| 00:06.06 | *** join/#asterisk DJ-Pyro (n=abrezins@12-208-113-85.client.insightBB.com) |
| 00:08.12 | cripito | protected virtual void OnUnParkedCall(managerEventsArgs e) |
| 00:08.12 | cripito | { |
| 00:08.12 | cripito | if (UnParkedCall != null) |
| 00:08.13 | cripito | { |
| 00:08.13 | cripito | UnParkedCall(this, e); |
| 00:08.13 | cripito | } |
| 00:08.15 | cripito | } |
| 00:08.17 | cripito | oops |
| 00:08.19 | cripito | sorry |
| 00:08.21 | cripito | wrong place |
| 00:09.33 | n3u7 | hey people:the following is the modules.conf that I had to impliment to get asterisk to run on Sarge |
| 00:09.38 | n3u7 | http://www.darkphiber.ca/asterisk/modules.conf |
| 00:09.51 | n3u7 | it's rendered asterisk useless |
| 00:10.23 | *** join/#asterisk Sp3ciaL_K (n=alex@d141-139-99.home.cgocable.net) |
| 00:10.24 | n3u7 | good news is that it would compile on Sarge from tonights cvs |
| 00:10.26 | Sp3ciaL_K | good eve |
| 00:10.29 | n3u7 | which is new |
| 00:11.59 | cripito | :) |
| 00:12.38 | Sp3ciaL_K | im trying to get app_rx|txfax working on freebsd |
| 00:13.45 | Sp3ciaL_K | but i keep getting undefine symbol lrint |
| 00:17.05 | FuriousGeorge | if i send my 240 local calling extension-exchanges to an included file in my dialplan, can i call macros in the original file from the included file? |
| 00:17.45 | *** join/#asterisk Los415 (n=los415@64.201.104.186) |
| 00:18.38 | FuriousGeorge | it would be pretty annoying if you couldnt. |
| 00:18.51 | FuriousGeorge | would you have to include the original file in the included file? |
| 00:19.19 | n3u7 | onto knoppix |
| 00:19.48 | alephcom | wunderkin: The Local change you suggested works perfectly. Thanks again. |
| 00:20.58 | *** join/#asterisk brent21 (n=Brent@pcp0010945096pcs.arlngt01.va.comcast.net) |
| 00:21.02 | *** join/#asterisk dudes (n=dudes@12-215-32-62.client.mchsi.com) |
| 00:21.11 | brent21 | does anyone know how to reset an IAXY? the instructions on the wiki dont seem to work |
| 00:22.00 | FuriousGeorge | can someone database all the areacode and exchanges, and write a canned script that generates a local calling via pots dialplan. its so freakin tedious |
| 00:23.40 | brent21 | these iaxy things are cheap |
| 00:24.55 | Katty | FuriousGeorge: hewwo. |
| 00:26.26 | FuriousGeorge | Katty: how goes it :) |
| 00:26.49 | Katty | FuriousGeorge: my brain hurts a little from unscrambling a 4d rubik's cube. |
| 00:27.16 | *** join/#asterisk coppice (n=chatzill@219.199.17.210.dyn.pacific.net.hk) |
| 00:27.19 | FuriousGeorge | 4d rubix cube? you gotta order it by side color and time? |
| 00:27.54 | FuriousGeorge | or does that mean its 4X4X4? |
| 00:29.38 | Katty | uhh |
| 00:29.39 | Katty | hold on |
| 00:29.54 | Katty | http://www.brick.net/~izaah/backrotate.jpg |
| 00:30.05 | Katty | 4 d because you can't see all the cubes at once |
| 00:30.19 | Katty | every single cube you click on will effect a rotation of some sort (except the center cubes which you can't move) |
| 00:30.34 | Katty | the solution is to ctrl click, which will bring the last cube into place so you can see it |
| 00:30.37 | cripito | my brain hurts even just looking it |
| 00:30.43 | Katty | rotate your view, then you can solve puzzle |
| 00:30.50 | FuriousGeorge | madness |
| 00:30.54 | FuriousGeorge | would ensue |
| 00:30.55 | Katty | it's quite fun. |
| 00:31.16 | Katty | it's quantum theory, basically (= |
| 00:31.27 | Katty | stepping out of your dimension to accomplish something |
| 00:31.34 | Katty | or to explain something which is not possible mathematically |
| 00:31.39 | Katty | (imaginary numbers) |
| 00:31.48 | Katty | yet, completely possible in another. |
| 00:31.49 | cripito | more pain ..... |
| 00:32.24 | FuriousGeorge | Katty: whenever you want i'd be willing to trade tasks, i just found out there are 240 area code exchanges that are "local" that need to be in my dialplan |
| 00:32.33 | cripito | :D |
| 00:33.10 | FuriousGeorge | Katty: you into advanced mathematics? |
| 00:33.14 | cripito | george u can apply the quantum theory to that.... |
| 00:33.17 | Katty | FuriousGeorge: no, i just turned 21 |
| 00:33.27 | Katty | FuriousGeorge: i've always been good with math though ;) |
| 00:33.55 | FuriousGeorge | cripito: if it would make it easier for me i'd love to know how, i got about 200 to go i think |
| 00:33.59 | Katty | http://www.superliminal.com/cube/cube.htm <- for much mindboggling fun! |
| 00:34.12 | cripito | divide the numbers in 8 differents group so and do 30 at the time from a different shair ;) |
| 00:34.26 | FuriousGeorge | Katty: math is fun |
| 00:34.40 | cripito | george question |
| 00:34.40 | FuriousGeorge | i shoulda studied something math related in college |
| 00:34.46 | cripito | did u have php in your machine? |
| 00:34.47 | Katty | http://www.superliminal.com/cube/halloffame.htm <- ubersmartness. |
| 00:35.13 | cripito | the numbers are consecutives george? |
| 00:37.05 | FuriousGeorge | ok, guys, i gotta really fun math related asterisk problem. in since im getting into open source software, i wanna submit this to the community |
| 00:37.07 | FuriousGeorge | http://pastebin.ca/26121 |
| 00:37.14 | coppice | if maths is fun, why are most people so scared of it? :-) |
| 00:38.09 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
| 00:39.27 | *** part/#asterisk sgorilla (n=tlp@cpe-24-160-119-179.houston.res.rr.com) |
| 00:39.46 | FuriousGeorge | ok here's one: two bikers are 25 miles appart, approaching at 10 mph. a fly on the front tire of 1 travels to the other front tire at 20 mph. Upon reaching the second tire he flys back to the original and repeats, until the bikes collide and crush him |
| 00:39.51 | FuriousGeorge | how far does the fly travel |
| 00:41.06 | MikeJ[Laptop] | yes |
| 00:44.06 | FuriousGeorge | well i just thought it would be fun |
| 00:45.08 | cripito | :) that's is a good one |
| 00:45.14 | FuriousGeorge | the bikes meet in 2.25 hrs, the fly's going 20 mph. so he can fly for 45 miles till he gets crushed |
| 00:45.33 | *** join/#asterisk brent21 (n=Brent@pcp0010945096pcs.arlngt01.va.comcast.net) |
| 00:45.56 | FuriousGeorge | most people think its more complicated cuz they gotta figure some decaying average with logs and factorials or something |
| 00:47.23 | cripito | 2.25 hrs or 2.5? |
| 00:47.59 | FuriousGeorge | oops |
| 00:48.29 | cripito | and i think that less. b/c the travelles only travel half |
| 00:48.35 | FuriousGeorge | the bikes meet in 1.25 hrs not 2.25, so the answer is 25 not 45 |
| 00:48.40 | cripito | :) |
| 00:48.43 | cripito | now ok |
| 00:49.19 | cripito | we always forgets those details ;) |
| 00:49.27 | FuriousGeorge | lol |
| 00:50.50 | cripito | i am happy my parked calls are working :) |
| 00:50.53 | FuriousGeorge | d'oh |
| 00:51.24 | FuriousGeorge | cripito: you using the standard call parking? |
| 00:51.54 | cripito | :) i am parking the calls from a manager appl george |
| 00:52.05 | cripito | not from the phones... |
| 00:52.13 | FuriousGeorge | cripito: i see |
| 00:52.13 | *** join/#asterisk mcadory (n=mcadory@208.149.64.28) |
| 00:52.16 | cripito | so u can said no |
| 00:52.20 | *** join/#asterisk kippi (n=kippi@cpc4-hatf3-6-0-cust33.lutn.cable.ntl.com) |
| 00:52.21 | *** part/#asterisk mcadory (n=mcadory@208.149.64.28) |
| 00:52.33 | kippi | what do people use for time servers? |
| 00:52.33 | *** join/#asterisk mcadory (n=mcadory@208.149.64.28) |
| 00:52.39 | cripito | ntpd |
| 00:53.14 | FuriousGeorge | cripito: i use vanilla call parking, was wondering if there would be some way to park calls on ${PARKEREXTEN}# or something like that |
| 00:55.09 | cripito | i never use vanilla call parking.. i am using the normal park system |
| 00:55.19 | FuriousGeorge | cripito: normal=vanila |
| 00:55.20 | cripito | and i just redirect to the park extension |
| 00:55.25 | cripito | :) |
| 00:55.27 | cripito | good 2 known |
| 00:55.28 | cripito | ;) |
| 00:56.22 | cripito | in that case i don't think that vanilla allow to choose the extension to park |
| 00:56.25 | FuriousGeorge | im up to 2016*. got 3 area codes to go |
| 00:56.49 | cripito | just pickup the first free from the parking lot |
| 00:57.18 | FuriousGeorge | assuming you know which the first free is |
| 00:57.19 | cripito | anyway was really easy just redirect to the parking extension |
| 00:57.31 | cripito | :) no need to known |
| 00:57.40 | cripito | u redirect to the parking extension definited in features.conf |
| 00:57.57 | cripito | and u can get the programmatically parsing the answer from the command show features |
| 00:58.09 | cripito | uff |
| 00:58.18 | cripito | i speak chinese sometimes |
| 00:58.29 | FuriousGeorge | lol |
| 00:58.43 | cripito | u can get that one parsing the answer of command show feautures |
| 00:59.31 | FuriousGeorge | this is what i would like: when i put a call on park, the person cal "pull" that call from me by dialing my extension + something. |
| 00:59.32 | cripito | that was really fun ;) |
| 00:59.37 | tehdely | ass turd dicks |
| 00:59.46 | FuriousGeorge | tehdely: well said |
| 01:00.38 | NewSole2 | hmm shitty dip stick |
| 01:04.52 | brent21 | Anyone have any experience with the PA168V? |
| 01:04.58 | NewSole2 | ya |
| 01:04.59 | brent21 | thinking about picking one up |
| 01:05.32 | NewSole2 | hmm must be a digium fan.... got money to waist |
| 01:05.47 | brent21 | who me? |
| 01:05.53 | brent21 | i have two IAXY's here |
| 01:06.36 | Nugget | Eye think it's time for the heeling to begin. |
| 01:06.45 | Nugget | We kneed tolerance. |
| 01:06.51 | brent21 | i just dont like the fact that the IAXY only supports Ulaw and ADPCM |
| 01:07.02 | Nugget | Everyone nose that |
| 01:07.02 | brent21 | it doesn't work well in my environment |
| 01:07.49 | brent21 | i missed a lot in the last year i take it |
| 01:07.56 | NewSole2 | the PA168V are Shit |
| 01:08.06 | cripito | question... did any figureout what means the status in the |
| 01:08.14 | cripito | agent memberstatus |
| 01:08.14 | brent21 | hmm, so no real good IAX ata out there yet? |
| 01:08.25 | NewSole2 | yes there is |
| 01:08.35 | brent21 | which one? |
| 01:09.17 | NewSole2 | we have them.... made with same ship set... but better hardware.... the PA168V are known to die with out warning.... I know I have a box full of them |
| 01:10.11 | NewSole2 | Nugget.... Personaly I think its about time Digium got a slap in the face |
| 01:10.14 | brent21 | hmm, NewSole2, so is there anything better? we need an IAX ATA with low bandwidth codec support |
| 01:10.26 | brent21 | and from what I read, 168V and IAXY are the only ones |
| 01:10.30 | NewSole2 | g729 low enough |
| 01:10.31 | Katty | NewSole2: why? |
| 01:10.31 | *** join/#asterisk Pete_Largo (n=PeteLarg@225-196.35-65.tampabay.res.rr.com) |
| 01:10.33 | brent21 | atleast from what I could find |
| 01:10.36 | brent21 | g729 is perfect |
| 01:10.47 | brent21 | g729 or GSM |
| 01:10.50 | brent21 | either or |
| 01:11.09 | Katty | brent21: they're not /perfect/ |
| 01:11.16 | Nugget | NewSole2: I look forward to seeing your hardware when it reaches the marketplace. |
| 01:11.20 | *** join/#asterisk Sed[SS] (i=brandon@smartserv.ipv6.smart-serv.net) |
| 01:11.39 | Katty | Nugget: way to ignore me. |
| 01:11.47 | NewSole2 | for the last year there has been more problems with asterisk... and no one wanted to fix it and kram and every one never did anything till they were confronted in public then and only then did they do somthing |
| 01:11.47 | Katty | Nugget: son of a silly person. |
| 01:12.12 | Katty | NewSole2: and how is this different than every other industry, again? |
| 01:12.12 | Sed[SS] | Ok.... I'm at a loss.. I've looked through the wiki.. and I've tried a few things... how can I lower the volume of music on hold? |
| 01:12.15 | NewSole2 | its not my hard ware |
| 01:12.53 | brent21 | this is crazy, i left one job that I deployed asterisk, did something else for a year, came back about two weeks ago |
| 01:13.03 | brent21 | and now it seems all hell has broken loose in the community |
| 01:13.16 | Katty | brent21: blame Nugget |
| 01:13.40 | Katty | and foil. |
| 01:13.57 | Katty | foil is to asterisk as tupperware is to communism. |
| 01:14.06 | NewSole2 | it not... but Digium was taking the same development idea from microsoft.... and more abilities and forget fixing bugs so people can play with new abilities and not be able to use it cause of older bugs |
| 01:14.25 | *** join/#asterisk nobell (n=Jared@160.7.249.18) |
| 01:14.59 | Katty | NewSole2: microsoft has a hell of a marketing campaign. |
| 01:15.01 | nobell | is there a way to provision a new Iaxy without doing DHCP first? |
| 01:15.04 | NewSole2 | lets add more abilities... and hope people forget about the bug |
| 01:15.41 | NewSole2 | ya... buy more upgrades or hardware.... |
| 01:15.59 | alephcom | Human nature is that way. I'm the same myself although I t ry not to be. |
| 01:16.21 | Katty | NewSole2: complain all you want, but your not providing an alternative. |
| 01:16.43 | NewSole2 | echo problem on asterisk.. when buy new card with echo cancel... forget the fact that u just bought two of their last cards cause of digium's say so... and no refunds |
| 01:16.49 | alephcom | I think they are still looking for bug marshalls if somebody has time. Hint,hint.... |
| 01:16.51 | cripito | katty did u live in a communism country or lived? |
| 01:17.08 | Katty | cripito: omgwtfulolzkthxbi!!!11oneone |
| 01:18.39 | cripito | tupperware is no so related..... |
| 01:18.50 | Katty | cripito: it is. |
| 01:18.55 | Katty | cripito: you're just not thinking. |
| 01:19.15 | cripito | i spend 35 years in one... ;) |
| 01:19.28 | Katty | then you obviously understand it was kept contained. |
| 01:20.00 | cripito | more or less. |
| 01:20.08 | *** join/#asterisk desktophero (n=chatzill@ip24-56-30-250.ph.ph.cox.net) |
| 01:20.13 | Katty | then obviously, tupperware falls into place |
| 01:20.17 | NewSole2 | brent21.... http://www.yntx.com/en/productygw20.htm |
| 01:20.20 | Katty | way to not think outside the box. |
| 01:20.23 | desktophero | hello all |
| 01:21.45 | *** join/#asterisk ixx (i=foobar@cpe-24-27-44-163.austin.res.rr.com) |
| 01:22.21 | ixx | anyone here using Telasip? After signup and paying I never was given the opportunity to choose a DID |
| 01:22.42 | ixx | no response to phone or email |
| 01:23.08 | nobell | is there a way to provision a new Iaxy without doing DHCP first? |
| 01:23.12 | *** join/#asterisk Tili (n=Tili@218.20.52.122) |
| 01:23.54 | *** join/#asterisk schuylerdigium (n=Bosco@pcp03052091pcs.huntsv01.al.comcast.net) |
| 01:24.19 | brent21 | nobell not that i know of |
| 01:25.30 | cripito | did anyone have the source code about the iaxprov? |
| 01:25.44 | cripito | i mean the one that u use for set the iaxy |
| 01:26.18 | nobell | http://www.netxusa.com/products/Digium/docs/Iaxy_Installation_Guide.pdf |
| 01:26.31 | brent21 | i just got two IAXY's yesterday |
| 01:26.34 | brent21 | one was dead, other works |
| 01:26.55 | brent21 | i hope the include support for GSM or G729 |
| 01:27.17 | nobell | anybody using FWD ? How is that working for you? |
| 01:27.43 | cripito | thanks nobell |
| 01:28.10 | tehdely | FWD has been all sorts of buggered today |
| 01:28.14 | cripito | that will be very helpfull |
| 01:28.17 | tehdely | about 1 in 10 calls went through for me |
| 01:28.26 | tehdely | of course i use their notoriously buggy iax trunking :P |
| 01:28.31 | cripito | going home |
| 01:28.48 | *** join/#asterisk mcadory (n=mcadory@208.149.64.28) |
| 01:28.48 | nobell | anybody know about that Iaxy pdf? looks like an older model.. the newer models have any newer features? |
| 01:29.02 | *** part/#asterisk Sed[SS] (i=brandon@smartserv.ipv6.smart-serv.net) |
| 01:29.55 | brent21 | nobell, dont think so |
| 01:29.58 | brent21 | im in the same boat as you |
| 01:30.02 | brent21 | just got these today |
| 01:31.17 | cripito | is easy to build a windows appl to setup iaxy |
| 01:31.37 | cripito | anyone interested? |
| 01:32.04 | Tili | cripito: yes its easy. i think some such app is available |
| 01:32.25 | cripito | i am interested in get it.... |
| 01:32.35 | cripito | or build 1 |
| 01:33.06 | Tili | i ported the linux one an year ago. let me check if anything has changed in iaxyprov |
| 01:33.16 | cripito | iaxyprov needs a linux locally for set the iaxy or access from the asterisk system to the phone.... |
| 01:35.00 | Tili | cripito: no iaxyprov is independent of asterisk. |
| 01:35.06 | cripito | no |
| 01:35.09 | cripito | yes |
| 01:35.09 | cripito | and no |
| 01:35.12 | brent21 | does anyone know if it would be impossible (or how hard) to incorporate a low bandwidth codec for the iaxy? |
| 01:35.18 | cripito | u need a linux machine |
| 01:35.37 | cripito | my asterisk server is in a colo.... |
| 01:35.46 | cripito | no linux at home :( for now |
| 01:35.50 | Tili | cripito: for asterisk we need. but for iaxyprov it just needs to be ported to windows, which is quite simple. |
| 01:35.50 | cripito | i sold it :D |
| 01:36.04 | Chuji | brent21 : Not much cpu horsepower in the iaxy |
| 01:36.11 | Chuji | brent21 : don't think it would handle it |
| 01:36.30 | cripito | yeap.. i was looking the code |
| 01:36.35 | brent21 | bummer, unfortunately going to have to return these puppies :( |
| 01:37.10 | Chuji | Yeah, iaxy's have so much potential |
| 01:37.20 | Tili | so cripito simply add WSAStartup in the beginning of main and change some headers which are not in wind32 SDK. link it ws2_32.lib |
| 01:37.27 | Tili | and you should have iaxyprov on windows |
| 01:37.42 | cripito | :) console. yup. |
| 01:37.47 | Chuji | Too bad Sipura was purchased. From what I heard, they were considering iax on them |
| 01:38.01 | brent21 | yeah I know, that does suck |
| 01:38.14 | brent21 | i have like 20 sipura 2000's |
| 01:38.15 | cripito | if iaxy manage to bring g729 that will be trememdous... |
| 01:38.31 | Tili | yeah console |
| 01:38.31 | cripito | i have an entire call center with 300 phones :(((((((( |
| 01:38.32 | brent21 | digium would sell a lot more g729 lic's |
| 01:38.44 | cripito | snif snif |
| 01:38.54 | cripito | spa841 |
| 01:39.04 | Tili | then they will have iaxura |
| 01:39.04 | cripito | cisco was smart... |
| 01:39.26 | Tili | cripito: why g729. why not speex. speex at 8kbps is almost as good as g729 |
| 01:39.27 | brent21 | im still amazed at the infancy of much of the VoIP hardware in the market... With the exception of the cisco and polycom phones (Polycom's were buggy at first), everything seems very amateurish |
| 01:40.05 | Chuji | brent21 : The other traditional pbx ip phones are good |
| 01:40.06 | cripito | :) speex is ok too... |
| 01:40.07 | *** join/#asterisk xkev (n=kevin@orbit.xmission.com) |
| 01:40.14 | *** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net) |
| 01:40.16 | Tili | speex is free also |
| 01:40.29 | Chuji | Lucent, Nortel, Toshiba, etc |
| 01:40.40 | cripito | the 841 is a great cheap phone... |
| 01:40.54 | Chuji | But they designed more for their own systems |
| 01:41.11 | brent21 | yeah, i wonder if anyone will ever come in and try to purchase Digium? |
| 01:41.32 | cripito | :) will be funny about asterisk |
| 01:41.39 | Chuji | Heh, won't be long I would bet |
| 01:42.02 | mog_home | i dont think mark would sell to a nortel |
| 01:42.08 | brent21 | they have an excellent idea, an excellent pbx so far, but the potential is huge |
| 01:42.14 | Sp3ciaL_K | does anyone use spandsp? |
| 01:42.14 | brent21 | its a very diruptive technology |
| 01:42.15 | mog_home | i mean its mark |
| 01:42.22 | Chuji | Either Digium or Pingtel |
| 01:42.28 | brent21 | but everyone is going to start eating at Digium's margins (namely hardware business) |
| 01:43.00 | Chuji | Yeah, I got to believe Sangoma is putting a dent in hardware |
| 01:43.09 | [hC] | Who is pingtel anyways? |
| 01:43.16 | mog_home | they make sip foundry |
| 01:43.18 | Chuji | It's probably growing fast enough where both prosper though |
| 01:43.20 | [hC] | I saw one of their phones at astricon, and.... eeeeacgh. |
| 01:43.21 | mog_home | an opensource sip pbx |
| 01:43.25 | mog_home | heh |
| 01:43.28 | mog_home | we had one |
| 01:43.38 | [hC] | one with jog-dial on the side |
| 01:43.42 | [hC] | y u c k |
| 01:43.45 | mog_home | yeah |
| 01:43.51 | brent21 | was there talk of the fork at astricon? |
| 01:43.58 | mog_home | yes |
| 01:43.59 | brent21 | im curious how this is going to affect * |
| 01:44.12 | brent21 | especially with deploying it in a business i work in |
| 01:44.13 | [hC] | either way it will be good, i think. |
| 01:44.23 | [hC] | I dont see how a fork has anything to do with deployment, though |
| 01:44.24 | brent21 | I now have a lot of FUD |
| 01:44.29 | [hC] | heh |
| 01:44.42 | mog_home | fud about what brent |
| 01:44.46 | [hC] | if something that small is going to scare your PHB's, youll have a hard time giving them an open source product period. |
| 01:44.57 | [hC] | Its not like someone can 'take it away from you' |
| 01:49.45 | brent21 | it just seems that nothing has really changed with * over the past two years |
| 01:49.55 | brent21 | in terms of hardware reliability |
| 01:50.06 | brent21 | believe me, i give digium all the business i can |
| 01:50.14 | brent21 | i just bought two IAXY's over 168V's |
| 01:50.32 | brent21 | we just bought a te100 card, a decision i thought long and hard about over sangoma |
| 01:50.33 | mog_home | asterisk has gotten more reliable lately |
| 01:50.35 | brent21 | went with the te100 |
| 01:50.44 | mog_home | and we are about to make the jump |
| 01:50.50 | mog_home | to high reliability |
| 01:50.51 | brent21 | and i also have 3 tdm cards with FXO mods that just stop working after 20 days |
| 01:51.04 | mog_home | what revs brent |
| 01:51.13 | brent21 | let alone the two IAXY's i just bought yesterday, one of them was DOA |
| 01:51.17 | brent21 | i think rev h |
| 01:51.30 | brent21 | its not a big deal, i jsut reboot the * server every week |
| 01:51.34 | *** join/#asterisk Jhereg_Veshar (i=DEMON@cpe-024-074-113-129.carolina.res.rr.com) |
| 01:51.34 | brent21 | and the client doesnt care/know |
| 01:51.50 | mog_home | well i mean those things shouldnt happen |
| 01:51.51 | brent21 | the rebooting is re: the tdm400B's locking up |
| 01:52.12 | *** part/#asterisk Jhereg_Veshar (i=DEMON@cpe-024-074-113-129.carolina.res.rr.com) |
| 01:52.14 | brent21 | im just concerned, especially with the fork, it seems a lot of the people involved have left |
| 01:52.14 | [hC] | brent21: you gotta be kidding me, asterisk has changed a TON in like, the past 6 months |
| 01:52.25 | brent21 | yet i still to this day still give digium the benefit of the doubt |
| 01:52.37 | brent21 | hC, from a hardware perspective, im not sure it has |
| 01:52.39 | [hC] | who left? |
| 01:52.44 | mog_home | anthm |
| 01:52.47 | mog_home | and company |
| 01:52.56 | mog_home | asterlink in general |
| 01:53.04 | brent21 | im by no means bashing asterisk/digium |
| 01:53.08 | mog_home | i wish we could come back together |
| 01:53.16 | brent21 | i like digium a lot, they have a great idea |
| 01:53.26 | [hC] | i only knew about anthm and bkw |
| 01:53.28 | *** join/#asterisk wolfson (n=hehe@usr-kdh-208-6-58-26.beachlink.com) |
| 01:53.30 | brent21 | its just at the point where this really needs to start being run like a business |
| 01:53.31 | mog_home | whats up file |
| 01:53.39 | msw | mog! |
| 01:53.40 | file[laptop] | nada, u? |
| 01:53.40 | xkev | anyone hooked an EM trunk to 911 tandem switch yet? |
| 01:53.44 | mog_home | msw! |
| 01:53.49 | xkev | beh, not EM, MF |
| 01:54.00 | [hC] | brent21: seen asterisk business edition? |
| 01:54.04 | mog_home | just kicking ass taking names |
| 01:54.25 | brent21 | hC, yeah I have, but that doesn't address the hardware issues does it? |
| 01:54.35 | brent21 | the open source business model is still in its infancy |
| 01:54.38 | mog_home | brent wait a week or so |
| 01:54.58 | brent21 | i am, definitly, and i continue to support and tell the world how great * is |
| 01:55.06 | brent21 | i have too much time and $ invested to do other wise |
| 01:55.12 | NewSole2 | seen asterisk business edition?.... ask for support... its none of your business |
| 01:55.43 | brent21 | I just can't wait for some of the great things that Digium offers to get more mature in the market place with resolutions to a lot of different things |
| 01:55.52 | brent21 | i respect mark a lot, he knows a heck of a lot more than I do |
| 01:56.02 | brent21 | im not even in the same league |
| 01:56.04 | brent21 | never will be |
| 01:56.33 | *** join/#asterisk Inkubot (i=inkubot@120-238-28.dial.terra.cl) |
| 01:56.48 | brent21 | but i just continue to look forward to the day that * becomes a more professional product |
| 01:56.50 | brookshire | brent21: new tdm card is coming out in the next few weeks |
| 01:56.55 | *** join/#asterisk yartelecom (n=no-email@62.33.182.162) |
| 01:56.56 | mog_home | brooks |
| 01:56.59 | brookshire | :) |
| 01:56.59 | mog_home | you spilled the beans |
| 01:57.03 | mog_home | you suck |
| 01:57.04 | brookshire | awh.. damn |
| 01:57.08 | brent21 | haha |
| 01:57.20 | file[laptop] | bad brookshire bad |
| 01:57.25 | brookshire | everyone knows about it! |
| 01:57.30 | brookshire | :) |
| 01:57.41 | mog_home | duh |
| 01:57.49 | mog_home | but its mark's job to spill beans |
| 01:58.03 | mog_home | it would be like if leaked powe |
| 01:58.13 | mog_home | Power over Wireless Ethernet |
| 01:58.19 | xkev | tesla |
| 01:58.21 | brookshire | it does do that |
| 01:58.21 | brookshire | :) |
| 01:58.29 | e-Hernick | unfortunately, wireless power also fries your brain |
| 01:59.44 | brent21 | we just ordered a PRI, this is going to be a fun learning process |
| 01:59.46 | mog_home | who uses those old things anyway |
| 01:59.49 | mog_home | brains that is |
| 01:59.53 | *** join/#asterisk Netgeeks (n=Chris@68-185-24-2.static.mdfd.or.charter.com) |
| 01:59.59 | mog_home | pris are so nice compared to otehr garbage |
| 02:00.10 | e-Hernick | how much for the PRI |
| 02:00.14 | e-Hernick | and did you get a full PRI ? |
| 02:00.15 | xkev | we run our call center on *, it works great. got TDM410P, but analog via channel bank |
| 02:00.33 | xkev | pri > * |
| 02:00.48 | e-Hernick | * > * |
| 02:00.58 | brent21 | pri ended up being like 530 a month |
| 02:01.01 | brent21 | w/ 20 DID's |
| 02:01.03 | xkev | ouch |
| 02:01.08 | brent21 | unlimited local |
| 02:01.10 | e-Hernick | that's nice |
| 02:01.22 | brookshire | brent21: how much for just 20 phone lines? |
| 02:01.27 | e-Hernick | only 26$ per line |
| 02:01.27 | brent21 | .03 LD |
| 02:01.41 | xkev | we pay $250 with 200 DIDs, but then we have 60 PRI too |
| 02:01.45 | brent21 | brookshire, we needed DNIS informatrion |
| 02:01.55 | brookshire | ahh.. ok |
| 02:01.56 | xkev | but I hear you can get ELI PRI for cheap round the $300-350 range |
| 02:01.56 | brent21 | like what number was dialed |
| 02:02.02 | Netgeeks | Question: Is there a method to determine the channel/device used when a dial command is completed? For example, lets say I use Dial(SIP/1&SIP/2&SIP/3), is there a dialplan accessible variable or application that will tell me which SIP device actually answered the dial? |
| 02:02.05 | brookshire | the us needs better bri :( |
| 02:02.10 | e-Hernick | 300-350 ? |
| 02:02.12 | e-Hernick | wow |
| 02:02.19 | brent21 | yeah BRI is a joke, its dead in US |
| 02:02.28 | brookshire | bri would be great though... |
| 02:02.30 | mog_home | yup |
| 02:02.32 | xkev | $150 in quantity :) |
| 02:02.35 | mog_home | bri is pretty lame |
| 02:02.38 | mog_home | in states |
| 02:02.51 | brookshire | mog: i hear in TN isn't not so bad |
| 02:02.51 | schuylerdigium | but in europe... |
| 02:03.07 | brookshire | isn't not |
| 02:03.07 | brookshire | lol |
| 02:03.08 | Vco | anyone know if fritzcard works on bri in japan? |
| 02:03.16 | brookshire | double negatives #1 |
| 02:03.17 | e-Hernick | anybody here got a megalink with bell canada |
| 02:04.01 | brent21 | anyone have any good experiences w/ Tier 1 co-location providers? |
| 02:04.17 | brent21 | i think we have outgrown our managed hosting provider |
| 02:04.19 | xkev | define tier 1 |
| 02:04.31 | brent21 | ATT, Level 3, etc |
| 02:04.45 | xkev | ..there aren't really tiers anymore :) |
| 02:04.53 | brookshire | brent21: i like internap |
| 02:04.59 | brookshire | just because i know half the people that work there |
| 02:05.11 | xkev | I actually recommend something like internap or a smaller multihomed/peering provider, as you'll get best path to * |
| 02:05.20 | brent21 | It seems like w/ managed hosting, we have a ton of linux knowledge in house, and we were troubleshooting and solving issues our Managed hosting provider should have been doing |
| 02:05.22 | brent21 | a waste of money |
| 02:05.23 | xkev | dallas area is good for latency |
| 02:05.37 | brookshire | pheonix is better for everything |
| 02:05.39 | brent21 | it seems like managed hosting is really a rip off |
| 02:05.58 | xkev | brookshire, but dfw is the center of everything |
| 02:06.07 | JunK-Y | MOOO |
| 02:06.08 | brent21 | they sell you on the service, but when it actually comes to performing, e.g. hacking our kernel to make soemthing work, load balancing, failover solutions, etc, managed hosting is no longer the way to go |
| 02:06.19 | brookshire | xkev: but more bandwidth in pheonix ;) |
| 02:06.26 | brent21 | and you pretty much have to take things in to your own hands |
| 02:06.30 | xkev | there's what? :) |
| 02:06.34 | brookshire | easynew's datacenter r0xx0rs ;) |
| 02:06.56 | xkev | easynews has pipe yes, but dallas has every pipe going across the country :) |
| 02:07.22 | brookshire | easynew's network is just rediculous |
| 02:07.32 | xkev | brent21, shameless plug: www.xmission.com/datacenter/ |
| 02:07.45 | brookshire | and because they have like 1,000s of customers, they know instantly if there is a problem |
| 02:08.00 | *** join/#asterisk w0w0 (n=w0w0@36.Red-83-50-231.dynamicIP.rima-tde.net) |
| 02:10.27 | [hC] | We have stuff in puregig (easynews) |
| 02:10.34 | [hC] | We only have about 15gb of transit there though |
| 02:11.08 | brookshire | only 15gb? |
| 02:11.33 | [hC] | In miami we have 35 individual gig peers |
| 02:12.06 | [hC] | DoS attacks are a bandwidth hungry business. :P |
| 02:12.08 | brookshire | miami + wilma ;) |
| 02:12.10 | brookshire | OH NO! |
| 02:12.16 | [hC] | Yeah, im gonna be killed in like, 3 days. |
| 02:12.33 | [hC] | Thats why im moving back to vancouver.. |
| 02:12.38 | *** part/#asterisk schuylerdigium (n=Bosco@pcp03052091pcs.huntsv01.al.comcast.net) |
| 02:12.47 | e-Hernick | you have no chance to survive make your time |
| 02:12.53 | mog_home | so brooks you want to show off res_xmpp? |
| 02:13.27 | brookshire | mog: let me hook up my new box to be the gateway at my house :) |
| 02:13.29 | xkev | alephcom, enough voip is a dos attack :) small packets moving fast to/from distributed places :) |
| 02:13.36 | xkev | s/alephcom/hc/ |
| 02:13.41 | mog_home | sounds good |
| 02:13.56 | [hC] | Yeah, it skewed our shit a little bit when we first deployed * |
| 02:14.03 | *** join/#asterisk kdb (n=kdb@203-206-36-92.dyn.iinet.net.au) |
| 02:14.10 | [hC] | I actually had an entire load balancer 'go away' when i placed a call |
| 02:14.18 | [hC] | because mitigation hardware decided that it was being attacked |
| 02:14.36 | [hC] | Learning that placing a call caused one segment of london to vanish was an interesting 'oops' |
| 02:14.44 | xkev | I've got a colo doing call brokering, he has 100kpps to his rack that moves around the network depending on time of day and where the sun is on the planet |
| 02:16.03 | brookshire | xkev: that must be intresting to bill |
| 02:16.25 | kdb | hi everyone |
| 02:16.41 | *** join/#asterisk shakuhashi (n=shaku@200.103.114.52) |
| 02:16.49 | brookshire | mog: brb |
| 02:17.08 | kdb | i dont suppose anyone could advise me why asterisk isnt allowing me to dial an extension after i've connected to it via voip? |
| 02:17.28 | xkev | brook we just bill him on GB usage, although tiny packets need more cpu in routers, so he kinda costs more than he should |
| 02:17.33 | *** join/#asterisk twisted (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
| 02:17.33 | *** mode/#asterisk [+o twisted] by ChanServ |
| 02:17.39 | e-Hernick | kdb, maybe because your DTMF isn't going through ? |
| 02:17.50 | kdb | any reasons for that? |
| 02:17.54 | e-Hernick | kdb, make sure that your DTMF transmission method is set up correctly. |
| 02:18.11 | kdb | like dtmfmode=rfc2833? |
| 02:18.18 | Netgeeks | kdb:L the sarcastic answer to your question is, because you haven't programmed it correctly to do so.... Otherwise, if you are looking for a better answer, you'll need to provide alot more data |
| 02:18.23 | e-Hernick | yeah, if rfc2833 is appropriate |
| 02:18.38 | e-Hernick | what's your other voip device speaking? |
| 02:18.55 | kdb | like sip, iax, etc? |
| 02:19.04 | e-Hernick | among other things |
| 02:19.07 | e-Hernick | what's your other device |
| 02:19.07 | kdb | sip |
| 02:19.20 | e-Hernick | softphone, sipura, whatever |
| 02:19.21 | e-Hernick | what is it |
| 02:19.26 | kdb | iinetphone |
| 02:19.46 | e-Hernick | no idea what that is, can't help you. you could try a few different dettings for dtmfmode |
| 02:19.55 | desktophero | anyone setup a tieline to a nortel meridian? having problems with that circuit and the immediate setting...thanks |
| 02:20.04 | e-Hernick | what's a tieline |
| 02:20.07 | *** join/#asterisk juice (n=juice@mo-65-40-248-239.dyn.sprint-hsd.net) |
| 02:20.29 | desktophero | i have a t1 connected between a nortel loop and a T1 on a quad 405p card |
| 02:20.30 | Netgeeks | what exactly are you trying to do, kdb? are you trying to dial into asterisk and use DISA to get a internal dial tone? are you daling to some remote number via asterisk and wanting to pass through dtmf to the far end? |
| 02:21.08 | desktophero | for some reason, changing the immediate setting in zapata.conf does not change the behavior of calls going across |
| 02:21.10 | desktophero | that t1 |
| 02:21.11 | kdb | i'm trying to dial into my asterisk box via voip and use a simple ivr |
| 02:21.27 | kdb | works for the softphone i'm using within the lan |
| 02:21.39 | kdb | and when i dialed through a x100p clone |
| 02:23.15 | kdb | and i think when i had asterisk dial my softphone after picking up the call, then using the softphone to redirect or transfer (i dont remember which) the call to, say, voicemailmain |
| 02:26.28 | kdb | anyone? |
| 02:26.39 | kdb | oh well. thanks anyway |
| 02:31.48 | *** join/#asterisk Jzalae (n=sk@216-220-249-131.midmaine.com) |
| 02:32.36 | *** part/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net) |
| 02:32.52 | *** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net) |
| 02:35.49 | marc324 | k |
| 02:37.30 | *** join/#asterisk ACiDV (n=acidvici@26-121.dr.cgocable.ca) |
| 02:38.55 | ACiDV | Hi all =) I have a "problem" with MWI... does it's normal that, when I leave a voicemail, the MWI light on the phone go on before I finish to record the message ? |
| 02:39.12 | alephcom | Has anybody here done anything was an asterisk statistics app? For example, I want to monitor how many calls are regected from various providers and track calls per hour in a bar or line graph. Any thoughts before I write my one? |
| 02:39.59 | ACiDV | alephcom: scopserv have report for call per hour/day/weekday/whatever you want but it's not a free apps |
| 02:40.00 | *** join/#asterisk PBXtech (i=nik@81.sub-70-218-18.myvzw.com) |
| 02:40.31 | *** join/#asterisk asteriskDOTbz (n=logger@64.5.53.45) |
| 02:40.31 | asteriskDOTbz | <PROTECTED> |
| 02:40.57 | Katty | mew? |
| 02:41.41 | alephcom | ACiDV: ok, thanks. Do you think it would make sense to write something? I'm considering it but if it's not needed then... |
| 02:41.53 | alephcom | Probably pull the data from the manager interface. |
| 02:42.07 | *** join/#asterisk docelm0 (n=docelm0@pool-68-238-171-126.tampfl.fios.verizon.net) |
| 02:42.13 | Katty | Netgeeks: :< |
| 02:42.20 | docelm0 | ?? |
| 02:42.23 | docelm0 | What did I walk into |
| 02:42.30 | file | an orgy |
| 02:42.38 | Katty | docelm0: obviously, what file said. |
| 02:43.18 | docelm0 | ya buddy! |
| 02:44.50 | Netgeeks | a very quiet orgy at that |
| 02:45.58 | docelm0 | ohh well.... |
| 02:46.06 | docelm0 | screaming gives me a headache anyhow.. :) |
| 02:47.10 | Katty | Netgeeks: soymilk, plskthx. |
| 02:47.26 | Netgeeks | I'm working on a fairly complex dial plan that is bordering on about 10k lines (40% of which is in-line documentation), and I now take two excedrin before starting even.... heh |
| 02:47.31 | Katty | Netgeeks: warm and with a bit of cinnamon. |
| 02:47.45 | Netgeeks | Soymilk it is |
| 02:47.50 | Katty | :> |
| 02:48.04 | Netgeeks | Hrm, all my cinamon is mixed with sugar right now |
| 02:48.06 | desktophero | 10K lines - whoa... |
| 02:49.26 | ACiDV | Any idea for my MWI "problem" ? I can set checkmwi=60 (default: 10) but dont think it's the best solutin |
| 02:49.33 | docelm0 | ACK! What are you building? |
| 02:49.36 | ACiDV | solutions oops |
| 02:49.58 | docelm0 | I thought my office's PBX was gonna be a pain in the ass.. And anyone looking to buy phones the Grandstream GXS-2000 are SWEET |
| 02:50.00 | xkev | netgeeks I hope you are splitting in multiple files |
| 02:50.13 | xkev | pbx:/etc/asterisk# wc -l ext* |
| 02:50.15 | xkev | <PROTECTED> |
| 02:50.20 | desktophero | ACiDV - i didn't see what the problem actually is. are you looking to change the MWI polling? |
| 02:50.32 | xkev | and I thought that was big |
| 02:50.43 | Katty | solu-tin. now with more solu than ever! comming soon to your grocer's freezer! |
| 02:50.47 | xkev | docelm0, polycom ip600 all the way |
| 02:50.48 | Netgeeks | I'm building a clustering system to provide commercial ITSP and Centrex services using Asterisk... |
| 02:50.57 | xkev | fear |
| 02:51.02 | desktophero | docelm0 - we have about 10 in a remote office. they are not too bad as soon as I updated the firmware |
| 02:51.03 | xkev | want to sell it? :) |
| 02:51.36 | Katty | Netgeeks: do you think someone has already patented the idea of piggybacking onto a pilot wave? |
| 02:51.37 | ACiDV | desktophero, the problem is in the current MWI polling implementation ? must write msg in a temp file before but it in inbox ? |
| 02:51.46 | xkev | I'm building residential class 5 on openser, asterisk as media, but delaying centrex til I can spend lots and lots of time pulling my hair out |
| 02:52.07 | Netgeeks | Katty: knowing the dismal status of our patent system here in the US, I would say yes |
| 02:52.11 | *** join/#asterisk brookshire (n=mbrooks@esbrooks3.traveller.com) |
| 02:52.13 | docelm0 | I got a killer Class4 switch built out of Asterisk and 19 servers |
| 02:52.16 | brookshire | mog: you still here? |
| 02:52.17 | Katty | Netgeeks: :< |
| 02:52.39 | brookshire | hey file! |
| 02:52.42 | file[laptop] | hihi |
| 02:52.52 | desktophero | ACiDV - what type of phone are you using? the grandstreams seem to work well with the default implementation. |
| 02:53.10 | Katty | file[laptop]: :< |
| 02:53.15 | file[laptop] | Katty: eep |
| 02:53.29 | ACiDV | desktophero try with Cisco 7940, Polycom IP600 and BT101 |
| 02:53.37 | Katty | file[laptop]: no point in eeping over spilled soymilk. |
| 02:54.26 | desktophero | i have a Cisco 7960...but not the others. I will have to give that a shot. For the Cisco, what proto are you using to connect to *. I used skinny...but that was hit or miss with some of the features |
| 02:54.46 | xkev | sip of course |
| 02:54.47 | desktophero | from everything I hear, i would love to try out a poly ip600 |
| 02:54.49 | xkev | :) |
| 02:54.57 | xkev | desktophero I have 40 of em |
| 02:55.09 | Netgeeks | hrm, yep, no cinnamon here, but I have a cinnamon stick you can use as a straw for the soy milk |
| 02:55.09 | xkev | they have their quirks, but what doesn't |
| 02:55.43 | Katty | those with quarks. |
| 02:55.46 | desktophero | 40 poly ip600s? cool. unfortunately, budget is a concern ;) And so far, everything I have tried - hard phones and soft phones - have had some sort of issue |
| 02:55.55 | xkev | welcome to sip |
| 02:56.30 | xkev | ip600 @ tritechcoa.com ~250 |
| 02:56.39 | e-Hernick | if you want the phones to work right, you must sacrifice a beefling to the hurricane god of voip |
| 02:56.42 | desktophero | we have had our best results using - dare i say - SJ Phones (free version). XLite was pretty good, but the CPU usage was too high for the rest of the apps we need to run |
| 02:56.46 | xkev | cheap phone == ip300 ~120 |
| 02:56.49 | desktophero | xkev - our budget is ~100 |
| 02:57.16 | e-Hernick | well, for 100$ you can go with either cheap sip phones or cheap analog phones and spa-2002s |
| 02:57.17 | desktophero | ip300 may be an option - thanks....is that price from the same URL? I'll check it out later on |
| 02:57.39 | xkev | yeah they keep lots in stock |
| 02:57.46 | file[laptop] | Katty: how is my favorite Katty doing tonight? |
| 02:57.49 | xkev | if they're out, they'll have more next week. |
| 02:57.53 | desktophero | i hate to break the flow - but anyone set up a nortel meridian to a T1 span on a 405p card? ;) |
| 02:58.00 | xkev | if you wnat to know for sure, call and ask for steven, tell em kevin @xmission sent ya :) |
| 02:58.24 | desktophero | copy and pasting the last line - THANKS xkev! |
| 02:58.28 | e-Hernick | desktophero, is your meridian T1 lines or extensions |
| 02:58.35 | e-Hernick | desktophero, or it doesn't matter ? |
| 02:59.18 | desktophero | T1 lines - I am having an issue with the immediate setting in zapata.conf - or at least i think that is what it is |
| 02:59.33 | *** join/#asterisk Aughey (n=jha@ns1.washucsc.org) |
| 02:59.33 | desktophero | i am trying to get calls to go back and forth |
| 02:59.37 | desktophero | from * to nortel - no issues |
| 02:59.41 | desktophero | nice and easy |
| 02:59.47 | e-Hernick | but when the nortel calls * |
| 02:59.52 | e-Hernick | * doesn't pick up the phone ? |
| 02:59.53 | desktophero | nortel to * - ugh |
| 02:59.57 | marc324 | whats openser? |
| 03:00.04 | e-Hernick | marc324, it's a fork of SER |
| 03:00.07 | desktophero | this may take a couple of lines in irc to explain |
| 03:00.13 | desktophero | here goes |
| 03:00.14 | e-Hernick | desktophero, pastebin ? |
| 03:00.36 | desktophero | from console - i get "starting simple switch on Zap/1-1 |
| 03:00.46 | xkev | openser has lots of pimp shit the ser crew should have got out the door |
| 03:00.48 | e-Hernick | desktophero, if you are hoping to get help, you're going to have to make a nice description of your setup and your problem; IRC isn't appropriate |
| 03:01.03 | e-Hernick | desktophero, put your description in a pastebin and post the link here |
| 03:01.10 | desktophero | as soon as someone starts to dial an exten - lets say 8028, asterisk only picks up the first 8 |
| 03:01.13 | xkev | e.g. LCR module, per-branch flags, UAC module, etc |
| 03:01.13 | nexis | is it possable to set flags with a call file like with the dial string, such as H or L |
| 03:01.39 | desktophero | there is no 8 extension, so i get the common error mesage of no context |
| 03:01.40 | *** join/#asterisk Tili (n=Tili@218.20.52.122) |
| 03:01.50 | desktophero | immediate = yes or immediate = no...same thing |
| 03:02.01 | ACiDV | desktophero problem is with context=... |
| 03:02.03 | e-Hernick | desktophero, that doesn't sound so bad.. why don't you adapt your context |
| 03:02.11 | ACiDV | ? |
| 03:02.12 | desktophero | e-Hernick...i will do that tomorrow - sorry but the T1 is in production |
| 03:02.25 | e-Hernick | desktophero, well, set one or two lines aside |
| 03:02.35 | e-Hernick | desktophero, and use that line for testing, with a different context |
| 03:03.10 | *** join/#asterisk shyru (n=shyru@200.94.130.194) |
| 03:03.11 | desktophero | good point - but don't some zapata.conf changes take a full restart and not just a reload? |
| 03:03.19 | shyru | hi all |
| 03:03.22 | shyru | hi all!! |
| 03:03.31 | e-Hernick | desktophero, well, why don't you keep this to the extensions.conf ? |
| 03:03.34 | Katty | file[laptop]: mewish? |
| 03:03.36 | e-Hernick | desktophero, can you check the callerid ? |
| 03:03.53 | e-Hernick | desktophero, you can bounce to another context based on callerid |
| 03:04.11 | desktophero | e-Hernick - this is a regular T1 and no CID is passed. We are out of PRIs on Nortel to go that route - unless of course there is a way around that ;) |
| 03:04.26 | e-Hernick | ok then |
| 03:04.48 | e-Hernick | can you isolate a line away for testing so that nobody can call it from the nortel |
| 03:04.51 | e-Hernick | except you of course |
| 03:05.03 | e-Hernick | then you use extensions.conf to send this particular line to your test context |
| 03:05.34 | e-Hernick | first thing you want is a way to test your changes without fucking up the system for the rest of your users |
| 03:05.47 | e-Hernick | among other things, does DTMF signaling get through at all ? |
| 03:06.01 | shyru | guys i have a problem :S , when i wake up zaptel's modules, appear this error: unable open /dev/zap/ctl and make reference into /et/zaptel.conf. Any idea? |
| 03:06.02 | marc324 | is openser or iptel-ser the more reliable? |
| 03:06.11 | desktophero | i'll have to see about isolating a single channel based on the prefix code...right now '777' is how you route calls on this particular T1. perhaps I can get our nortel guy to give me another code that points a channel specifically |
| 03:06.13 | e-Hernick | marc324, both are equally reliable |
| 03:06.15 | shyru | i have TDM04B Digium card |
| 03:06.21 | e-Hernick | marc324, I suggest that you flip a coin |
| 03:06.23 | desktophero | DTMF is going through |
| 03:06.45 | marc324 | e-hernick -- did you flip a coin with that answer? |
| 03:06.53 | *** join/#asterisk santiago (n=santiago@208.195.215.158) |
| 03:07.17 | desktophero | shryu - i ran into that problem before...i had to recompile zaptel files, modprobe zaptel and the appropriate module I was using...that seemed to work for me |
| 03:07.23 | e-Hernick | Hi marc324, I'm <rolls dice> ... very happy to see you ! |
| 03:07.37 | desktophero | more like a magic 8 ball ;) |
| 03:07.39 | e-Hernick | desktophero, that's good. So, let's go through the process from the start. I'm a user, I have a nortel digital phone on my desk |
| 03:07.49 | e-Hernick | desktophero, I dial an extension which is routed to the * through the T1 |
| 03:07.50 | desktophero | e-Hernick - cool! |
| 03:07.53 | shyru | desktophero, oki thanks! |
| 03:07.54 | desktophero | yes |
| 03:07.58 | xkev | old iptel ser 0.8.x is most reliable, but missing good features |
| 03:08.01 | e-Hernick | desktophero, * gets the first digit of the extension, nothing else. |
| 03:08.06 | desktophero | exactly |
| 03:08.18 | e-Hernick | does * get the first digit of the extension after the user has finished dialing |
| 03:08.23 | desktophero | here is the work around - we also have symposium |
| 03:08.23 | e-Hernick | or before he has finished dialing |
| 03:08.38 | desktophero | the user does not get done dialing |
| 03:08.47 | e-Hernick | what do you mean ? |
| 03:08.52 | xkev | openser 0.9.x good for both concerns, but a recent post to openser-devel says there may be an issue which was fixed in cvs recently |
| 03:08.52 | Aughey | what's the range for txgain in /etc/asterisk/zaptel.conf? I've adjusted it a lot without any noticable difference. |
| 03:09.01 | desktophero | from my example, the 8 is trapped and the call drops because there is no 8 |
| 03:09.07 | e-Hernick | ok |
| 03:09.12 | e-Hernick | can I see your extensions.conf context |
| 03:09.27 | desktophero | symposium - when a person dials an extension, symposium is able to 'dial fast enough' to get the call through |
| 03:09.30 | desktophero | works all the time |
| 03:09.37 | desktophero | so - dial 8028 from your nortel phone |
| 03:09.49 | desktophero | symposium then dials 7778028 and the call goes through |
| 03:09.58 | Katty | nortel seems like a good name for a turtle |
| 03:10.21 | e-Hernick | so the second I finish pressing 8 on my end-user phone, * gets the call at extension 8 ? |
| 03:10.25 | Katty | oh, i know what i'm thinking |
| 03:10.28 | desktophero | Katty - that reminds me of a Dana Carvey movie - am I too turtley for the turtle club |
| 03:10.29 | Katty | norvil, from oddballz. |
| 03:10.30 | *** join/#asterisk brookshire (n=mbrooks@esbrooks3.traveller.com) |
| 03:11.02 | desktophero | e-Hernick - yep...keep in mind that you have to dial 777 first...so 777 then 8028. The 8 is the only thing that gets trapped |
| 03:11.12 | e-Hernick | ah |
| 03:11.26 | e-Hernick | ok, does * get the call when I press the last 7 or when I press the first 8 |
| 03:11.33 | desktophero | symposium can dial the digits fast enough for * to recognize |
| 03:11.43 | desktophero | the last 7 - 'starting simple switch on...' |
| 03:11.52 | e-Hernick | ah |
| 03:11.57 | shyru | desktophero, does not ruls :S |
| 03:12.00 | e-Hernick | can I see your extensions.conf context into which this get routed ? |
| 03:12.12 | e-Hernick | Cause afaik your setup is nearly working fine, as you say that DTMF gets through |
| 03:12.14 | desktophero | shryu - huh? |
| 03:12.19 | e-Hernick | I feel that your extensions.conf is broken |
| 03:12.49 | desktophero | e-Hernick - working on a pastebin right now |
| 03:13.04 | e-Hernick | good |
| 03:13.11 | shyru | desktophero, im trying fixs my problem with zaptel and nothig :S |
| 03:13.12 | shyru | :S |
| 03:13.19 | e-Hernick | desktophere, copy-paste from our little chat |
| 03:13.37 | e-Hernick | you've defined your problem pretty well |
| 03:13.58 | e-Hernick | how good is your asterisk-fu anyway ? |
| 03:14.11 | *** join/#asterisk trelane_ (n=trelane@66.93.203.199) |
| 03:14.43 | brookshire | shyru: what kinda problems you having? |
| 03:15.13 | *** join/#asterisk SwK (i=dzybec@12-219-144-126.client.mchsi.com) |
| 03:16.31 | *** join/#asterisk Insanity5 (n=feaw@ip68-105-215-25.tl.dl.cox.net) |
| 03:16.31 | desktophero | e-Hernick - I have been dealing with * for about 4 months. I have about 6 servers with LCR and about 120 calls....I think I did pastebin right (1st time) http://pastebin.com/400536 |
| 03:16.34 | Insanity5 | Is there anyway when you do a goto(submenu) to keep out the extensiosn from the main menu? |
| 03:16.54 | trelane_ | someone want to help me find my iaxy? It's on the network, I've reset the config via the little button and yet it is not attempting to pull DHCP |
| 03:17.56 | shyru | brookshire, i did compile zaptel with make linux26&&make install then i did put modprobe wcfxs modprobe ztdummy modprobe ztdynamic on rc.local; i have FC3 and config files for udev permissions and whe i reboot FC3 box i have error: unable open /dev/zap/ctl |
| 03:17.59 | e-Hernick | who wait what is this crazy thing |
| 03:18.03 | desktophero | please be kind re: my extensions.conf. so far I have been able to get stuff working ;) |
| 03:18.12 | e-Hernick | okay okay now that's your first problem |
| 03:18.17 | e-Hernick | you are trying to debug a very complex context |
| 03:18.29 | e-Hernick | you need to do as I suggested; set one line aside and give it its own context |
| 03:18.36 | e-Hernick | then you can start from scratch |
| 03:18.39 | desktophero | i will work with the nortel guy on that |
| 03:18.41 | e-Hernick | cause this context is a mess |
| 03:18.46 | e-Hernick | you don't need the nortel guy |
| 03:18.56 | e-Hernick | wait, maybe you do |
| 03:18.57 | shyru | brookshire, any idea? |
| 03:19.05 | e-Hernick | can't you dial using the line of your choice ? |
| 03:19.11 | e-Hernick | just use the last line in the pool |
| 03:19.40 | brookshire | shyru: run modprobe zaptel |
| 03:19.45 | brookshire | does it give you any errors? |
| 03:19.46 | *** join/#asterisk Darwin35 (n=kvirc@c-67-190-93-187.hsd1.co.comcast.net) |
| 03:20.24 | desktophero | but the call comes from nortel...i could force the call going TO nortel on my side. i need him to be able to go code based within a T1 - like 777 is for channels 1-23, 778 is channel 24 - that is what you are thinking, right? |
| 03:20.44 | e-Hernick | yes, for testing purposes |
| 03:20.49 | shyru | brookshire, no |
| 03:20.53 | Insanity5 | can anyone hlep me with dialplan 101? |
| 03:21.02 | e-Hernick | I can help you with dialplan 102. |
| 03:21.04 | desktophero | cool - i'll see if i can get that going tomorrow - finger crossed |
| 03:21.10 | shyru | brookshire, does not appear error |
| 03:21.14 | brookshire | shyru: what does dmesg say? |
| 03:21.28 | e-Hernick | desktophero, also, I strongly recommend that you plan your dialplan in plain english |
| 03:21.30 | Insanity5 | I have a menu, and submenu. with a goto submenu when you dail an extnension. For same reason, it dials out of the main menu and/or default instead of the submenu. |
| 03:21.31 | desktophero | shryu - can you modprobe your card's module? |
| 03:21.38 | shyru | brookshire, Registered tone zone 0 (United States / North America) |
| 03:21.46 | e-Hernick | desktophero, make a clear description of your dialplan and then translate it to extensions.conf |
| 03:21.49 | desktophero | e-Hernick - what do you mean? |
| 03:21.55 | desktophero | pseudo-code? |
| 03:21.58 | e-Hernick | Yes. |
| 03:22.06 | brookshire | shyru: try to modprobe your card now |
| 03:22.14 | e-Hernick | You *need* pseudo-code, because your extensions.conf is poo. |
| 03:22.15 | desktophero | i will have to go back to the drawboard a little for that...i totally agree though! |
| 03:22.21 | shyru | oki brookshire |
| 03:22.22 | *** join/#asterisk gorauskas (n=gorauska@69.111.163.160) |
| 03:22.22 | desktophero | ouch! |
| 03:22.28 | desktophero | ;) |
| 03:22.38 | e-Hernick | I mean, it's copy-paste heaven with half the code commented out |
| 03:22.42 | e-Hernick | And useless comments |
| 03:22.55 | e-Hernick | You want to learn how to use variables |
| 03:23.04 | e-Hernick | And context switches |
| 03:23.11 | e-Hernick | Goto |
| 03:23.16 | wundaboy | how do i setup a menu system? |
| 03:23.25 | e-Hernick | It's awful I know, gotos suck, but it's much better than your current thingy |
| 03:23.31 | e-Hernick | much better |
| 03:23.36 | e-Hernick | 1000* better |
| 03:23.54 | e-Hernick | wundaboy, if you're not aiming for a production system just yet, I highly suggest * 1.2-beta1 and AEL |
| 03:24.00 | desktophero | on my other servers I do use variables as well as Goto and other functions....i need to spend some time and revamp some of my older contexts vs. new stuff I learn |
| 03:24.08 | wundaboy | no, my phone is on this guys pbx |
| 03:24.11 | e-Hernick | AEL is so awesome compared to extensions.conf |
| 03:24.11 | Insanity5 | AEL? |
| 03:24.11 | wundaboy | and i want to setup a menu |
| 03:24.13 | wundaboy | and i dont know how |
| 03:24.18 | wundaboy | he said it was gsm format? |
| 03:24.20 | e-Hernick | Yeah, AEL rules |
| 03:24.24 | Insanity5 | What is it? |
| 03:24.27 | wundaboy | gsm |
| 03:24.28 | e-Hernick | extensions.ael |
| 03:24.32 | wundaboy | i think? |
| 03:24.35 | e-Hernick | it replaces extensions.conf in asterisk 1.2.0-beta1 |
| 03:24.39 | wundaboy | great |
| 03:24.44 | wundaboy | but how do i setup the menu system? |
| 03:24.47 | Insanity5 | I did made head like a month ago and I don't have it. |
| 03:24.48 | desktophero | shryu - did the modprobe on the module for your card work? |
| 03:25.03 | wundaboy | like its a document right |
| 03:25.07 | wundaboy | like press 1 for this |
| 03:25.09 | wundaboy | and press 2 for that |
| 03:25.16 | Netgeeks | actually it doesn't replace extensions.conf, you can use either |
| 03:25.21 | Insanity5 | e-Hernick - can you put a sample .ael on pastebin? |
| 03:25.38 | e-Hernick | http://pastebin.com/400544 |
| 03:25.42 | e-Hernick | this is a part of my IVR system |
| 03:25.42 | Insanity5 | e-Hernick - Anyways, can you help me with extension hell? |
| 03:25.43 | desktophero | http://www.voip-info.org/wiki/index.php?page=Asterisk+AEL |
| 03:25.59 | e-Hernick | my full IVR is 514 lines of AEL |
| 03:26.12 | Insanity5 | e-Hernick - More programmer like, less like common english. |
| 03:26.16 | e-Hernick | Yes, it rules. |
| 03:26.18 | Netgeeks | it allows you to write dial plan code ins something that looks much more like programming code than the old dialplan method |
| 03:26.22 | e-Hernick | It's 1000 times better than extensions.conf |
| 03:26.49 | Insanity5 | e-Hernick - Anyawys, can you help me with extension 101? |
| 03:26.59 | e-Hernick | Insanity5, what's your question ? |
| 03:27.08 | desktophero | e-Hernick - reminds me of JavaScript or PHP - more inline code as opposed to exten =>...that makes more sense to me. I will ahve to check that out |
| 03:27.19 | e-Hernick | desktophero, yes, it rules |
| 03:27.21 | e-Hernick | desktophero, a lot |
| 03:27.28 | Insanity5 | e-Hernick - I have 3 extensions, 1 2 and 3. My IVR has an option -- press 8 and it goes a goto(submenu). that submenu has an option 2 -- but it's going to extension 2, not that option. |
| 03:28.08 | desktophero | only beta 1.2 though - right? |
| 03:28.08 | e-Hernick | yes |
| 03:28.08 | e-Hernick | soon it won't be beta no more |
| 03:28.08 | desktophero | i'll have to set up a beta machine with that on it |
| 03:28.08 | Insanity5 | e-Hernick - some day :) |
| 03:28.08 | FuriousGeorge | if i send my 240 local calling extension-exchanges to an included file in my dialplan, can i call macros in the original file from the included file? |
| 03:28.09 | desktophero | any timelines? |
| 03:28.09 | desktophero | for it going prod? |
| 03:28.09 | e-Hernick | desktophero, when it's ready |
| 03:28.12 | desktophero | gotcha |
| 03:28.41 | e-Hernick | Insanity5, okay. Are you using multiple contexts and Goto ? |
| 03:28.47 | e-Hernick | Insanity5, pastebin your extensions.conf |
| 03:29.01 | xkev | one plenty depends on what you're doing. there is no turnkey solution in open source land |
| 03:29.21 | *** join/#asterisk mcf3782 (n=mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net) |
| 03:29.23 | xkev | <PROTECTED> |
| 03:29.49 | desktophero | shryu - any luck on the second modprobe? |
| 03:29.59 | e-Hernick | I mean, my extensions.* programming style involves lots of contexts and lots of gotos |
| 03:30.05 | *** join/#asterisk websae (n=websae@CPE-24-167-204-30.wi.res.rr.com) |
| 03:30.11 | e-Hernick | Each user action means a context switch |
| 03:30.47 | e-Hernick | I wouldn't try to do it any other way. |
| 03:30.54 | Insanity5 | e-Hernick - http://pastebin.com/400553 |
| 03:31.10 | e-Hernick | okay okay okay |
| 03:31.12 | e-Hernick | start with a clean extensions.conf |
| 03:31.19 | e-Hernick | don't simply modify the example file |
| 03:31.24 | e-Hernick | keep *only what you need* |
| 03:31.30 | e-Hernick | otherwise it's impossible to understand |
| 03:31.43 | *** join/#asterisk supaigtr (n=yurplsl@152.53.17.1) |
| 03:31.50 | desktophero | i am totally agreeing with e-Hernick right about now |
| 03:31.56 | Insanity5 | Do you need all that crap at the top for it to work? |
| 03:32.07 | e-Hernick | if you use it |
| 03:32.07 | desktophero | the default is/was a little chinese to me - no offense....;) |
| 03:32.10 | e-Hernick | most of it you don't |
| 03:32.23 | e-Hernick | do you want freeworlddialup and iaxtel ? |
| 03:32.25 | mcf3782 | I need to go through my extensions.conf file and strip out all the example stuff. e-Hernick is right, it's gotten a bit confusing trying to sort through it. |
| 03:32.52 | e-Hernick | You cannot use * with the default files. They're only examples. If you don't strip out the useless parts, you'll not be able to understand how your own * works |
| 03:33.03 | e-Hernick | strip out all you can |
| 03:33.07 | e-Hernick | keep a copy of the original file |
| 03:33.10 | mcf3782 | The examples were very helpful to me to get up and running. But now I need a little less noise. |
| 03:33.14 | Insanity5 | e-Hernick - stand by :) |
| 03:33.16 | e-Hernick | if you stripped out something important, add it back |
| 03:33.20 | supaigtr | Hello ppl. |
| 03:33.23 | Insanity5 | hehe |
| 03:33.25 | xkev | someone needs to cook up a good extensions.conf sample and actually contribute it back for inclusion |
| 03:33.34 | xkev | mine is too much spaghetti |
| 03:33.41 | e-Hernick | What we need is an example configuration bank |
| 03:33.45 | xkev | but something that actually implements the features listed on asterisk.org :) |
| 03:33.45 | e-Hernick | And extensions.conf is dead |
| 03:33.50 | e-Hernick | AEL is the real deal |
| 03:33.55 | xkev | yeah ael++ |
| 03:33.58 | Insanity5 | e-Hernick - I dont' know if AEL is for everyone :) |
| 03:34.07 | mcf3782 | AEL? |
| 03:34.09 | e-Hernick | Nobody is gonna use extensions.conf in a few months except for legacy users |
| 03:34.27 | Insanity5 | e-Hernick - Why is that? Is there anything you can't do with extensiosn.conf that you can with ael? |
| 03:34.31 | supaigtr | AEL is the format all other softswitches use. |
| 03:34.31 | desktophero | based on e-Hernick's pastebin...I will be switching over as soon as 1.2 is prod |
| 03:34.34 | e-Hernick | Nobody who has a choice between AEL and extensions.conf has the right to use extensions.conf without being declared insane and shunned by the support community |
| 03:34.51 | Insanity5 | e-Hernick - Well how do I know if my build supports ael? |
| 03:34.52 | desktophero | don't hold back e-Hernick |
| 03:35.16 | trelane | someone want to help me find my iaxy? It's on the network, I've reset the config via the little button and yet it is not attempting to pull DHCP |
| 03:35.20 | supaigtr | Anyone know a trick way to debug IAX2 - IAX2 dropped calls. Very random and really a problem. |
| 03:35.24 | e-Hernick | extensions.conf is a piece of shit, the syntax is more ugly than the bastard child of Bill Gate's GWBASIC mixed with LISP, FORTRAN and MS BOB SCRIPT |
| 03:35.44 | e-Hernick | I hate extensions.conf... with fire |
| 03:35.55 | Insanity5 | e-Hernick - Ok, well would you rather help me with ael or .conf? |
| 03:35.59 | mcf3782 | I wouldn't go so far as to suggest shunning someone just because they prefer one format over another.. that's getting very close to the "emacs versus vi" religious wars, IMO. |
| 03:36.08 | e-Hernick | Well, if you're using 1.2 already |
| 03:36.20 | Insanity5 | I don't know if it's b1, but it's 1.2 |
| 03:36.22 | e-Hernick | I highly suggest that you upgrade to extensions.ael |
| 03:36.36 | e-Hernick | I mean, your dialplan isn't very complex, but it's a mess |
| 03:36.40 | e-Hernick | Okay, first rule of making a dialplan. |
| 03:36.44 | e-Hernick | Write it in pseudo-code. |
| 03:36.49 | Insanity5 | I have a blank document. |
| 03:36.51 | e-Hernick | Or in plain english, whichever you prefer. |
| 03:37.12 | e-Hernick | But don't try to write your dialplan in AEL straight away if you haven't touched AEL before. |
| 03:37.34 | e-Hernick | You want to decide what it's gonna do and then find out how you can do it. If you try writing the code straight away, you'll do what you find easier to do with the code. |
| 03:37.42 | e-Hernick | Rather than what you want to do. |
| 03:38.06 | mcf3782 | Where can I see an example of AEL as it pertains to Asterisk (and not some other switch/pbx)? |
| 03:38.13 | supaigtr | Reality is * needs a RAD package to control the system. |
| 03:38.32 | e-Hernick | mcf: http://pastebin.com/400544 |
| 03:38.38 | e-Hernick | this is part of my * dialplan |
| 03:38.40 | mcf3782 | thanks |
| 03:39.27 | mcf3782 | eww |
| 03:39.45 | e-Hernick | I know, it looks like the bastard child of Perl, BASH and extensions.conf |
| 03:39.52 | e-Hernick | And BASIC, too |
| 03:39.56 | e-Hernick | But it's better than extensions.conf |
| 03:40.25 | mcf3782 | No offense intended, but I think "better" is relative and a preference. |
| 03:40.31 | e-Hernick | No. |
| 03:40.51 | Insanity5 | e-Hernick - ok hows this look? http://pastebin.com/400563 |
| 03:40.55 | e-Hernick | My preference is absolute; in my world, AEL is not just a good idea, it's the law. |
| 03:41.16 | mcf3782 | ;) |
| 03:41.19 | *** join/#asterisk brookshire (n=mbrooks@esbrooks3.traveller.com) |
| 03:41.38 | e-Hernick | Insanity5, much better |
| 03:41.54 | e-Hernick | So, what are you trying to do |
| 03:42.50 | Insanity5 | e-Hernick - 3 people living in a house -- asterisk. |
| 03:43.11 | Insanity5 | e-Hernick -call in, 1 for paul, 2 for fred, etc |
| 03:43.21 | mcf3782 | I like your use of Festival for prompting the user. |
| 03:43.22 | e-Hernick | Okay, what's not working ? |
| 03:43.57 | Insanity5 | e-Hernick - dial 8600, the recorder is broke. press 1 to save and it dials paul |
| 03:44.21 | e-Hernick | Don't you mean 8400 ? |
| 03:44.30 | Insanity5 | ya, whatever it its. |
| 03:44.56 | xkev | "Recordings" != "recordings" |
| 03:45.20 | mcf3782 | D'oh. case sensative. |
| 03:45.24 | mcf3782 | I hate it when I do that. |
| 03:45.24 | e-Hernick | try that first, yes. |
| 03:45.33 | e-Hernick | And also, group your extensions by numerical order |
| 03:46.09 | mcf3782 | numerical order within a context sounds like a good idea. |
| 03:46.25 | e-Hernick | Me, when I have any extension that requires more than a single line in extensions.conf, I give it its own context |
| 03:47.09 | shyru | damm!! unable to open master device /dev/zap/ctl :S |
| 03:47.10 | xkev | also I suggest you never put dial-out rules in [default] |
| 03:47.54 | mcf3782 | Does anyone know if there is a person/place/site that is collecting "cookbook" or "how to do ......" type examples of things to go in extensions.conf? Sounds like a good idea to me. |
| 03:47.56 | xkev | I could use your nufone just by calling SIP/you/a_number if you didn't change sip.conf default context |
| 03:48.16 | e-Hernick | I don't know if you have programming experience, but extensions.* is just another programming language. Don't use line numbers for functions. Each context is a function, make lots of them |
| 03:48.26 | shyru | :'( |
| 03:48.34 | xkev | in fact, I never use default. I change sip.conf context=anonymous and put something that warns me loudly that I missed something that fell back to default |
| 03:49.02 | xkev | let's say I send a manager redirect that barfs, and then it continues to fall back until it drops into default. I ran into that many times |
| 03:50.18 | xkev | a friend of mine is setting up a small home thing like you, and I pointed him in the direction of a [default] context for incoming calls, then a [home] context that the sip phones "dial" into, then branch off from there |
| 03:50.45 | xkev | so default has extens 's' and his DIDs, which jump over to a menu context, etc. |
| 03:51.09 | *** join/#asterisk bmg505 (n=leon@rndf-146-18-84.telkomadsl.co.za) |
| 03:51.10 | *** join/#asterisk Netgeeks_ (n=Chris@68-185-24-2.static.mdfd.or.charter.com) |
| 03:52.20 | *** part/#asterisk Netgeeks_ (n=Chris@68-185-24-2.static.mdfd.or.charter.com) |
| 03:54.17 | *** join/#asterisk gorauskas (n=gorauska@adsl-69-224-19-95.dsl.pltn13.pacbell.net) |
| 03:55.01 | desktophero | shryu - what is the zaptel device you are trying to load - a TDM? if so, run 'lspci -vvvv' and see if anything shows up about Tiger Jet Network..or something like that |
| 03:59.07 | supaigtr | Anynoe know what iax2 notransfer is? |
| 03:59.21 | Aughey | just curious, what if it showed a Tiger jet Network device? |
| 03:59.36 | e-Hernick | oh gnoes the tiger jet has taken over your asterisk ! |
| 03:59.42 | e-Hernick | run, run for the hills ! |
| 04:00.26 | desktophero | e-hernick - my TDM400P shows as a Tiger Jet Network device |
| 04:00.47 | e-Hernick | too late ! |
| 04:00.52 | desktophero | i have a x100p clone on another box that shows as an unknown device |
| 04:00.56 | desktophero | that is my home box |
| 04:01.06 | desktophero | the calls on the TDM400P are actually not too bad.... |
| 04:01.16 | desktophero | the x100p clone SUCKS! |
| 04:01.20 | desktophero | at least the one i have |
| 04:01.51 | desktophero | supaigtr - from my understanding, that is equivalent to the sip setting of canreinvite |
| 04:01.52 | Aughey | my x100p is quiet (other people here me about 1/2 volume). I've tried to adjust txgain, but that doesn't seem to do anything |
| 04:02.46 | desktophero | Aughey - i get noise and sounds from my PSTN line that make the sound quality T-E-R-R-Ible. ztmonitor shows constant noise on the receiving end |
| 04:02.50 | mcf3782 | I'm saving my pennies to buy a TDM400P with an FXO and an FXS module on it. If anyone here has one they're not using or want to sell, let me know. |
| 04:03.12 | e-Hernick | I like the SPA-2002 and SPA-3000 |
| 04:03.20 | e-Hernick | So cheap, and yet so cuddly. |
| 04:03.45 | desktophero | e-Hernick - over any of the TDM cards, right? |
| 04:03.53 | mcf3782 | Those less expensive than the TDM cards? |
| 04:03.58 | e-Hernick | Oh yeah. |
| 04:04.09 | e-Hernick | 80$ buys you two FXS in an SPA-2002 |
| 04:04.16 | mcf3782 | wow |
| 04:04.21 | e-Hernick | and 100$ buys you one FXS, one FXO in a SPA-3000 |
| 04:04.24 | mcf3782 | Got a source? |
| 04:04.35 | e-Hernick | I get mine at ncix or voxilla |
| 04:04.40 | e-Hernick | but just froogle for it |
| 04:05.19 | e-Hernick | works good with * too |
| 04:05.35 | desktophero | any luck with a fax line and an SPA? |
| 04:05.54 | e-Hernick | no attempt yet |
| 04:05.56 | supaigtr | desktophero: Any chance its causing dropped call on PRI - IAX - IAX - SIP? |
| 04:06.04 | supaigtr | T.38 passthru? |
| 04:06.33 | e-Hernick | don't like fax, don't care much for fax. usually give it a pariah dedicated phone line for it, the DSL and the alarm system |
| 04:07.01 | desktophero | any chance the TDM card is causing dropped calls? or fax? |
| 04:07.05 | e-Hernick | small price to pay for not dealing with fax crap |
| 04:07.07 | mog_home | i have fax over ip, its called email |
| 04:07.09 | mog_home | it rocks |
| 04:07.24 | e-Hernick | wow where can i so obtain this new fangled "electronic mail" thing |
| 04:07.37 | SwK | T.37 Rules! |
| 04:07.38 | supaigtr | desktophero: I've tried everything. I can't even debug it. |
| 04:07.47 | e-Hernick | V.42bis sucks! |
| 04:08.00 | desktophero | supaigtr - debug what...sorry I am not sure what you are refering to? |
| 04:08.00 | SwK | T.37 > V.* |
| 04:08.06 | supaigtr | Other TDM calls don't seem to get dropped. Just when they goto another box. |
| 04:08.13 | e-Hernick | X.2? > T.?7 |
| 04:08.37 | supaigtr | I don't have a good way to debug or even find the problem. |
| 04:08.55 | desktophero | supaigtr - when in our remote office, we have a SIP server for local calls then use an IAX2 trunk to our main SIP server |
| 04:08.56 | e-Hernick | supa_thygar, what is the problem |
| 04:09.05 | desktophero | are you dropping calls between the servers over the trunk? |
| 04:09.14 | mog_home | e-Hernick it is beta though |
| 04:09.22 | supaigtr | I can't really tell. |
| 04:09.31 | supaigtr | I've never caught a call. |
| 04:09.49 | e-Hernick | supaigtr, first thing you gotta do is find a way to reproduce the problem. devise a torture test that reproduces the problem for sure |
| 04:09.57 | desktophero | you mentioned PRI -are you using a TE11XP card for the PRI? |
| 04:10.01 | e-Hernick | supaigtr, with no test, there ain't no fix |
| 04:10.06 | supaigtr | TE411p |
| 04:10.36 | desktophero | so a call comes into your PRI circuit, goes from one server to another and then ends at a SIP destination? |
| 04:10.39 | supaigtr | e-Hernick: I realize that. All it get is about 15 complaints a day with no way to figure out what the problme is. |
| 04:10.40 | xkev | mcf3782 cookbook/howto see voip-info.org and post em up |
| 04:10.53 | supaigtr | desktophero: Exactly |
| 04:11.07 | desktophero | are you using the 'full' log from logger.conf? |
| 04:11.22 | desktophero | are you getting any errors that the iax2 peer is 'unreachable'? |
| 04:11.22 | e-Hernick | supaigtr, first things first then. make a test. make it crap on you at will - can't fix it before then |
| 04:11.30 | mcf3782 | xkev I'll look around on voip-info.org. Thanks. |
| 04:11.48 | supaigtr | I am also using that box with direct sip no iax and don't have problems with that. Seems to be only pri IAX2 - IAX2 - Sip calls. Its intermittent and seems to happen more on cell calls. |
| 04:12.33 | desktophero | what about 'unreachable' errors between the IAX peers? |
| 04:12.43 | supaigtr | e-Hernick: I've tried everything. Loaded up calls. Generated traffic. All I've even gotten in a controled envirnoment is packet loss but not a dropped call. But I know it happening. |
| 04:13.05 | e-Hernick | okay then, can you pinpoint the moments in time where the dropped calls happen ? |
| 04:13.19 | e-Hernick | you need to get a user to tell you the exact time where his call drops |
| 04:13.24 | supaigtr | desktophero: I've never seen that and tcp/ip testing with ping ever 2 or 3 seconds shows no loss of connectivity. |
| 04:13.26 | e-Hernick | then you can go see that precise call in the logs |
| 04:13.59 | e-Hernick | 2-3 second ping ? |
| 04:14.06 | e-Hernick | do you think maybe it's the other end that's dropping the call |
| 04:14.11 | mcf3782 | Enough for me for the night. I need to go crawl between the sheets. G'nite folks. Thanks for the suggestions. |
| 04:14.32 | e-Hernick | like, the other end see something it don't like, weird silence or such, and drops out |
| 04:14.39 | supaigtr | every 2 or 3 secs I have process that ping tests connecitivy. |
| 04:14.52 | supaigtr | e-Hernick: Thats what I am thinking. |
| 04:14.54 | *** part/#asterisk mcf3782 (n=mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net) |
| 04:15.13 | e-Hernick | when in the call does the dropping occur |
| 04:15.16 | e-Hernick | at random ? |
| 04:15.35 | desktophero | are you using the 'full' log? sometimes that shows stuff that i don't catch on the console |
| 04:15.46 | supaigtr | very random. So random I've tested for days without it occuring myself. Some users also report going to on-hold music the the call returns. |
| 04:15.49 | e-Hernick | yeah, and do you know where the call gets dropped first |
| 04:16.11 | e-Hernick | weird. |
| 04:16.18 | supaigtr | This system has like 4 PRI's mostly loaded during the day. Full log is unreadable unless I know what I am looking for. |
| 04:16.23 | e-Hernick | 1. you need to sync the clocks of all your servers very precisely with ntp |
| 04:16.30 | e-Hernick | 2. you need to gather full logs from all systems |
| 04:16.38 | supaigtr | sync? |
| 04:16.46 | e-Hernick | 3. get from a user a timestamp and call information for a dropped call |
| 04:17.01 | e-Hernick | yeah, you can't have 3 seconds of drifts on the server clocks |
| 04:17.05 | desktophero | good call on ntp syncing...i ended up witha circuit not being able to 'connect' at all until i got the timing between the two servers sync'd |
| 04:17.16 | e-Hernick | you wanna know where it drops first, you need servers with very close clocks |
| 04:17.18 | xkev | how about load/interrupt handling, crons, disk i/o? |
| 04:17.21 | xkev | is pri zaptel? |
| 04:17.23 | supaigtr | Hmm. Thats likely an issue. Lemme check. |
| 04:17.33 | e-Hernick | but you want the clock in sync so the logs make sense |
| 04:17.58 | desktophero | supaigtr - if you get an extension and perhaps a CID you might be able to grep out a section of the log to see if there are any errors prior in the log itself |
| 04:18.02 | e-Hernick | so you get full logs, and once you get a user report with necesary info (call destination, user extension, timestamp) you can trace the call in the logs |
| 04:18.02 | supaigtr | the PRI is zaptel a 411p. Other machine has a 410 but its disabled and using ztdummy at the momemnt. same result if I enable the card in it. |
| 04:18.13 | xkev | are you doing any manager port voodoo that might be sending the wrong channels around? |
| 04:18.22 | supaigtr | No manager port |
| 04:19.15 | desktophero | xkev - i ran a dual cpu server without the smp process for about 2 days - could not figure out where my IRQ errors where coming from...drove me nuts! grub "somehow" defaulted to the non-smp kernel...just commenting on the interrupt handling |
| 04:19.17 | supaigtr | Basically I have a single system with PRIS and remote sites via fiber. |
| 04:19.44 | supaigtr | It may be interupts but how to check? |
| 04:19.54 | desktophero | if i could reach my own butt with a foot- ida kicked myself |
| 04:20.15 | desktophero | more /proc/interrupts |
| 04:20.28 | *** join/#asterisk alephcom (n=Miranda@207.34.97.130) |
| 04:20.30 | desktophero | from your linux command line |
| 04:20.39 | xkev | I have a manager app that runs an operator panel and queue call picker thing. the old code used to lose track of channels (thank you Event: Rename) and fuck shit up real good |
| 04:20.40 | supaigtr | CLocks off by an hour. |
| 04:20.42 | supaigtr | :) |
| 04:20.48 | desktophero | DOH! |
| 04:21.00 | supaigtr | Wonder if that could be the problem with dropped calls? |
| 04:21.03 | xkev | operator would transfer the same call through the system over and over |
| 04:21.06 | e-Hernick | clocks off probably ain't your problem |
| 04:21.14 | e-Hernick | but with clocks synced you can merge logs |
| 04:21.22 | e-Hernick | and then you can know what servers drops the call, why and how |
| 04:22.09 | desktophero | i had issues between servers when the clocks were off seven minutes...the clock on the server does not provide any timing? that seems strange to me... |
| 04:22.41 | desktophero | maybe i just lucked out... |
| 04:22.48 | supaigtr | http://pastebin.ca/26138 |
| 04:22.53 | supaigtr | My interrupts |
| 04:23.21 | xkev | yuck, XT-PIC |
| 04:24.04 | desktophero | can you force your eth0 to another IRQ in the bios - give it its own? do you need the ps/2 mouse? |
| 04:24.24 | supaigtr | BIOS isn't that powerful. |
| 04:24.50 | marc324 | ne1 can help for ser installation? |
| 04:25.04 | trelane | should iaxy's respond to ping? |
| 04:25.08 | FuriousGeorge | everything works but how do i find out what this referrs to: Oct 21 00:21:19 WARNING[28429]: pbx.c:651 ast_extension_match: Wrong usage of [] in the extension |
| 04:25.08 | supaigtr | I don't think I can disable the ps/2 mouse. I'm not having dropped calls on the 411p enable machine from PRI - SIP if that makes a difference. |
| 04:25.25 | supaigtr | marc324: I kinda have ser working. |
| 04:25.31 | desktophero | supaigtr - your ide is only at 4. that seems strange.... |
| 04:25.43 | supaigtr | Its SATA |
| 04:26.00 | supaigtr | I don't think anything but cdrom is IDE. |
| 04:26.07 | desktophero | on the megaraid card? |
| 04:26.26 | supaigtr | ??? Its a MPC 3700 rackmount server. |
| 04:26.43 | supaigtr | Supposed to work with digium hardware well. |
| 04:27.09 | marc324 | supaigtr-- do you use openser or iptel-ser? |
| 04:28.44 | supaigtr | openser but I don't think theres much difference in getting it running. |
| 04:35.34 | *** join/#asterisk Igbothom (n=HiltonT@static-84.217.240.220.dsl.comindico.com.au) |
| 04:37.28 | brookshire | trelene: problems finding your iaxy? |
| 04:38.43 | drumkilla | brookshire: !!!!!!!!!!!!!!!! |
| 04:38.59 | brookshire | z0mg it's killad |
| 04:39.06 | drumkilla | w00t! |
| 04:39.23 | brookshire | wussell :) |
| 04:39.33 | e-Hernick | z0mg w00t r0r! |
| 04:39.42 | Igbothom | wasn't wussel wandering wiv da wabbits? |
| 04:41.34 | brookshire | drumkilla: have you seen astjab yet? |
| 04:41.46 | drumkilla | no |
| 04:42.13 | e-Hernick | yay netsplit ? |
| 04:42.19 | brookshire | :( |
| 04:42.36 | drumkilla | that's hot |
| 04:42.48 | brookshire | that's wild ;) |
| 04:43.26 | wunderkin | bitchin |
| 04:44.07 | drumkilla | brookshire: I gave a talk for the Clemson LUG tonight, and nobody wanted the digium calculators I had, hah |
| 04:44.16 | drumkilla | pens and cds, sure |
| 04:44.23 | drumkilla | but they were not so impressed with the calculator. |
| 04:44.31 | *** part/#asterisk mcadory (n=mcadory@208.149.64.28) |
| 04:44.32 | e-Hernick | calculators suck |
| 04:44.33 | orlok | did it have a subnet function? |
| 04:44.35 | e-Hernick | digium abacus would rule |
| 04:44.37 | orlok | if not, then thats why :) |
| 04:44.43 | brookshire | yeah.. i agree |
| 04:44.47 | e-Hernick | hex abacus |
| 04:44.48 | drumkilla | nah man, just 4 function |
| 04:45.11 | *** join/#asterisk mcadory (n=mcadory@208.149.64.28) |
| 04:45.34 | desktophero | how about codec bandwith device and an rj45 to sniff SIP traffic...anything like that? |
| 04:46.35 | brookshire | a Digium abacus would be cool, lol |
| 04:47.18 | FuriousGeorge | to calculate how many tdm u need |
| 04:53.31 | Tili | how a network timestamp guaranteed to be same anywhere in the world |
| 04:55.36 | trelane | anyone using asterisk with cisco vic cards? |
| 04:58.26 | *** join/#asterisk rene- (n=rene-@dsl-200-67-134-156.prod-empresarial.com.mx) |
| 04:58.30 | rene- | hello all |
| 04:58.51 | *** join/#asterisk djin_ib (n=djin_ib@gridfox.xs4all.nl) |
| 04:59.21 | *** join/#asterisk argos73 (i=1000@jason.argos.org) |
| 05:02.10 | rene- | when using iax realtime is the output of iax2 show registry available? |
| 05:07.29 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 05:07.48 | orlok | hey Bhaal |
| 05:10.36 | *** join/#asterisk ptblank (n=MURDER1@68-169-161-61.lmdaca.adelphia.net) |
| 05:14.08 | Tili | rene-: my guess is yes |
| 05:16.24 | *** join/#asterisk juice (n=juice@mo-65-40-248-239.dyn.sprint-hsd.net) |
| 05:16.24 | *** join/#asterisk dudes (n=dudes@12-215-32-62.client.mchsi.com) |
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| 05:16.25 | *** join/#asterisk mogorman (n=mogorman@gateway.digium.com) |
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| 05:16.41 | trelane | is there an awake digium employee? |
| 05:16.49 | trelane | I need to know who to contact about a semi-dead iaxy |
| 05:17.07 | brookshire | support@digium.com |
| 05:17.41 | brookshire | they can get you a rma # |
| 05:18.02 | *** part/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 05:18.08 | *** join/#asterisk BhaalWTF (n=bhaal@CPE-141-168-108-119.qld.bigpond.net.au) |
| 05:18.32 | *** join/#asterisk BhaalHTF (n=bhaal@CPE-141-168-108-119.qld.bigpond.net.au) |
| 05:19.03 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 05:22.53 | *** join/#asterisk mjmac (n=mjmac@pdpc/supporter/active/mjmac) |
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| 05:23.43 | FuriousGeorge | is there any way to interface an * server with one of them broadband wireless cards from cell phone providers? |
| 05:24.03 | brookshire | bluetooth |
| 05:24.04 | brookshire | :) |
| 05:24.24 | brookshire | but that's more of not an asterisk thing |
| 05:24.32 | FuriousGeorge | do they have external cell broadband network reveivers |
| 05:24.44 | brookshire | yeah.. they do |
| 05:24.56 | brookshire | gsm sip gateways, etc |
| 05:25.25 | *** join/#asterisk tecnico (n=tecnico@user-24-236-120-2.knology.net) |
| 05:25.25 | brookshire | there are a lot out there for data as well |
| 05:25.26 | FuriousGeorge | brookshire: think about this though: if you could do it, and all ur employees/friends use cell phones, certain networks allow free calling |
| 05:26.33 | brookshire | FuriousGeorge: i get free calling as it is to my friends who use cingular, lol |
| 05:26.42 | FuriousGeorge | so when calling from home or work you could use the receiver to take calls through asterisk |
| 05:26.45 | FuriousGeorge | we use verizon |
| 05:26.52 | FuriousGeorge | in-calling, same concept |
| 05:27.29 | FuriousGeorge | so i could interface one of these receivers to asterisk via a bt usb dongle? you sure about that? |
| 05:29.52 | brookshire | pretty sure |
| 05:29.59 | brookshire | you can get data via bluetooth |
| 05:30.20 | brookshire | i don't know how much bandwidth you can pull though |
| 05:30.48 | brookshire | verizon apparently upgraded their network so it's faster than it use to be |
| 05:31.11 | brookshire | also.. usb is to interface with a phone |
| 05:31.28 | brookshire | you can buy serial, pci, or network gsm adapaters |
| 05:31.32 | brookshire | there are plenty of them out there |
| 05:32.02 | brookshire | but no promise the provider will hook them up |
| 05:44.30 | *** join/#asterisk oej (n=Olle@apollo.webway.se) |
| 05:57.52 | *** join/#asterisk tekati (n=captain@cpe-66-75-215-63.bak.res.rr.com) |
| 05:57.53 | Insanity5 | Can someone tell me if 66.225.32.67 is pingable? |
| 05:58.13 | FuriousGeorge | Insanity5: no |
| 07:56.30 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 07:56.30 | *** topic/#asterisk is Asterisk 1.2.0 Beta1 - http://www.asterisk.org || Astricon 2005 was a great success! || Level3 has depeered the internet (also known as: Al Gore turned the internet off) |
| 07:57.08 | *** join/#asterisk GXTi_ (n=omgwtfbb@freenode/developer/GXTi) |
| 07:57.45 | RoyK | ~beer |
| 07:57.47 | jbot | rumour has it, beer is ummm, ummm good!, or good for you!... not just for breakfast anymore |
| 07:57.48 | MGSsancho | lol |
| 07:58.31 | *** join/#asterisk XTR (n=xtr@staff-nat.netnation.com) |
| 07:58.40 | RoyK | ~sex |
| 07:58.41 | jbot | updatedb; locate; talk; date; cd; strip; look; touch; finger; unzip; uptime; gawk; head; apt-get install condom; mount; fsck; gasp; more; yes; yes; yes; more; umount; apt-get remove --purge condom; make clean; sleep, or super extractor, http://sf.net/projects/sex/ |
| 07:59.03 | MGSsancho | ~unf |
| 07:59.04 | jbot | it has been said that unf is the Universal Noise of Frustration, it is also washort's favourite word, after heh . something else related to pr0n movies |
| 07:59.23 | *** join/#asterisk Delvar (n=irc@host-83-146-53-34.bulldogdsl.com) |
| 07:59.58 | MGSsancho | ~asterisk |
| 07:59.59 | jbot | extra, extra, read all about it, asterisk is the symbol that looks like a star (shift-8 on north american English keyboards), or a definative book on the future of telephony written by jsmith |
| 08:00.08 | MGSsancho | booo |
| 08:00.14 | MGSsancho | ~level3 |
| 08:00.29 | MGSsancho | nothing hahahahahha so true |
| 08:01.19 | *** join/#asterisk ohsix (i=ohsix@bc111224.bendcable.com) |
| 08:01.27 | SwK | hah |
| 08:01.33 | SwK | l3 sucks |
| 08:02.20 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 08:04.37 | *** part/#asterisk KaBewM (n=DA-MAN@66-215-7-106.dhcp.psdn.ca.charter.com) |
| 08:05.05 | *** part/#asterisk nobell (n=Jared@160.7.249.18) |
| 08:07.58 | niZon | jbot, level3 is crap |
| 08:07.59 | jbot | niZon: okay |
| 08:08.04 | niZon | ~level3 |
| 08:08.05 | jbot | methinks level3 is crap |
| 08:08.08 | niZon | :D |
| 08:08.42 | MGSsancho | lol |
| 08:09.08 | MGSsancho | jbot, twisted is hawt ans secksi |
| 08:09.10 | jbot | MGSsancho: cannot alter locked factoids |
| 08:09.15 | MGSsancho | :( |
| 08:09.18 | MGSsancho | *and |
| 08:13.38 | *** join/#asterisk FastJack (i=fastjack@2001:8d0:20ff:3:0:0:0:1) |
| 08:26.56 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 08:28.28 | RoyK | anyone around that's heard of a 2.6.13 update for zaptel yet? |
| 08:29.31 | *** join/#asterisk GXTi (n=omgwtfbb@freenode/developer/GXTi) |
| 08:30.06 | syle2 | i heard same thing alot jbot |
| 08:30.26 | SwK | http://blog.ryandubois.net/archives/2005/10/21/level3-fails-upgrade/ |
| 08:30.28 | syle2 | err nizon |
| 08:30.49 | syle2 | and their customer support is the worst |
| 08:31.18 | *** join/#asterisk skrusty (i=muad@xdev.net) |
| 08:31.27 | niZon | yeah i'd imagine |
| 08:31.42 | niZon | if they screw up this bad.. |
| 08:33.52 | *** join/#asterisk RaYmAn-Bx (i=rayman@x1-6-00-40-63-da-39-3f.k191.webspeed.dk) |
| 08:40.25 | *** join/#asterisk jac]Z[oby (n=me@62.218.44.178) |
| 08:40.49 | jac]Z[oby | bristuff plus msn help needed |
| 08:40.54 | jac]Z[oby | for the 234234 time |
| 08:40.57 | jac]Z[oby | pleeeeeeez |
| 08:41.02 | jac]Z[oby | im begging here ;) |
| 08:41.45 | *** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 08:43.18 | *** join/#asterisk Teeli (n=Tili@219.136.219.242) |
| 08:46.31 | *** join/#asterisk Beirdo (n=gjhurlbu@unaffiliated/beirdo) |
| 08:46.36 | *** join/#asterisk alrs (n=lars@dsl092-033-090.lax1.dsl.speakeasy.net) |
| 08:47.16 | jac]Z[oby | is a capi needed in order to let the card listen to a specific MSN? |
| 08:48.26 | clive- | jac, sometimes the telco doesnt send the msn |
| 08:48.44 | jac]Z[oby | sure it sends it |
| 08:48.46 | jac]Z[oby | the problem is |
| 08:49.18 | jac]Z[oby | that i have to configure the * to only accept calls to msn#1 and not to msn#2 |
| 08:49.50 | jac]Z[oby | 1 BRI line with 2 MSNs |
| 08:50.09 | jac]Z[oby | 1st MSN is connected to a MODEM which listens to a certain number |
| 08:50.32 | jac]Z[oby | problem is that the asterisk also tries to connect to this number and thus blocking the modem |
| 08:51.11 | jac]Z[oby | i can already use the line |
| 08:51.23 | jac]Z[oby | phone incoming and outgoing |
| 08:51.39 | jac]Z[oby | bristuffed acer ISDN active pci card |
| 08:52.27 | jac]Z[oby | do i have do setup the junghanns chan_capi on top of it to st those things (e.g. incomingmsn=XXX) |
| 08:54.51 | clive- | acer active pci?...what is that, never heard of that |
| 08:55.12 | jac]Z[oby | errrm sorry no active card |
| 08:55.20 | jac]Z[oby | but bristuffable ;) |
| 08:55.24 | jac]Z[oby | cologne chipset |
| 08:55.37 | jac]Z[oby | very easy to get to isdn with a@h |
| 08:55.58 | jac]Z[oby | just the MSN prob. remains |
| 08:56.46 | clive- | i pulled all my hfc-s cards |
| 08:57.17 | clive- | too many hassles |
| 08:57.18 | *** join/#asterisk KriS83 (n=KriS@212.202.141.92) |
| 08:57.36 | jac]Z[oby | what do you mean |
| 08:57.46 | KriS83 | Hello |
| 08:57.49 | clive- | mine now collect dust |
| 08:58.07 | jac]Z[oby | ah.. hmm well here its the cheapest solution |
| 08:58.19 | clive- | where you at? |
| 08:58.25 | jac]Z[oby | dont want to pay 260€ for an active ISDN card |
| 08:58.30 | jac]Z[oby | austria |
| 08:58.45 | KriS83 | I have a question. How can I have a digits a caller enters saved in a var? |
| 08:58.54 | clive- | the hassle factor is much more expensive |
| 08:58.56 | clive- | imho |
| 08:59.07 | jac]Z[oby | :) |
| 08:59.22 | jac]Z[oby | well thats y i'm now working with thsi acer card |
| 08:59.42 | jac]Z[oby | because normal fritz fuggin AVM card too complicated |
| 08:59.55 | KriS83 | jac]Z[oby, not really... its quite easy.. ;) |
| 09:00.05 | jac]Z[oby | well we thaught so too |
| 09:00.07 | jac]Z[oby | problem is |
| 09:00.14 | clive- | i'd go for a fritz or eicon anyday |
| 09:00.15 | jac]Z[oby | heres no linux pro available |
| 09:00.52 | jac]Z[oby | tried about 10 linux distributions |
| 09:00.54 | KriS83 | jac]Z[oby, basicly its just compiling the drivers... including them into the kernel.. |
| 09:00.58 | jac]Z[oby | jepp |
| 09:01.00 | jac]Z[oby | i know |
| 09:01.03 | KriS83 | jac]Z[oby, tried ubuntu? ;) |
| 09:01.08 | jac]Z[oby | thats what we tried everytime |
| 09:01.18 | jac]Z[oby | ubuntu sux even more |
| 09:01.22 | KriS83 | ubuntu has packages even ready I think |
| 09:01.29 | jac]Z[oby | yea like suse |
| 09:01.36 | jac]Z[oby | and then comes the mysql |
| 09:01.38 | jac]Z[oby | HATE |
| 09:01.42 | jac]Z[oby | doesnt matter |
| 09:01.43 | KriS83 | ? |
| 09:01.46 | jac]Z[oby | works perfectly now |
| 09:01.57 | jac]Z[oby | only msn problem remains |
| 09:02.12 | KriS83 | jac]Z[oby, msn problem? you want to pass back to an msn? |
| 09:02.29 | jac]Z[oby | i just want the asterisk to take a certain incoming msn |
| 09:02.34 | jac]Z[oby | i BRI 2 MSns |
| 09:02.40 | jac]Z[oby | 1 is reserved for a modem |
| 09:02.52 | *** part/#asterisk wmandra (n=me@pcp04943183pcs.verona01.nj.comcast.net) |
| 09:02.52 | jac]Z[oby | but asterisk connects to this msn too |
| 09:06.14 | *** join/#asterisk MikeJ__ (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
| 09:06.52 | *** join/#asterisk ful|work (n=fulgas@209.8.233.170) |
| 09:07.11 | ful|work | hey |
| 09:08.58 | *** join/#asterisk kll (i=ply@ircgeeks.net) |
| 09:11.18 | KriS83 | Anyone one a hint on my wish? -> Call -> Hello X, Please enter your CustID -> how do I get the CustID in a variable? |
| 09:11.58 | kll | I'm having some problems with a zaptel E1/ISDN PRI card, I'm trying to configure crc4 but it doesn't seem to have any effect. Is there a way to actually look at how the card is currently configured? |
| 09:13.32 | clive- | does anyone have any pointers on fastagi ? |
| 09:16.40 | jac]Z[oby | or on bristuff MSN ? ;)) |
| 09:16.41 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net) |
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| 09:21.35 | puzzled | morning all |
| 09:22.43 | wibblewobble | Hi, I know this is off-topic but I was wondering if anyone knew of a good VoIP carrier in the Florida area? |
| 09:24.54 | tzafrir_laptop | kll, what do you mean "has no effect"? Could you pastebin the output of ztcfg -vv ? |
| 09:26.32 | *** join/#asterisk duckz (n=duckz@193.192.46.26) |
| 09:26.36 | duckz | hi |
| 09:27.34 | newl | KriS83: check the docs for Read() |
| 09:29.27 | KriS83 | newl, Thanks |
| 09:29.57 | newl | np |
| 09:31.09 | darkskiez | is there a function to change a meetme conference pin ? |
| 09:34.01 | *** join/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8) |
| 09:34.14 | clive- | what is a VNAK ? |
| 09:35.12 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 09:35.48 | clive- | ~VNAK |
| 09:36.15 | clive- | !VNAK |
| 09:36.24 | clive- | i guess the bot doesnty know either |
| 09:41.56 | *** join/#asterisk cypromis (n=michael@83.149.70.59) |
| 09:42.50 | clive- | hi Mike |
| 09:43.28 | cypromis | morn |
| 09:43.30 | cypromis | :) |
| 09:44.02 | Teeli | morning cypromis |
| 10:07.18 | *** join/#asterisk zaffa (n=x@cs158252.pp.htv.fi) |
| 10:07.40 | zaffa | Anyone using chanspy? I'm using 1.0.9 *-version.. do I need to upgrade or are there patches anywhere? |
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| 10:09.26 | *** part/#asterisk ringe (n=runar@dsl-253-122.monet.no) |
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| 10:12.45 | zaffa | Anyone? |
| 10:16.13 | tzafrir_laptop | I'm not sure it works with 1.0 |
| 10:20.00 | supa_thygar | need some help please - My asterisk server is running with no problems on the server side but i have people complaining the they cannot hear me and that i sound far away any ideas on what this might be ?? |
| 10:20.23 | *** join/#asterisk Delvar (n=irc@host-83-146-53-34.bulldogdsl.com) |
| 10:20.25 | supa_thygar | o they sound clear to me though |
| 10:20.53 | *** join/#asterisk konfuzed (n=konfuzed@H129.C72.B0.tor.eicat.ca) |
| 10:21.49 | supa_thygar | :( -- really need some help |
| 10:21.59 | zaffa | tzafrir_laptop: Do you recommend to upgrade to 1.2.0 beta 1? |
| 10:23.21 | supa_thygar | ! |
| 10:24.00 | *** join/#asterisk christo (n=chris@brezhnev.spiration.co.uk) |
| 10:24.06 | christo | hi all :) |
| 10:24.14 | supa_thygar | hi |
| 10:27.42 | supa_thygar | need some help please - My asterisk server is running with no problems on the server side but i have people complaining the they cannot hear me and that i sound far away any ideas on what this might be ?? |
| 10:27.49 | supa_thygar | they sound clear to me though |
| 10:28.53 | christo | what are you using? E1/T1, SIP, H323 ? |
| 10:29.03 | tzafrir_laptop | zaffa, I didn't say that |
| 10:32.02 | christo | over IAX, a 'user' makes calls, a 'peer' receieves calles, but what does a 'friend' do? (apart from phone you up at midnight drunk having split up with $girlfriend)? |
| 10:35.57 | X-Rob | christo - both. |
| 10:36.06 | christo | X-Rob - cool thanks |
| 10:37.25 | *** part/#asterisk Dr_Ray (i=drray@dsl254-011-243.sea1.dsl.speakeasy.net) |
| 10:37.49 | *** join/#asterisk Dr_Ray (i=drray@dsl254-011-243.sea1.dsl.speakeasy.net) |
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| 10:40.40 | zaffa | tzafir: hmm... |
| 10:41.13 | KriS83 | Anybody working with the cmd MYSQL with asterisk here? |
| 10:41.25 | christo | KriS83 - yep |
| 10:41.51 | KriS83 | ahh good, may I ask you a simple question? I must be doping something wrong... |
| 10:42.06 | christo | Kris83 - don't ask to ask. Just ask |
| 10:43.27 | KriS83 | http://nopaste.bysteini.de/index.php?id=76ba3912c0 |
| 10:43.49 | KriS83 | Have a look there... I have pasted the part of my extensions.conf |
| 10:44.09 | KriS83 | What I want to do is ask the caller for his customer ID and PIN |
| 10:44.24 | KriS83 | coming from the MySQL DB |
| 10:44.53 | christo | what errors do you get in your console? |
| 10:45.16 | KriS83 | none.. the query works fine.. its more a problem with the line: exten => 123,5,MYSQL(Fetch foundRow ${resultid} number) |
| 10:45.31 | KriS83 | I think or with exten => 123,6,GotoIf($["${foundRow}"!=${pin}]?123,1:205,1) ; leave loop if no row found |
| 10:46.21 | KriS83 | I'm not yet to firm with the MySQL Syntax in general... |
| 10:46.50 | KriS83 | So I'm not sure if the GotoIf is ok? |
| 10:47.23 | christo | your row numbers don't look right.. 123,1:205,1 I've not seen ',1' on the end of a row number before.. so perhaps that's some features I don't know about, but I would just specify 123:205 |
| 10:48.16 | KriS83 | ahh I thought this is the redirect to 123,1 if wrong and 205,1 is right? |
| 10:48.27 | KriS83 | Thats bassicly what I wanted to achive |
| 10:48.41 | christo | also, foundRow just returns a bool as far as I know, so test foundRow like this: exten => 123,6,GotoIf($["${foundRow}" = "1"]?123:205) |
| 10:48.43 | KriS83 | If {PIN} == the PIN from mysql |
| 10:48.59 | christo | hint - this stuff is far easier with AGI |
| 10:49.29 | KriS83 | This is only the beginning though.. ;) And I'd still like to learn it... |
| 10:52.04 | KriS83 | christo, ok, how would it look if I wanted the result in a varable? |
| 10:52.09 | KriS83 | variable* |
| 10:53.54 | KriS83 | Ok, have you got an example for an AGI for me? I found this and thought it would be what I need |
| 10:54.15 | Chuji | What's your language of choice? Perl, C, php? |
| 10:54.19 | KriS83 | Best would be some example AGI with any MySQL query.. no matter what it queries for.. |
| 10:54.21 | christo | KriS83 - here's an example of capturing a value from the mysql result row.. and then using it on a later line... |
| 10:54.22 | christo | exten => s,10,MYSQL(Fetch foundRow ${trackidres} tr_lead) ; |
| 10:54.23 | KriS83 | PHP please |
| 10:54.33 | christo | exten => s,18,Playback(ps/leads/${tr_lead}) |
| 10:55.37 | Chuji | christo : There are a couple php examples in the * agi folder I believe. If not, just google "asterisk php agi" |
| 10:55.42 | christo | KriS83 - I also recommend doing AGI in PERL, because of the very useful Asterisk::AGI module (which is probably in CPAN by now) |
| 10:56.02 | christo | KriS83 - Chuji's last message there was probably intended for you :) |
| 10:56.10 | bsdfreak | heh |
| 10:56.20 | KriS83 | right... |
| 10:56.27 | Chuji | yeah, lol... it's early |
| 10:56.31 | christo | :) hehe |
| 10:57.08 | Chuji | Speaking of which... I need to get ready for work! |
| 10:57.39 | KriS83 | one really silly question then.. sorry for such stupid questions, but I have only compiled ASterisk so far and this is my first contact with it... where do I find the AGI folder? ;) |
| 10:58.01 | KriS83 | ahh.. var/libs/agi-bin? |
| 10:58.18 | Chuji | tada |
| 10:58.32 | christo | KriS83 - you'll probably need Asterisk-addons too for the mysql stuff |
| 10:58.39 | KriS83 | christo, I have them |
| 10:58.41 | christo | okay |
| 10:58.44 | KriS83 | they work fine ;) |
| 10:59.04 | christo | my agi path is /var/lib/asterisk/agi-bin/ just for comparison |
| 10:59.11 | Chuji | /var/lib/asterisk/agi-bin actually |
| 10:59.17 | christo | :) aye |
| 11:01.11 | *** part/#asterisk Dr_Ray (i=drray@dsl254-011-243.sea1.dsl.speakeasy.net) |
| 11:06.01 | KriS83 | right.. none in PHP though.. but I'll check google for that |
| 11:15.29 | christo | Guys, I'm having trouble passing CLI over an IAX connection.. ${EXTEN:1} is showing as serverA@serverA. How can I get it to pass in the extension dialled by the caller? |
| 11:16.09 | jac]Z[oby | so |
| 11:16.11 | jac]Z[oby | work suxx |
| 11:16.15 | jac]Z[oby | im going home |
| 11:16.27 | jac]Z[oby | cyas guyses nexts weeks |
| 11:16.47 | jac]Z[oby | *quote: gross guy from futurama:)* |
| 11:17.44 | *** join/#asterisk sivana (n=sivana@mixdown.ca) |
| 11:20.12 | christo | er.. I didn't mean CLI, I meant the _dialled_ number |
| 11:22.00 | kaldemar | christo: Dial(IAX2/remote/${EXTEN}) |
| 11:26.22 | christo | Kaldemar - sorry.. that bit does work.. I can see now witn 'agi debug' that the dialled extension is being passed in.. it's just that the agi isn't running. I'll see if I can work out what's going on here. thanks |
| 11:35.59 | *** join/#asterisk fenlander (n=fenlande@82.152.81.57) |
| 11:37.26 | christo | this is weird.. it looks like the IAX bit is working, then at the remote end, the AGI is launched and runs normally (according to debug), but the caller just hears a ringing tone and the call doesn't appear ever to be answerd :S |
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| 11:38.15 | christo | the only difference between a 'working' agi debug and a 'not working' agi debug is that the agi_dnid value is unknown in the latter |
| 11:39.36 | RoyK | hrmf |
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| 11:41.24 | anguis | hi all |
| 11:41.26 | christo | RoyK - hrmf? |
| 11:41.27 | RoyK | can someone take a look at this? |
| 11:41.28 | RoyK | http://pastebin.ca/26153 |
| 11:41.32 | RoyK | sip register works |
| 11:41.35 | RoyK | but not the invite |
| 11:43.00 | ful|work | ser ? |
| 11:43.39 | RoyK | asterisk |
| 11:46.56 | anguis | maybe someone in here who can help with some strange problems with 3 isdn-hfc-cards? |
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| 11:59.55 | sivana | anyone using HEAD as of now? |
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| 12:11.30 | Little-L | HI |
| 12:11.48 | Little-L | http://lists.digium.com/pipermail/asterisk-bsd/2005-October/001250.html -I've got some problems solving a driver issue |
| 12:16.24 | KriS83 | In the Beta Version with chan-capi-0.6-cm how can I define the outgoing MSN? "outgoingmsn=27" in /etc/asterisk/capi.conf in the ISDN1 section doesn't work? |
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| 12:17.14 | littleball | hello |
| 12:17.20 | RoyK | http://www.hypotheticals.co.uk/images/SofitelLoo2.jpg |
| 12:17.47 | gaggaman | anguis whats the prob? |
| 12:19.37 | littleball | i have setup a conference call service by using meetme. THe system use E1 interface connnect to center office. Sometimes, i think the center office doesn't detect the remote participant(especially oversea users) tear down the call. So, asterisk system cannot recover that channel. ANy suggestions? |
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| 12:21.27 | littleball | i think the problem happend to other asterisk users also:http://lists.digium.com/pipermail/asterisk-users/2003-December/030353.html |
| 12:23.58 | anguis | gaggaman: I'm using three HFC-S cards, the first two are working w/o any problem, but now i'm trying for about 3 days to get the third one working, without any success |
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| 12:27.55 | Little-L | Any ideas which driver to use with TE410P (2nd Gen) and Freebsd 5.4-Stable |
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| 13:09.03 | *** join/#asterisk hatems (n=hatems@196.203.15.218) |
| 13:09.19 | hatems | hi all |
| 13:14.26 | *** join/#asterisk fugitivo (n=ajf@209.13.241.66) |
| 13:16.24 | *** join/#asterisk shadebob (n=shadebob@ll81-144-114-192-81.ll81.iam.net.ma) |
| 13:16.28 | shadebob | hi |
| 13:17.47 | hatems | hi shadebob |
| 13:17.58 | shadebob | someone have info on xorcom astribank8? In Xorcom site the is only a small PDF |
| 13:18.25 | *** join/#asterisk CANO-1982 (i=alejandr@201.255.51.193) |
| 13:19.33 | CANO-1982 | I have a plroblem with my TDM400 fxo module |
| 13:19.49 | fugitivo | what problem? |
| 13:20.01 | CANO-1982 | have to unplug and plug the telephone to answer an incoming call |
| 13:20.15 | fugitivo | what telephone? |
| 13:20.18 | CANO-1982 | do you speak spanish? |
| 13:20.24 | fugitivo | yes |
| 13:20.27 | littleball | hi, what is this variable ${MEETME_EXIT_CONTEXT} for? and how to use? |
| 13:20.42 | CANO-1982 | its an analog line provided by a hicom 100 |
| 13:21.09 | CANO-1982 | de esa forma me atiende la llamada. |
| 13:21.28 | fugitivo | te referis a la linea telefonica, no a un telefono? |
| 13:21.37 | CANO-1982 | ahá |
| 13:21.58 | CANO-1982 | la línea viene de una Hicom 100 (PBX) |
| 13:21.59 | *** join/#asterisk nexis (n=nexis@12-207-56-108.client.mchsi.com) |
| 13:22.15 | *** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 13:22.16 | CANO-1982 | puedo realizar llamadas pero no puedo atenderlas |
| 13:22.18 | Ariel_ | Morning |
| 13:22.23 | fugitivo | morning |
| 13:22.38 | fugitivo | ves algo en el CLI cuando entra una llamada? |
| 13:23.14 | CANO-1982 | no, ni se mueve, y lo corro con por lo menos 5 v |
| 13:23.41 | fugitivo | que signalling estas usando? |
| 13:23.51 | *** join/#asterisk supa_thygar (i=thygar@tpr-165-12-169.telkomadsl.co.za) |
| 13:23.52 | Katty | sadly, i'm catching about half of this conversation. |
| 13:23.59 | Katty | thanks to my bilingual mother. |
| 13:24.07 | CANO-1982 | cuando saco el cable me dice que detecta una inversión de polaridad, probé con kewl y loop |
| 13:24.12 | Ariel_ | Katty, well he is asking him now which type he is using. |
| 13:24.26 | Katty | Ariel_: yes. i caught what signalling is |
| 13:24.29 | fugitivo | CANO-1982: fxsks? |
| 13:24.31 | Ariel_ | all fxs/fxo so he is on a zap channel |
| 13:24.33 | shadebob | nobody from xorcom here? |
| 13:24.34 | Katty | Ariel_: i can pretty much guess the rest of it |
| 13:24.47 | fugitivo | Katty: :) |
| 13:24.56 | nexis | is it possable to set flags with a call file like with the dial string, such as H or L |
| 13:25.00 | supa_thygar | guys i need some help people say that i sound soft and far away when they speak to me over the phone i use TDM400P with asterisk 1.2 Beta with a artdio2000l sip phone |
| 13:25.05 | *** part/#asterisk Insanity5 (n=feaw@ip68-105-215-25.tl.dl.cox.net) |
| 13:25.10 | Katty | all can understand that, despite their speaking language ;) |
| 13:25.11 | hatems | can any one have a commercial link with digium |
| 13:25.11 | CANO-1982 | si (es que se me da vuenta en la cabeza porque son al revés) |
| 13:25.26 | Katty | hatems: what do you mean by Commercial Link? |
| 13:25.30 | hatems | iwould like to bye a complet system for call center |
| 13:25.30 | Ariel_ | supa_thygar, use txgain and rxgain settings in the zapata.conf |
| 13:25.38 | hatems | hi katty |
| 13:25.44 | hatems | like email |
| 13:25.46 | Katty | hatems: i don't think. i believe you need a reseller. |
| 13:25.57 | supa_thygar | Ariel_ i have tried what would good settings be ? |
| 13:26.01 | hatems | yes a resseller |
| 13:26.04 | Ariel_ | CANO-1982, que tiens TDM400 or x100p ? |
| 13:26.19 | CANO-1982 | Ariel_ una TDM400p |
| 13:26.21 | Ariel_ | supa_thygar, you can use ztmonitor to check your volume |
| 13:26.39 | supa_thygar | Ariel_ on the ztmonitor it looks fine |
| 13:26.57 | fugitivo | CANO-1982: si probas llamar al asterisk, te da tono de ocupado? |
| 13:27.16 | Ariel_ | CANO-1982, las plaqeta's son rojas or verde que tiene instalada? |
| 13:27.21 | CANO-1982 | no, me indica que está llamando |
| 13:27.42 | fugitivo | CANO-1982: estas seguro que son FXO? |
| 13:28.03 | Ariel_ | supa_thygar, you just have to play with the settings until you get the right corret sound |
| 13:28.19 | CANO-1982 | inclñuso ayer probé de llamar desde elAsterisk cuando hy una llamada entrante y en el teléfono desde el que realizo las llamadas escucho el dtmf de Asterisk |
| 13:28.41 | CANO-1982 | si, son 2 fxs y 2 fxo |
| 13:28.41 | fugitivo | CANO-1982: si conectas el asterisk directo a la linea telefonica (no pbx), funciona? |
| 13:28.43 | supa_thygar | Ariel_ do you know what the max is for tx rx gains ? |
| 13:28.57 | CANO-1982 | las luces son todas rojas ?! |
| 13:29.04 | Ariel_ | -12 +18 |
| 13:29.18 | CANO-1982 | es que no tengo acceso a línea externa |
| 13:29.26 | supa_thygar | Ariel_ my problem is that it happens with the sip as well |
| 13:29.37 | CANO-1982 | pero lo probé en una Hicom 300 y anda perfecto |
| 13:29.44 | supa_thygar | when i phone from one ext to another |
| 13:29.58 | CANO-1982 | tuve que ajustarle el answeronpolarity... |
| 13:30.10 | Katty | answeronpolarity? |
| 13:30.14 | Katty | Ariel_: translate! |
| 13:30.17 | *** join/#asterisk rking (n=rking@ip68-105-231-56.lu.dl.cox.net) |
| 13:30.20 | Ariel_ | that is wrong |
| 13:30.31 | CANO-1982 | perdón fujitivo, las luces son todas verdes |
| 13:30.32 | Ariel_ | but I think he is trying to ajust the polarity |
| 13:31.00 | Ariel_ | las luces no indica que tiene los fxs/fxo correcto |
| 13:31.13 | CANO-1982 | OK, I also have a Hicom 300, but the asterisk serve handles that line very well |
| 13:31.30 | christo | is there any reason why an AGI shouldn't run on the remote end of an IAX trunk? |
| 13:31.55 | Ariel_ | remote end of an iax trunk??? |
| 13:32.00 | CANO-1982 | i made it work by tweaking the hangupon.. and tthe answeronpol... parameters |
| 13:32.36 | CANO-1982 | what are the ligths for? |
| 13:33.16 | Ariel_ | CANO-1982, only when there is a major problem do they go red on the TDM400p. |
| 13:33.22 | CANO-1982 | are all green |
| 13:33.40 | christo | Ariel_ sorry, perhaps I'm using the wrong terminology. I have set up an IAX connection btw two servers. ServerA receives a call and dials over IAX to ServerB. At that point, I would like to run an AGI script. My problem is that the console says the AGI is launching, but in fact it does nothing |
| 13:33.41 | CANO-1982 | oh thanks, I didnt know |
| 13:33.44 | christo | if I dial direct, the AGI works |
| 13:34.12 | Ariel_ | christo, your not passing the correct parameters in your dial string then |
| 13:35.39 | christo | Ariel_ I'm passing ${EXTEN} so that I can tell which extension is dialled, and the remote dialplan is working fine, picking up the call and it reports that it's launching the AGI. The agi header also reports the correct extension (altho the dnid is empty) |
| 13:36.01 | CANO-1982 | here is a clue, when I measure the Voltage on the phone line with the incoming line, it reads detween 120 and 48 (using the fxstest tool) |
| 13:36.46 | CANO-1982 | the problems could be related to a bad ground? |
| 13:36.46 | *** join/#asterisk szer (n=Miranda@217.116.36.22) |
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| 13:36.50 | szer | hi all |
| 13:36.56 | christo | So my dialplan launches the AGI, the console reports that the AGI is running and shows it going through the various stages, but nothing happens.. the caller just hears ringing and nothing happens |
| 13:37.23 | szer | I've got a problem with res_agi |
| 13:37.36 | *** join/#asterisk file[laptop] (n=jcolp@mctnnbsa31w-142166095097.nb.aliant.net) |
| 13:37.55 | CANO-1982 | fugitivo, los problemas podrían estar relacionados a una mala masa? |
| 13:37.58 | szer | sometimes when it's called it's drop a bad file descriptor error |
| 13:38.11 | christo | Ariel_ my dial string looks like this: exten => 810400,1,Dial(IAX2/nielsen/${EXTEN},30,r) and then at the remove end, in the incoming context it says: exten => 810400,1,agi,logger.pl |
| 13:38.31 | szer | anybody knows how to solve this problem? |
| 13:40.03 | CANO-1982 | szer no, i dont |
| 13:40.12 | *** join/#asterisk lehel (n=lehel@82.79.20.17) |
| 13:40.18 | lehel | hello |
| 13:40.22 | christo | HELO |
| 13:41.11 | *** join/#asterisk CoolAcid (n=jason@216.99.98.39) |
| 13:41.20 | fugitivo | CANO-1982: creo que si hubiera una mala masa, la tdm daria tono de ocupado |
| 13:42.14 | CANO-1982 | ah gracias, es que todavía no me informo bien sobre las distintas señalizaciónes |
| 13:42.20 | CANO-1982 | además puedo realizar llamadas |
| 13:42.37 | lehel | ?? |
| 13:42.46 | lehel | i missed the server? |
| 13:42.58 | lehel | this is spanish huh/ |
| 13:42.59 | fugitivo | podrias medir la tension cuando entra una llamada para ver si es la suficiente para que la tdm detecte el llamado |
| 13:43.06 | fugitivo | lehel: :) |
| 13:43.21 | lehel | i was in spain this summer.. |
| 13:43.29 | lehel | where in spain do u are? |
| 13:43.49 | CANO-1982 | y ?uánto tiene que medir? |
| 13:43.58 | CANO-1982 | yo soy de Argentina |
| 13:44.02 | fugitivo | spain isn't the only country where people speaks spanish :) |
| 13:44.05 | lehel | comprende |
| 13:44.12 | lehel | oups :) |
| 13:44.15 | CANO-1982 | j aj a |
| 13:44.17 | lehel | of course not |
| 13:44.18 | CANO-1982 | es verdad |
| 13:44.23 | fugitivo | i'm from argentina too |
| 13:44.28 | lehel | i know..ok;) |
| 13:44.30 | CANO-1982 | de donde sos fujitivo? |
| 13:44.36 | fugitivo | argentina |
| 13:44.50 | CANO-1982 | puta, estamos acá nomás |
| 13:44.52 | CANO-1982 | ja j a |
| 13:44.56 | fugitivo | bs as? |
| 13:45.11 | CANO-1982 | no, san juan |
| 13:46.02 | CANO-1982 | hace mucho que trabajás con Asterisk? |
| 13:46.50 | lehel | fugitivo: "bs as?" means buenos airos? |
| 13:46.54 | lehel | ;P |
| 13:46.58 | fugitivo | lehel: Buenos Aires |
| 13:47.12 | CANO-1982 | yes, good airs, ja ja |
| 13:47.37 | CANO-1982 | or healthy airs |
| 13:48.29 | CANO-1982 | che, cuánto tiene que medir la tensión? |
| 13:48.45 | CANO-1982 | me da entr 127 y 40 más o menos |
| 13:48.57 | CANO-1982 | casi en ningún momento es negativa |
| 13:49.17 | *** join/#asterisk rudi_ (n=rudi@karfire.megabit.net) |
| 13:49.21 | rudi_ | hey there |
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| 13:50.09 | lehel | fugitivo and CANO-1982.. i think you should create a new channel, like "asterisk-ar" or something.. just a thought? |
| 13:50.49 | CANO-1982 | we are very few people |
| 13:52.30 | Ariel_ | sorry been on the phone. There is an #asterisk-spanish But only used when you fine someone here that speaks and is willing to go there to help. |
| 13:52.50 | *** join/#asterisk ful|work (n=fulgas@209.8.233.170) |
| 13:52.51 | Ariel_ | argh another call. what my phone just is ringings and rings. |
| 13:52.52 | CANO-1982 | ah thaks |
| 13:53.11 | lehel | Ariel_ :D |
| 13:53.51 | CANO-1982 | fujitivo are you there? |
| 13:53.56 | *** join/#asterisk theNOTO (n=biggs@69-165-25-59.clvdoh.adelphia.net) |
| 13:55.55 | CANO-1982 | well , bye everybodie |
| 13:56.34 | *** join/#asterisk bob_too (n=chris@rrcs-24-153-179-246.sw.biz.rr.com) |
| 13:56.36 | *** join/#asterisk EvilRick (n=Ev1lRick@196-28-86-129.wdsl.co.za) |
| 13:56.48 | EvilRick | can anyone help me out on getting my PRI line to dial.. |
| 13:56.56 | *** join/#asterisk shido6 (n=curtis@d221-68-210.commercial.cgocable.net) |
| 13:56.58 | Little-L | Is there anyone who can help with a TE410P and Fbsd 5.4?? |
| 13:56.59 | EvilRick | I ahve it up and the sman configured |
| 13:57.18 | CANO-1982 | bye |
| 13:57.22 | EvilRick | I have it up and the span configured |
| 13:58.23 | EvilRick | I jsut continously get this from pri debug |
| 13:58.23 | EvilRick | Ext: 1 Cause: Call Rejected (21), class = Normal Event (1) |
| 14:00.03 | tzanger | cause code 21 is just "Call Rejected" |
| 14:00.15 | tzanger | the switch doesn't like your number, EvilRick |
| 14:00.21 | tzanger | do you have wrong TON? |
| 14:00.31 | e-Hernick | visdn or zaphfc ? |
| 14:00.36 | e-Hernick | which is better ? |
| 14:00.50 | *** join/#asterisk yxa (i=empty@cm121.gamma228.maxonline.com.sg) |
| 14:00.56 | EvilRick | TON? |
| 14:00.59 | christo | okay - so my AGI script is working. I have it logging lines to a file just to be sure, but the caller just hears ring ring ring instead of the AGI stream.. this problem only occurs when routed over IAX. If I call direct it's fine.. how come? |
| 14:01.09 | EvilRick | tzanger, sorry I am new to PRI lines |
| 14:01.31 | Ariel_ | christo, remove the r from the dial string |
| 14:01.35 | EvilRick | tzanger, whats the usual numbering scheme? |
| 14:01.44 | christo | Ariel_ what does that do? |
| 14:01.55 | Ariel_ | gives a fake ring |
| 14:02.13 | christo | okay.. let's try that :) |
| 14:02.57 | christo | Ariel_ I love you |
| 14:04.44 | yxa | hi guys what's a good POE-enabled sip phone that doesn't cost a bomb? |
| 14:05.01 | Ariel_ | Polycom 601 or the IP-501 with a poe cable |
| 14:05.13 | tzanger | EvilRick: pridialplan=unknown |
| 14:05.18 | tzanger | EvilRick: prilocaldialplan=unknown |
| 14:05.38 | tzanger | christo: never use 'r' unless you have a SPECIFIC reason to |
| 14:05.43 | yxa | Ariel_ thanks. checking it out now... |
| 14:05.50 | christo | tzanger - gotcha |
| 14:05.52 | tzanger | christo: I'm also guessing that your AGI is not Answer()ing the channel, which is why the fake ring didn't end |
| 14:06.04 | christo | tzanger - well it is actually |
| 14:06.10 | christo | I find that weird also |
| 14:06.25 | tzanger | I use 'r' when I dial my cell phone because if I'm out of service or whatnot I don't want it to play the "the person you're calling is out of service" message, just ring and drop to my own voicemail |
| 14:06.41 | tzanger | christo: Answer() before AGI(youragi) -- I found the AGI answer func ot be flakey |
| 14:06.48 | tzanger | Ariel_: nah it has its uses |
| 14:07.56 | Ariel_ | tzanger, yes it does but I still think it's a hack |
| 14:08.06 | tzanger | Ariel_: what would the alternative be? |
| 14:08.46 | tzanger | IMO Dial() needs to be broken up into PreDial() and Dial() where PreDial() sets up all the fucked up parameters (call duration, acknowlege, transferability, etc.) and Dial() just attempts to connect |
| 14:09.15 | yxa | Ariel_ any others you can recall off hand? |
| 14:10.30 | Ariel_ | The Cisco has some phones. Take a look at the listing on voipsupply.com there are many good phones there. |
| 14:11.53 | yxa | Ariel_ cisco phones are way above my budget :) |
| 14:12.42 | Ariel_ | yxa, for the price the Polycom's are best. But only the 600/601 is native PoE. The IP-500/501 are with a special cable from Polycom. |
| 14:12.54 | *** join/#asterisk w0w0 (n=w0w0@36.Red-83-50-231.dynamicIP.rima-tde.net) |
| 14:12.58 | Ariel_ | The Snom phones are good PoE as well but not the 190 |
| 14:14.02 | yxa | Ariel_ I see |
| 14:14.43 | yxa | went to a cisco conference today. they were flaunting their 7970s |
| 14:15.39 | Ariel_ | yxa, it's a nice phone but it's not a sip phone |
| 14:16.07 | Ariel_ | niether is there new CP-7920 wireless phone. Which I would love to have as SIP instead of skinny |
| 14:16.23 | yxa | * can handle skinny, no? |
| 14:16.36 | lehel | yea |
| 14:16.37 | Ariel_ | yes but it's not complete |
| 14:16.53 | Ariel_ | Sip is better |
| 14:17.15 | *** join/#asterisk festr__ (n=festr@ns.regnet.cz) |
| 14:17.17 | festr__ | hello |
| 14:17.27 | yxa | iaxtalk.com has phones which can do iax. anyone have any experience with those? |
| 14:17.39 | festr__ | is it possible to pass through ,SetCallerPres(prohib_not_screened)? |
| 14:17.43 | festr__ | over IAX? |
| 14:17.54 | *** join/#asterisk MPro (n=mpro@82.108.126.172) |
| 14:22.15 | festr__ | i've two boxes, stable <- IAX -> head. if i set on head's dialplan ,SetCallerPres(prohib_not_screened) it will not pass through zaptel E1 on stable |
| 14:22.31 | festr__ | but if i set ,SetCallerPres(prohib_not_screened) on stable it is working |
| 14:22.36 | festr__ | whats wrong? |
| 14:24.56 | *** join/#asterisk tecnico (n=tecnico@user-24-236-120-2.knology.net) |
| 14:25.03 | *** join/#asterisk slePP (n=slepp@S0106000f663692da.ed.shawcable.net) |
| 14:27.07 | MPro | Has anyone got hangup detection of an incoming call working on a TDM400P card (FXO), in the UK on a BT line? I've been struggling with this for close to 2 weeks now and have exhausted all my options. Running with debug=1 (and opermode=UK) on the wcfxs driver I see the voltage drop to zero which I assume is disconnect supervision, and the driver correctly outputs "Battery loss:" and "NO BATTERY", but asterisk does not hang up and keeps playing queue music do |
| 14:27.07 | MPro | wn the line indefinately. |
| 14:29.35 | christo | tzanger - thanks |
| 14:29.52 | christo | (from 23 mins ago :)) |
| 14:30.51 | *** join/#asterisk santiago (n=santiago@208.195.215.158) |
| 14:31.35 | MPro | I've tried everything I can find including options like callprogress, busydetect,hanguponpolarityswitch, all to no avail :( I still can't get hang-ups of the incoming PSTN line to be detected by asterisk. |
| 14:31.43 | festr__ | anyone here use SetCallerPres over IAX channels? |
| 14:31.49 | festr__ | (CLIR) |
| 14:32.22 | KriS83 | Does anybody know if I can make app_capiECT shut up on connect? I don't want to know the number who is calling me. I'd like to be able to connect straight away? |
| 14:34.52 | *** join/#asterisk supaigtr (n=yurplsl@152.53.16.10) |
| 14:35.45 | KriS83 | Nevermind... found it in the source code |
| 14:38.12 | *** join/#asterisk Seyr (n=seyr@cpe-67-10-139-141.houston.res.rr.com) |
| 14:38.39 | *** join/#asterisk Theuni (n=ctheune@alphastar.gocept.com) |
| 14:38.41 | Theuni | howdi |
| 14:39.13 | Seyr | im trying to ring two phones and ringing individual is ok, but if i Dial(SIP/200&SIP/201) only 200 rings. Anyone know what I am doing wrong? |
| 14:41.21 | supaigtr | 10:39:07 Warn IAX Packet 44515 from circuit ids 14->9 conflicts with earlier call with circuit ids 9->10 |
| 14:41.29 | supaigtr | Anyone know what that could be? |
| 14:41.33 | Theuni | I have a weird problem. I have a grandstream budgetone connected to asterisk and an x-lite softphone. i can dial out using some standard isdn hardware. When I use the budgetone everything sounds ok. When I use the softphones calling the budgetone (or vice versa) the softphone has bad output (low voice level) but the budgetone has a good voice level. Any idea what to look out for? |
| 14:41.52 | supaigtr | softphone often sound like crap. |
| 14:42.02 | Theuni | hmm |
| 14:42.03 | Theuni | darn |
| 14:42.19 | Theuni | easy and simple (but unsatisfactory) answer ;) |
| 14:42.26 | Seyr | . |
| 14:42.38 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-218-202.rockynet.com) |
| 14:42.44 | supaigtr | softphone = unsatisfactory is the point |
| 14:42.58 | supaigtr | I've had best luck with firefly. |
| 14:43.07 | Seyr | Softphone is ok, depending on your headset |
| 14:43.11 | jake1932 | i heard diax has been good - don't know first hand though |
| 14:43.19 | *** join/#asterisk mcn_ (n=mcn@ext-gw.newtoncomputing.co.uk) |
| 14:43.35 | Seyr | Xlite with a $20.00 headset can sound like crap, but the same Xlite with a $150.00 headset sounds as good as a 7960 |
| 14:43.55 | supaigtr | Seyr: Exactly, You have to have the right combination of headset, sound card , and machine. |
| 14:43.59 | Theuni | (whatever 7960 is) |
| 14:44.07 | Seyr | Cisco 7960 |
| 14:44.09 | Theuni | ok |
| 14:44.15 | supaigtr | Xlite craps out sometimes. The 7960 not as much. |
| 14:44.20 | Theuni | (isn't that siemens relabeled?!?) |
| 14:44.27 | Little-L | Which drivers do people in here use for their TE410P cards on FBSD 5.4? |
| 14:45.06 | supaigtr | zaptel is the only driver. Don't know about fbsd |
| 14:45.33 | Little-L | Then I need a version and a link to the file? |
| 14:45.45 | file[laptop] | Little-L: FreeBSD isn't an officially supported platform |
| 14:45.45 | *** join/#asterisk just4fun (n=edwin@209.151.137.139) |
| 14:45.56 | file[laptop] | Little-L: so you'll have to get the drivers from ports, but I believe they're old... so - have fun! |
| 14:46.00 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:46.29 | Little-L | file[laptop], Okay. |
| 14:47.51 | *** join/#asterisk Uther_P (n=uther_p@66.180.120.82) |
| 14:48.07 | *** join/#asterisk mhnoyes (n=mhnoyes@user-38lc0d2.dialup.mindspring.com) |
| 14:49.36 | supaigtr | Warn IAX Packet 44515 from circuit ids 14->9 conflicts with earlier call with circuit ids 9->10 Is this a problem????? |
| 14:49.54 | *** join/#asterisk paryl (n=paryl@209.236.78.59) |
| 14:52.13 | paryl | i can't figure out why my message lights aren't working |
| 14:52.50 | paryl | according to all of the info i find, it's solely dependant on the mailbox line in sip.conf |
| 14:52.51 | just4fun | hi everything, I was trying to compile asterisk for arm but have trouble getting the byte order in IAX datagram for (char *) to (int *) casting (eg in iax2-parser.c). Is there a compiler option that correctly convert those castings? |
| 14:53.11 | *** join/#asterisk vtsherwood (n=rushowr@cpe-24-26-150-142.columbus.res.rr.com) |
| 14:53.16 | Seyr | paryl: are you using Realtime? |
| 14:53.17 | vtsherwood | hello all |
| 14:53.40 | paryl | seyr: relatime? |
| 14:53.44 | paryl | realtime* |
| 14:53.49 | paryl | what is that? |
| 14:54.28 | Seyr | paryl: sql |
| 14:54.34 | paryl | ah, no... flat files |
| 14:54.51 | *** join/#asterisk scanna (n=scannach@81-174-16-211.f5.ngi.it) |
| 14:56.01 | just4fun | anyone know of using special flag to fix those (char *) to (int *) castings to compile for arm processor? |
| 14:56.25 | *** join/#asterisk SpaceBass (n=sp@c-24-125-184-203.hsd1.va.comcast.net) |
| 14:56.27 | SpaceBass | morning |
| 14:56.55 | SpaceBass | im having an issue with DISA... I get a dial tone but when i try and dial an extension or external number, i get a fast busy... nothing shows on the cli |
| 14:56.58 | Seyr | paryl: If you have the mailbox in /etc/asterisk/voicemail.conf and you have it in /etc/asterisk/sip.conf pointing to the correct Contect in /etc/asterisk/voicemail.conf, it should work :-) |
| 14:57.12 | Seyr | Context/Contect |
| 14:57.33 | *** join/#asterisk DarthClue (i=user76@wsip-68-99-73-32.tu.ok.cox.net) |
| 14:57.43 | paryl | if i don't specify a context, it assumes the context of the sip entry, correct? |
| 14:58.01 | Seyr | no, it assumes Default in /etc/asterisk/voicemail.conf |
| 14:58.16 | Seyr | i could be mistaken though |
| 14:58.24 | vader-wrk | anyone have luck setting up xlite to work with asterisk? |
| 14:58.26 | Seyr | but thats the way it works :-) |
| 14:58.36 | SpaceBass | vader-wrk it works well |
| 14:58.38 | vader-wrk | i can make phone calls using xlite but i can't seem to get it to recieve calls |
| 14:58.41 | SpaceBass | DISA help... anyone? |
| 14:58.48 | SpaceBass | vader-wrk same LAN as the asterisk box? |
| 14:58.48 | Seyr | if you have "mailbox = 200", it goes to 200 in default context in voicemail |
| 14:58.50 | vader-wrk | i get a registration error on the asterisk console |
| 14:58.50 | vader-wrk | ya |
| 14:58.55 | paryl | wait... in sip.conf i am specifying the context |
| 14:59.00 | Uther_P | vader-wrk: xlite sucks... but it sounds like yourregistry |
| 14:59.13 | paryl | but do i have to say something like "mailbox=1111@context" ? |
| 14:59.32 | SpaceBass | vader-wrk try using pastebin.ca and send it to everyone |
| 14:59.33 | Seyr | mailbox=1111 = default VOICEMAIL context |
| 14:59.47 | paryl | so how do i specify which context? |
| 14:59.51 | Seyr | mailbox=1111@MyContext goes to the MyContext in voicemail.conf |
| 14:59.58 | paryl | gotcha... i'll try that |
| 15:00.24 | Seyr | use default context in voicemail [default] and dont specify a context on the mailbox line in sip |
| 15:00.27 | Seyr | easiest way to test |
| 15:01.27 | SpaceBass | im having an issue with DISA... I get a dial tone but when i try and dial an extension or external number, i get a fast busy... nothing shows on the cli |
| 15:01.41 | *** join/#asterisk rene- (n=rene-@dsl-200-67-134-156.prod-empresarial.com.mx) |
| 15:01.46 | vtsherwood | any * developers on here today? |
| 15:01.49 | rene- | hello all |
| 15:02.05 | just4fun | anyone have successfully compile asterisk on arm? |
| 15:02.27 | *** join/#asterisk fiber0pti (n=johndoe@pcp01876618pcs.sandia01.nm.comcast.net) |
| 15:02.41 | fiber0pti | Anyone have comments on the grandstreams (budgeton, gxp-2000) and asterisk? Any problems with integration? |
| 15:03.40 | paryl | i just rolled out 4 gxp-2000's, had one defective, the other 3 i'm replacing |
| 15:03.57 | paryl | after using them for a week, they're just too cheap feeling |
| 15:04.00 | Seyr | fiber0pti: Have you looked at the Wiki? |
| 15:04.12 | Seyr | paryl: yeh :-) |
| 15:04.23 | Seyr | I like the polycoms and ciscos |
| 15:04.30 | fiber0pti | seyr: i bit.. but I wanted to see if I could get all kinds of opinions.. |
| 15:04.33 | *** join/#asterisk jontow (i=jontow@bsd.adminforrent.com) |
| 15:04.35 | paryl | i've been looking at polycom 501's |
| 15:04.48 | fiber0pti | 501's a fantastic |
| 15:04.50 | fiber0pti | I love 'em |
| 15:04.59 | fiber0pti | think they are btter than the ciscos |
| 15:05.06 | paryl | the grandstream has major volume issues |
| 15:05.09 | Seyr | for the price, polycoms about the best as far as look/feel and realability |
| 15:05.20 | Seyr | realiability |
| 15:05.24 | Seyr | err.. |
| 15:05.26 | paryl | what kind of headset jack do they have? |
| 15:05.36 | paryl | is it a normal db11? |
| 15:05.49 | fiber0pti | nod |
| 15:05.52 | paryl | the gxp2000 has a 3.5mm jack! |
| 15:05.58 | fiber0pti | weird |
| 15:06.02 | paryl | try finding a headset that works with it :\ |
| 15:06.17 | fiber0pti | haha.. I bet |
| 15:06.55 | rene- | i am having issues with a Zaptel Card a |
| 15:07.51 | vtsherwood | anyway...I'm looking for input on how I might change asterisk's sip.conf setting "call-limit" as it causes problems when doing RT config with MySQL |
| 15:09.07 | SpaceBass | anyone using DISA? |
| 15:09.17 | paryl | Seyr: changing the context in voicemail.conf to [default] worked, thanks. for some reason i thought it had to be the same at the one in sip.conf |
| 15:10.23 | *** join/#asterisk Xen^ (n=linux@202.63.195.80) |
| 15:10.28 | Xen^ | hello all |
| 15:11.31 | rene- | never mind i was doing something stupid last night with the dp it is ok no |
| 15:11.34 | rene- | hello |
| 15:12.37 | paryl | how is the polycom 501 as far as software goes? is it updated regularly? |
| 15:12.52 | paryl | i read somewhere that they don't really like the 'community' |
| 15:12.54 | Xen^ | can some one help me in setting up an IVR ? |
| 15:13.38 | paryl | Xen^: http://www.voip-info.org/wiki-Asterisk+tips+ivr+menu |
| 15:13.46 | vtsherwood | anyone? this would be a serious issue for anyone wanting to set client configs in MYSQL like my co. .. |
| 15:13.52 | scanna | hi all, can someone help me with queues? |
| 15:14.28 | SpaceBass | anyone using DISA? |
| 15:14.40 | Xen^ | paryl : well i did it but not working :( |
| 15:15.12 | paryl | what's the problem? |
| 15:15.39 | Xen^ | well no idea :( |
| 15:15.45 | Xen^ | nothing happen |
| 15:17.11 | vtsherwood | ok, well have a good one mates... |
| 15:17.55 | scanna | when a queue exten is called if the caller hangs up when queue-youarenext is playing the phones members of the queue show one call not answered |
| 15:18.09 | *** join/#asterisk steve____ (n=steve@store-fw.porchlight.ca) |
| 15:19.17 | scanna | is it normal? |
| 15:19.38 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 15:19.38 | *** join/#asterisk juice (n=juice@mo-65-40-248-239.dyn.sprint-hsd.net) [NETSPLIT VICTIM] |
| 15:19.38 | *** join/#asterisk dudes (n=dudes@12-215-32-62.client.mchsi.com) [NETSPLIT VICTIM] |
| 15:19.38 | *** join/#asterisk newl (n=newlook@203-59-214-216.dyn.iinet.net.au) [NETSPLIT VICTIM] |
| 15:19.38 | *** join/#asterisk mogorman (n=mogorman@gateway.digium.com) [NETSPLIT VICTIM] |
| 15:19.38 | *** join/#asterisk crash3m_ (i=crash3m@unaffiliated/crash3m) [NETSPLIT VICTIM] |
| 15:19.57 | festr__ | anyone here using setcalleridpres? |
| 15:20.16 | MPro | Aha, fixed my undetected hangup problem with TDM400P at last! After 2 weeks! By lowering the BATT_DEBOUNCE time in the driver wcfxs.c :) |
| 15:21.14 | festr__ | MPro: what is BATT_DEBOUNCE? |
| 15:23.24 | MPro | It's a setting in the zaptel FXO (FXS) driver, the number of milliseconds the line voltage has to drop to zero (hangup) for it to be registered and sent to asterisk as an event |
| 15:24.04 | MPro | 'Debouncing' means a very short event is considered a 'blip' and ignored by the driver. |
| 15:24.22 | snitt | and what is your correct value? |
| 15:24.34 | MPro | It was 64ms, I lower it to 24. |
| 15:25.11 | MPro | Now it works perfectly (so far) on a basic BT analog line in the UK which after searching the web for 2 weeks I didn't think was possible./ |
| 15:25.14 | MPro | :) |
| 15:25.51 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 15:25.57 | MPro | bbi5m got to inform a fellow asterisk hacker |
| 15:26.07 | Seyr | im trying to ring two phones and ringing individual is ok, but if i Dial(SIP/200&SIP/201) only 200 rings. Anyone know what I am doing wrong? |
| 15:26.09 | *** join/#asterisk JonR800 (i=jon@p1mp.org) |
| 15:26.15 | *** join/#asterisk mcf3782 (n=mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net) |
| 15:30.42 | Uther_P | Seyr: what does the cli say when it executes the dial? |
| 15:30.42 | paryl | Seyr: is 201 registered? |
| 15:30.42 | paryl | because that works for me |
| 15:30.44 | *** join/#asterisk hugo-v6 (n=hugo@ns1.bundesunixminister.de) |
| 15:30.52 | hugo-v6 | hi gals |
| 15:32.56 | *** join/#asterisk mkrufky (n=mk@68.160.103.77) |
| 15:33.02 | *** join/#asterisk Tili (n=Tili@211.147.234.5) |
| 15:38.22 | snitt | MPro: i tried it |
| 15:38.27 | snitt | everything went wrong |
| 15:38.34 | snitt | not it does not answer zap channels |
| 15:38.34 | *** join/#asterisk myiagy (n=myiagy@200.138.215.78) |
| 15:38.35 | *** join/#asterisk fugitivo (n=ajf@209.13.241.66) |
| 15:38.42 | snitt | chan_zap.c:3445 zt_handle_event: Ring/Off-hook in strange state 6 on channel 1 |
| 15:40.12 | *** join/#asterisk af_ (n=af@ip-142-250.sn1.eutelia.it) |
| 15:41.44 | *** join/#asterisk kisu (n=kisu@2001:618:400:0:0:0:da26:a0d9) |
| 15:42.35 | *** join/#asterisk Inkubot (n=inkubot@200.75.4.7) |
| 15:45.01 | paryl | when a second line rings, is it possible to ring muted, so that the ring isn't so loud, or is that a function of the phone itself? |
| 15:46.55 | jarrod | anyone here used the CAC 3104 |
| 15:46.56 | jarrod | ? |
| 15:48.27 | vader-wrk | anyone in here work with xlite before? |
| 15:48.37 | jarrod | yea xlite works great for a desktop phone |
| 15:48.44 | vader-wrk | im getting an error Call failed: 484 Address incomplete |
| 15:50.02 | *** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com) |
| 15:50.09 | vader-wrk | this is my sip.conf entry |
| 15:50.09 | vader-wrk | http://pastebin.ca/26177 |
| 15:54.52 | Inkubot | vader-wrk: i think the problem is in you extensions file |
| 15:55.42 | vader-wrk | [softphones] |
| 15:55.42 | vader-wrk | exten => _[123456789]XXXX,1,NoOp(.call for .${EXTEN}) |
| 15:55.42 | vader-wrk | exten => _[123456789]XXXX,2,Dial(SIP/${EXTEN},60,tr) |
| 15:55.42 | vader-wrk | exten => _[123456789]XXXX,3,Congestion |
| 15:56.04 | vader-wrk | thats the only thing i changed in the extensions file |
| 15:57.06 | *** join/#asterisk JerJer (n=JerJer@pdpc/supporter/bronze/jerjer) |
| 15:57.58 | JerJer | Al Gore created the internet and Al Gore can break the internet |
| 15:58.19 | JerJer | sounds like, "its my toy I an break it if i want" |
| 15:58.24 | JerJer | an = can |
| 16:01.33 | vader-wrk | hmm i can't figure out anything on this address incomplete |
| 16:01.51 | vader-wrk | the only file i modified when this happen was my sip.conf |
| 16:03.12 | EvilRick | has anyone got any tips on how I should be dialing on my PRI line. I have it hooked up and all and my PRI Channel 1 is up |
| 16:03.20 | EvilRick | just cant dieal |
| 16:03.24 | EvilRick | dial |
| 16:03.38 | EvilRick | the exchange keeps refusing the number |
| 16:07.06 | IronHelix | dial 11 digit (1-xxx-xxx-xxxx), also check your callerid and ani, perhaps the exchange is refusing it because the call isnt coming from a number assigned to you? |
| 16:07.10 | IronHelix | *shrug* |
| 16:10.34 | wunderkin | EvilRick: do you have pridialplan set in zapata.conf? |
| 16:11.11 | *** join/#asterisk iax_user (n=iax_user@217.9.235.43) |
| 16:11.32 | EvilRick | wunderkin, I tried leaving it default and setting it to unknown |
| 16:12.49 | wunderkin | you want it set to unknown |
| 16:13.07 | wunderkin | you unloaded the modules and did ztcfg afterwords? |
| 16:13.26 | *** part/#asterisk santiago (n=santiago@208.195.215.158) |
| 16:14.31 | EvilRick | no I'll try |
| 16:14.45 | EvilRick | but I thought that wasnt necessary only for /etc/zaptel.conf |
| 16:15.46 | FuriousGeorge | and changes to zapata.conf |
| 16:16.28 | FuriousGeorge | i gotta set up an asterisk door entry system |
| 16:17.00 | FuriousGeorge | i have no idea how the "electromagnetic" door latch is gonna install |
| 16:17.13 | EvilRick | same problem.. |
| 16:18.07 | *** join/#asterisk whitefunk (n=ben@shell.reach.net) |
| 16:18.50 | *** topic/#asterisk by twisted[asteria] -> Asterisk 1.2.0 Beta1 - http://www.asterisk.org || Astricon 2005 was a great success! |
| 16:18.58 | wunderkin | what is the problem |
| 16:19.01 | jake1932 | is a computer science degree worth anything today? |
| 16:19.16 | twisted[asteria] | jake1932, as Level 3 proved last night: no. |
| 16:19.20 | jake1932 | heh |
| 16:19.32 | twisted[asteria] | just kidding |
| 16:19.34 | twisted[asteria] | it all depends |
| 16:19.45 | jake1932 | as a small business owner |
| 16:19.55 | Katty | twisted[asteria]: hi. |
| 16:20.09 | Katty | twisted[asteria]: i have a job for you. |
| 16:20.27 | sigterm | ut oh |
| 16:21.00 | whitefunk | hey guys, quick question, what is the trend now for people needing ss7 support within asterisk? |
| 16:22.37 | *** join/#asterisk ender (n=me@fedora/ender) |
| 16:22.39 | jake1932 | http://www.voip-info.org/tiki-index.php?page=Asterisk+bounty+SS7 the bounty is at $4000 |
| 16:22.45 | EvilRick | what abut the calling numebr plan ? |
| 16:22.53 | EvilRick | shoudl taht be unknown as well? |
| 16:23.54 | wunderkin | EvilRick: i have an LD outgoing only line and i have everything commented out except for the essentials: signalling, switchtype, callerid, group, context, channel |
| 16:24.36 | jake1932 | twisted[asteria]: i could ask the admissions guys at the univ, but i'm thinking they might be a bit biased |
| 16:24.51 | coppice | jake1932: if QWest made you do full interop tests, that alone would cost upwards of $100k :-) |
| 16:25.34 | jake1932 | coppice: i'm sorry - i was referring to the previous question (NOT ss7) |
| 16:25.35 | whitefunk | jake1932 : looks like there isnt much documentation on the integrating |
| 16:25.40 | drumkilla | and the chances of getting that money is pretty slim ... |
| 16:25.44 | whitefunk | jake1932 : integration* |
| 16:25.45 | FuriousGeorge | can someone take a loock at this, i'm having trouble "including" a file in my extensions.conf: http://pastebin.ca/26185 |
| 16:26.12 | FuriousGeorge | do i have to start the included file with "[included_file_name]" or something |
| 16:26.21 | bsdfreak | boh? |
| 16:26.22 | bsdfreak | BOH |
| 16:26.29 | FuriousGeorge | ? |
| 16:26.30 | coppice | jake1932: I mean a $4k bounty is pretty irrelevant for SS7 |
| 16:27.13 | EvilRick | wunderkin, you wouldnt happen to knwo it that works on South African PRI's ? |
| 16:28.47 | wunderkin | EvilRick: i dont know a whole lot about it, i can only tell you what works for me, i asked for the real problem but you never said |
| 16:28.47 | jake1932 | coppice: wouldn't the trend of people needing asterisk support for ss7 be relevent to the bounty? |
| 16:28.57 | IronHelix | doesnt it have to be #include <filename.conf>? |
| 16:28.57 | *** join/#asterisk brettnem (n=brettnem@72.29.102.158) |
| 16:29.15 | jake1932 | IOW if it was really needed, the bounty would be higher? |
| 16:29.18 | whitefunk | so do these cats just get the stack workong on the platform ? |
| 16:29.29 | bsdfreak | i dont care if you dont |
| 16:30.06 | coppice | jake1932: americans tell me all the major telcos there insist on certification. I'm skeptical about that, but if its really a hard rule someone needs to come up with at least $100k |
| 16:30.32 | FuriousGeorge | is this not how you include a file in your dialplan? http://pastebin.ca/26185 |
| 16:30.33 | jake1932 | we're 4% of the way there already |
| 16:31.33 | IronHelix | ah, well, sort of |
| 16:31.48 | IronHelix | you can #include <file.conf> any files |
| 16:31.58 | IronHelix | but then you also have to include => contexts inside them |
| 16:32.03 | twisted[asteria] | haha yay |
| 16:32.14 | *** join/#asterisk Astinus (i=iBook@freenode/staff/gentoo.astinus) |
| 16:32.18 | twisted[asteria] | A250 + BT120 + OSX + X-Lite == cool |
| 16:32.36 | EvilRick | wunderkin, well.. basically I get a line busy message every time I try dial |
| 16:32.42 | IronHelix | what i usually do is at the top of the file #include all the extra files i will be using |
| 16:32.43 | EvilRick | I can send you my pridebug |
| 16:33.02 | EvilRick | but it seems like the exchange just hangs up my line when I dial |
| 16:33.09 | IronHelix | then have my dialplans include => them when necessary or have the channels refer to them directly |
| 16:33.19 | Inkubot | damn disconnect supervision |
| 16:33.39 | jake1932 | you guys are all computer pros - does a degree matter today? |
| 16:33.58 | FuriousGeorge | IronHelix: i have to include contexts from the original file in the included file? |
| 16:34.04 | IronHelix | other way |
| 16:34.13 | IronHelix | you have to include contexts from the included file in the original file |
| 16:34.39 | IronHelix | once you #include a file, it treats that file and extensions.conf as one big text |
| 16:34.49 | FuriousGeorge | oh i see, so the included file probably just needs to start with [context_name] and i'll be all set |
| 16:35.17 | IronHelix | well if the included file doesnt start with a [context] header |
| 16:35.18 | FuriousGeorge | with a few other minor tweaks, brb |
| 16:35.25 | IronHelix | then you might be able to get away with what you're doing |
| 16:35.38 | IronHelix | but you still need to change #include "file.conf" to #include <file.conf> |
| 16:35.40 | wunderkin | EvilRick: i dont know, maybe wait awhile and see if someone else can help, or maybe you can call your telco and see if they see the calls going through maybe theres a problem or can help you figure it out, my experience with something like that has been they wont but maybe you have a better telco |
| 16:36.00 | FuriousGeorge | IronHelix: k, brb |
| 16:37.12 | FuriousGeorge | so the original "#include ,file.conf> could go in general section and i can include it like a normal context below |
| 16:37.33 | FuriousGeorge | in dialplan |
| 16:38.01 | IronHelix | yup |
| 16:38.15 | IronHelix | just make sure the file.conf starts with a [context] header so there is something to include |
| 16:38.42 | EvilRick | wunderkin, rofl.. in this country... I dobpt it. |
| 16:38.49 | EvilRick | doubt |
| 16:39.41 | FuriousGeorge | IronHelix: yeah, i think its gonna work, i'll et you know "when convenient" :) |
| 16:39.51 | IronHelix | hehe |
| 16:40.02 | IronHelix | i'm about to run off but you can zap me a /msg or something |
| 16:40.03 | IronHelix | later |
| 16:40.29 | FuriousGeorge | later |
| 16:43.00 | *** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) |
| 16:43.40 | BladeRunner05 | Ciao HironHelix |
| 16:44.25 | FuriousGeorge | http://pastebin.ca/26188 <-- this include doesnt seem to be working either :| |
| 16:49.16 | FuriousGeorge | the wiki is really skimpy on the include command |
| 16:50.03 | *** join/#asterisk SplasPood (i=jwb@ludicrous.paravolve.net) |
| 16:50.25 | *** join/#asterisk erickj_az (n=erickj_a@wsip-68-98-222-74.ph.ph.cox.net) |
| 16:50.55 | erickj_az | is there someone here that can help me with iax and/or iax trunking? |
| 16:57.20 | jake1932 | erickj_az: http://lists.digium.com/pipermail/asterisk-dev/2003-October/001927.html |
| 16:57.26 | kll | how powerful machine do I need to be able to use all my four channels on my E1 card? Is a single cpu machine enough? |
| 16:57.47 | erickj_az | I thought I could dial into another context by appending @context to the iax2 dial string |
| 16:58.14 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 16:58.17 | erickj_az | i.e. IAX2/trunk_demo:trunk_d3m0@192.168.0.70/${EXTEN}@context |
| 16:58.22 | erickj_az | is that corrent? |
| 16:58.31 | jake1932 | never did that |
| 16:59.01 | file[laptop] | you can yes, but you need to be allowed to do it |
| 16:59.19 | file[laptop] | if it was written so you could use any context, that would open up many ways of like abuse |
| 16:59.32 | erickj_az | Ahh... How do I go about allowing that to happen? |
| 16:59.54 | file[laptop] | never done it but you might try multiple context lines in the iax.conf user entry, or no context line |
| 17:00.10 | erickj_az | OK I'll try. |
| 17:01.42 | FuriousGeorge | shoot thats my problem. what did i do wrong here: exten => _120120[0,4,7-9]XXXX,1,macro(local_call,${EXTEN},zap/G1) |
| 17:01.46 | jake1932 | kll: http://www.voip-info.org/wiki/view/Asterisk+dimensioning |
| 17:02.14 | jake1932 | kll: 1 processor (chassis) (example: 2.8Ghz P4) for every 4 E1's (example: one TE405P card) |
| 17:02.14 | FuriousGeorge | can you not use the [ ] with the XXXX? |
| 17:02.35 | erickj_az | no context line did not work |
| 17:04.24 | erickj_az | multiple context lines was successful |
| 17:04.54 | *** join/#asterisk nain (n=nain@202.154.245.18) |
| 17:05.16 | FuriousGeorge | exten => _120120[0,4,7-9]XXXX,1,macro() < Asterisk is parsing that as extension 1201200 priority 4 |
| 17:05.44 | nain | Hi |
| 17:06.42 | rene- | do i have to include the macro context in a macro in the calling context? |
| 17:07.13 | rene- | i meant do i have to include the macro context in the macro call calling context? |
| 17:07.26 | rene- | because it isnt reaching it |
| 17:07.45 | FuriousGeorge | rene-: no, but you have to call the macro context [macro-BLAH] |
| 17:07.54 | FuriousGeorge | and call it with macro(blah) |
| 17:08.07 | *** join/#asterisk ful|work (n=fulgas@209.8.233.170) |
| 17:08.24 | nain | I am unable to ping my Gateway from Linux Machine having Asterisk Installed in same network and on other side Gateway is pinging in same network from Windows Machine ? |
| 17:08.29 | FuriousGeorge | exten => _120120[0,4,7-9] I CANT DO THIS!!!! |
| 17:08.48 | snitt | FuriousGeorge: then do [0,4,7,8,9] |
| 17:09.16 | FuriousGeorge | snitt: i know, i just wanted to make sure that was the problem, because i need to correct 300 patterns now |
| 17:09.25 | snitt | uh |
| 17:09.29 | *** join/#asterisk shmundler (n=nunya@mtnuser.icgws.com) |
| 17:09.38 | snitt | FuriousGeorge: i dont know if that is the problem |
| 17:09.50 | snitt | [7-9] and [7,8,9] works ok, but not [7,8-9] |
| 17:10.04 | shmundler | •ursuspacificus• e |
| 17:10.33 | shmundler | •supaigtr• u here |
| 17:10.47 | rene- | furious im using realtime extensions, but iam getting this in the cli No such context 'macro-stdexten,5250,SIP/5250' |
| 17:11.00 | nain | shmundler: fedora3 without firewall having problem in pinging Gateway from fc3 but not from windows xp in same network ? |
| 17:11.05 | rene- | for macro 'stdexten,5251,SIP/5251' |
| 17:11.39 | rene- | that was another line but anyways |
| 17:11.48 | rene- | does the macro works with realtime extensions? |
| 17:12.27 | *** join/#asterisk Inv_arp (i=junya@adsl-144-134-75.mia.bellsouth.net) |
| 17:12.59 | rene- | im using this as arguments to Macro (appdata) stdexten,${EXTEN},SIP/${EXTEN} |
| 17:13.26 | rene- | should i be using () around parameters for this ? |
| 17:15.05 | *** join/#asterisk doctor_za_ljubav (i=bkwyg@195.252.90.250) |
| 17:15.15 | doctor_za_ljubav | Hello everyone i have problem with False anwer on Asterisk, can anyone help me with configuration? What do i need to change |
| 17:17.49 | shmundler | doctor_za_ljubav what do you mean by false answer? |
| 17:18.05 | doctor_za_ljubav | when i dial the number it starts billing right away |
| 17:18.34 | *** join/#asterisk dalabera (n=Dalabera@pmr.pmrtechnologies.com) |
| 17:18.38 | shmundler | dont Answer() the call |
| 17:18.57 | doctor_za_ljubav | i need a parameter to change in asterisk that will send a signal to switch that call is answered |
| 17:19.32 | doctor_za_ljubav | it should not be billed right after dialing the call, but when call is answered |
| 17:19.38 | file[laptop] | doctor_za_ljubav: analog? |
| 17:19.42 | doctor_za_ljubav | thanks shmundler*** |
| 17:19.43 | doctor_za_ljubav | :) |
| 17:19.53 | file[laptop] | shmundler: nice. |
| 17:20.04 | shmundler | thnx |
| 17:20.13 | doctor_za_ljubav | analog or digital it should be the same parameter |
| 17:20.22 | doctor_za_ljubav | i tried callprogress in zapata conf but its the same |
| 17:20.35 | file[laptop] | call progress on analog is... not that good/non-existant |
| 17:21.09 | doctor_za_ljubav | on digital? |
| 17:21.27 | file[laptop] | on digital it's beautiful because it's usually out of band |
| 17:21.41 | file[laptop] | thus why I love digital |
| 17:21.46 | dalabera | guys What I have to do in order to debug faxes with spandsp? Would to see what's going on under the hood, during the transmission... |
| 17:21.49 | file[laptop] | PRIs are tasty, D3s are nice |
| 17:23.01 | FuriousGeorge | hmmm exten => _120120[0,4,7,9]XXXX??? |
| 17:23.02 | coppice | dalabera: what is wrong with your faxes? |
| 17:23.21 | *** join/#asterisk DarthClue (i=user76@wsip-68-99-73-32.tu.ok.cox.net) |
| 17:23.22 | FuriousGeorge | whats wrong with this: exten => _120120[0,4,7-9]XXXX,1,macro(blah) |
| 17:23.46 | FuriousGeorge | i mean _120120[0,4,7,8,9]XXXX,1,macro(blah) |
| 17:23.56 | dalabera | Sending fax to a Ricoh 2035sp and the pages are cut off or someones get blank |
| 17:24.45 | FuriousGeorge | file[laptop]: how much is that usb phone on your site? |
| 17:24.57 | file[laptop] | uh wha? |
| 17:25.12 | FuriousGeorge | asterlink usb phone |
| 17:25.15 | *** join/#asterisk fiXXXerMet (n=Kyle@ip67-154-236-201.z236-154-67.customer.algx.net) |
| 17:25.19 | FuriousGeorge | i thought you worked for asterlink? |
| 17:25.26 | file[laptop] | yes |
| 17:25.29 | file[laptop] | we don't sell a usb phone |
| 17:25.48 | fiXXXerMet | I'm trying to setup a Linksys PAP2-VN with asterisk, but have no idea how to configure it to use my box... Can anyone point to some docs/help? |
| 17:26.06 | FuriousGeorge | file[laptop]: sorry, confused you with sineapps |
| 17:26.19 | *** join/#asterisk pavelnedved (n=afso7523@freedom.rd.francetelecom.com) |
| 17:26.22 | file[laptop] | ah I just host Sineapps |
| 17:26.28 | Inv_arp | dont get this, sometimes my * box just gets unregistered from sip provider , sip reload fixes it |
| 17:26.28 | pavelnedved | hello |
| 17:27.59 | FuriousGeorge | hey file[laptop] since i got your attention can you tell me why this: _197399[1,2,8]XXXX is being parsed as -- Added extension '_197399[1' priority 2 to local |
| 17:28.21 | file[laptop] | _197399[128]XXXX |
| 17:28.22 | nain | Any Linux Expert here ? |
| 17:28.58 | Nugget | Do you really need an expert, or do you just need someone who knows the answer to your question? |
| 17:28.59 | JerJer | nope, we all run Amiga |
| 17:29.05 | FuriousGeorge | i want 1973991XXXX 1973992XXXX and 1973998XXXX |
| 17:29.22 | file[laptop] | FuriousGeorge: did you even try what I said? |
| 17:29.24 | JerJer | FuriousGeorge: you just typed the regex you need - minus the _ |
| 17:29.27 | nain | Nugget: I just need the person to help me in a question |
| 17:29.33 | Nugget | so ask the damn question. |
| 17:29.39 | Nugget | none of us are mindreaders. |
| 17:29.43 | nain | Nugget: Thanks, |
| 17:29.55 | file[laptop] | trial/error, it's a good thing. |
| 17:30.10 | jake1932 | unless it's a fatal error |
| 17:30.16 | jake1932 | that's a bad thing |
| 17:30.18 | FuriousGeorge | file[laptop]: gotcha |
| 17:30.20 | file[laptop] | then you're dead, minor detail |
| 17:30.24 | Nugget | I have to wonder, if nain was choking on a piece of steak, if he'd gesture violently to find out if anyone in the restaurant was an expert in cardiology. :) |
| 17:30.26 | pavelnedved | dear friends, i would like to know when you are installing asterisk on a linux x86 server, you are obliged to install specific drivers for dialogic D/300JCT-1E1 E1 + 30 voice card, or are the drivers provided by asterisk ? |
| 17:30.28 | dalabera | Sending fax to a Ricoh 2035sp and the pages are cut off or someones get blank |
| 17:30.38 | jake1932 | hah |
| 17:30.39 | *** join/#asterisk clyrrad (n=ddd@CPE001195f553c7-CM0011aea484a4.cpe.net.cable.rogers.com) |
| 17:30.41 | nain | Nugget: the problem is very simple that i am not able to ping Gateway from Linux Machine in a same network but pinging from Windows Machine . No Firewall |
| 17:30.44 | file[laptop] | pavelnedved: you have to contact Digium for dialogic drivers, they cost money |
| 17:31.12 | Nugget | can you ping any of the other IPs on the subnet? |
| 17:31.19 | *** join/#asterisk b0xii (i=b0xii@pool-70-110-88-241.dfw.dsl-w.verizon.net) |
| 17:32.01 | nain | Nugget: Yes i can ping DSL Gateway IP as well as DNS but not the particular IP. |
| 17:32.16 | Nugget | I'm confused, I thought you said you could not ping the gateway ip. |
| 17:32.34 | pavelnedved | is there any hardware card that do not cost money file[laptop] concerning the drivers ? |
| 17:32.51 | FuriousGeorge | file[laptop]: is _197355[124-8] allowed? |
| 17:32.52 | file[laptop] | pavelnedved: digium cards, or sangoma cards |
| 17:33.06 | nain | Nugget: here the gateway i am talking about is the voip gateway, and the dsl gateway is pinging and other ip in the subnet |
| 17:33.34 | Nugget | I do not understand your situation, then |
| 17:33.38 | nain | Nugget: The voip gateway ip is not pinging from linux machine but it is pinging from the windows machine and outside the world |
| 17:33.52 | pavelnedved | so that means if i buy digium cards i do not have to buy the drivers ? |
| 17:34.01 | jake1932 | and you get install support |
| 17:34.11 | Nugget | so there is just a single IP you cannot ping from the linux machine, and the others are pingable? |
| 17:34.25 | supaigtr | Anyone know how to capture IAX traffic and decode it or possilbiy play it back? |
| 17:34.29 | nain | Nugget: 100% right other ips are pingable |
| 17:34.55 | *** join/#asterisk Seyr (n=seyr@cpe-67-10-139-141.houston.res.rr.com) |
| 17:35.02 | Nugget | what are the IPs? Is there a chance you have a bad netmask on either the linux box or the unpingable machine? |
| 17:35.05 | nain | Nugget: when i ping the particular ip i got "Do you want to ping broadcast? Then -b" |
| 17:35.13 | pavelnedved | are drivers included in Asterisk for digium cards ? |
| 17:35.21 | Nugget | that sounds like your netmask is bad on the linux box. |
| 17:35.49 | nain | Nugget: let me check it again but i think both machine have same netmask |
| 17:35.49 | file[laptop] | FuriousGeorge: try, and, see |
| 17:35.55 | file[laptop] | pavelnedved: they're a separate download but they're free |
| 17:35.57 | jake1932 | supaigtr: http://lists.digium.com/pipermail/asterisk-dev/2005-April/011004.html |
| 17:36.03 | pavelnedved | or we have to install first drivers on a linux server for the digium cards, then installing asterisk ? |
| 17:36.14 | jake1932 | pavelnedved: the drivers are in asterisk |
| 17:36.36 | pavelnedved | thanks a lot dear collegues |
| 17:36.47 | JerJer | this makes me sick: http://tinyurl.com/aefk6 |
| 17:37.07 | nain | Nugget: I check both machines have same subnetmask |
| 17:37.28 | Nugget | what's the ip/netmask of the linux box and what's the ip you cannot ping? |
| 17:40.23 | Ariel_ | wow AT&T is being taken over by SBC |
| 17:41.11 | *** join/#asterisk X-Rob_ (n=xrob@dsl-202-173-151-24.qld.westnet.com.au) |
| 17:41.12 | supaigtr | jake1932: I have the dissector for ethereal but I don't see an option to follow the call. My problem is one way loss of audio for a few seconds and I'm trying to debug where the audio is getting lost. |
| 17:41.47 | jake1932 | if you're only doing one call, shouldn't be difficult to figure out |
| 17:42.13 | jake1932 | or does it only happen on multiple calls |
| 17:43.04 | pavelnedved | wow AT&T is being taken over by SBC ? |
| 17:43.13 | pavelnedved | what is SBC ? |
| 17:44.05 | supaigtr | jake1932: During day it happens on less than 10 out of 1000. |
| 17:45.27 | *** join/#asterisk cripito (n=ncripito@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 17:45.29 | cripito | hi |
| 17:45.39 | Inkubot | anyone know why my * is not showing the caller id |
| 17:45.47 | Inkubot | i've got two fxo |
| 17:45.49 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 17:45.49 | supaigtr | IAX2/pbx-1 4 2 88 26 0 8 0 2 0 59 0 0 0 0 0 |
| 17:46.01 | supaigtr | does that look normal for an iax2 show netstats? |
| 17:46.22 | FuriousGeorge | file[laptop]: the answer is no, you cant do that :) |
| 17:46.43 | pavelnedved | have a nice week end and thanks a lot for the answers again. |
| 17:47.07 | cripito | O:-) |
| 17:47.31 | FuriousGeorge | file[laptop]: i mean, yes u can |
| 17:48.32 | dalabera | cripito let me try... |
| 17:48.42 | cripito | uploading the installation |
| 17:49.11 | cripito | and i will help u with the installation too |
| 17:49.52 | *** join/#asterisk redhate (n=redhate@201.18.141.194) |
| 17:52.33 | *** join/#asterisk kippi (n=kippi@cpc4-hatf3-6-0-cust33.lutn.cable.ntl.com) |
| 17:52.34 | kippi | hey |
| 17:53.19 | cripito | hey |
| 17:53.24 | cripito | i scared the guy? |
| 17:53.27 | *** part/#asterisk pavelnedved (n=afso7523@freedom.rd.francetelecom.com) |
| 17:53.36 | kippi | I am trying to get onto my asterisk box remotely. Sometimes I can get on, other times it won't register. I don't get anything in the log from asterisk. Has anyone had this before? |
| 17:54.08 | bsdfreak | heh |
| 17:54.49 | cripito | well the offer is there ... http://www.cripiland.com/screenshots/manager4.jpg ... only * with realtime (mysql) |
| 17:55.52 | kippi | is there anything on asterisk to block people from connecting? |
| 17:55.54 | cripito | there is anyone here from asterisk.net? |
| 17:57.35 | vader-wrk | what does congestion mean in the extensions.conf? |
| 17:57.58 | file[laptop] | it means Congestion... makes the congestion sound... |
| 17:59.44 | Aughey | is actually doesn't make the congestion sound....Playtones(congestion) would do that. It just sends the signal to indicate congestion |
| 18:00.51 | file[laptop] | it depends |
| 18:00.58 | file[laptop] | if the call is answered, Asterisk will generate it |
| 18:01.06 | file[laptop] | if SIP is set for inband progress, Asterisk will generate it |
| 18:01.17 | file[laptop] | if the call is not answered, it's signalled to indicate it |
| 18:09.31 | cripito | mmm this is quiet like a cementary today!!! |
| 18:09.41 | JerJer | yep, they are all dying to get in here |
| 18:09.54 | cripito | :) hi jerjer |
| 18:11.19 | JerJer | moo |
| 18:11.48 | *** join/#asterisk pussfeller (n=todd@12.150.129.170) |
| 18:12.22 | supaigtr | moo moo |
| 18:12.55 | supaigtr | arrgh. IAX2 is a pain to debug when you don't know what yer doing. :( |
| 18:13.10 | JerJer | figure it out then |
| 18:14.11 | lesouvage | is there a way to delay the reaction of an asterisk box to an incoming call so a normal phone can ring 5 times before asterisk handles the phone. |
| 18:14.20 | JerJer | Wait,5 |
| 18:14.29 | JerJer | although thats seconds |
| 18:14.35 | JerJer | so Wait,25 |
| 18:14.41 | supaigtr | :) easier said than done. I'm already in a crash course for about 10 other technologies related to getting IAX2 there. |
| 18:15.03 | *** join/#asterisk MacRohard (i=rm@80.68.89.47) |
| 18:15.14 | lesouvage | jerjer: should I add this to the zapata.conf for every channell? |
| 18:15.23 | MacRohard | I am Originate'ing a call through the manager interface and I set some variables there with Variable: foo=bar, etc. In the context specified in the originate it goes into a context that runs an AGI. This works fine, but for some reason the AGI isn't seeing the variables I set in the Variable: section of the manager request... any ideas? |
| 18:15.42 | JerJer | lesouvage: um no |
| 18:16.02 | *** join/#asterisk pussfeller (n=todd@12.150.129.170) |
| 18:16.08 | lesouvage | Jer,Jer: where should I add to make it work? |
| 18:16.16 | JerJer | the dialplan |
| 18:16.16 | *** join/#asterisk pooh_ (n=hfwang@cust.15.241.adsl.cistron.nl) |
| 18:16.29 | Aughey | you add it to the entry point for your dialin context |
| 18:16.45 | JerJer | yeah that |
| 18:17.18 | lesouvage | Aughey: like exten => s,1,Wait(25) |
| 18:17.42 | JerJer | if your zap channel does not provide DNIS |
| 18:17.46 | JerJer | then yes |
| 18:18.23 | *** join/#asterisk Juggie (i=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
| 18:20.13 | *** join/#asterisk wolfson (n=hehe@usr-kdh-208-6-58-26.beachlink.com) |
| 18:21.10 | lesouvage | JerJer: I will try it. Is there an option when there seem to be DNIS? |
| 18:21.28 | JerJer | dnis is a service |
| 18:21.54 | JerJer | you are going to want to first learn about telcom in general before diving into Asterisk |
| 18:22.09 | JerJer | google is your friend |
| 18:22.15 | JerJer | abuse her |
| 18:22.17 | Aughey | voip-info.org is too |
| 18:22.29 | JerJer | voip-info is full of incorrect information |
| 18:22.35 | Aughey | and old info |
| 18:22.37 | JerJer | and no i am not gonna fix it |
| 18:23.05 | Aughey | is the Asterisk O'Riley book any good? I haven't seen it yet |
| 18:23.12 | JerJer | yes |
| 18:23.13 | JerJer | very |
| 18:23.43 | Aughey | I read the chapter on amazon on extensions. That was good but I didn't know how indepth it goes |
| 18:23.56 | CoffeeIV_ | I have it, it's good (the O'Reilly book) (haven't read it cover to cover yet) |
| 18:24.21 | *** join/#asterisk pussfeller (n=todd@12.150.129.170) |
| 18:26.25 | doctor_za_ljubav | is there anywhere in asterisk configuration answer delay option? |
| 18:26.26 | JerJer | Aughey: well read the whole line onine: http://www.asteriskdocs.org |
| 18:26.36 | JerJer | Click on "read the book online" |
| 18:26.47 | JerJer | then if you like it buy a printed copy - I did |
| 18:27.06 | *** join/#asterisk PupenoL (n=pupeno@200.123.183.89) |
| 18:27.07 | Aughey | hay, that's great. I got it. |
| 18:27.22 | *** join/#asterisk coppice (n=chatzill@194.143.17.210.dyn.pacific.net.hk) |
| 18:27.32 | lesouvage | Yoy can , legely, download the book on this link http://geekgazette.com/index.php?option=com_content&task=view&id=40&Itemid=2 . It's a great book but DNIS isn't explained. |
| 18:28.01 | *** join/#asterisk jjhall (n=chatzill@94-253.69-92-cpe.cableone.net) |
| 18:28.35 | *** join/#asterisk ejo1 (n=ejo1@kinetic-host39.dsl.visi.com) |
| 18:28.40 | ender | anybody know of a headset for Polycom phones that would allow remote pickup and hangup of lines? |
| 18:29.40 | JerJer | lesouvage: old news |
| 18:29.50 | JerJer | that was soo 2 minutes ago |
| 18:30.13 | Aughey | close to 3 |
| 18:30.14 | *** join/#asterisk Tili (n=Tili@211.147.234.5) |
| 18:31.26 | *** join/#asterisk watchy (n=watchy@adsl-69-152-41-249.dsl.ltrkar.swbell.net) |
| 18:31.30 | jjhall | Does anyone have any recomendations of an outbound fax service? I just want to be able to deposit a couple of dollars into a prepaid account and be able to send once in a while. I don't need it near often enough to warrant a monthly service like eFax. |
| 18:31.48 | JerJer | i would but it would be biased |
| 18:32.04 | *** join/#asterisk n0rf- (n=n0rf@ip152.66.1311D-CUD12K-03.ish.de) |
| 18:32.12 | n0rf- | hey guys |
| 18:32.38 | n0rf- | where can i find an updated musiconhold.conf? |
| 18:32.52 | JerJer | asterisk/configs/ |
| 18:33.28 | n0rf- | cheers |
| 18:33.47 | lesouvage | doctur_za_ljabav:exten => s,1,Wait(25) if your zap channel does not provide DNIS |
| 18:33.54 | Aughey | the latest is here if you don't want to download it all http://washucsc.org/~jha/musiconhold.conf.sample |
| 18:34.53 | *** join/#asterisk CANO-1982 (i=alejandr@201.255.51.193) |
| 18:35.01 | drumkilla | you can blame me, i'm the one that changed the code so it would complain and tell you to update your config :) |
| 18:36.07 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 18:36.16 | JerJer | but your not in Canada |
| 18:36.30 | drumkilla | crap |
| 18:36.38 | JerJer | 404 |
| 18:36.45 | drumkilla | 404!!! |
| 18:36.50 | synthetiq | how can u check what ocdecs are in use? |
| 18:36.50 | drumkilla | you can't find me. |
| 18:37.14 | JerJer | synthetiq: like available on your box? |
| 18:37.15 | synthetiq | codecs |
| 18:37.35 | drumkilla | show codecs? something like that ... |
| 18:37.37 | JerJer | show translations |
| 18:37.49 | JerJer | show codecs just list the bit mapping |
| 18:37.55 | drumkilla | the translation table is sweet |
| 18:38.02 | JerJer | yes, yes it is |
| 18:38.30 | JerJer | <PROTECTED> |
| 18:38.53 | synthetiq | cute table |
| 18:39.39 | n0rf- | down to 2 warnings yet still * won't start after the update |
| 18:39.53 | JerJer | asterisk -vvvgc |
| 18:40.00 | JerJer | and read update.txt |
| 18:40.23 | ender | no headset users w/ Polycom phones? |
| 18:40.37 | synthetiq | weird i can do show trnsaltion on one box and not the other |
| 18:41.11 | synthetiq | ahh there it is |
| 18:41.45 | JerJer | speeling it write is halph da battel |
| 18:42.21 | n0rf- | hmm |
| 18:42.26 | n0rf- | getting a segfault |
| 18:42.33 | n0rf- | "Ouch ... error while writing audio data: : Broken pipe" |
| 18:42.42 | JerJer | that's not an error |
| 18:42.48 | JerJer | look higher in the output |
| 18:42.59 | n0rf- | it happened after loading skinny before... i've noloaded that one |
| 18:43.37 | X-Rob_ | n0rf-, that error is from mpg123 |
| 18:43.45 | X-Rob_ | that's not the problem |
| 18:45.36 | n0rf- | weird.. rest of the output looks ok to me |
| 18:46.02 | *** join/#asterisk znoG (n=gs@200.115.218.81) |
| 18:47.44 | *** join/#asterisk HuNTER_SC (i=Junior@200-138-246-239.fnsce7006.dsl.brasiltelecom.net.br) |
| 18:48.00 | *** join/#asterisk geryon2005 (n=Steve@69.212.137.6) |
| 18:48.29 | *** join/#asterisk HuNTER_SC (i=Junior@200-138-246-239.fnsce7006.dsl.brasiltelecom.net.br) |
| 18:53.46 | *** join/#asterisk Hunter_SC (i=Junior@201-2-230-36.fnsce7006.dsl.brasiltelecom.net.br) |
| 18:54.33 | *** join/#asterisk shmaltz (n=chatzill@69.28.255.210) |
| 18:54.53 | shmaltz | anybody have any clue if something is wrong with the list? |
| 18:55.08 | *** join/#asterisk ohad (n=ohad@19-231-13-72.cosmoweb.net) |
| 18:55.17 | Hunter_SC | they alguem knows to say me if I to make a connection of my HT with asterisk I saw vpn I go to give lag very? Connection 300k! |
| 18:55.22 | ohad | hi, is there a way to retrieve del. VM's? |
| 18:55.29 | *** join/#asterisk Primer (n=primus@sh.nu) |
| 18:55.37 | ohad | one of my users just deleted them now. |
| 18:56.00 | Primer | Anyone working on wrt asterisk here? I'm curious if builidng something outside of the asterisk tree will be...challenging... |
| 18:56.23 | Primer | I'd like to setup asterisk on my wrt54g, but I need chan_sccp (chan-sccp.org) |
| 18:56.28 | *** join/#asterisk dalabera (n=Dalabera@pmr.pmrtechnologies.com) |
| 18:56.57 | ohad | anyone? |
| 18:57.36 | shmaltz | anybody here tried ADSI with asterisk? |
| 18:57.44 | CoffeeIV_ | ohad: check the /var/spool/asterisk/voicemail/XXX/ dir, where XXX is the extension. If the files are still available they will be somewhere in there |
| 18:59.27 | CoffeeIV_ | ohad: if you have an ext2 filesystems instead of ext3, and are prepared to stop the server now, mount the disk read-only with a live CD like knoppix, and fiddle around for hours, you *might* be able to undelete the files. However I suspect you are screwed, sorry. |
| 19:00.54 | *** join/#asterisk toddf (n=toddf@ns0.fries.net) |
| 19:01.23 | ohad | CoffeeIV_, the files are not there. it removed them completely. |
| 19:03.15 | CoffeeIV_ | ohad: I think they are gone. If your asterisk is set up to monitor every call that goes through it you might have a recording of hte whole voicemail session where they left the message. (this is unlikely, most people don't set it up that way) Recovering the files now is probably too expensive, and might not be possible |
| 19:03.39 | *** join/#asterisk jpm_SD (n=jpm@207-40-115-38.sugardog.com) |
| 19:03.45 | doctor_za_ljubav | i need asterisk help please pvt |
| 19:04.01 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:04.18 | jpm_SD | doctor_za_ljubav, perhaps you could be a little more specific? |
| 19:05.26 | Lathos42 | Quick Question.. Hyperthreading, good or bad? |
| 19:05.34 | *** part/#asterisk mcf3782 (n=mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net) |
| 19:05.36 | ender | Lathos42: depends |
| 19:05.39 | jjhall | ohad: Do you have Asterisk set to e-mail voice mail messages to the user? Perhaps you can recover the files from your mail server if so. |
| 19:05.55 | *** part/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 19:06.05 | Lathos42 | ender: Depends on what phase the moon is in? |
| 19:06.07 | Lathos42 | :) |
| 19:06.16 | ender | Lathos42: depends on what you're using the CPU for. |
| 19:06.21 | doctor_za_ljubav | <doctor_za_ljubav> i have problem with false answer |
| 19:06.43 | lesouvage | doctor_za_ljubav: what 's the problem? |
| 19:06.48 | Lathos42 | ender: It'll be dedicated to *, with little to no transcoding |
| 19:06.50 | doctor_za_ljubav | unanswered calls should not be billed, but my asterisk is not sending that signal |
| 19:07.03 | doctor_za_ljubav | that call is connected |
| 19:07.06 | doctor_za_ljubav | or ringing |
| 19:07.11 | Lathos42 | ender: Sangoma T1 card |
| 19:07.21 | file[laptop] | doctor_za_ljubav: is it a PRI, Analog line, what? |
| 19:07.23 | ender | Lathos42: *shrug* it most likely won't hurt (kernel 2.6 I assume?). Won't help a whole lot. |
| 19:07.34 | doctor_za_ljubav | it is connected to Sonus |
| 19:07.43 | doctor_za_ljubav | no zaptel devices |
| 19:07.53 | Lathos42 | ender: Ok, I just wanted to make sure that it wasnt going to cause any problems to leave it enabled |
| 19:08.13 | ender | shouldn't. |
| 19:08.18 | doctor_za_ljubav | connected to closed system |
| 19:08.22 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:09.16 | lesouvage | doctor_za_ljubav: what does the cli show? (asterisk -vvvvvvvr) |
| 19:15.05 | doctor_za_ljubav | call is going thru without a problem |
| 19:15.18 | doctor_za_ljubav | all calls are billed no metter if someone answered it or not |
| 19:16.02 | doctor_za_ljubav | i need to change something so there would be some answer supervision or something and i cant find where that parameter is |
| 19:16.06 | shmaltz | anybody here ever tried adsi phone with asterisk? |
| 19:16.34 | shmaltz | doctor_za_ljubav, what type of hardware you using? |
| 19:16.52 | doctor_za_ljubav | on asterisk? or? |
| 19:17.03 | shmaltz | yeah on asterisk? |
| 19:17.09 | shmaltz | or what? |
| 19:17.10 | doctor_za_ljubav | just a server |
| 19:17.57 | shmaltz | and how you connecting? |
| 19:18.07 | doctor_za_ljubav | its connected to sonus system |
| 19:18.13 | doctor_za_ljubav | no zaptel devices |
| 19:18.20 | shmaltz | how is it connected? |
| 19:18.56 | shmaltz | doctor_za_ljubav, how is it connected to the sonus system? |
| 19:19.29 | doctor_za_ljubav | its connected lan to lan |
| 19:19.40 | doctor_za_ljubav | just to pass the call on termination system |
| 19:20.17 | file[laptop] | what protocol? SIP? |
| 19:20.21 | shmaltz | so maybe the sonus sytem is answering it? |
| 19:20.23 | doctor_za_ljubav | h323 |
| 19:21.01 | shmaltz | thank you doctor_za_ljubav, I am assuming that you are either from england, or from a country that was under the british for a long time. Am I wrong? |
| 19:21.25 | doctor_za_ljubav | no u r wrong |
| 19:21.33 | doctor_za_ljubav | why do you think that? |
| 19:21.56 | doctor_za_ljubav | anyway its not suppose to be realy problem, just one parameter change, i tried callprogress yes, but it didnt work |
| 19:21.59 | shmaltz | so where you from? |
| 19:22.06 | doctor_za_ljubav | serbia originally |
| 19:22.57 | jjhall | doctor_za_ljubav: My guess is Sonus systems is sending the signal that the call is connected, it more than likely is not an issue with your box, but of the provider and/or the destination network. |
| 19:23.01 | shmaltz | where are you now? |
| 19:23.38 | shmaltz | well, that fact that I had to grill you for 10 minutes until you answered h323 makes me think that |
| 19:23.50 | doctor_za_ljubav | US |
| 19:24.00 | doctor_za_ljubav | thanks jjhall i will check that too |
| 19:24.16 | doctor_za_ljubav | just want to exclude that something is wrong with asterisk box |
| 19:24.29 | jjhall | doctor_za_ljubav: For example, when I dial my cell phone, my cellular company sends the answer signal while it is ringing the phone. This is outside the control of my asterisk box, and my VoIP providers. It is a pain because I can |
| 19:24.52 | jjhall | 't ring my cell phone and my desk phone simultaneously as it will always show my cell as immediately answering. |
| 19:24.55 | shmaltz | doctor_za_ljubav, the IP address you are using right now disagrees with what you said |
| 19:25.05 | jake1932 | caught! |
| 19:25.07 | doctor_za_ljubav | yeah |
| 19:25.41 | doctor_za_ljubav | i am out thru another system |
| 19:25.51 | doctor_za_ljubav | because i am trying to configure this remotly |
| 19:26.10 | shmaltz | jjhall, you could use a macro to do the dialing for you, and have not answring unless you press a button |
| 19:26.22 | *** join/#asterisk riksta (n=rick@62.6.163.90) |
| 19:26.32 | jjhall | shmaltz: Good idea, I may look into that. |
| 19:26.49 | shmaltz | jjhall, check this out: |
| 19:26.50 | shmaltz | http://www.voip-info.org/wiki-asterisk+cmd+dial |
| 19:27.20 | shmaltz | look at the M option |
| 19:27.42 | shmaltz | http://www.breitbart.com/news/2005/10/20/D8DC4N8O0.html |
| 19:28.30 | *** join/#asterisk rob314 (n=rob314@207.58.194.2) |
| 19:29.37 | *** join/#asterisk afrosheen (n=test@txprotoa2.august.net) |
| 19:30.27 | afrosheen | woot finally got my polycoms playing custom ring tones |
| 19:31.09 | jjhall | Funny that article mentions Post Falls, Idaho. I'm in Idaho myself, though about 8 hours away from Post Falls. |
| 19:32.45 | afrosheen | wow who handed out valiums today |
| 19:36.15 | ender | shmaltz: I've experienced that on my index finger from tapping Mc Donald's computer screens too much. I worked at MC D's for 3~ years. I had to start using the backend of a pencil to do my tapping. |
| 19:36.47 | shmaltz | ender, a teller at my local bank had the same thing |
| 19:39.10 | jjhall | Seeing that wiki page for the Dial function reminds me of another question. Using the W/w parameter, I see that the sound files end up in /var/spool/asterisk/monitor as unixtime.wav (basically.) Does it create the two separate audio files for each channel or does it combine them? |
| 19:40.09 | shmaltz | jjhall, I never tried, but I am assuming that it is 2 files |
| 19:40.14 | hugo-v6 | valium? *wantsome* |
| 19:40.23 | shmaltz | you need something like sox to combine them |
| 19:40.58 | jjhall | That is kind of what I was guessing. I'll try it this evening and see what I can do. |
| 19:45.09 | *** join/#asterisk escribzz (n=escribzz@h-64-105-157-50.phndaz91.covad.net) |
| 19:48.02 | JerJer | shmaltz: http://www.breitbart.com/news/2005/10/21/D8DCHJ087.html |
| 19:48.07 | JerJer | that one is funnier :) |
| 19:50.39 | *** join/#asterisk PupenoL (n=pupeno@200.123.183.89) |
| 19:50.40 | shmaltz | JerJer, I remember back around 2 years ago one of these stories made it into the papers |
| 19:50.59 | shmaltz | its not new (as the articles says) |
| 19:51.36 | shmaltz | I live about 60 miles south of the city so I get the New York Post around here, they reported it |
| 19:52.47 | shmaltz | c ya guys |
| 19:54.31 | JerJer | then there's this: http://tinyurl.com/95dnf |
| 19:55.22 | jjhall | I saw that this morning, that is crazy |
| 19:55.53 | *** join/#asterisk fulgas (n=fulgas@a81-84-116-219.cpe.netcabo.pt) |
| 19:56.13 | PupenoL | what free software do you use/recommend to benchmark voip devices ? particulary Asterisk servers and SIP. |
| 19:57.54 | PupenoL | what free software do you use/recommend to benchmark voip devices ? particulary Asterisk servers and SIP. |
| 19:57.57 | PupenoL | sorry! |
| 19:59.32 | *** join/#asterisk ejo1 (n=ejo1@kinetic-host39.dsl.visi.com) |
| 20:00.01 | JerJer | PupenoL: Asterisk |
| 20:01.47 | *** join/#asterisk GXTi (n=omgwtfbb@freenode/developer/GXTi) |
| 20:05.35 | groogs | PupenoL: i've heard sipp but never used it myself |
| 20:05.55 | Inkubot | bye |
| 20:09.02 | *** join/#asterisk tekati (n=captain@cpe-66-75-215-63.bak.res.rr.com) |
| 20:09.12 | *** join/#asterisk docE (n=docE@66.237.242.41.ptr.us.xo.net) |
| 20:09.44 | docE | Say anyone know how I can manually force a code in Asterisk Dialplan? I have a unique situation |
| 20:10.04 | docE | Can I use something like servar(CODEC= or something? |
| 20:11.26 | docE | nevermind I finally found it. was hiding.. :) |
| 20:11.48 | ejo1 | Anyone have ideas for me... I have a new TE410P card installed in a dual XEON, running Asterisk@Home v1.3, Asterisk 1.0.9. When I run ztcfg I get messages like "Putting 1 in register 31 on spam 1" "Putting 255 in register 2f on span 1" "Putting 192 in register 30 on span 1" I have only 10 channels defined, but channel 24 for the 'd' channel. |
| 20:12.02 | *** join/#asterisk wh|tey (n=wh|tey@c-24-8-158-104.hsd1.co.comcast.net) |
| 20:13.50 | *** join/#asterisk rocket (n=rocket@gentoo/developer/rocket) |
| 20:15.30 | n0rf- | still no luck starting my updated *, i get a segfault but don't see any error. even after clearing out my modules dir and recompiling. any ideas would be appreciated |
| 20:15.56 | ejo1 | Source code in file /usr/src/zaptel/wct4xxp.c looks something like if oldnotclear, & if(debug) then print these messages.... I don't know what to check from here. |
| 20:16.14 | *** join/#asterisk cripito (n=ncripito@67.96.197.99) |
| 20:16.29 | cripito | hi again |
| 20:22.57 | *** join/#asterisk kiwnix (n=egarcia@120.red-82-158-158.user.auna.net) |
| 20:26.35 | *** join/#asterisk shamo999 (n=shamo@ool-4578565c.dyn.optonline.net) |
| 20:27.16 | shamo999 | hello yall, this is a newbie: can someone help me? |
| 20:27.35 | blitzrage | shamo999: ask an actual question, and we'll try :) |
| 20:29.46 | *** join/#asterisk jsaunders (i=js@S01060060971c5817.va.shawcable.net) |
| 20:30.10 | shamo999 | I just set up asterisk, but only with soft phones (xlite). In the extensions.conf, when I set exten => s,1,Playback(hello-world) in the context, it does not work. But if I set a specific extension number, like exten => 111,1,Playback(hello-world) it works. Is there a reason? Thanks! |
| 20:30.13 | jsaunders | Is it possible to disable gtk usage when compiling asterisk? |
| 20:30.19 | n0rf- | blitzrage: i could use some help as well :/ |
| 20:30.24 | *** join/#asterisk BladeRunner05 (n=gianni@adsl-201-217.37-151.net24.it) |
| 20:30.40 | n0rf- | i get == Parsing '/etc/asterisk/enum.conf': Found |
| 20:30.40 | n0rf- | Asterisk Ready. |
| 20:30.52 | n0rf- | *CLI> Ouch ... error while writing audio data: : Broken pipe |
| 20:30.52 | n0rf- | Segmentation fault (core dumped) |
| 20:31.15 | n0rf- | but last entry in log is "asterisk ready" as above |
| 20:31.39 | snitt | ;) |
| 20:31.43 | snitt | strace it |
| 20:32.10 | snitt | strace -o asterisk-log -ff -- <your usual asterisk startcommand here> |
| 20:32.12 | CoffeeIV_ | shamo999: it matters what context (the sections named [likethis]) that it's in. I would advise finding that link to the free PDF download of the O'Reilly Asterisk book and reading it -- the chapter on dialplans will help you figure this out |
| 20:32.47 | n0rf- | snitt: so if i usually start it from etc/init.d just that command? |
| 20:32.48 | *** part/#asterisk _Thor (i=CS@user-vc8fl7n.biz.mindspring.com) |
| 20:33.09 | snitt | no. |
| 20:33.22 | snitt | find out what does that init script use |
| 20:33.32 | snitt | probably start-stop-daemon |
| 20:33.43 | shamo999 | CoffeeIV: I read the whole book last night, that is how I set it up. I am actually using the examples from that book. But they are using analog phones for the examples, so I am not sure if that makes a difference. |
| 20:33.47 | snitt | anyway, start it with 'asterisk -U asterisk -G asterisk -vvfc' |
| 20:33.54 | snitt | if you use * with username asterisk |
| 20:34.54 | CoffeeIV_ | shamo999: softphones versus hardphones does not make a difference in this case, I think, but it would be hard to know for sure without seeing all your conf files |
| 20:35.19 | n0rf- | startproc -u [uname] -g [gname] -p $PIDFILE $ASTERISK_BIN <- that's how i've put it in the start scrip |
| 20:35.23 | n0rf- | gonna strace that now |
| 20:37.15 | ejo1 | anyone from digium online now, i've been on hold with dead air for quite some time. Is your system down? |
| 20:37.38 | *** join/#asterisk loud (n=ariel@blaqhat.com) |
| 20:38.37 | shamo999 | Coffee: My conf files are very simple: in sip.conf I have [general] with ulaw=allow and [sfone] type=friend;qualify=yes;secret=xxxx;nat=no;host=dynamic;canreinvite=yes;context=internal |
| 20:38.41 | n0rf- | snitt: where do i get the trace output? |
| 20:38.49 | snitt | n0rf-: at $PWD |
| 20:38.58 | snitt | $PWD/asterisk-log* |
| 20:38.59 | jsaunders | Why does * not have a ./configure scripts? Sheesh. |
| 20:39.25 | jsaunders | err, script. singular |
| 20:40.04 | shamo999 | then, in extensions.conf i have [general]static=yes[global][internal]exten =>s,1,Playback(hello-world) |
| 20:40.49 | shamo999 | it works if I change in context [internal] exten => 111,1,Playback(hello-world) and then dial 111 in the phone. |
| 20:40.54 | n0rf- | snitt: ? |
| 20:40.58 | snitt | n0rf-: ? |
| 20:41.14 | n0rf- | $PWD? |
| 20:41.18 | snitt | yeah |
| 20:41.20 | n0rf- | homedir? |
| 20:41.22 | snitt | environment variable |
| 20:41.24 | snitt | echo $PWD |
| 20:41.29 | snitt | mostly the current dir |
| 20:41.43 | n0rf- | ah ok |
| 20:41.44 | *** join/#asterisk nobell (n=Jared@160.7.249.18) |
| 20:41.50 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 20:41.56 | n0rf- | well, my ssh crashed |
| 20:41.57 | n0rf- | ffs |
| 20:42.01 | snitt | try an 'ls ./asterisk-log*' |
| 20:42.21 | harryvv | thats no good it its a remote Pc with ssh running |
| 20:42.51 | CoffeeIV_ | shamo999: you have to dial something for it to go into asterisk and start looking for an extension/context that matches, I think |
| 20:42.53 | *** join/#asterisk Exomorph (i=Exomorph@216.251.134.2) |
| 20:43.05 | Exomorph | Hey peoples. |
| 20:43.34 | n0rf- | ok. i can get as far as starting * now |
| 20:43.34 | harryvv | how many attended astricon? |
| 20:43.43 | escribzz | anyone want to sell a ser config? lol |
| 20:43.48 | n0rf- | as soon as i try to call anything it crashes |
| 20:44.04 | n0rf- | and it takes the ssh session with it |
| 20:44.08 | shamo999 | coffee: yes I tried to dial various numbers from the softfone but the CLI does not show anything. |
| 20:44.11 | snitt | that's strange |
| 20:44.22 | snitt | my * crashes only when i dial the voicemail |
| 20:44.29 | snitt | the voicemailmain() app |
| 20:44.35 | harryvv | snitt what version |
| 20:44.45 | snitt | 1.0.9 and 1.2_beta1 |
| 20:44.49 | harryvv | k |
| 20:44.53 | n0rf- | nope, i've dialed an outbound context |
| 20:44.57 | BladeRunner05 | How can I recognize a callerid and play a personalized menu ? |
| 20:45.04 | snitt | strange, 1.0.9 worked well before |
| 20:45.11 | harryvv | I am only interested in the most stable version of asterisk if im going to deploy it. |
| 20:45.27 | shamo999 | coffee: do you know if incoming call from sip is treated differently by extensions.conf than from analog channels? |
| 20:45.28 | snitt | harryvv: don't count me. im hacking it all the time |
| 20:45.39 | n0rf- | hm, at least i got some logs this time |
| 20:45.40 | snitt | im not using vanilla * |
| 20:46.38 | n0rf- | i can't derive anything from these logs |
| 20:46.44 | Exomorph | Just wondering... I know Asterisk didn't do T.38 faxing (as of 6 months ago)... Does anyone know if it can now, or if someone is working on it? |
| 20:46.53 | n0rf- | snitt: shall i pate one or several of those somewhereß |
| 20:46.55 | n0rf- | *? |
| 20:47.07 | n0rf- | *paste |
| 20:47.10 | snitt | n0rf-: just look after the of the last one |
| 20:47.16 | snitt | the of = the end of |
| 20:48.09 | n0rf- | i think |
| 20:48.10 | n0rf- | _exit(0) = ? |
| 20:48.15 | n0rf- | won't be of much use? |
| 20:48.46 | *** part/#asterisk geryon2005 (n=Steve@69.212.137.6) |
| 20:49.30 | n0rf- | what i get in the log before is |
| 20:49.30 | n0rf- | nanosleep({1, 0}, 0x4093ee64) = ? ERESTART_RESTARTBLOCK (To be restarted) |
| 20:49.30 | n0rf- | +++ killed by SIGSEGV +++ |
| 20:49.37 | harryvv | so who here has the longest running asterisk with no problems in a production enviroment. |
| 20:49.48 | snitt | ok, paste/mail them |
| 20:52.46 | *** part/#asterisk mkrufky (n=mk@68.160.103.77) |
| 20:52.47 | *** join/#asterisk glm2k (n=GLM@rrcs-24-199-11-41.west.biz.rr.com) |
| 20:53.43 | *** join/#asterisk santiago (n=santiago@208.195.215.158) |
| 20:54.08 | NewSole2 | how to do (Maridian Switch -> TE410P -> Asterisk -> voip) |
| 20:54.25 | Inv_arp | NewSole2: how to do what |
| 20:54.32 | *** join/#asterisk damned (n=vpol@damned.vpol.org.ru) |
| 20:54.49 | NewSole2 | we have a Maridian PBX...... |
| 20:55.05 | NewSole2 | want to get asterisk to emulate PRI for box |
| 20:55.11 | *** join/#asterisk dinkdinkdink (n=d@pool-68-238-147-186.dllstx.fios.verizon.net) |
| 20:55.19 | harryvv | so there is pri in the meridian box |
| 20:55.21 | harryvv | ? |
| 20:55.25 | NewSole2 | yes |
| 20:55.48 | NewSole2 | what setting do I need in asterisk and on Maridian |
| 20:55.52 | harryvv | im looking at also taking a asterisk box and putting it on the backside of a ATT partner PBX |
| 20:56.05 | shamo999 | Can someone help? Basically, I have a sip phone defined in sip.conf, which works when the relevant context in extensions.conf refers to a specific number (example exten => 111,1,Playback(hello-world). But, it does not work if I make it general by instead having the line exten => s,1,Playback(hello-world). THanks! |
| 20:56.42 | file[laptop] | shamo999: s is only used when the extension dialed isn't known, in the case of SIP - it's always known |
| 20:57.30 | shamo999 | Thanks file! |
| 20:57.36 | NewSole2 | can anyone point to direction |
| 20:58.36 | shamo999 | Now, then, how can I get asterisk to answer no matter what extension I call in the sip phone. Is the only way to have exten => _X.,1,Playback(...)? |
| 20:58.37 | *** join/#asterisk Cybertoy (n=Cybertoy@dsl254-123-241.nyc1.dsl.speakeasy.net) |
| 20:58.41 | *** join/#asterisk santiago (n=santiago@208.195.215.158) |
| 20:58.48 | file[laptop] | shamo999: pretty much |
| 20:59.29 | *** join/#asterisk brookshire (n=pfffft@216.215.208.188.nw.nuvox.net) |
| 20:59.56 | Little-L | file[laptop], Hi - look at this: http://qubix.dk/zgw01/kerne.txt - I almost got my card up and running |
| 20:59.56 | NewSole2 | harryvv... any ideas |
| 21:00.07 | *** join/#asterisk Jabroni (n=Jabroni@red-corp-201.143.46.226.telnor.net) |
| 21:00.12 | harryvv | nope |
| 21:00.13 | harryvv | :) |
| 21:00.29 | bsdfreak | A WITH A TEETH A |
| 21:00.31 | bsdfreak | A WITH A TEETH A |
| 21:00.36 | *** part/#asterisk wh|tey (n=wh|tey@c-24-8-158-104.hsd1.co.comcast.net) |
| 21:00.56 | shamo999 | Now, I am trying to configure incoming IAX calls from the internet. Freeworlddialup is continuously saying "all circuits busy". Are there other services that give free IAX connections? |
| 21:00.59 | Little-L | bsdfreak, ? |
| 21:02.08 | Little-L | bsdfreak, since you are calling you self a bsdfreak.... Have you tried setting up a TE410P on FBSD 5.4? |
| 21:02.10 | Jabroni | bummer... ive been checking docs for almost a week, and I still cant find how asterisk detects outgoing anwsered calls :( |
| 21:02.20 | bsdfreak | heh |
| 21:02.26 | bsdfreak | negatory |
| 21:02.27 | Little-L | :) |
| 21:02.34 | Little-L | Bummer .... |
| 21:02.36 | bsdfreak | TE410P is condition red |
| 21:02.37 | bsdfreak | over. |
| 21:02.38 | glm2k | shamo999: hmmm, looks like FWD is busy |
| 21:03.07 | Little-L | bsdfreak, http://qubix.dk/zgw01/kerne.txt - it almost happened |
| 21:03.13 | *** join/#asterisk vtsherwood (n=rushowr@cpe-24-26-150-142.columbus.res.rr.com) |
| 21:03.18 | vtsherwood | hello again all! |
| 21:03.18 | bsdfreak | d'oh |
| 21:03.20 | lesouvage | shamo999: I have fwd up and running soit can work |
| 21:03.26 | Cybertoy | anyone good with enum lookup? |
| 21:03.26 | *** join/#asterisk JonR800 (i=jon@p1mp.org) |
| 21:03.35 | vtsherwood | any -devs on here? |
| 21:03.50 | Cybertoy | I'm trying to dial a e164.org number but get naptr regex match failed errors in asterisk |
| 21:04.13 | Cybertoy | but I tried multiple numbers listed in e164.org so I don't think it's a problem in the dns lookup |
| 21:05.12 | shamo999 | lesouvage: it does not work for me mostly. Only a few connections through to 612 in the last hour or so. And no call me back. |
| 21:05.56 | glm2k | shamo999: lesouvage: it is busy at the moment. must be the weekend. |
| 21:06.18 | shamo999 | thanks glm2k. Is there other similar servers? |
| 21:06.40 | glm2k | shamo999: free DIDs? |
| 21:06.43 | glm2k | gimme a sec |
| 21:06.50 | *** join/#asterisk RaYmAn-Bx (i=rayman@x1-6-00-40-63-da-39-3f.k191.webspeed.dk) |
| 21:07.04 | shamo999 | OK |
| 21:07.50 | glm2k | shamo999: http://www.0xdecafbad.com/Free-VoIP-Providers.html |
| 21:08.10 | shamo999 | thanks! |
| 21:11.05 | pooh_ | anybody know why vmail.cgi does not display existing vmail messages |
| 21:11.25 | escribzz | permissions probably |
| 21:11.41 | pooh_ | tried that, both root and www(apache) |
| 21:11.57 | escribzz | the upper dirrectories too |
| 21:12.06 | escribzz | try chmod 777 everything |
| 21:12.07 | *** join/#asterisk hohum (i=corbe@snoop.burghcom.com) |
| 21:12.08 | pooh_ | hmmm let me check |
| 21:12.13 | escribzz | each parent directory too |
| 21:13.31 | pooh_ | escribzz: thx |
| 21:13.52 | escribzz | :) |
| 21:14.47 | *** join/#asterisk Astinus (i=iBook@freenode/staff/gentoo.astinus) |
| 21:16.08 | Inv_arp | dont get this, sometimes my * box just gets unregistered from sip provider , sip reload fixes it |
| 21:16.15 | pooh_ | escribzz: chown to apache was sufficient |
| 21:16.31 | escribzz | great |
| 21:16.52 | escribzz | the problem is new messages |
| 21:16.57 | escribzz | I believe there is a patch |
| 21:17.19 | *** part/#asterisk santiago (n=santiago@208.195.215.158) |
| 21:17.27 | pooh_ | escibzz, let me try that |
| 21:18.28 | pooh_ | escribzz: right, the new vm will list but can not be played |
| 21:19.05 | malverian[work] | Why would I be getting a "403 Use Proxy" from a snom phone? |
| 21:19.07 | malverian[work] | Any ideas? |
| 21:20.26 | escribzz | you can always change the perms with that call external program command or run the patch |
| 21:21.46 | pooh_ | escribzz: right, meaning system command? Is the patch in the bugtracker? |
| 21:24.36 | *** join/#asterisk brookshire (n=pfffft@216.215.208.188.nw.nuvox.net) |
| 21:25.15 | *** join/#asterisk darwin_35 (n=darwin35@208.139.193.178) |
| 21:25.24 | darwin_35 | Stella |
| 21:25.31 | darwin_35 | you bitch I'm home |
| 21:25.50 | watchy | lets fuck |
| 21:25.55 | JerJer | Wilma |
| 21:26.10 | darwin_35 | get down on all 4's and give me the rear baby |
| 21:26.21 | watchy | haha |
| 21:26.38 | *** join/#asterisk glm2k (n=GLM@rrcs-24-199-11-41.west.biz.rr.com) |
| 21:26.43 | darwin_35 | and wheres barney I want him here with me |
| 21:26.59 | darwin_35 | h's my special buddy |
| 21:27.53 | darwin_35 | alone |
| 21:28.04 | lesouvage | malverian[work] : Is it registered properly? sip show channels on the cli should show the snom phone. |
| 21:29.34 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 21:30.44 | *** join/#asterisk PupenoL (n=pupeno@200.123.183.89) |
| 21:38.52 | malverian[work] | lesouvage, Yeah, it's registered. |
| 21:39.07 | malverian[work] | lesouvage, I turned off "filter registrar" and it works now... |
| 21:40.30 | lesouvage | malverian[work]: OK, I have two snom phones running and they are working great. |
| 21:40.45 | *** join/#asterisk CANO-1982 (i=alejandr@201.255.51.193) |
| 21:41.18 | *** part/#asterisk CANO-1982 (i=alejandr@201.255.51.193) |
| 21:41.27 | *** join/#asterisk cripito (n=ncripito@67.96.197.99) |
| 21:43.31 | *** join/#asterisk robbie2 (n=rob@CPE-60-231-143-195.qld.bigpond.net.au) |
| 21:43.39 | robbie2 | helllooooooooooo |
| 21:43.52 | Aughey | Does anyone have a voip setup that supports paging/intercoms well? |
| 21:43.54 | darwin_35 | helllloooooooooooooooooooooooooooooooooooooooooo |
| 21:44.21 | robbie2 | anyone have the time and patience to step me through getting x-lite to connect to a xorcom rapid box i just set up ? |
| 21:45.07 | robbie2 | hi Darwin35 |
| 21:45.33 | Aughey | robbie2: try reading http://www.automated.it/asterisk/lah-3-6-05_1.html |
| 21:45.42 | robbie2 | ok |
| 21:46.09 | Aughey | you need to set type=friend in the sip.conf file (that's not stated in the doc) |
| 21:46.20 | darwin_35 | hey |
| 21:47.43 | robbie2 | Aughey: in the blind setup? |
| 21:47.51 | robbie2 | for xlite ? |
| 21:48.38 | Aughey | in the /etc/asterisk/sip.conf file |
| 21:48.44 | robbie2 | ta |
| 21:52.01 | darwin_35 | no its /usr/local/etc/asterisk |
| 21:52.24 | darwin_35 | or is it usr/local/share/examples/asterisk |
| 21:52.47 | darwin_35 | or is it asteriak/local/usr/ |
| 21:53.25 | cripito | hi.. anyone from colorado? |
| 21:53.26 | malverian[work] | Aughey, I plugged ours into our overhead paging system :) |
| 21:53.34 | darwin_35 | some where in one of those |
| 21:55.38 | darwin_35 | paging is a pain |
| 22:00.30 | *** join/#asterisk tekati (n=captain@cpe-66-75-215-63.bak.res.rr.com) |
| 22:02.28 | *** join/#asterisk Corndawg_ (i=whoisit@71.196.61.13) |
| 22:05.08 | *** join/#asterisk darenw (n=daren@rrcs-67-79-20-162.sw.biz.rr.com) |
| 22:05.52 | darenw | Any one know how to fix this error.. chan_oss.c:257 sound_thread: Read error on sound device: Resource temporarily unavailable |
| 22:06.02 | darenw | can't seem to find anything on google for it |
| 22:08.28 | *** part/#asterisk Cybertoy (n=Cybertoy@dsl254-123-241.nyc1.dsl.speakeasy.net) |
| 22:09.23 | kuku5 | How the heck to do i setup ulaw on a gxp 2000 - its no in the settings of the phone! |
| 22:10.13 | Nugget | kuku fought the ulaw and the ulaw won. |
| 22:16.12 | robbie2 | Aughey: thanks, get x-lite conencted however i get "Call not approved" when i dial the echo test |
| 22:16.46 | *** part/#asterisk darwin_35 (n=darwin35@208.139.193.178) |
| 22:16.49 | *** join/#asterisk Uberbot (n=Uberbot@69.252.219.76) |
| 22:16.55 | Uberbot | Hi all. |
| 22:17.20 | Uberbot | Anyone tell me what this means when I start asterisk: Ouch ... error while writing audio data: : Broken pipe |
| 22:17.47 | *** join/#asterisk JASON-0 (n=jason@jason.unitz.ca) |
| 22:18.12 | JASON-0 | Hello, I've install zaptel and when I do a modprobe I get the following.. |
| 22:18.30 | *** join/#asterisk Los415 (n=los415@64.201.104.186) |
| 22:18.41 | JASON-0 | lib/modules/2.4.21-32.ELsmp/misc/zaptel.o: couldn't find the kernel version the module was compiled for |
| 22:18.42 | JASON-0 | lib/modules/2.4.21-32.ELsmp/misc/zaptel.o: insmod /lib/modules/2.4.21-32.ELsmp/misc/zaptel.o failed |
| 22:18.42 | JASON-0 | lib/modules/2.4.21-32.ELsmp/misc/zaptel.o: insmod wanpipe failed |
| 22:18.53 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 22:19.10 | JASON-0 | anyone have idea what this means ? |
| 22:19.42 | FuriousGeorge | i gotta get an el cheapo softphone |
| 22:20.00 | FuriousGeorge | i mean "el cheapo" ipphone |
| 22:20.24 | *** join/#asterisk lunk (n=lunk@negative-influence.com) |
| 22:20.25 | FuriousGeorge | isnt there one that everyone just luvs to use with asterisk. i believe it starts with an a |
| 22:20.28 | CoffeeIV_ | JASON-O: it means you are trying to load zaptel.o into a different kernel from the one it was compiled for |
| 22:20.41 | JASON-0 | but I just compiled it |
| 22:20.57 | JASON-0 | i even deleted the zaptel.o |
| 22:21.00 | JASON-0 | recompiled |
| 22:21.09 | JASON-0 | and same thing |
| 22:21.14 | lunk | is there a way to use a javabean on an application server somewhere as a remote AGI script? |
| 22:21.33 | drumkilla | look at FastAGI |
| 22:21.49 | lunk | i'm using it, but it still requires that the AGI server be running |
| 22:22.02 | lunk | as an application independent of the application server |
| 22:22.42 | FuriousGeorge | anyone wanna recommend an economic ipphone? |
| 22:22.45 | GXTi | fastagi? |
| 22:22.45 | GXTi | heh. |
| 22:22.54 | GXTi | im guessing its analogous to fastcgi |
| 22:23.00 | lunk | let me double check |
| 22:29.33 | morale | how do you generate patches in linux with diff properly? diff [src] [dst]? |
| 22:29.52 | harryvv | FuriousGeorge one pbx i may be integrating my asterisk into is a ATT partner pbx. Ever hear of those? |
| 22:30.26 | FuriousGeorge | harryvv: i know nothing of proprietary pbx. call att ask if it can understand fxs signalling for incoming lines, or something |
| 22:30.40 | FuriousGeorge | i mean for analog lines |
| 22:31.07 | FuriousGeorge | i think im gonna go with the sipura 841 unless anyone gives me a better ida |
| 22:31.09 | FuriousGeorge | idea |
| 22:31.37 | lunk | i think I was approaching the problem wrong |
| 22:31.45 | harryvv | FuriousGeorge there is actually a manual left behind. its a lucent/avaya pbx with att logo on it. |
| 22:32.00 | lunk | instead of just creating an action and sending it to the dial plan, i should write the dialplan in the manager api \o/ |
| 22:32.17 | lunk | duh |
| 22:32.46 | *** part/#asterisk just4fun (n=edwin@209.151.137.139) |
| 22:33.06 | harryvv | FuriousGeorge this is one such system on ebay |
| 22:33.08 | harryvv | http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=5818358106&indexURL=0&photoDisplayType=2#ebayphotohosting |
| 22:36.15 | *** join/#asterisk meowwuff (i=mreith@snoid.gfed.net) |
| 22:36.20 | *** part/#asterisk bob_too (n=chris@rrcs-24-153-179-246.sw.biz.rr.com) |
| 22:38.20 | cripito | george the 841 works pretty well for me |
| 22:39.27 | *** join/#asterisk alephcom (n=Miranda@207.34.97.130) |
| 22:40.42 | kuku5 | How the heck to do i setup ulaw on a gxp 2000 - its not in the settings of the phone! |
| 22:40.55 | harryvv | so if i can integrate the existing ATT lucent/avaya pbx into a asterisk box then no need for new phones. |
| 22:41.21 | alephcom | Is it referred to by a different name, perhaps g711? |
| 22:41.30 | mogorman | hi |
| 22:43.13 | kuku5 | nope |
| 22:43.23 | *** join/#asterisk _tekati_ (n=captain@cpe-66-75-215-63.bak.res.rr.com) |
| 22:43.31 | malverian[work] | Why would Asterisk return a 503 server error on subscribe? |
| 22:43.34 | kuku5 | 723 |
| 22:43.36 | kuku5 | 729 |
| 22:43.51 | kuku5 | gsm |
| 22:43.53 | kuku5 | pcmu |
| 22:43.59 | kuku5 | no 711 |
| 22:44.02 | cripito | pcmu |
| 22:44.21 | kuku5 | what about pcmu |
| 22:44.22 | *** join/#asterisk kippi (n=kippi@cpc4-hatf3-6-0-cust33.lutn.cable.ntl.com) |
| 22:44.24 | kippi | hey |
| 22:44.35 | cripito | try that one like ulaw |
| 22:44.42 | kuku5 | ok |
| 22:44.50 | kippi | what softphone support IXA2? |
| 22:46.06 | *** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 22:47.19 | JerJer | kippi: Asterisk |
| 22:48.17 | cripito | :D |
| 22:48.35 | kippi | at the mo i am using X-lite but sip is causing problems with NAT so I belive away round this is IXA? |
| 22:48.42 | cripito | i don't think that asterisk fit well in the description of a softphone |
| 22:48.51 | kuku5 | so pcmu is 711? |
| 22:48.56 | JerJer | why not? |
| 22:49.11 | cripito | kippi firefly 3thparty or see in the wiki there is a BIG list |
| 22:49.43 | kippi | ah ok, how about handsets? |
| 22:50.00 | JerJer | TDM400P |
| 22:50.01 | kuku5 | How woudl I setup a backup provider ( i'm already registered to 2 ) in extensions.conf |
| 22:50.20 | JerJer | by looking at the ${DIALSTATUS} value |
| 22:55.45 | kippi | wot port does ixa2 use? |
| 22:56.27 | meowwuff | What is dundi? |
| 22:56.29 | ender | kippi: udp:4569 |
| 22:59.54 | *** part/#asterisk Uberbot (n=Uberbot@69.252.219.76) |
| 23:00.06 | *** join/#asterisk dlucas (n=root@Broadband-Dynamic-Western84.connect.com.fj) |
| 23:01.42 | meowwuff | are newer asterisk versions backwards compat with older/ |
| 23:02.09 | kippi | anyone know if avaya 4620 handsets work with IXA2? |
| 23:03.52 | ender | meowwuff: I've iax2 connected the current w/ the beta. |
| 23:03.52 | dlucas | malverian ... how goes the diagnostic? |
| 23:04.04 | ender | 1.0 w/ 1.2 |
| 23:05.53 | cripito | anyone from colorado? |
| 23:09.40 | FuriousGeorge | one of my servers isnt registering with the other ones and i cant seem to figure out why. other servers can log into it. its register commands are right. |
| 23:09.46 | FuriousGeorge | its gotta be something stupid |
| 23:10.39 | FuriousGeorge | keeps timing out |
| 23:11.05 | kippi | can you get asterisk to work in a culster? |
| 23:11.10 | kippi | cluster? |
| 23:11.22 | FuriousGeorge | i heard you could |
| 23:11.49 | FuriousGeorge | but like anything else with a cluster its supposed to be complicated to do right |
| 23:12.04 | FuriousGeorge | but what do i know, ive never done it :) |
| 23:12.29 | kippi | or have it so that if one server dies the other server takes all the calls |
| 23:12.41 | hardwire | anybody using jb + ulaw? |
| 23:13.02 | FuriousGeorge | kippi: keep a hot spare, and swapem |
| 23:13.23 | kippi | hehe |
| 23:13.44 | FuriousGeorge | seriously |
| 23:13.57 | FuriousGeorge | its a lot less complicated than running a cluster for redundancy |
| 23:14.12 | GXTi | how are cisco SIP phones configured? |
| 23:14.19 | FuriousGeorge | i guess if you need a cluster for the computation thats one thing, but how many extensions can you need? |
| 23:14.25 | GXTi | do they have the IP of the SIP router programmed into them or do they use DHCP? |
| 23:15.09 | FuriousGeorge | GXTi: ask voip-info.org, it knows about all those configurations |
| 23:15.16 | kippi | how many extensions will asterisk take at one time? |
| 23:15.30 | GXTi | school recently replaced its old PBX with cisco stuff |
| 23:15.37 | GXTi | wondering how i can screw around with it :p |
| 23:16.10 | FuriousGeorge | what specs are your server, maybe someone in here can tell you. depending on codecs, and what kind of client, i would assume you could do 50 on a modern computer safely |
| 23:16.17 | FuriousGeorge | modern *server |
| 23:16.47 | cripito | i have a 330 in a dual xeon at 2.80 Mhz and 2Gb the ram |
| 23:16.58 | cripito | without TDM cards |
| 23:17.15 | FuriousGeorge | GXTi: i know some ip phones use tftp, but im waiting for my first one to arrive and its a sipura |
| 23:17.24 | FuriousGeorge | cripito: whats a 330? |
| 23:17.39 | cripito | 330 extensions |
| 23:17.46 | JerJer | two 3's and a zero |
| 23:17.53 | cripito | with usually 200 - 250 calls concurrent calls |
| 23:17.54 | kippi | server is: 2.80 GHZ, 1GB ram, 300 harddrive (RAID) 1x TE100P card |
| 23:18.02 | FuriousGeorge | ive never done more than 10 extensions in production :) |
| 23:18.16 | FuriousGeorge | what kind of clients |
| 23:18.21 | cripito | call center |
| 23:18.27 | FuriousGeorge | sip phones? |
| 23:18.40 | cripito | gst 101 and polycom for the managers |
| 23:18.43 | GXTi | aha, TFTP |
| 23:19.07 | GXTi | FuriousGeorge: im thinking about bringing a laptop with a softphone and seeing if i can call school phones :) |
| 23:19.30 | kippi | so I should be able to get 200+ concurrent sip calls? with out any quailty problems? |
| 23:19.34 | jpm_SD | GXTi do they use Cisco Call Mangler...er Manager? |
| 23:19.42 | X-Rob | GXTi - unlikely, unless they're using SIP. |
| 23:19.46 | GXTi | jpm_SD: probably |
| 23:19.48 | cripito | no quality issues even using ulaw |
| 23:20.09 | FuriousGeorge | GXTi: all depends on their server, i guess |
| 23:20.11 | cripito | 3 dif servers additionally for digium cards |
| 23:20.12 | jpm_SD | yeah.. then they are most likely using skinny... |
| 23:20.17 | GXTi | right, skinny |
| 23:20.22 | X-Rob | SCCP |
| 23:20.23 | GXTi | i bet i can get _something_ to work |
| 23:20.34 | X-Rob | I bet you can get your laptop to boot. |
| 23:20.40 | GXTi | yay! |
| 23:20.42 | GXTi | :p |
| 23:20.44 | jpm_SD | haha |
| 23:20.58 | X-Rob | GXTi - I suggest you RTFM on SCCP and SIP and voip and stuff. |
| 23:20.59 | FuriousGeorge | GXTi: search the model of your phone on voip-info |
| 23:21.07 | GXTi | FuriousGeorge: ill have to look on monday what model it is |
| 23:21.17 | JerJer | yeah and stuff |
| 23:21.22 | GXTi | its grey and black and has a large red LED in the top of the handset, but i dont know the exact model no |
| 23:21.27 | cripito | :) |
| 23:21.41 | GXTi | guess google images could help |
| 23:21.51 | X-Rob | sounds like a 7920 |
| 23:21.53 | cripito | jerjer is talking a lot today :D |
| 23:21.57 | GXTi | ... first try. |
| 23:22.19 | jpm_SD | The Cisco phones are sexy... but I' |
| 23:22.25 | jpm_SD | ve heard people don't like em |
| 23:22.38 | GXTi | well they're loud as hell |
| 23:22.52 | GXTi | its definately not a 7920 |
| 23:23.01 | GXTi | 7960 i think' |
| 23:23.11 | jpm_SD | loud is good... if you're at a metal concert. |
| 23:23.21 | GXTi | jpm_SD: but it's a high school :) |
| 23:23.27 | X-Rob | uh |
| 23:23.30 | *** join/#asterisk folsson (n=filip@lund-meje-sr0-vl101-249.perspektivbredband.net) |
| 23:23.38 | X-Rob | brainfart. |
| 23:23.43 | X-Rob | been thinking too much about wireless |
| 23:23.46 | GXTi | yeah, 7920 is a wireless deal |
| 23:24.15 | GXTi | okay, so i need to get some kind of skinny/sccp-enabled softphone |
| 23:24.18 | jpm_SD | Speaking of wifi ... I got one the UTstarcom F1000 phones... Simple... but it works well.. |
| 23:24.19 | GXTi | or maybe use asterisk to relay it |
| 23:24.24 | cripito | anyone have make anything with the prov for the 841? |
| 23:24.54 | cripito | specially with the xml setup? |
| 23:25.11 | JerJer | XML is patented |
| 23:25.20 | cripito | xml |
| 23:25.32 | cripito | mmm jerjer is looking like a bot today |
| 23:25.34 | cripito | ok. |
| 23:25.41 | X-Rob | GXTi - http://www.voip-info.org/tiki-index.php?page=VOIP+Phones |
| 23:26.47 | cripito | mmm voipsupply have a ip500 polycom in ebay... |
| 23:27.18 | *** join/#asterisk opus__ (n=opus@dahphish.org) |
| 23:27.22 | opus__ | anyone here use res_js? |
| 23:28.11 | GXTi | X-Rob: i was already reading the 79xx page |
| 23:28.15 | GXTi | X-Rob: (convenient) |
| 23:29.43 | meowwuff | anyone here have a patch to get gastman to compile on OpenBSD? |
| 23:29.51 | *** join/#asterisk Vco (n=Vco@S01060050da6df072.sc.shawcable.net) |
| 23:31.01 | X-Rob | yeah, but it involves installing CentOS. |
| 23:37.59 | meowwuff | heh |
| 23:38.08 | meowwuff | i need a home computer.. |
| 23:38.29 | Vco | PC? yea those are cool |
| 23:38.37 | Vco | one day lots ofpeople will have then |
| 23:38.38 | Vco | them |
| 23:39.09 | Vco | same all kinds of time doing math and stuff |
| 23:42.46 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-61-165.cybersurf.com) |
| 23:44.38 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 23:47.23 | *** join/#asterisk rayvd (i=rayvd@arthur.bludgeon.org) |
| 23:51.51 | *** join/#asterisk BigJoe1 (n=BigJoe1@dsl-20-177.cofs.net) |
| 23:53.17 | BigJoe1 | hello folks newbie here |
| 23:53.31 | BigJoe1 | so what is asterisk suppose to do or be for? |
| 23:54.11 | Juggie | for getting your best friends hot sister in bed |
| 23:54.15 | morale | it is a complete phone system |
| 23:54.38 | BigJoe1 | Juggie, now that is great news |
| 23:55.32 | BigJoe1 | not much traffic here tonight? |
| 23:55.47 | *** join/#asterisk Tili (n=Tili@211.147.234.5) |
| 23:56.01 | morale | im trying to get dundi working between two asterisk machines |
| 23:56.06 | BigJoe1 | Has anyone used Asterisk@Home? is it a good thing to try? |
| 23:56.43 | X-Rob | BigJoe1- asterisk@home support is on #amportal |
| 23:57.15 | *** join/#asterisk deezed (i=none@adsl-065-006-189-182.sip.bct.bellsouth.net) |
