54.53 | VeNoMouS_ | Heimidal whats ure issue? |
| 02:57.29 | VeNoMouS_ | heh ive been connected to this irc for a week |
| 02:57.32 | VeNoMouS_ | not bad |
| 02:57.35 | VeNoMouS_ | : idle : 0 hours 2 mins 21 secs (signon: Mon Mar 13 14:11:14 2006) |
| 02:58.28 | *** join/#asterisk L|NUX (n=linux@202.5.145.58) |
| 03:00.34 | *** join/#asterisk newmember (n=username@S010600036d1139bb.cg.shawcable.net) |
| 03:04.40 | riksta | so i'm trying to call a box over IAX with a TDM30B that dials a Zap channel, but i can only use the codec ilbc not g729, can someone explain why that is please? |
| 03:06.17 | *** join/#asterisk juice (n=juice@mo-69-68-106-145.dyn.sprint-hsd.net) |
| 03:06.27 | SwK | riksta: do you have the g729 codec? |
| 03:06.54 | SwK | if you didnt license it you dont have it and you cant transcode G729 -> tm |
| 03:06.56 | SwK | tdm even |
| 03:07.38 | *** join/#asterisk evilphil (n=phil@dsl001-170-166.nyc1.dsl.speakeasy.net) |
| 03:07.54 | evilphil | hello all |
| 03:08.51 | evilphil | could anyone here help me with a really strange problem i'm having with the g729a codec? i'm at my wit's end.... |
| 03:09.46 | FuriousGeorge | do you have a licence to transcode it? |
| 03:09.59 | evilphil | yes |
| 03:10.14 | FuriousGeorge | what doesnt work? |
| 03:10.21 | evilphil | and basically everything is working except prompts generated by asterisk....such as the OGM, or the voicemail system |
| 03:10.51 | evilphil | i can call in and out, but when i call in, i don't hear the OGM...but if i dial my extension, the call goes through fine |
| 03:11.52 | FuriousGeorge | sounds like your licence isnt installed right, are you sure it works to call, say, another softphone using GSM |
| 03:11.53 | evilphil | plus i've checked to make sure the codec is properly registered, and i see "0/0 encoders/decoders of 5 licensed channels are currently in use" |
| 03:12.41 | evilphil | call another softphone using gsm? hrm.... |
| 03:12.54 | FuriousGeorge | seems to me like * isnt taking your prompts and turing'em into g729 for your device |
| 03:13.34 | evilphil | yeah, that's what i thought....the weird thing is that when i'm at an OGM (and not hearing anything), the show g729 output says "1/0 encoders/decoders of 5 licensed channels are currently in use" |
| 03:13.54 | FuriousGeorge | hey can i install multiple sound cards and have multiple dsp channels |
| 03:13.57 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 03:13.57 | *** mode/#asterisk [+o russellb] by ChanServ |
| 03:14.02 | FuriousGeorge | evilphil: im not sure what to tell you |
| 03:14.06 | FuriousGeorge | obviously gsm works right |
| 03:14.23 | FuriousGeorge | try to call a device that isnt using g729 |
| 03:14.35 | evilphil | well, i have a cisco phone that only supports ulaw and g729....but ulaw worked.... |
| 03:15.30 | FuriousGeorge | so call a phone that is ulaw and see of you ehar it |
| 03:15.33 | *** join/#asterisk NetrixTardis (n=leoem@cpe-24-28-92-172.austin.res.rr.com) |
| 03:16.09 | FuriousGeorge | anyone know if one can install multiple sound cards for multiple dsp channels? |
| 03:16.48 | NetrixTardis | anyone seen some fool named "gigagod" in the last year or so? |
| 03:17.11 | FuriousGeorge | ~seen gigagod |
| 03:17.13 | jbot | gigagod <~test@24-155-122-21.dyn.grandenetworks.net> was last seen on IRC in channel #asterisk, 466d 19h 42m 21s ago, saying: 'got kicked'. |
| 03:17.23 | FuriousGeorge | lol |
| 03:17.55 | FuriousGeorge | ask his isp who had that ip 466 days ago then go ask for your money back :) |
| 03:18.05 | FuriousGeorge | j/k |
| 03:18.36 | NetrixTardis | FuriousGeorge: you know this <sorry excuse for the living> ? |
| 03:18.58 | FuriousGeorge | no, i just like making jbot do tricks |
| 03:19.04 | NetrixTardis | ah |
| 03:19.09 | FuriousGeorge | ~seen FuriousGeorge |
| 03:19.11 | jbot | furiousgeorge is currently on #asterisk. Has said a total of 16 messages. Is idling for 2s, last said: '~seen FuriousGeorge'. |
| 03:19.24 | FuriousGeorge | ~FuriousGeorge |
| 03:19.25 | jbot | well, furiousgeorge is a knife-fighting monkey last seen with The Man with the Yellow Bat |
| 03:22.06 | *** join/#asterisk Op3r (n=op3r@202.71.189.90) |
| 03:22.19 | Op3r | anyone tried chanspy? |
| 03:24.12 | *** part/#asterisk NetrixTardis (n=leoem@cpe-24-28-92-172.austin.res.rr.com) |
| 03:24.38 | *** join/#asterisk Utah_Dave (n=boucha@c-67-166-71-219.hsd1.ut.comcast.net) |
| 03:25.28 | *** join/#asterisk weinerk (n=irc@88.153.4.52) |
| 03:25.36 | Op3r | anyone can help me with chanspy? |
| 03:28.04 | *** join/#asterisk Wi_Fi (n=OUT@c-24-127-12-85.hsd1.ca.comcast.net) |
| 03:28.07 | riksta | ask the Qn |
| 03:32.25 | *** join/#asterisk Kumbang (n=kumbang@167.205.24.5) |
| 03:33.42 | Op3r | I tried to do this |
| 03:33.43 | Op3r | ;### call barging (Edwin is your daddy!)##### |
| 03:33.43 | Op3r | exten => *888,1,Answer |
| 03:33.43 | Op3r | exten => *888,2,Wait(1) |
| 03:33.43 | Op3r | exten => *888,3,ChanSpy(SIP/,q) |
| 03:33.43 | Op3r | exten => *888,4,Hangup |
| 03:34.16 | riksta | i think the comment broke it |
| 03:34.27 | Op3r | its the comment? |
| 03:34.41 | Op3r | I thought ; is just a comment? |
| 03:34.42 | Op3r | :( |
| 03:34.44 | riksta | </sarcasm> |
| 03:34.47 | [av]bani | http://unrule.info/files/linux_is_bad.gif |
| 03:35.00 | Op3r | riksta: any idea? |
| 03:35.09 | Op3r | because we have 24 extension |
| 03:35.22 | Op3r | I tried dialing *888 then extension number |
| 03:35.25 | Op3r | its busy |
| 03:35.26 | Op3r | :( |
| 03:35.27 | VeNoMouS_ | Op3r |
| 03:35.30 | VeNoMouS_ | lol |
| 03:35.33 | Op3r | VeNoMouS_: jj! |
| 03:35.40 | VeNoMouS_ | did that shit work that i gave u? |
| 03:35.45 | VeNoMouS_ | u toook off b4 i got home |
| 03:35.45 | Op3r | VeNoMouS_: yes it did |
| 03:35.48 | VeNoMouS_ | sweet |
| 03:36.13 | Op3r | VeNoMouS_: I just changed it to go to outgoing directory |
| 03:36.20 | *** join/#asterisk bkw_ (n=bkw_@adsl-70-142-62-199.dsl.tul2ok.sbcglobal.net) |
| 03:36.22 | Op3r | not on the monitor folder itself |
| 03:36.23 | Op3r | :D |
| 03:36.42 | VeNoMouS_ | kool |
| 03:36.49 | VeNoMouS_ | but u understand what i ment |
| 03:49.23 | *** join/#asterisk Op3r (n=op3r@202.71.189.90) |
| 03:49.24 | Op3r | y0 |
| 03:49.26 | Op3r | VeNoMouS_ |
| 03:49.45 | Op3r | sorry I got disconnected |
| 03:49.47 | Op3r | :( |
| 03:51.24 | *** join/#asterisk bmg505 (n=leon@165.146.59.47) |
| 03:52.02 | Op3r | any one up? |
| 03:52.12 | Op3r | anybody familiar with chanspy? |
| 03:52.19 | Op3r | any other way to barge calls? |
| 03:53.55 | VeNoMouS_ | yo |
| 03:53.56 | VeNoMouS_ | sorry |
| 03:54.01 | VeNoMouS_ | just sitting here listening to ppls calls |
| 03:54.02 | VeNoMouS_ | heh |
| 03:54.12 | VeNoMouS_ | im doing chanspy atm bro |
| 03:54.12 | VeNoMouS_ | lol |
| 03:54.15 | VeNoMouS_ | what u wanna know |
| 03:54.28 | Darwin_35 | how to set it up |
| 03:54.34 | VeNoMouS_ | well |
| 03:54.40 | Darwin_35 | a good cut and pastebin |
| 03:54.46 | *** join/#asterisk kuku5 (n=kuku5@c-71-201-217-245.hsd1.il.comcast.net) |
| 03:54.54 | VeNoMouS_ | exten => somenumber,1,ChanSpy(Sip/|q) |
| 03:55.00 | VeNoMouS_ | done |
| 03:55.03 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 03:55.07 | Darwin_35 | ok |
| 03:55.12 | Op3r | thats it? |
| 03:55.18 | kuku5 | why was there a fork ? |
| 03:55.22 | Op3r | no other way like this |
| 03:56.06 | Op3r | exten => *888,1,Answer |
| 03:56.07 | Op3r | exten => *888,2,Wait(1) |
| 03:56.07 | Op3r | exten => *888,3,ChanSpy(SIP/,q) |
| 03:56.07 | Op3r | exten => *888,4,Hangup |
| 03:56.14 | VeNoMouS_ | no , |
| 03:56.16 | VeNoMouS_ | put | |
| 03:56.18 | Darwin_35 | man I still need a callback if busy setup |
| 03:56.20 | VeNoMouS_ | SIP/|q |
| 03:56.22 | Qwell | doesn't matter |
| 03:56.35 | Qwell | and you kinda need a channel name |
| 03:56.50 | Op3r | so that stuff is wrong? |
| 03:57.04 | VeNoMouS_ | no |
| 03:57.16 | rpm | exten => a888,1,ChanSpy(SIP/xxx,q) |
| 03:57.24 | Qwell | rpm: a? |
| 03:57.35 | rpm | a == * isn't it? |
| 03:57.35 | VeNoMouS_ | u dont need xxx |
| 03:57.39 | Qwell | no.. |
| 03:57.52 | VeNoMouS_ | i know i was saying u dont |
| 03:58.20 | VeNoMouS_ | rpm u can just do SIP/|q |
| 03:58.36 | rpm | ah. |
| 03:59.08 | VeNoMouS_ | q means no beep |
| 03:59.10 | Op3r | VeNoMouS_: that stuff is wrong? |
| 03:59.14 | Qwell | hmm, can't say I've ever used chanspy |
| 03:59.23 | Qwell | VeNoMouS_: Does it let you cycle through channels or something? |
| 03:59.24 | VeNoMouS_ | op3r just do |q not ,q |
| 03:59.27 | VeNoMouS_ | yea press * |
| 03:59.31 | Qwell | neat |
| 03:59.33 | VeNoMouS_ | it will cycle the channel |
| 03:59.37 | Qwell | and | or , would work |
| 03:59.47 | VeNoMouS_ | Qwell not according to doc |
| 03:59.51 | VeNoMouS_ | http://www.voip-info.org/wiki-Asterisk+cmd+ChanSpy |
| 03:59.54 | Qwell | according to the code, you can ;) |
| 04:00.06 | Qwell | bbl, tv |
| 04:00.08 | VeNoMouS_ | heh ure prob right |
| 04:00.51 | VeNoMouS_ | u have to ave an established sip |
| 04:01.33 | VeNoMouS_ | btw |
| 04:02.39 | Op3r | god damn wiki |
| 04:03.14 | *** part/#asterisk Utah_Dave (n=boucha@c-67-166-71-219.hsd1.ut.comcast.net) |
| 04:03.58 | VeNoMouS_ | Op3r u got it yet? |
| 04:04.18 | *** join/#asterisk kainam (n=Jake@202.137.160.110) |
| 04:04.43 | Op3r | exten => *888,3,ChanSpy(SIP/ |q) |
| 04:04.44 | Op3r | ? |
| 04:04.49 | Op3r | thats correct? |
| 04:04.52 | VeNoMouS_ | no space |
| 04:04.56 | Op3r | ok |
| 04:05.00 | VeNoMouS_ | SIP/|q |
| 04:05.04 | VeNoMouS_ | do that and reload |
| 04:05.08 | VeNoMouS_ | when u dial *888 |
| 04:05.13 | VeNoMouS_ | it will just sit there until a sip connects |
| 04:05.26 | VeNoMouS_ | if u ave more then 1 sip call running |
| 04:05.32 | VeNoMouS_ | press * to cycle the channels |
| 04:05.52 | VeNoMouS_ | so just dial *888 and goto another fone |
| 04:05.54 | VeNoMouS_ | and call some 1 |
| 04:06.03 | *** join/#asterisk Nukemizer (n=Nuke@160.7.249.15) |
| 04:06.09 | Op3r | i thought you can do this. *888 then dial extension # |
| 04:06.10 | VeNoMouS_ | u will hear it on the fone u dial *888 on |
| 04:06.13 | VeNoMouS_ | np |
| 04:06.14 | VeNoMouS_ | no |
| 04:06.27 | VeNoMouS_ | it will listen to all channels |
| 04:06.40 | VeNoMouS_ | u could setup so enter in an ext |
| 04:06.44 | Op3r | you cant just do a locking on an extension? |
| 04:06.58 | VeNoMouS_ | and that will do ChanSpy(SIP/{monitorex}|q) |
| 04:07.14 | Op3r | ;### call barging (Op3r is your daddy!)##### |
| 04:07.14 | Op3r | exten => *888,1,Answer |
| 04:07.14 | Op3r | exten => *888,2,Wait(1) |
| 04:07.14 | Op3r | exten => *888,3,ChanSpy(SIP/|q) |
| 04:07.14 | Op3r | exten => *888,4,Hangup |
| 04:07.18 | Op3r | how about that? |
| 04:08.06 | VeNoMouS_ | that will listen for all sip |
| 04:08.15 | Op3r | :( |
| 04:08.19 | VeNoMouS_ | why :( |
| 04:08.25 | VeNoMouS_ | if u had 3 ppl on calls |
| 04:08.30 | VeNoMouS_ | u press * until u get the right channel |
| 04:08.35 | VeNoMouS_ | u wont hear all 3 channels @ the same time |
| 04:08.39 | Op3r | oh ok |
| 04:08.48 | VeNoMouS_ | gimmie a sec i'll write u something so u can enter just the ext if u want |
| 04:08.50 | Op3r | no other way to lock into a channel? |
| 04:08.52 | Op3r | ok |
| 04:08.53 | Op3r | <PROTECTED> |
| 04:12.08 | VeNoMouS_ | maybe something liket his |
| 04:12.10 | VeNoMouS_ | [spy] |
| 04:12.10 | VeNoMouS_ | exten => s,1,BackGround(please-enter-the) |
| 04:12.10 | VeNoMouS_ | exten => s,n,BackGround(extension) |
| 04:12.10 | VeNoMouS_ | exten => s,n,Set(TIMEOUT(digit)=5) ; Set Digit Timeout to 5 seconds |
| 04:12.10 | VeNoMouS_ | exten => s,n,Set(TIMEOUT(response)=10) ; Set Response Timeout to 10 seconds |
| 04:12.11 | VeNoMouS_ | exten => s,n,WaitExten(10) |
| 04:12.13 | VeNoMouS_ | exten => _XXXX,1,ChanSpy(SIP/${EXTEN}|q) |
| 04:12.15 | VeNoMouS_ | exten => i,1,PlayBack(bad) |
| 04:12.17 | VeNoMouS_ | exten => i,2,PlayBack(extension) |
| 04:12.19 | VeNoMouS_ | exten => i,3,Goto(s,1) |
| 04:12.33 | Qwell | ~pb |
| 04:12.42 | jbot | pb is, like, a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 04:12.42 | VeNoMouS_ | heh yea sorry bout that |
| 04:12.42 | *** join/#asterisk amdtech (n=ditto@ip70-179-174-151.dl.dl.cox.net) |
| 04:13.10 | VeNoMouS_ | op3r then just ave ure *888 instead of 3,Chan do 3,Goto(spy,s,1); |
| 04:14.09 | Op3r | oh ok |
| 04:14.22 | Op3r | bot |
| 04:14.36 | VeNoMouS_ | u get the idea? |
| 04:14.58 | VeNoMouS_ | <VeNoMouS_> exten => s,n,Set(TIMEOUT(digit)=5) ; Set Digit Timeout to 5 seconds |
| 04:15.02 | *** join/#asterisk niZon (n=ilt@S010600080db4ab60.wp.shawcable.net) |
| 04:15.04 | VeNoMouS_ | will listen for 4 digits |
| 04:15.32 | VeNoMouS_ | op3r well take this to msg |
| 04:15.35 | *** join/#asterisk Cation (n=rafnorwi@user-0cev7pb.cable.mindspring.com) |
| 04:15.38 | Op3r | oh ok |
| 04:28.31 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:33.40 | Darwin_35 | why i am in on tis |
| 04:33.44 | Darwin_35 | this |
| 04:35.03 | VeNoMouS_ | on what |
| 04:35.23 | Darwin_35 | the chanspy |
| 04:35.31 | Darwin_35 | more to add to my dialing plan |
| 04:35.53 | Darwin_35 | but my big thhis is a call back if busy setup |
| 04:36.09 | harlequin516 | When you originate a call asterisk essentially callouts to the Specified channel and the when answers connects the the context,extension,priority. What if I want my dial plan to make the origination call and the destination call. What would I specify for my dialplan/callout file? Do I have to specify extension/priority twice, once in the local channel string and then again in the extension and priority? |
| 04:36.10 | Darwin_35 | wich I have yet to find |
| 04:38.05 | harlequin516 | Is my question clear? |
| 04:38.22 | VeNoMouS_ | wtf |
| 04:38.24 | VeNoMouS_ | no |
| 04:40.10 | harlequin516 | Okay ya know when you originate a call from astreisk, commonly you need to specify the channel (zap/1/623666-7777), context(callout), extension (2000), priority(1) |
| 04:40.57 | harlequin516 | But I want the channel speicied later in the the agi script called from the dialplan. |
| 04:41.14 | harlequin516 | I was told to use local channel to do this. |
| 04:41.52 | harlequin516 | The callout file still needs the four parameters as exemplied above. |
| 04:42.04 | harlequin516 | So I do channel |
| 04:43.06 | harlequin516 | <PROTECTED> |
| 04:43.49 | harlequin516 | Won't this connect two calls to exten 2000 of context callout? |
| 04:44.31 | harlequin516 | Anyone know what I am talking about/ |
| 04:44.33 | harlequin516 | ? |
| 04:46.12 | Darwin_35 | sex on the beach ? |
| 04:46.26 | *** join/#asterisk inv_Arp (i=junya@adsl-10-132-83.mia.bellsouth.net) |
| 04:46.31 | tsume | sex on the beach with a bitch dog ;) |
| 04:46.49 | VeNoMouS_ | see if u said with a turtle that would've been funny |
| 04:46.58 | VeNoMouS_ | but what u said makes ppl want to back away slowly from u |
| 04:47.00 | harlequin516 | Maybe that's easier than what i am trying to do... |
| 04:47.19 | tsume | VeNoMouS_: would have to be a tourtoise :) |
| 04:47.21 | VeNoMouS_ | ya just had to take it too far didnt u |
| 04:47.46 | harlequin516 | Well have you ever had sex on the beach? |
| 04:47.51 | VeNoMouS_ | yea |
| 04:47.55 | VeNoMouS_ | not with a dog tho |
| 04:47.57 | tsume | VeNoMouS_: would be intresting to see someone having sex with a gator :) |
| 04:47.58 | VeNoMouS_ | or a turtle |
| 04:48.06 | harlequin516 | Sand don't feel too good in sensitive places... |
| 04:48.17 | VeNoMouS_ | pft get a blanket or a jacket foolio |
| 04:48.23 | tsume | they have a size of a humans after all, just a little bend :P |
| 04:48.27 | VeNoMouS_ | what are u stupid? |
| 04:48.40 | tsume | harlequin516: no kidding :P |
| 04:48.42 | *** join/#asterisk kfuq (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 04:49.37 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 05:12.06 | *** join/#asterisk Chai_Sangeen (n=aljishi@c-24-147-123-10.hsd1.ma.comcast.net) |
| 05:12.17 | Chai_Sangeen | hello everyone |
| 05:13.46 | Chai_Sangeen | can anyone help how can I trigger the following URL from linux console: http://serverip/indigoControl.php?actionGroup=Normal&command=Trigger |
| 05:14.09 | VeNoMouS_ | wtf is serverip |
| 05:14.30 | Chai_Sangeen | its the ip of the computer running the service |
| 05:14.38 | glm2k | what do you mean by "trigger"? |
| 05:14.45 | VeNoMouS_ | browse to it? |
| 05:14.54 | VeNoMouS_ | oh i know what he wants |
| 05:14.59 | glm2k | w3, links, links-graphic... |
| 05:15.03 | VeNoMouS_ | Chai_Sangeen do lynx --dump http://serverip/indigoControl.php?actionGroup=Normal&command=Trigger |
| 05:15.10 | glm2k | or lynx yes |
| 05:15.11 | rpm | System(wget http://serverip/indigoControl.php?actionGroup=Normal&command=Trigger) |
| 05:15.17 | glm2k | lol |
| 05:15.19 | glm2k | that works |
| 05:15.21 | VeNoMouS_ | nah wget dont do posts |
| 05:15.28 | VeNoMouS_ | and if it needs a post then its scripted |
| 05:15.28 | Chai_Sangeen | let me try |
| 05:15.38 | glm2k | i forgot lynx |
| 05:15.55 | VeNoMouS_ | if u want it to just quit afterwards just do --dump |
| 05:16.14 | rpm | wget can do post requests. |
| 05:16.21 | rpm | <PROTECTED> |
| 05:16.29 | rpm | --post-data=string |
| 05:16.30 | rpm | --post-file=file |
| 05:18.51 | Chai_Sangeen | VeNoMouS_, do i have to install lynx? -bash: lynx: command not found |
| 05:19.06 | VeNoMouS_ | yes |
| 05:19.21 | rpm | cat stuff > /dev/net/tcp/host/port |
| 05:19.23 | rpm | :P |
| 05:22.00 | Chai_Sangeen | VeNoMouS_, when i execute "lynx --dump http://192.168.1.232/indigoControl.php?actionGroup=Normal&command=Trigger" i get: No Command Received.; |
| 05:22.10 | VeNoMouS_ | <Chai_Sangeen> VeNoMouS_, do i have to install lynx? -bash: lynx: command not found |
| 05:22.12 | VeNoMouS_ | <VeNoMouS_> yes |
| 05:22.24 | Chai_Sangeen | VeNoMouS_, yeah i installed it |
| 05:22.34 | VeNoMouS_ | then put full realitive path |
| 05:22.54 | VeNoMouS_ | ie /usr/bin/lynx ....... |
| 05:26.11 | Chai_Sangeen | VeNoMouS_, /usr/bin/lynx --dump http://192.168.1.232/indigoControl.php?actionGroup=Normal&command=Trigger gives me:[5] 1348 |
| 05:26.22 | VeNoMouS_ | cause of the & |
| 05:26.23 | *** join/#asterisk Darwin_35 (n=Darwin@c-24-9-75-234.hsd1.co.comcast.net) |
| 05:26.27 | VeNoMouS_ | wrap ure url in "" |
| 05:26.30 | VeNoMouS_ | so |
| 05:26.34 | VeNoMouS_ | <PROTECTED> |
| 05:26.39 | VeNoMouS_ | <PROTECTED> |
| 05:27.15 | Chai_Sangeen | VeNoMouS_, YES!! it worked thank you so much ! |
| 05:28.06 | Chai_Sangeen | VeNoMouS_, what is the best way to use it in extentions.conf |
| 05:28.15 | VeNoMouS_ | system(.... |
| 05:28.41 | Qwell | chances are, this could easily be solved by app_curl |
| 05:30.03 | Corydon76-home | Yay! Use my app! |
| 05:30.13 | Corydon76-home | or my function, which isn't deprecated |
| 05:31.37 | VeNoMouS_ | weinerk prob, but he didnt say it was for asterisk @ first |
| 05:31.49 | VeNoMouS_ | err Qwell |
| 05:31.52 | VeNoMouS_ | even |
| 05:34.15 | *** part/#asterisk SpaceBass (n=SpaceBas@static-71-251-230-2.rcmdva.fios.verizon.net) |
| 05:35.33 | omal | hm. i can't seem to add a working extension to this sample config |
| 05:36.49 | VeNoMouS_ | which context u adding it too |
| 05:37.03 | omal | hm, actually the one i added for FWD worked for their test |
| 05:37.05 | omal | default |
| 05:37.28 | omal | i just want to add an extension 2600 and have it play the hello-world right now |
| 05:37.44 | omal | then i'll see about getting that extension to refer over to my other asterisk server |
| 05:38.04 | omal | i get a 404 error from xlite on it. hm. |
| 05:38.15 | omal | because of the issues, i reduced to something very simple |
| 05:38.47 | omal | exten => 2600,1,Playback(hello-world) |
| 05:48.31 | VeNoMouS_ | heh come on d/l eyebeam and crack it fuck xlite! |
| 05:48.33 | VeNoMouS_ | :P |
| 05:48.44 | VeNoMouS_ | and 404 is normally user not found |
| 05:48.47 | VeNoMouS_ | check ure sip.conf |
| 05:48.53 | VeNoMouS_ | or ure extentions |
| 05:49.06 | omal | xlite is the same for linux/osx/windows, and seems to be used in most examples |
| 05:49.18 | omal | other extensions work fine |
| 05:49.27 | omal | i can run the asterisk demo from 500 |
| 05:49.35 | omal | i can dial FWD with 613, which i added |
| 05:49.35 | VeNoMouS_ | does the file hello-world exiost? |
| 05:49.43 | VeNoMouS_ | hello-worldl.gsm |
| 05:49.47 | VeNoMouS_ | in /var/lib/asterisk/sounds/ |
| 05:49.51 | omal | for some reason i can't make this third one do a thing |
| 05:49.59 | VeNoMouS_ | err hello-world.gsm |
| 05:50.14 | omal | yup |
| 05:50.31 | VeNoMouS_ | when u asterisk -vvvr |
| 05:50.33 | VeNoMouS_ | err |
| 05:50.39 | VeNoMouS_ | when u run asterisk -vvvr |
| 05:50.43 | VeNoMouS_ | and u dial 2600 |
| 05:50.45 | VeNoMouS_ | on xlite |
| 05:50.50 | VeNoMouS_ | wat does asterisk say |
| 05:51.19 | omal | hm. nothing. |
| 05:51.40 | VeNoMouS_ | well its gotta say something |
| 05:51.51 | VeNoMouS_ | unless ure xlite isnt registered |
| 05:52.13 | omal | wehn i type 500 i get output, and i hear the demo |
| 05:52.17 | omal | 2600, nothing |
| 05:52.21 | omal | hmf |
| 05:52.32 | omal | i _must_ have that extension in the wrong place or something |
| 05:56.23 | omal | man, wtf |
| 05:56.31 | *** join/#asterisk tengulre (n=tengulre@222.90.66.4) |
| 05:56.45 | omal | i ought to start from scratch, part of my problem is that sample configs are loaded with stuff |
| 05:57.02 | omal | typically thats a good way to get your feet wet, uncomment just whats needed |
| 05:57.10 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
| 05:57.10 | VeNoMouS_ | who are u talking to? |
| 05:57.14 | omal | but man...i can't explain why i'm getting nothing here |
| 05:57.28 | omal | apparently, myself :D |
| 05:59.01 | *** join/#asterisk clive- (n=pirch@dsl-146-83-101.telkomadsl.co.za) |
| 05:59.33 | *** join/#asterisk lorinc (n=ang@caracas-1172.adsl.interware.hu) |
| 06:08.13 | VeNoMouS_ | anyways im out |
| 06:08.17 | VeNoMouS_ | time to go home |
| 06:08.20 | VeNoMouS_ | Mon Mar 27 18:06:26 NZST 2006 |
| 06:13.08 | *** join/#asterisk mcnobody (n=laaksola@laaksola.net) |
| 06:14.54 | omal | AHA |
| 06:15.06 | omal | i had context=demo under the channel in sip.conf |
| 06:15.37 | omal | HELLO WORLD |
| 06:17.00 | *** join/#asterisk badfish (i=dfdsf@43-87.69-92-cpe.cableone.net) |
| 06:17.02 | badfish | http://www.challenge-tv.com/index.php?mode=demodetail&demo=31023&dl=3 |
| 06:17.10 | badfish | nice article on microsoft, future plans |
| 06:19.27 | *** join/#asterisk adelas (n=booger@rrcs-24-199-21-141.west.biz.rr.com) |
| 06:21.49 | *** part/#asterisk badfish (i=dfdsf@43-87.69-92-cpe.cableone.net) |
| 06:23.06 | *** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim) |
| 06:24.44 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 06:25.00 | kmilitzer | Morning everyone ... |
| 06:26.43 | *** join/#asterisk tehdely (n=delysiid@home.teambarry.org) |
| 06:27.11 | tehdely | hi @ all |
| 06:27.53 | ManxPower | No matter how old I get, I still adore the show Daria |
| 06:29.55 | FuriousGeorge | anyone know if its possible to have multiple sound cards and multiple dsp channels? |
| 06:30.58 | ManxPower | FuriousGeorge, If you can do multiple soundcards with Linux you should be able to with Asterisk |
| 06:32.00 | `Sauron | long live also |
| 06:32.18 | `Sauron | err |
| 06:32.19 | `Sauron | ALSA |
| 06:33.13 | FuriousGeorge | i guess its safe to assume it can be done with linux/alsa :) |
| 06:33.19 | *** join/#asterisk Falle (n=falle@falle.se) |
| 06:33.43 | `Sauron | Yep |
| 06:35.25 | tehdely | james are you in here? :P |
| 06:35.56 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 06:35.56 | *** mode/#asterisk [+o denon] by ChanServ |
| 06:37.10 | *** join/#asterisk Greek-Boy (n=grb@193.220.93.162) |
| 06:38.22 | *** join/#asterisk nayyares (n=Nayyar@mbl-65-151-218.dsl.net.pk) |
| 06:42.14 | FuriousGeorge | can someone translate this for me: chan_iax2.c:7810 iax2_poke_peer: Still have a callno... |
| 06:42.22 | FuriousGeorge | ~callno |
| 06:47.26 | FuriousGeorge | anyone? |
| 06:49.08 | *** join/#asterisk JffMRIII (n=JffMRIII@c-67-167-205-28.hsd1.il.comcast.net) |
| 06:49.29 | JffMRIII | hello all |
| 06:49.41 | FuriousGeorge | it seems to be preventing one of my boxes from resolving the other ones |
| 06:50.36 | JffMRIII | Question: I am attempting to get a cisco 7960 to talk to my openwrt asterisk box without sucess. Can anyone point me in the right direction? |
| 06:51.19 | FuriousGeorge | damn, here we go again with the unreachable peers |
| 06:51.29 | FuriousGeorge | JffMRIII: what happens when you try to register? |
| 06:51.32 | FuriousGeorge | just nothing? |
| 06:51.37 | JffMRIII | correct |
| 06:51.44 | Abydos313 | JffMRIII how well does asterisk run on a router? can it handle multiple lines |
| 06:51.47 | JffMRIII | configuring CM list |
| 06:51.48 | FuriousGeorge | can you log any client into your asterisk |
| 06:52.06 | JffMRIII | it is running ok |
| 06:52.16 | JffMRIII | I havent really banged on it yet |
| 06:52.27 | JffMRIII | because I have to get this cisco huked up |
| 06:52.32 | Abydos313 | let us know when you find out :) |
| 06:52.37 | FuriousGeorge | try with x-lite or any other client |
| 06:52.42 | FuriousGeorge | im assuming you can ping the thing |
| 06:52.48 | JffMRIII | then I will order a voip provider which I need a recommendatino if any one is using |
| 06:52.54 | FuriousGeorge | i ehar they can handle 2-3 calls |
| 06:52.56 | FuriousGeorge | at a time |
| 06:53.02 | JffMRIII | ping = yes |
| 06:53.21 | Abydos313 | FuriousGeorge that's what i read but wanted to hear real world trials |
| 06:54.11 | JffMRIII | I was at wispcon with mark had it running on the rt3 |
| 06:54.17 | JffMRIII | was running great |
| 06:54.32 | Abydos313 | nice |
| 06:54.36 | tehdely | JffMRIII: does this cisco have the SIP firmware or the CM firmware |
| 06:54.37 | JffMRIII | just now getting back into it |
| 06:54.46 | JffMRIII | CM |
| 06:54.57 | JffMRIII | i think if I need the sip I have to flash it |
| 06:55.00 | tehdely | you do |
| 06:55.03 | JffMRIII | but unknown |
| 06:55.04 | tehdely | but asterisk supports SCCP |
| 06:55.11 | JffMRIII | yeppers |
| 06:55.12 | tehdely | how do you have it configured on asterisk's end |
| 06:55.19 | JffMRIII | probably now |
| 06:55.25 | JffMRIII | in the sip.cong |
| 06:55.30 | tehdely | you wouldn't configure it there :P |
| 06:55.53 | JffMRIII | where would I find that |
| 06:55.59 | tehdely | /etc/asterisk/skinny.conf |
| 06:56.05 | tehdely | is where you define SCCP users |
| 06:56.06 | JffMRIII | ahh good call |
| 06:57.42 | Abydos313 | anyone use a spa3k? i can make calls perfectly, just can't receive them. i do with xlite though |
| 06:57.56 | omal | i'm working on setting one up actually |
| 06:58.05 | omal | but in an odd configuration |
| 06:58.43 | Abydos313 | the calls thru it have great quality so far |
| 06:59.03 | Abydos313 | i'm using telasip |
| 06:59.33 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 06:59.53 | FuriousGeorge | this is pretty bad. it appears every time my ip changes im no longer able to contact any of my peers |
| 06:59.59 | FuriousGeorge | including my provider |
| 07:00.15 | FuriousGeorge | i have a catch in the dialplan if provider is unavailable, but it is still a pain in the ass |
| 07:00.34 | Abydos313 | what does the catch do? |
| 07:00.36 | FuriousGeorge | and the only way to get them back seems to be to STOP asterisk for a while |
| 07:00.50 | FuriousGeorge | Abydos313: uses pots depending on dialstatus |
| 07:01.20 | Abydos313 | maybe you need to remove registration before you reregister. saw that option in the spa3k |
| 07:01.36 | FuriousGeorge | remove registration before i register? what does that mean |
| 07:01.49 | FuriousGeorge | oh like comment it out |
| 07:02.19 | FuriousGeorge | you know that used to work, but since then i took out the register all together, set the boxes up as "friends" and host=thebox.dynu.org |
| 07:02.56 | Abydos313 | why doesn't that work anymore |
| 07:02.58 | FuriousGeorge | so now there is no register statement to comment and uncomment, but the peers become unreachable less often |
| 07:03.22 | FuriousGeorge | because there is no register statement. i removed it because it is actually more reliable to set them up as static friends |
| 07:03.33 | FuriousGeorge | but obviously not THAT reliable |
| 07:03.35 | FuriousGeorge | b/c here i am |
| 07:03.40 | Abydos313 | :) |
| 07:03.40 | *** join/#asterisk Kumbang (n=kumbang@167.205.24.5) |
| 07:04.00 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-18.claranet.co.uk) |
| 07:04.24 | FuriousGeorge | i cant even find out what the f*ck "still have a callno" means. google doesnt seem to know |
| 07:04.47 | Abydos313 | i've never seen it |
| 07:04.48 | FuriousGeorge | no one in here knows either |
| 07:04.56 | FuriousGeorge | oh, i have |
| 07:05.15 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 07:06.35 | FuriousGeorge | i guess its a bug |
| 07:06.48 | FuriousGeorge | can u imagine if ssh were this unreliable |
| 07:07.16 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:07.38 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 07:07.40 | Abydos313 | that would suck |
| 07:08.54 | riksta | what's the delay like when sending SIP data over a vpn like openvpn? |
| 07:09.15 | wasim | depends |
| 07:09.21 | FuriousGeorge | riksta: encryption? |
| 07:09.33 | riksta | FuriousGeorge: just curious |
| 07:10.14 | FuriousGeorge | anyway, im sure if i had a static ip at all 4 locations it wouldnt be an issue, but why should i have to pay 1000 USD a year for that. either * will interface with among dynamic ips or it wont, but this half and half crap is starting to annoy me |
| 07:10.32 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 07:10.32 | *** mode/#asterisk [+o denon] by ChanServ |
| 07:10.35 | riksta | why do you need a static ip? |
| 07:10.50 | riksta | oh not for me, i'll read up |
| 07:10.54 | FuriousGeorge | b/c when my ip changes * shits the bed |
| 07:11.09 | MikeJ__ | ewww |
| 07:11.10 | MGSsancho | use a dynamic dns service |
| 07:11.14 | FuriousGeorge | i am |
| 07:11.16 | riksta | use register and dynamic dns |
| 07:11.23 | FuriousGeorge | register is even less reliable |
| 07:11.25 | MGSsancho | dyndns.org |
| 07:11.34 | FuriousGeorge | i do host=box.dyndns.org |
| 07:11.37 | FuriousGeorge | among friends |
| 07:11.41 | riksta | use qualify then |
| 07:11.45 | riksta | if that works |
| 07:11.46 | FuriousGeorge | riksta: i do |
| 07:11.51 | FuriousGeorge | it helps |
| 07:12.50 | FuriousGeorge | any of you guys know what a "callno" is |
| 07:13.10 | riksta | the extension number? |
| 07:13.35 | FuriousGeorge | jbot_: no, a callno is part of an undocumented cryptic error |
| 07:13.57 | FuriousGeorge | riksta: no thats not it |
| 07:14.08 | FuriousGeorge | iax_poke: Still have a callno... |
| 07:14.19 | riksta | more information is needed |
| 07:14.54 | X-Gen | use the source Luke (or FuriousGeorge) |
| 07:15.14 | riksta | wasim: care to elabourate on openvpn pls? |
| 07:15.28 | X-Gen | was that source or force, i cant remeber |
| 07:15.36 | FuriousGeorge | X-Gen: the source? what to look up what this error could mean? |
| 07:15.46 | riksta | FuriousGeorge: of course |
| 07:16.14 | FuriousGeorge | if i found it i wouldnt understand |
| 07:16.40 | X-Gen | FuriousGeorge: do a grep through the source for that string in the source and look for some clues |
| 07:16.49 | FuriousGeorge | dont speak C here |
| 07:16.55 | omal | <PROTECTED> |
| 07:17.19 | FuriousGeorge | hmmm |
| 07:17.22 | omal | when in doubt, google the error ;) |
| 07:17.25 | riksta | yeah thats like IAX2/host/CALLNO |
| 07:17.31 | FuriousGeorge | omal: i totally googled |
| 07:17.38 | FuriousGeorge | what terms did you put it |
| 07:17.39 | FuriousGeorge | in |
| 07:17.44 | omal | "asterisk callno" |
| 07:17.48 | FuriousGeorge | i just put "Still have a callno" |
| 07:17.56 | omal | that was the error? |
| 07:17.58 | riksta | duh~ |
| 07:18.03 | FuriousGeorge | omal: yeah |
| 07:18.07 | *** join/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
| 07:18.26 | riksta | you can't just put still have a callno |
| 07:18.28 | FuriousGeorge | http://www.mail-archive.com/asterisk-users@lists.digium.com/msg19716.html |
| 07:18.28 | riksta | that's just dumb |
| 07:18.46 | FuriousGeorge | when did searching google for a string become dumb |
| 07:18.49 | FuriousGeorge | i put it in quotes |
| 07:18.54 | Assid | woopass |
| 07:19.17 | Assid | in 2 hrs.. i start working on creating agents and shit |
| 07:19.31 | X-Gen | Assid: u bragging ? |
| 07:19.38 | Assid | nah |
| 07:19.45 | Assid | i did it once.. dont remember what to do |
| 07:19.56 | X-Gen | it will come to you |
| 07:20.05 | omal | i'm still figuring out WTF i'm doing |
| 07:20.09 | Assid | easiest way to find someone whose done the same thing is to mention what you are doing |
| 07:20.32 | Assid | fastest way instead of asking. 'anyone know how to do this' |
| 07:22.03 | *** join/#asterisk |||sLaSh||| (n=ehje@203.215.100.96) |
| 07:22.12 | |||sLaSh||| | is it ok to upgrade to Cisco SIP 8.2 |
| 07:23.41 | JffMRIII | do you have sip 8.2 that you can share with others looking for it tonight |
| 07:30.59 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 07:31.48 | FuriousGeorge | ok, so i got 4 asterisk peers that are friends. when my ip changed yesterday on one of them it became unreachable and it couldnt reach anyone else. so i had to reload iax2 on all three friends so that they could see the one who's ip changed again. |
| 07:32.03 | FuriousGeorge | i gotta write a script or something to do that automatically |
| 07:32.08 | FuriousGeorge | but |
| 07:32.28 | FuriousGeorge | when i reloaded iax2 on the one who ipchanged all its peers were /still/ unreachable |
| 07:32.41 | FuriousGeorge | so i stopped asterisk and restarted it, and 2 peers came back |
| 07:33.12 | FuriousGeorge | one of them was the provider though |
| 07:33.23 | FuriousGeorge | so i stopped asterisk, waited, and started it again |
| 07:33.39 | *** join/#asterisk Poincare (n=jefffnod@195.207.137.89) |
| 07:33.42 | FuriousGeorge | 1 peer came back, one still unreachable |
| 07:33.51 | FuriousGeorge | my questions is: wtf? |
| 07:34.49 | *** join/#asterisk RES2 (n=res-1@gateway1.nemox.net) |
| 07:34.59 | RES2 | Hi. |
| 07:35.20 | RES2 | I have a big problem. |
| 07:35.24 | JffMRIII | big? |
| 07:35.28 | RES2 | Yes. :-) |
| 07:35.34 | riksta | wow, big. |
| 07:35.39 | JffMRIII | Well how big? |
| 07:35.40 | FuriousGeorge | pour water on it before it spreads |
| 07:35.50 | JffMRIII | windex |
| 07:36.20 | FuriousGeorge | im gonna have to join the mailing list |
| 07:38.05 | RES2 | We have six asterisk-servers. They all are absolutly stable. But one (it is configured very simple) crashes regularly. |
| 07:38.16 | FuriousGeorge | test your memory |
| 07:38.20 | RES2 | The server have one Digium E1-Card. |
| 07:38.59 | JffMRIII | 6 |
| 07:39.04 | JffMRIII | wow that is great |
| 07:39.07 | JffMRIII | all in the same office |
| 07:39.09 | JffMRIII | or location |
| 07:39.17 | RES2 | But the rest of the system woks fine. Only asterisk craches. Do you think, the memory is the reason? |
| 07:39.19 | JffMRIII | what hardware is the one crasking |
| 07:39.26 | JffMRIII | could be |
| 07:39.28 | tehdely | did you enable core dumps |
| 07:39.31 | Assid | RES2: try upgrading too |
| 07:39.37 | JffMRIII | switch mem to the another machine |
| 07:39.43 | JffMRIII | and see if that starts crshing |
| 07:39.56 | tehdely | in the term from which you start asterisk |
| 07:39.58 | tehdely | ulimit -c unlimited |
| 07:39.58 | Assid | i only got 5 boxes having asterisk on them |
| 07:40.00 | tehdely | then pass the -g flag |
| 07:40.02 | tehdely | when you start asterisk |
| 07:40.06 | tehdely | the next time it crashes it will dump core |
| 07:40.12 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 07:40.20 | tehdely | chuck that into gdb; if it's always crashing in the same spot, it's probably no thardware |
| 07:40.27 | Assid | of which 3-4 actually used |
| 07:40.29 | tehdely | if it's somewhere random, start swapping components out until it stops :> |
| 07:40.41 | Assid | 2 are test boxes for a whole lotta shit |
| 07:42.49 | Poincare | if i setup a call to an extension that is busy i want asterisk to setup a call between that and my extension as soon as the other extenstion is free. how is that functionality called? |
| 07:42.57 | RES2 | tehdely: Thank you. So how can i analyse the dump? |
| 07:43.02 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 07:45.04 | tehdely | RES2: gdb -c title-of-core-file `which asterisk` |
| 07:45.13 | tehdely | gdb is a black art |
| 07:45.14 | tehdely | but at the very least |
| 07:45.16 | tehdely | you can type 'bt' |
| 07:45.17 | tehdely | and get a backtrace |
| 07:45.19 | tehdely | and chuck it on the pastebin |
| 07:46.49 | JffMRIII | ok I have skinny.so |
| 07:47.05 | JffMRIII | chan_skinny.so that is |
| 07:47.25 | RES2 | tehdely: Thank's again. So i will try, if I have the first dump. |
| 07:47.29 | Qwell | JffMRIII: good luck |
| 07:47.35 | JffMRIII | lol |
| 07:47.41 | JffMRIII | thank you Qwell |
| 07:47.45 | RES2 | bye and thank's folks. |
| 07:47.46 | *** part/#asterisk RES2 (n=res-1@gateway1.nemox.net) |
| 08:01.57 | *** join/#asterisk fourcheeze (n=rich@westbury.doilywood.org.uk) |
| 08:02.22 | fnordian | good morning |
| 08:03.36 | *** part/#asterisk tehdely (n=delysiid@home.teambarry.org) |
| 08:11.14 | *** join/#asterisk kamuix (n=kamuix@195.78.4.174) |
| 08:13.06 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 08:19.36 | *** join/#asterisk oej_ (n=Olle@apollo.webway.se) |
| 08:19.51 | *** part/#asterisk oej_ (n=Olle@apollo.webway.se) |
| 08:20.54 | *** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt) |
| 08:21.25 | *** join/#asterisk stoffell (n=stoffell@fw.catsanddogs.com) |
| 08:29.17 | *** join/#asterisk apardo (n=apardo@87.218.44.228) |
| 08:30.40 | VeNoMouS_ | hrm is there anyway to hangup a sip call from the console |
| 08:31.00 | VeNoMouS_ | cause i left chanspy() running on speak @ the office by mistake |
| 08:31.00 | VeNoMouS_ | lol |
| 08:31.50 | VeNoMouS_ | n/m |
| 08:31.51 | VeNoMouS_ | <PROTECTED> |
| 08:45.29 | MGSsancho | lol |
| 08:49.03 | *** join/#asterisk chris_ast (n=Administ@59.93.56.163) |
| 08:49.13 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 08:51.52 | chris_ast | Hi people |
| 08:51.53 | stoffell | tzafrir, any idea why the 8-bank gives me crackling noises on analogue phones? |
| 08:53.07 | *** part/#asterisk |||sLaSh||| (n=ehje@203.215.100.96) |
| 08:53.18 | *** join/#asterisk |||sLaSh||| (n=ehje@203.215.100.96) |
| 09:00.34 | Strom_C | hello channel |
| 09:00.53 | backblue | morning all |
| 09:00.54 | harlequin516 | When I originate a call how do I get around to knowing whether or not the Zap channel has answered or not? When I call my cell phone the dialplan is halfway through before I even answer it. |
| 09:01.08 | harlequin516 | Hi backblue |
| 09:01.13 | Strom_C | harlequin516, calling out over an FXO port? |
| 09:01.20 | harlequin516 | Yeah |
| 09:01.32 | Strom_C | There is no way to know unless you order answer supervision from your telephone company |
| 09:01.54 | harlequin516 | Ick |
| 09:02.46 | Strom_C | either call out using voip, or have a system that waits for you to touchtone back at it before starting |
| 09:03.55 | tzafrir | stoffell, what adapter? |
| 09:04.18 | harlequin516 | What's the technical problem with finding out from the signals? |
| 09:04.44 | Strom_C | harlequin516, what do you mean? |
| 09:04.50 | stoffell | tzafrir, the astribank8, usb2 port, analogue phone ( a dect handset) |
| 09:05.01 | kmilitzer | Does anyone have an idea how I can implent an round robin dial command for outgoing calls to two PSTN-Gateways with checking if the destination is still alive and usable? |
| 09:05.08 | harlequin516 | Why shouldn't an FXO be able to figure out when the call is answered? |
| 09:05.14 | *** join/#asterisk festr_ (n=festr@ns.regnet.cz) |
| 09:05.15 | stoffell | tzafrir, is the zttest useful on this? |
| 09:05.25 | *** join/#asterisk af_ (n=af@ip-164-94.sn2.eutelia.it) |
| 09:05.28 | festr_ | hello anyone use jitter for sip with 1.2? |
| 09:05.36 | Strom_C | harlequin516, the FXO card /can/ figure that out - if the phone company is reversing your line polarity when the other party answers. |
| 09:05.45 | backblue | kmilitzer: chanisavaliable() |
| 09:06.01 | tzafrir | stoffell, if you use host synchronization, try to get sync from the device . do you have HOST in 'cat /proc/xpp/sync' ? |
| 09:06.09 | Strom_C | inferred answer supervision is unreliable - if you connect to a recording, for example, it will assume the line has supervised |
| 09:06.20 | harlequin516 | Strom_C: So the phone company isn't doing what it is supposed to do? |
| 09:06.41 | kmilitzer | backblue: Does it work with SIP-Channels? |
| 09:06.43 | Strom_C | harlequin516, the phone company is doing exactly what it's supposed to do - answer supervision is something you have to explicitly order from them |
| 09:07.09 | stoffell | tzafrir, yes, I use host |
| 09:07.30 | harlequin516 | Oh. Is it available for a single pots line? Expensive? |
| 09:07.35 | tzafrir | try: echo 0 0 >/proc/xpp/sync # mind the space after the second 0 |
| 09:07.53 | Strom_C | harlequin516, who is your telephone company? |
| 09:08.00 | harlequin516 | Qwest |
| 09:08.04 | stoffell | tzafrir, okay; doing that and checking the website |
| 09:08.09 | Strom_C | hardwire, which state? |
| 09:08.11 | Strom_C | er |
| 09:08.12 | Strom_C | harlequin516, |
| 09:08.16 | Strom_C | damned autocomplete :) |
| 09:08.20 | harlequin516 | Arizona, phoneix |
| 09:08.22 | tzafrir | that page lacks a "troubleshooting" section, though |
| 09:08.44 | Strom_C | harlequin516, yes, you should be able to order it, although I don't know whether it's tarriffed as an option for commercial service only in AZ |
| 09:08.57 | stoffell | tzafrir, do i need to unload anything? because putting 0 0 leaves me without dialtone |
| 09:09.20 | tzafrir | what do you have on /proc/xpp/xbuses ? |
| 09:09.28 | tzafrir | (should be 1 or two lines) |
| 09:09.54 | harlequin516 | That sucks, I would have thought that kind of basic thing would be a standard feature... |
| 09:10.03 | Strom_C | harlequin516, no, it isnt a standard feature. |
| 09:10.08 | backblue | kmilitzer: voip-info.org check for that func, and your question will be awnser! :D |
| 09:10.18 | Strom_C | there are lots of polarity-sensitive phones and modems out there |
| 09:10.34 | stoffell | tzafrir, ack, 1 line: XBUS-0: CONNECTOR=usb-0000:00:1d.7-7 STATUS=connected bus_type=2 |
| 09:10.41 | Strom_C | sometimes the pulse of the polarity reversal will screw things up...so you have to explicitly order it from the telco |
| 09:10.58 | harlequin516 | I have a cheapy zaptel fxo |
| 09:11.03 | harlequin516 | single |
| 09:11.12 | Strom_C | the clone card? |
| 09:11.16 | harlequin516 | yeah |
| 09:11.19 | Strom_C | yeeech |
| 09:11.30 | harlequin516 | hahah, what? |
| 09:11.37 | harlequin516 | Is there a known problem ? |
| 09:11.39 | Strom_C | horrid little card :) |
| 09:11.45 | Strom_C | no, it's just a piece of junk |
| 09:11.52 | tzafrir | stoffell, so it is basically OK |
| 09:12.05 | Strom_C | I had one and used it about three times. ended up throwing it away |
| 09:12.32 | harlequin516 | Well, It al the very least got me interested in Asterisk.. I can't afford much else yet. |
| 09:12.43 | stoffell | tzafrir, if I use host, dialtone is okay; but sometimes i get crackling noise, is the zttest a good tool also? |
| 09:13.21 | tzafrir | zttest may show a good result and you still get some cracks. |
| 09:13.25 | kmilitzer | backblue: OK, could have looked there directly ... ;) works for SIP, so I'll give it a try ... ;) |
| 09:13.45 | tzafrir | However I wonder why device sync won't work. This is bad |
| 09:13.48 | Strom_C | hahahaha |
| 09:13.52 | Strom_C | that's the greatest quit message ever |
| 09:14.12 | stoffell | tzafrir, okay, i will try on a different server, just a sec :) |
| 09:14.23 | tzafrir | when you set the card to device sync, does the sync led (the second led) still blink regularily? |
| 09:15.33 | harlequin516 | So what's the next best card cheapest buy? |
| 09:16.35 | Strom_C | harlequin516, I've had good experiences with the digium tdm400p |
| 09:16.54 | Strom_C | but these days I keep everything digital - no analog line interfaces at all |
| 09:17.22 | stoffell | tzafrir, i used device sync; and then no dialtone (only crackling), but sync blinks on the device |
| 09:17.40 | harlequin516 | I have to stay zaptel, as I am intending to use the TDD mode for communications. |
| 09:18.59 | Strom_C | harlequin516, are you hearing impaired? |
| 09:19.21 | harlequin516 | Not me personally, but the inteded users of my project are. |
| 09:19.31 | Strom_C | what is your project? |
| 09:20.08 | harlequin516 | I'm trying to build an internet telnet bridge to POTS TTY. |
| 09:20.33 | Strom_C | so people can talk to TTY users through web browsers? |
| 09:20.38 | *** join/#asterisk eset (n=eset@ip545186e3.direct-adsl.nl) |
| 09:20.57 | harlequin516 | Yeah or from Mobile PDA or CEll phones |
| 09:21.10 | Strom_C | harlequin516, you don't need a zap interface for that |
| 09:21.25 | eset | hey, was wondering if anyone had clues why xten would work one day and the next day give 'loggin failed' error (no config has changed) |
| 09:21.39 | harlequin516 | How else can I do it? |
| 09:21.51 | Strom_C | harlequin516, you can do it all via voip |
| 09:22.12 | Strom_C | assuming that software TDD interfaces exist |
| 09:22.14 | harlequin516 | But it has to terminate one end of the connection to a TTY/TDD |
| 09:22.29 | tzafrir | stoffell, take a look at /proc/xpp/sync again: is tick rate steady at 1000 and the 'tick' counter keep increasing? |
| 09:22.40 | *** join/#asterisk nayyares (n=Nayyar@mbl-65-151-218.dsl.net.pk) |
| 09:22.52 | harlequin516 | Zaptel does software TDD, that's why I am trying to use that. |
| 09:22.57 | tzafrir | watch -n1 -d cat /proc/xpp/sync |
| 09:23.06 | Strom_C | harlequin516, point me at the spec sheet |
| 09:23.18 | Strom_C | is it the card itself that does it or is it the zaptel module? |
| 09:23.20 | tzafrir | TDD == ? |
| 09:23.34 | Strom_C | tzafrir, telecommunications device for the deaf |
| 09:23.38 | Strom_C | ~tdd |
| 09:23.45 | harlequin516 | No spec sheet, but all basic functions are claimed to be available. |
| 09:23.59 | Strom_C | harlequin516, please show me where it claims that |
| 09:24.14 | harlequin516 | http://www.voip-info.org/wiki/view/tdd+mode |
| 09:24.36 | tzafrir | Strom_C, so it's basically only the signally and no PCM? |
| 09:24.45 | Strom_C | tzafrir, what? |
| 09:25.15 | Strom_C | harlequin516, ah ok, so it's TDD mode from within AGI scripts |
| 09:25.26 | harlequin516 | Right |
| 09:25.36 | harlequin516 | sendtext recvtext |
| 09:26.04 | harlequin516 | I'm guessing its buffered for American 45.5 baud |
| 09:26.37 | tzafrir | stoffell, what kernel version BTW? |
| 09:26.54 | Strom_C | harlequin516, I'm looking to see if you can create pseudo zap interfaces |
| 09:26.59 | harlequin516 | Though Cisco docs say that TDD signals can pass through VOIP G711?? or some codec, decently with proper setup. |
| 09:27.10 | Strom_C | G711 only |
| 09:27.19 | Strom_C | I wouldnt trust data over anything but G711 |
| 09:28.42 | stoffell | tzafrir, debian sarge; but with 2.6.15 kernel |
| 09:28.43 | harlequin516 | What's a pseudo channel? |
| 09:28.52 | Strom_C | harlequin516, exactly what it sounds like |
| 09:28.57 | harlequin516 | Haha |
| 09:28.58 | Strom_C | you know what pseudo means, right? |
| 09:29.09 | harlequin516 | Yeah, but I'm really new to asterisk.. |
| 09:29.25 | harlequin516 | I only recently figured out the diff between a channela nd an extension |
| 09:29.36 | Strom_C | so? that doesnt mean you need to stop thinking |
| 09:29.49 | *** join/#asterisk kippi (n=chris@untrust-gct.equinoxit.net) |
| 09:29.51 | kippi | hey |
| 09:29.54 | Strom_C | hi |
| 09:30.06 | kippi | are there rmps for asterisk on redhat |
| 09:30.14 | harlequin516 | I'm sure |
| 09:30.25 | Strom_C | kippi, who cares? download the source from digium and compile it yourself |
| 09:30.26 | *** join/#asterisk exten123 (n=exten@60.49.6.190) |
| 09:30.32 | stoffell | tzafrir, oke; i'm gettin further. plugged in on other server, much better now |
| 09:30.33 | wasim | about friggin' time they bowled them out ... |
| 09:30.37 | harlequin516 | Yeah its surprisingly easy |
| 09:30.54 | tzafrir | stoffell, what's the difference between the two servers? |
| 09:30.55 | *** join/#asterisk tehdely (n=delysiid@home.teambarry.org) |
| 09:31.23 | tzafrir | kippi, there are some RPMs at freshrpms, IIRC |
| 09:31.30 | harlequin516 | I haven't quite figured out what a local channel can do. |
| 09:31.54 | tzafrir | harlequin516, highly useful for originating calls |
| 09:31.57 | harlequin516 | I read it, but telephony jargon, is only starting to make sense for me |
| 09:32.10 | kippi | Strom_C: can't get seem to download the source for redhat so the complie fails |
| 09:32.28 | tzafrir | kippi, what redhat? |
| 09:32.41 | Strom_C | harlequin516, from the wiki: "chan_local is a pseudo-channel. Use of this channel simply loops calls back into the dialplan in a different context. Useful for recursive routing; it is able to return to the dialplan after call completion. " |
| 09:32.47 | kippi | tzafrir: yeah |
| 09:32.57 | kippi | tzafrir: redhat fc4 |
| 09:33.07 | backblue | redhat != fc4 |
| 09:33.16 | tzafrir | fc4 is not exactly "RedHat [tm]", tou know... |
| 09:33.38 | harlequin516 | Yeha I read teh wiki page, but I didn't understand the examples, though it refernced some hardware that probably made it obvious to people who were familiar. |
| 09:34.21 | Strom_C | harlequin516, essentially, you place a call to another part of the dialplan |
| 09:34.26 | Strom_C | it sets up a virtual call segment |
| 09:34.48 | Strom_C | it basically says "go do this, but come back here when that finishes" |
| 09:36.22 | harlequin516 | Yeah that makes sense, but I didn't get how you could use it to originate a call. |
| 09:36.47 | Strom_C | well, it's very simple |
| 09:37.07 | Strom_C | when asterisk is originating a call, it sets up two calls |
| 09:37.17 | harlequin516 | Cause you always have to specify Channel Context and extension right? |
| 09:37.20 | Strom_C | hang on |
| 09:37.21 | harlequin516 | Yeah 2 calls |
| 09:37.32 | tehdely | so chan_local is a goto that returns |
| 09:37.33 | tehdely | neat |
| 09:37.46 | Strom_C | one call goes out to the far end and the other call goes to something within your dialplan |
| 09:37.51 | Strom_C | asterisk bridges the two together |
| 09:37.56 | stoffell | tzafrir, difference: the one is a celeron, the other Xeon. But itlooks like the 'sequence' of loading is important |
| 09:38.24 | harlequin516 | Yeah, I'm learning a lot today. |
| 09:38.32 | Skid | if i call multiple sip usrs, and my mobile phone at the same time from an extension - and hang up on my mobile, how can i make asterisk not realise this (aka i press the busy button on mobile) and still ring |
| 09:39.41 | Strom_C | harlequin516, telephony is very simple. it's just the setting up of voice paths between two points. |
| 09:39.50 | *** join/#asterisk MichaelPHines (n=MichaelP@hh-1-109.flexabit.net) |
| 09:39.50 | harlequin516 | Where I am confused about originating with chan local, it would be setting up two internal connections right? So one dialplan communicating with another. |
| 09:40.07 | Strom_C | harlequin516, why the hell would you do that? |
| 09:40.09 | *** join/#asterisk Aurs (n=aurs@a217-118-40-143.bluecom.no) |
| 09:41.00 | harlequin516 | That's a good question, but originally I thought I could control the entire call through agi, but apparently this doesn't seem possible to me. |
| 09:41.17 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 09:41.20 | MichaelPHines | hey, I'm looking to set up a simple VXML server with regular old telephone lines, is asterisk a good tool for handling the SIP? |
| 09:41.25 | Strom_C | harlequin516, perhaps if you tell me what you're trying to do, I can help you. |
| 09:41.30 | Strom_C | MichaelPHines, yes. |
| 09:41.42 | harlequin516 | So I have implemented using stadard Zap channel dialout from Java-asterisk manager |
| 09:42.01 | MichaelPHines | does it understand VXML or do I need something else to work together with it |
| 09:42.16 | Strom_C | harlequin516, no no, don't get into implementation specifics - tell me what your end goal is. |
| 09:42.41 | Strom_C | MichaelPHines, i've personally never worked with asterisk and vxml at the same time |
| 09:42.41 | x86 | what are some cool things i can do to give more features to my dialplan? |
| 09:42.49 | x86 | as far as star codes, etc? |
| 09:42.56 | Strom_C | x86: www.nanpa.com |
| 09:43.01 | Strom_C | those are called "vertical service codes" |
| 09:43.01 | x86 | also, how do i setup 3-way calling? |
| 09:43.03 | Strom_C | look em up |
| 09:43.17 | harlequin516 | The dialplan will connect the ZAP cahnnel (PoTS TDD) to my java agi which will pipe the data to/fro a socket to internet my end user |
| 09:43.47 | MichaelPHines | Strom_C: what have you used for VXML? just services like TellMe or anything hardware based? |
| 09:43.51 | harlequin516 | Oh End goal is simple Telnet to a Pots TDD |
| 09:44.17 | Strom_C | harlequin516, why TELNET?! No one in their right mind uses Telnet anymore |
| 09:45.07 | MichaelPHines | harlequin516: lol, its all about the ssh now |
| 09:45.12 | harlequin516 | Well, telnet is the abstarction I am using to describe the datapath which could be more advaced (Java applet, IM client, Cell phone telnet app .....) |
| 09:45.21 | Strom_C | MichaelPHines, why do you want to use vxml? |
| 09:45.41 | Strom_C | harlequin516, just a random question: is English not your first language? |
| 09:45.53 | nettie | Hi guys, anyone know why moh seems to work (debugwise) but I cant hear any sound please? Thanx in adv. |
| 09:45.57 | harlequin516 | Actually in the Deaf world telnet is what connected a lot of folks to relay service up until recently. |
| 09:46.11 | tzafrir | stoffell, what do you mean by "sequence"? |
| 09:46.34 | harlequin516 | Strom_C : English is practally my firest language. |
| 09:46.43 | Strom_C | harlequin516, "practically"? |
| 09:46.45 | harlequin516 | I'm just sloppy on IRC |
| 09:46.56 | MichaelPHines | Strom_C: its a project im doing at my univerisity, we want to be able to have people call in and interact with the building, (eg. turn lights on and off, adjust temperature in certain rooms, and change video wall information) |
| 09:46.57 | harlequin516 | I was born in Illinois |
| 09:46.59 | Strom_C | harlequin516, your sentence constructions are absolutely bizarre |
| 09:47.07 | stoffell | tzafrir, it looks like i must put the sync to device before i do ztcfg and load asterisk |
| 09:47.12 | harlequin516 | Realy? |
| 09:47.14 | Strom_C | MichaelPHines, you don't need vxml for that :) |
| 09:47.36 | harlequin516 | I took latin for 7 years. I think that might have a large influnce on my writing styles. |
| 09:47.41 | *** join/#asterisk Modcuts (n=bob@proporta.gotadsl.co.uk) |
| 09:47.46 | MichaelPHines | Strom_C: whats the alternative, C? |
| 09:47.53 | tzafrir | stoffell, that seems strange. We have no problem changing sync source at e.g. in the middle of a call |
| 09:48.09 | Strom_C | MichaelPHines, perl, python, c, whatever you want to use - they can all interface with asterisk through AGI |
| 09:48.10 | harlequin516 | Ceasarian Prose, its longwidned like a chapter of one paragraph, with never ending suffixed clauses. |
| 09:48.37 | tzafrir | is there any other zaptel card on that system? |
| 09:48.38 | Strom_C | harlequin516, actually it's the failure to use the proper parts of speech at certain times that I'm noticing :) |
| 09:48.54 | Strom_C | words being dropped... |
| 09:48.55 | Strom_C | etc |
| 09:49.06 | harlequin516 | hmm, I'll have to review... Oh |
| 09:49.28 | harlequin516 | Weird, cause normally I'm the grammar nazi. I went to a Jesuit High School. |
| 09:49.49 | MichaelPHines | Strom_C: so when you say any language, can I do it in...C#? |
| 09:49.59 | Strom_C | perhaps you just have extremely awkward phrasing :) |
| 09:50.02 | stoffell | tzafrir, indeed; changing sync while calilng goes okay |
| 09:50.10 | harlequin516 | Yes I am extremely awkward. |
| 09:50.11 | Strom_C | MichaelPHines, Assuming there's a C# AGI interface... |
| 09:50.11 | harlequin516 | Hahah |
| 09:50.38 | MichaelPHines | ie. are there APIs for these languages? |
| 09:51.06 | Strom_C | MichaelPHines, I don't know off the top of my head. |
| 09:51.22 | Strom_C | you'll have to see whether there's a C# AGI interface |
| 09:51.30 | harlequin516 | Did you find out about pseudo Zap channels? |
| 09:51.47 | harlequin516 | I was looking but couldnt see anything relevant... |
| 09:51.55 | *** join/#asterisk nayyares (n=Nayyar@mbl-65-151-218.dsl.net.pk) |
| 09:52.08 | MichaelPHines | Strom_C: this AGI looks like just what I was looking for, thank god I wont have to use VXML |
| 09:52.12 | Strom_C | harlequin516, I still have absolutely no idea why it makes a difference or what you're trying to do. You still haven't really effectively communicated to me what it is that you're trying to do |
| 09:52.13 | *** join/#asterisk Hermis (n=guitarug@85.21.204.146) |
| 09:52.16 | MichaelPHines | what an abomination of the standard |
| 09:52.43 | MichaelPHines | whoever decided that a scripting language should be written in XML was out of their mind |
| 09:52.46 | tzafrir | stoffell, so you say that you have no problem if the sync source was HOST at ztcfg time? |
| 09:52.48 | Hermis | Hello everybody |
| 09:53.12 | x86 | how do i setup three-way calling? |
| 09:53.18 | Strom_C | x86, on a zaptel channel? |
| 09:53.22 | x86 | SIP |
| 09:53.28 | tzafrir | Meetme |
| 09:53.36 | Hermis | Help me please with configuring zap CO interface... |
| 09:53.36 | harlequin516 | Well simply communicate text between a TTY/TDD and an IP socket. |
| 09:53.38 | Strom_C | your SIP device will have to do the three-way and bridge itself |
| 09:53.38 | stoffell | tzafrir, indeed |
| 09:53.52 | Strom_C | Hermis, just ask your question |
| 09:54.07 | x86 | tzafrir: that will allow Phone A to call Phone B, put Phone B on hold, dial Phone C, then connect all three call legs? |
| 09:55.00 | Strom_C | x86, meetme for a three-way is a really bad way of doing it |
| 09:55.00 | Hermis | What loadzone must I configure for Russian PSTN to correctly identify ringtone,busytone etc. |
| 09:55.00 | x86 | Strom_C: what are my options? |
| 09:55.00 | Strom_C | x86, your SIP device should be capable of threeway itself |
| 09:55.00 | x86 | Strom_C: what if it's not? |
| 09:55.00 | tzafrir | strange. Our current init scripts tend to change the sync source to the device just before running ztcfg |
| 09:55.04 | Skid | none of my hardware is Strom_C .. |
| 09:55.05 | Strom_C | x86, then it blows donkeys for quarters |
| 09:55.08 | Skid | and its all cisco :) |
| 09:55.13 | Strom_C | Skid, what?? |
| 09:55.22 | Strom_C | what hardware? |
| 09:55.23 | Skid | capable of doing 3 way |
| 09:55.26 | Skid | 7940/60's |
| 09:55.27 | x86 | Strom_C: aka, i want to terminate both the Phone B and Phone C legs at the asterisk box, and only have one leg going to Phone A (it's a really dumb phone) |
| 09:55.30 | RoyK | http://centos.org/modules/news/article.php?storyid=127 |
| 09:55.41 | Strom_C | Skid, my 7960 is capable of three-way; I don't know what you're smoking |
| 09:55.50 | Skid | mmmmmm ? |
| 09:56.00 | Strom_C | Skid, step 1: dial call |
| 09:56.14 | Strom_C | step 2: press "CONFRN" button |
| 09:56.19 | Strom_C | step 3: dial second call |
| 09:56.24 | Strom_C | step 4: press "join" |
| 09:56.26 | Strom_C | SIMPLE |
| 09:56.29 | Strom_C | EASY |
| 09:56.33 | Strom_C | A BABY COULD DO THIS |
| 09:56.43 | Strom_C | (albeit with a lot more drooling) |
| 09:57.08 | Hermis | and what about zap? |
| 09:57.16 | Skid | ho hum |
| 09:57.18 | Skid | i do apologise |
| 09:57.20 | Strom_C | Hermis, im looking |
| 09:57.27 | MichaelPHines | Strom_C: can I do voice recognition with the AGI? |
| 09:57.30 | Hermis | I must configure it, but i't not working correctly |
| 09:57.44 | tzafrir | RoyK, also search in http://lwn.net/Articles/177085/ for "has happened before" and a bit below |
| 09:57.46 | Strom_C | MichaelPHines, not until someone gets Sphinx working with asterisk :) |
| 09:57.53 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 09:58.09 | MichaelPHines | Strom_C: see thats really why i needed vxml support |
| 09:58.17 | Assid | hrmm.. i have this 'gateway' box which is designed to just take calls and forward them to my other boxes |
| 09:58.39 | Strom_C | Hermis, I can't find a setting explicitly for Russia; does your tone plan mimic another country's tone plan? |
| 09:58.41 | Assid | but.. when im trying to forward the calls i get 'Mar 27 15:26:45 WARNING[2254]: chan_iax2.c:6985 socket_read: Call rejected by internalip: No such context/extension' |
| 09:59.14 | Assid | but i have the number setup on this box as well |
| 09:59.38 | Hermis | Strom_C I don't know this |
| 09:59.50 | Assid | Rejected connect attempt from gateway.ip, request 's@wtn-inbound' does not exist |
| 09:59.52 | Strom_C | Hermis, well, now you know what you need to find out |
| 09:59.56 | exten123 | may I know when I need enable NAT in sip? |
| 10:00.09 | Strom_C | exten123, when your SIP client is behind NAT |
| 10:00.11 | *** join/#asterisk Nix (n=Nix@81.214.255.57) |
| 10:00.32 | Assid | anyone know what could be the issue here? |
| 10:00.47 | exten123 | Strom_C, it's mean that only when is out of enternet only need NAT? |
| 10:00.51 | MichaelPHines | Strom_C: this seems to imply that Sphinx already works with asterisk: http://www.voip-info.org/wiki-Sphinx |
| 10:00.51 | Hermis | Strom_C Thanks, may i reconpile Zaptel with changet zonedata.c with my country default tones, will it work correctly |
| 10:01.12 | Strom_C | exten123, I cannot understand your question |
| 10:01.19 | Strom_C | MichaelPHines, well good - last I heard it didnt work |
| 10:01.24 | Strom_C | Hermis, sure, whatever |
| 10:01.54 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 10:01.58 | Hermis | Strom_C Thanks |
| 10:02.19 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 10:04.27 | Hermis | My FireFly client establish connection but i can' hear anything. What's the problem? RTP and Codecs work correctly. |
| 10:05.04 | x86 | gah |
| 10:05.14 | x86 | having a hard time getting call pickup to work |
| 10:05.26 | x86 | directed call pickup, not group call pickup... |
| 10:06.03 | MichaelPHines | this looks like a pain to set up. Thank god I'm project leader, I think i'll just offload the rest of this to my hardware guy |
| 10:07.33 | x86 | i put pickupexten => *8 in the [featuremap] context in features.conf and reloaded asterisk, but when i dial *8100 from my BT101 (extension 103) to pickup a ringing call on my X-Lite (extension 100), i get a 404 error |
| 10:08.03 | x86 | what am i doing wrong? |
| 10:08.09 | x86 | http://www.voip-info.org/wiki/view/Asterisk+config+features.conf |
| 10:08.13 | x86 | this is what i'm going off of |
| 10:08.24 | x86 | do i need to throw anything into my extensions.conf to recognize that? |
| 10:09.15 | MichaelPHines | x86: = *8 instead of => *8 maybe? |
| 10:10.16 | x86 | nope |
| 10:10.21 | x86 | still get a 404 error |
| 10:10.48 | Assid | <PROTECTED> |
| 10:11.07 | wasim | and ? |
| 10:11.23 | MichaelPHines | x86: 404 is HTTP not found? |
| 10:11.25 | Assid | i have iax->gateway(iax) -> another asterisk server on iax |
| 10:12.43 | x86 | MichaelPHines: also SIP not found ;) |
| 10:13.05 | x86 | MichaelPHines: as the SIP protocol shares a few of the HTTP return codes hehe |
| 10:13.22 | MichaelPHines | haha, im new at all this |
| 10:13.47 | MichaelPHines | i wonder if the univeristy will let me just use some of their lines from the T1 |
| 10:15.43 | *** join/#asterisk Winkie (n=urmom@cpc2-stre1-6-0-cust119.bagu.cable.ntl.com) |
| 10:17.53 | backblue | anyone with hylafax integration? |
| 10:19.33 | konfuzed | ok a small but important piece of the picture has left my head space.. what sort of gear do I use to hook up 40 rj11 outlets to an asterisk box as zapata ports ? |
| 10:20.06 | wasim | either a channel bank or digium/sangoma fxs pci cards |
| 10:22.20 | *** join/#asterisk GolobTGG (n=GolobTGG@193.2.154.246) |
| 10:25.17 | *** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt) |
| 10:26.10 | wasim | err, sorry fxo cards ... you can also use some fxo ata's |
| 10:30.58 | *** join/#asterisk shiznatix (n=shiznati@213-35-237-100-dsl.end.estpak.ee) |
| 10:36.07 | fnordian | can anyone tell me the email-address of olle/oej? |
| 10:36.18 | MikeJ__ | nope! |
| 10:36.20 | MikeJ__ | :P |
| 10:36.33 | MikeJ__ | I'm sure it's all over the mailing lists |
| 10:37.05 | fnordian | ah |
| 10:37.11 | fnordian | it's in the svn-logs also |
| 10:39.56 | MikeJ__ | that;s not right |
| 10:44.04 | fnordian | you're right, it differs |
| 10:45.03 | fnordian | eah |
| 10:45.16 | fnordian | he's using different ones |
| 10:50.49 | *** join/#asterisk gfox (n=vince@calvix/staff/gfox) |
| 10:56.00 | *** join/#asterisk chris_ast (n=Administ@59.93.56.163) |
| 10:57.11 | festr_ | anyone use jitter for sip with 1.2? |
| 10:57.17 | festr_ | in production env? |
| 10:59.44 | *** join/#asterisk kainam (n=Jake@202.137.160.110) |
| 11:04.46 | backblue | festr_: why do you need it? |
| 11:05.01 | festr_ | backblue: cause of jitter network |
| 11:05.18 | Strom_M | festr_, you should solve your network's jitter problems :) |
| 11:05.56 | RoyK | I have tried it in production, but got an enormous amount of complaints from customer |
| 11:05.56 | festr_ | Strom_M: unsolvable |
| 11:06.04 | RoyK | Strom_C: you can't do that |
| 11:06.04 | backblue | festr_: why? |
| 11:06.05 | Strom_M | festr_, why? |
| 11:06.14 | festr_ | we are ISP |
| 11:06.21 | RoyK | customers on DSL etc |
| 11:06.23 | festr_ | and have many wireless links |
| 11:06.35 | festr_ | shaper problems etc.. |
| 11:06.51 | festr_ | RoyK: which version have you tried? |
| 11:07.02 | festr_ | RoyK: and what problems did you expirience? |
| 11:07.11 | Strom_M | festr_, what kind of latency can you expect between your sip terminals and your pbx? |
| 11:07.30 | festr_ | Strom_M: 50~300 |
| 11:07.32 | chris_ast | festr_,RoyK,backblue: Can you please help me on musiconhold? |
| 11:07.48 | Strom_M | ouch....300ms |
| 11:07.57 | festr_ | Strom_M: yes it is high but reality |
| 11:08.03 | kmilitzer | I am still looking for a way to spread calls in a round robin fashin to two SIP-PSTN-Gateways and check if they are reachable. If one is not, only the other should be used. ChanIsAvail as backblue suggested is not a soultion ... |
| 11:08.06 | Strom_M | that's almost entirely unsuitable for voip |
| 11:08.28 | x86 | i have ~260ms clients that work fine ;) |
| 11:08.29 | RoyK | festr_: i've tried them all. i've paid zoa a whole lot to write this, and the current code is unusable |
| 11:08.46 | wasim | kmilitzer: +1 prio should handle that |
| 11:09.16 | festr_ | RoyK: thank you for this info you safe me a lot of work |
| 11:09.38 | kmilitzer | wasim: How would that work? |
| 11:10.06 | wasim | kmilitzer: 1,Dial(SIP/blah)\n2,Dial(SIP/bleh) or +101 depends |
| 11:10.48 | festr_ | RoyK: anyway, what kind of problems is with this jitter? |
| 11:11.27 | kmilitzer | wasim: Really bad idea. It is IMHO possible that the call works for the first dial command and is then build up a second time ... |
| 11:12.36 | kmilitzer | wasim: And furthermore this way I will not find out if the destination is reachable. Asterisk fires and forgetts SIP Invites ... (that's the way it should be, I know) |
| 11:12.54 | backblue | festr_: cable isp? |
| 11:13.06 | backblue | chris_ast: not now, sorry |
| 11:13.17 | festr_ | backblue: cable, wireless etc. |
| 11:18.01 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 11:19.58 | *** join/#asterisk cybergypsy (n=cybergyp@APoitiers-156-1-64-186.w86-217.abo.wanadoo.fr) |
| 11:20.18 | cybergypsy | anyone using asterisk on ubuntu in here ? |
| 11:20.20 | backblue | festr_: i hate cable isp's! :D |
| 11:20.32 | festr_ | backblue: which country? |
| 11:20.33 | festr_ | :) |
| 11:20.36 | backblue | portugal |
| 11:20.48 | cybergypsy | is it best to install from the repo`s or from source ? |
| 11:22.35 | x86 | a repo is source ;) |
| 11:22.59 | x86 | unless you're talking about a package repository, like a RPM or DEB repository for example |
| 11:24.00 | *** join/#asterisk tuxinator_linux (n=tuxinato@netblock-68-183-112-100.dslextreme.com) |
| 11:24.08 | cybergypsy | yea - i am using ubuntu and its in the repositories |
| 11:24.25 | cybergypsy | just not sure which would be best to use |
| 11:24.37 | cybergypsy | i`m an asterisk noob |
| 11:24.50 | Strom_M | which version is in the repositories? |
| 11:25.29 | cybergypsy | 1:1.0.9.dfsg-1 |
| 11:25.50 | x86 | holy damn that's old ;) |
| 11:25.56 | x86 | you'll want source for sure |
| 11:25.57 | x86 | lol |
| 11:26.06 | cybergypsy | great! |
| 11:26.10 | cybergypsy | now the fun starts |
| 11:26.18 | cybergypsy | thanks x86 |
| 11:26.35 | *** join/#asterisk Winkie (n=urmom@cpc2-stre1-6-0-cust119.bagu.cable.ntl.com) |
| 11:27.20 | x86 | get 1.2.5 |
| 11:27.23 | x86 | it's stable |
| 11:27.57 | cybergypsy | ok |
| 11:29.36 | *** join/#asterisk paanz (n=Paanz@60.51.180.130) |
| 11:32.32 | *** join/#asterisk [swb] (n=swb@cornelyn.force9.co.uk) |
| 11:37.31 | *** join/#asterisk pedros09 (n=pedros09@p50865850.dip.t-dialin.net) |
| 11:38.09 | pedros09 | anyone successfully compiled iaxClient samples on windows? |
| 11:38.22 | pedros09 | I am having some compatibility issues: |
| 11:38.53 | pedros09 | ../../lib/libiaxclient.a(rpe.o):/cygdrive/d/Downloads/Tools/Telephony/VOIP/Asterisk/developer/iaxclient_SVN/trunk/iaxclient/lib/gsm/src/rpe.c:405: more undefined references to `__a |
| 11:38.53 | pedros09 | ssert' follow |
| 11:38.53 | pedros09 | collect2: ld returned 1 exit status |
| 11:38.53 | pedros09 | make: *** [iax2slin.exe] Error 1 |
| 11:38.59 | *** join/#asterisk elfo (n=elfo@tremplin.epfl.ch) |
| 11:43.55 | *** part/#asterisk cybergypsy (n=cybergyp@APoitiers-156-1-64-186.w86-217.abo.wanadoo.fr) |
| 11:47.47 | x86 | is it possible to store a user's area code in a mysql database, and retrieve it prior to a Dial() within a dialplan? |
| 11:48.11 | x86 | (trying to make it so my users can 7-digit dial within their area code) |
| 11:50.52 | *** join/#asterisk zotz (n=zotz@24.231.32.85) |
| 11:53.57 | Aurs | x86: short answer: yes ;) |
| 11:55.05 | x86 | Aurs: cant find it in the wiki :( |
| 11:56.36 | Aurs | x86: create a table with 2 cols. first column has all the area codes, 2nd has the extensions |
| 11:56.48 | Aurs | then do a realtime() on that table |
| 11:57.40 | Aurs | realtime(familynamefromextconfig|col2|extension) - gives you the area code as $col1 |
| 11:58.27 | Aurs | or it would make more sense to have the extensions as col1 |
| 11:59.57 | kippi | when I have been doing make install on asterisk I am getting this error |
| 11:59.58 | kippi | /usr/bin/ld: cannot find -lssl |
| 11:59.58 | kippi | collect2: ld returned 1 exit status |
| 11:59.58 | kippi | make: *** [asterisk] Error 1 |
| 12:00.41 | x86 | Aurs: hmm, i found info on the MYSQL() command |
| 12:02.31 | Aurs | x86: never used that one, but RealTime will do the trick in this case, me thinks |
| 12:04.08 | Aurs | kippi: check that you've got all the needed dependencies. looks like you haven't got openssl installed? |
| 12:05.12 | Aurs | kippi: wild guess: install openssl and associated -devel package(s) |
| 12:08.57 | *** join/#asterisk umay (n=chris@70-101-61-50.dsl2-plymouth.roc.ny.frontiernet.net) |
| 12:08.59 | *** join/#asterisk apardo (n=apardo@62-14-56-136.inversas.jazztel.es) |
| 12:09.11 | x86 | hmm |
| 12:09.31 | x86 | i'm trying to setup 10 digit dialing without the need for an access code... |
| 12:09.55 | x86 | how is that possible? i tried exten => _XXXXXXXXXX,1,Dial(IAX2/trunk/${EXTEN}) |
| 12:10.02 | x86 | but i get a 404 code when i try it |
| 12:12.22 | wasim | how can you get a 404 on IAX? |
| 12:13.15 | Aurs | missing context? |
| 12:18.06 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 12:18.13 | PakiPenguin | afternoon |
| 12:19.31 | *** join/#asterisk lunaphyte_ (n=lunaphye@70.90.148.1) |
| 12:24.32 | *** join/#asterisk Eitch (n=hugo@unaffiliated/eitch) |
| 12:30.08 | x86 | PakiPenguin: !!!! |
| 12:30.19 | x86 | PakiPenguin: what the hell is going on with your ToDial? :) |
| 12:30.47 | *** join/#asterisk TUplink (n=sdfgkjm@68-232-82-147.chvlva.adelphia.net) |
| 12:30.57 | TUplink | loader.c:325 __load_resource: /usr/local/lib/asterisk/modules/chan_zap.so: Undefined symbol "pri_restart" whats wrong??? |
| 12:31.03 | PakiPenguin | x86 :) |
| 12:31.16 | x86 | PakiPenguin: is ToDial down? |
| 12:31.23 | x86 | PakiPenguin: i havent been able to use it for a week :( |
| 12:31.34 | x86 | PakiPenguin: also, we need to talk in private about some things |
| 12:31.40 | PakiPenguin | x86, pvt please :) |
| 12:31.42 | PakiPenguin | sure |
| 12:33.00 | TUplink | any one know anything bout my prob? |
| 12:44.28 | *** join/#asterisk _Paulo_ (n=Paulo@200-168-112-132.dsl.telesp.net.br) |
| 12:53.01 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 12:54.29 | *** join/#asterisk Ad-Hoc (n=Nimbus@ppp71-adsl-177.ath.forthnet.gr) |
| 12:55.15 | *** join/#asterisk sergeus (n=s@195.112.98.13) |
| 12:57.40 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 12:58.54 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 12:59.04 | *** join/#asterisk Lino` (i=Lino@i577BC5C5.versanet.de) |
| 13:00.46 | *** join/#asterisk Skarmeth (n=Skarmeth@201008193102.user.veloxzone.com.br) |
| 13:01.12 | Skarmeth | hi all |
| 13:01.20 | _Paulo_ | hi |
| 13:02.38 | *** join/#asterisk coppice (n=chatzill@45.193.17.210.dyn.pacific.net.hk) |
| 13:03.16 | Skarmeth | _Paulo_, are you from L5 networks? |
| 13:03.36 | _Paulo_ | Skarmeth, no. |
| 13:03.44 | Skarmeth | does anyone here uses HP servers for asterisk projects? Something like a T110P and TDM04B with small number of transcoding ( < 10 channels )? Dell mid-machines for rack environment it's a pain |
| 13:03.46 | _Paulo_ | Skarmeth, Im from Braslink |
| 13:03.52 | Skarmeth | nice |
| 13:05.03 | *** join/#asterisk brockj49464 (n=brockj49@56.108.dhcp.hope.edu) |
| 13:05.52 | *** join/#asterisk Malthus (n=admin@uslec-66-255-41-2.cust.uslec.net) |
| 13:06.39 | Malthus | Strom_C hi, I figured out the problem with my T1 cards not doing zttest |
| 13:06.57 | *** join/#asterisk pigpen2 (n=mark@207.71.48.222) |
| 13:07.07 | _Paulo_ | Skarmeth, I use home brew beige box only. Supermicro sometimes. |
| 13:07.10 | Malthus | it doesn't enable the timer if no spans are configured |
| 13:08.45 | *** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
| 13:08.52 | Malthus | hmm, guess he not home |
| 13:10.38 | *** part/#asterisk chris_ast (n=Administ@59.93.56.163) |
| 13:21.55 | *** join/#asterisk vaw (n=vaw@195.Red-83-60-228.dynamicIP.rima-tde.net) |
| 13:21.57 | vaw | hello |
| 13:22.42 | vaw | I'm using supplied h323 channel to receive calls |
| 13:22.43 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
| 13:22.56 | dpryo | lol |
| 13:23.15 | vaw | and I wanted to switch context using the ip where the call is originated |
| 13:23.46 | vaw | so I've made a user in h323.conf telling the ip address in "host=" parameter |
| 13:24.01 | vaw | and "context=" to the one I want |
| 13:24.12 | vaw | but it allways goes to default context |
| 13:24.19 | *** join/#asterisk epablo (n=epablo@200.84.7.239) |
| 13:24.54 | vaw | can't I switch contexts using origing IP? |
| 13:29.07 | *** topic/#asterisk by russellb -> Asterisk: The Open Source PBX -=- Asterisk 1.2.6 and Zaptel 1.2.5 Released! (March 27, 2006) -=- http://www.asterisk.org/ -=- AMP/FreePBX users should join #freepbx for support |
| 13:31.49 | epablo | Hi people |
| 13:32.00 | PakiPenguin | hi epablo |
| 13:32.11 | epablo | Hows it going? |
| 13:32.31 | PakiPenguin | am alright how about you? |
| 13:32.51 | *** join/#asterisk viperdude (n=jon@borat.enta.net) |
| 13:34.20 | epablo | All good |
| 13:34.42 | sivana | russellb: are you going to copy the change log? |
| 13:35.30 | *** join/#asterisk Utah_Dave (n=boucha@0-1pool149-100.nas31.salt-lake-city1.ut.us.da.qwest.net) |
| 13:37.52 | tdonahue | hi all, we are having some DTMF problems with our Polycoms after updating to the latest 1.6 firmware. does anyone know of any changes that have been made to the phones that might affect DTMF? |
| 13:38.02 | russellb | sivana: the changelog should be there |
| 13:38.23 | sivana | russellb: nope |
| 13:38.28 | russellb | lies, i see it! |
| 13:38.29 | sivana | /pub/telephony/asterisk/ChangeLog-1.2.5 was not found |
| 13:38.31 | sivana | hehe |
| 13:38.38 | russellb | that's the old one, goofball |
| 13:38.45 | russellb | http://ftp.digium.com/pub/asterisk/ChangeLog-1.2.6 |
| 13:38.57 | sivana | oh.. I'm looking on the main page of asterisk.org |
| 13:39.06 | russellb | yeah, announcement hasn't been made yet |
| 13:39.14 | sivana | ah.. so you broke that page! :) |
| 13:39.21 | russellb | ok ok, i'll fix that link :) |
| 13:39.24 | sivana | hehe |
| 13:39.49 | russellb | ok, it's fixed, hehe |
| 13:39.54 | sivana | :) |
| 13:42.29 | epablo | i'm setting up an SER with asterisk.. The first does the load balancing for my * servers using a redirect. But the IP I'm getting is SERs and not the clients. Question: from where does chan_SIP take the peers info, sip packets or IP packets? |
| 13:44.10 | backblue | epablo: why do you need to use SER? |
| 13:45.33 | tamp4x | epablo are you doing a forward()? |
| 13:45.38 | epablo | backblue:I have to many SIP clientes. The server can't handle the load. So I need to put in another server.. But i don't want to segregate users.. |
| 13:45.45 | GolobTGG | hello all, does anyone know of a GUI for managing asterisk using asterisk realtime (so that all the data is stored in mysql only)? |
| 13:45.49 | tamp4x | you have to send sl reply 300 or 203 dont remmebr of "REDIRECT" |
| 13:46.27 | tamp4x | 203 i mean 302 |
| 13:47.02 | epablo | tamp4x: Let me check what I'm using now.. have done tests with both.. ;) |
| 13:48.42 | epablo | tamp4x: I'm doing this:: append_branch("sip:test1.cv.com:5060"); sl_send_reply("300", "Redirect"); |
| 13:50.07 | *** join/#asterisk atta (n=ansgar@p54B6E2D2.dip.t-dialin.net) |
| 13:50.26 | *** part/#asterisk atta (n=ansgar@p54B6E2D2.dip.t-dialin.net) |
| 13:50.30 | *** join/#asterisk corruptor (n=andrew55@www.tae.ru) |
| 13:50.54 | sivana | epablo: what's your number of SIP clients, if you don't mind. I'm curious on what your load is |
| 13:51.21 | epablo | Over 300 using g729 |
| 13:51.25 | sivana | ok |
| 13:51.48 | tamp4x | the append branch is not neccessary |
| 13:53.19 | vaw | I'm using supplied h323 channel to receive calls and I wanted to switch context using the ip where the call is originated. I've made a user in h323.conf telling the ip address in "host=" parameter and "context=" to the one I want but it allways goes to default context. Can't I switch contexts using origin IP? |
| 13:54.00 | *** join/#asterisk apardo (n=apardo@62-14-56-136.inversas.jazztel.es) |
| 13:55.13 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 13:56.58 | *** join/#asterisk octothorpe (n=octothor@unaffiliated/octothorpe) |
| 13:57.44 | backblue | epablo: you can do it only with asterisk servers. |
| 13:57.59 | *** join/#asterisk TinoW (n=tino@living-examples.com) |
| 13:58.03 | TinoW | hell-o |
| 13:58.05 | backblue | i think we should improve SIP channel quality in asterisk, to stop using ser. |
| 13:58.26 | epablo | backblue: ??? |
| 13:58.53 | backblue | epablo: i understand why you use SER. |
| 13:59.06 | tamp4x | dont listen to backblue epablo |
| 13:59.18 | epablo | LOL |
| 13:59.19 | backblue | tamp4x: why not? |
| 13:59.38 | backblue | please, tell me |
| 14:00.22 | russellb | backblue: they serve different purposes. |
| 14:00.41 | epablo | You said I don't need to do the append branch. How should I pass the new SIP servers IP's with the redirect? |
| 14:01.14 | epablo | russellb: I agree but the SER model is a pain in the ass. |
| 14:01.33 | viperdude | hi guys i have a TE110P card and as soon as i load wcte11xp then i lose audio on my sip channels, any ideas? |
| 14:02.41 | tamp4x | use rewriteuri |
| 14:03.25 | vaw | can someone help me with h323? |
| 14:03.29 | epablo | So rewriteuri and the redirect.? How will my packets end up. |
| 14:03.46 | TinoW | q: does Dial() return to the dialplan when the connection is picked up? |
| 14:04.00 | epablo | vaw: i use oh323.. have never used the supplied h323 channel |
| 14:04.17 | vaw | epablo, ok, thanks |
| 14:04.56 | x86 | i'm trying to setup 10-digit dialing without the need for a trunk prefix... |
| 14:05.12 | x86 | so a SIP user can just dial 10 digits and not have to request an outside trunk... |
| 14:05.35 | *** join/#asterisk x3me (n=x3me@201.11.226.147) |
| 14:05.47 | *** part/#asterisk x3me (n=x3me@201.11.226.147) |
| 14:06.15 | x86 | i'm trying to use exten => XXXXXXXXXX,1,Dial(IAX2/trunkname/${EXTEN},100,tr), but when i dial a 10 digit number, i get a "404" error |
| 14:06.18 | [TK]D-Fender | x86 : Plenty easy... |
| 14:06.36 | [TK]D-Fender | x86 : You forgot the "_" preceeding your X's .... |
| 14:06.42 | [TK]D-Fender | x86 : its a PATTERN. |
| 14:07.53 | tamp4x | how will your packets en dup? what does that mean |
| 14:07.54 | tamp4x | just do it |
| 14:10.15 | x86 | [TK]D-Fender: even with a _ in front of the 10 X's it doesnt work |
| 14:10.22 | backblue | russellb: yes they do, but as i said, why not improve asterisk to do the job? that's the way to go. |
| 14:10.43 | x86 | [TK]D-Fender: if i put _N in front of the 10 X's, i can get it to show me a 484 message instead of a 404, but still no dice... |
| 14:11.01 | backblue | epablo: we are trying to implement sip redirects on asterisk. |
| 14:11.02 | russellb | backblue: no, Asterisk will not and never will be a SIP proxy |
| 14:11.34 | russellb | it's not a matter of improving :) |
| 14:11.45 | backblue | russellb: we dont need sip proxy, mostly asterisk cant handle sip redirects as SER does, so we can just redirect the sip trafic. |
| 14:15.44 | *** join/#asterisk cuco (n=diego@local.xorcom.com) |
| 14:18.07 | [TK]D-Fender | x86 : Pastebin your dialplan |
| 14:18.16 | x86 | i got it :) |
| 14:18.21 | x86 | PakiPenguin helped me ;) |
| 14:18.25 | [TK]D-Fender | I didn't say add an "N", just the "_" |
| 14:18.31 | PakiPenguin | haha :) |
| 14:18.37 | epablo | I think both aproches are correct.. It would be nice for * to handle redirects.. |
| 14:20.29 | Darwin_35 | NxxNxxxxxxx |
| 14:20.38 | kmilitzer | Any idea why hangupcauses don't get translated correctly into SIP Causes? |
| 14:20.41 | Katty | anyone ever wake up one morning with their ear having a faint ring to it? |
| 14:21.08 | Hmm-work | the signals all are flashing red, it doesn't matter what was said, this bit is much to big without me and you |
| 14:21.15 | cuco | Katty, lol |
| 14:21.24 | Darwin_35 | its called tenitus |
| 14:21.34 | Katty | Darwin_35: and the cause? |
| 14:21.44 | Katty | Darwin_35: more importantly, is it dangerous? |
| 14:21.48 | Darwin_35 | nerve damage |
| 14:21.48 | *** join/#asterisk Bambr (n=Bambr@213-35-236-110-dsl.end.estpak.ee) |
| 14:22.12 | Darwin_35 | mmost people have slight damage and get ringing once in awhile |
| 14:22.24 | Katty | it's been ringing for 30 minutes now. |
| 14:22.28 | Katty | granted, it's quite faint... |
| 14:22.35 | Darwin_35 | but if it hhruts and is loud you need to go get it checked |
| 14:22.36 | TinoW | does the dialplan continue after Dial() when the call succeeds? |
| 14:22.43 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 14:22.44 | Assid | heya tkd! |
| 14:22.46 | Assid | whats happenin |
| 14:22.54 | Darwin_35 | hurts |
| 14:23.00 | Darwin_35 | fat finger |
| 14:23.05 | [TK]D-Fender | klsjadhgflaksjdhflkjashdfpiouewryt; ewrt dfg fgds sdafkgj hlsdkjfgh |
| 14:23.19 | Katty | Darwin_35: it does not hurt.. |
| 14:23.23 | Katty | Darwin_35: nor is it loud... |
| 14:23.26 | kmilitzer | Katty: If it's still rining tonite (+6 hours) see a doctor, otherwise it's possible that you'll never get rid of it ... |
| 14:23.46 | Darwin_35 | still should go to a audioligist and have your hearing checked |
| 14:23.49 | Katty | kmilitzer: i have the option of seeing an ENT at 2:30 this afternoon... |
| 14:24.07 | TinoW | ah I guess nobody knows ( |
| 14:24.10 | kmilitzer | Katty: What time is it now at your location? |
| 14:24.27 | Katty | kmilitzer: i noticed it at 7:45am this morning, it's now 8:15am |
| 14:24.47 | Katty | or 8:30, something like that |
| 14:25.21 | *** join/#asterisk b0xii (n=b0xii@cpe-70-116-68-157.houston.res.rr.com) |
| 14:26.05 | Darwin_35 | Katty I would call your DR and let him know and see what he says |
| 14:26.15 | *** join/#asterisk brockj49464 (n=brockj49@56.108.dhcp.hope.edu) |
| 14:26.43 | *** part/#asterisk cuco (n=diego@local.xorcom.com) |
| 14:26.52 | Darwin_35 | but I would still make apt with a audioligest and make a apt also |
| 14:26.59 | Darwin_35 | shower time brb |
| 14:27.13 | epablo | tamp4x: Do you have an example on how to do that redirect? do you mind sharing it? |
| 14:27.56 | kmilitzer | A tinitus can happen from time to time ... it's no cause to worry if it fades after a while (like after a few) hours ... the reason is, that the blood circulation in your ear is not working right. Could happen from too much stress or loud noises (music, eg.) |
| 14:28.01 | loko | hey Darwin_35 |
| 14:28.16 | Assid | damn |
| 14:28.30 | Assid | stupid sipdiscount dogs banned my asterisk server |
| 14:28.56 | backblue | epablo: using asterisk with ser, how do you handle billing? |
| 14:29.04 | kmilitzer | But I still would like to know if anyone else has problems with the translation of causecodes into SIP reasons |
| 14:29.23 | Assid | err.. what port should i telnet to? |
| 14:29.29 | Katty | Darwin_35: i did...but it would cost me the same to go to the clinic than it would to simply see an ENT |
| 14:29.42 | Katty | Darwin_35: at the clinic they just give you the next available doctor. |
| 14:29.47 | Assid | for sip |
| 14:29.58 | TinoW | hello? |
| 14:30.36 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 14:31.32 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 14:31.41 | backblue | russellb: why this http://bugs.digium.com/view.php?id=6721, its not applyed in asterisk-1.2.6? just asking! not big deal! |
| 14:32.03 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 14:32.27 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 14:32.44 | *** join/#asterisk guyboertje (n=guy@213-131-125-116.onyx.net) |
| 14:32.45 | *** join/#asterisk coppice (n=chatzill@174.198.17.210.dyn.pacific.net.hk) |
| 14:32.49 | *** part/#asterisk brockj49464 (n=brockj49@56.108.dhcp.hope.edu) |
| 14:33.01 | TinoW | again... can I use some commands after Dial() to run while the connection is up? |
| 14:33.20 | [TK]D-Fender | TinoW : not really. |
| 14:33.50 | TinoW | [TK]D-Fender: whats the DIALSTATUS "ANSWER" good for then? |
| 14:34.30 | *** join/#asterisk malverian (n=malveria@adsl-065-005-207-210.sip.gnv.bellsouth.net) |
| 14:34.51 | epablo | backblue: Asterisk does it.. or at least that is the idea. |
| 14:35.16 | backblue | epablo: so what you handle in SER? |
| 14:35.22 | epablo | backblue: I wan't to use SER only as an load balancer.. |
| 14:35.25 | backblue | just register trafic? |
| 14:35.41 | backblue | epablo: i'm asking because maybe, i will have to do it this week |
| 14:35.52 | [TK]D-Fender | TinoW : So you know what happened to the dial attempt AFTER the call ... |
| 14:35.58 | backblue | we are bringing the cluster up now, but we will try to do it all without ser. |
| 14:36.25 | [TK]D-Fender | TinoW : So you cantinue on in context to log the call perhaps, rename the monitor file, etc, and other "stuff" |
| 14:36.27 | backblue | but if it does not work well, i will have to use SER. |
| 14:36.31 | epablo | backblue: He should redirect on-net traffice to the correct asterisk box and off-net is handled by asterisk.. At least thats the idea |
| 14:36.45 | TinoW | [TK]D-Fender: what can I do to do something when the connection starts? |
| 14:36.49 | backblue | epablo: yes, i understand. |
| 14:37.22 | epablo | backblue: We could work as a team.. I'm doing the same over here.. |
| 14:37.24 | backblue | epablo: but does ser have somehow to know, when one asterisk server have many calls and forward to other? (i dont know ser) |
| 14:37.42 | x86 | how can i provide E911 services to my customers? |
| 14:37.49 | [TK]D-Fender | TinoW : You'd have to run some sort of manager script to monito for the channel starting.... not easy for sure. |
| 14:38.08 | x86 | i cant find info about it anywhere, except the government site saying it's mandatory, but they fail to mention how to go about it |
| 14:38.16 | Hmm-work | there is no good load balancing module for SER right now |
| 14:38.40 | TinoW | [TK]D-Fender: too bad. :( what can I do beside hacking the source? |
| 14:38.49 | backblue | how better its load balancing with ser? its the same as round robin dns? |
| 14:39.30 | *** join/#asterisk cytrak (n=kvirc@adelphi.geofocus.com) |
| 14:40.03 | cytrak | does anyone here know what causes this problem? chan_zap.c:8203 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 |
| 14:40.36 | cytrak | my audio quality gets really bad when I'm talking to someone and sundenly I get a bunch of those errors |
| 14:41.10 | cytrak | I also get this one: pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span |
| 14:42.54 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 14:42.55 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:43.43 | *** join/#asterisk guyboertje (n=guy@213-131-125-116.onyx.net) |
| 14:45.16 | [TK]D-Fender | TinoW : What are you trying to do? |
| 14:45.42 | [TK]D-Fender | cytrak : Sounds like a clocking problem. |
| 14:45.59 | TinoW | [TK]D-Fender: I want to playback a greating when the callee takes the call |
| 14:46.19 | *** join/#asterisk groogs (n=greg@d221-73-237.commercial.cgocable.net) |
| 14:46.24 | *** join/#asterisk powerchip (n=powerchi@197.80-202-229.nextgentel.com) |
| 14:47.05 | powerchip | how can i know what agent so call the server ? |
| 14:47.25 | cytrak | [TK]D-Fender: clocking ? in the /etc/zapta.conf file ? |
| 14:48.18 | [TK]D-Fender | TinoW : Theres an option code for that in the dial command! |
| 14:48.32 | cytrak | I just found an article on asteriskgurus that also says it could be IDE interrupts problem or that Then probably the PRI you are using is not using PRI signalling but maybe some other type of signalling like E&M. |
| 14:48.47 | viperdude | anyone know what this means? "ZT_CHANCONFIG failed on channel 26: No such device or address (6)" |
| 14:48.51 | cytrak | I'm pretty sure I'm using PRI signalling though |
| 14:49.04 | TinoW | [TK]D-Fender: M(x) ? |
| 14:49.06 | *** join/#asterisk chapeaurouge (n=chap@vilhost1.vision.lu) |
| 14:50.23 | powerchip | I must know the varable to know what aget so call? any now? |
| 14:50.37 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 14:50.37 | powerchip | if agent call to the server |
| 14:51.06 | Darwin_35 | agents.conf |
| 14:51.22 | Darwin_35 | look at it |
| 14:51.22 | [TK]D-Fender | TinoW : Go read and test... |
| 14:51.42 | TinoW | [TK]D-Fender: yes, I was reading but aparently overlooked M(x) |
| 14:52.20 | cytrak | do I need to remove mods for my digium card to get the timesync reloaded ? or ztcfg -vvv should do it ? |
| 14:52.44 | x86 | how can i provide E911 services to my customers? |
| 14:52.55 | x86 | can i do it myself or do i need a provider? |
| 14:53.17 | x86 | i run an ITSP and all my lines are over IAX2 and SIP, I have no connected POTS lines |
| 14:54.43 | epablo | tamp4x: are you still around' |
| 14:54.48 | tamp4x | ? |
| 14:54.49 | cytrak | [TK]D-Fender: don't think its timingsync |
| 14:55.08 | epablo | have a couple minutes to help finnish that asterisk / SER integration. I must be missing something' |
| 14:55.12 | powerchip | Darwin_35: i know the filne but i will make so if ($agent == "2"){ // not call } else { //call } |
| 14:55.54 | tamp4x | just go to iptel.org and get the sip intro manual |
| 14:56.12 | tamp4x | go to the section using ser as a redirect server |
| 14:56.21 | epablo | tamp4x: Ok.. Thanks.. |
| 14:58.31 | cytrak | I think the siemens guys that take care of my PBX box suck |
| 14:58.43 | cytrak | sorry that was just a thought |
| 15:00.05 | TinoW | [TK]D-Fender: funny, aparently it does something - but the caller does not hear a sound... |
| 15:00.16 | *** join/#asterisk umay (n=chris@70-101-61-50.dsl2-plymouth.roc.ny.frontiernet.net) |
| 15:00.28 | *** part/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 15:00.58 | Hmm-work | nothing is more annoying than getting told the exact same thing twice |
| 15:01.14 | Hmm-work | "blah blah blah blah" |
| 15:01.51 | Strom_M | Hmm-work, also, nothing is more annoying than getting told the exact same thing twice |
| 15:03.25 | Nugget | heh |
| 15:04.27 | *** join/#asterisk kpettit (n=keith@69.15.174.114) |
| 15:04.43 | Hmm-work | funny guy |
| 15:04.47 | kpettit | anybody know if there is a way to turn call-fowarding off on a polycom phone remotly? |
| 15:05.40 | [TK]D-Fender | TinoW : A(x): Play an announcement (x.gsm) to the called party. |
| 15:06.03 | TinoW | [TK]D-Fender: thats the wrong direction. I want to play the announcement to the calling party |
| 15:06.05 | [TK]D-Fender | kpettit : Force a reboot. |
| 15:06.36 | *** join/#asterisk octothorpe (n=octothor@unaffiliated/octothorpe) |
| 15:06.39 | kpettit | cool. I thought it stored that in a file. It just dose it in memory? |
| 15:06.53 | TinoW | [TK]D-Fender: and aparently the Playback() in the M() macro also plays to the callee and not to the caller. Bummer :( |
| 15:07.06 | cytrak | is this normal on zttool ? IRQ misses 38387 |
| 15:08.19 | *** join/#asterisk jaike (n=a@203.131.137.76) |
| 15:09.22 | [TK]D-Fender | TinoW : READ ABOVE |
| 15:09.36 | TinoW | [TK]D-Fender: pardon? |
| 15:09.38 | jaike | anyone know the link to the 1.2.6 changelog? |
| 15:09.43 | [TK]D-Fender | [10:05] <[TK]D-Fender> TinoW : A(x): Play an announcement (x.gsm) to the called party. |
| 15:09.47 | jaike | website still shows 1.2.5 |
| 15:10.01 | TinoW | [TK]D-Fender: yes please. I already told you I want to play to the _calling_ party |
| 15:10.20 | *** join/#asterisk brc_ (n=brian@pdpc/supporter/basic/brc) |
| 15:10.24 | brettnem | M^^ |
| 15:10.34 | jaike | oh..found it |
| 15:11.09 | x86 | how can i provide E911 services to my VoIP network? |
| 15:11.24 | brettnem | x86: partner with a provider that properly supports E911 |
| 15:11.27 | x86 | are there companies that will let me setup an IAX2 trunk to them? |
| 15:11.31 | brettnem | or go with someone like Dash911 |
| 15:11.32 | x86 | brettnem: do you know any? |
| 15:11.36 | brettnem | well, me. :) |
| 15:11.43 | brettnem | where are you located? |
| 15:11.43 | x86 | dash911 is expensive and uses dedicated circuits |
| 15:11.50 | brettnem | yes, it is very expensive.. |
| 15:12.04 | x86 | my company is just starting out and cant afford dedicated circuits for E911 |
| 15:12.18 | brettnem | you shouldnt need it unless you are a CLEC |
| 15:12.22 | brettnem | where are you located? |
| 15:12.28 | x86 | i'd rather maintain the customer's address and pass it to a PSAP, or pass it to a provider that routes to the proper PSAP |
| 15:12.35 | x86 | illinois, usa |
| 15:12.41 | brettnem | hmm.. can't help you there |
| 15:12.51 | brettnem | x86: what you want is frequently called PS/ALI |
| 15:13.01 | x86 | oh yeah? |
| 15:13.03 | brettnem | in the traditional 911 world |
| 15:13.10 | x86 | whats it mean? |
| 15:13.15 | brettnem | yeah, and many CLECs will sell it to you |
| 15:13.26 | brettnem | Private Switch/Automatic Location Identification |
| 15:13.52 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 15:14.06 | brettnem | the concept is that you send 911 calls to a provider that takes the NENA (911 records) from you and passes it onto the 911 database for you |
| 15:14.08 | x86 | so i can maintain a list of addresses, and when one of my extensions dials 911 it will pass it to the proper PSAP? |
| 15:14.15 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 15:14.17 | x86 | ah ok |
| 15:14.24 | x86 | you know of any providers? |
| 15:14.46 | brettnem | no exactaly, you pass the call to the provider, the provider passes it to the incumbant and their route selector passes it to the appropriate PSAP.. but you get the basic idea. |
| 15:14.50 | *** join/#asterisk fu3 (n=kaa@234-200-29-134.hcc.mnscu.edu) |
| 15:14.51 | brettnem | s/no/not |
| 15:14.55 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 15:15.03 | x86 | ah ok |
| 15:15.04 | brettnem | not in your area |
| 15:15.10 | x86 | where are you? |
| 15:15.14 | brettnem | I'm in texas |
| 15:15.16 | brettnem | yehaw |
| 15:15.22 | x86 | are you an ITSP? |
| 15:15.22 | TinoW | [TK]D-Fender: I guess you are out of ideas? Any variable I could use to redirect the playback to the calling channel? |
| 15:15.35 | brettnem | I'm a CLEC. we have an ITSP company too. |
| 15:16.00 | x86 | brettnem: do i need a PS/ALI provider for every PSAP area? or why does it matter where my PS/ALI provider is located? |
| 15:16.04 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 15:16.07 | brettnem | but I do a lot of CLEC functions.. like interconnection, 911, SS7 translations.. |
| 15:16.35 | brettnem | x86: you must have 911 access for each area you have.. ie: you can't use a ps/ali provider in one area for customers in another. |
| 15:16.44 | x86 | oh man... |
| 15:16.44 | |Vulture| | anyone here using a Dell Power Edge 850 for *? |
| 15:16.50 | x86 | my customers are all over the nation ;) |
| 15:16.52 | brettnem | unless of course the ps/ali provider is in both areas. |
| 15:17.05 | brettnem | x86: that's why people typically partner with companies like dash911 |
| 15:17.10 | x86 | there is no one stone to kill all birds? :P |
| 15:17.18 | brettnem | Intrado may have an offering now to people like you, you should definately call them. |
| 15:17.33 | x86 | brettnem: yeah, but dash911 only does dedicated TDM circuits :( |
| 15:17.39 | brettnem | x86: btw, I think sellvoip.net has a portal to do this as well, but they are rediculously hard to get a hold of |
| 15:17.40 | x86 | url? |
| 15:17.50 | brettnem | dash911 doesn't only do dedicated TDM circuits.. that's nonsense. |
| 15:18.05 | x86 | http://www.dash911.com/how_it_works.htm |
| 15:18.10 | x86 | according to this, that's all they do |
| 15:18.15 | brettnem | let me see.. |
| 15:18.27 | x86 | "Dash911 uses dedicated TDM/PSTN circuits to carry 9-1-1 call traffic" |
| 15:18.30 | x86 | direct quote |
| 15:18.35 | brettnem | look, there are two important parts to 911.. 1) ALI database 2) 911 interconnection. |
| 15:18.46 | brettnem | x86: duh, but not tdm circuits to you |
| 15:18.47 | brettnem | ;) |
| 15:18.53 | x86 | ah :) |
| 15:19.13 | *** join/#asterisk SwK[Work] (n=SwK@64.89.118.139) |
| 15:19.20 | brettnem | I'm pretty sure you can send them a 911 call SIP.. virtually positive.. |
| 15:19.20 | x86 | just $5000 up front and like ~$2/mo per customer then? |
| 15:19.28 | nettie | Hi guys, I'm having problem bridging calls. From phone 'a' I call a pSTN number trought my voip carrier, when the user picks up I then try to transfer the call to another phone in my lan and on asterisk console I see messages flood all saying "Attempting native bridge of local calls<->voip call". The local phone rings but when my collegue picks up it's mute. Any idea please? |
| 15:19.43 | brettnem | yeah, it's a total ripoff |
| 15:19.57 | Hmm-work | heart breaker |
| 15:19.59 | Hmm-work | heart breakerrrrr |
| 15:20.02 | brettnem | x86: check out the drawing on that webpage.. |
| 15:20.31 | brettnem | x86: try to flag down sellvoip.net |
| 15:22.03 | brettnem | wow it's awfully quiet in here this morning.. everyone still asleep?? |
| 15:22.08 | tzanger | yep |
| 15:22.27 | |Vulture| | I wish |
| 15:22.46 | TinoW | *snore* |
| 15:22.48 | brettnem | kids got me up at 6:30 this morning.. ugh |
| 15:24.42 | *** join/#asterisk file (n=file@mctnnbsa24w-142167058031.pppoe-dynamic.nb.aliant.net) |
| 15:24.54 | *** join/#asterisk brockj49464 (n=brockj49@56.108.dhcp.hope.edu) |
| 15:25.06 | *** join/#asterisk oej (n=oej@gateway.digium.com) |
| 15:25.22 | *** part/#asterisk brockj49464 (n=brockj49@56.108.dhcp.hope.edu) |
| 15:25.44 | RoyK | anyone using SIPCHANINFO? |
| 15:27.08 | Hmm-work | i need another website besides fark to waste time at work |
| 15:29.23 | ManxPower | Hmm-work, I suggest http://www.bellsouth.com/tariffs/?sbs_dd=tarrifs for light reading |
| 15:29.37 | Hmm-work | lol |
| 15:30.11 | Hmm-work | I could use a good php photo album |
| 15:31.21 | TinoW | [TK]D-Fender: I see in the log something like Playback("SIP/foobar-3de8","Soundfile") while this is the called party. Can I replace this somehow with the calling party? |
| 15:31.37 | [TK]D-Fender | TinoW : Not sure. |
| 15:31.52 | *** part/#asterisk epablo (n=epablo@200.84.7.239) |
| 15:31.58 | TinoW | [TK]D-Fender: I was expecting I can play to both directions of a connection? |
| 15:32.29 | Hmm-work | can anyone recommend one? |
| 15:32.55 | ManxPower | TinoW, See "show application dial" |
| 15:33.00 | Nugget | I use gallery (1, not 2) but I don't think that extends as far as an endoresement. |
| 15:33.08 | Nugget | I'm really unhappy with the direction that gallery is heading |
| 15:33.57 | TinoW | ManxPower: yes, it shows me the manual (which I also read online ;) But still it does not help much I guess... |
| 15:34.07 | *** join/#asterisk azzie (n=az@azzie.net) |
| 15:34.53 | ManxPower | TinoW, So the "A" option to Dial was not what you were looking for? |
| 15:35.08 | TinoW | ManxPower: no :) |
| 15:35.28 | ManxPower | <PROTECTED> |
| 15:35.46 | Strom_M | wasting all my time time |
| 15:36.00 | Hmm-work | yeah I was looking at gallery |
| 15:36.02 | TinoW | ManxPower: yes, I read that about 1000 times. But I want to play to the _calling_ party. Not to the called! :) |
| 15:36.34 | ManxPower | TinoW, um, for the CALLING party you can use playback |
| 15:36.42 | TinoW | ManxPower: ah! |
| 15:37.08 | ManxPower | TinoW [TK]D-Fender: I see in the log something like Playback("SIP/foobar-3de8","Soundfile") while this is the called party. Can I replace this somehow with the calling party? |
| 15:37.22 | ManxPower | ah, I see where the confusion on my part is. |
| 15:37.39 | TinoW | ManxPower: ah well, yes I tried Playback as well. But it does not what I want :( |
| 15:37.49 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
| 15:38.40 | nettie | Hi Manx, how's going? Just wondeirng if you had experience any problems with call bridging? When I transfer a call to another phone withing my LAN I see a flood of attempts on asterisk console, the phones destination phone rings but the call gets lost. Any idea please? |
| 15:39.11 | ManxPower | nettie, Only if you have NAT involved. |
| 15:39.12 | tzanger | how does the new DB function work? Set(DB(key)=value) ? |
| 15:39.22 | nettie | ManxPower unfortunately I do |
| 15:39.35 | ManxPower | nettie, then things get VERY complicated. |
| 15:39.43 | *** join/#asterisk The_X (i=chris@true.fiberpimp.net) |
| 15:40.15 | nettie | ManxPower the server is in my colo and the phones are on a local netowrk behind nat. I would prefer not ending configurig a vpn if possible.. |
| 15:40.18 | The_X | I call from a phone to a direct line (Did) behind asterisk and even if I flush the call from the cell, it keeps ringing on the 7960 for a good 5 seconds |
| 15:40.22 | The_X | is there a way to fix it |
| 15:40.29 | fnordian | oej: hi |
| 15:40.54 | ManxPower | nettie, SIP/RTP puts the address inside the data part of the packet. NAT only works on the headers, which would have the public address. |
| 15:40.56 | [TK]D-Fender | tzanger : Thats right |
| 15:41.07 | tzanger | [TK]D-Fender: danke. I haven't had to use it much :-) |
| 15:41.10 | tzanger | ... at all |
| 15:41.12 | ManxPower | nettie, do the phones work if you are not trying to do a transfer? |
| 15:41.13 | tzanger | ... ever. :-) |
| 15:41.22 | ManxPower | 'morning tzanger |
| 15:41.55 | tzanger | morning ManxPower, how are things? |
| 15:42.14 | ManxPower | tzanger, they could be better, they could be worse. |
| 15:42.26 | tzanger | hmm. sounds... mediocre |
| 15:43.25 | nettie | ManxPower they do |
| 15:43.48 | nettie | I had issues before when calling within my LAN |
| 15:44.12 | ManxPower | nettie, and you have nat=yes in sip.conf for each of the devices? |
| 15:44.12 | nettie | but after I disabled SIP ALG on the router it worked perfectly |
| 15:44.18 | nettie | ManxPower yes |
| 15:44.21 | nettie | I do |
| 15:44.57 | ManxPower | nettie, then you either need to disable reinvites or put in VPN tunnel. |
| 15:44.59 | nettie | ManxPower I also have qualify=yes |
| 15:45.57 | nettie | I think I'll disable reinvite .. if I do it local calls will be routed trough asterisk I suppose |
| 15:46.02 | nettie | more than routed |
| 15:46.04 | nettie | proxied |
| 15:46.07 | ManxPower | nettie, correct. |
| 15:46.16 | nettie | ManxPower let's try |
| 15:46.26 | ManxPower | nettie, you don't have to do an encrypted tunnel. |
| 15:46.32 | *** join/#asterisk bweschke (n=bweschke@c-68-36-51-213.hsd1.nj.comcast.net) |
| 15:46.35 | ManxPower | A simple GRE tunnel would work just fine for this. |
| 15:47.07 | nettie | yeah I know the problem is that the router at the warehouse is a zyxel |
| 15:47.17 | nettie | not sure it support unenc |
| 15:47.31 | nettie | I normally use cisco's and kame/ipsec-tools serverside |
| 15:47.37 | ManxPower | you might start by getting a real router. 8-| |
| 15:47.41 | nettie | eheeh |
| 15:47.42 | nettie | :) |
| 15:47.45 | nettie | yeah I know .. |
| 15:48.05 | nettie | ehehe |
| 15:48.12 | nettie | I have a 1840 at home |
| 15:48.19 | nettie | and here a zyxel |
| 15:48.22 | nettie | I shoul dbe killed:) |
| 15:48.23 | nettie | ehe |
| 15:48.26 | nettie | anyway |
| 15:48.32 | nettie | I'll definitely upgrade it |
| 15:48.32 | ManxPower | I use 1750 for hime and my clients usually use 2600s |
| 15:48.34 | x86 | i have a pix 501 at home, asterisk behind it ;) |
| 15:49.04 | nettie | ManxPower on all my phones I have careinvite=no |
| 15:49.06 | x86 | i'm pondering getting an 871 for home :) :) :) |
| 15:49.18 | GolobTGG | do you use ciscos for PSTN connectivity too? |
| 15:49.32 | nettie | is careinvite the same as reinvite |
| 15:49.34 | ManxPower | I should have a Cat 5505 by the middle of this coming month 8-) |
| 15:49.35 | nettie | uhmm |
| 15:49.42 | ManxPower | GolobTGG, not at all. |
| 15:49.42 | nettie | ManxPower you're loaded :) |
| 15:49.43 | *** join/#asterisk veto (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com) |
| 15:49.46 | ManxPower | We use routers to route. |
| 15:49.51 | ManxPower | we use servers to serv. |
| 15:49.53 | x86 | i've got a cat 2912 at home right now ;) |
| 15:49.55 | nettie | ManxPower I did a better job on the switch though |
| 15:50.01 | x86 | i love it, even tho it's EOS / EOL |
| 15:50.03 | ManxPower | nettie, Cat 550xs are CHEAP on ebay |
| 15:50.06 | nettie | ManxPower I got a procurve 2650 I think |
| 15:50.16 | GolobTGG | cisco's big with all the "integrated services" on their routers |
| 15:50.25 | nettie | ManxPower around 600ish new |
| 15:50.29 | ManxPower | I'm wireing up a campground and we need something decent. |
| 15:50.35 | nettie | ManxPower limited layer3 capabilities.. |
| 15:50.39 | ManxPower | and cheap. |
| 15:50.47 | nettie | ManxPower it's impressive how cool it is.. |
| 15:51.03 | GolobTGG | procurve switchs are great |
| 15:51.17 | *** join/#asterisk saftsack (n=saftsack@p54A7D3CA.dip.t-dialin.net) |
| 15:51.24 | GolobTGG | I have a couple of 2650s and a 3400 |
| 15:51.37 | nettie | yeah |
| 15:51.40 | ManxPower | my largest client has something like 18 offices. We finally got Cisco routers at most of the offices after 5 years of working on it, now comes standardizing the switches at these offices. |
| 15:51.45 | nettie | impressive for the amount of money |
| 15:51.50 | Hmm-work | well yappa-ng was insanely easy to install |
| 15:51.51 | saftsack | hi |
| 15:52.04 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6) |
| 15:52.08 | GolobTGG | 1/2 the price of a similar cisco (2960 or such) |
| 15:52.11 | GolobTGG | here |
| 15:52.30 | saftsack | if someone dials my number and if the channel is busy it does do congestion, but the dialer doesnt hear anything :( |
| 15:52.52 | ManxPower | Customer: "But we can get SMC switches at 1/2 the cost!" Me: "And you have found a consultant that knows how to manage them, right?" |
| 15:53.02 | ManxPower | they usually then shut up and order the cisco. |
| 15:53.38 | tzanger | :-) |
| 15:54.38 | RoyK | ManxPower: use D-Link! |
| 15:54.40 | kend | Hey -- I'm using "viewfax" to view fax TIF files, and it comes out fine -- if I don't use viewfax, it looks all squished. Okay. However, is there an equivalent to "viewfax" on Windows, where the faxes *won't* look squished? |
| 15:55.35 | RoyK | ManxPower: works wonderfully as long as you only need to utilise them 10% and not use anything fancy like multicasting or something |
| 15:55.35 | *** join/#asterisk freezer (i=freeze@66.29.46.127) |
| 15:55.39 | freezer | hi guys |
| 15:56.17 | kend | RoyK: what do you mean by the "10%" bit? Bandwidth? PoE? |
| 15:56.48 | *** join/#asterisk Dandre (n=Dandre@was59-3-82-236-48-30.fbx.proxad.net) |
| 15:56.52 | RoyK | I just meant that cpu-wise, it's crap |
| 15:56.54 | Dandre | Hello all, |
| 15:56.56 | RoyK | and stability-wise |
| 15:56.59 | RoyK | and so on |
| 15:57.15 | RoyK | but it works well as long as you don't expect too much |
| 15:57.16 | nettie | ManxPower didnt work :( |
| 15:57.28 | nettie | I got a nice Incoming call: Got SIP response 500 "Internal Server Error" back from 172.31.253.51 |
| 15:57.36 | ManxPower | canreinvite=no in each section of sip.conf? |
| 15:57.45 | nettie | now I dont get the attempts flood.. |
| 15:57.49 | ManxPower | nettie, you are using polycoms |
| 15:57.54 | nettie | yeah |
| 15:57.55 | nettie | you got me |
| 15:57.56 | nettie | eheheh |
| 15:57.59 | nettie | damn |
| 15:58.02 | Dandre | I have a little problem when I do consultative transfer with sip. The callerid is set to the transferer and not the transferee. I there anything I could do? |
| 15:58.19 | nettie | one is polycom the other is a pap2 |
| 15:58.33 | nettie | which is there just for testing |
| 15:58.48 | [TK]D-Fender | Dandre : Thats what a cosultative transfer is SUPPOSED to do. |
| 15:58.48 | jaike | nettie: might be a firewall problem...noticed polycoms sending packets to other ports aside from 5060 |
| 15:59.32 | nettie | well on the asterisk server only port 5060 is open |
| 15:59.55 | Dandre | Yes but the phone whose the call is transfered to hasn't the real callerid |
| 15:59.59 | Katty | anyone use wget for ftp? |
| 16:00.08 | *** join/#asterisk inv_arp[work] (i=junya@adsl-10-132-83.mia.bellsouth.net) |
| 16:00.08 | nettie | ManxPower canreinvite is on all devices in sip.conf |
| 16:00.16 | blitzrage | RTP (media/audio) is on ports other than 5060 -- 5060 is only signaling |
| 16:00.17 | malverian | Anyone know where the default Asterisk MOH files came from? |
| 16:00.33 | [TK]D-Fender | Dandre : Then start a a consultative transfer, then take it back and go blind. |
| 16:00.45 | [TK]D-Fender | Katty : Thats how I do all of my * installs. |
| 16:01.05 | Dandre | it is not very convenient for the end suser |
| 16:01.21 | [TK]D-Fender | Dandre : Sorry, thats just how these transfers are defined. |
| 16:01.54 | Katty | [TK]D-Fender: think you can give me a hand with my syntax? |
| 16:02.02 | Dandre | ok |
| 16:02.04 | blitzrage | [TK]D-Fender: you coming to VON Canada in TOronoto? |
| 16:02.36 | nettie | ManxPower what's wrong with polycoms? |
| 16:02.38 | saftsack | http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+congestion is the example on this page the onliest way to signal a congestion to the caller? |
| 16:02.46 | [TK]D-Fender | Katty : nothing much to say .... "wget [url]" |
| 16:02.54 | [TK]D-Fender | blitzrage : When? |
| 16:03.02 | ManxPower | nettie, nothing, but they commonly five that error when transfering a call, but the call still goes thru |
| 16:03.09 | Katty | [TK]D-Fender: well that's http. |
| 16:03.19 | Katty | [TK]D-Fender: not ftp with username and password.....but apparently mozilla does ftp. |
| 16:03.24 | [TK]D-Fender | Katty : Works for FTP too.... |
| 16:03.26 | Katty | [TK]D-Fender: so i'll just use that for now (= |
| 16:03.35 | blitzrage | [TK]D-Fender: Apr 3-5 -- I think JunK-Y is coming down too |
| 16:04.01 | [TK]D-Fender | Katty : use the standard FTP notation like "wget ftp://user:pass@server.suffix" |
| 16:04.01 | Dr-Linux | anybody is using >> http://astguiclient.sourceforge.net/ |
| 16:04.05 | tzanger | not me |
| 16:04.16 | [TK]D-Fender | blitzrage : don't think I can... |
| 16:04.29 | Katty | [TK]D-Fender: okies. |
| 16:04.36 | TinoW | .oO(or lukemftp ;) |
| 16:04.46 | *** join/#asterisk darkm20 (i=andrea@217.221.121.92) |
| 16:04.59 | darkm20 | Anyone here who has access to Cisco CCO ? |
| 16:05.49 | eric_hill | darkm20: yes |
| 16:06.17 | darkm20 | Eric_hill: Hi |
| 16:08.51 | *** join/#asterisk salviadud (n=ralfalfa@201.138.132.150) |
| 16:09.46 | *** join/#asterisk hfb (n=hfb@pool-71-116-252-188.lsanca.dsl-w.verizon.net) |
| 16:10.41 | *** join/#asterisk nDuff (n=ccd@64.128.31.220) |
| 16:11.56 | Hmm-work | Not sure if i'm going this year |
| 16:12.08 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 16:13.25 | blitzrage | tzanger: you coming out to the VON/TAUG next week? |
| 16:13.35 | Katty | Hmm-work: hi. |
| 16:13.44 | tzanger | ooh I forgot about that |
| 16:13.47 | *** join/#asterisk Cation76 (n=rafnorwi@user-0cev7pb.cable.mindspring.com) |
| 16:13.51 | tzanger | I'm going to make a real effort to get out there |
| 16:13.59 | tzanger | maybe I'll bring Alina... you'll get to meet her :-) |
| 16:14.09 | blitzrage | tzanger: schweet -- weather should be a lot better this time of year too |
| 16:14.15 | tzanger | absolutely |
| 16:14.17 | blitzrage | tzanger: schweet -- bring her hot sister too |
| 16:14.35 | kend | iDunno: not knowing about resolv.conf *is* a pretty big sin -- at least, for a *nix admin. |
| 16:14.38 | tzanger | she's got 4 sisters and a couple of them are pretty damn hot... but they're all married and in Romania |
| 16:15.09 | kend | Dr-Linux: I fired it up last week; don't know much about it, but two heads can be better than one. What's wrong? |
| 16:15.10 | blitzrage | tzanger: hahaha |
| 16:15.17 | iDunno | kend: well, indeed. Not knowing that, or anything about DNS, or being able to configure their firewall... |
| 16:15.33 | iDunno | kend: oh, and sending root passwords as plain text BY E-MAIL. |
| 16:15.43 | kend | iDunno: sounds like a top-notch sysadmin to me. ;-) |
| 16:16.09 | Hmm-work | Hi Katty |
| 16:16.17 | salviadud | gotta use pgp for that |
| 16:16.18 | Hmm-work | married and in Romania, that sure doesn't help me |
| 16:22.57 | *** join/#asterisk darkm21 (i=andrea@217.221.121.92) |
| 16:26.32 | *** join/#asterisk ManxPower (n=ewieling@dpc6745150107.direcpc.com) |
| 16:27.53 | brettnem | ugh.. I've got an unmatched } in my extensions.conf and I can't find it.. anyone got a way to find it? |
| 16:28.15 | ManxPower | brettnem, it should tell you the line number. |
| 16:28.22 | brettnem | nah |
| 16:28.26 | brettnem | let me show you |
| 16:28.41 | brettnem | Mar 27 04:17:04 NOTICE[20998]: pbx.c:1476 pbx_substitute_variables_helper_full: Error in extension logic (missing '}') |
| 16:29.12 | brettnem | any ideas? |
| 16:30.26 | [TK]D-Fender | brettnem : Try pasting the the line that CAUSED the error.... |
| 16:30.42 | brettnem | heh, if I knew that.. |
| 16:30.54 | [TK]D-Fender | brettnem : You should have an idea where it is.... |
| 16:31.06 | ManxPower | Do a search for ] or ) |
| 16:31.16 | [TK]D-Fender | brettnem : pastebin the whole damn thing then. |
| 16:31.31 | *** part/#asterisk darkm21 (i=andrea@217.221.121.92) |
| 16:32.48 | brettnem | this was the line right before it.. |
| 16:32.50 | brettnem | -- Executing Macro("SIP/t3vn-lubbock-d9e2", "Ring1Ring1Ring1Return|SIP/4325551212|SIP/4325551213|SIP/4325551214|10|10|") in new stack |
| 16:33.10 | brettnem | could that null arg be causing it? |
| 16:34.11 | *** part/#asterisk Dandre (n=Dandre@was59-3-82-236-48-30.fbx.proxad.net) |
| 16:34.20 | [TK]D-Fender | brettnem : PASTEBIN! |
| 16:34.46 | salviadud | pastebin for the love of honey maple syrup |
| 16:34.49 | salviadud | and jesus |
| 16:34.50 | *** join/#asterisk brif8 (n=Techno@lazyjtrainingcenter.com) |
| 16:34.57 | [TK]D-Fender | We'll settle for the syrop! |
| 16:35.10 | salviadud | hehe |
| 16:35.19 | nettie | ManxPower I'm getting a "noaudio available message" |
| 16:35.31 | nettie | ManxPower when I try meetme for example |
| 16:35.48 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 16:35.52 | brettnem | oh give me a break, it's one freakin line.. :P |
| 16:35.59 | nettie | ManxPower on the console.. and of course I dont hear any audio feedback on the phone |
| 16:36.13 | brettnem | I'd use up as much IRC realestate as the link to the pastebin |
| 16:36.40 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 16:37.02 | [TK]D-Fender | brettnem : Ok, stop talking about your coding flaws and SHOW US! |
| 16:37.04 | brif8 | Hi All, using a mix of Cisco 7960s and Cisco 7920s which is better (a) set all using SCCP not SIP, (b) set the 7960 using SIP and 7920 SCCP ? Also anyone had any experience with the 7920s, I am using the 7960 at present under SIP. |
| 16:37.23 | *** join/#asterisk angler_ (n=johnb@199.227.185.58) |
| 16:37.41 | brettnem | hmm it doesn't seem cause any problems.. what the heck.. :-/ |
| 16:37.45 | *** join/#asterisk HamYaI (n=HamYai@125.24.1.170) |
| 16:37.49 | brettnem | ok, let me cut it out.. |
| 16:38.29 | HamYaI | where can I find a "Change Log" or" New Features" od 1.2.6? |
| 16:39.03 | RoyK | in the source! |
| 16:39.05 | RoyK | :) |
| 16:39.29 | brettnem | [TK]D-Fender: http://pastebin.ca/47181 |
| 16:39.35 | RoyK | http://ftp.digium.com/pub/telephony/asterisk/ChangeLog-1.2.6 |
| 16:40.11 | salviadud | *added prankster support for MixMonitor |
| 16:40.28 | salviadud | yep... asterisk 1.2.6 is da bomb dude |
| 16:41.07 | brettnem | prankster support? |
| 16:41.38 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 16:41.43 | salviadud | i'm kidding |
| 16:42.37 | salviadud | though i would like to know how to make mixmonitor do my recordings on mp3 |
| 16:42.42 | salviadud | that would be fantastic |
| 16:43.12 | *** join/#asterisk angom_w (n=angom@red-corp-200.38.16.10.telnor.net) |
| 16:43.45 | loko | Does Asterisk @home actually run without timing issues under vmware? |
| 16:44.36 | Qwell | loko: not really |
| 16:45.08 | Nugget | heck, asterisk barely runs under freebsd without timing issues. vmware is optimistic. :) |
| 16:45.08 | RoyK | loko: asterisk or asterisk at home or whatever realtimesystem does not run well under vmware |
| 16:45.29 | RoyK | vmware is beyond optimistic |
| 16:45.53 | brettnem | Nugget: I'd remove the "under freebsd" from your sentense and try again. :) |
| 16:46.10 | Nugget | heh |
| 16:46.23 | brettnem | and the word "timing" |
| 16:46.25 | *** join/#asterisk mog_work (n=mogorman@gateway.digium.com) |
| 16:46.26 | tzanger | it runs pretty damn well under Xen |
| 16:46.27 | Damin | I'd spell sentence correctly.. |
| 16:46.31 | Nugget | comedy is not pretty. |
| 16:46.40 | brettnem | Damin: poetic justice |
| 16:46.48 | salviadud | it runs pretty good under linux imho |
| 16:46.53 | tzanger | salviadud: heh |
| 16:47.21 | brettnem | anyone have any experience with lighttpd? |
| 16:47.29 | mog_work | why mess with good thing |
| 16:47.31 | brif8 | anyone using Cisco 7920 and did it go ok or are there better wireless IP phones to use ? |
| 16:47.39 | Nugget | because linux isn't a good thing. :) |
| 16:47.50 | tzanger | mog_work: because there's no sense of adventure in that :-) |
| 16:47.53 | salviadud | are you saying linux is awesome? |
| 16:47.54 | brettnem | yay war |
| 16:47.54 | mog_work | billions of nerds must be wrong.... |
| 16:48.02 | salviadud | i agree with Nugget |
| 16:48.08 | Nugget | it wouldn't be the first time. |
| 16:48.30 | salviadud | brettnem, what's a yay war? |
| 16:48.53 | mog_work | are you a bsd hippie nugget? ^_^ |
| 16:49.04 | Nugget | I use what makes sense. Sometimes that's bsd. |
| 16:49.28 | Nugget | I run a little bit of everything here, even linux (for asterisk and gridmp) |
| 16:49.35 | tzanger | I love that... 8192 samples in 8191 sample intervals |
| 16:49.47 | tzanger | we squeezed an extra one in there, they'll never notice |
| 16:49.48 | nDuff | Nugget: howdy |
| 16:50.01 | Nugget | moo |
| 16:51.43 | *** join/#asterisk Eggplant (i=No@dsl-448.cascadeaccess.com) |
| 16:53.01 | *** join/#asterisk GerbilNut (i=GerbilNu@65.88.144.41) |
| 16:53.10 | _Paulo_ | I like BSD but cant bear Theos atitude. |
| 16:53.30 | *** join/#asterisk signal-eleven (n=evan@lion.ragga-jungle.com) |
| 16:53.48 | Nugget | Theo's a lot less annoying than RMS. |
| 16:53.55 | Nugget | but it's a fair point nontheless |
| 16:54.04 | *** join/#asterisk octothorpe (n=octothor@unaffiliated/octothorpe) |
| 16:54.17 | signal-eleven | hey all, anyone know how to make asterisk respond with a specific sip response code from an AGI script? ie. force the return of a 404 not found. |
| 16:54.20 | file | Nugget, what is today's speciality sauce? |
| 16:54.35 | angler_ | g love and special sauce |
| 16:54.38 | Nugget | mango habañero |
| 16:54.42 | [TK]D-Fender | # |
| 16:54.42 | [TK]D-Fender | exten => s,1,NoOp(Ringing ${ARG1} then ${ARG2} then ${ARG3} for ${ARG4} seconds then rolling to ${ARG5) - ${MACRO_EXTEN} from ${MACRO_CONTEXT} ${SIP_CODEC} ${CALLERIDNUM}) |
| 16:54.52 | [TK]D-Fender | brettnem : look at ARG5 |
| 16:55.18 | brettnem | wahoo.. nice catch.. |
| 16:55.23 | brettnem | my eyes were crossing looking for those. |
| 16:56.06 | tzanger | hahaha |
| 16:56.17 | tzanger | I have to do something a little similar |
| 16:56.24 | [TK]D-Fender | Great Andrew sees all.... knows nothing..... |
| 16:56.42 | tzanger | I need to send a SIP INFO message to a bunch of ip501s... but hte SIP INFO header is different for some of htem |
| 16:56.44 | _Paulo_ | who listen to RMS? RMS is into politics, not on IT. |
| 16:56.45 | brettnem | hmm I used to have that rainbow parens plugin for vi.. where'd that goo.. |
| 16:56.49 | brettnem | -o |
| 16:56.56 | tzanger | so I have to Dial(Local/) and split it off from there |
| 16:57.01 | tzanger | I can't think of a different way to do it |
| 16:57.08 | tzanger | I see all and know nothing? |
| 16:57.09 | tzanger | ouch :-) |
| 16:57.30 | brettnem | tzanger: I'm afraid of the local channel. :) |
| 16:57.41 | ManxPower | Bell sheduled the conversion from our CLEC to the ILEC for 10am tomorrow morning. |
| 16:57.48 | tzanger | Local/ and I are great friends |
| 16:58.16 | nDuff | I'm trying to figure out exactly how voicemail.conf's externpass works. Apparently voicemail.conf is no longer regenerated when voicemail.conf is in use (which is good) -- but where is it documented how the new mailbox/password combo is given to the invoked externpass process? |
| 16:58.22 | tzanger | Dial(Local/${EXTEN}@SipInfoFoo&Local/${EXTEN}@SipInfoBar) |
| 16:58.34 | tzanger | and then [SipInfoFoo] sends Foo and [SipInfoBar] sends Bar |
| 16:58.47 | tzanger | there's nothing quite like a dialplan that works as a tree :-) |
| 16:59.06 | brettnem | tree? |
| 16:59.08 | MooingLemur | my dialplan is more like velcro |
| 16:59.23 | MooingLemur | hooks and loops |
| 16:59.26 | [TK]D-Fender | brettnem : My guess : Poison Oak :) |
| 16:59.28 | tzanger | MooingLemur: heh |
| 16:59.40 | *** join/#asterisk nettie (i=esivieri@85-18-54-38.ip.fastwebnet.it) |
| 16:59.42 | tzanger | brettnem: yeah... call comes in and splits off to two different contexts that run at the same time |
| 17:00.11 | nettie | ManxPower the problem is that I can hear asterisk back only is SIP ALG is enabled on the router.. |
| 17:00.12 | tzanger | actually it'd be Local/${EXTEN}@OfficePhones&Local/${EXTEN}@RemotePhones) |
| 17:00.12 | brettnem | yeah, I like the local channel driver.. I've just had some deadlock issues in the past that still haunt me |
| 17:00.36 | tzanger | and [RemotePhones] changes the SIP INFO message depending on time of day and stuff, because he doesn't want his home phones ringing for business calls after a certain time, phase of the moon, etc. |
| 17:05.23 | *** join/#asterisk SplasPood (n=jwb@206.252.198.100) |
| 17:06.52 | *** join/#asterisk nahirean (n=nahirean@67.132.43.2) |
| 17:06.56 | nahirean | Anyone know what NOTICE[2027]: pbx_spool.c:232 attempt_thread: Call failed to go through, reason 1 |
| 17:07.00 | nahirean | would indicate? |
| 17:07.56 | *** join/#asterisk SkalTura (i=none@a85-156-173-3.elisa-laajakaista.fi) |
| 17:08.14 | signal-eleven | nahirean: reason 1 is AST_CONTROL_HANGUP |
| 17:08.44 | ManxPower | you are assuming "reason" == "hangup cause". I don't know if that is true. |
| 17:10.28 | *** join/#asterisk edobe (n=shigueta@69.65.149.190) |
| 17:10.30 | nahirean | hmm, ok - thanks for your help.. ill take a look .. have a good one |
| 17:11.38 | edobe | hi all, i´m testing asterisk with a softphone... the softphone shows buttons for different lines, how can i configure asterisk for this multi-line support, so I press Line 1 and it uses 1 line, Line 2 and it uses other, and so on |
| 17:11.58 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 17:12.11 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 17:12.16 | ManxPower | I see this is "newbie questions day" today |
| 17:12.30 | Hmm-work | whats the deal, with my brain, why am I so obviously insane |
| 17:13.38 | salviadud | for starters, you are asking in the wrong chan buddy |
| 17:13.51 | salviadud | maybe #psychiatry would help |
| 17:14.01 | *** join/#asterisk newmember (n=username@ptr-66-11-81-65.ptr.terago.ca) |
| 17:14.36 | SkalTura | ;) |
| 17:14.44 | salviadud | edobe, that softphone, what brand is it? |
| 17:14.56 | salviadud | did you get it in a box of Frosted Flakes? |
| 17:15.13 | *** join/#asterisk mdo_ (n=13h7@p508A24E7.dip0.t-ipconnect.de) |
| 17:15.15 | mdo_ | hi |
| 17:15.17 | brettnem | awesome.. My wifi connection works outside my office. |
| 17:15.23 | brettnem | wahoo |
| 17:15.24 | mdo_ | i am looking for a system that can call a number via voip and play prerecorded sound files, further react on keypad input for e.g. repeat a sound file. is asterisk for this or does someone know a more qualified system? |
| 17:15.39 | salviadud | planning on watchin pr0n outside the office eh? |
| 17:15.40 | brettnem | mdo_: very easy to do that |
| 17:15.43 | edobe | i thought this was asterisk channel, not clownswannabe |
| 17:15.57 | jbalcomb | asterisk really should come with a /working/ samples config. say set your phone to ext 1234 and you can test it right off. |
| 17:15.58 | signal-eleven | this is an asterisk channel, not a channel for your softphone |
| 17:16.02 | brettnem | salviadud: heh, I'm in the cafe downstairs.. ;) |
| 17:16.19 | brettnem | asteisk does come with working samples! |
| 17:16.20 | edobe | signal-eleven: so i don´t need to configure anything on asterisk for this? |
| 17:16.25 | [TK]D-Fender | jbalcomb : I'm working on that.... |
| 17:16.26 | brettnem | hmm.. this is newbie day eh? |
| 17:16.29 | wasim | AMR! |
| 17:16.30 | X-Gen | u hear about the chick that was killed, she had Frosted Flakes sprinkled around her. they think it was a cereal killer |
| 17:16.34 | signal-eleven | edobe: nope, just make your softphone register each line as a different account |
| 17:16.42 | salviadud | hahaha, cereal killer |
| 17:16.47 | brettnem | X-Gen: HAR HAR! |
| 17:16.48 | brettnem | ;) |
| 17:17.00 | X-Gen | :D |
| 17:17.00 | signal-eleven | or the same account depending on the protocol (for instance you can register the same sip account multiple times) |
| 17:17.10 | salviadud | brettnem, you laugh like a pirate maaaaaan |
| 17:17.12 | jbalcomb | [TK]D-Fender right on |
| 17:17.41 | jbalcomb | what does this mean/do? astdb=chan2ext/SIP/grandstream1=1234 ; ensures an astDB entry exists |
| 17:17.56 | [TK]D-Fender | X-Gen : Did you you hear that the Energizer Bunny was just found dead? The doctor cited the cause of dead as "acute sexual exhaustion". Apparently someone put his batteries in BACKWARDS.... |
| 17:18.13 | [TK]D-Fender | jbalcomb : That line is BS. |
| 17:18.16 | edobe | signal-eleven: but asterisk is the one that routes the calls through trunks... how do i define that Line1 is a line and Line2 is another? I mean to use a specified line |
| 17:18.18 | brettnem | jbalcomb: yes |
| 17:18.29 | *** join/#asterisk sysdebug (n=sysdebug@200.250.222.8) |
| 17:18.54 | X-Gen | lol @ fender |
| 17:18.56 | signal-eleven | edobe: depends on the protocol. if you have a sip account registered to 4 lines when a call comes in serial forking occurs and all the lines ring at the same time, the first one to answer winds |
| 17:19.01 | signal-eleven | winds = wins |
| 17:19.19 | signal-eleven | if you use multiple accounts for the lines then it's pretty obvious how it works |
| 17:19.49 | *** join/#asterisk Cooltalk (n=io@203.91.145.184) |
| 17:20.37 | salviadud | my social engineering is strong |
| 17:20.50 | salviadud | i have a direct line with the president of a phone company |
| 17:20.53 | mdo_ | brettnem: thnx |
| 17:20.54 | salviadud | located in nevada |
| 17:21.00 | salviadud | and she's a girl |
| 17:21.08 | salviadud | i wanna ask her out or something |
| 17:21.22 | salviadud | she sounds kinda worried and financially insecure |
| 17:22.07 | salviadud | all thanx to asterisk... the funkiest pbx |
| 17:22.40 | signal-eleven | any agi coders know howto generate a specific sip response code from a fastagi script? ie. send a 404 not found back? |
| 17:22.46 | rpm | this is going to turn into the story.. "How asterisk got salviadud laid." |
| 17:23.18 | salviadud | rpm, i like you're thinking dude |
| 17:23.40 | salviadud | i got into this business just for that |
| 17:23.57 | salviadud | some say... it's because jabbering on the phone is cool |
| 17:24.02 | salviadud | i saw, it's the ladies |
| 17:24.12 | salviadud | we are rockstars in our own way |
| 17:24.18 | salviadud | we pay the bills |
| 17:24.30 | salviadud | therefore, we get the bitches and hoes |
| 17:24.37 | signal-eleven | lol |
| 17:24.38 | salviadud | we are the asterisk pimps |
| 17:24.49 | justinu | w3rd |
| 17:24.56 | signal-eleven | asterisk - when you need a pbx or bitches & hoes |
| 17:25.11 | salviadud | yeah man! it's like this |
| 17:25.16 | salviadud | i go to a company |
| 17:25.18 | salviadud | setup a pbx |
| 17:25.28 | salviadud | and so i memorize which secretary is hot |
| 17:25.54 | salviadud | so, one day, i play with extensions.conf, and ALL her calls get transferred to me |
| 17:26.11 | salviadud | of course, i play it dumb "geez louise! you got me agaaaaain?" |
| 17:26.18 | salviadud | stuff like that |
| 17:26.28 | *** join/#asterisk bamp (n=iraklion@olon.ath.forthnet.gr) |
| 17:26.37 | *** join/#asterisk octothorpe (n=octothor@unaffiliated/octothorpe) |
| 17:26.44 | signal-eleven | i thought that was called stalking? |
| 17:26.51 | signal-eleven | hahaha |
| 17:26.58 | salviadud | no way man. i'm in mexico |
| 17:27.04 | salviadud | this is called latin love |
| 17:27.10 | signal-eleven | hahaha. nice. |
| 17:27.14 | eric_hill | mexico? I'm coming down there in two weeks to see you then... |
| 17:27.24 | [TK]D-Fender | ~>~ |
| 17:27.27 | salviadud | what part of mexico are you going to? |
| 17:27.32 | eric_hill | Guadalajara. |
| 17:27.49 | eric_hill | Which, btw, has too many "a"'s in the name. |
| 17:27.57 | salviadud | mmmmmm, i'm in monterrey, you won't be seeing me... |
| 17:28.35 | salviadud | yeah, it's a freaky name, and even i don't know what it means... |
| 17:28.51 | salviadud | it's probably an aztec name |
| 17:29.08 | *** join/#asterisk stoffell (n=stoffell@d51A4D49E.access.telenet.be) |
| 17:29.38 | eric_hill | I was in Cuatalincinco Puebla a few months ago. Nice city, really. Never learned how to spell the first name though... |
| 17:30.18 | rpm | i wonder how much of a pain in the ass it would be to write a voicemail system without using app_voicemail.c |
| 17:30.38 | eric_hill | X-Gen: if you type your password in, it comes out as stars! See: ********* |
| 17:30.47 | salviadud | i wonder if the guys at digium ever check the logs and go "damn... i thought those guys were devs..." |
| 17:31.07 | eric_hill | http://www.bash.org/?244321 |
| 17:31.21 | X-Gen | eric_hill: whenever i'm depressed i read bash.org |
| 17:31.40 | X-Gen | they should mention what # it was said on aswell |
| 17:32.19 | eric_hill | aye. |
| 17:32.38 | salviadud | poor fool |
| 17:32.40 | salviadud | haha |
| 17:32.56 | salviadud | i remember when i made a guy winnuke himself |
| 17:33.04 | salviadud | back in... 95 |
| 17:33.15 | salviadud | when windows was all i had |
| 17:33.26 | *** part/#asterisk bamp (n=iraklion@olon.ath.forthnet.gr) |
| 17:35.22 | *** join/#asterisk moprilo (i=whatMMx@201.192.107.57) |
| 17:35.45 | moprilo | i need to install the digium tdm24, but i can't find the module .. :S |
| 17:35.48 | moprilo | help ? |
| 17:36.06 | moprilo | i had a link with it, but i seem to have lost it |
| 17:36.41 | rpm | moprilo: www.asterisk.org |
| 17:36.46 | rpm | moprilo: zaptel drivers |
| 17:38.01 | mog_work | wctdm24xxp |
| 17:38.32 | *** join/#asterisk skkip (n=Skipper@216.160.91.91) |
| 17:38.44 | [TK]D-Fender | When are the rest of the * resources going to acknowledge the new * release? Wiki has no mention, and asterisk.org doesn't have a new article.... |
| 17:39.06 | *** join/#asterisk Eggplant (i=No@dsl-176.cascadeaccess.com) |
| 17:39.58 | jbalcomb | [TK]D-Fender yeah, how about the book that has examples that result in massive amounts of syntax errors? |
| 17:40.29 | ManxPower | [TK]D-Fender, that's why you should not rely on those. Rely on the docs that come with Asterisk |
| 17:40.38 | jbalcomb | ManxPower where are those? |
| 17:40.38 | rpm | _XXX. matches only 3 numbers right? |
| 17:40.58 | jbalcomb | rpm i dont think so.. isnt the . anything of any amount? |
| 17:41.12 | iDunno | no, there's a . |
| 17:41.13 | ManxPower | jbalcomb, in the "docs" directory of the asterisk source code. Also "show applications" at the Asterisk CLI, as well as the configs directory of the asterisk source tree. |
| 17:41.23 | iDunno | _XXX will match 3 numbers ;) |
| 17:41.48 | ManxPower | _XXX. will match FOUR or MORE numbers. |
| 17:42.10 | salviadud | TK, you know how i can get mp3 support for mixmonitor, i was reading and supposedly, i need soxmix to be able to encode to mp3, which unfortunately, it doesn't, and i have no real clue on how to compile soxmix or sox with mp3 for that matter |
| 17:42.28 | jbalcomb | ManxPower i thought the dot meant it didnt need to exist? so should be atleast but also anything more? |
| 17:42.43 | ManxPower | no, . means "one or more" |
| 17:42.51 | salviadud | compile with mp3 support, that is |
| 17:43.03 | salviadud | anybody with that same pickle? |
| 17:43.11 | salviadud | mixmonitor and mp3 support? |
| 17:43.33 | [TK]D-Fender | salviadud : no clue |
| 17:45.18 | *** join/#asterisk kardecallan (n=kardecal@ns1.pcma.com.br) |
| 17:45.23 | salviadud | allright, i'll do some more research |
| 17:45.24 | jbalcomb | ManxPower is there a ? for 'possible'? like _XXX? three or maybe four numbers but not anymore than four? |
| 17:45.25 | ManxPower | of course any time you use "." in a pattern match the person dialing will have to wait for DigitTimeout before their call will be processed |
| 17:45.35 | signal-eleven | salviadud: you need to have either libmad or libmp3lame installed when you build sox to get mp3 support |
| 17:45.40 | ManxPower | how about _XXX and _XXXX |
| 17:45.58 | jbalcomb | ManxPower shouldnt have to do two statements |
| 17:46.11 | ManxPower | jbalcomb, best of luck with fighting Asterisk. |
| 17:46.23 | jbalcomb | ManxPower I'm pretty sure Perkl regex uses the ? like that |
| 17:46.36 | ManxPower | jbalcomb, Asterisk does not have perl regex |
| 17:46.46 | ManxPower | or any kind of regex for exten patterns at all |
| 17:46.51 | jbalcomb | ManxPower asterisk has asterisk regex? ;) |
| 17:47.04 | signal-eleven | i wouldn't even call it regex, more like pattern matching |
| 17:47.14 | *** join/#asterisk PakiPenguin (i=uppal@linuxpakistan/admin/pakipenguin) |
| 17:47.17 | Hmm-work | cmd regex was messed up in 1.2.5 |
| 17:47.24 | jbalcomb | i would say that it may be low-level but that is still regex |
| 17:47.40 | salviadud | thanx signal-eleven |
| 17:47.41 | jbalcomb | Hmm-work yummy |
| 17:47.51 | Hmm-work | yummy? |
| 17:47.59 | skkip | anyone get Zaptel to compile on a FC5? I am getting You do not appear to have the sources for the 2.6.15-1.2054_FC5smp kernel installed. |
| 17:48.15 | signal-eleven | skkip: do you have the sources for you kernel installed? |
| 17:48.24 | jbalcomb | Hmm-work yeah, that's how i feel about the potential disaster that would result from a stable release blowing up on something fairly important. ;) |
| 17:48.36 | Hmm-work | ahh |
| 17:48.38 | *** join/#asterisk TinoW (n=tino@living-examples.com) |
| 17:49.07 | skkip | Sig: I thought I DL it and installed it but I am guessing the the BUILD dir did not get populated as it should. |
| 17:49.35 | jbalcomb | Hmm-work when I got hired to learn and admin asterisk here I kinda thought i would have it figured out in three to four months and then focus on more fun, progressive projects.. silly boy, I am. |
| 17:49.36 | ManxPower | unfortunatly that file does not seem to have the new patterns in it. |
| 17:49.46 | signal-eleven | skkip: run - rpm -qa | grep 2.6.15-1.2054 |
| 17:49.48 | umay | skkip: im getting errors building zaptel cvs on debian with 2.6.15 kernel |
| 17:49.48 | ManxPower | Of course you should be looking at the extensions.txt file in the asterisk doc directory |
| 17:49.56 | *** join/#asterisk Z0m81e (n=pault@85-210-190-236.dsl.pipex.com) |
| 17:50.04 | signal-eleven | skkip: you should see a src package if you've installed it |
| 17:50.04 | umay | i mean svn trunk not cvs |
| 17:50.48 | Z0m81e | Hey all, does anyone have experience of the SPA3000? I borrowed one and i'm trying to get it to work with * but at the mo I can't get it to do anything... I have a dect phone plugged into it but the **** commands don't do anything and there is no dialtone |
| 17:50.55 | skkip | sig: returns the followin - kernel-smp-2.6.15-1.2054_FC5 |
| 17:51.06 | signal-eleven | skkip: so you need the kernel source |
| 17:51.13 | jbalcomb | ManxPower how shall i start in docs? README-configuration? |
| 17:51.19 | skkip | i'll try again thanks |
| 17:51.24 | ManxPower | jbalcomb, I don't care. |
| 17:51.42 | jbalcomb | nice |
| 17:51.50 | umay | i think you can also link /lib/modules/2.6.x/build to /usr/src/linux-2.6 |
| 17:51.56 | kardecallan | Is there Anybody to know this error: [app_enumlookup.so]Mar 27 14:48:48 WARNING[16265]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/app_enumlookup.so: undefined symbol: option_priority_jumping |
| 17:52.37 | signal-eleven | kardecallan: something app_enumlookup depends on isn't loaded, or you have a compile problem |
| 17:53.20 | *** join/#asterisk lzhang (n=lewiszha@67.95.13.46) |
| 17:53.36 | ManxPower | kardecallan, sounds like you are using an app_enumlookup from 1,2 with the source from 1.0 |
| 17:55.04 | ManxPower | Did you ignore the warnings when you did a "make install" |
| 17:56.55 | kardecallan | How I make to know which are app_enumlookup depends? |
| 17:57.33 | ManxPower | Did you ignore the warnings when you did a "make install"? |
| 17:58.12 | ManxPower | the easiest thing to do is remove /usr/lib/modules/asterisk and reinstall Asterisk |
| 17:58.49 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 18:00.54 | kardecallan | ok ManxPower, I will make this |
| 18:01.32 | ManxPower | if the problem still happens after reinstalling asterisk, then your asterisk source is corrupted and you should download it again |
| 18:03.20 | kardecallan | This happened later that I installed the library unicall. |
| 18:03.34 | *** join/#asterisk zapa (n=zant@200.66.19.194) |
| 18:05.28 | *** join/#asterisk malverian (n=malveria@adsl-065-005-207-210.sip.gnv.bellsouth.net) |
| 18:06.05 | *** join/#asterisk unixgeek (n=unixgeek@12.45.238.189) |
| 18:08.31 | *** join/#asterisk asterboy (n=kevin@S010600485480f4be.ed.shawcable.net) |
| 18:09.22 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 18:09.27 | asterboy | Is anyone here using the call forward features of asterisk? |
| 18:09.37 | Qwell[] | asterboy: Ask your real question |
| 18:09.42 | ManxPower | asterboy, which ones? |
| 18:09.50 | ManxPower | Yeah, ask a real question. |
| 18:10.05 | asterboy | ya, thats just it, there are a few features of call forward I need to drill out. |
| 18:10.12 | asterboy | Here is the situation. |
| 18:10.25 | ManxPower | asterboy, only zaptel has call forward features, so I assume your question is about Zap. |
| 18:10.36 | asterboy | yes |
| 18:10.53 | asterboy | I'm setting up a telephone system for a crisis center. |
| 18:11.08 | asterboy | After hours, the volunteers need to have the system forward calls. |
| 18:11.19 | asterboy | I have a 3 line rotary. |
| 18:11.36 | asterboy | call comes in and needs to forward to volunteer #1 |
| 18:11.41 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:11.53 | asterboy | if that does not answer, the call needs to then goto volunteer #2 |
| 18:11.58 | asterboy | and so on... |
| 18:12.00 | ManxPower | asterboy, forward or transfer? |
| 18:12.26 | asterboy | well a forward because the number is off site like a ccell phone. |
| 18:12.36 | blitzrage | I use dialplan logic to do that... |
| 18:13.03 | ManxPower | asterboy, You would have to look it up but *72 is what is usually used on your Zap FXS lines. |
| 18:13.12 | asterboy | yes that is true. |
| 18:13.27 | asterboy | my question is this: |
| 18:13.32 | ManxPower | and it works pretty much just like the telco does, except of course that it takes 2 lines per call. |
| 18:14.09 | *** join/#asterisk astra^^ (n=muhajir_@59.145.104.74) |
| 18:14.32 | asterboy | If I have call forward setup on my line and a call comes to *... * will have to pickup ANOTHER line to make the call out to the cell...then the telco drops the original line, correct? |
| 18:14.50 | ManxPower | asterboy, no, both lines will be in use. |
| 18:14.59 | ManxPower | your telco call forward us not used when you do a zaptel call forward. |
| 18:15.26 | ManxPower | so pick one and use it. |
| 18:15.28 | Z0m81e | Does anyone know if you can use a DECT phone to access the IVR on a SPA3000? |
| 18:15.28 | asterboy | ok, so I don't need the telco call forward? |
| 18:15.54 | asterboy | or like you say, pick one. |
| 18:16.11 | ManxPower | asterboy, you have 3 lines. A call comes in, is forwarded by asterisk out a 2nd line and asterisk then bridges both lines. Then a 2nd call comes in and cannot be forwarded. |
| 18:16.37 | asterboy | but on a rotary it could |
| 18:16.44 | asterboy | start another bridge anyway. |
| 18:17.09 | ManxPower | umay, the 2nd call will come in on the 3rd line, then there will be no free line to send the call out of. |
| 18:17.21 | asterboy | unless you provision for it. |
| 18:17.26 | asterboy | like have a 4th line. |
| 18:17.33 | ManxPower | asterboy, then you don't have 3 lines anymore. |
| 18:17.50 | asterboy | <PROTECTED> |
| 18:18.28 | asterboy | right? |
| 18:18.40 | ManxPower | obviously |
| 18:19.08 | asterboy | yes, obvously. :) |
| 18:19.19 | asterboy | ok, thanks guys, I get the picture now. |
| 18:19.28 | blitzrage | since you can run Asterisk on a bunch of little embedded systems, has anyone found one that can around 8 sim. channels (regular, or MeetMe conferences) -- not too sure if anyone of them have the power to do transcoding or not though |
| 18:19.51 | asterboy | I'll need to adjust my proposal to accomodate the maximum call forwarding load. |
| 18:20.03 | Qwell[] | blitzrage: cluster some netgear routers :P |
| 18:20.05 | asterboy | basically doubling the line capacity. |
| 18:20.16 | asterboy | lol, clust netgear. |
| 18:20.22 | Qwell[] | asterboy: Just get a SIP provider |
| 18:20.25 | blitzrage | Qwell[]: not a bad idea... wonder if those can do 2-3 transcoded calls :) |
| 18:20.29 | signal-eleven | i've got asterisk on a wrt54l and I can get 6 calls using pass through or 1 using transcoding |
| 18:20.32 | blitzrage | I heard Mix Networks is a good one :) |
| 18:20.36 | Qwell[] | blitzrage: alaw<>ulaw maybe :) |
| 18:20.42 | asterboy | Qwell, totally would love to, but the client has it out for VOIP. |
| 18:20.43 | blitzrage | </shameless_self_promotion> |
| 18:20.57 | blitzrage | Qwell[]: heh :) |
| 18:20.58 | asterboy | typicall, shy away from new technology they don't understand. |
| 18:20.59 | Qwell[] | blitzrage: How many cookie jars do you have your hands in, exactly? |
| 18:21.09 | blitzrage | Qwell[]: to f'n many |
| 18:21.12 | Qwell[] | heh |
| 18:21.14 | blitzrage | s/to/too |
| 18:21.22 | file | blitzrage, *cough* you can use... you know what... to transcode... |
| 18:21.31 | blitzrage | file: AHA!!!!!!!!!!!!!!!!!!! |
| 18:21.42 | Qwell[] | "you know what"? |
| 18:21.51 | blitzrage | Qwell[]: super secret stuff :) |
| 18:21.57 | Qwell[] | sheesh |
| 18:22.06 | asterboy | g739? |
| 18:22.07 | file | well, I'll be putting it into a branch in my team folder soon... |
| 18:22.10 | asterboy | :P |
| 18:22.23 | file | so it won't be super secret for long |
| 18:22.23 | blitzrage | file: can I do it off-site? Would be cool to deploy the little netgears, then centralize the transcoding from a few clients... |
| 18:22.24 | ManxPower | OH YES, get a SIP provider for your crisis center. |
| 18:22.41 | file | blitzrage, in theory yeah... using the P2P stuff... |
| 18:22.49 | signal-eleven | anyone know howto generate a specific sip response code from a fastagi script? ie. send a 404, 484 or 500 back |
| 18:22.49 | file | you just wouldn't have the features of a community setup |
| 18:22.51 | blitzrage | file: sweet cookies :) |
| 18:23.03 | Qwell[] | community transcoding? |
| 18:23.11 | Qwell[] | woops, did I blow the secret? :D |
| 18:23.35 | file | it's not like an uber uber 1337 secret |
| 18:23.51 | Qwell[] | is it something you talked about at VON? :p |
| 18:23.52 | *** join/#asterisk Jon335 (n=jon335@ottawa-hs-209-217-84-138.d-ip.magma.ca) |
| 18:23.55 | blitzrage | file: thats cool -- I have a situation where I could deploy a few small boxes into some offices in Sarnia (they are cheap and behind technology about 5 years) -- but bandwidth won't be very much -- so offloading the transcoding to a CO (i.e. my house :)) would be wicked |
| 18:23.58 | Qwell[] | when you heard about the super secret PCI card? |
| 18:24.05 | file | yesssssssssss |
| 18:24.06 | blitzrage | Qwell[]: yah -- if it was, we'd not be talking about it in here :) |
| 18:24.11 | Qwell[] | heh |
| 18:24.40 | Qwell[] | all they need to do, is make that card minipci |
| 18:24.44 | ManxPower | you guys have alot more confidence in the internet than I do. |
| 18:24.49 | [av]bani | the interweb rocks |
| 18:25.14 | ManxPower | signal-eleven, I don't believe you can. |
| 18:26.16 | Qwell[] | ManxPower: It's not like it's a life/death crisis thing |
| 18:26.25 | Qwell[] | I mean, unless it's like a suicide hotline or something |
| 18:26.26 | tzanger | community transcoding? odd |
| 18:26.43 | mutilator | eleventy seven! |
| 18:26.47 | signal-eleven | ManxPower: that sucks... is there any way I can generate any responses by like setting a variable or anything... I've been digging through res_agi and chan_sip but nothing's stickin out |
| 18:26.56 | ManxPower | signal-eleven, no. |
| 18:27.04 | file | tzanger, Stay tuned! |
| 18:27.07 | ManxPower | Asterisk does not expose the protocol internals to AGI |
| 18:27.20 | tzanger | that's perverse |
| 18:27.43 | signal-eleven | crap-tastic... alright, i'll have to figure out another way to do this |
| 18:27.49 | ManxPower | now, you can disconnect with a specific HANGUPCAUSE, and those hangupcauses are mapped to specific SIP responses. |
| 18:27.57 | signal-eleven | ahhh |
| 18:27.58 | ManxPower | signal-eleven, SER is a SIP Proxy |
| 18:28.00 | Qwell[] | tzanger: Will be fun to ChanSpy() them |
| 18:28.05 | tzanger | Qwell[]: indeed |
| 18:28.10 | Qwell[] | the ultimate in privacy invasion :P |
| 18:28.12 | ManxPower | Asterisk is not a SIP proxy. |
| 18:28.17 | signal-eleven | ManxPower: I know, that's what I'm sending the responses back to |
| 18:28.17 | tzanger | file does use a lot of call-girl services |
| 18:28.19 | ManxPower | It is a protocol agnostic PBX |
| 18:28.26 | Qwell[] | tzanger: really? |
| 18:28.29 | *** join/#asterisk Utah_Dav1 (n=boucha@0-1pool149-179.nas31.salt-lake-city1.ut.us.da.qwest.net) |
| 18:28.36 | Qwell[] | file: bad, bad, bad |
| 18:31.17 | signal-eleven | ManxPower: HANGUPCAUSE will work nicely, thanks |
| 18:31.38 | *** join/#asterisk areski (n=areski@polar.es6.egwn.net) |
| 18:33.38 | Hmm-work | i remember the way you curled your toes on side of the stage at all our shows, and the glow on your face just because of one rose, and I wake up in the morning and you're wearing my clothes |
| 18:34.19 | *** join/#asterisk SuperLag (n=aaron@gentoo/developer/SuperLag) |
| 18:35.32 | weinerk | Any scathing criticisms against a plan to put |
| 18:35.32 | weinerk | about 20 telemarketers with XTEN-lite phones with USB headsets? |
| 18:35.52 | salviadud | you bastard |
| 18:36.07 | salviadud | 1 telemarketer is evil enough |
| 18:36.10 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 18:36.17 | salviadud | 20 is an army |
| 18:36.49 | salviadud | i remember the last time they called me, i hung up on they're middle speech about what they were offering me |
| 18:37.03 | salviadud | i feel sorry for those guys, worst job ever |
| 18:37.06 | asterboy | We should donate to telemarketers! |
| 18:37.16 | salviadud | hauling trash is a lot more fun... |
| 18:37.22 | asterboy | a big pile of poo. |
| 18:37.34 | *** part/#asterisk Utah_Dav1 (n=boucha@0-1pool149-179.nas31.salt-lake-city1.ut.us.da.qwest.net) |
| 18:37.41 | [av]bani | yeah, a job where you know people hate you |
| 18:38.06 | asterboy | I've been approached by them to offer VOIP phones...told them no. |
| 18:38.19 | [TK]D-Fender | [av]bani : That'd be dentistry, one of the highest suicide rates of any profession... |
| 18:38.51 | [TK]D-Fender | weinerk : Softphones suck, get them minimal hardphones & headsets. |
| 18:39.42 | Hmm-work | specially when you're watching a movie, burning a dvd and playing quake4 all at once while trying to talk on it |
| 18:40.16 | weinerk | [TK]D-Fender, thanks. from your experience which cheap ones would you recommend ? |
| 18:40.42 | asterboy | Hmm-work, lol all at once. |
| 18:40.55 | [TK]D-Fender | weinerk : best one for that use : Polycom IP 301 + Plantronics M12 amplifier + H261 headset. |
| 18:41.35 | Hmm-work | asterboy: we call that multitasking |
| 18:42.00 | asterboy | no way I'd burn a cD while doing some quake. |
| 18:42.00 | [TK]D-Fender | Hmm-work : Studies show multitasking is becoming increasingly counter-productive. |
| 18:42.01 | weinerk | Hmm-work :-) Are you saying that unless you spike CPU - its ok? |
| 18:42.10 | asterboy | I like the detail h-res. |
| 18:42.23 | [TK]D-Fender | asterboy : What, and sacrifice a few FPS? NEVER! |
| 18:42.32 | asterboy | The only thing I have found that can handle it is the dual SLI . |
| 18:43.10 | Hmm-work | [TK]D-Fender: yeah especially when one of your tasks is reading threads on fark.com |
| 18:43.21 | asterboy | :) |
| 18:43.23 | tzanger | yeah there was actually a good article |
| 18:43.35 | tzanger | "(Some) Attention Must Be Paid!" |
| 18:43.45 | tzanger | all about that |
| 18:44.12 | brif8 | anyone using Cisco 7920 and did it go ok or are there better wireless IP phones to use ? |
| 18:44.13 | asterboy | britney spears has a gunt now...natures cruel |
| 18:44.43 | Qwell[] | brif8: 7920 works well. It'll work better with chan_skinny when I get my hands on one, to fix it |
| 18:44.45 | *** join/#asterisk brodiem (i=1000@cpe-66-69-222-36.austin.res.rr.com) |
| 18:44.55 | tzanger | a gunt? |
| 18:45.02 | asterboy | gut + cunt |
| 18:45.24 | brif8 | Qwell[]: Thanks |
| 18:45.51 | tzanger | interesting... |
| 18:45.55 | tzanger | I'd just say she's pregnant but ok |
| 18:46.09 | asterboy | oh the pouch has been stretched |
| 18:46.35 | asterboy | let's see her in a thong now. |
| 18:47.53 | *** join/#asterisk zapa (n=zant@200.66.19.194) |
| 18:50.22 | asterboy | Just landed an Asterisk install! |
| 18:50.26 | asterboy | Yippeee. |
| 18:50.41 | salviadud | what ya mean, they hired ya? |
| 18:50.48 | *** join/#asterisk cryptnix (n=andrew@zero.levelsync.com) |
| 18:50.59 | salviadud | or, you just installed * on a box? |
| 18:51.12 | asterboy | No, I quoted a telephone system and beat out the other guys using * |
| 18:51.12 | salviadud | in any case, cheers man :) |
| 18:51.15 | cryptnix | Anyone here having issues with Asterisk@Home with an IAX setup and trying talk between extensions ... ? |
| 18:51.29 | asterboy | I'm providing the system. |
| 18:51.36 | asterboy | Couldn't get them to go for Polycom. |
| 18:51.44 | asterboy | Have to use the Grandstream GPX-2000 |
| 18:51.49 | asterboy | oh well. |
| 18:51.52 | salviadud | grandstream is good |
| 18:51.59 | salviadud | i like it better than sipura |
| 18:52.02 | salviadud | which i own, and hate |
| 18:52.08 | *** join/#asterisk TinoW (n=tino@living-examples.com) |
| 18:52.10 | Qwell[] | salviadud: send it here |
| 18:52.11 | asterboy | I love my Polycom phones though. |
| 18:52.50 | *** join/#asterisk sigwerk (n=sigwerk@cyclone.sigterm.net) |
| 18:53.00 | *** join/#asterisk trbldwine (i=trbldwin@vpn164060.vpn.northwestern.edu) |
| 18:53.35 | salviadud | i'd trade a sipura 3000 for a grandstream handytone |
| 18:53.38 | salviadud | anytime |
| 18:53.47 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 18:53.47 | tzanger | you don't like hte SPA3k? |
| 18:53.54 | salviadud | it works |
| 18:54.01 | salviadud | it has loads of functions... |
| 18:54.01 | *** join/#asterisk livesNbox (n=livesNbo@68-76-129-2.ded.ameritech.net) |
| 18:54.09 | salviadud | which i hate configuring... |
| 18:54.10 | cryptnix | Anyone here having issues with Asterisk@Home with an IAX setup and trying talk between extensions ... ? |
| 18:54.17 | livesNbox | hey guys -- I am running asterisk at home --- how can I setup one-touch recording for my sip extensions ? |
| 18:54.29 | Qwell[] | livesNbox: see the channel topic |
| 18:54.30 | PakiPenguin | how do you guys rate broadvoice |
| 18:54.35 | salviadud | i don't hate it cause it sucks, i hate it cause it's too much |
| 18:54.39 | Qwell[] | PakiPenguin: when it's up...5 |
| 18:54.50 | livesNbox | Qwell[], this is a FreePBX issue ? |
| 18:54.58 | Qwell[] | livesNbox: yes |
| 18:54.58 | salviadud | the handytone is practical and simple |
| 18:55.07 | PakiPenguin | it goes down a lot? |
| 18:55.07 | livesNbox | I thought this was just standard asterisk configuration. |
| 18:55.26 | Qwell[] | livesNbox: not if you're using AMP/freePBX |
| 18:55.29 | salviadud | the sipura seems to be created for a more "tight-ass" environment |
| 18:55.37 | PakiPenguin | Qwell[], which service do you use? |
| 18:56.07 | salviadud | the thing i dislike the most is the dialplan on the sipura, the fact that it actually has one, because it needs to match *'s dialplan |
| 18:56.25 | salviadud | and if anyone cares... i recommend you buy a handytone |
| 18:56.26 | Qwell[] | I love it when phones in the office randomly ring |
| 18:56.32 | Qwell[] | phones without users |
| 18:56.46 | salviadud | yet, if you wanna get tight with the employees, buy a sipura |
| 18:56.56 | livesNbox | Qwell[], can you give me advice on how i would do it if I wasn't running AMP? I'm sure I can figure it out from there.. |
| 18:57.04 | livesNbox | what needs to go in what config file.. |
| 18:57.07 | *** join/#asterisk backblue (n=moo@87-196-72-98.net.novis.pt) |
| 18:57.11 | Qwell[] | livesNbox: features.conf |
| 18:57.22 | *** join/#asterisk octothorpe (n=octothor@unaffiliated/octothorpe) |
| 18:57.25 | salviadud | is amp recommended? |
| 18:57.27 | livesNbox | i have automon => *1 |
| 18:57.28 | salviadud | i only use the console |
| 18:57.34 | salviadud | and um... no problems here |
| 18:57.38 | docelm0 | AMP SUCKS! |
| 18:57.38 | livesNbox | salviadud, I don't think it matters |
| 18:57.46 | *** join/#asterisk bkw__ (n=brian@adsl-70-142-62-199.dsl.tul2ok.sbcglobal.net) |
| 18:57.55 | rpm | Executing Festival("Zap/4-1", "Domo Arigato, Mr. Roboto, Domo Arigato, Mr. Roboto") in new stack |
| 18:57.58 | rpm | I love it! |
| 18:58.06 | Qwell[] | rpm: use lpc10 |
| 18:58.36 | salviadud | rpm, you got iaxtel, or FWD? |
| 18:58.49 | salviadud | so i can call you, and be greeted by festival? |
| 18:58.56 | salviadud | i've never heard that thing... |
| 18:59.01 | livesNbox | Qwell[], ok is that all I need in features.conf? when I dial *1 during a call it doesn't do anything but pass it along to the caller |
| 18:59.06 | salviadud | is it worth a try? |
| 18:59.15 | Qwell[] | livesNbox: show application dial - w or W |
| 18:59.25 | *** join/#asterisk Utah_Dave (n=boucha@0-1pool149-179.nas31.salt-lake-city1.ut.us.da.qwest.net) |
| 18:59.52 | rpm | fwd 712906 should work |
| 19:00.00 | salviadud | yeah |
| 19:00.28 | livesNbox | Qwell[], ok I see that -- so I need to add Ww to my dial command ? |
| 19:00.36 | livesNbox | now I just have to find my Dial command :) |
| 19:00.51 | Qwell[] | livesNbox: AMP will overwrite it |
| 19:00.57 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 19:01.07 | salviadud | hahaha |
| 19:01.07 | livesNbox | Qwell[], perhaps it's in mySQL ? |
| 19:01.14 | salviadud | rpm, that's pretty good |
| 19:01.23 | Qwell[] | livesNbox: #freepbx |
| 19:01.27 | salviadud | sounds kinda like stephen hawking's voice |
| 19:01.28 | rpm | heh :) |
| 19:02.35 | cryptnix | hmm, i have some iax s100-fx's connected via IAX to my asterisk box whenever i try to communicate between extensions it rings and goes to VM without even ringing the phones ... i am using the dummy gui with asterisk@home any idea |
| 19:03.07 | Qwell[] | cryptnix: See channel topic... |
| 19:05.52 | cryptnix | heh |
| 19:05.55 | *** join/#asterisk inv_Arp (i=junya@adsl-153-242-128.mia.bellsouth.net) |
| 19:06.52 | *** join/#asterisk wunderkin (i=wunderki@69.26.192.234) |
| 19:08.28 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-96-67.rockynet.com) |
| 19:09.08 | *** join/#asterisk ACiDV (n=acidv@modemcable247.11-37-24.mc.videotron.ca) |
| 19:09.15 | *** join/#asterisk heka (n=heka@82.114.68.126) |
| 19:09.20 | [TK]D-Fender | Qwell : See the tremendous payout of my having asked russellb for that? :) |
| 19:09.29 | Qwell[] | [TK]D-Fender: indeed |
| 19:11.00 | heka | Hello, I got a remote machine having about 100ms latency, using g729 codec, but the voice quality is not so good. any body can help me about the problem that is cousing bad voice quality? |
| 19:11.14 | ACiDV | Hi, I have a server without Zaptel cards, without Meetme, only SIP/IAX channels, server load is 5-10%CPU use and suddently go to 99% and doesnt lower unless I restart asterisk. Try with 1.2.4, 1.2.5, SVN 1.2 and this morning with 1.2.6. |
| 19:11.14 | ManxPower | heka, what is the jitter? |
| 19:11.55 | ACiDV | Any idea on how to check what can "eat" all CPU ? show modules doesnt show anything anormal, sip channels, etc all are ok (~20-30 SIP/IAX channels) |
| 19:11.55 | heka | ManxPower: well! I dont know much about jitter! an explanation would be good! |
| 19:12.22 | ManxPower | jitter is the VARIENCE in how long it takes packets to arrive. latency has nothing to do with voice quality |
| 19:12.31 | ManxPower | ~jitter |
| 19:12.32 | jbot | rumour has it, jitter is at http://www.handhelds.org/z/wiki/Kernel%20Documentation |
| 19:12.48 | *** join/#asterisk medusaXX (n=medusaxx@p54A983D4.dip0.t-ipconnect.de) |
| 19:12.53 | heka | ManxPower: and how can I check that? |
| 19:13.04 | ManxPower | how did you check the latency/ |
| 19:13.05 | tzanger | heka: are you using SIP or IAX2? |
| 19:13.34 | heka | tzafrir: Im using sip |
| 19:13.42 | heka | ManxPower: using ping |
| 19:13.51 | ManxPower | rtt min/avg/max/mdev = 887.062/1260.977/1495.336/217.566 ms, pipe 2 |
| 19:14.05 | ManxPower | as you can see the jitter is the difference between min and max |
| 19:14.10 | ManxPower | so AROUND 216ms |
| 19:14.34 | ManxPower | of course ping does not use RTP, which is what your audio uses, so your ping numbers may not be valid for RTP. |
| 19:15.03 | TinoW | use udping ;) |
| 19:16.37 | ACiDV | and my asterisk doesnt run with -p ... does exist tools to trace what loop in * ? It's not so funny to restart * at each hour :P |
| 19:17.47 | heka | ManxPower: is there any way to minimize the jitter? |
| 19:18.21 | jaike | heka: are u sharing bandwidth with other applications? |
| 19:18.46 | heka | jaike: no! using only asterisk, apache and mysql |
| 19:19.02 | heka | all for asterisk needs |
| 19:19.53 | ManxPower | heka, jitter is a function of the network. |
| 19:20.01 | ManxPower | fix the network, you'll get rid of jitter |
| 19:20.06 | Nugget | here jitter is a function of how strong I make the coffee. |
| 19:20.37 | heka | :) |
| 19:20.43 | tzanger | Nugget: hahaha |
| 19:20.49 | ManxPower | speaking of coffee..... |
| 19:20.53 | Z0m81e | Does anyone have experience of SPA3000 ? |
| 19:21.04 | ManxPower | On the phone with my former bank right now, I need something stronger than coffee |
| 19:21.18 | *** join/#asterisk redondos (n=redondos@190.48.62.91) |
| 19:22.07 | [TK]D-Fender | ManxPower : Looks like a job for Jolt Cola or Rev (added bonus of "stiff drink" attached) |
| 19:22.46 | _Paulo_ | get some guarana... |
| 19:22.56 | [TK]D-Fender | _Paulo_ : Rev has that covered :) |
| 19:24.44 | _Paulo_ | Here in Brazil you can buy guarana as a powder, like the indians use it. Strong stuff. |
| 19:24.52 | jbalcomb | hey, why does my phone default to RTP port 5004 but asterisk feaults to 10,000 thru 20,000? |
| 19:24.56 | *** join/#asterisk ruza (n=ruza@81.0.238.58) |
| 19:25.30 | Primer | guaraná rules |
| 19:25.52 | justinu | what does it do _Paulo_? |
| 19:26.10 | Primer | it makes you amped |
| 19:26.14 | [TK]D-Fender | justinu : Just like caffeine on non-regulated. |
| 19:26.19 | _Paulo_ | its 4 times stronger than cafeine, almost the same efects. |
| 19:26.34 | justinu | ok |
| 19:26.47 | justinu | no cool hallucinations tho? |
| 19:27.12 | _Paulo_ | no, its like amphetamines, make you sharp. |
| 19:28.00 | justinu | k |
| 19:28.13 | _Paulo_ | gives insomnia. |
| 19:28.18 | Primer | I recall having my head tingle once when I chewed on two dried guaraná berries |
| 19:28.41 | salviadud | paulo! finally somebody from south america i can relate to |
| 19:29.00 | salviadud | guarana is not that good, i prefer jalapeños |
| 19:29.04 | salviadud | that makes me wake up |
| 19:30.17 | lzhang | hey guys, I've got a polycom 300 that keeps ringing while it's on call, even though there is no other call coming in... what might cause that? |
| 19:30.34 | lzhang | the ringing is coming through the headset/handset |
| 19:30.52 | salviadud | i've never even seen a polycom... |
| 19:30.56 | *** join/#asterisk octothorpe (n=octothor@unaffiliated/octothorpe) |
| 19:31.15 | salviadud | poor lil' o' me |
| 19:31.21 | _Paulo_ | salviadud, I love peppers, I have vases with 4 varieties. |
| 19:31.23 | lzhang | hehe |
| 19:31.23 | *** join/#asterisk rob314 (n=root@207.58.194.55) |
| 19:31.47 | salviadud | paulo, you got free world dialup? |
| 19:31.54 | salviadud | or maybe iaxtel? |
| 19:32.57 | _Paulo_ | no, how does this free world dialup works? |
| 19:33.17 | salviadud | well, you go to www.freeworldialup.com |
| 19:33.25 | salviadud | then you subscribe for free |
| 19:33.37 | salviadud | you play around with asterisk |
| 19:33.47 | *** join/#asterisk sambal (n=ivo@sd5116ceb.adsl.wanadoo.nl) |
| 19:33.48 | salviadud | and you can connect via SIP or IAX2 |
| 19:33.52 | salviadud | i recommend IAX2 |
| 19:33.57 | sambal | hi, how can i count the number of characters / digits in a variable? |
| 19:34.09 | salviadud | then we can talk to each other via FWD |
| 19:34.18 | salviadud | it's a community |
| 19:34.20 | salviadud | basically |
| 19:36.14 | _Paulo_ | ok, checking out |
| 19:37.01 | *** join/#asterisk Defraz (n=t0tal@tim.mychoice.cc) |
| 19:37.05 | *** join/#asterisk trbldwine (n=trbldwin@adam.ur.northwestern.edu) |
| 19:39.49 | *** join/#asterisk eipi (n=eipi@OL17-54.fibertel.com.ar) |
| 19:39.52 | eipi | hi |
| 19:40.13 | *** join/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com) |
| 19:40.22 | qseek | hello all |
| 19:40.39 | eipi | im trying to compile asterisk 1.2.6 with spandsp and i get Makefile:110: *** missing separator. |
| 19:40.56 | eipi | how i can debug what's happening? |
| 19:41.17 | ACiDV | eipi... look around line 110... you have something wrong with your patch ... |
| 19:41.20 | jbalcomb | [TK]D-Fender do i need to change the port settings in rtp.conf because my gxp-2000s are set to RTP port 5004? |
| 19:41.39 | Qwell[] | jbalcomb: yes |
| 19:42.24 | qseek | hi jbalcomb |
| 19:42.27 | sambal | hi, how can i count the number of characters / digits in a variable? |
| 19:42.31 | [TK]D-Fender | jbalcomb : No, just throw out the GXP's and buy more Polycom! NEXT!!!! (c) BKW |
| 19:42.33 | jbalcomb | Qwell[] whats a good default range or does it really matter? |
| 19:42.48 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
| 19:42.48 | jbalcomb | [TK]D-Fender what port does the ip501 use for rtp by default? |
| 19:42.58 | [TK]D-Fender | jbalcomb : Port is variable in RTP |
| 19:43.00 | jbalcomb | qseek hey, hows that project going? |
| 19:43.07 | [TK]D-Fender | thats why there is a range |
| 19:43.17 | qseek | jbalcomb: getting there slowly found a PRI with local loop for 630 |
| 19:43.25 | jbalcomb | [TK]D-Fender recommend me a good range to set asterisk to in rtp.conf please |
| 19:43.26 | _Paulo_ | salviadud, Congratulations Paulo Scardine, you are registered with FWD Number: 759683 |
| 19:43.41 | [TK]D-Fender | jbalcomb : The default is usually good (10000-20000) |
| 19:43.53 | jbalcomb | [TK]D-Fender ip501 default to inside that range? |
| 19:44.06 | salviadud | good paulo |
| 19:44.06 | *** join/#asterisk tainted_ (n=identd@ppp-71-134-51-75.dsl.irvnca.pacbell.net) |
| 19:44.10 | jeffgus | i'm looking to use an Adit 600 with asterisk, but have a question about the fxs ports |
| 19:44.11 | *** part/#asterisk Utah_Dave (n=boucha@0-1pool149-179.nas31.salt-lake-city1.ut.us.da.qwest.net) |
| 19:44.21 | jeffgus | does anyone have docs on the adit 600 fxs ports? |
| 19:44.30 | salviadud | let me give you a nice tip for your iax.conf and extensions.conf |
| 19:44.31 | salviadud | wait... |
| 19:44.36 | Z0m81e | Is anyone home with SPA3k experience? mine is still giving me abuse |
| 19:44.47 | jeffgus | i'd like to know if the fxs ports can generate a MWI using FSK for phones that support that |
| 19:45.01 | jeffgus | Z0m81e, what kind of abuse? |
| 19:45.08 | eipi | any wip300 user? |
| 19:45.12 | *** join/#asterisk [hC] (n=hardcore@S01060004e21ea953.vc.shawcable.net) |
| 19:45.18 | jeffgus | Z0m81e, i just starting playing with one, but haven't tweaked it yet |
| 19:45.29 | Z0m81e | jeffgus, at the mo I can't get a dialtone on the phone, its a pretty standard uk cordless dect phone |
| 19:45.39 | jeffgus | hmm |
| 19:45.41 | jeffgus | strange |
| 19:45.50 | jeffgus | that worked right out of the box |
| 19:45.57 | jeffgus | factory defaults |
| 19:46.13 | Z0m81e | i've tried **** and that does nothing either, unfortunately this one is borrowed so someone may have messed it around, can you factory reset them without using the IVR? |
| 19:46.15 | ACiDV | How I can check why Asterisk eat 99% of my CPU ? normally it use 5-10% then suddently raise to 99% and never lower... |
| 19:46.31 | Qwell[] | ACiDV: gdb should be able to help |
| 19:47.04 | ACiDV | I must restart * with gdb or I can trace running process ? |
| 19:47.24 | Qwell[] | ACiDV: I should know the answer to that, but I don' |
| 19:47.25 | Qwell[] | t |
| 19:47.29 | qseek | jbalcomb: are u going to be connecting directly to the polycom or over a remote connection |
| 19:47.31 | jeffgus | Z0m81e, i don't see a factory reset item on it's built-in web pages |
| 19:47.44 | ACiDV | :P ok will a man gdb :P |
| 19:47.51 | Z0m81e | jeffgus, me either but i did think I saw something about a factory reset url |
| 19:47.51 | *** join/#asterisk bkw__ (n=brian@adsl-70-142-62-199.dsl.tul2ok.sbcglobal.net) |
| 19:47.51 | qseek | jbalcomb because if you r just connecting directly to the asterisk box. you dont need to change it |
| 19:48.11 | jeffgus | Z0m81e, i haven't tested to see if line enable setting causes the tone to die |
| 19:48.28 | *** join/#asterisk terrapen (n=cjs@166.70.183.108) |
| 19:49.09 | [av]bani | anyone here knowledgable about asterisk queues? |
| 19:49.35 | *** part/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com) |
| 19:49.39 | Z0m81e | jeffgus, it seems ok with the pstn, it says it has a line voltage (-48v!) and when i ring it from my mobile the state changes to ringing, but there is nothing on the phone |
| 19:49.59 | *** join/#asterisk dzlabing (n=dzlabing@62.116.84.64) |
| 19:50.08 | *** join/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com) |
| 19:51.09 | rob314 | hello room |
| 19:52.33 | Z0m81e | i may have to take it to work tomorrow and try it with a non-dect phone tho people on the net seem to say they use dect |
| 19:53.20 | TinoW | nortel? hehe. |
| 19:53.23 | sambal | hi, how can i count the number of characters / digits in a variable? |
| 19:53.28 | *** part/#asterisk rob314 (n=root@207.58.194.55) |
| 19:55.10 | [TK]D-Fender | sambal : LEN |
| 19:55.14 | *** join/#asterisk octothorpe (n=octothor@unaffiliated/octothorpe) |
| 19:56.10 | *** part/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com) |
| 19:57.03 | dzlabing | has anyone a simple application which collects input via dtmf und puts the result into a csv-file (i need to collect results from a measurements this way, the interface should be a simple mobile phone. should somehow be possible via AGI, but is the any example for this? |
| 19:58.40 | rpm | [2006-03-27 12:53:45] AGI Tx >> 510 Invalid or unknown command |
| 19:58.40 | rpm | [2006-03-27 12:53:46] AGI Rx << STREAM FILE /var/lib/asterisk/festivalcache/c4ca4238a0b923820dcc509a6f75849b "" |
| 19:58.47 | rpm | booo. my AGI script sucks |
| 19:59.20 | rpm | is "stream file" case-sensitive? |
| 19:59.46 | *** join/#asterisk mtaht_ (n=mtaht@c-71-198-23-124.hsd1.ca.comcast.net) |
| 20:01.51 | jeffgus | Z0m81e, if you pull the power on the spa |
| 20:01.54 | jeffgus | then it'll ring the phone |
| 20:02.30 | jeffgus | Z0m81e, a relay closes and connects the pots port directly to the pstn |
| 20:02.39 | jeffgus | once you plug it in |
| 20:02.48 | jeffgus | the two are seperate ports |
| 20:03.02 | Z0m81e | Sorry, if i pull the power the phone will ring? |
| 20:03.10 | *** join/#asterisk websae (n=websae@h69-129-132-18.69-129.unk.tds.net) |
| 20:03.17 | zapa | hi all does anybody know another freee tool for agents reporter like Asteriskguru Queue Statistics ? |
| 20:03.29 | jeffgus | Z0m81e, if you pull the power to the spa |
| 20:03.39 | jeffgus | Z0m81e, if you call the pstn line, the phone should ring |
| 20:03.50 | Z0m81e | hmm, hold on will have to go downstairs, brb |
| 20:03.52 | jeffgus | Z0m81e, and if you pick up the phone you will get a dial tone from the pstn port |
| 20:04.19 | jeffgus | if the power to the spa is plugged in, then it treats the ports seperately |
| 20:04.19 | *** join/#asterisk kardecallan (n=kardecal@ns1.pcma.com.br) |
| 20:04.50 | *** join/#asterisk tzafrir_laptop (n=tzafrir@88.153.133.128) |
| 20:05.15 | *** join/#asterisk Cybertoy (n=maxim@dsl254-123-241.nyc1.dsl.speakeasy.net) |
| 20:06.03 | jeffgus | Z0m81e, dialing 73738 (RESET) on the POTS port is supposed to reset to factory defaults (make sure power is on) |
| 20:06.25 | websae | i am curious does anyone here have a call center or have high call volume each month? how is your reliability with your trunks? |
| 20:07.15 | kardecallan | _Paulo_ do have you use Asterisk@home? |
| 20:07.31 | _Paulo_ | kardecallan, I have debian |
| 20:07.32 | rpm | does anyone have any AGI scripts which work? |
| 20:08.01 | _Paulo_ | rpm, I have some writen in perl. |
| 20:08.17 | kardecallan | _Paulo_ ok |
| 20:08.17 | rpm | do you use any which use the $AGI->stream_file() function? |
| 20:08.22 | kardecallan | thanks, |
| 20:08.51 | *** part/#asterisk kardecallan (n=kardecal@ns1.pcma.com.br) |
| 20:09.11 | _Paulo_ | rpm yepz, all them. |
| 20:09.53 | rpm | how do you execute them.. im getting error code 510 |
| 20:12.03 | Z0m81e | jeffgus, with no combination of power plugged in or not plugged in to I get a dialtone on the phone or does the phone ring if I call the line??? |
| 20:12.46 | jeffgus | Z0m81e, both |
| 20:12.50 | jeffgus | Z0m81e, with no power |
| 20:13.06 | jeffgus | Z0m81e, the spa connects the fxo to the fxs port |
| 20:13.14 | jeffgus | Z0m81e, we're talking about the 3000, right? |
| 20:13.26 | Z0m81e | yes, but with no power i get nothing |
| 20:13.32 | Qwell[] | well...duh? |
| 20:14.02 | mog_work | i thought spa had hardware relay? |
| 20:14.06 | Qwell[] | probably does |
| 20:14.07 | jeffgus | Z0m81e, disconnect the spa from the wall pstn and connect your phone |
| 20:14.13 | jeffgus | Z0m81e, dial tone? |
| 20:14.18 | Qwell[] | failover is for chumps though :p |
| 20:14.36 | Z0m81e | also, i noticed, I have two leads that go from the master socket to the phone/spa with one lead the phone does not detect a dialtone at all, but with the other it does, it is the other way around with the spa3000 one lead shows line volts of -6v the other shows -48v |
| 20:15.30 | [av]bani | spa3k has a hardware relay |
| 20:15.42 | Z0m81e | yeah, when i power on the 3k i hear the relay click |
| 20:16.05 | [av]bani | if power is lost, it fails over. also, you can make it so if registration is lost it fails over too. |
| 20:16.20 | Z0m81e | the phone works only with the cable that came with it, is is possible it has some dodgy wiring going on? |
| 20:16.30 | [av]bani | no |
| 20:17.09 | *** join/#asterisk MacDome (n=eseidel@A17-255-98-160.apple.com) |
| 20:17.12 | Z0m81e | would you agree -48v sounds like the correct line volts as opposed to -6? |
| 20:19.11 | *** join/#asterisk octothorpe (n=octothor@unaffiliated/octothorpe) |
| 20:21.19 | *** join/#asterisk MacDome (n=eseidel@A17-255-98-160.apple.com) |
| 20:21.55 | eipi | anyone tried to compile 1.2.6 with spandsp? |
| 20:22.10 | eipi | or exactly to execute 1.2.6 with spandsp |
| 20:22.21 | jeffgus | -6 is phone off hook |
| 20:22.24 | [av]bani | spandsp might need to be upgraded to compile for 1.2.6 |
| 20:22.52 | jeffgus | 48 volts is normal phone on hook voltage |
| 20:23.00 | jeffgus | Z0m81e, do you have another phone off hook? |
| 20:23.09 | eipi | spandsp code? by developers, or by me? |
| 20:23.13 | eipi | im using lastest build |
| 20:25.06 | *** join/#asterisk octothorpe (n=octothor@unaffiliated/octothorpe) |
| 20:25.56 | Z0m81e | jeffgus I only have 1 phone, but I have noticed something... The cable that came with the dect phone has 4 pins at both ends, the other cable has one 2 pins at one end....? |
| 20:27.24 | Z0m81e | the 4 pin cable works with the dect phone, and the 2 pin cable works with the spa3000... Maybe the phone is setup with a non standard pin configuration? |
| 20:27.26 | *** join/#asterisk asterboy (n=kevin@S010600485480f4be.ed.shawcable.net) |
| 20:27.48 | asterboy | Anyone suggest the most reliable PC hardware to use for *? |
| 20:27.59 | asterboy | Motherboard? |
| 20:28.25 | asterboy | I prefer AMD CPUs and Western Digital HDs |
| 20:28.40 | Qwell[] | western digital? Why? |
| 20:28.45 | _Sam-- | i just got a supermicro dual opteron system, i like it |
| 20:28.52 | eipi | anyone tried to execute 1.2.6 with spandsp 0.2pre25? |
| 20:29.14 | asterboy | cause maxtor sucks and I've never had a problem with WD |
| 20:29.22 | Qwell[] | asterboy: seagate |
| 20:29.25 | _Sam-- | im using fujitsu drives, i like them too. |
| 20:29.28 | asterboy | They are good too. |
| 20:29.34 | Z0m81e | i've had all sorts of crap with maxtor drives, tho i admit their returns service is good :) |
| 20:29.37 | Qwell[] | seagate wins, hands down |
| 20:29.38 | asterboy | ya, never had a problem with fujitsu |
| 20:29.42 | Qwell[] | Z0m81e: not anymore! |
| 20:29.46 | _Sam-- | fujitsu outperforms the seagates now |
| 20:29.50 | _Sam-- | not that it matters |
| 20:29.50 | Qwell[] | They dropped the 3 year warranty |
| 20:29.55 | _Sam-- | fujitsu has a 5 |
| 20:29.59 | Qwell[] | whereas seagate comes with 5 |
| 20:30.02 | Z0m81e | probably because their drives don't last 3 years :) |
| 20:30.04 | *** join/#asterisk viLeR (i=1000@66.128.47.232) |
| 20:30.07 | Qwell[] | Z0m81e: indeed |
| 20:30.09 | *** part/#asterisk lemmy (n=lemmy@developer.g2gui.net) |
| 20:30.17 | asterboy | I'll be putting in 3 TDM cards |
| 20:30.22 | asterboy | 8 lines total |
| 20:30.31 | asterboy | so don't need a dual cpu |
| 20:30.31 | Qwell[] | excessive |
| 20:30.31 | *** join/#asterisk gaspiz (n=gaspiz@86.34.6.164) |
| 20:30.40 | asterboy | excessive? |
| 20:30.44 | _Sam-- | i like it for redundancy |
| 20:30.47 | Qwell[] | That's a lot of interrupt handling |
| 20:31.01 | asterboy | oh ya guess I could use just the 1 card...the new ones they have at digium |
| 20:31.06 | gaspiz | does anyone of you know why the mysql cdr is not working in asterisk 1.2.1? |
| 20:31.22 | Qwell[] | gaspiz: got asterisk-addons installed? |
| 20:31.29 | asterboy | what do you suggest for motherboards? |
| 20:31.31 | backblue | gaspiz: because you dont have hands for it. |
| 20:31.39 | asterboy | ASUS has been good. |
| 20:31.42 | jeffgus | Z0m81e, could be... i don't know much about UK phones |
| 20:31.51 | _Sam-- | supermicro |
| 20:31.54 | backblue | asus it's very good, tyan its the best., |
| 20:32.05 | asterboy | tyan is good too. |
| 20:32.11 | backblue | tyan its the best |
| 20:32.12 | _Paulo_ | supermicro rulez |
| 20:32.23 | backblue | after tyan, you have asus, msi, supermicro... |
| 20:32.28 | backblue | and a couple of them |
| 20:32.33 | _Sam-- | this is the supermicro i have, i love it... <i think its nicer than any tyan>....http://www.supermicro.com/Aplus/motherboard/Opteron/8131/H8DA8.cfm |
| 20:33.31 | backblue | supermicro its far from tyan, and far from asus. |
| 20:33.43 | _Sam-- | to each his own..ive owned them all. |
| 20:33.46 | backblue | supermmicro it's midle range marquet. |
| 20:34.05 | _Sam-- | i think supermicro is more high end than you are acknowledging, but that is fine, its your opinion. |
| 20:34.08 | backblue | i'm not saying it does not fells your needs :D |
| 20:34.52 | backblue | _Sam--: take this clue, why tyan its the only motherboard used in high HIGH end linux clusters? |
| 20:35.04 | backblue | and why it's the BEST motherboard suporting linuxbios? :D |
| 20:35.22 | Qwell[] | and you can't beat serial console, from the bios |
| 20:35.34 | Qwell[] | (yes, I realize other boards do that too) |
| 20:35.36 | *** join/#asterisk paanz (n=Paanz@60.51.180.134) |
| 20:35.38 | _Sam-- | actually, i find there are a bunch of supermicro sclusters. |
| 20:35.40 | octothorpe | yay, mobo wars! |
| 20:35.43 | _Sam-- | tyan isnt the only one |
| 20:35.57 | _Sam-- | but i am not going to argue..its like having a debate with the special olympics...you cant win. |
| 20:35.59 | mtaht_ | I hate the raid on some supermicro mbs |
| 20:36.06 | backblue | _Sam--: yes, but tyan it's in the top, for years, i think you can see that. |
| 20:36.17 | _Sam-- | i disagree |
| 20:36.23 | _Sam-- | but thats fine |
| 20:36.34 | backblue | _Sam--: yeah, just use whatever you like, and do what you need! :D |
| 20:37.20 | _Sam-- | the raid on my supermicro board is adaptec 7902 raid...i dont know whats so bad about that. |
| 20:38.06 | eipi | sorry its not flood, but anyone tried to execute 1.2.6 with spandsp 0.2pre25? |
| 20:39.11 | mtaht_ | the raid on the supermicroboards I have is based on an adaptec (actually, marvel) chipset - and comes with a propriatary driver |
| 20:39.18 | asterboy | If tyan is Linux special...I'll go that route |
| 20:39.37 | backblue | _Sam--: that's god. |
| 20:39.38 | mtaht_ | my attempts to convert any given box to sata_mv have failed. |
| 20:40.06 | backblue | i dont have many experience with sata raid cards |
| 20:40.23 | _Sam-- | i have a tyan with sata on board raid |
| 20:40.24 | backblue | one of this days i build a linux server with 4 raid sata disks |
| 20:40.27 | _Sam-- | i just use linux soft raid instead |
| 20:40.41 | backblue | and raid5 was not avaliable with that sata chipset |
| 20:40.42 | _Sam-- | my supermicro has 8 hot swap U320 raid drives connected to it |
| 20:40.51 | backblue | ICH7 or something like that (asus board) |
| 20:41.12 | backblue | _Sam--: linux software raid its not allways the solution. |
| 20:41.23 | _Sam-- | neither is sata hardware raid. |
| 20:41.23 | backblue | i used raid 5 linux software raid |
| 20:41.37 | backblue | _Sam--: it can be |
| 20:41.39 | _Sam-- | eh i got some work to do, nice arguing with you folks. |
| 20:41.44 | backblue | but you have to have a god controler |
| 20:41.48 | backblue | with a lot of mem |
| 20:41.57 | backblue | _Sam--: you too. |
| 20:42.47 | *** join/#asterisk Strom_M (n=strom@66.159.243.59) |
| 20:43.03 | *** part/#asterisk eipi (n=eipi@OL17-54.fibertel.com.ar) |
| 20:44.45 | *** join/#asterisk gammacoder (n=chatzill@64-132-192-33.gen.twtelecom.net) |
| 20:44.50 | saftsack | i heard from digium that they will release a quadport bricard |
| 20:45.24 | gammacoder | is anyone using grandstream gxp-2000s and dhcp option 2 (timezone)? |
| 20:45.48 | asterboy | Digium has the working on the TDM2400 cards missleading. |
| 20:45.57 | saftsack | missleading? |
| 20:45.58 | asterboy | s/working/wording/g |
| 20:46.04 | brookshire | asterboy: how so? |
| 20:46.08 | gammacoder | i'm failing at getting the timezone override to work |
| 20:46.21 | asterboy | They say they support 6 FXS and FXO modules. |
| 20:46.33 | asterboy | s/modules/interfaces/g |
| 20:46.45 | saftsack | no |
| 20:46.48 | asterboy | but each module supports 4, no? |
| 20:46.55 | saftsack | there was a quad prort isdn card on the cebit |
| 20:46.56 | asterboy | so that will be way more. |
| 20:47.00 | saftsack | with onboard dsp |
| 20:47.17 | backblue | what i want its cheap gsm pci cards, like dual or quad gsm cards. |
| 20:47.17 | asterboy | http://www.digium.com/en/wheretobuy/digiumdirect/productview.php?category_id=18&product_code=TDM2401E |
| 20:47.33 | asterboy | The Wildcard TDM2400P is a full-length PCI 2.2-compliant card that supports up to 6 FXS and FXO station interfaces |
| 20:47.36 | Strom_M | backblue, GSM? |
| 20:47.40 | backblue | Strom_M: yes |
| 20:47.48 | Strom_M | why GSM? |
| 20:47.53 | backblue | i know beronet will do one. |
| 20:47.56 | stoffell | backblue, junghanns is coming with that |
| 20:48.01 | backblue | stoffell: yes |
| 20:48.05 | backblue | i know |
| 20:48.09 | asterboy | makes you think it only supports 6 of each when it should be 6x4 =24 |
| 20:48.22 | brookshire | oh yeah.. that should be 24 |
| 20:48.28 | backblue | Strom_M: because i use it?! sorry dont understand your question. |
| 20:48.28 | brookshire | thanks :D |
| 20:48.30 | *** join/#asterisk astra^^ (n=muhajir_@59.145.104.74) |
| 20:48.45 | Strom_M | backblue, what advantage does a GSM card have over a regular TDM or T1 card? |
| 20:48.52 | Strom_M | how is it different? |
| 20:49.18 | rpm | how much bandwidth does GSM use? |
| 20:49.19 | backblue | you will have to buy tdm or t1 card + gsm bridge |
| 20:49.20 | rpm | 32kbit? |
| 20:49.41 | astra^^ | can anyone tellme wen i place call..it goes failes inthe log it shows failed ..but i ambeen charged.. what might be the problem |
| 20:49.56 | backblue | you can buy IP gsm bridges insted |
| 20:50.09 | Strom_M | backblue, is it a GSM codec card? does it connect to the mobile phone network instead of the regular wireline network? |
| 20:50.30 | backblue | Strom_M: yes, it connects directly to gsm network. |
| 20:50.49 | Strom_M | why the hell would you want to do that? |
| 20:50.50 | *** join/#asterisk angler_ (n=johnb@199.227.185.58) |
| 20:50.51 | backblue | you can do something like a pbx, all inside the box, with gsm bri pri... |
| 20:51.06 | *** join/#asterisk bkw__ (n=brian@adsl-70-142-62-199.dsl.tul2ok.sbcglobal.net) |
| 20:51.07 | *** join/#asterisk starlein (i=star@fo0bar.de) |
| 20:51.23 | backblue | Strom_M: or you are not understands, or i'm not being explicit |
| 20:51.38 | backblue | understanding |
| 20:51.40 | Strom_M | backblue, no, you're not being specific at all. You're just barely answering my questions. |
| 20:52.21 | backblue | Strom_M: if i use a pci card, that connects directly to my gsm network, what its the big deal with this? i'm not understanding. |
| 20:52.21 | astra^^ | can anyone tellme wen i place call..it dosent connect but in the log it shows failed ..but i am been charged.. what might be the problem |
| 20:53.17 | backblue | Strom_M: http://www.junghanns.net/images/quadGSM_big.jpg |
| 20:53.20 | Strom_M | backblue, your computer is stationary. It makes no sense to have it connect to the mobile phone network. Why not provision an ISDN line instead? |
| 20:54.15 | astra^^ | :/ ? |
| 20:54.19 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 20:54.32 | backblue | isdn calls to mobile phones are expencive than with a gsm card to gsm phone? |
| 20:55.11 | Strom_M | backblue, I suppose. GSM audio quality blows donkeys for quarters |
| 20:56.12 | backblue | you can make a remote office solution with you gsm provider, and have unlimited calls in your gsm phones network (about 10 numbers) and just use disa to call every where. |
| 20:56.24 | backblue | Strom_M: easy english please. |
| 20:57.18 | *** join/#asterisk MacDome (n=eseidel@A17-255-98-160.apple.com) |
| 20:58.06 | Strom_M | backblue, "blows donkeys for quarters" means, literally, "performs oral sex on donkeys for a payment of twenty-five cents" - a very colorful way of saying "it's really bad" |
| 20:59.00 | *** part/#asterisk paanz (n=Paanz@60.51.180.134) |
| 20:59.11 | _Sam-- | could be worse, it could blow sheep for pennies |
| 20:59.20 | Strom_M | haha |
| 20:59.30 | Strom_M | _Sam--, that's LPC10 |
| 21:00.18 | [av]bani | \o> |
| 21:00.19 | [av]bani | <o/ |
| 21:00.20 | *** join/#asterisk ToTo (n=ToTo@host38-162.pool875.interbusiness.it) |
| 21:00.30 | [av]bani | lpc10 rocks if you have a 300 baud modem |
| 21:00.52 | _Sam-- | are they worth money yet as a collectors item? |
| 21:00.56 | _Sam-- | a 300 baud modem that is |
| 21:01.25 | _Sam-- | i think i have a new in box hayes 2400 baud |
| 21:01.33 | Cybertoy | I still have an acoustic coppler lying around... |
| 21:02.18 | _Sam-- | hows things, [av]bani |
| 21:03.10 | *** join/#asterisk nagl (n=nagl@vie-086-059-104-148.dsl.sil.at) |
| 21:04.29 | [av]bani | _Sam--: tried 1.1.0.1 yet? |
| 21:04.51 | backblue | i need athlon MP processors |
| 21:05.08 | _Sam-- | i have an dual athon MP server, its just ok |
| 21:05.14 | _Sam-- | barton 2.8's |
| 21:05.33 | _Sam-- | [av]bani: not yet....ive been fantastically busy at work, which is a good thing. |
| 21:05.45 | _Sam-- | had our busiest month on record at kneedraggers going |
| 21:06.20 | _Sam-- | i never even heard of that firmware, too |
| 21:07.12 | backblue | _Sam--: i need just like you. |
| 21:07.19 | backblue | i have a tyan board here |
| 21:07.24 | _Sam-- | model name : AMD Athlon(tm) MP 2800+ |
| 21:07.26 | backblue | and i'm not using it. |
| 21:07.36 | backblue | yes, exacly that ones. |
| 21:07.38 | _Sam-- | ya, this is a tyan |
| 21:07.44 | backblue | MPX? |
| 21:07.46 | _Sam-- | the 2882 or sometihng |
| 21:07.57 | backblue | tiger mpx? |
| 21:08.01 | backblue | i have a tiger mpx |
| 21:08.03 | _Sam-- | its a tiger something |
| 21:08.11 | _Sam-- | its a few years old now |
| 21:08.12 | backblue | yes it should be |
| 21:08.14 | asterboy | I had a coupler type modem |
| 21:08.32 | backblue | yes, but if i find processores, i would use it. |
| 21:08.34 | asterboy | back in the day when Tron the movie was a hit. |
| 21:08.40 | _Sam-- | i dont like the athlon MPs |
| 21:08.41 | [av]bani | athlon mp is ancient |
| 21:08.44 | [av]bani | amd64 baby |
| 21:08.46 | backblue | _Sam--: why? |
| 21:08.57 | _Sam-- | i feel they run hot and are just so/so compared to a xeon |
| 21:09.09 | backblue | [av]bani: who cares? i dont waste so much money. |
| 21:09.15 | _Sam-- | [av]bani " i just got a dual operton machine |
| 21:09.26 | _Sam-- | im scared to run 64bit |
| 21:09.27 | [av]bani | blitzrage: amd64 is now cheaper than athlon mp... |
| 21:09.42 | [av]bani | _Sam--: scared why? our * pbx is amd64 |
| 21:09.44 | [av]bani | works fine |
| 21:09.46 | backblue | i waste money in cars, not in machines! :P that's was in the oldies |
| 21:09.55 | _Sam-- | ive heard bad things about mysql 64 |
| 21:10.00 | [av]bani | s/blitzrage/backblue/ |
| 21:10.03 | _Sam-- | this is for a database server |
| 21:10.12 | [av]bani | eh? mysql works fine |
| 21:10.24 | [av]bani | i've been using mysql 64 for years now |
| 21:10.27 | [av]bani | zero problems |
| 21:10.27 | backblue | [av]bani: to have a dual amd64, how much mone will i waste? |
| 21:10.27 | _Sam-- | there is some specific bug regarding inserts on the |
| 21:10.28 | [av]bani | none |
| 21:10.32 | [av]bani | _Sam--: nope |
| 21:10.34 | backblue | about $500? |
| 21:10.41 | [av]bani | backblue: cheaper than dual athlonmp |
| 21:10.42 | _Sam-- | [av]bani : it was just announced like 5 days ago |
| 21:10.45 | _Sam-- | <PROTECTED> |
| 21:10.48 | [av]bani | lies |
| 21:10.59 | _Sam-- | http://bugs.mysql.com/bug.php?id=8555 |
| 21:11.06 | _Sam-- | er thats not the one |
| 21:11.08 | backblue | [av]bani: i allready have the tyan, just need the fucking processores, how can you say it will be cheaper? :o |
| 21:11.09 | _Sam-- | thats the old one |
| 21:11.41 | _Sam-- | it was from just this month, sec |
| 21:13.14 | _Sam-- | hmmm this is from november... i cant find the one i saw over the weekend: http://lists.debian.org/debian-amd64/2005/11/msg00291.html |
| 21:13.15 | [av]bani | backblue: what's fucking processores? |
| 21:14.31 | [av]bani | _Sam--: glibc bug with NPTL, and specifically debian. other distros not affected |
| 21:14.40 | _Sam-- | im debian |
| 21:14.46 | [av]bani | i use fedora :) |
| 21:14.51 | [av]bani | but also only happens with replication |
| 21:14.56 | _Sam-- | i replicate wildly |
| 21:15.13 | _Sam-- | literally, have about 4 machines replicating various directions |
| 21:15.31 | [av]bani | well, cant blame mysql for debian glibc bugs |
| 21:15.38 | *** join/#asterisk salviadud (n=dude@dsl-201-129-86-188.prod-infinitum.com.mx) |
| 21:15.42 | salviadud | well hello |
| 21:15.44 | _Sam-- | i just think i may stick with a 32bit kernel / setup |
| 21:15.44 | [av]bani | yell at debian to fix the glibc |
| 21:15.50 | _Sam-- | the performance gains arent that great in 64 |
| 21:15.54 | _Sam-- | 10-15%? |
| 21:15.56 | [av]bani | of course debian isnt known for being ... up to date |
| 21:16.09 | [av]bani | it depends, if you want >4g processes then 64 is the only way to go |
| 21:16.30 | _Sam-- | lessee...ps auxww | wc -l |
| 21:16.36 | salviadud | im having some trouble with mixmonitor, i already compiled SoX with mp3 support, and the darn thing gives me an error |
| 21:16.36 | _Sam-- | 123 |
| 21:16.42 | _Sam-- | i think i should be ok |
| 21:17.28 | salviadud | Mar 27 15:19:40 WARNING[26171]: file.c:981 ast_writefile: No such format 'mp3' |
| 21:17.33 | salviadud | that's baloney! |
| 21:17.46 | _Sam-- | salami is a better cut of meat |
| 21:17.54 | salviadud | haha |
| 21:17.56 | [av]bani | ok, app_queue is 100% busted |
| 21:17.59 | Qwell[] | salviadud: asterisk-addons has format_mp3 |
| 21:18.00 | [av]bani | totally |
| 21:18.02 | salviadud | i even recompiled asterisk |
| 21:18.10 | [av]bani | this is not good at all |
| 21:18.21 | salviadud | somegeek, i should compile asterisk-addons too? |
| 21:18.22 | _Sam-- | damn, glad i didnt try it then...we are 100% queue based now |
| 21:18.33 | [av]bani | no... this is app_queue period |
| 21:18.34 | [av]bani | not amd64 |
| 21:18.38 | salviadud | damn, im using xchat |
| 21:18.44 | Qwell[] | [av]bani: What's it doing? |
| 21:18.46 | [av]bani | it is completely and utterly busted |
| 21:18.47 | _Sam-- | i hear ya...i dont use the opterons for the * |
| 21:18.51 | salviadud | Qwell, should i download asterisk-addons |
| 21:18.53 | salviadud | ? |
| 21:18.58 | [av]bani | Qwell[]: it is not sending cancels to extensions |
| 21:19.00 | Qwell[] | salviadud: If you want mp3 support, yes |
| 21:19.03 | salviadud | so i can get mp3 support on mixmonitor |
| 21:19.05 | salviadud | alright |
| 21:19.07 | salviadud | will do |
| 21:19.08 | [av]bani | Qwell[]: thought it was just polycom, but it's borking grandstreams too |
| 21:19.20 | Qwell[] | so fix it :P |
| 21:19.32 | [av]bani | no thanks, no time to fuck around with this broken shit |
| 21:19.35 | [av]bani | back to dial() for me |
| 21:19.39 | _Sam-- | [av]bani : for the record, BLF still breaks my asterisk, but i worked around. |
| 21:20.01 | _Sam-- | if blf is on, the when calls come in, and a grandstream tries to answer it...fast busy on both sides when answered |
| 21:20.03 | *** join/#asterisk groogs (n=greg@d221-73-237.commercial.cgocable.net) |
| 21:20.08 | _Sam-- | i had to switch to an auto attendant to use BLF |
| 21:24.34 | *** join/#asterisk lzhang (n=lewiszha@67.95.13.46) |
| 21:24.37 | harlequin516 | Okay I am trying to test my fwd setup. How can I do this? |
| 21:24.48 | lzhang | if I have a 601 with the sidecar, what do I need to get the sidecar working? |
| 21:25.26 | lzhang | right now I a bunch of names in my "directory" but they are not spilling over into the sidecar list |
| 21:25.35 | harlequin516 | Is there a better way to test incoming calls than the call me thing ? |
| 21:28.23 | Strom_C | harlequin516, whats your fwd number? I can call you |
| 21:28.43 | Cybertoy | harlequin, there's also the "514" extensions which puts you into a conference room |
| 21:28.51 | Cybertoy | hardly anyone there thoguzh |
| 21:29.08 | Cybertoy | actually.. I think the callme button puts you into the same room |
| 21:30.23 | gammacoder | has anyone had luck using the Grandstream gxp-2000's dhcp options to override tfpt server (option 66) or timezone (option 2) ? |
| 21:30.41 | *** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be) |
| 21:32.12 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 21:32.36 | [TK]D-Fender | lzhang : they need to have a speed-dial index. |
| 21:34.43 | lzhang | can anybody point me to where I can mess with the speed dial index? |
| 21:34.43 | *** part/#asterisk dzlabing (n=dzlabing@62.116.84.64) |
| 21:35.19 | Qwell[] | lzhang: my guess is the xml file |
| 21:35.31 | Qwell[] | or whereever you do the directory stuff |
| 21:37.16 | *** part/#asterisk lzhang (n=lewiszha@67.95.13.46) |
| 21:37.19 | *** join/#asterisk lzhang (n=lewiszha@67.95.13.46) |
| 21:37.51 | lzhang | I'm looking around in the (MACADDR)-directory.xml and there is an entry for each contact, as well as an item <sd>1</sd> for each |
| 21:38.06 | lzhang | except each contact has a different number |
| 21:38.16 | Qwell[] | looks like a speeddial index to me |
| 21:38.16 | lzhang | I'm assuming sd stands for speed dial |
| 21:39.20 | lzhang | yea me too, except I've got entries in here for that, but only 5 contacts are showing (on the actual phone) the sidecar is still blank |
| 21:40.50 | lzhang | I wish I could find some documentation on these xml files |
| 21:41.32 | *** join/#asterisk cryptnix (n=andrew@zero.levelsync.com) |
| 21:42.50 | *** join/#asterisk Darwin35 (n=darwin35@sta-208-139-193-162.rockynet.com) |
| 21:44.42 | Darwin35 | http://pastebin.ca/47206 feed back and add fuctions.... |
| 21:44.52 | Darwin35 | major dialplan |
| 21:45.14 | Darwin35 | thus far it works |
| 21:45.25 | Darwin35 | total rewrite. |
| 21:45.43 | jarrod | what is the method of dialing a number, after connection having asterisk send more dtmf digits... |
| 21:46.07 | Grizzy | is there an include -file- syntax element for asterisk? |
| 21:46.19 | jarrod | #include "file.conf" |
| 21:46.28 | Grizzy | thanks. |
| 21:47.04 | ManxPower | jarrod, "show application dial" |
| 21:48.48 | jarrod | ah D([]) |
| 21:48.49 | jarrod | nice! |
| 21:49.54 | Grizzy | And I thought I was a command-line junkie; everything in asterisk seems to be in it's CLI. |
| 21:50.15 | Grizzy | I still want .asteriskrc |
| 21:50.50 | Darwin35 | dial needs to be replaced with lcdial |
| 21:52.42 | Grizzy | Arbitrary tone detection with duration wouild be nicee. |
| 21:52.50 | ManxPower | jarrod, just remember, analog FXO ports are considered answered as soon at the number is done dialing |
| 21:52.55 | *** join/#asterisk tuxinator_linux (n=tuxinato@netblock-68-183-112-100.dslextreme.com) |
| 21:53.51 | Darwin35 | man rewriting dialplans if fun but gives you a headack |
| 21:53.54 | jarrod | i dont see a way to insert pauses |
| 21:53.56 | jarrod | which kinda sux |
| 21:54.05 | Darwin35 | wait() |
| 21:54.09 | Qwell[] | w |
| 21:54.14 | Darwin35 | or w |
| 21:54.27 | Qwell[] | Dial(123w456w789) |
| 21:54.43 | *** join/#asterisk salviadud (n=dude@dsl-201-129-86-188.prod-infinitum.com.mx) |
| 21:54.46 | Darwin35 | ahh ok |
| 21:54.47 | ManxPower | Qwell, you know that only works on analog fxo ports, for all other types of ports, you need D() |
| 21:54.53 | Qwell[] | Dial(18005551212w*12345#) |
| 21:55.09 | Qwell[] | ManxPower: of course |
| 21:55.10 | salviadud | guys, i just want to mention that MixMonitor does not work with mp3 :( |
| 21:55.16 | salviadud | sadly, it is not yet implemented |
| 21:55.28 | salviadud | the addon is so asterisk can "play" mp3's, not record |
| 21:56.00 | salviadud | could i put that as a wishlist of some sort? |
| 21:56.22 | salviadud | well, in C, at least |
| 21:59.11 | salviadud | well, i'll record in wav i guess... |
| 22:01.17 | *** part/#asterisk jaike (n=a@203.131.137.76) |
| 22:06.29 | lzhang | my sidecar on my 601 is blank, and the lights are flickering between green and red... is that supposed to indicate anything? |
| 22:06.54 | asterboy | Any comments on the # of interrupts you should max out on a motherboard in an * install? |
| 22:07.09 | asterboy | I'd like to save a buck and go with 3 TDM400 cards. |
| 22:07.21 | asterboy | This board looks sweet: http://tyan.com/products/html/tomcatk8s.html |
| 22:07.22 | mog_work | get 2400p ^_^ |
| 22:07.28 | Qwell[] | mog_work: That's what I said |
| 22:07.35 | asterboy | mog_work, love to but they cost. |
| 22:07.36 | mog_work | shebus that is a board asterboy |
| 22:07.39 | Qwell[] | at 3 TDM400p's, you're at about the same price |
| 22:07.52 | mog_work | in fact i think it same cost Qwell[] |
| 22:07.55 | mog_work | but /me is not in sales |
| 22:08.09 | Qwell[] | pfft, only 6 pci slots? |
| 22:08.15 | asterboy | lol |
| 22:08.38 | Qwell[] | friend of mine is gonna do an install on a box that used to be a windows pbx... |
| 22:08.38 | Qwell[] | SIXTEEN pci slots |
| 22:08.42 | mog_work | JESUS |
| 22:08.47 | mog_work | how can it have that many slots |
| 22:08.48 | asterboy | never heard of such a beast! |
| 22:08.53 | mog_work | pics? |
| 22:08.59 | Qwell[] | he said it might be 24. he'd have to check |
| 22:09.00 | asterboy | who makes that board? |
| 22:09.02 | mog_work | or im calling bs |
| 22:09.03 | Qwell[] | no clue |
| 22:09.09 | Qwell[] | I'll ask him for pics tonight |
| 22:09.16 | harlequin516 | When I connect a call through free world dialup does the data go through their server, or is it just a directory service? |
| 22:09.19 | Qwell[] | he said it was "purpose built" for the old pbx |
| 22:09.43 | harlequin516 | I mean does the voice data actually pass through their servers? |
| 22:10.03 | eric_hill | http://www.mobl.com/expansion/products/pcie_expansion/6slot/index.html |
| 22:10.38 | eric_hill | Er, sorry... http://www.mobl.com/expansion/products/pci_expansion/P13RR-TEL/index.html |
| 22:10.48 | mog_work | that doesnt count thats cheating ^_^ |
| 22:10.48 | Qwell[] | eric_hill: it may be something like that... |
| 22:10.58 | mog_work | has to be all one board |
| 22:11.21 | asterboy | that's a great link. |
| 22:11.21 | eric_hill | That's pretty common in the industrial world I'm used to - many many digital and analog IO ports. |
| 22:11.33 | harlequin516 | Anyone know how IAX routes voice data? |
| 22:11.47 | mog_work | same way as data data harlequin516 |
| 22:11.49 | mog_work | its all udp |
| 22:11.54 | mog_work | one stream |
| 22:12.08 | *** join/#asterisk xp_prg (n=anonymou@67-102-228-17.adsl.lbdsl.net) |
| 22:13.01 | mog_work | besides they amx out at 13 |
| 22:13.09 | harlequin516 | So when a call is routed through free world dialup using IAX, is the data passing through freeworld dialup (Are they fitting a bill for my vaoice bandwidth)? |
| 22:13.13 | Qwell[] | mog_work: daisy chain! |
| 22:13.15 | Qwell[] | :D |
| 22:13.17 | mog_work | lol |
| 22:13.22 | *** join/#asterisk MacDome (n=eseidel@A17-255-98-160.apple.com) |
| 22:13.26 | Qwell[] | or, get the board above, and get 6 of those beasties |
| 22:13.28 | mog_work | possibly harlequin516 |
| 22:13.34 | mog_work | you can do reinvite |
| 22:13.42 | Qwell[] | mmm... |
| 22:13.48 | Qwell[] | 78 transcoder boards |
| 22:13.56 | mog_work | shebus |
| 22:14.00 | Qwell[] | :p |
| 22:14.03 | mog_work | thats alot of g729 to ulaw |
| 22:14.08 | Qwell[] | or 78 gbit NICs? |
| 22:14.14 | Qwell[] | half and half? |
| 22:14.18 | harlequin516 | Reinvite will rediect the asterisk server to go point-to-point instead of their party routed? |
| 22:14.20 | Qwell[] | or whatever the ratio needs to be, heh |
| 22:14.26 | mog_work | yes harlequin516 |
| 22:14.47 | harlequin516 | ok must lookup reinvite, thanks |
| 22:16.47 | Netgeeks | Hey folks |
| 22:17.39 | Netgeeks | Is there a dialplan accessible variable that is available after a call has been hungup (in the h extension or if my dial command allows continuation) that gives me the total connect time? |
| 22:17.42 | *** join/#asterisk subdolus (n=subby@subby.afraid.org) |
| 22:18.15 | *** join/#asterisk MacDome (n=eseidel@A17-255-98-160.apple.com) |
| 22:19.15 | *** join/#asterisk brc__ (n=brian@pdpc/supporter/basic/brc) |
| 22:19.21 | nDuff | Netgeeks: You could probably store the time when a call starts in a channel variable, and work from there. |
| 22:19.44 | nDuff | ...presuming there's not a better solution, of course. |
| 22:19.58 | Qwell[] | DIALEDTIME, ANSWEREDTIME, etc |
| 22:22.09 | *** join/#asterisk brettnem (n=brettnem@nemeroff.com) |
| 22:22.23 | *** join/#asterisk brettnem (n=brettnem@nemeroff.com) |
| 22:24.58 | Netgeeks | in 1.2 there is a CDR function that supposable you can call like CDR(billsec). Ever used that Qwell? |
| 22:26.07 | Qwell[] | no |
| 22:29.48 | *** part/#asterisk Cybertoy (n=maxim@dsl254-123-241.nyc1.dsl.speakeasy.net) |
| 22:30.27 | lzhang | doesn't CDR automatically store billable seconds? |
| 22:30.54 | websae | has anyone here done high call volume or do high call volume like in charge of call centers or anything like that? |
| 22:31.38 | zapa | :) i recive a lot call from a radio station is the oposite |
| 22:33.56 | websae | how does that work |
| 22:34.00 | websae | why ? |
| 22:35.25 | *** join/#asterisk delta34ooo (n=delta34o@global-sf.keen.com) |
| 22:35.59 | websae | anyone else have experience with high call volume? |
| 22:36.19 | Qwell[] | websae: When I call Sprint or Adelphia |
| 22:36.44 | websae | haha |
| 22:36.53 | websae | or broadvoice |
| 22:38.10 | *** join/#asterisk tehdely (n=delysiid@home.teambarry.org) |
| 22:38.43 | zapa | works well but, i need to ask help to the carrier because the E1 Lines get congetion |
| 22:38.47 | zapa | tu fast |
| 22:40.30 | [av]bani | ok... when asterisk dials multiple extensions at once (eg dial(sip/4000&sip/4001) it sometimes drops cancels |
| 22:40.33 | [av]bani | sux... |
| 22:41.04 | asterboy | digium.org is down? |
| 22:41.22 | delta34ooo | can someone clearify questions regarding moh native for 1.2 release? |
| 22:41.24 | mog_work | digium.com |
| 22:41.26 | mog_work | asterisk.org |
| 22:41.45 | *** join/#asterisk LapTop006 (n=laptop00@sparc006.chriskaine.com.au) |
| 22:41.50 | asterboy | ah asterisk.org |
| 22:42.41 | *** join/#asterisk angler_ (n=johnb@199.227.185.58) |
| 22:44.05 | delta34ooo | so in 1.2 for moh i dont need to use mpg123, if i want to play from files, what format does my music files need to be raw, gsm, mp3s? |
| 22:44.21 | asterboy | ok, already won an * sale today. |
| 22:44.31 | asterboy | Now I'm going for a bigger fish. |
| 22:44.47 | asterboy | Here is an email I'm compiling to try and win a big contract. |
| 22:44.53 | asterboy | http://pastebin.ca/47211 |
| 22:45.11 | asterboy | If you guys can add to that, be greatly apprecieated. |
| 22:45.39 | justinu | based on DARPA? that's creative... |
| 22:45.45 | asterboy | :) |
| 22:46.01 | mog_work | <PROTECTED> |
| 22:46.03 | mog_work | ? |
| 22:46.17 | mog_work | lots of mil. orgs and other gov orgs use asterisk |
| 22:46.32 | mog_work | but what do you mean darpa |
| 22:46.50 | eric_hill | Probably should say "based on DARPA research"... |
| 22:47.06 | mog_work | how is asterisk based on darpa research |
| 22:47.07 | eric_hill | Or maybe "paid for with DARPA funding"? |
| 22:47.11 | *** join/#asterisk |omni| (n=rob@c-67-185-96-86.hsd1.wa.comcast.net) |
| 22:47.12 | mog_work | or am i missing something |
| 22:47.21 | Qwell[] | or "uses the Internet" |
| 22:47.29 | eric_hill | It's not asterisk - it's the darpa funding for the SELinux distro |
| 22:47.31 | mog_work | but it doenst have to |
| 22:47.32 | asterboy | no, that more focuses on the *nix part |
| 22:47.34 | mog_work | ahh |
| 22:47.43 | eric_hill | Asterisk on SELinux == secured by DARPA funding :) |
| 22:47.48 | asterboy | lol |
| 22:47.55 | mog_work | i thought nsa did selinux |
| 22:48.06 | mog_work | or was it dod |
| 22:48.26 | eric_hill | Got me - give me debian. Just Works (TM) |
| 22:48.34 | _Paulo_ | I think you should add: No vendor lockin |
| 22:48.46 | mog_work | debian can be selinux distro i thought? |
| 22:48.57 | mog_work | selinux is only a set of specs |
| 22:49.17 | asterboy | no venfor lockin is a good one, basically non-proprietary. |
| 22:49.26 | asterboy | s/venfor/vendor |
| 22:49.28 | eric_hill | Really? I didn't know that. apt-get install super-dod-darpa-funded-linux-settings |
| 22:49.33 | eric_hill | Darn. didn't work. |
| 22:49.38 | asterboy | lol |
| 22:49.39 | Katty | eric_hill: everytime i read your /nick i read eric_clapton |
| 22:50.02 | eric_hill | If I could have his bank account, I'd change my name... |
| 22:50.02 | mog_work | heh its not quite that simple |
| 22:50.08 | asterboy | I use lfs so not rpm, no apt-get, no merge world |
| 22:50.43 | Nugget | I use slackware but I treat it like lfs. |
| 22:51.08 | Nugget | my ideal linux is the smallest, tiniest, least linux thing I can install and move past, that will stay out of my way as much as possible. |
| 22:51.17 | Nugget | hee |
| 22:51.21 | Qwell[] | I use Asterisk Complete :p |
| 22:51.34 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
| 22:51.45 | Qwell[] | I don't actually...but I should |
| 22:51.47 | eric_hill | Anyone use reiserfs4? Does it help with large VM file structures? |
| 22:51.57 | Nugget | ricerfs! :) |
| 22:52.08 | *** part/#asterisk mtaht_ (n=mtaht@c-71-198-23-124.hsd1.ca.comcast.net) |
| 22:52.21 | eric_hill | i.e. speed? Reliability? We have quite a few small gsm files. It can't be that efficient on space with ext... |
| 22:52.32 | Grizzy | it rices your files? : o ) |
| 22:52.44 | mog_work | Qwell, http://lists.digium.com/pipermail/asterisk-commits/2006-March/002586.html woohoo |
| 22:52.58 | Qwell[] | :D |
| 22:53.00 | Grizzy | I like FreeBSD ufs (berkeley fast file system) |
| 22:53.11 | eric_hill | rice - noun. Act of making things go faster by sticking a "Type R" sticker on them. |
| 22:53.17 | mog_work | but i want to see some mega commits |
| 22:53.24 | Qwell[] | mog_work: they're in the works |
| 22:53.33 | Qwell[] | I'm ripping shit apart |
| 22:53.44 | Qwell[] | die sub lines, DIE! |
| 22:53.50 | mog_work | heh |
| 22:53.51 | Grizzy | I'm thinking of a kitchen implement, a potato ricer for making smooth mashed potatoes. |
| 22:54.03 | Qwell[] | They're currently gone, and it compiles...I have yet to test it |
| 22:56.17 | mog_work | COMMIT IT! |
| 22:56.19 | mog_work | ^_^ |
| 22:56.20 | Qwell[] | just wait...by this time next year, I'll actually be able to dial :P |
| 22:56.46 | asterboy | broken link on Digium.com front page? |
| 22:56.48 | asterboy | http://lists.digium.com/pipermail/asterisk-commits/2006-March/002586.html |
| 22:56.57 | asterboy | oopss not that one. |
| 22:57.07 | asterboy | this one: http://lists.digium.com/pipermail/asterisk-commits/2006-March/002586.html |
| 22:57.20 | mog_work | same one |
| 22:57.29 | asterboy | this one! |
| 22:57.30 | asterboy | http://www.linuxpr.com/releases/8562.html |
| 22:57.48 | asterboy | Suppose to be recent news. |
| 22:58.04 | mog_work | thanks |
| 22:58.17 | [av]bani | 404 - File not found |
| 22:58.17 | [av]bani | Sorry, the file you have requested cannot be found on any of our servers. Please check the file name and try again, or try a search on search.internet.com. For your convenience, we have listed below an extended menu of Jupitermedia's sites. |
| 22:58.39 | asterboy | Is there a news archive? |
| 22:58.58 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.236) |
| 22:59.01 | *** join/#asterisk sssk (n=sssk@s55935276.adsl.wanadoo.nl) |
| 22:59.08 | asterboy | I'd like to put recent news in my quote as "growing community" |
| 22:59.19 | asterboy | You don't have permission to access /releases/ on this server. |
| 22:59.44 | mog_work | eep |
| 22:59.50 | mog_work | message brookshire asterboy |
| 23:00.16 | Qwell[] | /msg brookshire omg, the sky is falling!!! |
| 23:00.25 | mog_work | heh |
| 23:00.30 | asterboy | seen brookshire |
| 23:00.32 | mog_work | Qwell, have you met brooks? |
| 23:00.40 | Qwell[] | nope |
| 23:01.07 | Corydon-w | We'll get him to spoon Brooks at Astricon |
| 23:02.11 | mog_work | hah |
| 23:03.01 | *** join/#asterisk konfuzed (n=Konf@H211.C18.B96.tor.eicat.ca) |
| 23:04.00 | Corydon-w | btw, have you spoken to mattf recently? |
| 23:05.24 | mog_work | yes |
| 23:05.30 | mog_work | he is usually down hall |
| 23:05.58 | Corydon-w | I have a ? for him |
| 23:06.20 | oej | I have ????? for him |
| 23:06.28 | mog_work | he isnt in on mondays |
| 23:06.29 | Qwell[] | Want him to appear? |
| 23:06.31 | mog_work | you could email him |
| 23:06.33 | Qwell[] | I have $$$$ for him |
| 23:06.36 | mog_work | but he will be onlinelater |
| 23:06.38 | Corydon-w | libmatt |
| 23:06.40 | mog_work | ill take it for him Qwell |
| 23:06.43 | Qwell[] | mog_work: :p |
| 23:06.47 | oej | Qwell: No, the $$$$$ was for me |
| 23:06.54 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 23:06.59 | Qwell[] | oej: ahh |
| 23:07.21 | oej | But it's ok to paypal them to me |
| 23:07.22 | Corydon-w | Qwell[]: besides, he's not that kind of guy. You can't spoon him. |
| 23:07.31 | oej | As long as you don't send it with KLM |
| 23:07.44 | oej | hey mog_work!!! How are you? |
| 23:07.44 | Qwell[] | KLM? |
| 23:08.02 | mog_work | some silly swedish airline |
| 23:08.05 | mog_work | wronged oej |
| 23:08.08 | oej | KLM is officially erase from my list of Airline options |
| 23:08.08 | mog_work | big time |
| 23:08.11 | Corydon-w | KLM is the airline that lost oej's luggage |
| 23:08.25 | Qwell[] | ahh |
| 23:08.31 | justinu | klm is royal dutch airlines |
| 23:08.42 | Qwell[] | what an interesting abbreviation |
| 23:08.49 | ManxPower | Enterprize rent a car is officialy on my list of Companies to Destroy |
| 23:08.49 | shmaltz | anybody here tried the wifi service they offer on the planes for VoIP? |
| 23:08.50 | oej | ...and put me in Dutch jail (a roadside motel) instead of sending me to my destination |
| 23:08.54 | Qwell[] | I'd assume rda, but no...silly Sweeds :) |
| 23:09.10 | ManxPower | oej, what, you could not find anything fun to do in Amsterdam? |
| 23:09.12 | oej | shmaltz: KLM does not. I've used it many times on SAS - the wonderful and serviceminded Scandinavian choice |
| 23:09.30 | oej | ManxPower: On a motel in the middle of nowhere? |
| 23:09.41 | ManxPower | oej, Ah, perhaps not. |
| 23:09.46 | shmaltz | oej, and the VoIP calls were ok? what was the latency? |
| 23:09.50 | oej | shmaltz: 700 ms latency |
| 23:09.56 | oej | Walkie-talkie |
| 23:09.59 | oej | But it worked! |
| 23:10.03 | ManxPower | 700ms! That's better than MY internet service. |
| 23:10.09 | oej | The purser placed a call through my PC just to test |
| 23:10.15 | justinu | he's a few miles closer to the satellite :P |
| 23:10.15 | shmaltz | that is not really very bad, I thought that those connections make you want to yell out the window |
| 23:10.30 | ManxPower | somegeek, where are you now, oej |
| 23:10.31 | oej | No, it's actually pretty good |
| 23:10.40 | oej | I have committed patches from above greenland |
| 23:10.50 | ManxPower | ..er... oej, where are you now? |
| 23:10.52 | oej | Manxpower: Huntsville! |
| 23:10.57 | ManxPower | ah! |
| 23:11.05 | oej | All of the week |
| 23:11.08 | *** join/#asterisk gandhijee (i=HydraIRC@ip72-192-222-181.dc.dc.cox.net) |
| 23:11.22 | oej | ...and I got your package here... hint, hint |
| 23:11.23 | ManxPower | nifty, maybe I can drive up some afternoon and we can have dinner. |
| 23:11.29 | ManxPower | (nearer the end of the week. |
| 23:11.31 | oej | Absolutely |
| 23:11.35 | oej | Leaving saturday |
| 23:12.03 | SwK[Work] | oej check your pm's |
| 23:12.04 | oej | File last week, me this week. Who will take care of Asterisk development over here next week? |
| 23:12.08 | gandhijee | so i upgrade from 1.2.4 to 1.26 |
| 23:12.12 | gandhijee | 1.2.6 |
| 23:12.36 | gandhijee | now i get an error about chan_oss.c:533 sound_thread: select failed: Bad file descriptor |
| 23:12.44 | *** join/#asterisk fuzzbawl (n=fuzzy@69.44.205.70) |
| 23:12.57 | mog_work | ManxPower, you have to come visit |
| 23:13.12 | ManxPower | mog_work, as soon as my bank returns the money the stole from me. |
| 23:13.28 | mog_work | lol is wells fargo a bank? |
| 23:13.30 | Qwell[] | ManxPower: I had *nothing* to do with that...I swear |
| 23:13.35 | mog_work | i thought they just moved money around |
| 23:13.38 | Qwell[] | mog_work: :p |
| 23:13.43 | mog_work | like moneygram or something |
| 23:13.46 | mog_work | that and packages |
| 23:13.51 | Qwell[] | mog_work: not for like...100 years, heh |
| 23:13.57 | ManxPower | mog_work, they moved it into a closed account |
| 23:14.10 | mog_work | ew |
| 23:14.15 | mog_work | i have had something like that happen |
| 23:14.16 | mog_work | is a pain |
| 23:14.18 | Qwell[] | ManxPower: woops...was it wells? |
| 23:14.19 | ManxPower | my entire month's income |
| 23:14.26 | mog_work | i will never bank with compass bank again |
| 23:14.30 | ManxPower | Qwell, BankOne/Chase |
| 23:14.34 | Qwell[] | phew :p |
| 23:14.45 | mog_work | i dont understand banks these days |
| 23:14.53 | mog_work | they make all money out of screwing customer |
| 23:14.55 | Qwell[] | ManxPower: Why'd they move it? |
| 23:15.01 | Qwell[] | mog_work: yeah, pretty much |
| 23:15.03 | mog_work | why not just find a happy medium |
| 23:15.08 | mog_work | like dont give me free checking |
| 23:15.15 | mog_work | give me 3 dollars a month checking |
| 23:15.18 | ManxPower | Qwell, the person that does the deposits of my checks had an old deposit slip. |
| 23:15.24 | Qwell[] | wtf? |
| 23:15.25 | mog_work | and dont try to screw me every thirty seconds |
| 23:15.30 | ManxPower | the bank happily accepted the depoist to the closed account and even cleared the checks. |
| 23:15.35 | ManxPower | that happened 7 days ago |
| 23:15.38 | Qwell[] | SOBs |
| 23:16.06 | Qwell[] | they plan on returning it soon? |
| 23:16.07 | ManxPower | they CLAIM they rejected the check and mailed it to me, but it has not arrived yet. |
| 23:16.15 | mog_work | bs |
| 23:16.36 | Qwell[] | have them overnight a photocopy of the check. |
| 23:16.42 | Qwell[] | They have to keep it on file |
| 23:16.58 | websae | any canadian carriers in here |
| 23:16.59 | websae | ? |
| 23:17.03 | Qwell[] | Check 21, or some such law |
| 23:17.28 | Qwell[] | "Checking system for the 21st century"...something like that |
| 23:17.52 | Qwell[] | let's the equiv of an email be valid for check transfers, between banks |
| 23:17.57 | gandhijee | anyone ever get this error: chan_oss.c:533 sound_thread: select failed: Bad file descriptor? |
| 23:18.03 | Qwell[] | lets* |
| 23:18.13 | gandhijee | it only happend after i upgraded from 1.2.4. to .6 |
| 23:18.30 | Qwell[] | gandhijee: check the bug tracker...see if there is anything new, related to that |
| 23:18.35 | Qwell[] | if not - open one up |
| 23:18.53 | gandhijee | ok |
| 23:18.57 | gandhijee | thanks |
| 23:19.21 | Qwell[] | mog_work: I propose a new rule |
| 23:19.24 | gandhijee | Qwell: dumb question, but how do i get to the bugtracker? |
| 23:19.30 | gandhijee | nm |
| 23:19.31 | Qwell[] | in order to open a bug - you must test (and possibly close) 2 others :P |
| 23:19.33 | mog_work | bugs.digium.com gandhijee |
| 23:19.35 | mog_work | lol |
| 23:20.14 | Qwell[] | how incredibly useful would that be? heh |
| 23:20.18 | Netgeeks | not! |
| 23:20.22 | *** join/#asterisk Eitch (n=hugo@unaffiliated/eitch) |
| 23:20.26 | mog_work | make it work like that delicious site |
| 23:20.29 | Qwell[] | ? |
| 23:22.36 | shmaltz | interesting: |
| 23:22.38 | shmaltz | http://news.yahoo.com/s/nf/20060327/tc_nf/42395;_ylt=AuttYUnyeQl4k1IbWwO2RYz6VbIF;_ylu=X3oDMTBjMHVqMTQ4BHNlYwN5bnN1YmNhdA-- |
| 23:23.24 | fuzzbawl | i'm having a bit of trouble, but I'm not sure if it's asterisk or AMP causing the issue. Voicemails record (I can see the files in /var/spool/asterisk/voicemail/) but they are not attached to the email |
| 23:23.39 | ManxPower | shmaltz, page not foound |
| 23:23.58 | shmaltz | ManxPower, check your DNS settings |
| 23:24.16 | ManxPower | shido6, um, yahoo is telling me page is not found |
| 23:24.22 | ManxPower | ..er. |
| 23:24.26 | ManxPower | shmaltz, , um, yahoo is telling me page is not found |
| 23:24.33 | shmaltz | ManxPower, oh sorry, but it works for me |
| 23:24.37 | Darwin35 | everyone is fired |
| 23:24.44 | Darwin35 | go home and leave this place |
| 23:24.57 | ManxPower | Can anyone else here get to http://news.yahoo.com/s/nf/20060327/tc_nf/42395;_ylt=AuttYUnyeQl4k1IbWwO2RYz6VbIF;_ylu=X3oDMTBjMHVqMTQ4BHNlYwN5bnN1YmNhdA-- |
| 23:25.08 | Darwin35 | yes |
| 23:25.12 | ManxPower | shmaltz, I repasted and it may be working now. |
| 23:25.15 | Qwell[] | worked for me |
| 23:25.28 | wunderkin | oh my |
| 23:26.57 | shmaltz | I like the JaJah one |
| 23:27.12 | shmaltz | it's like placing 2 .call files one in UK and the other in NY |
| 23:27.28 | shmaltz | who here is a southern? |
| 23:27.47 | shmaltz | what does the expression *thats just a bummer* mean? |
| 23:28.32 | shmaltz | anybody from digum land here? |
| 23:28.48 | Darwin35 | thats coo |
| 23:28.50 | Darwin35 | cool |
| 23:28.54 | Darwin35 | I just tried it |
| 23:29.05 | shmaltz | Darwin35, what's cool? |
| 23:29.10 | Darwin35 | jajah |
| 23:29.42 | Darwin35 | you enter the numbers in the portal it calls you and then calls the other number |
| 23:29.56 | SwK[Work] | hey whats the link to Leif & Crew's asterisk book? |
| 23:29.59 | Darwin35 | and uses a voip connection for the call |
| 23:30.05 | ManxPower | ~docs |
| 23:30.06 | jbot | somebody said docs was probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 23:31.59 | shmaltz | JaJah doesn't have progress detection |
| 23:32.14 | shmaltz | I hung up around 50 seconds ago and it still says call is active |
| 23:32.29 | shmaltz | I'm using digital lines on bothe ends (SPRINT PCS, to PRI) |
| 23:32.52 | *** join/#asterisk KranZ (n=user@sme.bestline.net) |
| 23:32.52 | ManxPower | Huh? |
| 23:32.59 | ManxPower | PRIs ALWAYS have hangup supervision |
| 23:33.10 | shmaltz | ManxPower, but JaJah is not detecting it |
| 23:33.26 | ManxPower | Ah, the silly new service that will fail like all the others? |
| 23:33.27 | IronHelixz | ~amp |
| 23:33.28 | jbot | [amp] "amp is, like, NOT supported here! people using it should join #freepbx (the new name of amp)" |
| 23:33.37 | shmaltz | It's still saying call is active after like 2:30+ |
| 23:33.50 | ManxPower | shmaltz, more billing for them. |
| 23:34.10 | shmaltz | yeah, thats what it was |
| 23:34.44 | *** join/#asterisk lunaphyte_ (n=lunaphye@70.90.148.1) |
| 23:35.00 | Qwell[] | When you need a longer cat5 cable, but don't have a coupler, what do you do? |
| 23:35.00 | asterboy | ok, by bid for an asterisk install at a crisis center is on its way. |
| 23:35.03 | asterboy | hope I get it. |
| 23:35.19 | Qwell[] | You use a cisco 7960 as a switch, to double the cable length :D |
| 23:35.42 | [av]bani | :P |
| 23:35.42 | asterboy | for phones I just use 2 tin cans and a string. |
| 23:35.43 | shmaltz | nah its not that, it's just that the page is coded that it doesn't update once the call is active, it just uses a static timer |
| 23:35.45 | lunaphyte_ | Qwell: you get your cable stretcher out. |
| 23:35.56 | asterboy | hard to call forward though |
| 23:36.01 | shmaltz | Qwell, I had this problem today |
| 23:38.09 | edobe | any good cheap IP phone recommendation? |
| 23:38.17 | edobe | how about atcom? |
| 23:38.19 | shmaltz | Qwell, I ended up cutting one end and punching it down to a jack |
| 23:38.26 | shmaltz | I charged $25.00 for that |
| 23:38.30 | xp_prg | edobe just use a software phone |
| 23:38.37 | IronHelix | ~softphone |
| 23:38.38 | jbot | something that should be drug out into the street and shot |
| 23:38.38 | gandhijee | edobe: you want IAX or SIP |
| 23:38.40 | Qwell[] | shmaltz: I don't have any tools or any such thing here, heh |
| 23:39.00 | gandhijee | Qwell: there isn't an area to submit a bug for 1.2.6 yet... |
| 23:39.10 | Qwell[] | gandhijee: no? |
| 23:39.12 | shmaltz | Qwell, so you did the right thingy :-) |
| 23:39.35 | edobe | gandhijee: IAX |
| 23:39.46 | gandhijee | Qwell: yep, no place to submit 1.2.6 bug reports yet |
| 23:39.55 | edobe | shmaltz: i´m testing with softphone but i´d like to port this setup to an office |
| 23:40.18 | gandhijee | edobe: i recommend the iareaPhone X12 |
| 23:40.29 | shmaltz | edobe, what? |
| 23:40.33 | gandhijee | its a YUXIN phone |
| 23:40.42 | gandhijee | has the same chipset as the atacom |
| 23:42.11 | edobe | gandhijee: does it support multiple calls? |
| 23:42.28 | gandhijee | multiple lines? |
| 23:42.32 | Qwell[] | gandhijee: we'll have to find an administrator then... |
| 23:42.34 | Darwin35 | 1688 chipset rocks |
| 23:42.37 | IronHelix | did yuxin ever figure out the concept of not spamming the wiki? |
| 23:42.44 | gandhijee | rofl |
| 23:42.45 | shmaltz | what time zone is new zealand? |
| 23:42.48 | Darwin35 | opensrc phones the way to go |
| 23:42.52 | Qwell[] | mog_work: Can you add/change versions on mantis? |
| 23:43.14 | mog_work | ? |
| 23:43.20 | Qwell[] | asterisk versions - 1.2.6 |
| 23:43.39 | mog_work | yeah |
| 23:43.43 | mog_work | in advanced or something |
| 23:43.47 | edobe | gandhijee: where are those sold? |
| 23:44.43 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
| 23:44.48 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-43-154.cybersurf.com) |
| 23:45.00 | Qwell[] | mog_work: I can only see projects in manage, not any of the custom fields stuff |
| 23:45.17 | mog_work | what bug? |
| 23:45.23 | Qwell[] | all :D |
| 23:45.37 | Darwin35 | 1.2.6 wow what is this a release every 2 weeks now |
| 23:45.46 | mog_work | oh add versions |
| 23:45.48 | mog_work | no i cant |
| 23:45.54 | mog_work | email webmaster@digium.com |
| 23:45.57 | gandhijee | edobe: iareaphone.com |
| 23:45.58 | mog_work | he can do it for ya |
| 23:46.05 | mog_work | not quite Darkhalf |
| 23:46.08 | mog_work | err Darwin35 |
| 23:46.57 | *** join/#asterisk Shaun222 (i=Shaun@tina.ndcservers.net) |
| 23:47.07 | Darwin35 | hehhe |
| 23:47.17 | Darwin35 | When did my nick change |
| 23:47.27 | mog_work | it didnt tab complete faild me |
| 23:47.34 | Darwin35 | lol |
| 23:47.59 | mog_work | im a lazy typist |
| 23:48.16 | justinu | so I take it there's no way to take two channels, and bridge them together in AMI.... looks like I have to drop both channels into a meetme, if I want them to talk... |
| 23:48.26 | Qwell[] | yea<tab> m<tab> to<tab> |
| 23:48.42 | Darwin35 | lol |
| 23:48.44 | Darwin35 | brb |
| 23:48.49 | Qwell[] | I wish I could tab complete other words... |
| 23:49.07 | mog_work | its getting there Qwell |
| 23:49.17 | mog_work | i think there is a mode for that in oofice |
| 23:49.19 | Qwell[] | all we need... |
| 23:49.21 | mog_work | and several ides do it |
| 23:49.28 | Qwell[] | is a giant channel, with several thousand nicks |
| 23:49.33 | Qwell[] | which are all common words :P |
| 23:49.57 | mog_work | heh |
| 23:51.26 | *** join/#asterisk Kumbang (n=kumbang@167.205.24.5) |
| 23:51.45 | Darwin35 | quiting time ... chat from home.... |
| 23:51.48 | *** part/#asterisk Darwin35 (n=darwin35@sta-208-139-193-162.rockynet.com) |
| 23:52.16 | *** join/#asterisk ManxPower (n=ewieling@dpc6745150107.direcpc.com) |
| 23:53.38 | Shaun222 | is it possible to setup redundancy with asterisk? right now i want to use it for my primary system, but if for some reason the network at the DC fails, i would like to either use analog lines to receive/send calls or have another asterisk server at another location that everything will fail-over too |
| 23:54.06 | shmaltz | Shaun222, whats DC stand for? |
| 23:54.11 | Qwell[] | datacenter? |
| 23:54.35 | Shaun222 | yes |
| 23:54.54 | mog_work | district of columbia |
| 23:55.03 | Qwell[] | direct current |
| 23:55.08 | shmaltz | mog, thanks |
| 23:55.18 | shmaltz | ~dc |
| 23:55.19 | jbot | i heard dc is better known as dc_ |
| 23:55.20 | *** join/#asterisk bkw__ (n=brian@adsl-70-142-62-199.dsl.tul2ok.sbcglobal.net) |
| 23:55.23 | Nugget | marvel comics are way cooler than dc. |
| 23:55.26 | gandhijee | Shawn222: there was something on the voip-info wiki about setting up something like that using a hearbeat server |
| 23:55.32 | mog_work | bah Nugget dc is old school |
| 23:55.41 | shmaltz | this looks cool: |
| 23:55.42 | shmaltz | http://www.google.com/search?hl=en&q=define+dc&btnG=Google+Search |
| 23:55.47 | bkw__ | DC is better |
| 23:55.54 | mog_work | whats dc stand for |
| 23:55.59 | mog_work | D_ comics? |
| 23:56.00 | bkw__ | Direct Current |
| 23:56.05 | subdolus | lol |
| 23:56.10 | Qwell[] | < Qwell[]> direct current |
| 23:56.13 | Qwell[] | I win this round! |
| 23:56.21 | Qwell[] | Next on "What will bkw__ say, ..." |
| 23:56.38 | bkw__ | whats the point of going DC -> AC -> DC? |
| 23:56.40 | bkw__ | you generate HEAT |
| 23:56.42 | bkw__ | and thats bad |
| 23:56.43 | bkw__ | mmkay |
| 23:56.46 | bkw__ | NEXT!!!! |
| 23:56.48 | mog_work | umm yeah |
| 23:56.51 | mog_work | whats your point? |
| 23:56.56 | Qwell[] | AC -> DC -> AC makes cold |
| 23:57.00 | mog_work | lol Qwell |
| 23:57.07 | bkw__ | ok you save room because you don't waste space with power supplies |
| 23:57.29 | mog_work | yes i agree |
| 23:57.37 | mog_work | but what does this have to do with disussion |
| 23:57.52 | Qwell[] | discussion control? |
| 23:57.56 | mog_work | lol |
| 23:57.57 | Qwell[] | (dc...get it?) |
| 23:57.58 | bkw__ | oh I see how it is |
| 23:58.02 | mog_work | dire commie? |
| 23:58.09 | bkw__ | dumbass coder? |
| 23:58.17 | Qwell[] | well, I never |
| 23:59.23 | Qwell[] | we've officially killed the channel |
| 23:59.44 | mog_work | dead channel? |
| 23:59.49 | Qwell[] | ... |
