| 00:01.57 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 00:08.20 | *** join/#asterisk Money5ack (i=moneysac@wer.will.spontanficken.de) |
| 00:14.32 | Egonis | Asterisk won't start on my server anymore, it complains about chan_sip.so and ast_osp_terminate -- anyone have any ideas? |
| 00:16.11 | *** join/#asterisk foRza (n=tMs@194.19.7.195) |
| 00:16.38 | znoG | Egonis: did you recompile asterisk? |
| 00:26.59 | Egonis | znoG: Yes, found out the problem.. osputils was incorrectly installed |
| 00:27.27 | Egonis | When I try to Dial(Console/1) where CONSOLE=Console/alsa, I get the message Unable to create channel of type 'Console' (cause 66 - Channel not implemented) |
| 00:30.18 | SarahEmm | i think it's Console/dsp not Console/1, no? |
| 00:30.45 | *** join/#asterisk sapo_original (n=Fenix@201-25-180-168.ctame700.dsl.brasiltelecom.net.br) |
| 00:30.58 | sapo_original | hi guys... |
| 00:31.06 | MiniDevil | hello |
| 00:31.13 | SarahEmm | hi |
| 00:31.31 | sapo_original | anybody can help me? i looking for a auto provisioning solution for linksys pap2.... |
| 00:31.44 | Egonis | SarahEmm: for alsa, no... for oss it's Console/dsp, but when I use OSS I get chan_oss.c:585 setformat: Unable to re-open DSP device /dev/dsp: No such device |
| 00:31.46 | sapo_original | i searching.. but, i dont find anything... |
| 00:32.47 | SarahEmm | ooh right |
| 00:33.03 | Egonis | SarahEmm: Any suggestions on getting OSS to co-operate? I can use noxmms with no issue |
| 00:33.06 | SarahEmm | did you try ALSA/default ? |
| 00:33.10 | SarahEmm | they kept changing it... |
| 00:33.14 | Egonis | SarahEmm: I don't understand |
| 00:33.21 | SarahEmm | instead of Console/dsp |
| 00:33.23 | SarahEmm | try ALSA/default |
| 00:33.32 | Egonis | SarahEmm: i.e. Dial(ALSA/default)? |
| 00:34.10 | SarahEmm | yeah try that |
| 00:34.26 | *** join/#asterisk BZBW (i=BZBW@ip67-153-142-109.z142-153-67.customer.algx.net) |
| 00:34.33 | Egonis | app_dial.c:1059 dial_exec_full: Unable to create channel of type 'ALSA' (cause 66 - Channel not implemented) |
| 00:34.41 | SarahEmm | err |
| 00:34.50 | SarahEmm | from the * console, try using the 'dial' command |
| 00:34.56 | Egonis | how so? |
| 00:35.03 | SarahEmm | type 'dial #' where # is a phone number |
| 00:35.04 | Egonis | dial ALSA/default literal? |
| 00:35.05 | SarahEmm | from the console itsefl |
| 00:35.17 | Egonis | no such command 'dial' |
| 00:35.26 | SarahEmm | okay, you don't have ALSA set up i think. |
| 00:35.29 | Egonis | I am trying to use this as a PA System |
| 00:35.33 | SarahEmm | do you have an ALSA sound card driver loaded before you started * ? |
| 00:35.44 | Egonis | yes, I can even use noxmms, mpg123, etc. |
| 00:36.20 | SarahEmm | type 'show modules like alsa' |
| 00:36.22 | SarahEmm | anything? |
| 00:37.12 | Egonis | yes, it is shown as per my loaded modules |
| 00:37.14 | Egonis | chan_alsa |
| 00:37.49 | sapo_original | SarahEmm you know any software for auto provisioning? |
| 00:38.02 | SarahEmm | Egonis: you're sure xmms ,etc, are using ALSA not OSS? |
| 00:38.05 | SarahEmm | sapo_original: nope |
| 00:38.19 | Egonis | SarahEmm: not sure, but neither driver works in asterisk\ |
| 00:38.28 | *** join/#asterisk queso (n=josh@nr13-216-68-211-77.fuse.net) |
| 00:38.30 | Egonis | SarahEmm: OSS complains about '/dev/dsp' cannot be re-opened |
| 00:38.58 | SarahEmm | err |
| 00:39.03 | SarahEmm | try cat /dev/dsp from a commandline |
| 00:39.07 | queso | can anyone recommend a good tollfree number provider? |
| 00:40.51 | Egonis | i'm going to try OSS built into my kernel |
| 00:40.54 | Egonis | instead of using alsa-tools |
| 00:41.42 | SarahEmm | well |
| 00:41.49 | SarahEmm | the first step is to figure out which your system is using, OSS or ALSA |
| 00:41.53 | SarahEmm | then you can know which * module to use |
| 00:41.55 | sapo_original | any?? please... anybody know the auto provisioning solution?? |
| 00:42.11 | SarahEmm | sapo_original: if anyone knew, we'd have answered.. |
| 00:43.55 | *** join/#asterisk anthm (n=anthm@CPE-69-76-83-52.wi.res.rr.com) |
| 00:43.55 | *** mode/#asterisk [+o anthm] by ChanServ |
| 00:44.37 | sapo_original | SarahEmm |
| 00:44.38 | sapo_original | perhaps nor all have seen the question |
| 00:44.38 | type0 | I have a dream.. |
| 00:44.40 | sapo_original | hehehe |
| 00:44.50 | type0 | I dream that my dvd burner would burn over 1x and a few gigs didnt take an hour |
| 00:45.21 | type0 | sapo |
| 00:45.24 | type0 | what's your question here? |
| 00:45.33 | type0 | you want an autoprovisioning solution ? |
| 00:46.15 | *** part/#asterisk Egonis (n=Egonis@207.245.14.10) |
| 00:46.26 | sapo_original | type0 yes |
| 00:46.33 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 00:46.36 | generalhan | whats up all |
| 00:46.38 | sapo_original | you know any? |
| 00:46.43 | type0 | how big of an organization? |
| 00:47.17 | sapo_original | small... but i need a solution for configure the remote customers atas... |
| 00:47.33 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 00:47.40 | type0 | I dont think I've ever heard of something like that for a small business |
| 00:47.42 | generalhan | the company across the way asked me about min. requirements to run an * server ... they dont have a rack or anything like that ... what kind of specs of a tower computer would be good for a 10 person org. doing about 50 calls each a day ? |
| 00:49.10 | *** join/#asterisk Netgeeks (n=chris@68-185-24-8.static.mdfd.or.charter.com) |
| 00:49.17 | queso | anyone have a suggestion for an inbound toll-free provider? |
| 00:49.29 | queso | not finding much on voip-info.org |
| 00:52.14 | anthm | try asterlink.com |
| 00:52.19 | Netgeeks | hrm, can a macro be called from inside a macro? |
| 00:52.33 | sapo_original | type0 you know any? |
| 00:52.47 | type0 | nah.. nothing off the top of my head.. why dont you just write one yourself? |
| 00:53.38 | sapo_original | i think this... |
| 00:53.57 | sapo_original | but, i need one example.... |
| 00:55.06 | *** join/#asterisk lucasjb (n=lucas@mail.stabat.com) |
| 00:55.09 | *** join/#asterisk Skarmeth (n=Skarmeth@201008198019.user.veloxzone.com.br) |
| 00:55.37 | *** join/#asterisk Beighto (n=Kry5ta1@adsl-70-133-78-95.dsl.scrm01.sbcglobal.net) |
| 00:56.45 | Beighto | Anybody know how to make a random number in a dialplan? You used to be able to use RAND(min,max) but not with the newer version of asterisk. |
| 00:57.15 | lucasjb | Hiyas, running an Asterisk system with IAX2 soft-clients in a call centre environment receiving calls from the outside world. On some calls, the voice signal will drop out completely, then slowly fade back in over the course of around 15 seconds. Can anyone suggest where I might start looking to troubleshoot this problem? |
| 00:59.09 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 01:02.26 | MiniDevil | Does Asterisk have a web interface? |
| 01:02.49 | SarahEmm | MiniDevil: not built in. there are third party ones you can 'plug on' |
| 01:03.16 | MiniDevil | Hopefully they dont sacrifice security though. |
| 01:03.29 | SarahEmm | i'm not sure |
| 01:03.32 | SarahEmm | i not use |
| 01:03.36 | MiniDevil | ah |
| 01:03.37 | SarahEmm | i don't use them even |
| 01:03.47 | *** join/#asterisk ast_freak|Laptop (n=jesse@12.104.247.2) |
| 01:04.49 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 01:05.34 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 01:06.03 | *** join/#asterisk lpb (n=root@63.81.26.126) |
| 01:06.20 | *** join/#asterisk Telamon (i=telamon@blk-222-22-126.eastlink.ca) |
| 01:06.28 | *** part/#asterisk Telamon (i=telamon@blk-222-22-126.eastlink.ca) |
| 01:07.07 | Docelm0 | MiniDevil, there are no good web interfaces except for the one you code for your own purposed |
| 01:07.09 | Docelm0 | purposes |
| 01:11.01 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-42-234.losaca.adelphia.net) |
| 01:12.07 | MiniDevil | Docelm0: Yeah, all I can find is Asterisk Management Portal |
| 01:12.29 | *** join/#asterisk websae (n=websae@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net) |
| 01:12.34 | queso | I think that changed it to FreePbx now |
| 01:13.16 | websae | anyone familia r with polycom 301? |
| 01:13.43 | SarahEmm | i have one sae |
| 01:13.51 | websae | i am having issues getting it to pull a IP from my router...any suggestions? |
| 01:14.06 | SarahEmm | hrm. other devices DHCP fine? |
| 01:14.12 | websae | iyeah |
| 01:14.19 | SarahEmm | are you getting error? or just not finish? |
| 01:14.21 | websae | i have 3 - 301s |
| 01:14.26 | *** part/#asterisk FuriousGeorge (n=FuriousG@ool-43536ea8.dyn.optonline.net) |
| 01:14.31 | websae | and none of them get a dhcp address |
| 01:14.45 | websae | says "failed to get boot parameters via DHCP" |
| 01:15.11 | SarahEmm | oh |
| 01:15.23 | SarahEmm | what dhcp daemon are you using? |
| 01:15.32 | websae | a netgear router |
| 01:15.50 | websae | my 501 and 601 pull a dhcp address just fine |
| 01:15.57 | websae | is there a way to do a factory reset? |
| 01:16.26 | SarahEmm | hrm |
| 01:17.26 | SarahEmm | i'd sniff traffic in the middle if possible |
| 01:17.30 | SarahEmm | see what's going where... |
| 01:17.33 | SarahEmm | i'm not sure offhand other than that |
| 01:18.25 | websae | anyone else here have a polycom 301? |
| 01:19.04 | *** join/#asterisk Jabroni (n=Hercules@red-corp-200.76.249.142.telnor.net) |
| 01:19.12 | sapo_original | guys, i still serching a provisioning solution... if anybody know's please, pvt-me |
| 01:20.38 | Jabroni | m8s, was wondering if anyone has had problems with Asterisk apps not producing audio, for example VoiceMail() wont say anything, still it does it works (by checking the CLI i can see my tones are being reconized and processed) |
| 01:20.51 | websae | i did factory reset...just says waiting for network to initalize on the phone (polycom 301) |
| 01:21.18 | SarahEmm | Jabroni: if you 'set verbose 99' on the console then do it, any errors or issues or anything? |
| 01:21.21 | SarahEmm | also, what version/svn/what? |
| 01:21.50 | Jabroni | 1.2.9.1 |
| 01:22.00 | Jabroni | let me check with the verbose 99 |
| 01:23.46 | Jabroni | no errors on the logs/console |
| 01:24.49 | Jabroni | at first i though it was a nat problem, but I just hooked up an iax extension, and same happend, then i dialed the iax to sip phone... and i did had 2 way conversation |
| 01:24.49 | SarahEmm | no issues with audio outside of apps? |
| 01:24.56 | SarahEmm | hrm |
| 01:24.58 | SarahEmm | odd. |
| 01:26.22 | Docelm0 | say anyone in here a termination provider? |
| 01:36.30 | pdthome | Jabroni: do you have any hardware zaptel cards? |
| 01:37.28 | *** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 01:38.16 | websae | are there any softphones with g.729 support |
| 01:39.56 | Jabroni | •pdthome• yes.. a configured tdm400 and a unconfigured E1/t1 |
| 01:41.49 | pdthome | so if you do zap show channels those show up |
| 01:42.01 | Jabroni | yup |
| 01:42.36 | Jabroni | pseduo and the 4 analogic trunks |
| 01:42.44 | *** join/#asterisk Johnnie (n=john@pdpc/supporter/active/Johnnie) |
| 01:42.50 | znoG | enough with the NuFone thread !!!!!!!! |
| 01:44.22 | queso | Ok, silly question, I just purchased Asterlink for voip termination, do I need to copy both the sip and iax2 setup info? |
| 01:44.31 | websae | no |
| 01:44.36 | websae | choose either sip or iax2 |
| 01:44.47 | queso | is there any major difference? |
| 01:44.48 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 01:44.55 | websae | sip is better :) |
| 01:45.25 | mitcheloc | i heard iax2 was better |
| 01:45.33 | Jabroni | iax2 is better |
| 01:45.36 | Jabroni | no nat problems |
| 01:48.49 | Jabroni | pdthome any recommendation to fix my issue?? |
| 01:48.57 | websae | anyone else here with a polycom 301? |
| 01:49.05 | pdthome | i have had a similar issue before but I can't remember how I fixed it :/ |
| 01:49.29 | Jabroni | ive tried recompiling from source |
| 01:49.49 | Jabroni | asterisk and zaptel, ran genzaptelconf |
| 01:53.35 | queso | if I setup an iax2 trunk, my sip softphones can still use it, right? |
| 01:54.36 | pdthome | ya |
| 01:57.51 | queso | What does this dial pattern mean: _1NXXNXXXXXX |
| 01:57.51 | queso | ? |
| 01:58.35 | SarahEmm | N is 2-9, X is 0-9 |
| 01:58.41 | SarahEmm | 1 is.. err.. the number 1 |
| 01:58.53 | queso | what does the _ do? |
| 01:59.28 | MiniDevil | Does anyone here run the Xorcom Rapid Asterisk? |
| 02:00.13 | *** join/#asterisk mosty (i=mostynm@60-241-198-194.static.tpgi.com.au) |
| 02:00.39 | SarahEmm | queso: that tells * that it's a pattern, not an actual number |
| 02:00.59 | queso | oh |
| 02:01.00 | queso | ok |
| 02:02.30 | SarahEmm | meow |
| 02:03.06 | mosty | in my dialplan i have a couple of dial commands one after the other, i was expecting this to dial one extension, and then if that can't through it dials the next and so on, but instead after dialling the first extension it jumps straight to voicemail for that extension after getting SIP response 302 "Moved Temporarily". why is this happening? |
| 02:03.24 | SarahEmm | pastebin the dialplan? |
| 02:05.51 | queso | does anyone here use a macosx sip client? |
| 02:05.51 | mosty | here you go: http://pastebin.ca/85608 |
| 02:06.04 | queso | I can't seem to get gizmo to login to my server, it just hangs during login |
| 02:07.36 | mosty | SarahEmm: the sip phone seems to be forwarding the call directly to voicemail- how can i prevent sip clients from doing that? i want the dialplan to control that not the phones |
| 02:07.51 | znoG | i'm a bit shocked by the emails trelane is sending |
| 02:08.03 | znoG | didn't think he would get so involved in such a pointless thread |
| 02:08.03 | SarahEmm | what sip phone is this? |
| 02:08.18 | *** join/#asterisk Terlouw (n=Terlouw@80.126.223.172) |
| 02:08.28 | mosty | SarahEmm: Useragent : Sipura/SPA941-4.1.8 |
| 02:08.41 | Terlouw | people!! i have a nice mystery here :) |
| 02:09.33 | Terlouw | im getting a "all circuits are busy now" when trying to dial out. internal everything works fine, incomming calls also work fine, but the outgoing... pfff it is driving me MAD!!! anyone got a clue on where to look ?? |
| 02:09.43 | SarahEmm | mosty: is the phone set to DND or anything? |
| 02:09.47 | SarahEmm | or just rings? |
| 02:10.55 | mosty | SarahEmm: the phone seems to forward immediately. but i don't want the phones to be allowed to do that- i want this controlled from the dialplan |
| 02:11.19 | SarahEmm | mosty: okay, so you're saying even if the user sets the phone to forward, you don't want to honor that? |
| 02:11.29 | Terlouw | .... anyone .... |
| 02:11.41 | mosty | SarahEmm: yes |
| 02:11.51 | SarahEmm | ahh. i have no idea how to make * ignore SIP302 received |
| 02:11.54 | SarahEmm | sorry! |
| 02:14.03 | websae | anyone familiar with POLYCOM 301s? |
| 02:14.17 | websae | I am trying to figure out how to get it to get a DHCP address |
| 02:15.02 | mosty | SarahEmm: i am logged into the phone now, and i cant see any setting telling it to forward to 101 |
| 02:15.33 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.18.156.Dial1.SanJose1.Level3.net) |
| 02:16.14 | SarahEmm | it's not set to do not disturb right now or anything? |
| 02:17.17 | mosty | ahh wait, i see it now |
| 02:17.59 | mosty | i've turned off the call forwarding on the phone, will try again |
| 02:18.04 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.18.156.Dial1.SanJose1.Level3.net) |
| 02:19.45 | Terlouw | can anyone take a look at this and tell me what goes wrond here ? http://pastebin.ca/85634 |
| 02:22.56 | SarahEmm | there is no host called VoipBuster |
| 02:23.50 | Terlouw | well... that is the trunkname used to call out... it is only used as trunkname... nothing else... |
| 02:25.03 | SarahEmm | well, it doesn't exist, and it's being dialled out on |
| 02:25.09 | SarahEmm | so it's failing |
| 02:26.35 | queso | Ok, I installed loudhush and it seems to login ok (Registration excepted) but any time I try to dial it says "Phone 200 is unavailable) 200 is my extension. |
| 02:27.01 | queso | I set it up as an IAX2 extension |
| 02:27.06 | Terlouw | ok, so i created a trunk called "VoipBuster" i selected that trunk to be used as default voip-out gateway... and asterisk says that trunk does not excist? ... what could cause that? |
| 02:28.31 | *** join/#asterisk s0lid (n=s0lid@61.28.161.132) |
| 02:28.36 | Terlouw | ok, so i created a trunk called "VoipBuster" i selected that trunk to be used as default voip-out gateway... and asterisk says that trunk does not excist? ... what could cause that? |
| 02:29.13 | SarahEmm | Terlouw: pastebin your SIP config |
| 02:30.10 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 02:30.26 | trelane | znoG, just trying to sort things out |
| 02:30.43 | trelane | you do realize I spend much time inhabiting news.admin.net-abuse.* |
| 02:31.13 | Terlouw | my sip.conf = http://pastebin.ca/85645 |
| 02:32.38 | *** join/#asterisk mrtwister (n=mrtwiste@107.250.broadband5.iol.cz) |
| 02:32.40 | SarahEmm | sorry lost power |
| 02:32.42 | SarahEmm | one sec Terlouw |
| 02:33.22 | SarahEmm | uhh |
| 02:33.26 | SarahEmm | are you using amp or something Terlouw? |
| 02:33.36 | Terlouw | that is correct |
| 02:33.38 | SarahEmm | oh. |
| 02:33.41 | SarahEmm | i have noooo idea then. |
| 02:33.43 | SarahEmm | i'm anti-amp heh |
| 02:33.54 | Terlouw | oh :( |
| 02:33.56 | SarahEmm | pastebin sip_custom and sip_additional? |
| 02:34.01 | SarahEmm | i can try, but no guarantees |
| 02:36.59 | hads|home | Terlouw: #freepbx will probably be able to help more |
| 02:37.27 | Terlouw | ok this is the sip additional, i dont have the other one |
| 02:37.28 | Terlouw | http://pastebin.ca/85647 |
| 02:38.32 | SarahEmm | nowhere in any of these is anything called VoipBuster |
| 02:38.43 | SarahEmm | i think what hads|home said is good. try #freepbx, i have no idea about AMP/freepbx |
| 02:38.55 | Terlouw | ok ... thanks anyway... |
| 02:39.55 | znoG | trelane: you, as well as trixter, should lay off that thread. It's growing exponentially on the basis of a troll message sent in by Robert |
| 02:40.36 | [TK]D-Fender | ~centosbug |
| 02:40.38 | jbot | i guess centosbug is a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
| 02:40.57 | trelane | znoG, I'm off it, I spoke with file earlier and was under the impression that something was being done, and when nothing was done, and unprovable accusations still being made, I contacted kpfleming who has hopefully put an end to it |
| 02:41.15 | file | we sort of went out to dinner |
| 02:41.24 | file | people showed up... |
| 02:41.27 | trelane | file, I blame noone for you are file, the immutable :) |
| 02:57.02 | *** join/#asterisk ivanfm (n=ivanfm@201.52.162.52) |
| 02:57.06 | *** join/#asterisk SwK (n=Silik0nJ@12-218-74-89.client.mchsi.com) |
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| 03:01.49 | BugKham | anyone knows if minregexpire and maxregexpire are applicable to 1.2.x's iax.conf? |
| 03:02.01 | BugKham | I don't see them in the wiki |
| 03:03.03 | *** join/#asterisk SplasPood (n=jwb@ool-18b93b84.dyn.optonline.net) |
| 03:05.58 | *** join/#asterisk r0d3nt|m (n=RatMan@foster.stonedcoder.org) |
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| 03:10.58 | *** part/#asterisk santiago (i=santiago@debian/developer/santiago) |
| 03:12.01 | Dovid | ~pb |
| 03:12.06 | jbot | pb is, like, a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
| 03:13.06 | Netgeeks | hey does anyone know off the top of thier head whether or not you can nestle dialplan macros? in other words can a macro call a macro |
| 03:13.31 | Dovid | hmm |
| 03:13.33 | Dovid | y not |
| 03:13.34 | Dovid | ? |
| 03:13.39 | Dovid | try it and report back to the class :) |
| 03:13.57 | *** join/#asterisk jsaunders (n=jsaunder@S01060060971c5817.va.shawcable.net) |
| 03:16.18 | Jabroni | does asterisk use a dummy zap interface to play all the internal sounds? |
| 03:20.39 | Netgeeks | Jabroni: no the only place where dummy zap interfaces are required is places where timing is needed. MeetMe comes to mind, but there is something else too I believe |
| 03:21.21 | SarahEmm | MOH used to iirc.. |
| 03:21.24 | Jabroni | so asterisk plays back the sounds directly on the channels?? |
| 03:21.25 | SarahEmm | i could be wrong on that tho |
| 03:21.32 | Jabroni | for like voicemail ? |
| 03:21.53 | mosty | jabroni: why shouldn't it? |
| 03:22.17 | Jabroni | just trying to figure out a reason why asterisk wont play internal sounds |
| 03:22.44 | Jabroni | running 1.2.9.1 |
| 03:23.17 | mosty | what do the logs say? |
| 03:23.33 | Jabroni | nothing.. no errors |
| 03:23.51 | Jabroni | even with verbose set to 99 |
| 03:24.18 | mosty | do you see the command trying to play the sound in the console? |
| 03:24.30 | *** join/#asterisk beyond (n=beyond@201-1-205-226.dsl.telesp.net.br) |
| 03:24.35 | Jabroni | yup.. but no sound |
| 03:25.15 | mosty | sounds like a NAT vs SIP problem |
| 03:25.21 | Jabroni | audio between phones are ok, its just the apps that wont output sound |
| 03:26.09 | Jabroni | yeah thats what i thought first.. but i then i tested with an iax extension.. and soem result.. and dialing between the sip and iax works fine |
| 03:28.00 | mosty | hmm |
| 03:31.55 | *** join/#asterisk alphaque (n=alphaque@60.51.217.61) |
| 03:35.28 | alphaque | has anyone got counterpath's eyebeam softphone to work as a video phone with asterisk over SIP ? |
| 03:35.38 | alphaque | we seem to be facing problems, and would appreciate some guidance |
| 03:36.06 | *** join/#asterisk akant2 (n=arthur@ip24-252-29-236.om.om.cox.net) |
| 03:37.00 | akant2 | Can anyone tell me where I can buy a REAL X100P ? |
| 03:37.34 | [TK]D-Fender | akant2 : Discontinued. Good luck. Most of the ones claimed as real on ebay and elsewhere are just clones. |
| 03:37.48 | akant2 | damn |
| 03:38.07 | akant2 | I knew they were discontinued, I was hoping that someone picked up a huge lot and was selling htem |
| 03:38.09 | SarahEmm | why do you need a rel one? |
| 03:38.27 | akant2 | as I hear non real != good ? |
| 03:38.31 | SarahEmm | ... |
| 03:38.34 | SarahEmm | real != good either. |
| 03:38.39 | akant2 | lol |
| 03:38.40 | SarahEmm | the x100p isn't a shining example of a good card |
| 03:38.56 | mosty | akant2: they are all lousy |
| 03:38.59 | akant2 | What is recommended for a simple single line termination then? |
| 03:39.09 | SarahEmm | get someone else to terminate and use IAX? ;) |
| 03:39.14 | akant2 | lol |
| 03:39.18 | SarahEmm | or use a tdm400 with one FXO module in it |
| 03:39.19 | [TK]D-Fender | akant2 : TDM400P, A200, SPA-3000, SPA-3102 |
| 03:39.26 | SarahEmm | or a SPA, yeah |
| 03:39.33 | akant2 | SPA? |
| 03:39.34 | SarahEmm | you can try an x100p clone, some people use 'em |
| 03:39.34 | akant2 | hmm |
| 03:39.55 | akant2 | SaraEmm which one? |
| 03:39.59 | Netgeeks | Analog cards will cause you to go insane, I suggest avoiding them like ebola |
| 03:40.04 | SarahEmm | umm, ambient msomething chipset |
| 03:40.07 | SarahEmm | it's a clone |
| 03:40.40 | akant2 | TDM400P this is the quad analog card? |
| 03:40.52 | hads|home | akant2: Yes, 1-4 ports |
| 03:41.21 | akant2 | Sarah: I was thinking about getting one of those cards from www.x100p.com or where ever :) |
| 03:41.43 | hads|home | akant2: Where in the world are you? |
| 03:42.05 | SarahEmm | akant2: i got mine a ocuple years back, dunno where from. was an ebay seller |
| 03:42.09 | akant2 | Is the TDM400P a reliable card? I would like to find something for small office setups |
| 03:42.12 | akant2 | Nebraska |
| 03:42.42 | hads|home | People in the US have less trouble with the X100 cards, but it's still not usually worth it. |
| 03:43.13 | SarahEmm | but phone-wise i might as well be in the US. same deal. |
| 03:43.21 | akant2 | I used one of the quad cards awhile back and I thought it worked well.. but had trouble with asterisk releasing channels properly |
| 03:43.33 | akant2 | had to reboot every two weeks or so |
| 03:43.41 | akant2 | that was an older asterisk tho too |
| 03:43.43 | Netgeeks | akant2: the problem with asterisk releasing analog channels is endemic with any analog card |
| 03:43.50 | Netgeeks | tdm4Xx included |
| 03:44.09 | hads|home | Only if you are missing disconnect supervision |
| 03:44.23 | akant2 | hmm |
| 03:44.32 | Netgeeks | hads: heh, as are most folks in the US |
| 03:44.46 | akant2 | disconnect supervision? |
| 03:44.55 | akant2 | <--- learning telecom terms here :) |
| 03:45.20 | mosty | netgeeks: i have been wondering why any el cheapo answering machine can detect hangup on analogue lines, but tdm400p's etc have a real hard time |
| 03:45.21 | hads|home | Netgeeks: Yeah, the telco here (NZ) will enable it for you if you bug them. |
| 03:45.33 | SarahEmm | it's always on, here |
| 03:45.39 | SarahEmm | you get a battery reversal for it |
| 03:46.03 | hads|home | akant2: http://www.voip-info.org/wiki-Asterisk+Disconnect+Supervision |
| 03:46.03 | Netgeeks | mosty: I'm not sure why the x100p and tdm4xx have such a hard time with it |
| 03:46.19 | SarahEmm | well, the x100p is a modem |
| 03:46.23 | SarahEmm | err, designed as such |
| 03:46.28 | SarahEmm | modems don't really have a need to detect disconnect |
| 03:46.35 | SarahEmm | not at that layer |
| 03:47.13 | hads|home | I've never really used an old answer machine, do they do actual disconnect supervision or just listen for busy tones? |
| 03:47.36 | SarahEmm | i think they just wait for silence or busy tones |
| 03:47.46 | SarahEmm | they must, because we only got disconnect super a couple years ago |
| 03:47.50 | SarahEmm | and they worked fine before that |
| 03:48.06 | hads|home | Yeah, that sounds right. |
| 03:48.12 | SarahEmm | bell sent out this notice to everyone talking about disconnect supervision coming soon and battery reversals and such |
| 03:48.17 | SarahEmm | COMPLETELY confused 99.9% of people |
| 03:48.17 | *** join/#asterisk THX2000 (i=AgentFLY@adsl-66-51-192-221.dslextreme.com) |
| 03:48.37 | hads|home | Wow, no-one would have had a clue what they were talking about |
| 03:48.44 | SarahEmm | no... |
| 03:48.50 | *** join/#asterisk viking78 (n=chatzill@66-168-98-144.static.jcsn.tn.charter.com) |
| 03:48.51 | SarahEmm | i'm not sure who the heck decided ot send that out |
| 03:48.56 | hads|home | heh |
| 03:48.59 | SarahEmm | anyone who remotely knew me, asked me.. :P |
| 03:49.08 | SarahEmm | i can't imagine how their helpdesk felt |
| 03:49.52 | Netgeeks | Did you tell them that it meant anytime they made a call, someone from the phone company would listen in on the call and supervise them until they disconnected? I would have and let it go for a day just to have fun |
| 03:50.58 | SarahEmm | lol |
| 03:51.05 | SarahEmm | few years too late, doh! |
| 03:51.40 | THX2000 | What version of gcc do i need to use to compile SCCP? |
| 03:52.03 | *** join/#asterisk bmg505 (n=leon@c1-142-5.rndf.isadsl.co.za) |
| 03:57.24 | *** join/#asterisk DrkShdw (n=DrkShdw@fl-209-26-20-205.sta.embarqhsd.net) |
| 04:02.04 | BugKham | anyone knows if minregexpire and maxregexpire are applicable to 1.2.x's iax.conf? |
| 04:17.09 | *** join/#asterisk ivanfm (n=ivanfm@201.52.162.52) |
| 04:21.08 | *** join/#asterisk Qwell[\0] (n=Qwell[]@unaffiliated/qwell) |
| 04:21.38 | THX2000 | Has anyone here successfully compiled chan_sccp2? |
| 04:22.14 | Qwell[\0] | THX2000: chan-sccp-users@lists.berlios.de |
| 04:28.35 | THX2000 | Still cant find any info on the error im getting :/ |
| 04:35.05 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:35.36 | Docelm0 | Qwell how goes testing with the sun? |
| 04:36.14 | alphaque | has anyone got counterpath's eyebeam softphone to work as a video phone with asterisk over SIP ? |
| 04:36.56 | Qwell[laptop] | Docelm0: delayed |
| 04:37.10 | Docelm0 | they take it back or making you buy it? |
| 04:37.20 | Qwell[laptop] | neither, I'm in HSV this week |
| 04:37.28 | Docelm0 | HSV? |
| 04:37.31 | Qwell[laptop] | hard to bring an ~80lbs server on a plane |
| 04:37.35 | Qwell[laptop] | Docelm0: Digium |
| 04:37.47 | Docelm0 | ohh thats right.. |
| 04:37.59 | Docelm0 | I remember people talking bout that the other day there's a bunch of people out there |
| 04:38.04 | Qwell[laptop] | yeah |
| 04:38.09 | Docelm0 | Whats the deal? Anything great happening? |
| 04:38.13 | Qwell[laptop] | all the Digium devs...and me |
| 04:38.26 | Qwell[laptop] | Docelm0: nah, just a convenient time to get together |
| 04:38.32 | Docelm0 | ahh nice.. |
| 04:38.37 | Docelm0 | that works |
| 04:38.51 | trelane | anyone have a clue who michael workman is? |
| 04:39.15 | hads|home | I thought that thread was over? |
| 04:39.37 | Docelm0 | Why? What the hell is going on now? |
| 04:40.25 | hads|home | A general bitch session. |
| 04:41.24 | *** join/#asterisk santiago (i=santiago@debian/developer/santiago) |
| 04:41.47 | Qwell[laptop] | I think Kevin is gonna have a fit if it continues, heh |
| 04:42.06 | hads|home | So he should. |
| 04:46.02 | *** join/#asterisk ivanfm_ (n=ivanfm@201.52.162.52) |
| 04:47.47 | trelane | hads|home, I offered to step in and assist Michael in contacting nufone and got flamed for it |
| 04:49.51 | hads|home | I know. It still has nothing to do with using Asterisk though :) |
| 04:53.02 | *** join/#asterisk MatsK (i=MatsK@83.233.97.229) |
| 04:53.55 | orlock | ngah |
| 04:54.04 | orlock | softphone isnt setting tos bits like most real phones |
| 04:54.55 | *** join/#asterisk Trazz (i=Trazz@c-67-163-92-37.hsd1.il.comcast.net) |
| 04:55.35 | trelane | trixter: ping (dunno if you frequent here) |
| 04:56.08 | *** join/#asterisk alephcom (n=alephcom@h11.n66-112-182.mcsnet.ca) |
| 04:56.46 | bugz | do you guys recommend the digium certifications and boot camp, etc? |
| 04:57.12 | *** part/#asterisk BugKham (i=BugKham@202.8.86.164) |
| 04:57.20 | bugz | i work with a company that has over a hundred * boxen in production, but only half a dozen employees, 3 technical |
| 04:57.32 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 04:57.33 | bugz | im just wondering if its worth it to miss work |
| 04:57.52 | bugz | personally id like to attend the boot camp and the dev shop |
| 04:58.33 | *** join/#asterisk mog_home (n=mogorman@68.62.237.103) |
| 04:58.46 | Qwell[laptop] | mog: Long drive home? :) |
| 04:59.26 | mog_home | 800km |
| 04:59.32 | Qwell[laptop] | ha |
| 04:59.40 | Qwell[laptop] | I *totally* didn't see that coming |
| 04:59.47 | mog_home | ^_^ |
| 04:59.53 | Qwell[laptop] | mog: What's the college here? |
| 04:59.59 | mog_home | uah |
| 05:00.08 | Qwell[laptop] | expanded? |
| 05:00.11 | mog_home | university of alabama at/in huntsville |
| 05:00.14 | Qwell[laptop] | ahh |
| 05:01.28 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 05:02.33 | Docelm0 | bugz, how new are you to asterisk? |
| 05:03.58 | *** join/#asterisk babyju (n=babyju@h-67-102-255-186.nycmny83.covad.net) |
| 05:04.43 | *** join/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net) |
| 05:04.46 | bugz | Docelm0: ive been working on it daily since november |
| 05:04.48 | bugz | nightly too.... |
| 05:04.57 | bugz | fairly new i guess |
| 05:05.00 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 05:05.02 | TripleFFFF | hey |
| 05:05.06 | bugz | i can build a hell of a box though ! |
| 05:05.14 | TripleFFFF | anyway to make sure we log an ip of a sip device that logs on ? |
| 05:05.32 | Docelm0 | ok so you have 9 months behind you.. do you feel comfortable with the CLI configurations, advanced configurations, custom AGI scripts, etc? |
| 05:05.32 | TripleFFFF | some freaks hicjacked an office we have in toronto area pointing guns and emptied the office |
| 05:05.34 | bugz | 100% zttest B] |
| 05:05.39 | TripleFFFF | so when they log in i want to see where from |
| 05:05.47 | bugz | absolutely |
| 05:05.49 | TripleFFFF | since xten was on these cpu's it will relog them from current iusp |
| 05:06.00 | TripleFFFF | isp that is |
| 05:06.12 | TripleFFFF | i got full i n logger but not sure what else to put |
| 05:06.31 | bugz | Docelm0: im just missing a few pieces here and there |
| 05:06.35 | bugz | but mostly it makes sense |
| 05:06.40 | Docelm0 | The only way to track is turn your logger up and watch the shit roll into the server.. |
| 05:06.57 | Docelm0 | unless you use realtime then the SIP URI will show up in the DB |
| 05:07.08 | bugz | holy crap |
| 05:07.16 | Docelm0 | bugz, then no. Keep working at it and ask questions.. |
| 05:07.25 | Docelm0 | the camps are for complete newb's to the areas |
| 05:07.52 | bugz | thanks for the input. what about the dev stuff they offer? |
| 05:08.02 | bugz | im pretty close to calling some C with agi |
| 05:08.06 | Docelm0 | dont know.. learned on my own in a couple months |
| 05:08.44 | Docelm0 | dude.. Just use the app_skel.c to code your own C module and be done with it. Dont use AGI.. on a busy server it will load you down |
| 05:09.04 | Docelm0 | I know I wrote a LCR in PHPAGI for asterisk and it was crashing |
| 05:09.06 | bugz | sweet. its things like that i want to find out about. questions it is then. |
| 05:09.17 | Docelm0 | then I rewrote it in C and now Im pushing like 50+ calls/sec |
| 05:09.36 | bugz | nice |
| 05:09.39 | Docelm0 | Im usually here.. if you have questions research the web if you cant find it then ask here |
| 05:09.53 | *** join/#asterisk BugKham (i=BugKham@202.8.86.164) |
| 05:10.07 | bugz | ah, i posted a stupid bug report a while back |
| 05:10.09 | Docelm0 | But now Im gone.. To bed that is.. lates all.. |
| 05:10.26 | Docelm0 | its ok the DEV's will just get a laugh out of it.. :) |
| 05:10.32 | h3x | Docelm0: what kind of LCR did you do |
| 05:10.33 | BugKham | I did asterisk -rx "sip show peers" on two different servers |
| 05:10.34 | *** join/#asterisk dongs (n=HPUX@i249218.ppp.asahi-net.or.jp) |
| 05:10.36 | bugz | seeya |
| 05:10.41 | BugKham | one shows all peers |
| 05:10.47 | dongs | lol another dead fxs module |
| 05:10.50 | BugKham | and the other only show 109 peers |
| 05:10.53 | dongs | holy shit i'm never buying digium hardware ever again |
| 05:10.55 | bugz | btw i posted bug 7181 a while back and found a fix |
| 05:11.06 | Docelm0 | h3x, is there any other way to make a "LEAST COST" routing engine? |
| 05:11.10 | BugKham | both use the same version of * |
| 05:11.17 | h3x | well international or what |
| 05:11.26 | bugz | though it was closed as a user related issue .. if anyone comes across it on polycom phones you can use the config file for them to disable forwarding of calls |
| 05:11.29 | h3x | im working on a LRN based routing system |
| 05:11.30 | BugKham | anyone knows what makes the difference |
| 05:11.32 | Docelm0 | it routes e.164 compliant coalls |
| 05:11.34 | Docelm0 | calls |
| 05:11.58 | h3x | you cant really lcr us domestic calls with a flat table because of number portability |
| 05:12.11 | Docelm0 | uhh yes I can |
| 05:12.21 | Docelm0 | I have > 120k routes for US48 alone |
| 05:12.23 | h3x | your underlying carrier will charge you based on actual terminating OCN |
| 05:12.34 | Docelm0 | haha.. your new at this right? |
| 05:12.41 | h3x | no |
| 05:12.46 | h3x | im dipping NPAC |
| 05:12.51 | h3x | like how most switches do it |
| 05:13.16 | h3x | but im caching the data |
| 05:13.17 | Docelm0 | Well my providers give me a flat breakout of codes, prices, and rounding.. I route based on that |
| 05:13.27 | *** join/#asterisk delmar (n=delmar@203-114-178-231.dsl.sta.inspire.net.nz) |
| 05:13.31 | Docelm0 | well route on code then price |
| 05:13.37 | h3x | did you do some mediation? |
| 05:13.45 | Docelm0 | mediation? |
| 05:13.49 | wunderkin | meditation? |
| 05:13.57 | h3x | compare what you thought it would cost compared to the carrier charging yoi |
| 05:13.57 | h3x | you |
| 05:14.10 | h3x | you know 38% of the numbers in the US are now ported |
| 05:14.17 | Docelm0 | yes.. I do upside down so I dont loose money |
| 05:14.21 | Docelm0 | so? |
| 05:14.34 | h3x | well you would be better off with no LCR |
| 05:14.38 | h3x | than a flat file LCR |
| 05:14.51 | Docelm0 | who's using flat files.. I use mysql |
| 05:14.57 | h3x | you know what i mean |
| 05:15.18 | h3x | the carriers try to make it look easy so tehy make money on you misrouting calls |
| 05:15.18 | Docelm0 | says you. I get a deck from a carrier and route based on price. I dont deal with this tier shit |
| 05:15.35 | h3x | LATA by Class? |
| 05:15.38 | Docelm0 | no misrouting here. If they ask for 1813817 I send it to that carrier |
| 05:15.40 | Docelm0 | nope |
| 05:15.50 | h3x | who the hell does that besides voicepulse |
| 05:15.55 | Docelm0 | haha me |
| 05:16.03 | Docelm0 | and my pricing is better btw |
| 05:16.19 | h3x | qwest, global, sprint, broadwing, you name it |
| 05:16.26 | h3x | they all send you tables that say npa/nxx to class |
| 05:16.36 | h3x | and they bill on actual OCN of terminatign carrier |
| 05:16.56 | Docelm0 | Takes the guess work out of life.. you forgot one.. :) I use XO and GX for term and I get flat files with code, price, rounding none of this tier shit |
| 05:16.57 | BugKham | okie, got it |
| 05:17.07 | h3x | dude XO is even worse |
| 05:17.11 | h3x | they base it on LRN |
| 05:17.19 | h3x | thats why i had to rework my system |
| 05:17.21 | Docelm0 | LRN? |
| 05:17.31 | h3x | and by the way Global Crossing bills before answer supervision so fuck them |
| 05:17.36 | *** join/#asterisk daysmen3 (n=primus@host86-141-242-253.range86-141.btcentralplus.com) |
| 05:17.48 | Qwell[laptop] | GX bothers me |
| 05:17.49 | h3x | Yes, the LRN is a 10 digit number assigned to the switch when a number is ported |
| 05:17.55 | Docelm0 | haha |
| 05:17.55 | Qwell[laptop] | 10c/min for conf calls...pfft |
| 05:18.07 | Docelm0 | They dont route based on that |
| 05:18.09 | Qwell[laptop] | 10c/min PER connected user, that is |
| 05:18.10 | Docelm0 | sigh.. |
| 05:18.11 | h3x | yes they do |
| 05:18.16 | h3x | i know the VP of XO Carrier Services |
| 05:18.19 | Docelm0 | I HAVE THEIR RATE DECK |
| 05:18.23 | h3x | Roy Burdette, ask him how their rate deck works |
| 05:18.28 | Docelm0 | Trust me no they dont |
| 05:18.29 | h3x | that table is based on LERG 12 |
| 05:18.43 | h3x | its the NPA/NXX of the LRN not the WTN |
| 05:18.48 | wunderkin | i thought he was that hotel guy.. |
| 05:18.49 | h3x | when the number is ported |
| 05:18.51 | Docelm0 | sigh.. |
| 05:19.00 | Qwell[laptop] | wunderkin: the motel 6 guy? |
| 05:19.03 | wunderkin | hehe |
| 05:19.03 | Docelm0 | haha |
| 05:19.13 | Qwell[laptop] | no, that's...umm |
| 05:19.21 | *** join/#asterisk envoy (n=user@000-016-837.area3.spcsdns.net) |
| 05:19.24 | Docelm0 | roy handova |
| 05:19.26 | Docelm0 | or some shit |
| 05:20.02 | wunderkin | tom beaudette or however you spell it |
| 05:20.09 | h3x | are you on their portal or saville billing system? |
| 05:20.28 | h3x | saville gives you 6/1 increment but they bill for both unanswered calls |
| 05:20.38 | h3x | and at the moment digits are presented and the call starts ringing |
| 05:21.11 | Qwell[laptop] | wunderkin: yes |
| 05:21.11 | h3x | portal gives you 6/6 but its after answer supervision |
| 05:21.28 | Qwell[laptop] | Bodett.. |
| 05:21.33 | Docelm0 | portal |
| 05:21.37 | h3x | I started using XO CLDI in 2004 |
| 05:21.37 | h3x | heh |
| 05:21.43 | Qwell[laptop] | http://en.wikipedia.org/wiki/Tom_Bodett :P |
| 05:21.45 | h3x | CLDT |
| 05:22.05 | h3x | in any case, it dosent matter, even their tiered deck is rated based on LRN |
| 05:22.07 | Docelm0 | look I know what I get charged and dont get charged for.. Im not going to sit here and argue with you bout it. |
| 05:22.18 | Docelm0 | GX is flat and so is XO.. |
| 05:22.24 | Docelm0 | So my LCR is perfectly fine |
| 05:22.48 | h3x | it aint flat dude |
| 05:22.48 | h3x | heh |
| 05:23.00 | h3x | what you described is rated off the LERG |
| 05:23.16 | Docelm0 | sure looks flat to me.. I dont deal with OCN's or Tier's or Lata's or whatever. Its just NPANXX and a Price |
| 05:23.21 | h3x | you know why they have to do this right? reciprocal compensation |
| 05:23.22 | Docelm0 | how much flatter do ya get geesh |
| 05:23.36 | h3x | that is the NPA/NXX of the ported LRN number |
| 05:23.40 | h3x | not the dialed number |
| 05:23.54 | Docelm0 | ohh you mean intra/inter state? |
| 05:23.56 | h3x | no |
| 05:24.01 | h3x | here try this |
| 05:24.03 | Docelm0 | well thats the only other variable |
| 05:25.16 | h3x | XOs pricing can be different for the same rate center and the same LEC but a different CO |
| 05:25.25 | h3x | if somebody ports a number from one part of town to another part of town |
| 05:25.40 | h3x | it can be different because they may have DECT trunks to the COs switch in one place and not another |
| 05:25.59 | Docelm0 | sigh I dont care about porting.. your giving me a headache.. quit.. Cause I am not going to say your right or wrong.. I know what I use I know what I have I know how Im billed.. |
| 05:26.08 | h3x | Anyway, the company I am using for SS#7 database stuff installed and supports XO's SS#7 network |
| 05:26.31 | Docelm0 | Do you want a cookie by chance? |
| 05:26.32 | *** part/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net) |
| 05:26.49 | h3x | 1000 blocks are *all* ported |
| 05:26.51 | h3x | numbers |
| 05:26.54 | h3x | out of a 10000 block |
| 05:27.14 | h3x | if a carrier gets a 1000 block, they all show up as ported out of the 10,000 block A code holder |
| 05:27.29 | h3x | most wireless numbers are ported even if you didnt switch carriers |
| 05:27.50 | Docelm0 | And I care because.. I mean seriously.. This really isnt a BIZ channel.. more USER but seriously are you just trying to be annoying come to a point some day? seriously.. I dont care cause I know what I have.. and well I just dont care.. |
| 05:27.53 | h3x | for instance, I have a cingular wireless number that was on cingular blue before, now its on orange, but even though its on orange - cingular ported it again out to a different HLR |
| 05:28.00 | h3x | so they could load balance their network ebtter |
| 05:28.28 | h3x | Because, its totally futile to make a US Domestic LCR from a flat database instead of querying the NPAC data |
| 05:28.38 | h3x | because people are uneducated about it |
| 05:29.14 | Docelm0 | look I dont have to dip NPAC I get a deck I run with it every quarter.. Im happy.. And I am educated but I dont have access to the SS7 network yet. Everything is provided for me so I dont have to |
| 05:29.15 | h3x | once somebody finds holes in your rate deck and somebody like me comes along that finds it i'd send you all sorts of purposely misrouted calls |
| 05:29.20 | h3x | and make you run out of a bunch of money |
| 05:29.35 | Docelm0 | www.plainvoip.com go to town |
| 05:29.38 | Docelm0 | have a blast |
| 05:29.39 | Docelm0 | enjoy |
| 05:29.42 | h3x | im carrierone.net :P |
| 05:29.50 | Docelm0 | and I really care |
| 05:30.16 | h3x | it would be useful for you to use NPAC to determine number porting to DIDs too |
| 05:30.33 | Docelm0 | Congrats.. You have make it to my /ignore list.. Seriously.. ACK! |
| 05:30.52 | Docelm0 | what was that? I didnt hear you.. |
| 05:31.08 | Qwell[laptop] | bed |
| 05:31.15 | h3x | ported |
| 05:31.53 | h3x | Oh look, its the next nufone! |
| 05:31.54 | h3x | ghahahahahahah |
| 05:32.03 | h3x | jeremy rev 2 |
| 05:32.06 | *** join/#asterisk CoffeeKid (n=kirkalle@dsl093-224-026.slc1.dsl.speakeasy.net) |
| 05:32.12 | CoffeeKid | Hello :) |
| 05:32.47 | CoffeeKid | i'm getting alot of the following error in my asterisk system: WARNING[18706]: channel.c:787 channel_find_locked: Avoided initial deadlock for '0x60f850', 10 retries! |
| 05:33.01 | CoffeeKid | anyone see this before? |
| 05:33.20 | CoffeeKid | it seems the more of them i get, the more likely the system is to crash. |
| 05:35.10 | Docelm0 | ok now Im going to bed.. night all |
| 05:35.22 | CoffeeKid | night! :) |
| 05:39.14 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 05:45.27 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 05:47.52 | *** join/#asterisk clive- (n=pirch@dsl-165-184-66.telkomadsl.co.za) |
| 05:54.19 | *** join/#asterisk Phantom-X (n=foo@c-212-031-186-116.cust.broadway.se) |
| 05:54.25 | Phantom-X | hello there |
| 06:02.59 | stoffell | CoffeeKid, i also had something like that a while ago.. what version are you using? |
| 06:05.31 | *** join/#asterisk spooky7 (n=spooky@ipa121.208.tellas.gr) |
| 06:05.43 | spooky7 | Good morning to all |
| 06:06.17 | stoffell | morning spooky7 |
| 06:06.20 | Phantom-X | im new to asterisk |
| 06:06.29 | Phantom-X | installing it on my gentoo =9 |
| 06:06.35 | spooky7 | me too pahntom-x |
| 06:06.53 | spooky7 | what is your problem ? |
| 06:06.57 | Phantom-X | heard nice things about it , so im going to hook it to my sip service and boost my cisco ata 186 =) |
| 06:06.59 | stoffell | ever tried trixbox ? |
| 06:07.16 | Phantom-X | trixbox ? |
| 06:07.28 | Phantom-X | i ahve no problem yet |
| 06:07.41 | stoffell | it's sort of an almost running system, it's good to get a first impression before you get into the real stuff :) |
| 06:07.47 | Phantom-X | i bet there will be problems later when i ahve to configure the asterisk |
| 06:08.03 | Phantom-X | well |
| 06:08.07 | Phantom-X | im not in a hurry |
| 06:08.12 | Phantom-X | i know what asterisk can do |
| 06:08.20 | Phantom-X | so i prefer using that i guess |
| 06:08.48 | spooky7 | do you know anything about asterisk having a gui to manage it ? |
| 06:08.53 | stoffell | sure, it's perfect, and good that you're not in a hurry (almost everyone is, these days :p) |
| 06:09.04 | Phantom-X | hehe |
| 06:09.11 | Phantom-X | im used to console7terminal stuff |
| 06:09.16 | Phantom-X | im a tcl coder myself |
| 06:09.23 | stoffell | spooky7, have a look on voip-info.org, you'll find several links to gui's there |
| 06:09.27 | stoffell | hehe :) |
| 06:10.29 | spooky7 | I have already looked there , tried something but nothing hapenned . |
| 06:10.31 | Phantom-X | what i want to do is to script/program watever needed to construct a system to log CID data and timestamps etc to mail and call me or that i can call to to check and also to some more stuff |
| 06:10.41 | Phantom-X | i know asterist pussports this more or less |
| 06:11.13 | stoffell | spooky7, also try the docs |
| 06:11.15 | stoffell | ~docs |
| 06:11.20 | jbot | docs is, like, probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 06:11.30 | stoffell | Phantom-X, surely does |
| 06:11.33 | *** part/#asterisk dongs (n=HPUX@i249218.ppp.asahi-net.or.jp) |
| 06:11.42 | babyju | can asterisk act as a sip proxy for direct access from the internet...traversing firewalls? |
| 06:12.15 | Phantom-X | babyju: i guess so |
| 06:12.59 | Phantom-X | anyways im going to have asteriskt to interscept calls from my ata 186 , where i can make it accept dials to ip hosts |
| 06:13.10 | stoffell | babyju; or you can use one of the many sip proxies around, they fit the job... |
| 06:13.13 | Phantom-X | since the cisco box dont support this simply |
| 06:14.33 | Phantom-X | so hookup the phone dial a #0 and get into 'dial ip' mode |
| 06:14.45 | Phantom-X | since i know what ips my friends has on their cisco ata's |
| 06:15.07 | *** join/#asterisk af_ (n=af@ip-170-209.sn1.eutelia.it) |
| 06:15.11 | stoffell | hehe, that's possible :) |
| 06:15.21 | stoffell | well, anything's possible :) |
| 06:15.37 | spooky7 | Is thee any webmin module for asterisk ? |
| 06:15.55 | Phantom-X | spooky7: if i get satisfied with asterisk |
| 06:16.06 | Phantom-X | ill write an tcl/tk app |
| 06:16.11 | Phantom-X | as 'gui' |
| 06:16.24 | spooky7 | That's great . |
| 06:16.35 | Phantom-X | make sure ill att myself to it so i can call for free though your asterisk |
| 06:16.35 | spooky7 | I am looking forward for it . |
| 06:16.36 | Phantom-X | lol |
| 06:17.34 | Phantom-X | ok now the compile is done |
| 06:17.49 | Phantom-X | i did with minimals flags to make sure i dont need to install x for running it |
| 06:17.50 | spooky7 | My asterisk is not internet published / |
| 06:18.05 | Phantom-X | ok ic |
| 06:18.12 | Phantom-X | still u asked for webmin |
| 06:18.42 | spooky7 | yes . do you know anything ? |
| 06:18.47 | THX2000 | does anyone have chan_sccp2 running right now? |
| 06:19.02 | spooky7 | i don't |
| 06:19.07 | THX2000 | sweet :P |
| 06:19.15 | Phantom-X | i dont know |
| 06:19.15 | THX2000 | neither do i :P |
| 06:19.24 | spooky7 | he he |
| 06:19.29 | Phantom-X | i started with compiling asterisk when i joined here |
| 06:19.32 | stoffell | THX2000, uhm, i think I have chan_sccp running.. not the 2.. |
| 06:19.43 | THX2000 | I just can't get it to compile |
| 06:19.44 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 06:19.44 | *** mode/#asterisk [+o denon] by ChanServ |
| 06:19.48 | THX2000 | think its related to gcc |
| 06:19.50 | stoffell | THX2000, where's it's homepage.. ? |
| 06:20.00 | *** join/#asterisk Johnnie (n=john@pdpc/supporter/active/Johnnie) |
| 06:20.01 | THX2000 | http://chan-sccp.berlios.de/ |
| 06:20.50 | stoffell | THX2000, make sure to check the "how to debug" section, and try removing the asterisk include and module files.. (and after that reinstall them of course) |
| 06:20.55 | stoffell | it's very picky on that.. |
| 06:21.07 | THX2000 | its a fresh install of everything |
| 06:21.08 | tzafrir | Sponge_bob, there's a non-free one. The one in the asterisk ftp site is worthless. |
| 06:21.15 | THX2000 | sposidly that was to clean up upgraded versions |
| 06:21.33 | stoffell | mm, but stillm it's worth the try |
| 06:22.09 | tzafrir | spooky7, there's a non-free one. The one in the asterisk ftp site is worthless. |
| 06:23.50 | Phantom-X | ok |
| 06:24.21 | Phantom-X | now i need documentation , i couldnt manage ti onstall with documentation since then it wanted to isntall x because of pdf stuff |
| 06:25.00 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 06:25.25 | Phantom-X | hmm |
| 06:26.11 | *** join/#asterisk [hC] (n=hardcore@S01060004e21ea953.vc.shawcable.net) |
| 06:27.36 | bugz | drat!! defeated by Ganon again. |
| 06:28.21 | Phantom-X | <PROTECTED> |
| 06:28.31 | Phantom-X | i looked at google for documentation |
| 06:28.50 | Phantom-X | and there on asterisk documentation projec there was showing some gtk gui |
| 06:29.24 | THX2000 | No dice |
| 06:30.21 | Phantom-X | those damn pdf's |
| 06:30.53 | spooky7 | yes phantom |
| 06:31.54 | Phantom-X | i run console only almost |
| 06:32.16 | Phantom-X | i ahve no workstation to play with |
| 06:33.27 | stoffell | Phantom-X, what irc client are u using? :) |
| 06:34.13 | Phantom-X | irssi |
| 06:34.22 | Phantom-X | follows with gentoo liveCD |
| 06:34.42 | Phantom-X | and also something i use to install meanwhile i install computer without x |
| 06:34.52 | stoffell | Phantom-X, oh, okay.. i'm looking for something like that to let it run and allow me to attach to it from everywhere (with screen), i'll try it out |
| 06:35.00 | *** join/#asterisk dlynes_laptop (n=dlynes@S0106001217014b92.vc.shawcable.net) |
| 06:35.18 | Phantom-X | screen works well |
| 06:35.40 | Phantom-X | jsut do ctrl-a x or ctrl-a d |
| 06:35.44 | Phantom-X | while u rin irssi |
| 06:35.45 | Phantom-X | =) |
| 06:36.00 | Phantom-X | in gentoo i run screen on every 6 virtual consoles |
| 06:36.02 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 06:36.09 | Phantom-X | and those i dont use i ctrl-a x |
| 06:36.30 | Phantom-X | to make sure no unauthorized access can be done if they ahve phiscal access to the amchine |
| 06:36.30 | stoffell | yeah, screen's one of those wonderful apps.. :) |
| 06:36.45 | Phantom-X | since else they wil have a root shell without passwd resuirement |
| 06:36.46 | *** join/#asterisk astar` (n=astar@ANantes-154-1-105-111.w86-214.abo.wanadoo.fr) |
| 06:36.55 | Phantom-X | which mean they can plan whatever trojan>/malware |
| 06:37.18 | Phantom-X | before u have reached to ensure integrity wiith checksummings etc |
| 06:37.23 | Phantom-X | well |
| 06:37.28 | Phantom-X | i dont like screen with links |
| 06:37.39 | dlynes_laptop | did i join ##linux by mistake? |
| 06:37.43 | Phantom-X | since screen spoils the gpm data flow |
| 06:37.57 | Phantom-X | so u cannot navigate with mouse in it if u run screen |
| 06:38.12 | Phantom-X | asterisk runs on linux dont it ? =) |
| 06:38.24 | stoffell | dlynes_laptop, probably your laptop has a memory overflow ;) |
| 06:38.43 | stoffell | dlynes_laptop, oh, no, all the * problems are solved, THAT's it... ;) |
| 06:38.45 | dlynes_laptop | stoffell: nah....i've got 1/2 a screen of phantom-x's soliloquy |
| 06:39.18 | Phantom-X | lol |
| 06:40.10 | Phantom-X | that what im going to use asterisk for |
| 06:40.30 | dlynes_laptop | soliloquys? |
| 06:41.24 | dlynes_laptop | i didn't know asterisk allowed you to have one person conversations |
| 06:41.58 | Jabroni | anyone knows a way to get latest firmware for polycom.. their model of distribution of firmware is really wierd.. i tried contacting the vendor (voipsupply) and they refused to give out the latest one :/ |
| 06:42.37 | dlynes_laptop | Jabroni: that's fucked up |
| 06:43.30 | Phantom-X | i just say ackerman(4,4) |
| 06:44.44 | bugz | Jabroni: what firmware do you have now? |
| 06:45.13 | CoffeeKid | i'm getting alot of the following error in my asterisk system: WARNING[18706]: channel.c:787 channel_find_locked: Avoided initial deadlock for '0x60f850', 10 retries! |
| 06:45.23 | CoffeeKid | anyone see this before? |
| 06:45.37 | CoffeeKid | It seems to happen most when i'm running AGI scripts. |
| 06:45.53 | dlynes_laptop | CoffeeKid: do you have a loop in your agi scripts? |
| 06:46.19 | CoffeeKid | yeah, there's a few, but this one wasn't written by me, so i don't know all the details. |
| 06:46.36 | dlynes_laptop | CoffeeKid: i suspect one of those loops is probably causing the issue |
| 06:46.48 | dlynes_laptop | CoffeeKid: it's probably not yielding to the other threads |
| 06:46.52 | Jabroni | bugz let me look for the files... i cant remember which one was the last one i tried... |
| 06:47.07 | CoffeeKid | dlynes_laptop: hmm.. that makes sense.. what would be the best way to test this? |
| 06:47.10 | dlynes_laptop | CoffeeKid: try putting a sleep in the loops before they repeat |
| 06:47.25 | dlynes_laptop | CoffeeKid: if the thread locking issue disappears, that was the problem |
| 06:47.37 | *** join/#asterisk chiprunner (n=dominick@47-136.69-92-cpe.cableone.net) |
| 06:48.21 | CoffeeKid | dlynes_laptop: good idea.. i'll try that. |
| 06:49.09 | dlynes_laptop | CoffeeKid: use usleep; sleep will give you too much of a delay |
| 06:49.16 | dlynes_laptop | CoffeeKid: try a usleep of 200 or so |
| 06:49.25 | CoffeeKid | dlynes_laptop: i'm using php, does it have a usleep? |
| 06:49.37 | dlynes_laptop | CoffeeKid: don't know...i try to stay well away from php |
| 06:49.47 | dlynes_laptop | CoffeeKid: especially when it's not for page scripting |
| 06:49.59 | CoffeeKid | dlynes_laptop: i loves me some php :) |
| 06:50.30 | dlynes_laptop | if it doesn't have sleep or usleep |
| 06:50.37 | dlynes_laptop | it'll probably have something like Wait() |
| 06:50.59 | CoffeeKid | yeah, i'll check the manual |
| 06:51.14 | dlynes_laptop | actually...I didn't think anyone used php for anything other than page scripting, until I started using asterisk |
| 06:51.23 | dlynes_laptop | now i see people using it for command line programs |
| 06:51.24 | CoffeeKid | yep, usleep |
| 06:51.25 | dlynes_laptop | very strange |
| 06:51.42 | dlynes_laptop | CoffeeKid: anyways...if you still get the issue with a usleep of 200 |
| 06:51.42 | CoffeeKid | all of our cli programs are in php |
| 06:51.44 | CoffeeKid | it works great for that. |
| 06:51.52 | dlynes_laptop | CoffeeKid: try increasing it to 1000 or 2000 |
| 06:52.11 | CoffeeKid | dlynes_laptop: okay, i'll do some tests .. thank you for your help :) |
| 06:52.22 | dlynes_laptop | if it hasn't disappeared with a usleep of 100000, it's some other issue |
| 06:52.51 | dlynes_laptop | what happens is that when you've got a tight loop |
| 06:53.02 | Phantom-X | ehhe looks simple enough this stuff |
| 06:53.03 | dlynes_laptop | the current thread does not yield to other threads |
| 06:53.19 | Phantom-X | sayunixtime.so u kid me right ? |
| 06:53.46 | dlynes_laptop | a simple statement like for( i=0; i>=0; i++ ) ; will bring Windows to its knees so badly that even task manager won't come up |
| 06:53.58 | CoffeeKid | dlynes_laptop: anyway to set the thread prority low enough that it'll yeild when required? |
| 06:54.07 | *** part/#asterisk chiprunner (n=dominick@47-136.69-92-cpe.cableone.net) |
| 06:54.21 | dlynes_laptop | CoffeeKid: that just sets the priority low |
| 06:54.28 | mitcheloc | dlynes_laptop, and pressing alt-tab on my ubuntu machine brings it to it's knees as well |
| 06:54.32 | dlynes_laptop | CoffeeKid: you can still write bad code with low thread priorities, too |
| 06:54.32 | CoffeeKid | dlynes_laptop: hehe, yeah.. while(1); :) |
| 06:54.42 | Phantom-X | while 1 is also making windows go down ? |
| 06:54.50 | CoffeeKid | sure can |
| 06:54.57 | dlynes_laptop | Phantom-X: yeah...if that's all that's in the loop |
| 06:54.58 | CoffeeKid | while(1); |
| 06:54.59 | Phantom-X | heh |
| 06:55.09 | Phantom-X | thats interesting |
| 06:55.11 | Phantom-X | lol |
| 06:55.40 | Phantom-X | forhaps a non tail recursive function would be nice to to slurp memory fast |
| 06:55.40 | CoffeeKid | while(1) { $arr[] = "blahblahblah"; } |
| 06:55.47 | CoffeeKid | will take all the memory :) |
| 06:56.06 | CoffeeKid | granted you set your memory limit to all the memory of the computer.. |
| 06:56.21 | dlynes_laptop | Phantom-X: char * p=malloc(3490839482390483949023849038293) ; |
| 06:56.29 | Phantom-X | lets make an ordo ackerman 4 4 algorithm , then it dies within no time |
| 06:56.31 | dlynes_laptop | that'll bring your machien to its knees, too |
| 06:56.33 | Jabroni | bugz i got SoundPointIP_SIP_1_6_3.zip |
| 06:56.35 | CoffeeKid | yeah, thats the c++ verion way to do it :) |
| 06:56.43 | CoffeeKid | *version |
| 06:56.52 | Phantom-X | i have bounds checking |
| 06:56.53 | Phantom-X | =) |
| 06:57.05 | Phantom-X | so it will jsut say "iodiot, dont do that" lol |
| 06:57.12 | dlynes_laptop | heh |
| 06:57.35 | dlynes_laptop | while(1) { char *p=strdup("idiot") ; } |
| 06:57.57 | mitcheloc | what would this do: while (1) { break; } ?? |
| 06:58.02 | mitcheloc | would it break my machine? |
| 06:58.10 | Phantom-X | well ill attach festival and its singing mode to asterisk |
| 06:58.10 | dlynes_laptop | mitcheloc: start a loop and exit immediately |
| 06:58.19 | mitcheloc | so it would break my machine? |
| 06:58.25 | CoffeeKid | micheloc: no |
| 06:58.37 | mitcheloc | oh, so why is it called break? |
| 06:58.37 | CoffeeKid | mitcheloc: would basically do nothing, process it 0.00000000000000001 secs :) |
| 06:58.53 | *** join/#asterisk TeePOG (n=1234@196.207.45.253) |
| 06:59.02 | TeePOG | morning |
| 06:59.03 | CoffeeKid | mitcheloc: break means stop doing what your doing in the loop.. "break" out of the loop |
| 06:59.04 | dlynes_laptop | mitcheloc: breaks out of the loop |
| 06:59.43 | mitcheloc | erm uh are you guys sure? i wrote my a program for asterisk and i specifically avoided that command because i didn't want to hurt anyone's computers... =-/ |
| 06:59.45 | CoffeeKid | mitcheloc: what you just said is the same as doing if(1){} |
| 06:59.59 | CoffeeKid | mitcheloc: 100% sure ;) |
| 07:00.28 | CoffeeKid | mitchloc: try this: |
| 07:00.28 | dlynes_laptop | CoffeeKid: he's just fucking with your head |
| 07:00.37 | dlynes_laptop | CoffeeKid: I think he already knows what break does :) |
| 07:00.48 | mitcheloc | it breaks my machine right? |
| 07:00.51 | CoffeeKid | oh, then screw you :P |
| 07:01.06 | mitcheloc | sorry? |
| 07:01.06 | CoffeeKid | no more programming loving :) |
| 07:01.16 | CoffeeKid | nah, no offense taken :) |
| 07:01.24 | mitcheloc | *i don't understand? |
| 07:02.46 | dlynes_laptop | btw, mitch |
| 07:02.50 | CoffeeKid | i love you man. |
| 07:02.56 | dlynes_laptop | you a sales guy at digium, or a coder, or? |
| 07:03.10 | mitcheloc | CoffeeKid, <3 |
| 07:03.38 | TeePOG | did I walk into a beer-drinking, male-bonding session? |
| 07:03.38 | mitcheloc | dlynes_laptop, sorry? no? i'm not? |
| 07:03.38 | TeePOG | ô.) |
| 07:03.38 | mitcheloc | TeePOG, group hug :) |
| 07:03.38 | TeePOG | ô.O |
| 07:03.38 | dlynes_laptop | Jul 07 19:12:01 * mitcheloc (n=mitchelo@gateway.digium.com) has left #asterisk |
| 07:03.39 | CoffeeKid | TeePOG: probably. |
| 07:03.44 | mitcheloc | dlynes_laptop, your a spy! |
| 07:03.48 | dlynes_laptop | heh |
| 07:03.48 | mitcheloc | * you are |
| 07:03.50 | mitcheloc | :) |
| 07:03.53 | TeePOG | mmmkay... too early to drink beer here, but you guys go ahead ok? |
| 07:04.07 | CoffeeKid | TeePOG: i'll drink one for you :) |
| 07:04.25 | dlynes_laptop | mitcheloc: so you're coder then? |
| 07:04.40 | TeePOG | good on yer CoffeeKid mate |
| 07:04.46 | Phantom-X | i could join in beer drinkin |
| 07:04.49 | mitcheloc | dlynes_laptop, oooh, i was drunk and they kindly took me in for the night ;) |
| 07:04.53 | Phantom-X | i ahve to celebrate |
| 07:04.54 | Phantom-X | lol |
| 07:04.55 | dlynes_laptop | ah |
| 07:05.24 | TeePOG | beer drinking and asterisk coding, nice combo |
| 07:06.29 | mitcheloc | dlynes_laptop, i just happened to be a thousand or so miles from california and was in the area so i stopped by to visit and check my irc messages ;) |
| 07:06.56 | CoffeeKid | mitcheloc: lier! :) |
| 07:07.14 | CoffeeKid | *liar |
| 07:07.25 | dlynes_laptop | CoffeeKid: American! |
| 07:07.36 | CoffeeKid | dlynes_laptop: haha |
| 07:08.02 | dlynes_laptop | it wasn't anti-american rhetoric |
| 07:08.11 | dlynes_laptop | coffeekid stepped in his own goo :p |
| 07:08.32 | mitcheloc | CoffeeKid, why am i liar? |
| 07:08.36 | TeePOG | how can jou not love the blood-stained star-spangled banner? *shock&horror* |
| 07:09.04 | mitcheloc | TeePOG....??? |
| 07:09.17 | TeePOG | mitcheloc: irony |
| 07:09.23 | dlynes_laptop | TeePOG: Well, i'm sure some of the innocent souls in the last few years in the middle east could give you a multitude of answers :) |
| 07:09.34 | Phantom-X | all flags are used for war so what ? |
| 07:09.56 | mitcheloc | good for tying splints? |
| 07:09.56 | CoffeeKid | mitcheloc: just teasing :) |
| 07:09.56 | TeePOG | ok guys, seriously, I was commenting on the popular perception of America, not giving my opinion |
| 07:10.20 | Phantom-X | new name for beer and asterisk coding combo : bastardix |
| 07:11.04 | dlynes_laptop | seriously |
| 07:11.09 | dlynes_laptop | i don't give a crap about anything |
| 07:11.18 | dlynes_laptop | except for when slackware 11.0 is coming out |
| 07:11.23 | mitcheloc | CoffeeKid, just teasing, eh? -- http://www.azomed.com/?id=13266 |
| 07:11.41 | Phantom-X | u read art of war by sun tzu and machiavelli.. so i dont really get shocked by the thought of that flags are actually war symbols |
| 07:11.50 | TeePOG | get bastardix! besides Asterisk which you all know and love, it contains a mysql database of every beer in the known world! |
| 07:11.52 | Phantom-X | s/u/I/ |
| 07:12.10 | CoffeeKid | mitcheloc: back at ya ;) |
| 07:12.57 | dlynes_laptop | CoffeeKid, mitcheloc: children....grow up! :) |
| 07:13.18 | mitcheloc | dlynes_laptop, awww, must we daddy??! |
| 07:13.42 | Phantom-X | hello ? *delay* hello ! |
| 07:13.50 | TeePOG | growing old is mandatory. growing up is optional |
| 07:14.04 | CoffeeKid | TeePOG: agreed |
| 07:14.13 | Phantom-X | old weed never gets bad |
| 07:14.37 | CoffeeKid | but old tobbaco does. |
| 07:14.52 | mitcheloc | TeePOG, LAH AAAAIMM!!! |
| 07:15.02 | *** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com) |
| 07:15.37 | Phantom-X | i thought it would be ahrder making asterisk dancing along |
| 07:15.44 | TeePOG | hehehe |
| 07:15.53 | mitcheloc | you dance with your PBX? |
| 07:16.03 | Phantom-X | i direct how it should dance |
| 07:16.03 | TeePOG | hmm, easier said than sung in Russian huh? |
| 07:16.04 | Phantom-X | =) |
| 07:16.53 | mitcheloc | so how about that new enum spec? |
| 07:17.02 | CoffeeKid | dlynes_laptop: hey, back ot being serious :) ... Is having over 95 people on asterisk at the same time too much overload for a xeon duel proc system, 4GB mem? |
| 07:17.04 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
| 07:17.07 | Phantom-X | making it call my cisco ata 186 telephone adapter and use festival to mention its kernel verion |
| 07:17.10 | Phantom-X | lol |
| 07:17.11 | Phantom-X | wasnt that hard |
| 07:17.12 | Phantom-X | =) |
| 07:17.40 | mitcheloc | Phantom-X, next you'll graduate to a weather agi, *that'll be useful* |
| 07:18.00 | Phantom-X | hehe , no ill add my IDS to it |
| 07:18.07 | mitcheloc | IDS? |
| 07:18.14 | Phantom-X | intrusion detection system |
| 07:18.22 | Phantom-X | make it call my cellphone |
| 07:18.30 | TeePOG | Phantom-X: you use Snort? |
| 07:18.36 | Phantom-X | nopes |
| 07:18.39 | Phantom-X | libpcap |
| 07:18.42 | Phantom-X | and own stuff |
| 07:18.50 | Phantom-X | based upon libpcapå |
| 07:18.52 | Phantom-X | based upon libpcap |
| 07:19.12 | Phantom-X | to have ip-less ability to check |
| 07:20.01 | Phantom-X | so you can write own sniffing stuff by suing libpcap |
| 07:20.04 | [hC] | any of you installed a polycom soundstation ip 4000? |
| 07:20.36 | mitcheloc | why would you sue libpcap? |
| 07:20.56 | Phantom-X | hahaha |
| 07:21.03 | Phantom-X | s/sue/use/ |
| 07:24.45 | mitcheloc | everyone went away =/ |
| 07:25.50 | E-bola | Do anybody have a good guide to installing asterix from scratch? Preferbly on debian |
| 07:25.57 | E-bola | i dont need any voice hw, its pure ip-based |
| 07:26.32 | Phantom-X | perhaps u wanna use apt-get ? |
| 07:27.01 | Phantom-X | if u mean configureing |
| 07:27.11 | Phantom-X | thats something else =) |
| 07:30.00 | So3kris | E-bola: http://www.asteriskguru.com/tutorials/asterisk_installation_compilation_debian.html |
| 07:31.40 | bugz | hey! |
| 07:31.46 | bugz | i have a soekris box |
| 07:33.53 | *** join/#asterisk mmmmmToop (n=mmmmToop@196.211.55.101) |
| 07:34.29 | *** join/#asterisk Guest^DJ (i=me@211.24.146.11) |
| 07:34.59 | Guest^DJ | hi all, is there a way to route my calls to other SIP channel if my Zap lines are full automatically ? |
| 07:35.13 | Guest^DJ | out going calls i meant |
| 07:36.09 | *** join/#asterisk cybergypsy (n=mark@APoitiers-156-1-59-100.w86-217.abo.wanadoo.fr) |
| 07:36.53 | So3kris | bugz: cool m2 |
| 07:37.32 | E-bola | thanks so3kris |
| 07:38.44 | So3kris | yw |
| 07:38.59 | E-bola | "...Remember that Asterisk is very proccessor-intensive.." |
| 07:39.10 | So3kris | :D |
| 07:39.20 | E-bola | <PROTECTED> |
| 07:41.17 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 07:42.42 | Phantom-X | is inter astersisk stiff on IAX CPU intensive ? i eman having say a large place u could have an asterisks erver at eacv department serving say 8-16 phones and then use IAX to route to other departments and so |
| 07:42.48 | mmmmmToop | hi...any one here cracked an Asterisk trunk side integration with a Panasonic on a PRI level? |
| 07:43.00 | Phantom-X | so ther ewill be no singe point for all stress |
| 07:43.24 | E-bola | 8-16 phones? |
| 07:43.35 | E-bola | i thought u could put 40 phones easily on an old p3 box? |
| 07:43.43 | Phantom-X | sure u can |
| 07:43.56 | Phantom-X | but iw as thinking of a pålace with say 30 000 ppl |
| 07:44.11 | Phantom-X | s/pålace/place |
| 07:44.21 | E-bola | why have a server for each 16 phones then? |
| 07:44.22 | *** join/#asterisk TeePOG (n=1234@196.207.45.253) |
| 07:44.39 | Phantom-X | because then each deparment han have therir own admin |
| 07:44.47 | Phantom-X | and add their own specific services |
| 07:44.52 | E-bola | ahh ok |
| 07:45.00 | E-bola | i just read on the guide that asterix is very cpu intensive |
| 07:45.12 | E-bola | im planning to use an old dell server, like p3 1,2ghz and 512mb ram |
| 07:45.18 | E-bola | shouldnt that be able to handle 10 or so sip phones |
| 07:45.26 | E-bola | with max 8 concurrent calls? |
| 07:45.28 | Phantom-X | i think so |
| 07:47.38 | drray | transcoding is cpu intensive |
| 07:48.05 | drray | you can make the sip phones not route through asterisk |
| 07:48.05 | Phantom-X | sure |
| 07:48.10 | Phantom-X | but not necesarily |
| 07:48.17 | Phantom-X | depending on your ahrdware |
| 07:48.20 | E-bola | i have to route them through asterix |
| 07:48.31 | E-bola | why would transcoding be needed though? |
| 07:48.47 | E-bola | (sory if its a stupid question im new to asterix) |
| 07:49.07 | Phantom-X | u need to transcode into some kind of coicxe data , unless u have special ahrdware for that |
| 07:49.21 | drray | or if you record voicemails |
| 07:49.24 | drray | or monitor |
| 07:49.49 | drray | I *believe* if you set can reinvite, asterisk will route the call but then let the two sip phones talk to each otehr |
| 07:50.08 | E-bola | dont the sip phones encode the voice data? |
| 07:50.16 | E-bola | i'd asume asterix just passed it on? |
| 07:51.21 | Phantom-X | you know if u go from one protocol to another it could be necesary tto transcode |
| 07:51.47 | *** join/#asterisk L|NUX (n=linux@202.5.145.56) |
| 07:52.21 | E-bola | why would u change protocol in a live conversation? |
| 07:52.32 | Phantom-X | but IF thets a problem i would solve it by clustering several servers together and use IAX to connect them together and make bandwidath balancing in between them |
| 07:53.05 | Phantom-X | and the balancing is depending on their cpu-load and not internet bandwidth |
| 07:53.26 | E-bola | this is for a small company |
| 07:53.33 | E-bola | below 12ppl |
| 07:53.38 | Phantom-X | i wouldnt worry then |
| 07:53.49 | E-bola | so this is more a question of, should i try to get a new box, or will an older server do |
| 07:54.01 | Phantom-X | if its 12 ppl |
| 07:54.04 | drray | an old server will do for a start |
| 07:54.05 | Phantom-X | i think its all ok |
| 07:54.08 | E-bola | ok |
| 07:54.17 | drray | you can always"outgrow" it |
| 07:54.18 | E-bola | drray: ummm either it will do or it wont do |
| 07:54.18 | E-bola | hehe |
| 07:54.20 | Phantom-X | i think ive once heard of someone running in a p3 600 |
| 07:54.21 | Phantom-X | for 40 ppl |
| 07:54.25 | E-bola | my needs wont increase |
| 07:54.30 | E-bola | sounds good |
| 07:54.44 | drray | actually your needs *will* increase as you offer more services |
| 07:55.05 | drray | call parking, meet me |
| 07:55.06 | drray | etc |
| 07:55.16 | E-bola | well isnt it the conversations that require most cpu? |
| 07:55.25 | *** join/#asterisk e-ddie (n=oa@62.61.144.75.generic-hostname.arrownet.dk) |
| 07:55.25 | E-bola | i mean stuff like extra services should have a minimal impact? |
| 07:55.36 | Phantom-X | E-bola: the parts wher eu need rtanscoding |
| 07:55.37 | drray | if you have to transcode a conversation |
| 07:55.56 | Phantom-X | as he told its recording for vvoice mails , if u wanna ahve special services etc |
| 07:56.08 | E-bola | hmm ok, i guess the quick answer is , it shoudl work, but no way of knowing before i get it running |
| 07:56.09 | E-bola | hehe |
| 07:56.10 | drray | set up correctly the sip phones can talk to each other outside of asterisk |
| 07:56.31 | drray | you can run 1-2 sip calls on a wrtg54 router |
| 07:57.21 | Phantom-X | i wouldnt bet on that if 40 ppl wants to check their voicemail at the same moment that it will go on smoothly =) |
| 08:00.22 | drray | I run 96 zap extensions on a 1.8 athelon box |
| 08:00.25 | drray | it's fine |
| 08:00.27 | drray | for light usage |
| 08:00.51 | Phantom-X | ok |
| 08:01.00 | Phantom-X | if i was in charge for large company |
| 08:01.26 | Phantom-X | i would run asterisk on a dedicated server, and then personells might have diskless workstations to boot fgrom this one |
| 08:01.29 | Phantom-X | ans also |
| 08:01.38 | drray | I agree with that |
| 08:01.43 | Phantom-X | i would make sure each workstation has stranscoding for it |
| 08:01.44 | E-bola | ddray: u think it can handle if everyone calls at same time? |
| 08:01.57 | drray | well, I have 9 outside lines on my PRI |
| 08:02.03 | Phantom-X | sowhen someone check their voicemail the file is sent to the workstation and transcoded there |
| 08:02.14 | drray | amd tje most I've ever had was 20 channels at the same time |
| 08:02.20 | Phantom-X | this way the pbx never get unneceseary workload |
| 08:02.42 | drray | phantom - or you could trunk voicemail and have a voicemail only asterisk box |
| 08:02.48 | *** join/#asterisk AltnTab (n=ecs@nrjsoft13.networx-bg.com) |
| 08:02.58 | Phantom-X | drray: i wouldnt |
| 08:03.01 | drray | also you can set which format the voicemail is recorded in, and mail it |
| 08:03.05 | Phantom-X | i would make it ldap |
| 08:03.07 | Phantom-X | for each user |
| 08:03.28 | Phantom-X | ahdn write a client that can support IAX |
| 08:03.37 | drray | you work too hard ;) |
| 08:04.02 | Phantom-X | given that user has access from the ldap then the data will be send to the workstation though IAX andtranscoded locally |
| 08:04.19 | Phantom-X | this way i could manage to serve hundreds of ppl |
| 08:04.25 | Phantom-X | simultaneously |
| 08:04.35 | Phantom-X | with and 'old' machine |
| 08:04.39 | drray | I like the google model myself |
| 08:04.40 | AltnTab | Anyone know provider in Moscow who gives PSTN numbers ? |
| 08:05.01 | Phantom-X | yes the google model is nice |
| 08:05.04 | Phantom-X | having a cluster |
| 08:05.05 | drray | bunch of cheap crappy machines |
| 08:05.17 | Phantom-X | they work |
| 08:05.18 | Phantom-X | =) |
| 08:05.34 | Phantom-X | and dont consume too much electricity compared to newre machines |
| 08:07.32 | e-ddie | i know |
| 08:07.40 | drray | my problem is I have points where I can't be redundant (the zap cards) |
| 08:07.45 | e-ddie | i have a cluster of 150 386dx 20mhz |
| 08:07.50 | drray | not cost effective |
| 08:07.52 | *** join/#asterisk jetaway2009 (n=asd@60.48.162.205) |
| 08:07.59 | Phantom-X | those zqap cards es |
| 08:08.03 | Phantom-X | i use sip only |
| 08:08.15 | Phantom-X | the pstn problems are handed over to others |
| 08:08.15 | *** join/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek) |
| 08:08.22 | Phantom-X | =) |
| 08:08.25 | Phantom-X | in my case |
| 08:08.29 | *** join/#asterisk sob0l (n=sobol@213.173.229.83) |
| 08:08.46 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 08:11.22 | Phantom-X | with the cost of having sip service conected to pstn , i could easily have 10 different accounts for same prices as normal pstn cost |
| 08:11.28 | Phantom-X | at least hee where i live |
| 08:12.15 | Phantom-X | so zeptel cards are almost not intereting here |
| 08:12.35 | drray | I'm in a hotel, and voip to rooms is not practical |
| 08:12.54 | Phantom-X | i understand |
| 08:13.19 | drray | hell, we have such cruddy wiring we use DSL from the units to the "datacenter" |
| 08:13.22 | Phantom-X | and i think using those sip phones are unnecesary to |
| 08:13.25 | drray | being the phone closets |
| 08:13.27 | Phantom-X | since ppl steal those |
| 08:13.54 | Phantom-X | unless you hide them and only have the telephone wire visible |
| 08:14.07 | e-ddie | gotta be careful though |
| 08:14.11 | e-ddie | they might steal the wire |
| 08:14.19 | Phantom-X | or else if u offer internet connectivity over wifi/cable |
| 08:14.31 | Phantom-X | u could ahve the phones that way |
| 08:14.39 | Phantom-X | hehe |
| 08:14.41 | drray | we've never lost a dsl modem |
| 08:14.45 | Phantom-X | steal the wire ? |
| 08:14.56 | drray | wifi is not reliable enough for internet |
| 08:15.14 | drray | our building has metal siding, flourescent lighting, and metal door frames. |
| 08:15.20 | Phantom-X | u can make it more reliable |
| 08:15.43 | drray | not for cheaper than DSL |
| 08:15.45 | Phantom-X | oh u mean signal probs |
| 08:15.48 | drray | yeah |
| 08:15.55 | drray | Hotel Faraday |
| 08:16.10 | Phantom-X | but if u have internet connectivity thought rj45 |
| 08:16.17 | drray | we don't |
| 08:16.17 | Phantom-X | u could have voip there to |
| 08:16.29 | Phantom-X | and also have autoamted services |
| 08:16.39 | Phantom-X | controlled from the palce where the consierge use to be |
| 08:16.39 | Phantom-X | =) |
| 08:17.08 | Phantom-X | like wakiong up by phonecall etc |
| 08:17.30 | drray | that's what we use zaptel for |
| 08:17.33 | Phantom-X | order pronmovies for their tv |
| 08:17.38 | Phantom-X | on the phone |
| 08:17.43 | Phantom-X | thanks to asterisk |
| 08:17.43 | Phantom-X | =) |
| 08:17.47 | drray | sure |
| 08:18.04 | drray | our building was built in 1995 |
| 08:18.18 | drray | it is inexcusable how they did things |
| 08:18.45 | Phantom-X | where do you live ? |
| 08:18.54 | drray | seattle, wa |
| 08:18.57 | Phantom-X | ok |
| 08:19.03 | Phantom-X | i live in sweden |
| 08:19.12 | Phantom-X | we are the phone specialists |
| 08:19.13 | Phantom-X | =) |
| 08:19.25 | Phantom-X | ofcourse the finnish ppl say the same |
| 08:19.26 | Phantom-X | lol |
| 08:19.56 | Phantom-X | but we had ericsson with AXE long before GSM telephony were invented |
| 08:20.25 | Phantom-X | that was technology existig already in 1970-ies |
| 08:20.25 | Phantom-X | =) |
| 08:20.54 | mmmmmToop | has anyone integrated with a Panasonic legacy PBX before? |
| 08:20.56 | Phantom-X | sothose super PBX:es ericsso sells all over the world were specified already in the 70-ies |
| 08:20.57 | Guest^DJ | hi all, is there a way to route my calls to other SIP channel if my Zap lines are full automatically ? |
| 08:22.39 | mmmmmToop | not sure if there is a better way to do the overflow routing...but we wrote a script to do a 'show channels' and then count them & the pass onto another trunk... |
| 08:23.00 | e-ddie | heheh |
| 08:23.32 | *** join/#asterisk TeeBoo (n=thibault@pai34-3-82-231-33-117.fbx.proxad.net) |
| 08:23.42 | e-ddie | Phantom-X: that's why the ericsson z800 sucks balls? |
| 08:24.03 | Phantom-X | e-ddie: i dont use ericcson celphones =)) |
| 08:24.12 | e-ddie | Phantom-X: like, it works usable till you get to the western part of denmark |
| 08:24.15 | e-ddie | heheh |
| 08:24.19 | e-ddie | me neither |
| 08:24.31 | e-ddie | i dont use swedish technology, if i can |
| 08:24.39 | Phantom-X | e-ddie: well their cellphones maybe arent as good as the PBX systems they make etc |
| 08:25.12 | jbalcomb | swiss army knives, swiss watches, and swiss bicycles are all excellent |
| 08:25.13 | e-ddie | no, they arent |
| 08:25.20 | jbalcomb | oh, and so are swiss people |
| 08:25.26 | Phantom-X | e-ddie: hehe well u got to know ther eis diff between mainsrteam tech as cellphones and military tech or heavy duty tech =) |
| 08:25.27 | e-ddie | jbalcomb: but not swedish :d |
| 08:25.33 | drray | I had the best time in Bern |
| 08:25.54 | Zeeek | hey now.... |
| 08:26.06 | e-ddie | Phantom-X: yeah... and i'm not impressed of the umts antennas from ericsson either |
| 08:26.08 | jbalcomb | is it odd that i confuse the idea that swiss and swede are not the same thing? |
| 08:26.10 | Zeeek | anyone seen this? (or not) http://www.krugle.com/ |
| 08:26.16 | *** join/#asterisk ghenry (n=ghenry@ghenry.plus.com) |
| 08:26.34 | e-ddie | jbalcomb: you wouldnt, if you'd met 1 swiss and 1 swede |
| 08:26.35 | e-ddie | :D |
| 08:26.42 | Phantom-X | e-ddie: the military industry producte ericcson makes are far better |
| 08:26.43 | Phantom-X | =) |
| 08:26.48 | e-ddie | i guess |
| 08:26.52 | jbalcomb | i'm pretty sure i have, just not at the same time |
| 08:26.55 | ghenry | On our TDM400P module 3 fxo (incoming line) port, we show it offhook, using zap show channel 3 |
| 08:27.03 | Phantom-X | ther they are top notch |
| 08:27.04 | ghenry | Hookstate (FXS only): Offhook |
| 08:27.13 | e-ddie | Phantom-X: but if they only make good products for the military, then they can suck my balls |
| 08:27.14 | Phantom-X | sells to every sucker in the world |
| 08:27.15 | Phantom-X | lol |
| 08:27.19 | ghenry | There are no calls being placed or received at the moment |
| 08:27.23 | ghenry | could it be stuck? |
| 08:27.44 | e-ddie | i'm stuck in the interweb |
| 08:28.02 | drray | ghenry - how long has it been open? |
| 08:28.03 | ghenry | why would it not have hangup? |
| 08:28.04 | E-bola | is 1-2.9.1 the latest stable build? |
| 08:28.09 | Zeeek | ghenry that's normal |
| 08:28.15 | e-ddie | Phantom-X: i love it, how some swedish guys' look remind me of emil :D |
| 08:28.20 | drray | it would not hang up if it could not detect the line being closed |
| 08:28.27 | ghenry | ah drray |
| 08:28.36 | Phantom-X | e-ddie: really ? |
| 08:28.38 | ghenry | how can I tell how long the zap chan has been open for dreamless |
| 08:28.41 | Zeeek | fxo is normally offhook |
| 08:28.42 | ghenry | how can I tell how long the zap chan has been open for drray |
| 08:28.45 | e-ddie | Phantom-X: yeah... makes me laugh every time |
| 08:28.52 | drray | zeek ? |
| 08:29.01 | Phantom-X | e-ddie: i look like a pigmented kaucasion |
| 08:29.07 | Phantom-X | caucasian myself |
| 08:29.09 | ghenry | Zeeek: Ah, yes, I see that. |
| 08:29.13 | jbalcomb | was beaker swedish or swiss? |
| 08:29.15 | e-ddie | heheh |
| 08:29.18 | e-ddie | i'm just bloody white |
| 08:29.21 | Phantom-X | perhaps typical icelander |
| 08:29.24 | Zeeek | let me put it this way.... on MY box, the first two channels are FXO and they are offhook right now |
| 08:29.36 | drray | mine are never off hook |
| 08:29.37 | Zeeek | while the FXS are onhook |
| 08:29.39 | Phantom-X | many ppl say that i look icelandic |
| 08:29.43 | e-ddie | Phantom-X: you're from iceland? |
| 08:29.44 | drray | zeek - where are you? |
| 08:29.48 | e-ddie | Phantom-X: ah |
| 08:29.51 | jbalcomb | woot. i look dutch. |
| 08:29.54 | Phantom-X | and yes may be so ,s ince im norwegian.. and i have relatives in iceland to |
| 08:29.54 | Phantom-X | lol |
| 08:29.56 | e-ddie | Phantom-X: better than looking like a swede :D |
| 08:29.59 | ghenry | yeah, that looks right. |
| 08:30.01 | Zeeek | drray what hardware? I'm X100P in Paris |
| 08:30.07 | e-ddie | Phantom-X: hehe.. i'm norwegian too |
| 08:30.08 | ghenry | both our fxo ports are offhook |
| 08:30.13 | drray | I'm a tor2 in seattle |
| 08:30.14 | jbalcomb | Zeeek sorry about the world cup |
| 08:30.21 | Zeeek | join #asterisk-myspace |
| 08:30.23 | ghenry | fxs is onhook |
| 08:30.25 | Phantom-X | e-ddie: cool |
| 08:30.32 | E-bola | is 1-2.9.1 the latest stable build? |
| 08:30.34 | TeePOG | sheesh, a whole new form of ASL? |
| 08:30.39 | Phantom-X | sogn & fjordane orignally |
| 08:30.44 | Zeeek | I'm not into sports but it was a sad end to a career |
| 08:30.47 | e-ddie | TeePOG: ASL? pls? |
| 08:31.13 | Zeeek | then only good thing about the sport is that if you took away everyone of African descent, there'd be NO TEAM :) |
| 08:31.16 | jbalcomb | Zeeek indeed. i'm not into sports but i am into soccer. |
| 08:31.21 | TeePOG | lol! <Zeeek> drray what hardware? I'm X100P in Paris <drray> I'm a tor2 in seattle |
| 08:31.28 | *** join/#asterisk my007ms (n=noor@217.139.224.194) |
| 08:31.33 | my007ms | hello all |
| 08:31.37 | Zeeek | sleepless in Seattle apparaently |
| 08:31.58 | Zeeek | We can all join an SMS chatline then |
| 08:32.06 | jbalcomb | TeePOG: I'm coming to Pretoria, SA in 2010!! |
| 08:32.25 | my007ms | when i try to make zaptel it's end with exit |
| 08:32.27 | my007ms | make[2]: *** [/usr/src/zaptel-1.2.6/zaptel.o] Error 1 |
| 08:32.27 | my007ms | make[1]: *** [_module_/usr/src/zaptel-1.2.6] Error 2 |
| 08:32.27 | my007ms | make[1]: Leaving directory `/usr/src/kernels/2.6.9-34.0.2.EL-smp-i686' |
| 08:32.27 | my007ms | make: *** [linux26] Error 2 |
| 08:32.32 | my007ms | sorry |
| 08:32.41 | my007ms | i think i have to use paspian |
| 08:32.42 | Zeeek | ghenry in fact, I just noticed Hookstate (FXS only) |
| 08:32.50 | Zeeek | so it's meaningless in FXO |
| 08:33.01 | TeePOG | jbalcomb: the way they're behind with the preparations, the world cup is happening in 2012 |
| 08:33.21 | Zeeek | when the global warming will be at 2°/year? |
| 08:33.45 | jbalcomb | TeePOG: :( i hope its all well when i get there. i find it hard to plan 4 years in advance, let alone 6 |
| 08:34.23 | E-bola | what is libpri? |
| 08:34.27 | jbalcomb | teepog: a lady i work with is from Pretoria and a Tunsanian friend of mine is going there next year for grad school |
| 08:34.31 | E-bola | its just mentioned to install in the basic installation |
| 08:34.40 | E-bola | i know i dont need the zaptel stuff, buyt cant figure out what libpri is |
| 08:34.49 | drray | libpri is for the zaptel stuff |
| 08:34.52 | drray | tdm |
| 08:34.53 | jbalcomb | its the libraries for Prime(Primary) Rate ISDN |
| 08:34.54 | TeePOG | cool jbalcomb, where are you from? |
| 08:35.01 | *** join/#asterisk X-Rob (n=rob@CPE-58-164-139-115.qld.bigpond.net.au) |
| 08:35.03 | jbalcomb | TeePOG: Cleveland, Ohio, USA |
| 08:35.17 | Zeeek | you don't need libpri if no PRI |
| 08:35.21 | E-bola | jbalcomb: so in a pure ip setup, its not needed? |
| 08:35.34 | TeePOG | cool jbalcomb |
| 08:35.35 | jbalcomb | E-bola: As I understand it, that is correct. |
| 08:35.40 | E-bola | thank you |
| 08:35.48 | drray | no zap (or ztdummy) no libpri |
| 08:35.59 | *** join/#asterisk foRza (n=tMs@firewall.hikt.no) |
| 08:36.00 | Zeeek | except for the dummy timer for meetme |
| 08:36.08 | jbalcomb | TeePOG: thanks. ;) it's nice to think so everyone once in a while, being American and all. |
| 08:36.28 | TeePOG | jbalcomb: we have to organise trading of some distros at that time |
| 08:36.33 | *** part/#asterisk foRza (n=tMs@firewall.hikt.no) |
| 08:36.39 | jbalcomb | TeePOG: do you speak the strange british/ausie english? |
| 08:37.09 | *** join/#asterisk foRza (n=tMs@firewall.hikt.no) |
| 08:37.11 | *** join/#asterisk Drenec (n=devilkil@port-212-202-210-59.dynamic.qsc.de) |
| 08:37.32 | TeePOG | jbalcomb: it's british english |
| 08:38.00 | jbalcomb | TeePOG: also, is it true that there is a edge to comparing SA to Australia? |
| 08:38.05 | TeePOG | and anyway, my previous boss was from Portland, Oregon... so i can drawl with the rest of them |
| 08:38.31 | jbalcomb | TeePOG: nice |
| 08:38.34 | TeePOG | an edge, jbalcomb? i don't understand the reference |
| 08:39.18 | jbalcomb | TeePOG: i told the lady here that her accent sounded like a cross between British English and Australian English |
| 08:39.50 | TeePOG | it's South African English, an accepted form of the language. Like Strine in Oz |
| 08:39.52 | jbalcomb | TeePOG: She gave me a look and said that it wasn't anything like Australian |
| 08:40.26 | *** join/#asterisk zepmantra (i=what@203.215.100.96) |
| 08:40.51 | TeePOG | personally, jbalcomb, with my knowing Russian, it's a lot closer to sounding like Russian than to Aussie english |
| 08:42.08 | TeePOG | just my opinion though |
| 08:42.20 | jbalcomb | TeePOG: interesting. i'll have to listen more closely next time I talk with Ilya ('the russian guy') |
| 08:42.32 | my007ms | i get error i try install zaptel |
| 08:42.32 | my007ms | /usr/src/zaptel-1.2.6/zaptel.c:384: error: `RW__LOCK_UNLOCKED' undeclared here (not in a function) |
| 08:42.49 | my007ms | any idea ?? |
| 08:43.11 | jbalcomb | TeePOG: how long you been working with Asterisk? |
| 08:43.32 | TeePOG | but my English friends generally remark on how much better English I speak than most Americans or South African English speakers [English is not my mother tongue, but I've been steeped in it since a tender age] |
| 08:43.44 | TeePOG | jbalcomb: about 3 weeks now :-D |
| 08:43.49 | Zeeek | how do I find out who owns an 800 number - the provider I mean? |
| 08:44.10 | jbalcomb | TeePOG: haha.. ok, thats a short enough time. new job? |
| 08:44.15 | TeePOG | lol Zeeek, and /whois 800 doesn't work? |
| 08:44.41 | jbalcomb | Zeeek: call your telco and ask them |
| 08:44.46 | TeePOG | no jbalcomb, used to just admin the ipcop firewall and a couple mandriva desktops, still quite green here |
| 08:45.13 | Zeeek | my telco is France Télécom - they don't know their own numbers let alone US ones |
| 08:45.27 | jbalcomb | TeePOG was the asterisk set up already there or are you setting it up? |
| 08:45.54 | jbalcomb | Zeeek: how do you make those neat e' characters? |
| 08:46.40 | *** join/#asterisk ghenry (n=ghenry@ghenry.plus.com) |
| 08:46.41 | ghenry | ls |
| 08:46.43 | ghenry | HI me again |
| 08:46.46 | TeePOG | jbalcomb: i tried to do it on my mandriva samba fileserver, but usage was too heavy and affected performance, so i got xorcom-rapid distro and gave it its own box |
| 08:47.26 | ghenry | Why would ZAP/3 hang? A restart of asterisk has fixed it. How can I debug the cuase of the hang? |
| 08:47.43 | Zeeek | Ôn my kéyböàrd |
| 08:48.20 | jbalcomb | Zeeek that is too cool. can you mail me a french keyboard? American/US keyboards are as narrow minded as most of the people who use them. ;) |
| 08:48.22 | Zeeek | ghenry disconnecting and reconnecting a line sometimes does this (FXS as well) and power cuts/restore |
| 08:48.27 | TeePOG | jbalcomb: it's a french keyboard Zeeek is using, i think |
| 08:48.35 | ghenry | Zeeek: Ah |
| 08:48.52 | ghenry | Zeeek: On ASterisk of normal PBX systems? |
| 08:48.56 | Zeeek | jbalcomb you can install the virtual kbd on linux or windoze |
| 08:49.11 | Zeeek | ghenry is this #normal-pbx-systems? |
| 08:49.11 | TeePOG | lol jbalcomb: you do get stickers you put on... then you load the french kbd layout and use the stickers |
| 08:49.19 | *** join/#asterisk mosty (i=mostynm@60-241-198-194.static.tpgi.com.au) |
| 08:49.34 | jbalcomb | I want a Japanese keyboard too. It drives me batty trying to make all the syllables<SP> |
| 08:49.36 | ghenry | Zeeek: Fair enough. |
| 08:50.01 | Zeeek | ??·?????????????????????????? |
| 08:50.15 | Zeeek | that was japanese but I don't see it |
| 08:50.24 | jbalcomb | yeah, thats whack. |
| 08:50.25 | ghenry | So, how can I tell is that line has hung/stuck? |
| 08:50.49 | e-ddie | 10:44 <Zeeek> ??·?????????????????????????? |
| 08:50.50 | jbalcomb | I'd rather run VMWare with windows/linux installations for each language |
| 08:50.59 | e-ddie | very nice |
| 08:51.03 | jbalcomb | ghenry: did you check in the logs? |
| 08:51.18 | ghenry | ah, no. I forgot about all the logs!!!! |
| 08:51.20 | ghenry | Cheers |
| 08:51.23 | Zeeek | ghenry I have repeating issues with FXS lines that degrade and come back with a zaptel re-install (not make, modules) but only once was there an FXO problem. They froze open upon a power loss. MOdems sometimes do this |
| 08:51.38 | ghenry | Ah |
| 08:51.46 | ghenry | its a TDM400p card our client has |
| 08:51.47 | jbalcomb | umm... is that guy jerking my leg? |
| 08:52.02 | Zeeek | beteer your leg than other parts |
| 08:52.09 | ghenry | I can't believe I forgot about the logs |
| 08:52.11 | ghenry | bb. |
| 08:52.12 | ghenry | bbl |
| 08:52.14 | ghenry | chaeers |
| 08:52.16 | jbalcomb | depends on how hard hes jerking..... |
| 08:52.29 | jbalcomb | ewe. |
| 08:52.34 | *** join/#asterisk tparcina (n=tparcina@lns02-0599.dsl.iskon.hr) |
| 08:52.36 | Zeeek | I have a zaptel reload script ready now |
| 08:52.50 | jbalcomb | anyway, yeah, so fun stuff happens in here at 5 AM i see |
| 08:52.52 | tparcina | hi channel! |
| 08:53.06 | Zeeek | in fact I could cron it for nightly run but I'm trying to see what the repeat time is. |
| 08:53.15 | jbalcomb | Zeeek: does it do something like `asterisk -rx "reload zaptel"` |
| 08:53.29 | Zeeek | jbalcomb no |
| 08:53.43 | jbalcomb | chan_ |
| 08:53.45 | Zeeek | it remmod and modprobes |
| 08:53.49 | jbalcomb | ah. |
| 08:53.57 | tparcina | jbalcomb, earth is circle (is it right word in english?) you know ;)) (5AM) |
| 08:54.00 | hads|home | jbalcomb: I guess it all depends what timezone you are in. |
| 08:54.14 | jbalcomb | i have fun with that sometimes when we can get incoming calls but cant make out going.. |
| 08:54.15 | Zeeek | earth is flat - or a sphere |
| 08:54.28 | jbalcomb | or when all my channels are up except my D channels |
| 08:54.38 | jbalcomb | ovular? |
| 08:54.51 | Zeeek | jbalcomb our symptom is noisy FXS. It's the same phone too, I changed the modules and lines around |
| 08:54.57 | jbalcomb | tparcina "The earth is round." |
| 08:55.10 | Zeeek | only in 2D |
| 08:55.18 | tparcina | jbalcomb: thank you :)) |
| 08:55.45 | Zeeek | anyone with nufone these days? |
| 08:55.55 | jbalcomb | uh uh.. things are round in 3d as well but the term sphere is symantically more accurate and precise |
| 08:56.02 | jbalcomb | tparcina sho nuff |
| 08:56.18 | jbalcomb | s/sho nuff/sure enough/ |
| 08:56.20 | *** join/#asterisk Sonderblade (n=mah@static-213.131.147.169.addr.tdcsong.se) |
| 08:56.24 | Zeeek | Is it Hawaii or Havaii ? |
| 08:56.26 | jbalcomb | s/sure enough/your welcome/ |
| 08:56.31 | Zeeek | Havaii |
| 08:56.34 | Zeeek | thanks. |
| 08:56.37 | jbalcomb | Zeeek depends on your accent? |
| 08:56.38 | Zeeek | you're velcome |
| 08:56.58 | jbalcomb | habaii |
| 08:57.07 | Zeeek | so hows's 2.4 ? Rocks? |
| 08:57.14 | Zeeek | erm... 1.2 |
| 08:57.20 | *** join/#asterisk l-fy (n=pchitesc@yate/developer/l-fy) |
| 08:57.24 | Zeeek | no, 1.4 |
| 08:57.30 | Zeeek | I meant 1.4!!! |
| 08:57.32 | l-fy | where is 1.4? |
| 08:57.42 | tzafrir | in the future? |
| 08:57.46 | jbalcomb | it seems |
| 08:57.48 | Zeeek | end of July it should be out afaik |
| 08:57.59 | mosty | i have a problem with ROIC, when i turn off busydetect and turn on hanguponpolarityswitch, sometimes * hangs up for no reason about a second into the call |
| 08:58.04 | tzafrir | the beta of it? |
| 08:58.05 | Zeeek | since we're close i thought someone may have had it |
| 08:58.05 | l-fy | well, ok |
| 08:58.07 | jbalcomb | yeah, and the sweet new gxp-2000 firmware |
| 08:58.22 | l-fy | jbalcomb > gxp2000 can be fixeD? |
| 08:58.27 | jbalcomb | and perhaps the firmware for the gxp-2000 caddy |
| 08:58.33 | Zeeek | tzafrir no, the shareware version :) |
| 08:59.01 | jbalcomb | l-fy: somewhat. i'm checking out the latest firmware now. supposedly some audio improvements. |
| 08:59.10 | l-fy | wow, super cooo |
| 08:59.11 | l-fy | l |
| 08:59.24 | jbalcomb | l-fy: personally i think the hardware is cheap so theres only so much to do. |
| 08:59.45 | Zeeek | people alwasy bitch about the quality of GS yet I talk to two people often who have BT101 a,d they sound fine. |
| 09:00.03 | Zeeek | they sound better than I type even |
| 09:00.06 | jbalcomb | l-fy: it would be nice if they would fix retard things like documenting that TFTP is the preffered method for config/firmware updates but making the phone default to HTTP |
| 09:00.06 | l-fy | jbalcomb > gxp has a software issue |
| 09:00.12 | l-fy | i have an gxp here also |
| 09:00.18 | jbalcomb | i have 100+ |
| 09:00.35 | Zeeek | so they're not as bad as folks say? |
| 09:00.38 | l-fy | Zeeek > i have 2 gs and sound lauzy |
| 09:00.45 | l-fy | one bt and a gxp |
| 09:00.56 | l-fy | and if you access the website while talking, welll |
| 09:01.04 | jbalcomb | i've had to swallow my foot hole and walk on my ass cheeks to get these phones somewhat automated |
| 09:01.05 | l-fy | it even has small problems on playing his own tone |
| 09:01.13 | jbalcomb | s/hole/whole |
| 09:01.27 | Zeeek | the BT is a little klunky but I had very good luck with them. Until one just stopped registering and never worked again with any firmware version |
| 09:01.31 | mmmmmToop | anyone in Johannesburg/ Pretoria looking for a Asterisk job...we are hiring...? |
| 09:02.05 | jbalcomb | l-fy sounds like some asterisk issue as well. i took over the setup here about 7 months ago and was able to improve the sound quality on the gxp-2000 about 65% |
| 09:02.06 | Zeeek | post on the user list maybe? |
| 09:02.16 | l-fy | jbalcomb > i don't use asterisk |
| 09:02.23 | jbalcomb | l-fy: just on network, server, and line based issues |
| 09:02.24 | Zeeek | jbalcomb give us your secrets! |
| 09:02.36 | l-fy | and i know for sure that my software dosen't have an issue |
| 09:02.40 | jbalcomb | Zeeek: ztmonitor is usefull |
| 09:02.51 | jbalcomb | as is calling your telco and having them test the lines intrusively |
| 09:02.54 | l-fy | jbalcomb > we had problems on even his own tone |
| 09:02.54 | Zeeek | yes, levels cause a lot of problems, especially echo |
| 09:02.57 | l-fy | not on the network |
| 09:03.18 | jbalcomb | don't load services or modules you dont need |
| 09:03.22 | hads|home | ztomonitor has nothing to do with the audio quality of the phones though.. |
| 09:03.30 | Zeeek | it does with zaptel |
| 09:03.42 | hads|home | _of the phones_ |
| 09:03.45 | Zeeek | bad settings can make *any* phone sound like shit |
| 09:03.49 | jbalcomb | setup MRTG and monitor your network traffic, check latency from end to end on stations that have extra trouble |
| 09:03.51 | cybergypsy | anyone got callerid working with a x100p in france ? |
| 09:03.59 | Zeeek | cybergypsy yes |
| 09:04.07 | Zeeek | are you paying FT for the service? |
| 09:04.22 | cybergypsy | no idea - the wife looks after that |
| 09:04.29 | cybergypsy | i doubt it if its an extra |
| 09:04.32 | Zeeek | gotta go strain the potatoes... brb |
| 09:04.49 | Zeeek | cybergypsy it IS A PAID service - Ieu/month |
| 09:05.01 | cybergypsy | ahhhh - that would explain it then |
| 09:05.04 | Zeeek | and the number is one price and the name lookup is extra |
| 09:05.05 | cybergypsy | thanks Zeeek |
| 09:05.15 | cybergypsy | what a rip |
| 09:05.18 | Zeeek | see their excellent web site (NOT!!!!) |
| 09:05.20 | jbalcomb | mmmmmToop: Can I have your Asterisk job for about 4 weeks during the summer of 2010? |
| 09:05.35 | cybergypsy | they even charge you to see what numbers you have called on your bill ? |
| 09:05.38 | Zeeek | cybergypsy go with wengo - 7€/month callerid and all the services |
| 09:05.55 | Zeeek | cybergypsy no detailled buill is free but you miust ask for it |
| 09:06.12 | Zeeek | s/bull/bill/ |
| 09:06.20 | my007ms | i have 3 TDM card and one TE205P |
| 09:06.20 | Zeeek | s/buill/bill/ |
| 09:06.26 | Zeeek | facture détaillée quoi |
| 09:06.37 | Zeeek | pipi |
| 09:06.40 | my007ms | i see three Tiger Jet Network Inc in my lspci |
| 09:06.49 | jbalcomb | Je suis une petite fille, mettez vos mains sur moi |
| 09:06.50 | *** join/#asterisk Gunnar (n=gunnar@62.97.242.6) |
| 09:06.50 | cybergypsy | ok , thanks guys |
| 09:06.52 | my007ms | how can i make sure that i see TE card |
| 09:08.01 | E-bola | enum.c:133: warning: pointer targets in passing argument 1 of âparse_ieâ differ in signedness |
| 09:08.01 | E-bola | enum.c:140: warning: pointer targets in passing argument 1 of âparse_ieâ differ in signedness |
| 09:08.01 | E-bola | enum.c:147: warning: pointer targets in passing argument 1 of âparse_ieâ differ in signedness |
| 09:08.07 | E-bola | should i be worried about that when compiling? |
| 09:08.10 | Zeeek | jbalcomb where've you been hanging out? pr0ncity.com ? |
| 09:08.13 | *** join/#asterisk MatsK (n=mats@141.221.181.62.in-addr.dgcsystems.net) |
| 09:08.19 | jbalcomb | Zeeek: Ottowa, CA |
| 09:08.21 | jbalcomb | ;) |
| 09:08.32 | mmmmmToop | my007ms: that is perfecty normal...u should see the Tiger Jet for TDM cards... |
| 09:08.42 | jbalcomb | Va-t'ens mon ostie! Salut la visite! =) |
| 09:09.27 | Zeeek | 02:09.0 Communication controller: Tiger Jet Network Inc. Intel 537 |
| 09:09.44 | jbalcomb | Zeeek: I have one of those as well |
| 09:09.53 | *** join/#asterisk qdk (n=qdk@213.237.44.34) |
| 09:10.01 | Zeeek | that's the X100P |
| 09:10.19 | Zeeek | 02:0e.0 Network controller: Tiger Jet Network Inc. Intel 537 |
| 09:10.28 | Zeeek | and that is the TDM400P |
| 09:10.33 | jbalcomb | Zeeek: ah, I took that out four days ago. I'll miss seeing it in my lspci |
| 09:10.47 | Zeeek | which, the TDM4? |
| 09:11.16 | jbalcomb | no, the single line version |
| 09:11.32 | MatsK | I have a dialplan challange with pattern matching, Isn't a _! in extension.cong going to match any letter or digits ? |
| 09:12.14 | jbalcomb | we had a four line card for two PRI lines and the jacknut I replaced installed a single line card to run to our analog PBX for the fax machines. :/ |
| 09:12.40 | Zeeek | jbalcomb ironically, I've had a couple of FXO mods for the TDM400 but while things are working so well I decided not to pull out the two FXO cards |
| 09:12.55 | hads|home | MatsK: I don't thing _! will match anything. |
| 09:13.19 | jbalcomb | MatsKl: i agree that it would seem to be a bad regex |
| 09:13.36 | MatsK | hads|home: OK, but I tried with _. also and no luck, it only matches digits |
| 09:13.39 | my007ms | how can i make sure that my pc se the TE205p card |
| 09:13.40 | jbalcomb | MatsK: how about _. instead? |
| 09:13.41 | mmmmmToop | jbalcomb: sure no problem...you have the job ; ) |
| 09:13.45 | my007ms | i make sure form TDM |
| 09:13.56 | hads|home | _X will match a digit. |
| 09:14.10 | my007ms | mmmmmToop, what abut TE205P how it will be in lspci |
| 09:14.13 | Zeeek | MatsK what are you trying to match? |
| 09:14.38 | MatsK | I try to match sip addresses like mats@rebtel.com |
| 09:14.39 | jbalcomb | mmmmmToop: awesome!! 160 hours of international asterisk voip consultation at the nominal rate of $300/HR USD. =) |
| 09:14.55 | mmmmmToop | my007ms: think they are all the same...doesn't matter what modules you have installed |
| 09:15.12 | MatsK | and in sip.conf I have directed that domain into a context where I tri to make a reg exp |
| 09:15.25 | jbalcomb | mmmmmToop: $300/HR EU if that'd be easier for exchange purposes. ;) |
| 09:15.26 | mmmmmToop | jbalcomb: sure why suck a cheap hourly rate...you can't be verry good then!! ; ) |
| 09:16.10 | jbalcomb | mmmmmToop: well, it's normally $900/HR but I figured since your in here I'd give you the good referal discount. |
| 09:16.38 | jbalcomb | mmmmmToop: Plus I'm going to be at soccer games most of those hours so it seemed more fair. ;) |
| 09:16.40 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 09:16.45 | mmmmmToop | ;) |
| 09:17.20 | MatsK | if I uses a direct match like "exten => mats,1,Dial" that works but regexps doesn't |
| 09:17.32 | jbalcomb | mmmmmToop: not to worry though, my PDA has cellular internet access, SSH, and RDC |
| 09:18.26 | *** join/#asterisk kanelbullar (n=kanelbul@83.240.200.92) |
| 09:18.34 | jbalcomb | i wonder if they make a sip phone client for the windows mobile... |
| 09:18.49 | MatsK | no regexp experts here ? |
| 09:19.15 | jbalcomb | it be wierd if i could use my cellular internet connection on my pda to use it as a sip phone |
| 09:19.20 | hads|home | MatsK: I'm not sure about matching letters. You could try _[a-z] |
| 09:19.46 | jbalcomb | MatsK: you are looking for regex as it pertains specifically to asterisks dial plan? |
| 09:19.49 | MatsK | hads|home: Ahh, wait I will |
| 09:19.49 | my007ms | mmmmmToop, then how i know i don't have irq conflict |
| 09:20.19 | mmmmmToop | cat /proc/interupts |
| 09:20.32 | Zeeek | lspci |
| 09:20.37 | *** join/#asterisk Assid (i=assid@203.115.83.214) |
| 09:20.49 | jbalcomb | my007ms: lspci -v or -vb i think will show all your devices and their IRQ addresses. If any have the same number then they are /conflicting/ |
| 09:22.12 | my007ms | dam :D i have very big conflicting all my TDM in IRQ 11 |
| 09:22.21 | my007ms | this is so bad |
| 09:22.22 | MatsK | hads|home: Noop, _[a-z] doesn't work at all |
| 09:23.06 | hads|home | MatsK: OK, sorry I'm out of ideas for matching letters |
| 09:23.16 | MatsK | jbalcomb: Yes thats true |
| 09:23.26 | Zeeek | how about _.! |
| 09:23.33 | MatsK | hads|home: I think I will have to file a bug report on this |
| 09:24.08 | *** join/#asterisk thygar123 (n=bla@196.211.12.34) |
| 09:25.23 | MatsK | Zeeke: Noop a _.! doesn't do it either |
| 09:25.29 | thygar123 | hi wonder if any 1 could help me i'm using asterisk business edition , when some one phones and gets put on hold then trannsfered it does not hang up the zap channel at the end of the call |
| 09:26.04 | Zeeek | MatsK if you're in a hurry you may need to use 's' and write the logic |
| 09:26.04 | hads|home | thygar123: What zap interface are you using? Also, ABE comes with support doesn't it? |
| 09:26.30 | thygar123 | yes it does was just wondering if any 1 knew |
| 09:26.57 | E-bola | i cant find a good howto on configuring asterisk |
| 09:26.59 | E-bola | the wiki |
| 09:27.06 | E-bola | refers to a what is channels when is tart |
| 09:27.06 | thygar123 | TDM2400p |
| 09:27.12 | E-bola | do any1 have a good starting point? |
| 09:27.18 | Zeeek | yes: http://automated.it/guidetoasterisk.htm |
| 09:27.19 | E-bola | ive just compile dthe newest aterisk and it starts up fine |
| 09:27.23 | Zeeek | http://asteriskdocs.org |
| 09:27.28 | hads|home | thygar123:You may not be getting disconnect supervision from your telco. |
| 09:27.46 | Zeeek | E-bola did you write on the docs list yesterday? |
| 09:27.52 | thygar123 | hads|home how do i work around it ? |
| 09:27.52 | E-bola | Zeek: nope |
| 09:27.54 | MatsK | Zeeke: OK I will dig around on that "s" and see if it's feasible ... |
| 09:28.25 | Zeeek | E-bola you'll find startup info in the links above |
| 09:28.33 | E-bola | ok im browsing them now... |
| 09:28.34 | Zeeek | MatsK 's' may be your only choice |
| 09:28.35 | hads|home | thygar123: Ask the telco for disconnect supervision on your lines would be the best way. |
| 09:28.47 | E-bola | hmm the http://asteriskdocs.org |
| 09:28.47 | E-bola | <f~hads|home> thygar123:You may n |
| 09:28.57 | E-bola | sory, pasted a line too much, but the site times out |
| 09:29.00 | E-bola | can u access it? |
| 09:29.02 | thygar123 | hads|home lol i'm from SA we get lines from Telkom and they are crapp |
| 09:29.14 | Zeeek | no it appears to be down for now, but thatis very rare |
| 09:29.26 | MatsK | Zeeke: OK now I see what you are directing me at, thats a solution that works for me, thx |
| 09:29.33 | Zeeek | E-bola there are PDF docs you can download - a whole BOOK on how to set up asterisk |
| 09:29.38 | MatsK | But I will file a bug anyway |
| 09:29.42 | Zeeek | MatsK great! |
| 09:29.53 | E-bola | ile start poking through sip.conf i guess |
| 09:29.53 | Zeeek | good idea, they should know |
| 09:29.57 | hads|home | thygar123: Other than that you could try busydetect=yes in zapata.conf but it isn't nearly as good a solution |
| 09:30.07 | Zeeek | yes: http://automated.it/guidetoasterisk.htm |
| 09:30.13 | Zeeek | E-bola start here ^^^^^^ |
| 09:30.20 | hads|home | thygar123: http://www.voip-info.org/wiki-Asterisk+Disconnect+Supervision |
| 09:30.39 | thygar123 | hads|home yea i've done that my problem is that the lines do diconnect some times it's just when they where put on hold for a while |
| 09:30.58 | mosty | is Answer the first command that should be run in the context that zapata.conf sends incoming zap calls to? |
| 09:31.10 | *** join/#asterisk darviria (n=dvr@194-105-181-29.ifb.co.uk) |
| 09:31.15 | Zeeek | mosty Answer is rarely needed |
| 09:31.36 | hads|home | thygar123: Getting disconnect supervision from your telco may be the only solution. |
| 09:32.04 | thygar123 | hads|home thanks for the info will try |
| 09:32.14 | E-bola | zeek: seems to conver installation? |
| 09:32.16 | hads|home | thygar123: No probs |
| 09:32.23 | Zeeek | E-bolaa |
| 09:32.29 | E-bola | i need something walking you through setting up the actual pbx configuration |
| 09:32.34 | mosty | zeeek: well sometimes the * console shows the first command in that context (which isn't Answer) then immediately hangs it up, but the call keeps ringing on the callers side, * then picks up again |
| 09:32.41 | Zeeek | E-bola please read the table of contents and go read the parts about configuring |
| 09:33.20 | E-bola | i did, its about the zaptel stuff, mine is purely sip based |
| 09:33.42 | E-bola | ahh ok it does go over sip furthger down |
| 09:33.52 | Zeeek | E-bola yyou did eh? So you didn't see the section about si and iax2? |
| 09:34.37 | Zeeek | here's another beginning startup: http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
| 09:34.55 | MatsK | Zeeke: The "s" doesn't work, "SIP/2.0 404 Not Found" so I will write a bug report |
| 09:34.55 | darviria | I was wondering, what exactly is Zaptel? Is it some type of card? I'm also trying to setup just for sip and wondered if I actually need any of the Zaptel stuff |
| 09:34.58 | Zeeek | it won't get ypou very far if you just read the first page. Read means READ |
| 09:35.48 | Zeeek | darviria ZAPTEL is the technology for the drivers connecting harware to asterisk |
| 09:35.49 | E-bola | i actualy think a technical reference would be better |
| 09:35.54 | mosty | darviria: zaptel cards include certain winmodems (for analogue lines) and isdn cards (for digital lines) |
| 09:35.55 | E-bola | like something explaninig what each option do |
| 09:36.01 | E-bola | the sample files seem quite good for basic stuff |
| 09:36.09 | mosty | and to phones |
| 09:36.15 | E-bola | atm im trying to find out if i should set domain to anything |
| 09:36.24 | mosty | e-bola: try the o |
| 09:36.28 | Zeeek | E-bola I'm afraid I'm gonna have to ignore you now because you won't go read |
| 09:36.29 | mosty | o'reilly walkthrough |
| 09:36.41 | drray | why read when you axe in here |
| 09:36.54 | drray | whining solves more problems than work |
| 09:37.06 | Zeeek | drray not necessarily |
| 09:37.22 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 09:37.33 | backblue | morning all |
| 09:37.39 | drray | drray.conf sarcasm=yes |
| 09:37.50 | darviria | most - so if I only want to use Asterisk for some softphones and some wireless voip phones internally I wont need any special hardware? I'm not interested at this stage in connecting it all to an external line |
| 09:38.22 | Zeeek | drray when someone asks " how to setup" questions and then refuses to go fscking read two or three sites that descibe this in detail, that person will not get a lot of help in future - yes I know you were being SAR castik |
| 09:38.38 | mosty | darviria: depends how the wireless voip phones connect |
| 09:38.48 | drray | do it for me |
| 09:38.53 | Zeeek | darviria this is true if you connect to SIP |
| 09:38.58 | darviria | mosty - through a wireless router |
| 09:39.00 | drray | that way I won't know how to fix it next time it breaks |
| 09:39.05 | Zeeek | hahah |
| 09:39.23 | Zeeek | if the advice is sound, it won't break :) |
| 09:39.34 | drray | everything breaks |
| 09:39.42 | Zeeek | except lines on web pages |
| 09:39.44 | mosty | darviria: so they are sip phones? no need for zaptel then |
| 09:40.13 | darviria | mosty - yep, I've had them working independantly of any software, I was wanting to use Asterisk to manage things better |
| 09:41.15 | Zeeek | darviria that's something asterisk does very well |
| 09:42.18 | tparcina | jbalcomb: can gxp2000 forward all calls to another extension? |
| 09:43.04 | jbalcomb | tparcina: i believe so. you are referring to putting the phone on DND right? |
| 09:44.56 | Zeeek | fscking search engine pollution! |
| 09:44.57 | Zeeek | Asterisk Braces Free Overnight Shipping |
| 09:44.57 | Zeeek | Free asterisk under sleeves and free overnight air shipping with purchase of asterisk... |
| 09:45.22 | Phantom-X | asterisk is always free |
| 09:45.32 | Phantom-X | not free as in fere beer but fair enough |
| 09:45.34 | Zeeek | yes, but shipping is $1000 |
| 09:45.42 | Phantom-X | Zeeek: ? |
| 09:46.02 | Phantom-X | what about emerge asterisk or apt-get asterisk etc |
| 09:46.05 | Phantom-X | its still free |
| 09:46.30 | jbalcomb | Anyone loading thier queue_log into MySQL? |
| 09:46.58 | Phantom-X | i jsut use a queue =) |
| 09:47.05 | Phantom-X | push and pop it |
| 09:47.16 | Phantom-X | a fifo queue |
| 09:48.01 | tparcina | jbalcomb: DND only doesnt recive calls, but I would like also to forward them to another extension |
| 09:48.24 | tparcina | jbalcomb: i can use DND with mute bottun, right? but how to forward calls? |
| 09:48.27 | Phantom-X | ill make a musical experiment , ill play different kind of music and time how long time ppl wants to wait before they hang up the telephone |
| 09:50.29 | Zeeek | sipura 941 will do DND and FWD calls |
| 09:51.12 | *** join/#asterisk cuco (n=cuco@local.xorcom.com) |
| 09:52.13 | cuco | hi, questions about fax here |
| 09:52.40 | cuco | i understand that in order to send faxes, i should use hylafax+iaxcom |
| 09:52.56 | my007ms | i have TE205P and three othere TDM card when i see zttool i see three TDM400P what TE205P will be in zttool |
| 09:53.13 | cuco | however, hylafax can also recive. should i disable this and let asterisk handle the incomming? |
| 09:54.27 | mosty | cuco: do you have a dedicated fax line? |
| 09:54.55 | cuco | lets assume it's voice/fax -> shared |
| 09:55.05 | jbalcomb | tparcina: upon further research I realize that we are handling that matter server side. I don't know for sure that the gxp-2000 can handle this feature natively. |
| 09:55.33 | *** join/#asterisk evisu (n=hIRC@bzq-88-153-66-99.red.bezeqint.net) |
| 09:55.50 | *** join/#asterisk TwinTrunk (i=NA@c-69-246-130-20.hsd1.ga.comcast.net) |
| 10:01.48 | *** part/#asterisk kanelbullar (n=kanelbul@83.240.200.92) |
| 10:02.03 | *** join/#asterisk kanelbullar (n=kanelbul@83.240.200.92) |
| 10:02.48 | So3kris | hello i'm tring to get a good extentions.conf. I want to make a dail plan between 3 context objects. A,B,C the config must be: A<->B->C. The problem is that If I include C at B. A get C from the include with B. How can i handel that |
| 10:04.15 | mosty | so3kris: don't include contexts that are included themselves |
| 10:04.17 | jbalcomb | So3kris maybe switch a-b? have B load A and C? |
| 10:04.22 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 10:04.59 | jbalcomb | mosty: you can do that? seems you should be able to, ive just never seen it discussed. |
| 10:05.38 | mosty | jbalcomb: you can do it. s03kris was just complaining that they didn't want the results of doing it |
| 10:06.08 | mosty | so3kris: create a new context that includes the other three |
| 10:06.11 | jbalcomb | ah, is that where you'd have to include every context in each context? |
| 10:06.41 | *** join/#asterisk saftsack (n=saftsack@p54A7EF6D.dip.t-dialin.net) |
| 10:06.58 | mosty | it's hard to tell, from so3kris' description of the problem |
| 10:07.57 | jbalcomb | hrm. yeah, i guess there is a certain catch22 there |
| 10:08.01 | So3kris | If there is a context A and B. If you want to call form A to B and B to A. Must you include at A include B. And at B include A ? |
| 10:08.38 | mosty | So3kris: create context C that includes A and B. then simply start in context C |
| 10:08.44 | jbalcomb | couldn't you just use some other form of logic to make it work? |
| 10:10.09 | jbalcomb | perhaps duplicate some of the lines from each context calls from either side? |
| 10:10.25 | jbalcomb | s/calls/to handle calls/ |
| 10:10.47 | jbalcomb | jbot: you are a sweetheart pal |
| 10:10.50 | jbot | I think you lost me on that one, jbalcomb |
| 10:11.01 | So3kris | mosty: thanx i will try that |
| 10:11.06 | jbalcomb | ~jbot jbalcomb is Jim Balcomb |
| 10:11.08 | jbot | jbalcomb: okay |
| 10:11.20 | jbalcomb | ~jbot what is jbalcomb? |
| 10:11.22 | jbot | I think you lost me on that one, jbalcomb |
| 10:11.22 | *** join/#asterisk Falle (n=falle@falle.se) |
| 10:11.29 | jbalcomb | ~jbalcomb |
| 10:11.30 | jbot | i heard jbalcomb is Jim Balcomb |
| 10:11.38 | mosty | So3kris: or create two contexts, A_outgoing and B_outgoing that calls from A and B start in respectively. and include A and B in both |
| 10:11.39 | jbalcomb | ~yermom |
| 10:12.51 | *** join/#asterisk swytch (n=ezcall@LNeuilly-152-22-86-193.w193-251.abo.wanadoo.fr) |
| 10:16.27 | jbalcomb | ~jbot yermom is the sweetest, most beautiful hippopotamus ever to wade in the Tarangire River. |
| 10:16.29 | jbot | okay, jbalcomb |
| 10:17.16 | jbalcomb | that'll do jbot, that'll do. |
| 10:22.02 | Zeeek | lunch ! |
| 10:22.12 | Zeeek | ~zeeek |
| 10:22.14 | jbot | i guess zeeek is someone who once said "learning asterisk using a GUI is like learning sex through masturbation. You'll never get to the good stuff." |
| 10:24.40 | So3kris | :D |
| 10:27.19 | tparcina | jbalcomb: how do you handle forward of all calls on server side? could you please send me sample of extensions.conf? |
| 10:30.03 | jbalcomb | Call Forwarding: |
| 10:30.11 | jbalcomb | *21XXXX Call Forward to XXXX |
| 10:31.17 | jbalcomb | tparcina: http://pastebin.ca/85960 |
| 10:32.09 | jbalcomb | i think that was cut&paste coding by the admin before me. I'm fairly certain the syntax is deprecated but it still works. |
| 10:36.05 | tparcina | jbalcomb: thank you for code. one more thing, this is for forwarding. do i still have to make some adjustments for calling that sip extension ir does asterisk do it? because, asterisk database should be checked somhow, now is it done automaticly or there is something more that should be changed in extensions.conf |
| 10:36.45 | jbalcomb | Anyone know if the Java SDK package still exists or is Sun changed the name? |
| 10:37.17 | jbalcomb | This page [http://java.sun.com/javase/downloads/index.jsp] is all I see of the Java stuff I need for my QueueMetrics package |
| 10:37.36 | jbalcomb | This page [http://www.astrecipes.net/index.php?n=108] however says to be certain I get the SDK |
| 10:38.23 | jbalcomb | tparcina: not sure I follow you there. when someone dials *21XXXX it put XXX in the asterisk DB |
| 10:38.54 | tparcina | jbalcomb: ok, i copy that |
| 10:39.55 | tparcina | and when sombody dials - Dial(SIP/xxx) - it won't be forwarded, right? so, how should this dial part of extensions.conf look like? |
| 10:40.25 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 10:40.28 | jbalcomb | tparcina: http://pastebin.ca/85963 look at priority level 3 for the dnd check and forward |
| 10:41.01 | *** join/#asterisk r0d3nt (n=RatMan@ip68-108-184-243.lv.lv.cox.net) |
| 10:41.37 | So3kris | mosty: Thank you it's working |
| 10:44.47 | E-bola | I cant get my asterix server to connect to my providers |
| 10:44.50 | E-bola | the register command fails |
| 10:45.02 | E-bola | besides wrong password/ip etc, can there be other reasons? |
| 10:45.13 | E-bola | like some security compilation/configuration error on my side? |
| 10:46.42 | So3kris | reload |
| 10:46.44 | So3kris | shit |
| 10:47.31 | So3kris | manage youre asterisk server by the asterisk ircbot :D |
| 10:50.00 | tparcina | jbalcomb: thank you, i have checked you code. i have get the idea how you do it. i'll probably need to adjust it to my needs. |
| 10:50.44 | Zeeek | shit reload |
| 10:50.56 | Zeeek | No such command 'shit' (type 'help' for help) |
| 10:51.16 | Zeeek | help |
| 10:51.26 | Zeeek | No such command 'help' (type 'shit' for help) |
| 10:52.08 | So3kris | :D |
| 10:53.24 | tparcina | anybody else forwards calls on asterisk (instead on the phone - grandstream gxp2000 doesn't support it) |
| 10:54.26 | jbalcomb | tparcina: cool, good luck. |
| 10:54.27 | Zeeek | I have done this sometimes |
| 10:54.46 | Zeeek | I use a code for fwd by CID |
| 10:54.59 | Zeeek | for example dial *600numbertoforwardto |
| 10:55.15 | jbalcomb | I use a baseball bat and post it notes to have my calls forwarded |
| 10:55.32 | Zeeek | that's even better but enters meatspace which is out of the realm of #asterisk |
| 10:55.57 | drray | so if digium had a contract with a guy named "the bat" you'd be down with it zeek? |
| 10:56.01 | Zeeek | anyway, it was a simple process to set up key/value paris in asteriskdb |
| 10:56.16 | Zeeek | I once used TheBat! as an email client |
| 10:56.23 | jbalcomb | yeah, i swear our receptionist has some sort of mental syndrome. She keeps reminding me that my phone doesn't auto-answer when she calls to let me know i have a call. |
| 10:56.38 | drray | passive aggressive |
| 10:56.48 | tparcina | Zeeek, there isn't option like forward_all_calls predefined on asterisk. everything shoul be made thrue dialplan using asterisk_database, right? |
| 10:57.02 | jbalcomb | "Yeah, Debbie, I know. If I get a call, just transfer it to me. If I want it, I'll take it, if not, it will go to my voicemail. Ok? Thanks." |
| 10:58.01 | jbalcomb | Anyone else work for a company that thinks auto-answer should be the default behavior for every phone in the building? |
| 10:58.17 | drray | outside of a call center |
| 10:58.21 | tparcina | jbalcomb, you have receptionist? good for you! ;)) |
| 10:58.31 | drray | does not sound like he has one actually |
| 10:58.48 | jbalcomb | tparcina: yeah, but we don't need one because we have a $20,000 phone system (TCO anyway) |
| 10:59.22 | jbalcomb | they think people getting a person when they call here is a selling point for our services... |
| 10:59.25 | tparcina | so, why do you keep her? or is it a domb question ;)) |
| 10:59.37 | *** join/#asterisk MatsK (n=mats@141.221.181.62.in-addr.dgcsystems.net) |
| 10:59.48 | jbalcomb | tparcina: not my choice.. I'm pushing my luck having auto-answer disabled on my phone. ;) |
| 11:00.44 | jbalcomb | they also really really like the paging system here.. |
| 11:00.46 | drray | I've worked in places like taht |
| 11:00.59 | drray | but it seemed to work downhill |
| 11:01.06 | jbalcomb | i've never worked for another company that had auto-answer or a paging system. |
| 11:01.20 | Zeeek | ok, some real work got in the way bs'ing on IRC |
| 11:01.24 | benjk | has anybody here ever seen a dmesg "JBD: barrier-based sync failed on hdaN ..." |
| 11:01.25 | *** join/#asterisk queso (n=josh@nr13-216-68-211-77.fuse.net) |
| 11:01.41 | jbalcomb | benjk: not I. |
| 11:01.50 | saftsack | hello |
| 11:01.54 | jbalcomb | hello |
| 11:01.56 | benjk | I wonder if this is a hardware or software problem |
| 11:01.57 | tparcina | jbalcomb: does grandstream gxp have autoanswer option? |
| 11:01.57 | Zeeek | It is VERY simple to arrange call forwarding. Google for "asterisk follow me" for starters, I'm sure there's dialplans out there ready to go |
| 11:02.10 | saftsack | so i have to go now, will be back later, cYA |
| 11:02.14 | jbalcomb | tparcina: yes, default or by call-info SIP Header |
| 11:02.20 | benjk | all things considered, I think the 2.6 kernel isn't ready for prime time yet |
| 11:02.22 | jbalcomb | goodbye saftsack |
| 11:02.27 | Zeeek | does grandstream have a web sites? Why not go look at the spec sheet for the phone? |
| 11:02.53 | jbalcomb | better yet, go to the gxp-2000 page on the voip-info wiki |
| 11:03.20 | jbalcomb | i had to set up grandstream.com on my internal name server so my phones would auto provision... |
| 11:03.43 | jbalcomb | freakin phone supports DHCP option 66 but its off by default. :/ |
| 11:04.00 | tparcina | yes, there is everything on Internet. what the *ell, we even don't need this channel at all! - i have just asked, one short, easy question |
| 11:04.45 | jbalcomb | the hell we don't need this channel. how else can i find a way to look at my clock and be like "holy hell, its 4:30 PM already?! ... sweet." |
| 11:05.13 | jbalcomb | Franz Ferdinand helps with that too. |
| 11:06.38 | Zeeek | tparcina but you keep asking it, my own reflex would be to just go see. You aren't obliged to follow that if you have the time |
| 11:06.51 | Zeeek | ~time |
| 11:06.56 | jbot | extra, extra, read all about it, time is 1 dimensional, or everlasting, an illusion, or 2006.07.12 11:06:56 GMT |
| 11:06.56 | *** join/#asterisk James_boxhead (n=icechat5@203.94.42.234) |
| 11:07.00 | Zeeek | ~date |
| 11:07.02 | jbot | It is now 2006.07.12 11:07:02 GMT |
| 11:07.23 | Zeeek | ~do grandstream phones do cfwd? |
| 11:07.24 | jbot | ACTION does grandstream phones do cfwd. |
| 11:12.48 | *** part/#asterisk cuco (n=cuco@local.xorcom.com) |
| 11:13.12 | tparcina | ~jbot is a freek ;) |
| 11:14.31 | *** join/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com) |
| 11:18.32 | *** join/#asterisk daysmen3_ (n=primus@host86-141-242-253.range86-141.btcentralplus.com) |
| 11:20.22 | drray | ~jbot is not stupid? |
| 11:21.27 | tparcina | drray: please don't call anybody stupid or implicade that someone is |
| 11:22.11 | *** part/#asterisk tparcina (n=tparcina@lns02-0599.dsl.iskon.hr) |
| 11:23.40 | *** join/#asterisk P-NuT (n=P-Nut@CPE-60-227-93-75.nsw.bigpond.net.au) |
| 11:24.19 | benjk | jbot is a bot, it's designed to be silly |
| 11:24.51 | Zeeek | ~silly |
| 11:24.52 | jbot | hmm... silly is 'a good question, the answer to which is "yes, i am"' |
| 11:25.00 | Zeeek | nyuk, nyuk |
| 11:25.59 | *** join/#asterisk niZx (n=ilt@S010600080db4ab60.wp.shawcable.net) |
| 11:26.28 | *** join/#asterisk evisu (n=hIRC@bzq-88-154-40-110.red.bezeqint.net) |
| 11:28.32 | *** join/#asterisk bikola_c (n=bikola@243.058.dsl.brs.iprimus.net.au) |
| 11:29.05 | *** join/#asterisk Mr-packet (n=a@222-154-239-122.adsl.xtra.co.nz) |
| 11:30.10 | bikola_c | hey guys, new to asterisk, was wondering are there any free voip to pstn terminaiton services |
| 11:30.45 | *** part/#asterisk P-NuT (n=P-Nut@CPE-60-227-93-75.nsw.bigpond.net.au) |
| 11:31.42 | Zeeek | look at freeworlddialup and sipgate |
| 11:31.57 | Zeeek | free for many tollfree numbers |
| 11:34.22 | Mr-packet | im wondering if anyone has got any ideas on what the general capability of a Linksys WRT54GS might be, if it was runing asterisk |
| 11:34.54 | Mr-packet | just for use at home, to connect to an upstream provder |
| 11:35.10 | CoffeeKid | ~do jbot is a freak. |
| 11:35.11 | jbot | ACTION does jbot is a freak.. |
| 11:35.25 | CoffeeKid | hehe |
| 11:35.56 | Mr-packet | we are all freaks |
| 11:36.03 | CoffeeKid | its true |
| 11:36.32 | bikola_c | thx' |
| 11:37.00 | TwinTrunk | hey what was the command to see a live call, etc take place in the highest of detail on the console? i forgot :( |
| 11:38.33 | *** join/#asterisk cy3o3 (i=cy@absolut.reject.org) |
| 11:39.43 | Mr-packet | well, i guess i will just go and find out then |
| 11:44.36 | E-bola | 12 15:42:21 NOTICE[4630]: chan_sip.c:5372 sip_reg_timeout: -- Registration for '369158@sip.v3tel.com' timed out, trying again (Attempt #60) |
| 11:44.36 | E-bola | <PROTECTED> |
| 11:44.40 | E-bola | why do i keep getting that? |
| 11:45.48 | Nugget | because it's doing that. |
| 11:45.58 | *** join/#asterisk kay2 (n=key2@sd-420.dedibox.fr) |
| 11:46.33 | Nugget | are you confused because you don't know what DNS SRV is, or are you asking why v3tel.com is timing out? |
| 11:47.31 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 11:48.48 | niZx | TwinTrunk: show channel <channel id> |
| 11:49.38 | *** join/#asterisk bmg505 (n=leon@c1-201-9.rndf.isadsl.co.za) |
| 11:50.49 | bikola_c | hey, i remember a year ago i was really into this voip thing, now im sort of getting back at it, i remember though free IAX to PSTN termination to landlines in certain countries, the ones you guys allready provided (fwd) and sipgate arent it, any other guesses? |
| 11:51.07 | syle | how do you block non-callerid on zap line |
| 11:52.00 | X-Rob | bikola_c, I don't think any do free any more. |
| 11:52.29 | X-Rob | a couple of years ago some places were doing it to encourage customers, but don't need to these days. |
| 11:54.24 | bikola_c | damn, i have this empty spot in my head, just cant think of the name... OK, anyone know the cheapest place to buy minutes, without acctually buying bulk |
| 11:55.04 | TwinTrunk | okay... i'm getting a weird/fucked up-ness here, i can make a call just fine but when i try to receive a call... the asterisk log says http://pastebin.ca/85989 someone please help. :( |
| 11:59.03 | E-bola | nugget: why its timing out |
| 11:59.12 | E-bola | they made me move the server to a none natted ip |
| 11:59.18 | E-bola | but its still timing out, and i got no clue why |
| 11:59.27 | E-bola | i doublechecked that the hostname and login/pass are correct |
| 12:00.16 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 12:02.43 | TwinTrunk | how can i change asterisks IP? when installing i put just any IP but now I need to change it to 192.168.98.103 and don't really feel like deleting it then reinstalling it just to get to that prompt :( |
| 12:03.02 | X-Rob | ~asteriskbook |
| 12:03.20 | X-Rob | TwinTrunk, it's not asterisks IP, it's the linux machines ip. |
| 12:04.04 | *** join/#asterisk ariel_ (n=Ariel@70.46.87.158) |
| 12:04.15 | TwinTrunk | ah |
| 12:04.20 | *** join/#asterisk af_ (n=af@ip-170-209.sn1.eutelia.it) |
| 12:04.35 | *** join/#asterisk evisu (n=hIRC@bzq-88-152-186-222.red.bezeqint.net) |
| 12:04.48 | X-Rob | if you're using centos, you can hand edit /etc/sysconfig/network/ifcfg-eth0 or use sysconfig-network |
| 12:04.58 | X-Rob | (sysconfig-network is a program) |
| 12:05.14 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 12:05.36 | TwinTrunk | well then that is already set to what it should be because it says its using 192.168.98.103 |
| 12:06.44 | *** join/#asterisk tparcina (n=tparcina@lns02-0599.dsl.iskon.hr) |
| 12:07.44 | TwinTrunk | its just that when installing asterisk it asked The IP Address of your Asterisk Machine and i put 192.168.98.90 - i was thinking that doing that might be what is causing my current problem of not being able to receive calls but only make them |
| 12:07.49 | znoG | bikola_c: would that be www.voipuser.org? |
| 12:07.53 | E-bola | nugget? |
| 12:08.17 | nextime | is there any news about the release of * 1.4? |
| 12:08.50 | Ahrimanes | it's not there yet? |
| 12:10.03 | nextime | Ahrimanes : the latest release is 1.2.9.1 as i can see on asterisk.org web site, and, even from the same site, i can read that 1.4 is scheduled for "beginning of july 2006" |
| 12:10.06 | Zeeek | soon (rsn) |
| 12:10.27 | Ahrimanes | mmm real soon now |
| 12:10.32 | Zeeek | nextime Mark announced at Astricon that he assumed that meant end of July |
| 12:10.33 | bikola_c | ZnoG: ill go check it out now |
| 12:10.45 | nextime | zeek: ok, thanks |
| 12:11.43 | *** join/#asterisk snitt (i=endre@222-006.adsl.pool.ew.hu) |
| 12:12.26 | [TK]D-Fender | TwinTrunk : PLease understand that * does not HAVE an IP. Linux does. And for support of Trixbox please read the /topic |
| 12:12.34 | *** join/#asterisk mfdutra (n=marlon@200.208.130.16) |
| 12:12.40 | *** join/#asterisk sshadow (n=sshadow@213-84-101-107.adsl.xs4all.nl) |
| 12:12.51 | mfdutra | does anyone have good experience with txfax? |
| 12:13.14 | tparcina | grandstream gxp2000 - headset adapter, is it for microphone, speakaer or both? |
| 12:13.33 | bikola_c | ZnoG: thx man, i think that is it |
| 12:13.34 | tparcina | Fender, you are awake? :) |
| 12:14.22 | nextime | mfdutra : in my opinion, try iaxmodem instead of rxfax/txfax |
| 12:14.34 | mfdutra | hmm why? |
| 12:14.36 | mfdutra | is it better? |
| 12:14.49 | mfdutra | I've been using rxfax ok |
| 12:15.13 | tparcina | asterisk dewelopers are planing to go out with new version every 6 months |
| 12:15.28 | nextime | mfdutra : it permit to use hylafax with a clean asterisk without patches or versioning problem, and hylafax have many contributions like web interfaces, mail2fax and others |
| 12:15.48 | [TK]D-Fender | tparcina : Its for a HEADSET like it says |
| 12:16.00 | TwinTrunk | ([TK]D-Fender): I don't have Trixbox but thanks anyway :) |
| 12:16.11 | nextime | mfdutra : and more, with hylafax + iaxmodem i can say that 99% of devices are sending and receiving fax correctly, i can't say the same for txfax/rxfax |
| 12:16.21 | mfdutra | hmm. if txfax worked fine, it's pretty good for me. I made a couple of perl scripts with a web interface and fax2mail interface |
| 12:16.32 | mfdutra | hmm good to know |
| 12:16.39 | [TK]D-Fender | TwinTrunk : Well it sounds like some sort of packaged * (A@H, etc) if you word it like Asterisk asking for an IP..... |
| 12:16.59 | mfdutra | since rxfax is working ok and in production for me, I'll consider testing hylafax for sending faxes |
| 12:17.29 | mfdutra | can I send a fax through a command line with hylafax? |
| 12:17.37 | nextime | mfdutra : absolutly yes |
| 12:17.40 | mfdutra | great |
| 12:17.46 | TwinTrunk | ([TK]D-Fender): yeah it was freebox asking for it during the install |
| 12:17.50 | mfdutra | it doesn't use spandsp, does it? |
| 12:18.31 | tparcina | Fender, thank you. and my headset has two connectors - and this one on phone has only one. i have tried it out with one 2to1 converter and it didn't work |
| 12:18.49 | tparcina | have you used headset with gxp2000? |
| 12:19.12 | nextime | mfdutra : iaxmodem work on top of spandsp, but it don't need to be linked to *, iaxmodem work as an "external iax client" for the asterisk view, and for a "hardware modem" for the hylafax view. It act as a iax client that login in asterisk and create a "virtual" tty to talk with hylafax |
| 12:20.15 | nextime | so, * <--IAX-->iaxmodem<--TTY-->hylafax |
| 12:20.19 | mfdutra | I understood that. I read about it yesterday |
| 12:20.23 | mfdutra | it seems to be nice |
| 12:20.37 | mfdutra | my concern is about the quality of the fax sent |
| 12:20.50 | mfdutra | txfax drops the call almost all the times without finishing the fax |
| 12:21.16 | nextime | mfdutra : i use it in production from 1/2 year now, and i'm very happy about stability and quality |
| 12:21.23 | mfdutra | great |
| 12:21.41 | mfdutra | I'll hopefully test it |
| 12:22.26 | mfdutra | do you use packages or compiled from source? |
| 12:22.46 | nextime | mfdutra : i use hylafax from debian packages, asterisk and iaxmodem from source |
| 12:22.57 | mfdutra | Version: 1:4.2.1-5sarge3 |
| 12:23.11 | mfdutra | my asterisk is from source too |
| 12:23.16 | mfdutra | debian's is old |
| 12:23.27 | [TK]D-Fender | tparcina : Your headset is a COMPUTER headset, not a PHONE headset. Wired differently. |
| 12:23.49 | nextime | mfdutra : agree |
| 12:24.08 | mfdutra | did you have to care about txgain and rxgain in zapata.conf? |
| 12:24.55 | E-bola | failed for '213.50.48.226' - Not a local SIP domain |
| 12:25.03 | nextime | mfdutra : no, you only need to assure that you don't have echo problems, and even it is better to not use echo cancellation for the channels used for faxes |
| 12:25.04 | E-bola | asterix wont let me register |
| 12:25.24 | mfdutra | really? |
| 12:25.25 | E-bola | im calling from behind a nat device, but the asterix server is directly on a wan ip |
| 12:25.28 | nextime | mfdutra : yes |
| 12:25.34 | E-bola | do anybody know whats wrong? |
| 12:25.39 | mfdutra | my server is only a fax server, so I'll turn off echo cancel |
| 12:26.17 | tparcina | can you send me link to phone headset that you use with gxp2000, please? |
| 12:26.24 | tparcina | Fender: can you send me link to phone headset that you use with gxp2000, please? |
| 12:26.48 | mfdutra | nextime: thanks for the attention. let's see what I get here. I'll let you know |
| 12:27.46 | [TK]D-Fender | tparcina : I don't own a GXP (nor will I ever) and its a standard headset connector of the same style cell-phones use. 2.5 MM |
| 12:30.04 | tparcina | Fender: why don't you like grandstream? what phones do you use? |
| 12:30.30 | mut | rotary dial phones with an ata |
| 12:31.12 | dpryo | lol |
| 12:31.40 | mut | hey! get off my party line! |
| 12:32.05 | [TK]D-Fender | tparcina : Flakey firmware, cheap production quality. poor physical feel. There was a reason they're called BarbieTones. |
| 12:32.26 | *** join/#asterisk heison (n=heison@ns.somanetworks.com) |
| 12:32.52 | mut | alright! IRCing from the dns server |
| 12:33.00 | cy3o3 | heh |
| 12:33.30 | [TK]D-Fender | tparcina : I typically only suggest Polycom. Cisco is also excellent, but overpriced and only become a worthy contender in certain scenarios because of that difference. Snom is barely OK. Linksys isn't too bad, but in the US the price difference disqualifies them as well. |
| 12:34.39 | nextime | wow, new asterisk trunk configure script and menuselect feature is amazing |
| 12:35.28 | [TK]D-Fender | mut : I still haven't fount one with pulse support! Link me! ;) |
| 12:38.22 | [TK]D-Fender | tparcina : Aastra might be an interesting option as well, but Their style and feel is a little behind Polycom, but are interesting in the "budget" range". |
| 12:38.45 | tparcina | Fender: I must agree with your about grandstream. but it seams that they are improwing... anyway, we sell them for users who saves money on everything |
| 12:40.11 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 12:40.12 | *** mode/#asterisk [+o anthm] by ChanServ |
| 12:40.19 | cy3o3 | Anyone have experience with database driven backend for asterisk? |
| 12:40.30 | [TK]D-Fender | tparcina : "You get what you pay for" |
| 12:40.38 | tparcina | Fender: so, have you used cisco? |
| 12:41.02 | tparcina | Fender: no, my costumers get what they pay for :)) |
| 12:41.16 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 12:41.41 | [TK]D-Fender | tparcina : I've only had my hands on one once, but from the impression I got from thta and several other accounts and reviews they seem really good. SIP implementation isn't always stable though, and firmware stability isn't always good either |
| 12:42.01 | [TK]D-Fender | tparcina : And if they pay for a GXP they get it :) I pity them. |
| 12:42.12 | *** join/#asterisk gandhijee (n=gandhije@mail.win-ent.com) |
| 12:43.04 | tparcina | Fender: I'm hapy when they get cisco, because 10% of 100 is much more than 10% of 40 ;)) |
| 12:43.47 | *** join/#asterisk nitram (i=foo@superblob.com) |
| 12:44.05 | [TK]D-Fender | tparcina : I see basic math has not failed you so far at least... What is Aastra / Linksys pricing like where you are? |
| 12:44.27 | [TK]D-Fender | tparcina : Either wouldd be a better choice for you to offer your customers. |
| 12:44.42 | tparcina | Fender: then i should feal lucky - in my company we use Cisco (7905, 7912, 7940, 7960, 7970 and 7920) - and so far, so good |
| 12:44.44 | *** join/#asterisk evisu (n=hIRC@bzq-88-153-208-11.red.bezeqint.net) |
| 12:45.21 | babyju | Can asterisk act as a sip proxy, allowing client connections from the internet (traversing firewall) and allowing calls to be made without the use of a vpn? |
| 12:45.24 | tparcina | Fender: i don't know prices, that is something my cooworkers doo |
| 12:45.55 | MrChimpy | you work in a pigeon loft? |
| 12:46.30 | [TK]D-Fender | babyju : Yes * can work behind NAT and do all the things that * does without need of VPN. |
| 12:46.42 | [TK]D-Fender | tparcina : You should ask |
| 12:46.47 | gandhijee | babyju: why not use something that is an actual sip proxy |
| 12:46.58 | tparcina | Fender: i'll sugest to people who decaide (astra/linksys) but I don't think anything will change now since we have sell several of them. and once we do, they don't change products weary easy |
| 12:47.09 | gandhijee | SER seems to come to mind |
| 12:48.00 | [TK]D-Fender | babyju : describe the full call flow (where phones / * / other resources) would be located in your scenario so we can see if this is the right choice for you |
| 12:49.48 | babyju | remote user<->firewall<->Internet<->NOC-Firewall<->Asterisk<->PRI/POTS |
| 12:50.39 | [TK]D-Fender | babyju : Ok, that can work just fine. |
| 12:51.04 | [TK]D-Fender | babyju : And * is indeed the tool for the job (assuming you aren't looking for a HUGE system) |
| 12:51.09 | *** join/#asterisk Tili (n=Tili@cm109.gamma248.maxonline.com.sg) |
| 12:52.09 | babyju | [TK]D-Fender thanks....I also read that * does not support bridge extensions, is this true? |
| 12:53.53 | *** join/#asterisk bitboy (n=amit@adsl-065-012-197-229.sip.bct.bellsouth.net) |
| 12:54.13 | bitboy | hello |
| 12:56.19 | *** join/#asterisk myiagy (n=myiagy@mail.voffice.com.br) |
| 13:01.54 | [TK]D-Fender | babyju : Can you clarify "bridge extensions" to make sure we're thinking the same thing... |
| 13:04.44 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 13:06.01 | babyju | extension that is visible on at least 2 phones. You can see when someone is on a call. Secretaries loves this. |
| 13:06.28 | gandhijee | isn't that slated for 1.4? |
| 13:06.41 | *** join/#asterisk Katty (n=aisaacs@64.82.232.54) |
| 13:07.32 | Katty | morning. |
| 13:08.09 | file | Katty: how are we today? |
| 13:08.12 | [TK]D-Fender | babyju : That is referred to as "presence" and yes * supports it on many phone models. However * does not support SHARED APPEARANCES which is where you can PICK UP their line if it is ringing, etc. |
| 13:08.18 | [TK]D-Fender | Katty : mew. |
| 13:10.52 | Katty | file: trying to wake up. woke up freezing =< |
| 13:10.58 | Katty | [TK]D-Fender: mew. |
| 13:11.14 | file | eep |
| 13:11.41 | [TK]D-Fender | Katty : We're hitting the 90's by the end of this week at night. Gonna have to rush the AC in. |
| 13:12.09 | *** join/#asterisk TeePOG (n=1234@mtngprs4.mtn.co.za) |
| 13:12.12 | TeePOG | hi hi |
| 13:12.32 | Katty | [TK]D-Fender: well it's been raining here for 3 days straight. |
| 13:12.50 | Katty | [TK]D-Fender: even broke out the hot chocolate this morning. |
| 13:14.19 | [TK]D-Fender | Katty : Wow. |
| 13:15.46 | [TK]D-Fender | And "Shared Call Appearances" will not apparently make it into * 1.4 at release last I heard. |
| 13:16.16 | file | evil things... |
| 13:17.08 | gandhijee | damn that sucks |
| 13:18.39 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
| 13:20.29 | *** join/#asterisk feld_ (n=feld@12.148.212.157) |
| 13:22.30 | *** join/#asterisk evisu (n=hIRC@bzq-88-155-138-136.red.bezeqint.net) |
| 13:23.49 | Assid | sup tkd, file, katty |
| 13:24.14 | [TK]D-Fender | Assid : Still breathing... |
| 13:24.19 | file | not I... yet :( I should wobble around and take a shower... but it's early |
| 13:25.22 | my007ms | what sampel of /etc/zaptel look like when i use TE205P |
| 13:25.32 | my007ms | the line is E1 |
| 13:25.47 | my007ms | # ??: 62 TE2/0/2/31 |
| 13:25.57 | my007ms | so what ?? optian is ?? |
| 13:27.13 | nextime | asterisk trunk seem to work good ( i'm testing it on a non-production server ) |
| 13:27.47 | Katty | Assid: (= |
| 13:27.57 | *** join/#asterisk crich1999 (n=crich@pd956852e.dip0.t-ipconnect.de) |
| 13:31.55 | my007ms | pleas some one tell me what first sampel line in /etc/zaptel.conf when i use TE205P and E1 line |
| 13:31.57 | *** join/#asterisk brif8 (n=Administ@ns1.ttienterprises.org) |
| 13:32.06 | my007ms | i know i need to know line it's ccs |
| 13:32.28 | my007ms | and signale hdb3 |
| 13:32.35 | my007ms | so how like will look like |
| 13:32.36 | my007ms | ?? |
| 13:32.40 | trelane_ | my007ms, go find genzaptelconf |
| 13:32.50 | trelane_ | it's a bash script which will (hopefully) do it for you |
| 13:32.57 | my007ms | i run this and it make it like this ] |
| 13:33.00 | brif8 | Is there a limit on the extensions.conf file ?? I have 724 lines, but when I execute a "extensions reload" it only loads the first 602 lines ? and leaves the rest ? |
| 13:33.04 | my007ms | # ??: 62 TE2/0/2/31 |
| 13:33.14 | my007ms | so what is "??" |
| 13:33.24 | trelane_ | my007ms, did you see my answer? |
| 13:33.43 | my007ms | i have this and use it trelane _ |
| 13:33.53 | trelane_ | check zapata-channels.conf |
| 13:33.59 | trelane_ | copy it's contents into zapata.conf |
| 13:34.52 | my007ms | trelane_, i don't have file call zapata-channels.conf |
| 13:35.09 | myiagy | does anyone know if there's a softphone, sip or iax, for windows mobile/palm os/symbian ? |
| 13:35.19 | trelane_ | hrm, I usually get one when using genzaptelconf, perhaps your mileage varies on the digital interfaces. |
| 13:35.44 | my007ms | yes it is |
| 13:35.53 | my007ms | as i see it's TE205P |
| 13:36.09 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 13:38.46 | *** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com) |
| 13:40.05 | MrChimpy | afternoon asterisk-types |
| 13:40.51 | tparcina | where can i find what new features asterisk 1.4 will have? |
| 13:41.06 | [TK]D-Fender | ~book |
| 13:41.08 | jbot | methinks book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 13:42.12 | *** join/#asterisk Hmmhesays (i=negative@66.173.103.110) |
| 13:42.20 | Hmmhesays | Morning all |
| 13:42.30 | trelane_ | morning :) |
| 13:43.23 | *** join/#asterisk kristalino (n=kristali@84-50-114-166-dsl.trt.estpak.ee) |
| 13:44.48 | Hmmhesays | I'm still swearing at this as53000 |
| 13:44.52 | Hmmhesays | *as5300 |
| 13:45.01 | *** join/#asterisk MACscr (n=MACscr@66.73.154.70) |
| 13:46.24 | brif8 | is there an extensions.conf limit ? my * will only read the first 600+/- lines and not the rest of my extensions.conf file ? Also I'm now getting an Auto fallthrough, channel 'SIP/bacc-68eb' status is 'UNKNOWN' |
| 13:47.04 | trelane_ | brif8, I'm not aware of one, and I've had extensions.conf that went longer than that |
| 13:47.14 | trelane_ | I usually chop up my dialplans as much as possible now and #include |
| 13:47.48 | [TK]D-Fender | brif8 : Pastebin.... |
| 13:47.49 | brif8 | that might be a good idea, wonder why it's only reading 600 lines and then stops reading the file |
| 13:47.50 | mosty | is there an error in the file? |
| 13:47.57 | brif8 | not that I can see |
| 13:48.18 | mosty | what does the console say when you reload extensions? |
| 13:48.55 | tparcina | asterisk 1.4 - guys (and Katty :) where do you get informations what new features will it have? |
| 13:49.03 | trelane_ | tparcina, svn? |
| 13:49.20 | trelane_ | tparcina, the blogs on asterisk.org? |
| 13:49.57 | MACscr | does a website exist that allows you to test ping a server from different servers around the world? |
| 13:50.07 | tparcina | trelane: thanks', i'm checking it now |
| 13:50.20 | *** join/#asterisk Egonis (n=Egonis@207.245.14.10) |
| 13:50.38 | Egonis | What command would I issue to do a Playback(file) to Console/dsp? |
| 13:50.41 | *** join/#asterisk mog_home (n=mogorman@68.62.237.103) |
| 13:50.42 | my007ms | is that span=1,1,0,esf,b8zs right line if digital |
| 13:50.50 | my007ms | zaptel.conf |
| 13:51.41 | *** join/#asterisk kyborg (n=kyborg@201.200.65.230) |
| 13:51.59 | kyborg | hello fellas |
| 13:52.38 | kyborg | is there any voip provider offers multichannels support? |
| 13:53.27 | Hmmhesays | that is a good question Egonis |
| 13:53.31 | *** join/#asterisk DrMouse (n=karl@user-5784.l6.c2.dsl.pol.co.uk) |
| 13:54.10 | kyborg | -_- |
| 13:54.34 | tparcina | trelane_: i dont use svn and some informations can be found on blogs. is there (somwhere) list of all new features (or should i read all blogs on asterisk.org)? |
| 13:54.56 | trelane_ | tparcina, I doubt there will be until an official beta is released, and no there is not to my knowledge |
| 13:55.15 | brif8 | mosty: just lists the extensions and then stops no error no message just does extensions xxxxx,1 and ends |
| 13:55.18 | Egonis | Hmmhesays: Is it possible? |
| 13:55.20 | tparcina | trelane_: ok, thank you for helping |
| 13:55.28 | Hmmhesays | Egonis: i don't know |
| 13:55.31 | DrMouse | hi, a quick question. does the default law in the zaptel driver (shown by 'zap show channel x') affect FXO channels? if so, i read that the default for europe is alaw, but my card is set to ulaw. is that correct (i'm in the uk) and if so how do i change it? |
| 13:55.33 | trelane_ | tparcina, no problem, sorry I couldn't have been of more help |
| 13:55.55 | mosty | brif8: set verbose and debug to something higher? |
| 13:56.07 | brif8 | mosty: it is at 9 right now |
| 13:56.25 | *** join/#asterisk basic1 (n=basic1@190.48.146.146) |
| 13:56.50 | *** join/#asterisk Evisu (n=hIRC@bzq-88-155-204-11.red.bezeqint.net) |
| 13:56.59 | brif8 | trelane_: to add include it just include => file_name.conf right ? |
| 13:57.58 | basic1 | hi everybody ... someone can help me with a question with TE110P |
| 13:58.13 | DrMouse | btw, to be polite ill introduce myself. i'm mouse, an IT/Telephony engineer in Leeds, UK |
| 13:59.01 | brif8 | ~wiki |
| 13:59.10 | DrMouse | when ur banging ur head against a brick wall its easy to forget manners :) |
| 13:59.37 | [TK]D-Fender | Wow, the new iP 430 boots pretty fast...... |
| 14:00.28 | DrMouse | MrChimpy - (if that was to me) no it's north-east england, the beautiful county of yorkshire. a lovely city, sometimes... |
| 14:00.28 | mosty | drmouse: loadzone and defaultzone in /etc/zaptel.conf ? |
| 14:00.41 | DrMouse | loadzone and defaultzone are uk |
| 14:00.52 | *** part/#asterisk tparcina (n=tparcina@lns02-0599.dsl.iskon.hr) |
| 14:01.05 | MrChimpy | drmouse: no, me london. you leeds. |
| 14:01.08 | DrMouse | i have also put the opermode option in modules.conf |
| 14:01.52 | mosty | drmouse: and switchtype in /etc/zapata.conf maybe |
| 14:01.58 | *** join/#asterisk veepster_ (n=veepster@67.130.38.2) |
| 14:02.01 | DrMouse | okies chimpy. so ur based in france then :) (anything south of watford is france) |
| 14:02.29 | gandhijee | my uncle use to live in watford |
| 14:02.32 | Egonis | So is it possible to play files i.e. Playback(blah) to Console/dsp? How would I do this? I can't find a howto anywhere |
| 14:02.35 | gandhijee | nice place from what i remember |
| 14:02.52 | mosty | egonis: isn't there a system or exec command available? |
| 14:02.54 | MACscr | Am i being ignorant thinking i can use asteriask in a configuration where i can have extensions in the UK, when the voip server is at a DC in the US? |
| 14:03.00 | Egonis | mostly: I don't know, I can't find any |
| 14:03.09 | MACscr | if so, what are my options that arent outragious |
| 14:03.18 | DrkShdw | MACscr: no, it's not ignorant. yes, you can. |
| 14:03.19 | DrMouse | i have no switchtype. just to be clear i'm using 2 TDM400's with 4 and 2 FXO modules respectively |
| 14:03.27 | MrChimpy | my e1s have come up in exactly the wrong way |
| 14:04.06 | DrkShdw | MACscr: the only drawback to that scenario, would be the latency across the pond. especially during weekends and holidays when international traffic peaks |
| 14:04.15 | MrChimpy | the ports labelled 1 and 2 on my TE411s are coming up as Card 0 span 3 and 4 and card 1 span 3+4! |
| 14:04.18 | *** join/#asterisk smackus (n=smackus@63.149.122.93) |
| 14:04.26 | MACscr | DrkShdw: what would be the best way to handle that? |
| 14:04.51 | DrkShdw | the *only* way to handle it, would be using a local asterisk machine ;-) |
| 14:05.18 | DrMouse | chimpy - go on! @ destroy south england :) |
| 14:05.35 | DrkShdw | anyway, as far as options.. any SIP / IAX capable voip phone, or adapter would work as an extension |
| 14:06.44 | smackus | question... under what circumstances would I get a timeout when dialing outbound long distance numbers? http://pastebin.ca/86057 I understand the error, there is no rule to handle the call if a timeout occurs, I just don't know what constitutes a time out. Looking for some advice. thanks |
| 14:07.11 | *** part/#asterisk Egonis (n=Egonis@207.245.14.10) |
| 14:08.22 | Zeeek | who dials? The php script? |
| 14:08.25 | DrMouse | DrkShdw - we are considering a very similar set up, but with UK central server and clients in Spain, I am recommending a local server but they want to keep costs down. i think it would work, but QOS may be an issue, and I am sure latency would... |
| 14:08.47 | smackus | Zeeek: correct |
| 14:09.00 | Zeeek | then the value is in that script |
| 14:09.06 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 14:09.13 | *** join/#asterisk Bert- (n=bert@bas33-1-82-66-4-198.fbx.proxad.net) |
| 14:09.24 | DrkShdw | DrMouse: there are a couple people I know doing exactly that. asterisk on one side, extensions on other. it's not too bad typically. though they do notice some jitter, and latency during weekends/holidays |
| 14:09.43 | Bert- | hello there |
| 14:10.01 | DrkShdw | <PROTECTED> |
| 14:10.10 | my007ms | pleas some one help |
| 14:10.17 | smackus | Zeeek: ok, cool... I will look it over. I am not familiar with what is in the script. Other developer wrote it, so I did not think to look there. Thanks |
| 14:10.21 | my007ms | i need to know sampel zaptel.conf |
| 14:10.28 | my007ms | i have TE205P |
| 14:10.36 | smackus | make samples will show you sample zaptel.conf |
| 14:10.51 | my007ms | i know all info in http://kb.digium.com/entry/1/93/ |
| 14:11.07 | smackus | also, the values for the TE205P can be found on the asterisk wiki |
| 14:11.11 | Bert- | I have to convert a soundfile (.ulaw) into another format, in order to work on it. I really don't know wich converter can I use. I tried thoose mentionned on voip-info.org, but no one deals with ulaw format. Does someone here knows a way to convert from ulaw to wav or gsm please ? |
| 14:11.35 | my007ms | i use /usr/local/sbin/genzaptelconf |
| 14:11.44 | my007ms | smackus, pleas send link |
| 14:11.54 | smackus | http://www.google.com |
| 14:12.01 | smackus | search for asterisk wiki |
| 14:12.14 | my007ms | so get line like # ??: 6 TE2/0/1/6 |
| 14:12.27 | brif8 | sip show peers lists all SIP peers is there one to show all extensions registered ? from extensions.conf ? |
| 14:12.28 | my007ms | how i will edit this line to be like my need is |
| 14:12.48 | *** join/#asterisk af_ (n=af@ip-170-209.sn1.eutelia.it) |
| 14:12.52 | [TK]D-Fender | brif8 : "show dialplan" |
| 14:13.00 | Zeeek | Bert- http://linuxcommand.org/man_pages/sox1.html |
| 14:13.21 | DrMouse | tell you what, ill give you my problem (which my original question was about)... I have an * server set up, on a separate LAN to the PCs (we dont support the LAN so we keep it separate), and I am getting hiss/crackle on the lines connected to the TDM400s (6 of them). either it is a problem with my config or a problem between the socket and the card (no problem if i plug a phone straight into the socket). any ideas? i can private msg my zapat |
| 14:13.26 | my007ms | is it all line will be like this span=1,1,0,esf,b8zs |
| 14:13.51 | my007ms | and coz i have 2 i will have tow line span=1,1,0,esf,b8zs span=2,1,0,esf,b8zs is that right |
| 14:14.22 | brif8 | [TK]D-Fender: thanks that shows my extensions, what about my incoming numbers that are also in my extensions.conf file ? I don't see them listed |
| 14:15.03 | my007ms | why there is no man zaptel.conf |
| 14:15.18 | [TK]D-Fender | brif8 : what is an "incoming" number in reference to extensions.conf? |
| 14:15.19 | DrMouse | also, the problem gets worse if i up the gains, but with them as they are people sometimes complain the line is too quiet. |
| 14:15.29 | Zeeek | my007ms there is |
| 14:15.35 | MACscr | DrkShdw: sry, i got pulled away on a call. I need to have a central PBX for handling particular things like call recording, hmm |
| 14:15.36 | Bert- | hihi |
| 14:15.40 | MACscr | oh the planning |
| 14:15.47 | Bert- | I found a bug on sox |
| 14:15.47 | Bert- | :) |
| 14:15.54 | my007ms | Zeeek, i try |
| 14:15.56 | *** join/#asterisk ghenry (n=ghenry@80.229.93.1.plusnet.pte-ag2.dyn.plus.net) |
| 14:15.57 | Bert- | first time I use it, it segfault :p |
| 14:15.58 | MACscr | lol, i swear im never going to get my voip solution implemented |
| 14:16.22 | brif8 | [TK]D-Fender: in extensions.conf I have exten 18009051552,1,Answer() 18009051552 is my "incoming" number |
| 14:16.29 | MACscr | TK]D-Fender: can i PM you? |
| 14:16.43 | Bert- | just remove extension of the infile then sox myinfile myoutfile.gsm -> segfault |
| 14:16.45 | MACscr | i have a question about your consulting |
| 14:16.50 | znoG | brif8: you probably want exten => s/18009051552,1,Answer() .... |
| 14:18.03 | brif8 | I meant exten => 18009051552,1,Answer() it works when you dial it jumps to that line a follows on through, but how do I show this show dialplan did not ? |
| 14:18.39 | Bert- | anyway |
| 14:18.47 | Bert- | sox seems unable to dealswith ulaw :( |
| 14:19.02 | *** join/#asterisk Juggie (n=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
| 14:19.04 | Zeeek | Bert look at the man page there are flags to override file types |
| 14:19.25 | Bert- | I am into :) |
| 14:21.11 | *** join/#asterisk hi365 (n=hi365@bzq-88-155-37-190.red.bezeqint.net) |
| 14:21.12 | *** part/#asterisk my007ms (n=noor@217.139.224.194) |
| 14:21.14 | *** join/#asterisk my007ms (n=noor@217.139.224.194) |
| 14:22.01 | *** join/#asterisk mihaib (n=mihai@slim-eth0.horizonlive.net) |
| 14:22.03 | my007ms | where i can copy and past my config file to let othere see my Q |
| 14:22.16 | Zeeek | ~pb |
| 14:22.18 | jbot | rumour has it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
| 14:22.20 | Bert- | hhmm sox: Failed reading pv_hold.ulaw: Do not understand format type: ulaw |
| 14:22.38 | mosty | bert: you need -t .ul |
| 14:22.40 | Zeeek | Bert you have to change the file name |
| 14:22.53 | Bert- | Zeeek I tried without extensions : segfault |
| 14:23.04 | Bert- | ok let me try options |
| 14:23.15 | Zeeek | and the flags? |
| 14:23.32 | *** part/#asterisk brif8 (n=Administ@ns1.ttienterprises.org) |
| 14:23.36 | Bert- | no flags |
| 14:23.47 | Zeeek | <PROTECTED> |
| 14:23.47 | Zeeek | <PROTECTED> |
| 14:23.47 | Zeeek | <PROTECTED> |
| 14:23.47 | Zeeek | <PROTECTED> |
| 14:23.48 | *** join/#asterisk |oranjia| (n=kvirc@dsl-146-10-188.telkomadsl.co.za) |
| 14:23.48 | DrMouse | [mouse looks around hopefully] anybody got any ideas? |
| 14:23.56 | [TK]D-Fender | bri8 : Thats jsut an exten... you see it like you should any other... |
| 14:23.57 | |oranjia| | hello everyone :) |
| 14:24.03 | [TK]D-Fender | MACscr : Why? |
| 14:24.03 | my007ms | http://bzflag.pastebin.ca/86062 |
| 14:24.24 | Bert- | haaa ? |
| 14:24.27 | MACscr | just wanted to know your rates again |
| 14:24.35 | my007ms | someone help in this http://bzflag.pastebin.ca/86062 |
| 14:25.18 | [TK]D-Fender | MACscr : ok |
| 14:25.40 | Bert- | ok |
| 14:25.43 | Bert- | got only noise :( |
| 14:26.37 | Zeeek | if there is no one in the forest, what is the noise of a segfault? |
| 14:27.11 | DrMouse | zeeek - kaboom? |
| 14:27.14 | Bert- | ? |
| 14:27.33 | Zeeek | Bert are you reading about u-law support in the man page? |
| 14:27.54 | Bert- | I 'm looking into that : http://linuxcommand.org/man_pages/sox1.html |
| 14:28.15 | Zeeek | but are you reading it or just looking into it? |
| 14:28.59 | |oranjia| | has anyone got experience using asterfax? |
| 14:29.21 | Bert- | I'm READING about files types for now |
| 14:29.27 | Zeeek | time to go home |
| 14:29.30 | *** part/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek) |
| 14:29.37 | Bert- | especially about that : .ub, .sb, .uw, .sw, .ul, .al, .lu, .la, .sl |
| 14:29.38 | Bert- | :) |
| 14:29.48 | Bert- | be sure I'll find :) |
| 14:31.57 | smackus | I am listening to my music on hold. It seems to sound fine, but then all of a sudden it sounds like the old FM radio with the turn nobs, and the radio is between channels. Also, I don't know if this is related but I get this error about the same time: Jul 12 08:29:34 WARNING[27584] interface.c: Junk at the beginning of frame 00000000. Any thoughts? |
| 14:32.01 | Bert- | yes ! :) |
| 14:32.05 | Bert- | It works ;) |
| 14:32.31 | Bert- | <mosty> bert: you need -t .ul <- I should have read that :( |
| 14:32.40 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@208-45-120-111.dlth.qwest.net) |
| 14:32.48 | hi365 | Hi! im trying to install a Sangom A200. After many tries i got the drivers installed and everything seems to be running ok, except that for some reson it seems that asterisk doesnt see the cahnnels (no trunks in FOP, out going calls get "all circuits are busy") Logs here: http://pastebin.ca/86065 |
| 14:33.13 | mosty | hi365: did you configure zapata.conf ? |
| 14:33.46 | [TK]D-Fender | hi365 : pastebin CLI output of a failed call |
| 14:34.15 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 14:34.34 | DrMouse | does anyone have any ideas? and should i pastebin my zaptel and zapata configs? |
| 14:35.19 | smackus | DrMouse: what was your question? |
| 14:35.57 | hi365 | [TK]D-Fender: http://pastebin.ca/86068 |
| 14:36.04 | DrMouse | <DrMouse> tell you what, ill give you my problem (which my original question was about)... I have an * server set up, on a separate LAN to the PCs (we dont support the LAN so we keep it separate), and I am getting hiss/crackle on the lines connected to the TDM400s (6 of them). either it is a problem with my config or a problem between the socket and the card (no problem if i plug a phone straight into the socket). any ideas? i can private ms |
| 14:36.04 | DrMouse | <DrMouse> also, the problem gets worse if i up the gains, but with them as they are people sometimes complain the line is too quiet. |
| 14:36.33 | hi365 | mosty: its right here: pastebin.ca/86065 |
| 14:37.03 | smackus | DrMouse: ah, sorry... no experience there. |
| 14:37.25 | DrMouse | thanks anyway smackus |
| 14:38.39 | [TK]D-Fender | hi365 : Your zapata.conf is wrong and you have no defined channels. the line labeled "channels" should be "channel" |
| 14:38.59 | Mw3 | hi folks. a little bit offtopic: is it true that there is no sim card in gsm phones in the USA ? |
| 14:39.02 | gandhijee | DrMouse: what board are you running the card in? |
| 14:39.29 | gandhijee | DrMouse: i've also seen things on the wiki that say changing the slot the card is in could help as well |
| 14:40.01 | hi365 | <[TK]D-Fender> Thanks! |
| 14:40.07 | benjk | no, all GSM phones require SIMs |
| 14:41.09 | hi365 | <[TK]D-Fender>, any idea why the trunks arent showing in FOP? |
| 14:42.27 | [TK]D-Fender | hi365 : Like I said, they aren't defined right. |
| 14:42.30 | *** join/#asterisk littlejohn (n=little@213-140-22-71.fastres.net) |
| 14:42.59 | hi365 | group=1 |
| 14:43.00 | hi365 | channel=1-4 |
| 14:43.08 | hi365 | is that wrong? |
| 14:43.20 | smackus | I only see this error, once in a while, so i am not terribly concerned about it... but does anyone know what causes it: |
| 14:43.22 | smackus | Jul 12 08:39:48 NOTICE[25937] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 |
| 14:43.22 | [TK]D-Fender | Better, but should also be "channel => 1-4" |
| 14:43.40 | [TK]D-Fender | smackus : Looks like frame slips maybe. |
| 14:44.01 | smackus | [TK]D-Fender: what is that? |
| 14:44.12 | [TK]D-Fender | smackus : Timing issues with telco |
| 14:44.28 | smackus | periodically i get a file.c: faile to write frame, is that the same thing? |
| 14:44.57 | [TK]D-Fender | smackus : Not sure... never seen that one and the message is ambibuos. |
| 14:45.08 | smackus | so the timing issue is just a problem on the telco side maybe? or would that be something to look for and address on asterisk? |
| 14:45.20 | hi365 | [TK]D-Fender channel => 1-4 but nothing in FOP |
| 14:45.35 | [TK]D-Fender | check that your card has its own IRQ, and run zttest |
| 14:45.44 | [TK]D-Fender | hi365 : Did you completely reload *? |
| 14:46.00 | *** join/#asterisk af_ (n=af@ip-170-209.sn1.eutelia.it) |
| 14:46.01 | [TK]D-Fender | hi365 : Also I have no idea if FOP is even set up right. |
| 14:46.01 | smackus | hi365: was also going to ask about ztcfg -vv, have you run that? what is your output? |
| 14:46.34 | [TK]D-Fender | hi365 : He has and it was in his pastebin, but it seems his problem was Zapata related, not zaptel. |
| 14:46.45 | smackus | gotcha |
| 14:46.46 | [TK]D-Fender | smackus rather. |
| 14:46.51 | smackus | yeah |
| 14:46.53 | smackus | i knwo |
| 14:46.55 | smackus | know |
| 14:47.34 | *** join/#asterisk ToyMan (n=stuq@74-32-9-135.dsl1.mdl.ny.frontiernet.net) |
| 14:51.57 | *** join/#asterisk ionix (n=ionix@p2168-ipbfp03miyazaki.miyazaki.ocn.ne.jp) |
| 14:52.11 | DrMouse | gandhijee - i am running on a via epia M10000 board, and I cant change the slots as there are only 2 (well, one, but designed to run 2 cards with a riser) |
| 14:54.38 | *** join/#asterisk Qwell[laptop] (n=Qwell[]@unaffiliated/qwell) |
| 14:56.02 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
| 15:04.03 | *** join/#asterisk Ateboy (n=ugob@modemcable002.152-81-70.mc.videotron.ca) |
| 15:05.58 | Ateboy | Hi, is there a way to have an option in the voicemail of an user so that if someone presses a series of digits while the unavailable or busy messge pays, it does a dial to his cell phone? |
| 15:06.13 | ionix | not with comedian mail |
| 15:06.33 | *** join/#asterisk Modcuts (n=bob@lan.proporta.com) |
| 15:06.33 | Ateboy | ok, but there is an option to dial the 0 to get the receptionist, right? |
| 15:06.38 | [TK]D-Fender | Ateboy : Yes you can. |
| 15:07.04 | [TK]D-Fender | Use the "a" exten in the same context as your dial command and exit using "*" while the message is playing. |
| 15:07.21 | [TK]D-Fender | Ateboy : "o" is more complicated to set up. |
| 15:07.36 | *** join/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com) |
| 15:07.57 | [TK]D-Fender | Ateboy : et..... salut mon ostie! ;) |
| 15:08.13 | Ateboy | TK: tu t'en viens un vrai quebecois? |
| 15:08.41 | [TK]D-Fender | Ateboy: the "o" (operator) requires you to set up funny stuff in voicemail.conf on a per box basis. |
| 15:09.11 | [TK]D-Fender | Ateboy : Assez ;) |
| 15:09.51 | [TK]D-Fender | Ateboy : Faut rester ouvert au monde.... meme celui d'ou tu viens. |
| 15:12.04 | my007ms | someone help in this http://bzflag.pastebin.ca/86062 |
| 15:14.47 | mosty | why does * hangup the first time in this trace? the caller's phone just rings and eventually picks up. http://pastebin.ca/86089 |
| 15:16.27 | Ateboy | TK: where can I find a list of all those new extensions I don't know about? |
| 15:16.54 | *** join/#asterisk chapeaurouge (n=chapeaur@80.92.83.35) |
| 15:16.54 | [TK]D-Fender | Ateboy : Its all in the WIKI since forever. Just lookup "standard extensions". |
| 15:17.47 | Ateboy | ok thanks |
| 15:19.08 | *** part/#asterisk alephcom (n=alephcom@h11.n66-112-182.mcsnet.ca) |
| 15:19.09 | feld_ | I just installed a second TDM400P in my Asterisk box.... it's not showing up; it does have power... is this some of that IRQ conflicts crap most likely? |
| 15:19.44 | *** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com) |
| 15:19.48 | [TK]D-Fender | feld_ : modprobe'd it twice and doesn't show up in cat /proc/interrupts? |
| 15:19.54 | *** join/#asterisk hohum (n=dcorbe@12.195.58.237) |
| 15:23.26 | mosty | anyone have any ideas? |
| 15:23.29 | feld_ | hrm silly me.... it's no getting modprobe'd twice. let me play with that....... |
| 15:23.36 | feld_ | err wait |
| 15:23.45 | feld_ | [TK]D-Fender: shouldn't it show up in lspci regardless? |
| 15:24.31 | chapeaurouge | hi all... im looking for a fxs/fxo channel bank to use with *... any advice? |
| 15:24.35 | [TK]D-Fender | feld_ : Not sure, never looked for it that way before |
| 15:25.19 | feld_ | it _should_ show in lspci... thats how you find out what devices are before you modprobe the correct modules. |
| 15:25.33 | *** join/#asterisk _problem_ (n=lokesh_k@estrela.nortenet.pt) |
| 15:26.17 | [TK]D-Fender | feld_ : Well for these I alrady know the module names so I "just do it" and tend to be successful. I ended up double mod-probing for X100P's in a system I did some work on to pick up the 2nd card. |
| 15:26.34 | *** join/#asterisk NDT (n=nunya@cpe-24-195-66-214.nycap.res.rr.com) |
| 15:27.25 | feld_ | lspci |
| 15:27.30 | feld_ | wrong window..... |
| 15:28.14 | NDT | I am in a macro...and trying to jump out of it and go back to a specific place in context default... yet when I jump out of it with: exten => 411,1(denied),Set(MACRO_RESULT=GOTO:default^112^1) it takes me back to the beginning of default. Actually it reverses the call and puts the callee at the beginning of the call hehe...Am I jumping out of this thing correctly? |
| 15:28.22 | *** join/#asterisk argos73 (n=mike@w010.z208036240.chi-il.dsl.cnc.net) |
| 15:29.13 | *** join/#asterisk fgwaller (n=frank@65.105.5.40.ptr.us.xo.net) |
| 15:29.15 | [TK]D-Fender | NDT : What are you in anything other than "s" in a macro in the first place? |
| 15:29.26 | [TK]D-Fender | NDT : Pastbin the whole thing |
| 15:30.43 | NDT | http://pastebin.ca/86104 |
| 15:31.25 | *** join/#asterisk tlow (n=tlowe@bgp.terrorist.net) |
| 15:34.32 | *** join/#asterisk GyrosGeier (n=richter@p54995F5B.dip.t-dialin.net) |
| 15:34.34 | GyrosGeier | hi |
| 15:35.42 | *** join/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 15:35.53 | *** join/#asterisk knarfly_wk (n=bwatson@12.42.132.26) |
| 15:36.03 | eKo1 | Will TE410P work in a 2.2V PCI-X slot? |
| 15:36.19 | GyrosGeier | probably a FAQ, but I cannot find it right now: My SIP provider refuses to authenticate to me, so if I send back a 407 response to an INVITE, the call is dropped. How can I make asterisk accept incoming INVITEs? |
| 15:36.33 | tlow | It needs 3.3 |
| 15:36.43 | GyrosGeier | I already have "insecure = yes" with the peer definition, but that doesn't help |
| 15:38.02 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 15:38.10 | Dr-Linux | hi all |
| 15:38.29 | Dr-Linux | well Qwell |
| 15:38.37 | eKo1 | err, I meant to say 3.3V |
| 15:38.57 | eKo1 | I just want to know if it will work with PCI-X. |
| 15:39.00 | *** part/#asterisk smackus (n=smackus@63.149.122.93) |
| 15:39.02 | Ateboy | For the asterisk presence, if I want to add it to my config, it seems just as simple as adding a line like exten => 802,hint,SIP/username on each of my internal extension definition? |
| 15:39.16 | Dr-Linux | eKo1: you are telling about Windows Vista versiion? 3.3V |
| 15:39.36 | Strom_C | good morning |
| 15:39.50 | Dr-Linux | Strom_C: good morning to you |
| 15:40.05 | Dr-Linux | Strom_C: please tell me "good evening" |
| 15:40.16 | *** part/#asterisk littlejohn (n=little@213-140-22-71.fastres.net) |
| 15:41.03 | St1ckm4n | NDT: what does your asterisk cgi output show when when it gets to that step? |
| 15:41.28 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 15:41.50 | NDT | I may have fixed it one sec while I check heh |
| 15:42.58 | [TK]D-Fender | Ateboy : pretty much. However none of the devices you had at the point in time I did that install for you support it. You could use the info somwhere else though like for a web-panel for status, etc, but then again there are other ways of getting it also. |
| 15:43.21 | eKo1 | Dr-Linux: No no no, 3.3V PCI-X |
| 15:43.55 | Dr-Linux | eKo1: o i see |
| 15:44.28 | Ateboy | I see |
| 15:44.40 | fnordian | hi |
| 15:44.41 | Ateboy | TK: so there is no point doint that |
| 15:45.17 | fnordian | can anybody tell me, why voip-info.org suggests a field "customer_id" for the voicemail's realtime table? |
| 15:46.04 | Katty | allo. |
| 15:46.13 | NDT | blah...nope not fixed... MACRO_RESULT=GOTO:default^112^1 <--is that really the way to GOTO out of a macro? |
| 15:46.41 | *** join/#asterisk mountainm2k (n=mountain@cbit-98.bullseye9.com) |
| 15:47.02 | St1ckm4n | yeah it looks right, what does that denied do by 411,1 I haven't used that before |
| 15:47.51 | *** join/#asterisk smackus (n=smackus@63.149.122.93) |
| 15:48.04 | NDT | just a tag on the priority... |
| 15:48.25 | St1ckm4n | you can try MACRO_RESULT=ABORT to see if that does what you need, you're just trying to hangup the call right? |
| 15:49.25 | NDT | Well on one option yes...on another no... |
| 15:50.01 | smackus | what does the number in Warning [] represent? Is it process id? http://pastebin.ca/86120 beacuse my asterisk is running on pid 25921, and the warning on all of my deadlocks is 25925. |
| 15:50.10 | Katty | don't everyone say hi at once ;) |
| 15:50.16 | mountainm2k | hi |
| 15:50.17 | *** join/#asterisk JohnJacob (n=m00p@pool-71-127-102-43.aubnin.fios.verizon.net) |
| 15:50.24 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 15:50.30 | anthm | hi at twice |
| 15:50.55 | smackus | when i do a "ps -ef | grep 25925" i get: [root@slk-apbx-01 ~]# ps -ef | grep 25925 |
| 15:50.55 | smackus | root 1337 22046 0 09:50 pts/12 00:00:00 grep 25925 |
| 15:50.55 | smackus | [root@slk-apbx-01 ~]# |
| 15:50.55 | Katty | hi mountainm2k (= |
| 15:50.58 | Katty | anthm: hey hun. |
| 15:51.09 | Katty | anthm: fop is pretty neat. |
| 15:51.14 | Katty | anthm: when it works... |
| 15:51.37 | anthm | =D |
| 15:52.05 | St1ckm4n | NDT: have you tried setting it to something other than default,112,1 maybe make a new context just for that or try default^111^4 see if it still goes to the beginning |
| 15:52.37 | fgwaller | hi (slow) |
| 15:52.43 | Katty | fgwaller: (= |
| 15:52.54 | NDT | yeah I am going to try sending it to a new context now |
| 15:52.54 | *** join/#asterisk ahsan (n=ahsan@host-83-146-53-46.bulldogdsl.com) |
| 15:53.04 | Katty | it's possible to make asterisk send out voicemail notifications with the actual voicemail as an attachment, yes? |
| 15:53.30 | mountainm2k | yes |
| 15:53.39 | Katty | how big of file sizes are we talking? |
| 15:53.39 | Qwell[laptop] | attach=yes |
| 15:53.45 | Katty | Qwell[laptop]: thanks. |
| 15:54.16 | fgwaller | is anyone beside on me running current SVN HEAD here? |
| 15:54.55 | Qwell[laptop] | svn head of what? trunk? |
| 15:55.12 | Katty | anthm: what general size range would a voicemail attachment be? |
| 15:55.20 | fgwaller | current development |
| 15:55.35 | Katty | anthm: 300k, 2mb..?? |
| 15:55.50 | anthm | if you used gsm wav it would be like a few hundred k |
| 15:55.56 | Katty | kk |
| 15:56.09 | St1ckm4n | also NDT, I just saw a note about Macros check out this link Note #2 http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Macro |
| 15:56.34 | St1ckm4n | I would watch * cgi to see if you are making it past this step # |
| 15:56.34 | St1ckm4n | exten => s,n,GotoIf($[${ACCEPT} = 5 ]?411,1) |
| 15:59.53 | eKo1 | Will TE410P work in a 3.3V PCI-X slot? |
| 16:00.00 | I-MOD | yes |
| 16:01.11 | mountainm2k | Anybody know, offhand, of a good source of info for Realtime (isn't that the database driven config stuff)? Presumably something in the wiki, hoped somebody could point me towards it... |
| 16:02.21 | terrapen | anybody ever seen Call Parking suddenly stop working? |
| 16:02.31 | terrapen | we've made no configuration changes |
| 16:03.10 | terrapen | now, when the user dials #999 (our command for parking), the caller hears the DTMF tones |
| 16:03.17 | terrapen | and the call isn't parked |
| 16:03.23 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 16:03.27 | terrapen | hey brad |
| 16:03.49 | St1ckm4n | NDT: sorry gotta go |
| 16:04.03 | brad_mssw | terrapen: sup man |
| 16:04.08 | terrapen | oh nada |
| 16:04.17 | brad_mssw | terrapen: ever get that bike ? |
| 16:04.18 | terrapen | same old same old |
| 16:04.20 | terrapen | yeah! |
| 16:04.23 | brad_mssw | terrapen: what was it, a Fuel, right ? |
| 16:04.40 | *** join/#asterisk MatsK (i=MatsK@83.233.97.229) |
| 16:04.41 | terrapen | Fuel EX 9.5 project one |
| 16:04.52 | brad_mssw | sweet |
| 16:04.57 | eKo1 | I-MOD: if it runs on 66 MHz? |
| 16:05.09 | brad_mssw | terrapen: how's it ride? |
| 16:05.13 | I-MOD | the card will pull the bus down to 33mhz |
| 16:05.15 | terrapen | http://flickr.com/photos/defender90/145923844/ |
| 16:05.17 | I-MOD | but it will run |
| 16:05.21 | terrapen | dude, it's sweeeeeet |
| 16:05.29 | terrapen | but I have to get new brakes |
| 16:05.42 | So3kris | Is is poseble to get a good running system with remote users behind nat ? |
| 16:05.55 | terrapen | the other day, I climbed 3000' vertical...and on the descent, I boiled my brake fluid :) |
| 16:06.01 | brad_mssw | Got disc brakes I see, they wear out, or just not good quality? |
| 16:06.09 | brad_mssw | ah... heh |
| 16:06.41 | terrapen | they're Avil Juicys...supposed to be awesome...but apparently the smaller rotors that I got were not designed for long, steep descents |
| 16:06.53 | terrapen | so i'm ordering some 203mm rotors and adapters |
| 16:07.10 | brad_mssw | ah |
| 16:07.14 | terrapen | have you been riding |
| 16:07.32 | brad_mssw | yep, every day |
| 16:07.33 | fgwaller | after svn update menueselect does not recognice zaptel anymore but it is installed in current version |
| 16:07.51 | brad_mssw | terrapen: my good ol' Trek 4500 is hanging in there though |
| 16:07.59 | terrapen | sweeet |
| 16:08.03 | brad_mssw | terrapen: can't really justify a Fuel class bike |
| 16:08.10 | terrapen | i lent my 970 to my friend and he rides the hell out of it |
| 16:08.22 | terrapen | your 4500 is probably lighter than mine anyway, brad |
| 16:08.33 | terrapen | the dual suspension is *heavy* |
| 16:08.34 | *** join/#asterisk SplasPood (n=jwb@gate.lga2.us.voxel.net) |
| 16:09.20 | brad_mssw | terrapen: heh, don't know about that one ... I actually bought a used bike of cdf here (Gary Fisher Sugar 4 full suspension) b/c my girlfriends brother was in town and wanted to ride |
| 16:09.32 | terrapen | wow |
| 16:09.35 | terrapen | thats a sweet bike |
| 16:09.39 | eKo1 | I-MOD: thanks for the explanation |
| 16:09.51 | brad_mssw | terrapen: eh, it's old now though, still in good shape |
| 16:10.28 | brad_mssw | terrapen: don't care for the feel of it too much either, it has you more bent over than a Trek |
| 16:11.17 | terrapen | ah |
| 16:11.23 | terrapen | it has to be hot as all hell down there now |
| 16:12.03 | brad_mssw | terrapen: yeah, usually in the 90's when I ride, you get used to it though, and at least it's through woods where I ride, so it's not in full sun |
| 16:13.06 | terrapen | this is so odd....call parking just stopped working on this PBX |
| 16:13.57 | terrapen | it has something to do with DTMF detection |
| 16:14.03 | brad_mssw | heh, that's not as odd as asterisk going into a ring loop on us the other day ... no one could answer calls, but all the phones would ring, it was weird ... had to forcibly restart asterisk |
| 16:14.44 | terrapen | haha |
| 16:16.18 | *** join/#asterisk skraelings001 (n=skraelin@201.230.239.132) |
| 16:16.24 | skraelings001 | hi everyone |
| 16:17.06 | Hmmhesays | ip tables on a linksys |
| 16:17.10 | Hmmhesays | yeaahyaaah |
| 16:18.09 | darviria | do I need to put anything in the conf files about my wireless router (ie ip address) when i'm configuring asterisk for sip phones? |
| 16:19.45 | *** join/#asterisk froguz (n=xxxxx@pc-95-155-104-200.cm.vtr.net) |
| 16:20.16 | froguz | does the X100P clones support Rx and Tx gain feature? |
| 16:21.36 | darviria | ...actually forget that, i'm now getting failed requests come up on the server |
| 16:22.28 | skraelings001 | hm.. does "In Use" in AVAILSTATUS means that the're is a current call in that channel ? |
| 16:25.52 | smackus | so would WARNING[25925] refer to process id 25925? |
| 16:29.24 | froguz | please, can somebody tell me if X100P clones support TxGain and RxGain? |
| 16:29.30 | Strom_C | froguz: they should |
| 16:30.10 | Dovid | using a sangoma a200-r |
| 16:30.19 | Dovid | i do not install zaptel correct ? |
| 16:31.16 | Dovid | ? |
| 16:31.21 | froguz | thanks Strom_C |
| 16:31.50 | benjk | froguz, they do but if not without a catch 22 |
| 16:32.10 | benjk | if you increase gain, you are likely to get echo problems |
| 16:33.08 | [TK]D-Fender | Docelm0 : Yes you do. |
| 16:33.13 | [TK]D-Fender | Dovid : rather |
| 16:33.26 | Dovid | tk |
| 16:33.29 | [TK]D-Fender | Dovid : You compile zaptel, then wanpipe, then zaptel again |
| 16:33.29 | *** join/#asterisk meni (n=meni@62.90.116.95) |
| 16:33.37 | Dovid | ok |
| 16:33.42 | Dovid | so make install zaptel |
| 16:33.43 | froguz | benjk, what is a catch 22? |
| 16:33.56 | Dovid | then wanpipe then zaptel again ? |
| 16:34.34 | [TK]D-Fender | Dovid : No, don't install zaptel. do make then do wanpipe's install which will offer to rerun make for you after patching it. |
| 16:35.04 | Dovid | ok so let me get it straight |
| 16:35.09 | benjk | the catch is that if you lower the gain, the volume may be too low to have a normal conversation and if you raise it to the level you really need you will get echo issues and other sound artifacts |
| 16:35.33 | Dovid | i do zaptel make not make install then wan pipe which patches zaptel then make or make install zpatel ? |
| 16:35.50 | Dovid | any url for instructions on the web ? i dont like sangoma's gui |
| 16:35.52 | froguz | aha... so i'll need to play a lot with such values |
| 16:35.56 | Dovid | wiki* |
| 16:35.58 | froguz | thanks |
| 16:36.02 | [TK]D-Fender | Dovid : correct |
| 16:36.25 | MatsK | froguz: Google is your friend http://en.wikipedia.org/wiki/Catch-22_%28logic%29 |
| 16:36.26 | [TK]D-Fender | Dovid : their wiki is pretty well laid out and the install process rather simple. |
| 16:36.38 | Dovid | ok will poke arund |
| 16:36.39 | Dovid | :) |
| 16:37.49 | benjk | froguz, you can tweak settings for analog modems for weeks and months and never get it right |
| 16:37.58 | benjk | those cards are just no good |
| 16:38.16 | benjk | its ok for a home pbx or for testing in a lab |
| 16:38.22 | mosty | benjk: is there a good solution for analogue lines? hardware voice modems? |
| 16:39.01 | benjk | I should think that the proper multi-port analog cards designed to be voice cards from Digium and Sangoma deliver better quality |
| 16:39.57 | mosty | but aren't they just multiple instances of the same crappy hardware? |
| 16:40.05 | benjk | yet if it is for business phone system, I would rather thing analog shouldn't be used at all |
| 16:40.26 | benjk | not sure about the Digium TDM400 card, it may well be, yes |
| 16:40.42 | mosty | in some countries, it's very very expensive to get digital lines |
| 16:40.43 | *** join/#asterisk THX2000 (i=AgentFLY@adsl-66-51-192-221.dslextreme.com) |
| 16:40.50 | benjk | but I think the 2400 and also Sangoma's card use more professional parts |
| 16:41.09 | benjk | yes, I know, but analog is nevertheless a hit and miss thing |
| 16:41.18 | THX2000 | is west coast teliax down? |
| 16:41.34 | So3kris | The TDM400 is a good card. I have seen multiple working at companys |
| 16:42.44 | benjk | well, in Japan that card doesn't cut it at all |
| 16:42.57 | So3kris | Oh |
| 16:42.58 | benjk | but most analog gear doesn't do well here |
| 16:43.07 | So3kris | I can say it only for Holland :D |
| 16:43.07 | benjk | only the Japanese stuff from NEC |
| 16:43.14 | eKo1 | analog sucks |
| 16:43.19 | benjk | 6000 USD for 8 analog ports |
| 16:43.26 | benjk | that works rather well |
| 16:43.30 | eKo1 | gatorade is better |
| 16:45.45 | *** join/#asterisk chorlick (n=Chris@63.81.26.126) |
| 16:45.53 | *** join/#asterisk KranZ (n=user@imail.bestline.net) |
| 16:45.53 | *** part/#asterisk KranZ (n=user@imail.bestline.net) |
| 16:45.55 | *** join/#asterisk KranZ (n=user@imail.bestline.net) |
| 16:46.21 | *** join/#asterisk MonkeyHugs (n=MonkeyHu@63.149.122.93) |
| 16:46.44 | Strom_C | chorlick! |
| 16:46.54 | Strom_C | long time no see |
| 16:47.02 | chorlick | Dude im right next to you. |
| 16:47.25 | chorlick | :) |
| 16:47.28 | Strom_C | :D |
| 16:47.32 | Hmmhesays | Sure you are |
| 16:48.09 | *** join/#asterisk JackEStorm (n=thinkthi@ip68-225-72-125.no.no.cox.net) |
| 16:52.21 | smackus | where is monkey lover? |
| 16:52.26 | THX2000 | for a mission critical setup, who are you guys using for a voip provider? |
| 16:52.51 | Strom_C | THX2000: none. For mission-critical stuff, use a PRI |
| 16:52.52 | KranZ | iphone - > [g726] -> * -> [g726] -> sip provider would use less cpu than if ulaw was coming from the ipphone, right? |
| 16:53.18 | *** join/#asterisk tekati (n=captain@cpe-66-75-214-16.bak.res.rr.com) |
| 16:53.33 | KranZ | since there isn't any transcoding involved |
| 16:53.43 | Strom_C | KranZ: correct |
| 16:54.13 | KranZ | basically offloads it onto the phone |
| 16:54.30 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.128.Dial1.SanJose1.Level3.net) |
| 16:54.43 | THX2000 | is that just a limitation of the technology or is it just that no one has got it right yet? |
| 16:54.57 | vader-- | the first part |
| 16:55.04 | *** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
| 16:55.10 | KranZ | it's because tdm is more reliable than packet switched |
| 16:55.32 | Strom_C | THX2000: technology...with PRI, you've got a direct link to the PSTN. With VoIP, you're relying on the Internet to route your packets to the ITSP |
| 16:55.39 | KranZ | there are certain guarantees with tdm you can't get going over a public packet network |
| 16:56.02 | KranZ | like a direct link and bandwidth |
| 16:56.02 | Strom_C | THX2000: if you had a private packet link to the ITSP, then VoIP would work just as well |
| 16:56.20 | *** join/#asterisk piper69 (n=piper69@69.155.81.24) |
| 16:56.25 | eKo1 | KranZ: you mean with circuit switching vs packet switching |
| 16:56.25 | piper69 | hi all |
| 16:56.38 | *** join/#asterisk Assid (i=assid@203.115.83.214) |
| 16:56.42 | KranZ | eKo1: werd |
| 16:56.49 | piper69 | i am fairly new here and i would like some guide please |
| 16:56.53 | *** part/#asterisk Ateboy (n=ugob@modemcable002.152-81-70.mc.videotron.ca) |
| 16:57.17 | piper69 | i also want to ask for ur opinion on a matter |
| 16:57.22 | Strom_C | piper69: blue/white is pair 1. orange/white is pair 2. |
| 16:57.32 | THX2000 | Strom_C: Are people doing that? Doesn't sound like it would really be cost effective |
| 16:57.43 | KranZ | i laughed at this one person (to myself) who was using dsl and ipphones in texas which registered with a voip provider in virginia |
| 16:57.45 | Assid | you making a straihgt connection? or cross over? |
| 16:57.50 | KranZ | she was complaining about bad quality |
| 16:58.12 | piper69 | Strom_C: WHAT DO U MEAN? |
| 16:58.18 | *** part/#asterisk smackus (n=smackus@63.149.122.93) |
| 16:58.21 | piper69 | Strom_C: opps sorry for the CAPS |
| 16:59.18 | Assid | piper69: you making a straight patch cable or cross over? |
| 16:59.22 | *** part/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
| 16:59.24 | Strom_C | THX2000: cheaper means nothing if the calls are unreliable |
| 16:59.28 | Strom_C | Assid: it was a joke |
| 16:59.59 | Assid | oh |
| 17:00.08 | *** join/#asterisk rollergrrl (n=0x3e44d@71-213-5-22.slkc.qwest.net) |
| 17:00.12 | Strom_C | he asked for help without asking a specific question, so I answered with a specific answer |
| 17:00.14 | THX2000 | Strom_C: Agreed, i meant if you've got a direct link to the ITSP you're gunna be paying more then just stickin with pstn |
| 17:00.30 | Strom_C | THX2000: it depends on call volume |
| 17:01.06 | Assid | what do you guys use first.. orange-white orange or reverse? |
| 17:01.09 | KranZ | THX2000: and you can use the same link as your broadband connection |
| 17:01.10 | *** part/#asterisk ahsan (n=ahsan@host-83-146-53-46.bulldogdsl.com) |
| 17:01.10 | piper69 | i have put some $10 into my yahoo acct and tried to call an international call to a country ...it charged me $1 for less than a minute, but when i had my friend login from my acct in that country and called me on my phone it charged him only $0.03 / min????? |
| 17:01.12 | THX2000 | Strom_C: Are there providers who specialize in this sort of arena? |
| 17:01.38 | KranZ | THX2000: where are you? |
| 17:01.41 | *** join/#asterisk AlexCTI (n=alex@adsl-074-238-025-003.sip.mia.bellsouth.net) |
| 17:01.45 | THX2000 | KranZ: SoCal |
| 17:02.02 | Strom_C | piper69: yahoo? |
| 17:02.06 | KranZ | hmm... if you were in austin.. |
| 17:02.16 | Strom_C | piper69: I didnt know yahoo was an ITSP that integrated with asterisk |
| 17:02.22 | THX2000 | KranZ: I've just got some pretty decent sized clients that are really interested in voip, i just havnen't seen a voip provider that i would be willing to move them over to |
| 17:02.31 | Strom_C | THX2000: I'm in Los Angeles |
| 17:02.47 | *** join/#asterisk Assid (i=assid@203.115.83.214) |
| 17:02.54 | THX2000 | Strom_C: Orange County, Anaheim Hills |
| 17:02.56 | piper69 | <PROTECTED> |
| 17:03.09 | Strom_C | THX2000: ah cool |
| 17:03.19 | AlexCTI | Hi everyone, quick question Can I put spaces into the brakets [] when i create queues? eg: [my queue] or must me one word [my-queue]? |
| 17:03.19 | Strom_C | THX2000: how many stations per location? |
| 17:03.47 | KranZ | THX2000: you might want to look into colo'ing a * server somewhere and having them register to your server, then you have pri's connected to the pstn |
| 17:03.48 | THX2000 | Strom_C: One is a mortgage company that currently has 30 pstn lines coming in |
| 17:04.05 | KranZ | THX2000: are they T1? |
| 17:04.13 | THX2000 | THX2000: Yea |
| 17:04.18 | THX2000 | lol |
| 17:04.22 | KranZ | good, i was about to fall out of my chair |
| 17:04.22 | THX2000 | KranZ: yes :P |
| 17:04.38 | syzygybsd | mmm.. talking to yourself I see |
| 17:04.41 | THX2000 | lol |
| 17:04.46 | THX2000 | i do that occasionally |
| 17:05.11 | syzygybsd | i do it alot.. I call it talking to my computer, but that is a lie... |
| 17:05.32 | THX2000 | lol |
| 17:05.48 | Strom_C | THX2000: I have lots of experience dealing with those kinds of clients |
| 17:06.01 | THX2000 | Strom_C: He's mine, u can't have him :P |
| 17:06.18 | Strom_C | im not trying to steal your clients |
| 17:09.55 | syzygybsd | can i have him? |
| 17:11.02 | *** join/#asterisk bmg505 (n=leon@c1-203-1.rndf.isadsl.co.za) |
| 17:11.09 | So3kris | Has someone experiance with the spa-3102 from linksys. I recive on my asterisk box the pstn in comming calls. But i can't dial pstn out |
| 17:11.46 | Strom_C | So3kris: I was actually just about to ask that same question on behalf of a client :) |
| 17:11.51 | syzygybsd | lol.. sipura model numbers are still used in linksys, I woulda thought they renumberd everything |
| 17:12.45 | *** join/#asterisk p1p (i=pip_@64.200.16.26) |
| 17:13.29 | So3kris | Strom_C: ? |
| 17:13.37 | Strom_C | So3kris: ? |
| 17:13.50 | So3kris | The problem i don't understand it :D |
| 17:14.12 | Strom_C | I'm a consultant. A client gave me an SPA-3102 and said "make it work." |
| 17:14.22 | Strom_C | I have no experience with the SPA-3102 |
| 17:14.59 | *** join/#asterisk ahsan (n=ahsan@host-83-146-53-46.bulldogdsl.com) |
| 17:15.03 | So3kris | Strom_C: http://voxilla.com/spa3kasterisk.php there is a automatic config tool |
| 17:15.23 | So3kris | but works not fully good. but incomming you can put to youre asterisk |
| 17:15.23 | Strom_C | automatic config doesn't teach me a damned thing |
| 17:15.25 | syzygybsd | lol.. automatic config.. sounds wonderful like it works perfect everytime... |
| 17:15.41 | So3kris | Strom_C: you can see the config at the sipura and the asterisk |
| 17:15.42 | Strom_C | I want to learn how to do it, not have some tool do it for me :) |
| 17:16.48 | So3kris | Strom_C: it's help you with the. where mus i change the stuff |
| 17:17.14 | syzygybsd | So3kris: so which direction are you not able to make calls? |
| 17:17.30 | syzygybsd | sip phone -> asterisk -> pstn? |
| 17:18.11 | Strom_C | syzygybsd: yes, thats where I'm having problems |
| 17:18.13 | So3kris | pstn-> asterisk-> ATA@the3102 |
| 17:18.29 | Strom_C | the SPA keeps spitting back 404 |
| 17:18.58 | syzygybsd | lol.. k, that message seems like it is comming from the wrong direction... |
| 17:19.37 | [TK]D-Fender | Strom_C : You are using it as the primary PSTN-in for *? |
| 17:19.39 | syzygybsd | are you having the same issue? |
| 17:20.14 | syzygybsd | are you tring to make the 3102 ring? both of you.... |
| 17:20.57 | Strom_C | I'm trying to dial out over the POTS line |
| 17:21.11 | syzygybsd | k, so oposite problems.... |
| 17:21.19 | syzygybsd | Strom_C: where do you see the 404 message? |
| 17:21.24 | Strom_C | in the SIP debug |
| 17:21.39 | syzygybsd | um... k... can you pastebin that? |
| 17:21.45 | Strom_C | sure thing |
| 17:21.48 | Strom_C | one moment please |
| 17:22.11 | piper69 | Strom_C: i am back, what i was asking could it be done to embend yahoo with asterisk since now you can pay $2 /month and yahoo will assign u a number |
| 17:22.17 | syzygybsd | throw up your extensions.conf too, and the context in your sip.conf for your 3102 |
| 17:22.23 | [TK]D-Fender | Strom_C : have the PSTN line register to * as a user, and dial it like Dial(SIP/SPA-PSTN/${EXTEN}) |
| 17:23.05 | syzygybsd | OHHHHH.... is that how he has it setup... |
| 17:23.23 | syzygybsd | pstn -> 3102 -> asterisk -> ???? |
| 17:23.28 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 17:23.39 | [TK]D-Fender | syzygybsd : Thats the points of the 3102.... PSTN |
| 17:24.04 | Strom_C | syzygybsd: http://pastebin.ca/86203 |
| 17:24.09 | syzygybsd | heh, shows what I know.. thought it was to hookup to any analog device... |
| 17:24.27 | Strom_C | hold for extensions.conf |
| 17:24.36 | syzygybsd | was just thinking it was an ATA, should probably look it up first :( |
| 17:24.54 | syzygybsd | wait.. it is... |
| 17:25.07 | [TK]D-Fender | Strom_C : inbound routing takes 3-4 tiny options. You'll typically tell it what * exten to dial (like a DID) so that you know what device is calling into *, and set the dialplan (SPA) to automatically throw all calls direct to * using PSTN CID as its own. |
| 17:25.46 | Strom_C | no no, it's OUTBOUND that's the problem |
| 17:25.48 | Strom_C | gah! |
| 17:26.27 | Strom_C | http://pastebin.ca/86206 |
| 17:26.37 | *** join/#asterisk rene- (n=rene-@dsl-200-67-175-250.prod-empresarial.com.mx) |
| 17:26.39 | Strom_C | (sip.conf and extensions.conf) |
| 17:27.05 | rene- | why asterisk takes so long in reporting me manager events such as agentconnect and agentcomplete? |
| 17:27.22 | piper69 | so can i get any feed back regarding yahoo |
| 17:27.25 | [TK]D-Fender | Strom_C : OH, and the default port for the PSTN "line" on it is 5061, which I think is whats throwing you off. You would be dialing the FXS port with that setup if I'm not off my mark today... |
| 17:27.58 | *** join/#asterisk Johnsie (n=john@pdpc/supporter/active/Johnnie) |
| 17:28.52 | Strom_C | lemme try 5060... |
| 17:29.10 | *** join/#asterisk ghenry (n=ghenry@ghenry.plus.com) |
| 17:29.10 | Strom_C | it's set on 5060 |
| 17:29.18 | rene- | somebody told me that events sometimes get buffered in asterisk, |
| 17:29.34 | piper69 | also anyone here using Primus VOIP iAN-02EX my friend gave me a brand new one but for some resaon i can't get the right usr/password combination |
| 17:29.42 | dlynes_laptop | [TK]D-Fender: I remember you said something about not having to adjust gains on the sangoma cards, right? |
| 17:29.58 | rene- | could i force asterisk manager to spit events faster by issuing some pings to it? |
| 17:30.25 | Assid | spit events faster? |
| 17:30.36 | Assid | events are outputted to the cli as it is processed.. |
| 17:30.42 | [TK]D-Fender | dlynes_laptop : Not for EC reasons anyways |
| 17:30.44 | Assid | provided you are in bverbose ofcourse |
| 17:30.52 | rene- | i am getting the events |
| 17:30.57 | rene- | but they take too long |
| 17:31.05 | Assid | explain take too long? |
| 17:31.14 | dlynes_laptop | [TK]D-Fender: ah...so gains still need to be adjusted on it for caller id reasons then? |
| 17:31.16 | rene- | as in the call has been established or terminated by 10 - 20 secs and then i get the event |
| 17:31.29 | Assid | how are you connected to the astrerisk box? |
| 17:31.31 | dlynes_laptop | [TK]D-Fender: well, and call volume? |
| 17:31.35 | rene- | local box |
| 17:31.48 | Assid | shouldnt be a lag |
| 17:31.52 | Assid | its always almost immediate |
| 17:31.59 | Assid | whats your verbose level? |
| 17:32.03 | [TK]D-Fender | dlynes_laptop : Shouldn't for CID, only for std audio rates |
| 17:32.28 | dlynes_laptop | [TK]D-Fender: ah...just asking because i'm getting invalid caller id data every once in a while |
| 17:32.55 | dlynes_laptop | [TK]D-Fender: i remember running into that issue with the x100p's as well |
| 17:33.04 | dlynes_laptop | [TK]D-Fender: and a slightly higher gain fixed that |
| 17:33.08 | piper69 | which ver of Asterisk i should install if i'm a beginner |
| 17:33.15 | Strom_C | 1.2.9.1 |
| 17:33.24 | dlynes_laptop | piper69: the latest release (1.2.9.1) |
| 17:33.33 | Assid | always use that.. unless you wanna play with svn |
| 17:33.50 | dlynes_laptop | Assid: beginners don't wanna play with branch and trunk :) |
| 17:33.56 | piper69 | dlynes_laptop: i'm running Debian |
| 17:34.02 | dlynes_laptop | Assid: or test-this-branch for that matter |
| 17:34.14 | [TK]D-Fender | piper69 : And forget about packages for *. DL the source and compile. |
| 17:34.37 | Dr-Linux | hi all |
| 17:34.38 | rene- | i dont know ahts my verbose level |
| 17:34.52 | Dr-Linux | dlynes_laptop: hey |
| 17:35.05 | Assid | rene- : set verbose 20 and try |
| 17:35.05 | rene- | i am checking |
| 17:35.09 | dlynes_laptop | Dr-Linux: hey |
| 17:35.12 | rene- | ok |
| 17:35.15 | rene- | in cli?? |
| 17:35.18 | rene- | weird |
| 17:35.21 | rene- | ok |
| 17:35.21 | Assid | yeah |
| 17:35.34 | Dr-Linux | dlynes_laptop: anyone give me hand to help me setup iax trunk between two servers? |
| 17:35.39 | Dr-Linux | aww |
| 17:36.44 | dlynes_laptop | Dr-Linux: trunk=yes |
| 17:37.34 | dlynes_laptop | Dr-Linux: put that inside the definition for every friend |
| 17:37.43 | Dr-Linux | dlynes_laptop: great, so i have ServerA and ServerB |
| 17:37.48 | dlynes_laptop | Dr-Linux: well, every friend that you want to use trunking on |
| 17:38.08 | *** join/#asterisk Eggplant (i=No@dsl-72-19-47-026.cascadeaccess.com) |
| 17:38.18 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 17:38.29 | Dr-Linux | dlynes_laptop: friends will go there >> http://www.voip-info.org/wiki/view/Asterisk+-+dual+servers&view_comment_id=11327 :) |
| 17:38.43 | *** join/#asterisk mroth_imm (n=chatzill@63.65.26.220) |
| 17:39.06 | Assid | im thinking of using sip instead of iax to gain advantage of native connections from phone directly |
| 17:39.06 | mroth_imm | could anyone tell me if Asterisk is transmitting SIP via TCP or UDP? |
| 17:39.19 | Strom_C | UDP |
| 17:39.39 | mroth_imm | Strom_C: thank you for the answer |
| 17:39.57 | Strom_C | very welcome. |
| 17:39.59 | dlynes_laptop | Dr-Linux: check example #4 |
| 17:40.05 | Strom_C | Next time I charge my hourly rate ;) |
| 17:40.05 | mroth_imm | we are experiencing a problem where calls won't tear down if our WAN gets overloaded...that could explain a lot |
| 17:40.20 | mroth_imm | (RTP times out, but the calls persist) |
| 17:41.35 | dlynes_laptop | mroth_imm: could be a firewall timeout issue |
| 17:41.43 | mroth_imm | bill my employer :) |
| 17:42.04 | dlynes_laptop | mroth_imm: is this happening on long conversations? |
| 17:42.15 | mroth_imm | dlynes_laptop: we are not going through a firewall...it's an MPLS network between the offices |
| 17:42.23 | dlynes_laptop | MPLS? |
| 17:42.25 | *** join/#asterisk Johnnie (n=john@pdpc/supporter/active/Johnnie) |
| 17:42.41 | mroth_imm | no...it's happening if some of the phones don't get routed properly and end up going over the frame relay instead |
| 17:42.45 | Dr-Linux | dlynes_laptop: there are two "slave" and two "master" |
| 17:43.17 | mroth_imm | from wikipedia: Multiprotocol Label Switching (MPLS) is a data-carrying mechanism which emulates some properties of a circuit-switched network over a packet-switched network. |
| 17:43.34 | dlynes_laptop | mroth_imm: that still didn't tell me what mpls is |
| 17:43.43 | dlynes_laptop | mroth_imm: explain it to me in plain english |
| 17:43.58 | dlynes_laptop | mroth_imm: it sounds to me like another name for voip |
| 17:44.06 | Dr-Linux | dlynes_laptop: how about this? >> http://www.voip-info.org/wiki/view/Asterisk+IAX+clients |
| 17:44.18 | Assid | oay all of a sudden i feel like working |
| 17:44.30 | dlynes_laptop | mroth_imm: but it's voip over packet radio or something |
| 17:44.34 | Katty | query! |
| 17:44.35 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 17:44.37 | mroth_imm | long story short, it allows us to maintain high quality VoIP calls with our remote offices...our frame relay does not |
| 17:44.49 | dlynes_laptop | mroth_imm: i see |
| 17:44.56 | rene- | Assid: it does seems faster with verbose 20 |
| 17:45.01 | Katty | if i have a _xxxx catch all, is there a way to make an exception? |
| 17:45.12 | rene- | Assid: do you know why? |
| 17:45.16 | dlynes_laptop | mroth_imm: so basically it's a leased line with high latency? |
| 17:45.22 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-155-126-111.red.bezeqint.net) |
| 17:45.31 | dlynes_laptop | mroth_imm: no special equipment needed for it? |
| 17:45.32 | Assid | higher verbose levels |
| 17:45.40 | Katty | file, deary, are you around? |
| 17:45.41 | Assid | you get more outputs |
| 17:46.06 | dlynes_laptop | Dr-Linux: what part of try example #4 do you not understand? |
| 17:46.08 | mroth_imm | it aims to emulate a leased line, but it's not...and the routers have to be MPLS aware |
| 17:46.09 | THX2000 | Katty: I believe asterisk is supposed to choose the closest match...so _1XX would overwrite _XXX if 100 was pressed |
| 17:46.21 | dlynes_laptop | mroth_imm: so there are routers then |
| 17:46.45 | Dr-Linux | dlynes_laptop: actually i seen 3 different methods, so i want to adapt the good one. |
| 17:46.47 | dlynes_laptop | mroth_imm: see if you can adjust timeouts on the routers |
| 17:46.51 | mroth_imm | yes |
| 17:46.56 | *** join/#asterisk japerry (n=japerry@216.231.51.208) |
| 17:47.00 | dlynes_laptop | Dr-Linux: example #4 example #4 example #4 example #4 |
| 17:47.04 | Katty | THX2000: i want to make an exception for a 4 digit number. |
| 17:47.11 | Katty | THX2000: rather than _xxxx catching it |
| 17:47.16 | mroth_imm | well, the issue is not when the voice traffic is carried over the MPLS...it's when it drops back to the frame |
| 17:47.17 | [TK]D-Fender | Katty : What would you do with that 4 digit #? |
| 17:47.29 | Katty | [TK]D-Fender: my 4 digit numbers are routed to SIP/ |
| 17:47.31 | japerry | morning.. so anyone here have experience with recieving/sending FAXes with fax machines and asterisk? this system has one problem after another! =/ |
| 17:47.35 | dlynes_laptop | Dr-Linux: the others are not 'trunking'; they're merely trunks aka phone lines |
| 17:47.37 | Katty | [TK]D-Fender: i need one of them to be routed to iax |
| 17:47.44 | mroth_imm | the frame gets overloaded and drops packets...rtp timeouts occur...asterisk tries to disconnect the calls but the SIP packets get dropped...things snowball |
| 17:47.57 | [TK]D-Fender | Katty : I'd suggest a GotoIf in your dialplan logic. |
| 17:48.07 | THX2000 | Katty: Yea, so just put in: exten => 1234,1,RoutthistoIAX |
| 17:48.20 | [Airwolf] | dlynes_laptop, fyi with mpls something gets added to an ethernet frame to route very fast with low overhead. |
| 17:48.20 | dlynes_laptop | mroth_imm: ok, you never mentioned anything about dropping packets before |
| 17:48.28 | Katty | THX2000: so if it's explicitly defined, it won't go to xxxx? |
| 17:48.51 | mroth_imm | it seems that Asterisk should either transmit SIP via TCP, or forcibly terminate calls if it counts X number of disconnect failures |
| 17:48.51 | dlynes_laptop | [Airwolf]: ah...so it's like that fast ethernet bridging feature in linux 2.4 and higher then? |
| 17:48.51 | THX2000 | Katty: Pretty sure :P |
| 17:48.54 | Katty | fancy. |
| 17:48.57 | Katty | i'll try. thanks. |
| 17:49.00 | [TK]D-Fender | Katty : Oh and since you didn't really want "the answer". Yes, I empathize with you and understand your frustration and am behind you 100% ;) |
| 17:49.14 | [Airwolf] | dlynes_laptop, I'm not sure about that feature in de linux kernel. But it could be. |
| 17:49.22 | Dr-Linux | dlynes_laptop: i'm confused there are 2 [slave] and 2 [master] |
| 17:49.24 | mroth_imm | i guess both would be better, since a lost connection now seems to cause an endless loop |
| 17:49.29 | Dr-Linux | dlynes_laptop: do i know one? or what? |
| 17:49.36 | syzygybsd | k, I have 2 asterisk boxes that talk to eachother, when one crashes the calls on the other end never end |
| 17:49.41 | [TK]D-Fender | Katty : I'm uncertain about the sort order personally, but I usually suggest hard-coding extens unless you actually HAVE 10000 extensions :) |
| 17:49.58 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 17:50.00 | anthm | the _ ones always win |
| 17:50.02 | dlynes_laptop | [Airwolf]: with that feature, you need two network cards that support that specific feature, but it allows for fast routing between the two network cards |
| 17:50.08 | piper69 | [TK]D-Fender:what do u mean And forget about packages for *. DL the source and compile. |
| 17:50.10 | anthm | you need a top level context |
| 17:50.12 | Katty | [TK]D-Fender: we don't have a lot of extensions |
| 17:50.12 | dlynes_laptop | [Airwolf]: and it requires a fast cpu |
| 17:50.18 | [TK]D-Fender | piper69 I mean exactly what it says. |
| 17:50.26 | *** join/#asterisk twisla (i=twisla@lutin.jard.in) |
| 17:50.28 | anthm | that includes => one_with_underscores |
| 17:50.29 | [Airwolf] | dlynes_laptop, it seems then it's the same. |
| 17:50.32 | *** join/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net) |
| 17:50.36 | Katty | [TK]D-Fender: but right now it's setup as a sort of xxxx,1,Dial(SIP/${exten}) type thingy |
| 17:50.39 | anthm | then the explicit ones win |
| 17:50.56 | dlynes_laptop | mroth_imm: there's stcp support in trunk I think it was |
| 17:50.59 | TripleFFFF | hey guys.. seems asteris as problems to push both voices when on a 3 way conference.. if 2 talk in same time |
| 17:51.04 | TripleFFFF | normal /? |
| 17:51.05 | piper69 | [TK]D-Fender: and if i know what u mean i would 'v enot asked u again, i am just confuseed would u please explain to me |
| 17:51.07 | Katty | Voicemail(${EXTEN}) <- that would become Voicemail(therealextension) right? |
| 17:51.08 | [TK]D-Fender | Katty : You really want to change that then. Explicit is a much safer and more flexible means. |
| 17:51.10 | [Airwolf] | But anyway mpls doesn't do anything on layer3 so it should not do anything with IP traffic. |
| 17:51.16 | mroth_imm | dlynes_laptom: thanks for the pointer |
| 17:51.26 | Katty | [TK]D-Fender: no, i don't |
| 17:51.28 | *** join/#asterisk ast_freak|Laptop (n=jesse@c-66-41-163-210.hsd1.mn.comcast.net) |
| 17:51.29 | anthm | put the xxxx on in it's own context called [dynamic] and put the explcit on in one called [static] then in the top level one |
| 17:51.35 | anthm | include => static |
| 17:51.36 | Katty | [TK]D-Fender: that would turn into 30 lines, which now take 3 |
| 17:51.42 | anthm | include => dynamic |
| 17:51.51 | [TK]D-Fender | piper69 : Don't use APT-GET to try to install a Debian pre-packaged form of *. Download the source tar.gz's direct from digium's FTP |
| 17:51.52 | dlynes_laptop | mroth_imm: i was thinking 1.4 would get released this month, but now i'm figuring it might not be until next month |
| 17:51.54 | anthm | that is how you get the static to beat the dynamic |
| 17:52.00 | mroth_imm | [Airwolf]: the MPLS functions fine...it's really a general issue of how Asterisk is handling large numbers of dropped packets |
| 17:52.00 | dlynes_laptop | mroth_imm: trunk is what will become 1.4 |
| 17:52.00 | Katty | anthm: yeah you cleaned up my extensions.conf with that outlook |
| 17:52.16 | Katty | anthm: you called it [speed-dial] tho ;) |
| 17:52.27 | [TK]D-Fender | Katty : True, but offer more control per box, and guarantee results and stop bad things from happening when people dial non-valid (in reality) extensions. |
| 17:52.28 | mroth_imm | it appears that if the signalling packets get dropped asterisk just spins trying to disconnect the call forever |
| 17:52.34 | Katty | (${EXTEN}) would be (2000) right? |
| 17:52.37 | TripleFFFF | ? |
| 17:52.50 | Katty | ${EXTEN} is the full variable? |
| 17:53.05 | dlynes_laptop | mroth_imm: you could try asking anthm |
| 17:53.15 | dlynes_laptop | mroth_imm: i think he wrote a lot of the chan_sip.c code |
| 17:53.24 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 17:54.00 | Katty | anthm: how do i take (${EXTEN}) and specifiy a real number? Does it become (2000)? |
| 17:54.01 | piper69 | [TK]D-Fender: oh thank you , i was about to do APT-GET install |
| 17:54.13 | anthm | nah chan_sip would be a blemish on my resume that is one of the places i try and avoid but i did add a feature or 2 |
| 17:54.31 | anthm | katty what's the goal? |
| 17:54.33 | THX2000 | Katty: Yea |
| 17:54.37 | dlynes_laptop | anthm: ah...thought I heard someone the other day mentioning you had contributed a lot of code to it |
| 17:54.46 | Katty | anthm: exten => 2011,2,Voicmeail(2011) |
| 17:54.50 | mroth_imm | for instance, i've seen the same call fail to be disconnected over 5000 times over the span of 18 minutes |
| 17:54.52 | Katty | anthm: minus the typos |
| 17:54.53 | piper69 | [TK]D-Fender: what is the link to digium's FTP please |
| 17:55.07 | rene- | my asterisk is happier, can i issue the set verbose from within manager?? |
| 17:55.12 | dlynes_laptop | piper69: ftp://ftp.digium.com/pub/telephony/ |
| 17:55.14 | Hmmhesays | http://ftp.digium.com |
| 17:55.16 | Hmmhesays | for the web |
| 17:55.19 | Katty | Hmmhesays: hey hun |
| 17:55.21 | anthm | they meant asterisk in general not chan_sip lol |
| 17:55.22 | Hmmhesays | Hello Katty |
| 17:55.45 | mroth_imm | anthm: do you know why the decision was made to transmit SIP via UDP? |
| 17:55.52 | Hmmhesays | anyone using a wintv-pvr-250? i'm trying to decide what to put in this mythtv box |
| 17:55.56 | Katty | Hmmhesays: is this right? exten => 2011,2,Voicmail(2011) |
| 17:55.59 | anthm | here is the exact list =p http://www.cluecon.com/anthm.html |
| 17:55.59 | dlynes_laptop | anthm: ummm....that gives you asterisk, asterisk-sounds, asterisk-addons, libpri, zaptel, ... |
| 17:56.07 | Katty | Hmmhesays: plus e. |
| 17:56.08 | Hmmhesays | Katty: no |
| 17:56.13 | Katty | Hmmhesays: what's the syntax? |
| 17:56.35 | Hmmhesays | cmd voicemail(u2011) or voicemail(b2011) |
| 17:56.46 | Katty | u or b? |
| 17:56.47 | Hmmhesays | for unavail or busy if you want different messages |
| 17:56.56 | Hmmhesays | otherwise just use u |
| 17:56.57 | Katty | oh. |
| 17:57.01 | Katty | the others don't have one |
| 17:57.10 | anthm | katty i think i am lost what is it you want to do again? =D |
| 17:57.11 | Katty | it's just..._xxxx,2,Voicemail(${EXTEN}) |
| 17:57.19 | THX2000 | Katty: I think you want 2011,2,Voicemail(u2011@default) |
| 17:57.22 | Katty | anthm: i want to take _xxxx,2,Voicemail(${EXTEN}) and make it a real number. |
| 17:57.24 | Hmmhesays | oh yeah? I guess i've never tried that |
| 17:57.37 | Hmmhesays | is there a problem? |
| 17:57.46 | mroth_imm | anthm: or more simply, do you think that the decision to transmit SIP via UDP was a sound one? |
| 17:57.56 | Katty | Hmmhesays: by default with that xxxx line be u or b? |
| 17:58.00 | anthm | if you want you can keep it that way and just dont put a 2, on the static one they can share |
| 17:58.10 | Hmmhesays | u I believe |
| 17:58.14 | Katty | kk |
| 17:58.21 | anthm | no sip on udp is daft |
| 17:58.37 | Katty | here's go nothing |
| 17:58.56 | anthm | sip is already daft but sip on udp is double daft cos it forces sip to emulate reliable transport inside the protocol |
| 17:58.58 | Katty | didn't work )= |
| 17:59.04 | Katty | the xxxx is catching all 4 digits |
| 17:59.21 | syzygybsd | lol anthm |
| 17:59.36 | anthm | you need to move it into it's own context |
| 17:59.39 | THX2000 | Katty: what is the _xxxx for? and how many extensions does it cover? |
| 17:59.42 | anthm | so it's not as powerful |
| 17:59.46 | TripleFFFF | 9999 |
| 17:59.51 | Katty | THX2000: i had sip calls included before speed dial calls |
| 17:59.53 | anthm | make a [dyna] |
| 17:59.56 | [TK]D-Fender | Katty : Time to start cleaning up your dialplan.... |
| 17:59.58 | Katty | THX2000: i swapped the order of my includes around and it worked |
| 17:59.58 | anthm | and put your _XXXX in there |
| 18:00.00 | Hmmhesays | daft, i haven't heard anyone say that in awhile |
| 18:00.03 | dlynes_laptop | anthm: btw...www.cluecon.com is down |
| 18:00.07 | Katty | [TK]D-Fender: no it's fine. |
| 18:00.09 | anthm | then include => dyna where it used to live |
| 18:00.19 | Katty | [TK]D-Fender: i just included my exceptions before the catch alls |
| 18:00.22 | THX2000 | lol, too many people are talking to Katty at once :P |
| 18:00.24 | Hmmhesays | so my ac494 has been stuck in customs for a week now |
| 18:00.25 | syzygybsd | anthm: do you think sip should be over tcp? |
| 18:00.28 | Katty | THX2000: yeah i know, and it's insanely confusing |
| 18:00.32 | ghenry | anyone got any tips for call back on a webpage? Just create a .call file and populate details from a form and SCP it over, or similar? |
| 18:00.40 | anthm | down? |
| 18:00.40 | Katty | THX2000: which is why i usually only talk to one person at a time and ignore everyone else |
| 18:00.43 | anthm | nooooo |
| 18:00.45 | *** join/#asterisk kristalino (n=kristali@84-50-84-146-dsl.trt.estpak.ee) |
| 18:00.48 | anthm | it looks up on my end |
| 18:00.54 | anthm | where are you from |
| 18:00.56 | Hmmhesays | and i've been all itching to try your nsf mount and compile my stuff on the board itself |
| 18:01.10 | dlynes_laptop | anthm: Vancouver, BC |
| 18:01.21 | anthm | it would surely work more reliably on tcp |
| 18:01.29 | dlynes_laptop | anthm: coming in through shaw cable for an isp |
| 18:01.32 | syzygybsd | I don't see that at all |
| 18:01.44 | anthm | if you did every logical change to sip you could come up with the result would resemble h323 |
| 18:02.01 | anthm | paging bkw... |
| 18:02.07 | syzygybsd | I see you introducing a huge number of delays, waiting for packets to be retransmited etc |
| 18:02.11 | JonR800 | ghenry: that would work but i would use the manager interface. |
| 18:02.23 | ghenry | ah, k |
| 18:02.25 | mroth_imm | anthm: how well would you say that SIP is emulating reliable transport in Asterisk? |
| 18:02.33 | bkw_ | dlynes_laptop, what does it say? |
| 18:02.35 | ghenry | JonR800: just looking on voip-info |
| 18:02.50 | syzygybsd | if every lost packet took an additional 100ms to retransmit (not too high of latency) think about how quickly a call could get out of sync |
| 18:03.04 | anthm | i think it has the scheduler done ok |
| 18:03.13 | *** join/#asterisk FlyboySR22 (n=rsears@gateway.americanis.net) |
| 18:03.31 | mroth_imm | in other words...could multiple (thousands) of attempts to disconnect the same call over an overloaded link be the result of SIP over UDP? |
| 18:03.40 | dlynes_laptop | anthm: yeah...http request is getting sent out, but no http response being sent back |
| 18:03.46 | JonR800 | ghenry: check out http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Originate |
| 18:03.52 | bkw_ | dlynes_laptop, can you telnet to port 80 on cluecon.com? |
| 18:03.52 | Nugget | telnet is eeeeeeevil! |
| 18:03.58 | syzygybsd | that page came up great for me dlynes_laptop |
| 18:03.59 | ghenry | Thanks JonR800 |
| 18:04.07 | dlynes_laptop | bkw_: just did |
| 18:04.11 | dlynes_laptop | bkw_: i get a connection |
| 18:04.13 | MikeJ[Laptop] | telnet |
| 18:04.15 | dlynes_laptop | bkw_: but nothing comes back |
| 18:04.21 | anthm | it's kinda one of those age old dilemma |
| 18:04.28 | MikeJ[Laptop] | hmmm.. I guess it's not a bot... |
| 18:04.36 | dlynes_laptop | telnet www.cluecon.com 80\nGET / HTTP/1.0 |
| 18:04.43 | dlynes_laptop | nothing |
| 18:04.49 | syzygybsd | ya. but if you arn't losing any packets, UDP wouldn't matter anyway would it? |
| 18:05.01 | bkw_ | dlynes_laptop, what is your IP? msg it to me |
| 18:05.03 | japerry | does anyone know of a TCP/IP Digital Fax machine (rather than one that uses a modem to connect?) |
| 18:05.18 | anthm | udp is attractive because 1 socket connection can send data to infinate destinations |
| 18:05.39 | anthm | but has no guarentee |
| 18:05.43 | syzygybsd | so can tcp |
| 18:05.51 | dlynes_laptop | syzygybsd: no |
| 18:06.07 | dlynes_laptop | syzygybsd: he said one "connection" |
| 18:06.14 | THX2000 | Will TCP be implemented for SIP, id like to see some integration w/ Exchange 2006 |
| 18:06.22 | dlynes_laptop | syzygybsd: unless you're doing broadcast packets, or multicast, tcp is not capable of that |
| 18:06.27 | anthm | tcp you need 1 socket per connection but you get guarenteed delivery and you can tell by the socket going down that they are not there anymore |
| 18:06.31 | syzygybsd | also said infinate destinations... which wouldn't suggest a connection at all |
| 18:06.39 | Hmmhesays | i need a cheap cheap pc case, preferable from a store with 0 shipping |
| 18:06.43 | Hmmhesays | anyone? anyone? |
| 18:06.53 | anthm | sctp is the balance |
| 18:06.57 | anthm | which would be ideal |
| 18:06.58 | mroth_imm | anthm: yes...i agree...i think asterisk right now just dumbly tries to disconnect the same call as the RTP continues to time out |
| 18:07.22 | anthm | so the best approach is to design sip as a library first |
| 18:07.30 | anthm | with pluggable tranport layer |
| 18:07.40 | anthm | cos you never know how you may be happy doing the transport |
| 18:07.40 | syzygybsd | does sip have a heartbeat or anything to make sure the responding server still has the call up ? |
| 18:07.52 | mroth_imm | resulting in 1000s of disconnection attempts, and Asterisk ultimately failing under the load |
| 18:08.22 | bkw_ | syzygybsd, it has timers |
| 18:08.29 | bkw_ | but asterisk doesn't support them |
| 18:08.40 | anthm | it's set to some certian amount of retries i think but that packet scheduler stuff may end up costing you the benifits that less sockets has to offer |
| 18:08.49 | mroth_imm | calls not tearing down properly also affects licensed users (ABE) because the call count increments but never decrements |
| 18:08.50 | benjk | anthm ... Asterisk and layered architecture? that doesn't sound like its going to fly with the maintainers |
| 18:09.21 | mroth_imm | anthm: are you sure? i've personally seen close to 6000 attempts to disconnect the same call over an 18 minute period |
| 18:09.48 | mroth_imm | maybe it is different under ABE (i don't like ABE personally, but i don't get to make that decision) |
| 18:10.10 | anthm | not positive but there should be some kind of retry limit in the packet scheduler |
| 18:10.30 | piper69 | dlynes_laptop: will it be the same as the one on the Asterisk website |
| 18:10.48 | MikeJ[Laptop] | yeah.. last I heard.. oej was working on them.. or it was waiting on people to pay him for chan_sip2 or somthing? |
| 18:11.20 | *** part/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net) |
| 18:11.25 | mroth_imm | the next step is to try replicate the problem and submit it as a bug. thanks for all of the information!! |
| 18:12.06 | Katty | will fop interface with iax? |
| 18:12.06 | NDT | Are you not able to have more than one extension inside a macro? |
| 18:12.17 | benjk | use chan_ltp and you avoid all of that nonsense in the first place |
| 18:12.18 | Katty | Hmmhesays: do you know if fop can handle iax? |
| 18:12.36 | Hmmhesays | I believe you can have iax buttons on there |
| 18:12.38 | [TK]D-Fender | NDT : You are not supposed to have ANYTHING except "s" in a macro. |
| 18:12.43 | Katty | Hmmhesays: fancy. thanks. |
| 18:12.51 | NDT | ahhh |
| 18:13.13 | [TK]D-Fender | NDT : It is not for IVR's and it is not a "real" context. |
| 18:14.01 | anthm | you should be careful making res_stun that will end up being a cross module dependancy |
| 18:14.15 | dlynes_laptop | piper69: huh? |
| 18:14.22 | dlynes_laptop | piper69: to what are you referring? |
| 18:14.34 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 18:14.35 | *** mode/#asterisk [+o russellb] by ChanServ |
| 18:14.41 | anthm | since there is no plugin archetechure for stun client you shoud just link it to the core as an api |
| 18:14.42 | NDT | I tried to jump to various priorities in the macro and was trying to use continue to come back out of it and return to default context right after where I called the macro but that doesn't seem to work. I thought continue was for that? |
| 18:15.09 | NDT | For the macro result |
| 18:15.12 | [TK]D-Fender | NDT : when "s" runs out your diaplan continues where it left off. |
| 18:15.20 | Dr-Linux | dlynes_laptop: one thing i didn't understand, that's the >> register => slave:secret@iax-gw1.company.net |
| 18:15.35 | Dr-Linux | on master server is the same in extensions.conf |
| 18:15.47 | Dr-Linux | on slave server it's the same |
| 18:15.56 | NDT | [TK]D-Fender: but you can't force it back? |
| 18:16.02 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 18:16.08 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 18:16.09 | *** join/#asterisk jgoo (n=foo@ppp65-254.adsl.forthnet.gr) |
| 18:16.17 | Dr-Linux | what should be the >> hostname? instead of iax-gw1.company.net |
| 18:16.29 | [TK]D-Fender | NDT : You're not supposed to. If you want to abort, make the last line of "s" a NoOp, and just goto it. |
| 18:16.52 | jgoo | Hey guys, anyone up on the trixbox iso install? nothing about trixbox itself but the centos booting from an external usb drive |
| 18:17.11 | [TK]D-Fender | jgoo : Please read the /topic |
| 18:17.32 | NDT | [TK]D-Fender: hmmm that may work for what I need...thanks |
| 18:17.47 | Hmmhesays | bah i can't remember the cheap case website |
| 18:18.04 | MikeJ[Laptop] | cheap case? |
| 18:18.16 | dlynes_laptop | Dr-Linux: i don't understand your question |
| 18:18.29 | dlynes_laptop | Dr-Linux: could you be a bit more specific? |
| 18:18.38 | Hmmhesays | yeah I need a cheap atx case |
| 18:18.52 | Hmmhesays | there was a site I always used to go to for them, I can't remember it for the life of me now |
| 18:18.58 | Dr-Linux | dlynes_laptop: look at the example #4 |
| 18:19.17 | Dr-Linux | in all examples it's >> iax-gw1.company.net |
| 18:19.49 | Dr-Linux | dlynes_laptop: so i'm confused with this, where should i put what IP server's IP address |
| 18:19.51 | *** join/#asterisk feld_ (n=feld@12.148.212.157) |
| 18:20.12 | Dr-Linux | dlynes_laptop: i don't know understand this example logic |
| 18:20.43 | feld_ | does anyone know if a module or something for SugarCRM that can integrate with Asterisk... but I don't want them on the same box..... so it could read the mysql logs, I guess? |
| 18:20.55 | feld_ | not logs, but database |
| 18:21.00 | NDT | [TK]D-Fender: You can do a goto inside the macro to like s,1 right? |
| 18:21.02 | feld_ | sugarcrm logs in that mysql database. |
| 18:21.08 | piper69 | dlynes_laptop: i was reffeing to the new relase of Asterisk |
| 18:21.17 | Dr-Linux | dlynes_laptop: i have done all the setting on "master" server and put the switch => IAX2/master:secret@other-server-ip.com/outbound |
| 18:21.24 | [TK]D-Fender | NDT : yup |
| 18:21.27 | NDT | k |
| 18:22.27 | Dr-Linux | dlynes_laptop: then i came to slave server, here is the same register => slave:secret@iax-gw1.company.net , so i'm not sure what server's IP address should i put here |
| 18:22.39 | *** join/#asterisk salviadud (n=ralfalfa@201.137.164.143) |
| 18:22.44 | NDT | I have an ${ACCEPT} variable in the macro can I pass it's contents to default? |
| 18:22.51 | benjk | never use passwords in your dialplan |
| 18:23.03 | benjk | set up a peer entry in iax.conf instead |
| 18:23.04 | Dr-Linux | dlynes_laptop: also third place in iax.conf at slave server >> host=iax-gw1.company.net , what server ip should i put here |
| 18:23.28 | anthm | benjk is there a lib of that ltp i'll add it to freeswitch if there is |
| 18:23.46 | benjk | otherwise the password will show up a bazillion times in all your logfiles |
| 18:23.58 | benjk | anthm, yes |
| 18:24.07 | anthm | where is it |
| 18:24.09 | [TK]D-Fender | NDT : Variables are still local and don't need to be passed |
| 18:24.12 | salviadud | newb question here: if i want to run asterisk as a normal user, what is the chmod or chown i need to apply to asterisk, or am i that stupid and i just need to add it in PATH? |
| 18:24.14 | benjk | its on their site |
| 18:24.22 | benjk | spokn.net or something like that |
| 18:24.38 | feld_ | I couldn't find anything interesting about integrating SugarCRM and Asterisk so I started a new module from scratch. (the one I had was only doing log without any options). |
| 18:24.47 | *** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 18:24.47 | benjk | I got it working on the Mac (INtel only right now) |
| 18:24.53 | benjk | LTP ROCKS |
| 18:24.55 | feld_ | crap didn't mean to paste! |
| 18:24.58 | Nugget | I had to buy a macbook pro this week. |
| 18:25.09 | feld_ | salviadud: you mean you want to run as a normal user but it will execute as root? |
| 18:25.18 | *** join/#asterisk af_ (n=af@ip-170-209.sn1.eutelia.it) |
| 18:25.36 | salviadud | feld_, i want to run asterisk as a normal user, without being root |
| 18:25.59 | salviadud | is it 667? |
| 18:26.05 | salviadud | 666? |
| 18:26.18 | *** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
| 18:26.22 | salviadud | i forgot... and i don't remember why |
| 18:26.24 | wunderkin | 42 |
| 18:26.31 | salviadud | 42??? |
| 18:26.36 | feld_ | do the match salviadud. 1=execute, 2=write, 4=read |
| 18:26.36 | Nugget | 755 |
| 18:26.36 | salviadud | you gotta be kiddin me |
| 18:26.48 | wunderkin | i always just use the letters |
| 18:26.58 | salviadud | 755, yeaaah, thats the one |
| 18:27.02 | Nugget | or learn to use those silly "o+rx" style arguments to chmod. |
| 18:27.05 | *** join/#asterisk Qwell[laptop] (n=Qwell[]@unaffiliated/qwell) |
| 18:27.07 | Nugget | if numbers scare you that much |
| 18:27.18 | salviadud | Nugget, thanx |
| 18:27.23 | feld_ | I prefer the numbers.... easier to understand if you ask me :) |
| 18:27.31 | DrkShdw | much easier |
| 18:27.32 | Nugget | clearly this is not the case for salviadud |
| 18:27.48 | feld_ | well hey now..... it took me a long time to fully understand them :) |
| 18:28.10 | salviadud | i'll be going now, thanx a lot you guys, bye bye |
| 18:28.17 | feld_ | later |
| 18:28.22 | eKo1 | numbers are ok but if you don't know what permissions a file has and are too lazy to find out, the letters come in handy |
| 18:28.33 | DrkShdw | I thought I had a good grasp on them, then I saw a chmod 2755 and was like wtf?! (this was back in the day) |
| 18:28.48 | feld_ | hehe :) yeah those extra bits |
| 18:28.58 | eKo1 | hehe, yeah i was like that too at one point |
| 18:29.09 | feld_ | for clarity I usually put a 0 in front to make sure I'm getting what I want |
| 18:36.08 | Dr-Linux | nothing happend |
| 18:36.28 | Dr-Linux | anybody can help me to setup iax trunk between 2 servers? |
| 18:37.02 | Dr-Linux | i'm following this document, but don't understand a few things http://www.voip-info.org/wiki/view/Asterisk+-+dual+servers&view_comment_id=11327 :) |
| 18:37.24 | eKo1 | Do you know how to setup a sip trunk between two servers? |
| 18:37.50 | eKo1 | It is essentially the same. |
| 18:38.16 | Dr-Linux | eKo1: yes, i'm using already sip trunks |
| 18:38.18 | *** join/#asterisk ariel_ (n=ariel_@74.8.35.2) |
| 18:38.30 | Dr-Linux | eKo1: but someone said, iax trunk is very good, so i wanna do that |
| 18:38.36 | Dr-Linux | but i don't understand few things |
| 18:38.41 | Dr-Linux | i'm following this document, but don't understand a few things http://www.voip-info.org/wiki/view/Asterisk+-+dual+servers&view_comment_id=11327 :) |
| 18:38.43 | *** part/#asterisk ariel_ (n=ariel_@74.8.35.2) |
| 18:38.48 | *** join/#asterisk dlynes_laptop (n=dlynes@S0106001217014b92.vc.shawcable.net) |
| 18:38.53 | Dr-Linux | eKo1: look here for example #4 |
| 18:39.11 | Dr-Linux | dlynes_laptop: was helping me , but i think he got busy :) |
| 18:39.44 | [Airwolf] | Can anyone tell me what the native mp3 frequency bitrate thingie has to be ? |
| 18:39.59 | piper69 | [TK]D-Fender: just a question, why i can't use the apt-get install? why do u want me to compile it and install? is there is a diffrent |
| 18:40.00 | [Airwolf] | Because I found on the wiki that I have to resample it to 8kbit mono |
| 18:40.08 | [Airwolf] | But that sucks ofcourse |
| 18:40.17 | Dr-Linux | In extensions.conf: (on master) |
| 18:40.18 | Dr-Linux | [outbound] |
| 18:40.18 | Dr-Linux | switch => IAX2/master:secret@iax-gw1.company.net/outbound |
| 18:40.29 | Dr-Linux | ~switch |
| 18:40.30 | jbot | extra, extra, read all about it, switch is This refers to a hub that directs network packets to the port they are intended for, without broadcasting them to all connections. Switching is an alternative to moving to faster architectures. Switched 10Base-T can move data faster in some cases than a 100Base-T hub, because the 100Base-T hub takes up the hub's entire ... |
| 18:40.37 | Hmmhesays | Dr-Linux: boobies? |
| 18:40.50 | Dr-Linux | Hmmhesays: no, but iax trunk |
| 18:40.55 | Hmmhesays | Ahh |
| 18:41.02 | Dr-Linux | hhm.. |
| 18:41.15 | Dr-Linux | i think no one is much fimiliar with IAX trunk .. |
| 18:41.39 | eKo1 | I was familiar. But it was such a long time ago... |
| 18:41.42 | Dr-Linux | will give it a little try, then will fuck iax and i'll setup SIP trunk |
| 18:43.24 | piper69 | dlynes_laptop: after i download asterisk 129 what should i do next |
| 18:44.34 | Dr-Linux | eKo1: i'm going to setup sip trunk, let'a IAX is wasting of time nothing else |
| 18:47.11 | *** join/#asterisk saftsack (n=oliver@pD9E05619.dip.t-dialin.net) |
| 18:51.33 | nortex | Dr-Linux, Where are you getting stuck on the dual server setup with IAX? |
| 18:52.13 | Dr-Linux | <PROTECTED> |
| 18:52.15 | Dr-Linux | exten => _92XX.,1,Set(CALLERID(NUM)=16505935400) |
| 18:52.15 | Dr-Linux | exten => _92XX.,2,Dial(SIP/serverA/${EXTEN:2},30,r) |
| 18:52.15 | Dr-Linux | exten => _92XX.,3,Playback(all-outgoing-lines-unavailable) |
| 18:52.26 | Dr-Linux | nortex: opssssssssss sorry |
| 18:52.38 | Dr-Linux | i was pasting a link but .. mistake |
| 18:52.51 | Dr-Linux | http://www.voip-info.org/wiki/view/Asterisk+-+dual+servers&view_comment_id=11327 |
| 18:53.06 | *** join/#asterisk topping (n=topping@207.47.6.157.static.nextweb.net) |
| 18:53.17 | Damin | Anyone ever have a problem with aPolycom 501 that won't let you wipe it's config? Everytime I go to the "Reset User Settings" menu, it asks for the password and then Hangs there on "Please Wait" |
| 18:54.20 | *** join/#asterisk FuriousGeorge (n=FuriousG@ool-43536ea8.dyn.optonline.net) |
| 18:54.25 | FuriousGeorge | hey all |
| 18:54.54 | *** join/#asterisk feld_ (n=feld@12.148.212.157) |
| 18:54.56 | FuriousGeorge | anyone else using cable internet in the tri-state area? if so are you noticing lotsa packet-loss today for voip traffic |
| 18:54.58 | FuriousGeorge | ? |
| 18:55.15 | FuriousGeorge | we are having some wicked storms. i guess those in general can mess up voip |
| 18:56.49 | Katty | Damin: yeah |
| 18:56.59 | Katty | Damin: it was a bitch, but i think we ended up upgrading the firmware |
| 18:57.36 | Katty | Damin: otherwise it said something like $foo was busy blahblah |
| 18:58.51 | [TK]D-Fender | Damin : fACTORY RESET |
| 18:58.58 | nortex | Dr-Linux, Let me get a pastebin together. IAX would be best between these boxes since NAT and firewalls are hard on SIP and all the ports SIP+RTP opens up. |
| 18:59.46 | Dr-Linux | nortex: yes, i know that, as i'm already using SIP trunk, but someone said, IAX is very nice, but i don't know how can i do that |
| 19:00.09 | Dr-Linux | nortex: can i PM? |
| 19:02.10 | nortex | Dr-Linux, pm is fine |
| 19:02.32 | Dr-Linux | nortex: my SIP trunk is almost ready, but i wanna use IAX trunk |
| 19:02.34 | Dr-Linux | ok thanks |
| 19:07.35 | MACscr | maybe its just me, but AFAIK, IAX and IAX2 dont really have much commercial support |
| 19:08.45 | GyrosGeier | if an UDP packet is dropped, then it's lost, the remaining packets are not affected. If a TCP packet is dropped, all following packets are queued up until the missing packet has been retransmitted. If there is only little traffic, then this can take a looong time |
| 19:09.21 | *** join/#asterisk TrixVox (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net) |
| 19:10.35 | benjk | depends what you mean by commercial support |
| 19:11.26 | benjk | Global Crossing use IAX2 for their last mile, that's quite a big deployment |
| 19:13.29 | *** join/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net) |
| 19:13.32 | TripleFFFF | hey guys |
| 19:13.45 | TripleFFFF | how can ui push fax incoming to a number to another box ? |
| 19:14.07 | TripleFFFF | tried number,fax,1,dial(sip/${EXTEN}@otherbox) |
| 19:15.01 | [TK]D-Fender | TripleFFFF : format is quite wrong. fax is an exten |
| 19:15.51 | benjk | exten => fax,1,Dial(...) |
| 19:16.31 | benjk | also, it isn't such a good idea to send fax over SIP/RTP |
| 19:16.53 | benjk | you probably only want to do that on a LAN and with G711 |
| 19:17.08 | TripleFFFF | yeah |
| 19:17.12 | TripleFFFF | im pulling from realtime |
| 19:17.14 | TripleFFFF | BUT |
| 19:17.25 | TripleFFFF | it pushed to box2 but box then loops trying to repush it |
| 19:17.38 | benjk | just about any amount of latency will confuse the fax machine at the receiving end |
| 19:18.13 | TripleFFFF | they like got 0.0001ms |
| 19:18.17 | TripleFFFF | same lan |
| 19:18.20 | TripleFFFF | ulaw |
| 19:18.31 | benjk | no way you get .0001ms |
| 19:18.36 | TripleFFFF | so if i push to box2 . then box 2 pulls same realtime info so tries to resend to itself |
| 19:18.39 | TripleFFFF | get it ? |
| 19:18.46 | benjk | it takes longer to encode ulaw ;) |
| 19:18.58 | TripleFFFF | your right 0.4ms |
| 19:19.12 | TripleFFFF | hey.. not talking about qos of fax.. but dialplan ; |
| 19:19.13 | TripleFFFF | 0 |
| 19:19.23 | benjk | yeah, sure |
| 19:20.06 | TripleFFFF | so im getting |
| 19:20.15 | TripleFFFF | -- Executing dial("SIP/123-23ab", "sip/5141239297@fax01") |
| 19:20.18 | TripleFFFF | on box a |
| 19:20.21 | TripleFFFF | and B gets it then |
| 19:20.46 | TripleFFFF | <PROTECTED> |
| 19:20.48 | TripleFFFF | resends |
| 19:20.57 | TripleFFFF | coz it gots same thing |
| 19:20.58 | TripleFFFF | shit |
| 19:21.21 | *** join/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com) |
| 19:22.31 | TripleFFFF | ok this sucks |
| 19:22.33 | TripleFFFF | hmm |
| 19:22.41 | TripleFFFF | ah |
| 19:22.42 | TripleFFFF | then |
| 19:22.45 | benjk | fax sucks anyway |
| 19:22.57 | TripleFFFF | on box 1 if a fax... can i push normal to box2 first ? |
| 19:22.59 | wunderkin | ya, fax sucks, interweb sux, life sux, it all sux |
| 19:23.10 | TripleFFFF | hmm will do same as both boxes read same realtime |
| 19:23.15 | benjk | no, specifically fax suxx |
| 19:23.20 | TripleFFFF | so box 2 will try to send to box 2 |
| 19:23.37 | TripleFFFF | hmm unles i try different context |
| 19:23.47 | TripleFFFF | yeah |
| 19:23.51 | benjk | sending faxes should incur an obsoletion tax of 1 USD per page |
| 19:24.15 | TripleFFFF | ok so if box one sends as sip/${EXTEN}@box2 |
| 19:24.18 | TripleFFFF | can i specify context ? |
| 19:24.27 | GyrosGeier | isn't there an RTP payload type for G3 data? |
| 19:24.44 | benjk | and on faxes coming in from overseas there should also be a 1 USD/page tax on receiving faxes |
| 19:25.04 | benjk | that would solve all fax problems once and for all in less than a year |
| 19:25.09 | GyrosGeier | if so, one could hack spandsp into a codec conversion thingy |
| 19:25.23 | benjk | if not, just raise the tax to 5 USD per page |
| 19:25.50 | benjk | seriously, fax shouldn't be around anymore |
| 19:26.27 | *** join/#asterisk rajiv|work (n=rajiv@gentoo/developer/rajiv) |
| 19:26.46 | drray | we keep faxing for legacy support |
| 19:26.50 | drray | it's a waste |
| 19:27.01 | benjk | yeah, I have a fax machine still, too |
| 19:27.02 | drray | ours is 99% spam |
| 19:27.10 | benjk | I get mostly spam on it |
| 19:27.26 | benjk | and once or twice a year there is a situation where I actually use it |
| 19:27.36 | drray | there needs to be an efax machine |
| 19:27.48 | drray | that you can scan documents to a computer |
| 19:27.52 | benjk | like sending a passport copy to a consulate for a visa application or something like that |
| 19:28.08 | drray | our people refuse to consider a scanner |
| 19:28.17 | drray | even though we fucking have one built in to the copier |
| 19:28.18 | drray | :) |
| 19:28.28 | *** join/#asterisk kfudge (n=kzymxa@68-188-118-234.static.stls.mo.charter.com) |
| 19:28.35 | benjk | yeah, but then connect the damn fax machine to an isolated analog line and be done with it |
| 19:28.45 | benjk | don't run it through your PXB |
| 19:28.47 | benjk | PBX |
| 19:29.18 | kfudge | Hi all, I am having issues with the G729 codec. Can anyone assist me? |
| 19:31.38 | dlynes_laptop | kfudge: ? |
| 19:31.38 | *** part/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net) |
| 19:32.12 | *** join/#asterisk lpb (n=root@63.81.26.126) |
| 19:32.15 | Corydon-w | Have you bought enough licenses? |
| 19:32.32 | dlynes_laptop | Corydon-w: he hasn't even stated what his problem is yet :) |
| 19:32.44 | dlynes_laptop | ~suggestions |
| 19:32.46 | jbot | somebody said suggestions was 1) Don't ask to ask. Just say your problem, 2) Don't repeat until 5 mins after, 3) Read and re-read the docs first, then admit it if you REALLY don't understand. You're wasting your time and ours if you haven't at least tried. 4) If your problem ain't solved, come back in 12 hrs or 24 hrs later. We're very international. 5) Be polite ... |
| 19:32.46 | wunderkin | is the power plugged in |
| 19:33.22 | Corydon-w | Yes, but I hear that plea usually when somebody is trying to use the unlicensed version |
| 19:35.02 | FuriousGeorge | so anyone notice a correlation between electrical storms and packet loss? i use cable here |
| 19:35.22 | Idle | I have asterisk set to attach messages to my email, and delte them... I sometimes get my full messages, but I often get 4 second blank messages (probably just someone taking too long to hangup), but I often get a lot of '10' to 15 second messages, that are actually only about a 1 second message of a click... |
| 19:35.45 | Qwell[laptop] | Idle: It's just you |
| 19:35.53 | Qwell[laptop] | asterisk doesn't like you |
| 19:35.57 | Idle | aparently...... |
| 19:36.07 | kfudge | Sorry I was afk |
| 19:36.08 | Corydon-w | Have you tried setting the minimum length? |
| 19:36.14 | wunderkin | you have to be very happy for it to like you |
| 19:36.17 | Idle | Corydon-w: 3 seconds |
| 19:36.26 | kfudge | The codec works for the audio of a call but on hold music and such does not work |
| 19:36.28 | tzafrir_laptop | I'm having problems connecting Asterisk 1.0 and 1.2 via IAX. Connection is established, but no audio passes. |
| 19:36.30 | Idle | the 4 second ones dont bother me, cause theyre 4 seconds of silence |
| 19:36.56 | Corydon-w | kfudge: are you calling voip to voip? |
| 19:36.58 | tzafrir_laptop | I also see on 'iax debug' messages of the sort of: 'ignoring unknown information element 'Unknown IE' (46) of length 4' |
| 19:36.59 | Idle | its the 20 seconds of silence ones, and the '10' second ones with 1 second of silence that bother me |
| 19:37.15 | tzafrir_laptop | Any idea on which direction I should dig? |
| 19:37.21 | Qwell[laptop] | Corydon-w: Idle hasn't paid his Qwell tax this week |
| 19:37.26 | Corydon-w | Idle: analog lines? |
| 19:37.32 | *** join/#asterisk Splas (n=jwb@gate.lga2.us.voxel.net) |
| 19:38.02 | kfudge | voip to voip or voip to land line |
| 19:38.07 | *** join/#asterisk Assid (i=assid@203.115.83.214) |
| 19:38.19 | Idle | Corydon-w: wildcard tdm 400p |
| 19:38.24 | kfudge | or from a cell |
| 19:38.31 | Idle | Qwell[laptop]: pfft |
| 19:38.42 | Corydon-w | Analog... what a surprise |
| 19:39.01 | Corydon-w | kfudge: you using zaptel cards? |
| 19:39.44 | FuriousGeorge | does anyone _not_ notice a correlation between electrical storms and packet loss? what is your isp? |
| 19:39.58 | Juggie | FuriousGeorge, what kind of question is that. |
| 19:40.06 | Corydon-w | FuriousGeorge: only when the power goes out |
| 19:40.15 | Assid | hell.. i see a correlation between lightning and my router blowing up |
| 19:40.20 | FuriousGeorge | Juggie: no one responded when i asked it phrased positively |
| 19:40.41 | Corydon-w | and only for an extended period of time (i.e. when the UPS runs out of juice) |
| 19:41.04 | FuriousGeorge | we are being told left and right we sound "choppy" and we're breaking up today |
| 19:41.09 | kfudge | Nope. SIP and IAX trunks |
| 19:41.36 | Corydon-w | kfudge: I bet you don't have any g729 licenses |
| 19:41.50 | kfudge | But the in call audio works |
| 19:41.59 | Corydon-w | kfudge: no shit, really? |
| 19:42.13 | FuriousGeorge | ... yet we have more than ample bandwidth for a few phone calls. business grade cable ISP |
| 19:42.29 | kfudge | And I purchased and registered $400 bucks worth of licenses |
| 19:42.35 | Corydon-w | Gee, that wouldn't have anything to do with passthrough not transcoding, would it? |
| 19:43.17 | kfudge | was that meant for me? |
| 19:44.05 | Corydon-w | Please paste the output of: 'show g729' from the Asterisk CLI |
| 19:44.12 | Juggie | FuriousGeorge, coax can loose packets for a number of reasons |
| 19:44.18 | Juggie | if it gets wet, etc :) |
| 19:44.38 | Juggie | is your cable modem hooked directally up to a linux box or anything? |
| 19:44.47 | kfudge | 0/0 encoders/decoders of 20 licensed channels are currently in use |
| 19:44.51 | Juggie | or can you get a pc directally attached to it. |
| 19:45.13 | FuriousGeorge | Juggie: yeah, ipcop firewall |
| 19:45.19 | Juggie | whats that bsd? |
| 19:45.25 | FuriousGeorge | linux based |
| 19:45.25 | Corydon-w | Is that the same box where music on hold is being generated? |
| 19:45.30 | kfudge | Yes |
| 19:45.59 | Juggie | FuriousGeorge, what brand name modem do you know? |
| 19:46.04 | Corydon-w | kfudge: what happens when you use ulaw as the codec? |
| 19:46.04 | FuriousGeorge | Juggie: we just got the account so its one of the new ones. westell i believe |
| 19:46.10 | kfudge | Works fine |
| 19:46.31 | Juggie | FuriousGeorge, well one thing you can do is probe the modem to find out what the signal strenghts are and such |
| 19:46.36 | Juggie | and how much power its using to transmit |
| 19:46.48 | Juggie | your modem may have a built in web server, or you may have to nab it over snmp |
| 19:46.50 | *** join/#asterisk h3x (n=hex@CPE-24-208-84-116.neb.res.rr.com) |
| 19:46.59 | eKo1 | in 'show channels', are the newest show on the top or the bottom? |
| 19:47.00 | FuriousGeorge | Juggie: im not even sure its packet loss. whats a good tool for that? |
| 19:47.04 | kfudge | And I have another machine that has 4 licenses install but no one is using showing: 4/0 encoders/decoders of 4 licensed channels are currently in use |
| 19:47.16 | FuriousGeorge | Juggie: er, to test for that (packet loss) |
| 19:47.27 | Juggie | packet loss? :) |
| 19:47.28 | Juggie | ping? :) |
| 19:47.41 | Corydon-w | kfudge: are you using the latest g729 code from the Digium ftp site? |
| 19:47.46 | kfudge | yes |
| 19:47.51 | Juggie | FuriousGeorge, http://homepage.ntlworld.com/robin.d.h.walker/docsdiag/ |
| 19:48.07 | Corydon-w | kfudge: please call Digium tech support |
| 19:48.11 | wunderkin | FuriousGeorge, try http://192.168.100.1 first |
| 19:48.15 | kfudge | Ok. Thanks |
| 19:48.24 | drray | it's part of what you pay for |
| 19:49.28 | *** join/#asterisk Luhiwu (n=marsosa@200.63.89.212) |
| 19:50.16 | Luhiwu | hi all, anyone is using SER as redirect server before * ? |
| 19:50.55 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 19:50.58 | FuriousGeorge | wunderkin: my network is on the 10.0.0.1 scheme so thats not gonna be resolvable |
| 19:51.04 | FuriousGeorge | 10.0.0.x |
| 19:51.27 | FuriousGeorge | Juggie: well, i guess packet loss isnt my problem, cuz ping is working fine with any packet size |
| 19:52.07 | Juggie | FuriousGeorge, no idea then :) |
| 19:52.27 | FuriousGeorge | i just figgured if callers said we sounded choppy it must be packet loss |
| 19:52.43 | Juggie | where is your termination? |
| 19:52.44 | FuriousGeorge | and i figured it had to be the weather on account of the storms we had today |
| 19:52.45 | drray | test it with the milliwat app? |
| 19:53.04 | FuriousGeorge | Juggie: it goes to a closet. then 25 feet to a table with the modem |
| 19:53.28 | FuriousGeorge | drray: a voltimiter? |
| 19:53.35 | Juggie | FuriousGeorge, your pstn termination. |
| 19:53.37 | Juggie | not the cable modem |
| 19:54.01 | *** join/#asterisk viking78 (n=aherbert@cerberus.franklinamerican.com) |
| 19:54.07 | FuriousGeorge | Juggie: these calls are all voip, but the pstn goes to the nid then 50 feet over a piece of cat5 to * |
| 19:54.17 | FuriousGeorge | 4 lines |
| 19:54.41 | drray | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Milliwatt |
| 19:54.48 | *** join/#asterisk brif8 (n=Administ@ns1.ttienterprises.org) |
| 19:54.53 | Juggie | FuriousGeorge, no idea :) |
| 19:54.54 | drray | I use that to settle arguments that it has to be the network |
| 19:55.03 | Juggie | try doing a trace to one of the people who complained. |
| 19:55.18 | *** join/#asterisk benjamin7062 (n=benjamin@mailserver.photodex.com) |
| 19:55.24 | benjamin7062 | 'ello! |
| 19:56.16 | benjamin7062 | I'm sure this is a quick setting; we experience a mild amount of jitter testing out MeetMe using only SIP clients. I made the setting change for audiobuffers and made it the highest setting but still had the jitter? Any suggestions? |
| 19:57.23 | benjamin7062 | Not a biggy just curious |
| 19:57.43 | brif8 | what causes an "Auto fallthrough" status UNKNOWN on an incoming number is finds it in extensions.conf and processes line 1,2 and 3 but doesn't do 4,5,6 ? |
| 19:58.34 | wunderkin | show your dialplan |
| 19:58.43 | *** join/#asterisk crich1999 (n=crich@port-212-202-198-145.dynamic.qsc.de) |
| 19:59.00 | wunderkin | ~pb |
| 19:59.02 | jbot | pb is probably a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
| 19:59.07 | wunderkin | just that context |
| 19:59.32 | *** part/#asterisk knarfly_wk (n=bwatson@12.42.132.26) |
| 20:01.02 | FuriousGeorge | Juggie: the miliwatt test sounded fine to me using a remote sip client from home (same cable isp). then again the bad weather passed. ill have them call again and tell me if there still popping |
| 20:01.38 | brif8 | wunderkin: never mind I had the number wrong on the next line duhhhh |
| 20:02.32 | Qwell[laptop] | ~vi |
| 20:02.33 | jbot | it has been said that vi is the best text editor in the world. |
| 20:02.46 | wunderkin | ~pico |
| 20:02.47 | jbot | [pico] a non-free text editor. Try nano as a GPL'ed alternative, or if you insist on pico, download the source for pine from non-free, build it, and install the resulting pico_something.deb |
| 20:03.21 | drray | have them call the test app |
| 20:03.28 | drray | and channel spy |
| 20:03.43 | wunderkin | chanspy has never worked very well for me, unless some changes have been made |
| 20:03.57 | drray | I'm mainly zap, it works great |
| 20:04.00 | drray | for me |
| 20:04.08 | wunderkin | for zap just use zapscan |
| 20:04.12 | *** part/#asterisk brif8 (n=Administ@ns1.ttienterprises.org) |
| 20:04.27 | drray | zapscan must be new |
| 20:04.33 | wunderkin | no.. |
| 20:04.38 | drray | new from when I installed asterisk |
| 20:04.47 | wunderkin | umm before chanspy |
| 20:04.59 | *** join/#asterisk brif8 (n=Administ@ns1.ttienterprises.org) |
| 20:05.11 | drray | ok, so why should I use zapscan is chanspy works for me? |
| 20:05.21 | drray | what feature does it bring? |
| 20:05.31 | brif8 | 2,SetCDRUserField() should the text be in ' quotes " quotes or just (...) ? |
| 20:05.35 | wunderkin | if chanspy works for you great |
| 20:05.41 | *** join/#asterisk troytop (n=troyt@remote.activestate.com) |
| 20:06.41 | wunderkin | as far as features chanspy is better but it just hasnt worked well for me |
| 20:07.31 | So3kris | Strom_C: have you get it working ? |
| 20:08.06 | So3kris | My asterisk message is now - SIP/pstn-spa3k-92d3 is circuit-busy |
| 20:08.53 | drray | wunderkin, what has not worked? |
| 20:10.35 | wunderkin | choppy audio sometimes, even when the audio on the other side is fine |
| 20:10.40 | benjamin7062 | Anyone have experience with and/or use MeetMe? |
| 20:13.24 | Qwell[laptop] | benjamin7062: What about it? |
| 20:13.36 | FuriousGeorge | wunderkin: yeah they were saying that the could hear fine, but people were telling them they were breaking up and to repeat things |
| 20:13.56 | FuriousGeorge | the weather has subsided around us, and now its a lot better, but not gone |
| 20:13.58 | wunderkin | that was to drray |
| 20:14.13 | FuriousGeorge | wunderkin: i was just agreeing that it could be the case |
| 20:14.20 | benjamin7062 | Qwell[laptop], Hey, thanks... we experience jitter using only SIP phones. Direct calls work excellent. With Audio buffers enabled we see no real improvement. Have you experienced the same? |
| 20:14.35 | wunderkin | the audio on both sides was fine but chanspy was breaking up |
| 20:14.41 | FuriousGeorge | ah |
| 20:14.48 | benjamin7062 | Qwell[laptop], My bad, direct calls work great -- the jitter is only in MeetMe |
| 20:14.48 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.205.128.142) |
| 20:16.40 | Hmmhesays | timer timer timer |
| 20:17.55 | benjamin7062 | Hmmhesays, you think that is my problem? Perhaps our A104 is not working right with timing? |
| 20:18.15 | Hmmhesays | you get jitter when when you play ivr's too? |
| 20:18.29 | benjamin7062 | No sir |
| 20:18.41 | benjamin7062 | All announcements are perfect |
| 20:18.46 | Hmmhesays | odd |
| 20:19.26 | benjamin7062 | Sip -> Sip perfect... Just 'minor' jittor during meetme. AudioBuffers => 32 .. |
| 20:19.44 | benjamin7062 | 1.2.9.1 |
| 20:20.00 | *** join/#asterisk mk500 (n=mk500@netblock-72-25-104-105.dslextreme.com) |
| 20:20.22 | mk500 | hi all, I'm considering building asterisk system |
| 20:20.40 | mk500 | I'm in US and currently use an ISDN line for my two business lines |
| 20:20.55 | mk500 | just phones connected to an old 3com ISDN box |
| 20:21.14 | mk500 | is there a recommended card I could use to connect my ISDN line to an asterisk server? |
| 20:21.16 | benjamin7062 | mk500, then you will love *! |
| 20:21.30 | mk500 | benjamin: cool! I hope so :-) |
| 20:21.37 | *** join/#asterisk hohum (n=dcorbe@12.195.58.235) |
| 20:21.59 | mk500 | I'm a big fan of OS projects like this...and do a lot of MythTV stuff; so I'm hoping I can jump on Asterisk without too much pain |
| 20:22.35 | wunderkin | well it will depend if it is bri or pri |
| 20:22.54 | mk500 | I was using it for data originally....hmmm...how do I tell |
| 20:23.05 | benjamin7062 | We are rolling ours out as we speak after several weeks of testing to see if we wanted to replace our Avaya |
| 20:23.07 | mk500 | maybe I need to get into the box and look at config....haven't touched it in years |
| 20:23.15 | benjamin7062 | Very pleased! Wish you the best! |
| 20:23.35 | wunderkin | mk500, well is it 2 channels or more, how fast |
| 20:23.48 | mk500 | wunderkin: it is two channels |
| 20:23.52 | CunningPike | mk500: We are a local government with ~400 sets and are migrating from a Nortel - so far so good........ |
| 20:23.56 | mk500 | 128K |
| 20:23.56 | wunderkin | bri probably then |
| 20:23.58 | wunderkin | have fun |
| 20:24.08 | mk500 | benjamin: cool! thanks :-) |
| 20:24.20 | mk500 | wunderkin: is bri a nightmare? |
| 20:24.35 | wunderkin | i guess digium is releasing a bri card, there are other bri cards out there but mostly in europe |
| 20:24.52 | benjamin7062 | CunningPike, I suppose you guys didn't have problems with MeetMe... ;-) |
| 20:25.11 | CunningPike | benjamin7062: Not yet........ are you? |
| 20:25.32 | benjamin7062 | Yeah, just jitter... It's not a deal breaker anyway, just won't use the feature if it's something we have to deal with |
| 20:25.54 | drray | benjamin do you have the ztdummy stuff set? |
| 20:26.02 | znoG | benjamin7062: even with just 2 people in the conf you get jitter? |
| 20:26.03 | benjamin7062 | Although, the bug regarding warm sip transfers and the queue locking up... ouch |
| 20:26.12 | benjamin7062 | znoG, yes, unfortunately... |
| 20:26.44 | benjamin7062 | znoG, But sip->sip and ivr all sound excellent... so, meetme is just 'extra' really |
| 20:27.23 | CunningPike | benjamin7062: Sounds like you're missing a timing source |
| 20:27.24 | drray | benjamin - do you have the ztdummy stuff set? |
| 20:28.04 | znoG | benjamin7062: are you using X windows on the * box? |
| 20:28.21 | benjamin7062 | CunningPike, might be... We have a Sangoma A104 with the HWEC and use that for timing so... hmm? |
| 20:28.28 | benjamin7062 | znoG, No sir, we are all console.. =) |
| 20:28.39 | drray | hmmm |
| 20:28.57 | drray | that's a timing soruce |
| 20:29.03 | benjamin7062 | CunningPike, one thing to note, our PRI's aren't plugged in yet... is timing pulled from the actual PRI.. or just the card? |
| 20:29.08 | znoG | benjamin7062: what SIP clients are you using? |
| 20:29.09 | rob0 | Does framebuffer console (graphical video mode) have the same issues with Zaptel that X does? |
| 20:29.23 | benjamin7062 | znoG, All Polycom 601's |
| 20:29.47 | wunderkin | benjamin7062, yes i think that matters |
| 20:29.53 | znoG | benjamin7062: have you tried, for the sake of it, 2 DIAX clients or something into the meetme? |
| 20:30.01 | CunningPike | benjamin7062: Depends on your zaptel.conf - if you are set up to pull timing from the PRI and it's not plugged in, you're not getting any |
| 20:30.29 | CunningPike | rob0: The recommendation is to boot with the nofb option |
| 20:30.52 | nextime | rob0 : usually fb isn't problematic, anyway, it is better to not use fb if you have zaptel hardware to prevent any possible trouble |
| 20:31.06 | benjamin7062 | wunderkin, That very well could be my problem |
| 20:31.16 | rob0 | Thanks, yes, my * server is text mode only. |
| 20:31.22 | rob0 | just curious |
| 20:31.37 | wunderkin | benjamin7062, maybe still setup ztdummy? |
| 20:31.40 | piper69 | can someone help on where i can get AT&T command tutorial , i ve seen a PDF before but i can't recall where |
| 20:31.43 | benjamin7062 | znoG, No sir, only the sip clients. I assume since quality seems excellent for other applications that this is probably something trivial on my server |
| 20:32.13 | *** join/#asterisk setti (n=setti@191-79-112.adsl.terra.cl) |
| 20:32.21 | CunningPike | benjamin7062: In zaptel.conf, set span=1,0,0....... until you get your PRI plugged in |
| 20:32.34 | benjamin7062 | Awesome... will do that |
| 20:32.37 | CunningPike | benjamin7062: That will force your card to use its own timing |
| 20:32.58 | wunderkin | oh.. |
| 20:33.09 | benjamin7062 | Sorry for the two line post but we have two pri's... and here is there config: |
| 20:33.12 | benjamin7062 | span=1,1,0,esf,b8zs |
| 20:33.12 | benjamin7062 | span=2,2,0,esf,b8zs |
| 20:33.32 | wunderkin | 1,0 2,0 if they arent being used then |
| 20:33.34 | benjamin7062 | so the second column defines the timing source |
| 20:33.36 | benjamin7062 | gotcha |
| 20:33.39 | znoG | benjamin7062: when you dial the voicemail app on your * server from the Polycom, does it sound OK? |
| 20:33.50 | benjamin7062 | znoG, yes sir, sounds excellent |
| 20:34.16 | znoG | ok then listen to CunningPike because I'm outta questions and ideas |
| 20:34.39 | benjamin7062 | znoG, LOL -- thank you very much for the suggestions. I will try ztdummy also |
| 20:34.54 | wunderkin | this will probably work, never had to do it before, i dunno |
| 20:36.38 | drray | you don't need ztdummy |
| 20:36.44 | drray | you have a timing source |
| 20:36.55 | drray | but meetmee and certian other things, need that source |
| 20:36.58 | benjamin7062 | =( No go, changed the zaptel.conf appropriately, ran ztcfg -vvv etc, then restarted *... and bummer, still the minor minor jitter |
| 20:37.02 | benjamin7062 | no load on the machine either |
| 20:37.38 | drray | turn the volume down? ;) |
| 20:37.51 | drray | I solved some echo the other day by turning the line down |
| 20:38.33 | *** join/#asterisk s0lid (n=s0lid@61.28.166.66) |
| 20:39.18 | benjamin7062 | drray, you mean on the sip phones? or specific config somewhere? We don't get echo so much as ... slight stutter... Imagine talking through a fan... Very similar to that |
| 20:40.11 | mocker | Wow. |
| 20:40.13 | benjamin7062 | Since 'stutter' refers to the ring type for a message I assumed the correct word for that sound is jitter... |
| 20:40.24 | mocker | Just figured out a crazy problem w/ asterisk 1.2.5. |
| 20:40.37 | Strom_C | why are you running 1.2.5? :) |
| 20:40.39 | mocker | the Playback() command in my dialplan was rewritting my caller id. :) |
| 20:40.53 | mocker | Strom_C: Good question! |
| 20:41.31 | benjamin7062 | mocker, that is by design... because when you play a file.. it is assumed you want to change your caller idea.. err, wait, that's silly |
| 20:41.45 | benjamin7062 | =) |
| 20:42.04 | mocker | benjamin7062: Ahhh, that makes sense. :) |
| 20:42.11 | benjamin7062 | mocker, exactly! |
| 20:42.51 | benjamin7062 | Try this one, setup an agent queue... login as an agent... call the queue... have the answering agent do a warm transfer to any other phone and connect the call... |
| 20:43.05 | benjamin7062 | Wubam! |
| 20:43.09 | *** join/#asterisk feld_ (n=feld@12.148.212.157) |
| 20:43.19 | benjamin7062 | Next inbound queue call locks up * and requires restart |
| 20:43.34 | mocker | benjamin7062: Seriously? |
| 20:43.48 | mocker | I've avoided transfers because everyone I work with uses park. |
| 20:44.18 | benjamin7062 | mocker, direct transfers (#) work fine... only warm transfers (sip transfers) |
| 20:44.36 | mocker | benjamin7062: Is asterisk dropping out of the media path? |
| 20:44.38 | benjamin7062 | I have a ruler ready for our call center staff... I'm going to smack knuckles |
| 20:44.44 | feld_ | I have 5 outgoing lines over 2 TDM400Ps. What's the easiest way to mod my dial plan so that outgoing calls will try the next line? (Zap/5-8) |
| 20:45.04 | benjamin7062 | mocker, Nope, queue app locks up... it's hawt. There's already a bug but it isn't fixed yet which is eating my lunch |
| 20:45.08 | Strom_C | feld_: use the group setting in zapata.conf |
| 20:45.19 | feld_ | Strom_C: ahhh, the group. I knew I was forgetting something. |
| 20:45.30 | Strom_C | shebang |
| 20:45.31 | feld_ | thanks for the reminder. |
| 20:46.25 | benjamin7062 | Will Digium modify and/or fix bugs for $$$? |
| 20:46.30 | feld_ | I have them all in Group=1.... so is the context Group/1/ instead of Zap/1/ ? |
| 20:46.36 | Strom_C | benjamin7062: I believe so ;) |
| 20:46.42 | Strom_C | no, feld_ |
| 20:47.12 | Strom_C | feld_: if it's Group=1, use Dial(ZAP/G1/${EXTEN}) to hunt from the high numbered channel |
| 20:47.23 | feld_ | aha! thanks. |
| 20:47.29 | Strom_C | or use Dial(ZAP/g1/${EXTEN}) to hunt from the low numbered channel |
| 20:47.33 | feld_ | I've been out of * for like... 3 weeks. other projects have interrupted my work. |
| 20:47.42 | Strom_C | traitor! |
| 20:47.46 | feld_ | no kidding |
| 20:47.47 | benjamin7062 | To make sure I'm being clear with everyone, if the call sounds like one party is talking through a fan .. is the term Jitter? |
| 20:48.10 | Strom_C | benjamin7062: yes, unless the problem is packet loss |
| 20:48.25 | benjamin7062 | Strom_C, excellent.. thanks |
| 20:48.32 | znoG | benjamin7062: i actually had that problem once. I just hanged up all parties, dialed up again and it was OK again |
| 20:49.02 | benjamin7062 | You guys don't know of any other tweaks that might cause MeetMe jitter other than timing and/or audiobuffers? If not, I shall proceed without it and just avoid the feature whcih is fine for us |
| 20:49.25 | Strom_C | benjamin7062: how are people coming into the meetme, and what is your timing source |
| 20:49.37 | benjamin7062 | znoG, Yeah, I've tried many times.. Have two sips on my desk and can reliably reproduce with only two members |
| 20:49.44 | ph|ber | anyone know why on polycom 301, when you dial a # that answers and requires you to press options, that it doesnt send the # pressed? |
| 20:50.07 | Strom_C | ph|ber: wrong dtmfmode in sip.conf |
| 20:50.09 | benjamin7062 | Strom_C, All sip agents... (we've tried 5 but now I test with 2)... Timing source is a Sangoma A104D (with EC) |
| 20:50.20 | Strom_C | benjamin7062: SIP users are on your LAN? |
| 20:50.22 | ph|ber | rfc2833 |
| 20:50.35 | benjamin7062 | Strom_C, PRI's are not hooked up so as directed above have made the config point to the card for timing |
| 20:50.51 | benjamin7062 | Strom_C, Yes sir, latency is around 1ms ish.. stable |
| 20:51.39 | Strom_C | just for shits and giggles, disable the card and try using ztdummy |
| 20:52.05 | E-bola | Have anybody here setup an asterix server that works with nat clients? |
| 20:52.14 | benjamin7062 | is that as simple as commenting out the lines in zaptel.conf, running ztcfg and modprobe ztdummy? If necessary, I will read |
| 20:52.23 | Strom_C | benjamin7062: no |
| 20:52.27 | benjamin7062 | K, |
| 20:52.36 | benjamin7062 | I'll go read .. I know docs exist for ztdummy |
| 20:52.43 | Strom_C | benjamin7062: rmmod whatever the hell sangoma's zaptel hacks are, and then modprobe ztdummy |
| 20:52.52 | mocker | benjamin7062: Have you run zttest to see how your interrupts are running? |
| 20:53.05 | Strom_C | this is, of course, assuming sangoma doesnt mung up ztdummy too |
| 20:53.37 | benjamin7062 | I can cleanly remove their modules |
| 20:53.43 | drray | sangoma is tough on a zaptel module |
| 20:53.47 | ph|ber | Strom_C: i tried inband, auto, and rfc2833. none work. |
| 20:53.51 | ph|ber | anyother ideas? |
| 20:53.52 | drray | maybe that's his problem? |
| 20:54.00 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 20:54.04 | benjamin7062 | my zaptel conf can stay the same with ztdummy? |
| 20:54.09 | Strom_C | benjamin7062: yes |
| 20:54.12 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.128.Dial1.SanJose1.Level3.net) |
| 20:54.21 | benjamin7062 | kewl... is ztdummy just a wrapper of some type |
| 20:54.23 | *** join/#asterisk murf (n=murf@gateway.digium.com) |
| 20:54.29 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 20:54.31 | piper69 | it seems that i installed asterisk 1.0.7.dfsg.1-2sarg open source Private Branch Exchange (PBX)....how can i get 1.2.9 |
| 20:54.40 | benjamin7062 | or I suppose a dumb echo of some kind |
| 20:54.44 | Strom_C | benjamin7062: you're running a 2.6 kernel right? |
| 20:54.49 | benjamin7062 | Strom_C, yes sir |
| 20:54.52 | Strom_C | piper69: www.asterisk.org |
| 20:54.53 | drray | piper from svn |
| 20:55.01 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 20:55.17 | Strom_C | ph|ber: maybe your phone's config is screwed up |
| 20:55.37 | piper69 | Strom_C: i download it the 1.2.9 but i don't know how to compile it |
| 20:56.00 | Strom_C | piper69: untar and do the make commands |
| 20:56.10 | Strom_C | piper69: asterisk.org tells you what to do |
| 20:56.27 | piper69 | Strom_C: ok let me check i t |
| 20:58.59 | gandhijee | does zaptel not play nice with 2.6.13? |
| 20:59.03 | benjamin7062 | To save me the read, will someone shoot me a line that will work with ztdummy to replace (span=1,1,0,esf,b8zs) |
| 20:59.06 | *** join/#asterisk jijgeh (n=jijgeh@72.166.51.162) |
| 20:59.07 | Qwell[laptop] | gandhijee: should work fine |
| 20:59.12 | *** part/#asterisk jijgeh (n=jijgeh@72.166.51.162) |
| 20:59.18 | Strom_C | benjamin7062: you dont need to change anythig |
| 20:59.21 | gandhijee | Qwell: zaptel: no version for "struct_module" found: kernel tainted. |
| 20:59.25 | Strom_C | benjamin7062: just load ztdummy |
| 20:59.37 | benjamin7062 | ZT_SPANCONFIG failed on span 1: Invalid argument (22) |
| 20:59.37 | benjamin7062 | FATAL: Error running install command for ztdummy |
| 20:59.43 | Strom_C | benjamin7062: so comment it out |
| 20:59.46 | piper69 | Strom_C: can you please post me a link |
| 20:59.49 | benjamin7062 | Sorry for the posts to channel -- won't continue if they get long |
| 20:59.57 | Strom_C | piper69: http://www.asterisk.org/ |
| 20:59.59 | benjamin7062 | Strom_C, k.. thank you |
| 21:00.11 | gandhijee | nm |
| 21:00.17 | gandhijee | i screwed up |
| 21:00.49 | piper69 | Strom_C: i am there but i can't seem to find where it show me how to compile |
| 21:00.57 | Strom_C | click "download" |
| 21:01.17 | terrapen | this really sucks |
| 21:01.26 | terrapen | nobody has seen a problem where call parking stops working? |
| 21:01.27 | *** join/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net) |
| 21:01.31 | TripleFFFF | app_rxfax.c: In function `phase_e_handler': |
| 21:01.35 | TripleFFFF | darn |
| 21:01.42 | terrapen | where, when the IP phone user dials the parking code, the caller hears the DTMF? |
| 21:01.48 | TripleFFFF | any idea ? |
| 21:02.03 | TripleFFFF | dotn remember the problem |
| 21:02.04 | THX2000 | Anyone know of a devious way to get the cisco firmwares? |
| 21:02.13 | TripleFFFF | t30 i thinki need to diable could it be ? |
| 21:02.15 | dlynes_laptop | THX2000: www.ciscowarez.com |
| 21:02.18 | terrapen | heh |
| 21:02.18 | drray | insight, for $90 |
| 21:02.26 | THX2000 | hehe, sweet ;) |
| 21:03.42 | drray | which reminds me, I need to get the new firmwares before my support contract expires, august 1st |
| 21:04.03 | So3kris | Strom_C: can you call outbound with the pstn line |
| 21:04.14 | Strom_C | So3kris: not yet |
| 21:04.19 | So3kris | m2 |
| 21:04.25 | *** join/#asterisk topping (n=topping@207.47.6.136.static.nextweb.net) |
| 21:04.39 | So3kris | connect you at the 5060 port at the 3102 ? |
| 21:04.59 | TripleFFFF | anyone ? |
| 21:05.21 | THX2000 | drray: I'll let u store em on my server if u want ;) |
| 21:05.26 | So3kris | I read at some forms that there must be a 5062 port but i can't find it |
| 21:05.42 | benjamin7062 | Strom_C, That proved the be the issue; to confirm, I changed back to the Sangoma card for timing and experienced jitter once again in Meetme. Interesting I don't experience anywhere else. I will try to identify whether we have jitter while connected to PRI's. If we do, perhaps we'll switch cards |
| 21:05.58 | benjamin7062 | Strom_C, thank you kindly for pointing me down that path! |
| 21:06.13 | Strom_C | benjamin7062: very welcome! and next time buy digium ;) |
| 21:06.20 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 21:06.37 | feld_ | lol oh boy |
| 21:06.45 | THX2000 | Strom_C: still workin on the 3102? |
| 21:06.55 | Strom_C | THX2000: havent really had time yet |
| 21:06.57 | feld_ | Strom_C says to buy Digium, [TK]D-Fender says to buy Sangoma.... what's a man to do?! :) |
| 21:07.07 | benjamin7062 | Strom_C, I believe you guys just recently got the HWEC stuff, right? Don't worry, you guys will still (and already have) get our money... |
| 21:07.10 | THX2000 | Strom_C: finish your tea already :P |
| 21:07.14 | drray | govarion.com |
| 21:07.15 | drray | :) |
| 21:07.27 | Strom_C | benjamin7062: yeah, the new cards have the otasic echo cans |
| 21:08.35 | Strom_C | otasic? octasic? (it all looks like Ocasek to me) |
| 21:09.00 | benjamin7062 | =) |
| 21:09.06 | TripleFFFF | can one help on this matter ? http://pastebin.ca/86421 |
| 21:09.18 | piper69 | please can someone help to compile 1.2.9.1 in Debian |
| 21:09.43 | Strom_C | piper69: asterisk.org gives you instructions. it tells you what packages you need. |
| 21:09.59 | piper69 | Strom_C: where , can you show me |
| 21:10.07 | Strom_C | on the "download" page |
| 21:10.09 | TripleFFFF | darn |
| 21:10.16 | TripleFFFF | fax not compiling 1.2.9.1 |
| 21:11.37 | piper69 | Strom_C: SVN repository??? |
| 21:12.00 | Strom_C | piper69: read further please |
| 21:13.38 | piper69 | Strom_C: for god sack will it hurt you if you guide me where i can find it, i have been looking since morning and i am at the point where i can't see it at all |
| 21:13.42 | TripleFFFF | . |
| 21:13.43 | TripleFFFF | . |
| 21:14.04 | THX2000 | piper69: chill, asterisk.org > Download |
| 21:14.05 | piper69 | Strom_C: and i know it;s going to b infront of my darn face and i can't see it |
| 21:14.07 | Strom_C | piper69: at the bottom of the page it says "THESE ARE THE PACKAGES YOU NEED TO INSTALL" |
| 21:14.08 | THX2000 | its all right there |
| 21:14.32 | Strom_C | piper69: if you want me to hold your hand, pay me a consulting fee |
| 21:15.00 | Strom_C | and I will do all the handholding you want |
| 21:15.12 | piper69 | Strom_C: i will pay you to come and slap me on my face |
| 21:15.43 | Strom_C | woot |
| 21:16.20 | *** join/#asterisk heffernan (n=hwt@curb.thorkildssen.com) |
| 21:16.52 | piper69 | Strom_C: THX2000 :huh i can't see what you said |
| 21:17.55 | piper69 | i am sorry guys, but i feel like i am useles i can't even see infront of me |
| 21:17.59 | *** join/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com) |
| 21:18.08 | *** part/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com) |
| 21:18.11 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 21:18.14 | piper69 | and i don't know where to get the right guide to work with |
| 21:18.42 | piper69 | damn boy flying is a lot lot easier |
| 21:19.05 | heffernan | hi. i'm using realtime, and i have several users in a table in mysql. i don't want to make a dialplan rule for every number, and the numbers are inported and not a continuous series. how can i achieve that? |
| 21:19.06 | drray | well, you'll survive a crash with asterisk more often |
| 21:19.17 | heffernan | both DID and internal calls. |
| 21:19.29 | *** join/#asterisk X-Rob (n=rob@CPE-58-164-139-115.qld.bigpond.net.au) |
| 21:19.36 | heffernan | and when is it really a point in having extensions in a table? |
| 21:19.55 | heffernan | the way i see it, the extensions really ought to be as static as possible.. with matching and stuff. |
| 21:20.03 | Strom_C | piper69: you're frustrated. |
| 21:20.13 | Strom_C | piper69: relax and calm down and things will be more apparent to you |
| 21:20.20 | feld_ | piper69: got beer? |
| 21:20.56 | *** join/#asterisk DarKnesS_WolF (n=wolf@82.201.230.198) |
| 21:21.42 | Strom_C | ah, beer |
| 21:21.47 | Strom_C | beer is good |
| 21:22.32 | heffernan | anybody? |
| 21:22.40 | hads | mmm beer. |
| 21:22.45 | znoG | someone say beer? |
| 21:22.54 | hads | Pity it's the morning here. |
| 21:23.08 | drray | it is too early to drink? |
| 21:23.26 | So3kris | Strom_C: Is this bull shit ? the post from mberlant http://voxilla.com/index.php?name=PNphpBB2&file=viewtopic&t=9365 |
| 21:23.29 | hads | 0923 |
| 21:23.35 | [TK]D-Fender | drray : "Beer : Its not just for breakfast anymore" |
| 21:23.41 | znoG | drray: it depends. See, when it's too early to drink, I look at other countries' timezones and drink with them, where it's nice and late and normal to drink. |
| 21:24.28 | heffernan | beer.. the cause of, and solution to, all of life's problems. |
| 21:24.46 | drray | I thought life itself was the cause of all of lifes problems |
| 21:24.50 | drray | think about it |
| 21:25.05 | drray | that and marketting |
| 21:25.09 | heffernan | drray: life's the price to pay for living. |
| 21:25.18 | *** join/#asterisk MikeJ__ (n=vircuser@208-45-120-111.dlth.qwest.net) |
| 21:27.07 | hads | Yay for the -users list. |
| 21:28.14 | znoG | as if there weren't enough idiots in the $3,000 server thread, Harvey jumps in. |
| 21:28.15 | heffernan | but hey, any help on the realtime issue? |
| 21:28.24 | *** join/#asterisk digus (n=digus@206.222.110.30) |
| 21:28.40 | hads | It makes it easier to catch up in the morning when you can just "Mark all as read" |
| 21:28.46 | *** join/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com) |
| 21:29.00 | *** part/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com) |
| 21:29.02 | *** join/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com) |
| 21:29.05 | *** part/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com) |
| 21:30.41 | *** join/#asterisk HolyGod (i=nobody@got.securebinary.com) |
| 21:31.43 | So3kris | Strom_C: i got it a little futher |
| 21:32.05 | So3kris | At the pstn line i got a divertn port than line 1 |
| 21:32.12 | So3kris | than it wil work |
| 21:32.29 | So3kris | I got nauw a Jul 12 23:30:24 NOTICE[29662]: chan_sip.c:3691 process_sdp: No compatible codecs! |
| 21:32.33 | So3kris | error |
| 21:33.31 | *** join/#asterisk bmg505 (n=leon@c1-234-9.rndf.isadsl.co.za) |
| 21:34.27 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-155-126-111.red.bezeqint.net) |
| 21:35.06 | TripleFFFF | loader.c:325 __load_resource: libspandsp.so.0: cannot open shared object file: No such file or directory |
| 21:35.12 | TripleFFFF | anuyreason ? |
| 21:35.18 | *** join/#asterisk alexis101 (n=alexis@70.54.204.92) |
| 21:35.20 | TripleFFFF | its there in /usrlib |
| 21:35.21 | *** part/#asterisk brif8 (n=Administ@ns1.ttienterprises.org) |
| 21:35.45 | Hmmhesays | My manager has seriously lost it |
| 21:35.55 | *** join/#asterisk stoffell_h (n=kristof@pot.catsanddogs.com) |
| 21:36.56 | alexis101 | hi , im looking for a list of change or feature that will be in 1.4 ... i tried to google it but din't find anything ... anyone have a link |
| 21:37.19 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
| 21:37.49 | Hmmhesays | Seriously |
| 21:37.57 | Hmmhesays | wow |
| 21:38.37 | iax | hi! |
| 21:38.44 | ss7 | hi! |
| 21:38.45 | *** join/#asterisk stoffell_h (n=stoffell@pot.catsanddogs.com) |
| 21:38.58 | wunderkin | /nick mgcp |
| 21:39.37 | wunderkin | dork |
| 21:39.37 | ss8 | that's what I thought fool! |
| 21:40.01 | So3kris | iax2: its working |
| 21:40.25 | Hmmhesays | wow |
| 21:40.28 | So3kris | Strom_C its working out and inbound at the 3102 |
| 21:40.37 | SS7 | So3kris: oh cool, what was the trick? |
| 21:40.38 | mgcp | hehe |
| 21:40.39 | mgcp | :D |
| 21:41.09 | TripleFFFF | anyone ? |
| 21:41.13 | So3kris | SS7: the trick is that you must the second sip accound running at port 506X |
| 21:41.28 | SS7 | i had the second one running at 5061 and it wasnt working |
| 21:41.46 | So3kris | got you configured it at the 3102 ? |
| 21:41.54 | TripleFFFF | ok 5$ paypal to one who make it work |
| 21:41.58 | So3kris | and asterisk |
| 21:42.08 | SS7 | ok, ill dink with it |
| 21:42.29 | So3kris | http://voxilla.com/index.php?name=PNphpBB2&file=download&id=623 |
| 21:42.45 | So3kris | thats a good exemple of a 3XXX |
| 21:43.00 | So3kris | and look good at the port numbers |
| 21:45.58 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 21:47.06 | SS7 | ill be back shortly |
| 21:47.26 | digus | Hello all. I'm trying to associate an extension to a particular trunk so all calls placed from that extension only use that 1 trunk outbound. Haven't had any luck on google – any takers? Is there something simple I'm overlooking here? |
| 21:47.50 | drray | digus - groups? |
| 21:48.23 | drray | digus - contexts and or groups |
| 21:49.14 | digus | drray: thanx - i'll start digging |
| 21:49.19 | piper69 | feld_: am at work ...i wish i had beer |
| 21:49.40 | drray | basically, you'd create a context for the outbound trunk and put that extension in that context |
| 21:50.27 | digus | cool - thanks |
| 21:51.16 | terrapen | i'll buy a beer for whoever can figure this out |
| 21:51.28 | terrapen | why call parking stopped working this morning |
| 21:51.55 | drray | did you update your kernel or system? |
| 21:51.58 | terrapen | nope, nothing |
| 21:52.01 | terrapen | no config changes, nothing |
| 21:52.12 | drray | hd drive corruption? |
| 21:52.22 | terrapen | when a polycom phone user dials #999 (the code to park), the caller hears DTMF |
| 21:52.31 | terrapen | well, sort of like DTMF...they hear popping |
| 21:52.35 | *** join/#asterisk postel (n=jp@unaffiliated/postel) |
| 21:52.41 | terrapen | * is not detecting the DTMF for some reason |
| 21:55.07 | piper69 | ok so what do i need to compile 1.2.9.1 in debian |
| 21:56.47 | terrapen | ok, this is really odd: |
| 21:57.02 | terrapen | calls from one extension to another can be parked |
| 21:57.11 | terrapen | but for calls coming in over the PRI, it does not work |
| 21:57.27 | terrapen | I wonder if it is a driver problem with Sangoma card |
| 21:57.29 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 21:57.47 | *** join/#asterisk ingOmar (n=chatzill@remote.activestate.com) |
| 21:58.12 | benjamin7062 | piper69, from the README: |
| 21:58.15 | benjamin7062 | On many distributions, these files are installed by packages with names like |
| 21:58.15 | benjamin7062 | 'glibc-devel', 'ncurses-devel', 'openssl-devel' and 'zlib-devel' or similar. |
| 21:58.39 | benjamin7062 | piper69, so get your apt-cache search out and start digging |
| 21:58.42 | benjamin7062 | ;-) |
| 22:00.31 | piper69 | benjamin7062: but with apt i only get ver 1.0.7.1 |
| 22:00.38 | TripleFFFF | offer closed thanks |
| 22:01.38 | *** join/#asterisk type0h (i=type0@216-67-30-34-cdsl-rb2.cwc.acsalaska.net) |
| 22:01.44 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 22:01.53 | type0h | anyone read that nerdcore article on slashdot today? |
| 22:03.02 | benjamin7062 | piper69, 1.0.7.1 of what ? |
| 22:03.06 | heffernan | hm, i have a problem here. when i dial from sip phones, it does not try the realtime extensions in its own context before using the outbound trunk. |
| 22:03.33 | piper69 | benjamin7062: asterisk 1.0.7.dfsg.1-2sarg open source Private Branch Exchange (PBX) |
| 22:03.36 | heffernan | i have a switch => Realtime/internal@extensions, and a exten => _X.,1,Macro(internal,SIP/${EXTEN}) |
| 22:03.41 | heffernan | shouldn't that be sufficient? |
| 22:04.07 | benjamin7062 | piper69, Oh, you want to do the full install via debian? |
| 22:04.16 | benjamin7062 | piper69, I thought you wanted to compile on debian |
| 22:04.36 | benjamin7062 | piper69, Beh -- You get the version available... or you need to find someone maintaining a newer version |
| 22:04.59 | heffernan | piper69: check astersikguru.com, they have a nice tutorial on building * on debian. |
| 22:05.11 | benjamin7062 | piper69, Can't help you; I'm running on debian.. but 1.2.9.1 from source |
| 22:05.33 | heffernan | piper69: or, if you really want debs, i suggest you set up apt-pinning with packages from unstable. |
| 22:05.40 | heffernan | piper69: or look into debian backports. |
| 22:06.31 | THX2000 | piper69: why can't u just compile from the svn download? |
| 22:06.44 | heffernan | oh, yeah, and i don't have any 00 (or similar) prefix that separates outbound from internal calls. |
| 22:06.49 | heffernan | so it's impossible to match on. |
| 22:07.05 | heffernan | the only way i got it working is by using one line per extension. |
| 22:07.49 | piper69 | benjamin7062: what do you think about 1.2.7.1 |
| 22:08.04 | piper69 | benjamin7062: because i can get that in apt- |
| 22:08.05 | *** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 22:08.05 | heffernan | perhaps by using Local or something? |
| 22:08.16 | THX2000 | why are you so stuck on apt-get? |
| 22:08.26 | heffernan | piper69: that's pretty recent. |
| 22:08.48 | heffernan | personally, i wouldn't trust the debian asterisk package manager. |
| 22:08.51 | piper69 | THX2000: something you know is better that something you don't know...i tried to go the other route but there was no help at all |
| 22:09.07 | heffernan | since he is probably a debian guy, not an asterisk guy. |
| 22:09.25 | THX2000 | svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk |
| 22:09.26 | THX2000 | done |
| 22:09.33 | THX2000 | make clean; make install |
| 22:09.35 | THX2000 | run asterisk |
| 22:10.16 | heffernan | THX2000: make samples. ;) |
| 22:10.42 | THX2000 | lol... make progdocs ... if u wanna get tricky ;) |
| 22:11.08 | Dr-Linux | heffernan: he needs to play with configs :P |
| 22:11.28 | piper69 | THX2000: get tricky are you serious ...lol |
| 22:11.35 | Dr-Linux | nortex: what's up dude? |
| 22:12.29 | piper69 | THX2000: the url didn't work...sorry I KNOW you want to shoot me by now |
| 22:12.36 | tzafrir_laptop | heffernan, what's your issue with it? |
| 22:12.52 | THX2000 | if u have internet access on this box then u mistyped it |
| 22:13.16 | heffernan | tzafrir_laptop: it tries calling it through the outbound trunk |
| 22:13.19 | *** join/#asterisk nagl (n=nagl@86.59.54.237) |
| 22:13.23 | heffernan | tzafrir_laptop: i can pastebin it if you want? |
| 22:13.54 | piper69 | THX2000: where should i type that < THX2000> svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk |
| 22:14.06 | THX2000 | cd /usr/src |
| 22:14.09 | THX2000 | then run that |
| 22:14.14 | tzafrir_laptop | I was actually referring to your issue with the "debian asterisk package manager". But pastebin away... |
| 22:14.38 | tzafrir_laptop | If you're looking for stability, trunk is still not the place |
| 22:15.27 | piper69 | THX2000: i can get that to work |
| 22:15.31 | heffernan | tzafrir_laptop: http://pastebin.ca/86471 |
| 22:15.41 | heffernan | tzafrir_laptop: that's the dialplan with the realtime issue. |
| 22:15.51 | heffernan | tzafrir_laptop: although i do not think it's realtime related. |
| 22:15.56 | THX2000 | ok so now u have a /usr/src/asterisk directory? |
| 22:16.31 | heffernan | tzafrir_laptop: i don't have any issues with the debian package, it's just that it usually lags behind upstream asterisk. |
| 22:16.54 | tzafrir_laptop | depends what package. |
| 22:17.01 | piper69 | THX2000: nope i have the extract dir for astersik |
| 22:17.36 | *** join/#asterisk daysmen3 (n=primus@host81-158-206-221.range81-158.btcentralplus.com) |
| 22:18.45 | *** join/#asterisk carl0s- (n=carl0s@compsup.demon.co.uk) |
| 22:18.48 | THX2000 | so what happens when u type: cd /usr/src/asterisk |
| 22:18.56 | Luke-Jr | anyone setup Asterisk to dial by decimal IP address? |
| 22:19.28 | piper69 | THX2000: i can't get there i have this /home/piper69/asterisk-1.2.9.1 |
| 22:20.01 | THX2000 | so then u didn't run the svn from /usr/src |
| 22:20.29 | carl0s- | can anybody tell me where to look to diagnose my X100P not being recognised? I had four cards in before, and two were being deteced ("card 1 & 3") with alarm status RED, but not working. Now I have a single card in, and "zap show status" only shows ZTDUMMY. /proc/pci shows the device is there. I have compiled zaptel-1.2.6 and done rebuild_zaptel. I'm using trixbox btw, is that frowned upon? |
| 22:20.35 | Luke-Jr | eg, dial 4294967295 to get 255.255.255.255 (ffffffff) |
| 22:20.51 | heffernan | <PROTECTED> |
| 22:20.54 | wunderkin | 4 cards, wow |
| 22:20.55 | tzafrir_laptop | red alarm: cable disconnected? |
| 22:21.23 | carl0s- | tzafrir_laptop: could be, but regardless.. there were only two cards shown so I'm down to a single card while I work it out. But now none are showing. |
| 22:21.29 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 22:21.33 | CoffeeKid | i keep getting the warning: WARNING[25925]: channel.c:787 channel_find_locked: Avoided initial deadlock for '0x737650', 10 retries! ... anyone seen this before? |
| 22:21.39 | carl0s- | wunderkin: they're only single-port modems though. nothing fancy. |
| 22:21.41 | tzafrir_laptop | carl0s-, what do you have in /proc/zaptel ? |
| 22:21.52 | piper69 | THX2000: bash: svn: command not found |
| 22:21.52 | carl0s- | tzafrir_laptop: ah. let me see. |
| 22:22.13 | tzafrir_laptop | piper69, you need auto-apt |
| 22:22.23 | THX2000 | piper69: apt-get install subversion |
| 22:22.27 | carl0s- | tzafrir_laptop: I have '1' which reports: Span 1: ZTDUMMY/1 "ZTDUMMY/1 1" |
| 22:22.50 | tzafrir_laptop | only '1'? |
| 22:22.58 | carl0s- | tzafrir_laptop: yup. |
| 22:23.07 | tzafrir_laptop | lsmod | grep zaptel |
| 22:23.15 | carl0s- | tzafrir_laptop: so that means the wxfxo module is only seeing one useful card? |
| 22:23.36 | tzafrir_laptop | no. ztdummy is unrelaetd to wcfxo |
| 22:23.44 | carl0s- | zaptel 206852 9 ztdummy,wctdm,wcfxo,wcte11xp,wct1xxp,wct4xxp,tor |
| 22:23.44 | carl0s- | 2 |
| 22:23.44 | carl0s- | crc_ccitt 2113 1 zaptel |
| 22:24.09 | tzafrir_laptop | wcfxo has not found the card. |
| 22:24.22 | heffernan | tzafrir_laptop: did you look at my extensions.conf? |
| 22:24.23 | tzafrir_laptop | Do you see any of the cards in lspci> |
| 22:24.35 | piper69 | THX2000: ok i got it to work |
| 22:24.42 | THX2000 | apt-get or svn? |
| 22:24.43 | *** join/#asterisk alt (n=alt@202.144.252.170) |
| 22:24.46 | alt | greetings. |
| 22:24.56 | alt | can anyone point me to a script to generate SIP calls from asterisk? |
| 22:24.58 | tzafrir_laptop | heffernan, couldn't figure anything useful from there. However, it may help to give a trace of a call |
| 22:25.03 | carl0s- | tzafrir_laptop: ok. I'll try another PCI slot. These cards are poor. They're Motorola. The kernel detects them as 'Wildcard X100P'. One of them came with a capacitor floating around in the anti-static bag! I had to resolder that one. Shanghai specials :D |
| 22:25.04 | alt | I need to load test a gateway |
| 22:25.15 | tzafrir_laptop | alt: call file or manager interface Originate |
| 22:25.17 | carl0s- | tzafrir_laptop: yeah, one sec. |
| 22:25.39 | Dr-Linux | file is away i think |
| 22:25.51 | heffernan | tzafrir_laptop: so, in theory, this should work. |
| 22:26.08 | piper69 | THX2000: both |
| 22:26.10 | heffernan | tzafrir_laptop: and by work, i mean that it will try local extensions before calling out using the outbound trunk. |
| 22:26.21 | THX2000 | piper69: ok, now do u have a /usr/src/asterisk folder? |
| 22:26.35 | THX2000 | piper69: if not, from what path did u run svn? |
| 22:26.49 | tzafrir_laptop | Dr-Linux, then it's time to interface to his manager? |
| 22:27.03 | piper69 | THX2000: now THX2000 i have asterisk there |
| 22:27.08 | carl0s- | tzafrir_laptop: lspci shows: 00:0c.0 Communication controller: Motorola Wildcard X100P |
| 22:27.28 | THX2000 | piper69: k, cd /usr/src/asterisk > make clean; make install |
| 22:27.31 | Dr-Linux | tzafrir_laptop: here is midnight :) |
| 22:28.01 | tzafrir_laptop | gee. It actually shows as "Motorola Wildcard X100P"? somebody bothred updating the pci cards database? |
| 22:28.32 | piper69 | THX2000: it's working :) |
| 22:28.41 | carl0s- | tzafrir_laptop: I guess so :) |
| 22:28.42 | THX2000 | piper69: tada |
| 22:29.11 | heffernan | tzafrir_laptop: ? |
| 22:29.20 | tzafrir_laptop | what? |
| 22:29.33 | piper69 | THX2000: so now just type astersik |
| 22:29.35 | heffernan | tzafrir_laptop: so, in theory, this should work. |
| 22:29.37 | carl0s- | is there a newer version of zaptel than 1.2.6 which maybe will recognise this card properly? |
| 22:29.38 | piper69 | to see if it's working |
| 22:29.43 | heffernan | carl0s-: no. |
| 22:29.51 | tzafrir_laptop | piper69, before you do that, apt-get build-dep asterisk |
| 22:30.01 | carl0s- | ok. Can I force insmod to try to associate the module with the specific pci dev/vendor ID of this card? |
| 22:30.02 | tzafrir_laptop | to make sure you have what it takes to build it |
| 22:30.16 | THX2000 | its already compiled |
| 22:30.19 | piper69 | tzafrir_laptop: and i'll do that from the root |
| 22:30.27 | THX2000 | yea, just run asterisk, and asterisk -r to view it |
| 22:30.44 | tzafrir_laptop | carl0s-, take a look at the kernel logs first |
| 22:30.55 | carl0s- | dmesg? |
| 22:31.01 | tzafrir_laptop | Maybe the driver identified the card but failed to load |
| 22:31.07 | tzafrir_laptop | dmesg will also do |
| 22:31.22 | piper69 | THX2000: E: Build-dependencies for asterisk could not be satisfied. |
| 22:31.22 | tzafrir_laptop | rmmod wcfxo; modprobe wcfxo |
| 22:31.37 | tzafrir_laptop | piper69, what's the problem? |
| 22:31.55 | carl0s- | tzafrir_laptop: here we go.. here's the problem during bootup: Failed to initailize DAA, giving up... |
| 22:31.55 | carl0s- | wcfxo: probe of 0000:00:0c.0 failed with error -5 |
| 22:32.06 | piper69 | tzafrir_laptop: apt-get builed-dep asterisk |
| 22:32.09 | carl0s- | hmm. that's a funny address isn't it? 0000:00:0c.0 ? |
| 22:32.19 | piper69 | tzafrir_laptop: returns "E: Build-dependencies for asterisk could not be satisfied." |
| 22:32.42 | THX2000 | what happens when u type asterisk and hit enter? |
| 22:32.55 | tzafrir_laptop | piper69, what is the error about, specifically? do you have deb-src sources for your 'deb' lines in sources.list? |
| 22:33.29 | piper69 | tzafrir_laptop: i don't know |
| 22:33.59 | tzafrir_laptop | carl0s-, I had this in one system and this was an issue with the interrupts controller. I added the boot prameter 'pci=noacpi' and the card worked |
| 22:34.25 | tzafrir_laptop | you need to mess a bit in the hardware/bios/whatever area |
| 22:34.45 | carl0s- | tzafrir_laptop: funny you say that.. I just googled and got : "his is a troubles if the card shares interrupt with something on MOtherBOard. configure the slot to other irq, or move the card to other irq.". I'll try pnp/acpi bios and boot options, and other slot. |
| 22:34.54 | tzafrir_laptop | piper69, if you don't know, check your /etc/pt/sources.list |
| 22:34.55 | carl0s- | thanks for the help. I know what to look for now which is good. |
| 22:35.30 | tzafrir_laptop | /etc/apt/sources.list , that is |
| 22:35.51 | carl0s- | the machine still has a 2003 bios.. I could start there I suppose. |
| 22:35.59 | CoffeeKid | i keep getting the warning: WARNING[25925]: channel.c:787 channel_find_locked: Avoided initial deadlock for '0x737650', 10 retries! ... anyone seen this before? |
| 22:36.29 | TripleFFFF | fuck asterisk |
| 22:36.31 | TripleFFFF | going openpbx |
| 22:36.33 | TripleFFFF | exit |
| 22:36.50 | tzafrir_laptop | "initial deadlock" is not a real problem, right? |
| 22:36.50 | carl0s- | lol |
| 22:36.56 | tzafrir_laptop | or am I wrong? |
| 22:37.07 | piper69 | Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k, Copyright (C) 1999-2004 Digium. |
| 22:37.07 | piper69 | Written by Mark Spencer <markster@digium.com> |
| 22:37.07 | piper69 | ========================================================================= |
| 22:37.07 | piper69 | Connected to Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k currently running on dell1 (pid = 25894) |
| 22:37.10 | piper69 | dell1*CLI> |
| 22:37.13 | tzafrir_laptop | ~pb |
| 22:37.15 | jbot | pb is probably a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
| 22:37.44 | CoffeeKid | tzafrir_laptop: well, that was my inital though.. but after so many, our server becomes very unstable. |
| 22:38.08 | CoffeeKid | tzafrir_laptop: its indicates an iminate crash.. in our server at least. |
| 22:38.15 | piper69 | THX2000: so it's working right |
| 22:38.32 | THX2000 | well yea...i dont know how u ended up w/ 1.0.7 though |
| 22:38.41 | tzafrir_laptop | piper69, that prompt was from the default Debian Sarge package |
| 22:39.07 | tzafrir_laptop | You want a 1.2.9.1 deb? |
| 22:39.17 | tzafrir_laptop | (for Sarge) |
| 22:40.02 | piper69 | tzafrir_laptop: how can i get that? |
| 22:40.36 | *** join/#asterisk chorlick (n=root@63.81.26.126) |
| 22:40.41 | tzafrir_laptop | I have my temporary repo with the misleading name 'deb http://rapid.tzafrir.org.il/rapid unstable main' |
| 22:41.21 | piper69 | troytop: so what are you requesting me to do |
| 22:42.31 | piper69 | tzafrir_laptop: so what r u requesting me to do |
| 22:43.06 | *** join/#asterisk knarfly (n=bdavis@c-69-180-98-189.hsd1.fl.comcast.net) |
| 22:43.08 | tzafrir_laptop | what do you want, exactly? |
| 22:43.18 | *** join/#asterisk saftsack (n=saftsack@p54A7E220.dip.t-dialin.net) |
| 22:43.22 | saftsack | good evening |
| 22:45.00 | piper69 | tzafrir_laptop: i have to leave now i'll logback in within 2 hrs |
| 22:45.16 | piper69 | tzafrir_laptop: THX2000 :thank you so much for all your help |
| 22:45.45 | CoffeeKid | why is it that everyone seems to know about the error i'm seeing, but no one can explain why its happening or how to fix it? |
| 22:46.02 | CoffeeKid | i keep getting the warning: WARNING[25925]: channel.c:787 channel_find_locked: Avoided initial deadlock for '0x737650', 10 retries! . |
| 22:46.25 | CoffeeKid | it seems to only happen on a higher load. |
| 22:46.31 | knarfly | CoffeeKid: Is your * system working? |
| 22:46.33 | CoffeeKid | 20+ calls |
| 22:46.40 | *** join/#asterisk RageMax (n=max@tw39.tele.pitt.edu) |
| 22:46.52 | CoffeeKid | knarfly: yes, but its unstable... and i'm pretty sure its because of these errors. |
| 22:47.28 | knarfly | CoffeeKid: Could be...but my system reports Warnings all the time too but it runs like a MF |
| 22:47.35 | tzafrir_laptop | CoffeeKid, if you'll look at the code there (channel.c, line 787) you'll see that this is an indication that some locking is used, but not of a real problem. |
| 22:47.51 | tzafrir_laptop | Now, this may be related to a separate issue |
| 22:48.05 | CoffeeKid | thats what i'm concerned about. |
| 22:48.20 | CoffeeKid | i know the error itself isn't a big deal, but it indicates something else is wrong... |
| 22:48.37 | knarfly | CoffeeKid: Are you running on Linux? |
| 22:48.45 | tzafrir_laptop | Or maybe it is just an indication that there is a load? |
| 22:48.59 | CoffeeKid | knarfly: yes, RHEL4 |
| 22:49.10 | CoffeeKid | dual proc xeon 3.0Ghz .. 4GB mem |
| 22:49.11 | tzafrir_laptop | and hence a certain thread has failed to gain a certain lock right away |
| 22:49.46 | tzafrir_laptop | CoffeeKid, I really haven't looked there, so take everything I say with a grain of salt |
| 22:49.46 | CoffeeKid | tzafrir_laptop: yes, but the same memory addresses keep coming up. |
| 22:49.50 | saftsack | grrr there is nobody on the sales sect @ digium |
| 22:50.07 | tzafrir_laptop | CoffeeKid, same lock? |
| 22:50.10 | knarfly | CoffeeKid: Wow...I see lots of post about stability but mainly with Linus users. I run FreeBSD and it's not been a problem. |
| 22:50.11 | CoffeeKid | same addresses, yes. |
| 22:50.40 | tzafrir_laptop | CoffeeKid, in what country are you? |
| 22:50.48 | CoffeeKid | knarfly: USA |
| 22:50.58 | CoffeeKid | er, tzafrir_laptop: USA |
| 22:51.10 | saftsack | benjk, evening |
| 22:51.21 | tzafrir_laptop | Funny. I see many posts from people in the USA with problems. I'm from Israel and I have no problems |
| 22:51.24 | tzafrir_laptop | ;-) |
| 22:51.31 | benjk | hi saftsack |
| 22:51.40 | CoffeeKid | hehe, i guess i'll just move the server to isreal :) |
| 22:51.57 | benjk | isreal? |
| 22:52.08 | CoffeeKid | but seriously, this is something that indicates an issue.. |
| 22:52.11 | heffernan | tzafrir_laptop: how is the situation down there now? |
| 22:52.12 | CoffeeKid | sorry, *israel |
| 22:52.13 | tzafrir_laptop | About the same distance as Japan |
| 22:52.15 | knarfly | CoffeeKid: Don't do that...someone will just blow it up! |
| 22:52.16 | saftsack | benjk, i phoned a master @ digium today over my budgetel and the sound quality was amazing compared with my isdn cards |
| 22:52.17 | benjk | I thought these days virtualisation was the big hit |
| 22:52.23 | benjk | should be isvirtual instead |
| 22:52.48 | CoffeeKid | ;) |
| 22:53.01 | tzafrir_laptop | heffernan, there are other types of problems there. actually today (yesterday). But this is no place for politics... |
| 22:53.11 | CoffeeKid | anyways, the error seems to happen more and more when there is more channels/calls being made.. |
| 22:53.24 | CoffeeKid | at the end of the night, even with no channels, we see TONS of these errors. |
| 22:53.31 | heffernan | oh well, going to bed. |
| 22:54.21 | CoffeeKid | any ideas on how to better diagnose the problem? |
| 22:54.34 | tzafrir_laptop | CoffeeKid, at the end of the night there are many calls? |
| 22:54.48 | CoffeeKid | no, at the end of the night, 1-2 calls max |
| 22:54.52 | CoffeeKid | but the locks still show |
| 22:54.55 | alt | I found a call sim. http://sipp.sourceforge.net/ |
| 22:55.10 | CoffeeKid | its seems to be compounding, or something. |
| 22:55.26 | tzafrir_laptop | name of debian package: sip-tester |
| 22:55.42 | *** join/#asterisk Egonis (n=Egonis@207.245.14.10) |
| 22:55.50 | Egonis | How do I playback sound files to Console/DSP? |
| 22:56.11 | saftsack | benjk, whats with your hfc card? still alive? |
| 22:56.28 | Egonis | when I call Console/DSP, and press 1 for instance, it says << Console Received digit 1 >> is there any way I can work with this? |
| 22:56.54 | *** join/#asterisk nettie (i=[U2FsdGV@85-18-54-38.ip.fastwebnet.it) |
| 22:57.22 | nettie | Hey guys, anyone have news regardign 1.4 release pls? |
| 22:57.27 | *** join/#asterisk Tall-guy (i=tall-guy@207-195-103-110.regn.hssx.sasknet.sk.ca) |
| 22:57.29 | Qwell[laptop] | nettie: "soon" |
| 22:57.36 | *** join/#asterisk SimoAmi (n=SimoAmi@207-38-248-221.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com) |
| 22:58.00 | SimoAmi | hi there |
| 22:58.10 | SimoAmi | my first time here |
| 22:58.46 | nettie | Qwell[laptop] this means trunk could be defined stable or it will just move to 1.4Alpha then beta then RC and so ? |
| 22:58.46 | nettie | so on |
| 22:58.46 | benjk | yeah, HFC is still rocking |
| 22:58.46 | Tall-guy | ok, semi-urgent request...I upgraded from a "VERY OLD" CVS to 1.2.9 this pm......and my zapata.conf file is causing asterisk to puke....error is "chan_zap.c 6890 mkintf unable to get parameters" |
| 22:58.56 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 22:59.44 | *** join/#asterisk MiniDevil (n=MiniDevi@71.224.106.188) |
| 23:00.10 | saftsack | humm :( mine not. the voice quality is still crap |
| 23:00.22 | MiniDevil | whats up people |
| 23:00.24 | SimoAmi | Can anyone provide some advice on what affordable service to use with Verizon. We would like to eliminate the multiple POTS lines and go digital |
| 23:00.28 | saftsack | does anyone know what digium uses? do they use * too? |
| 23:00.51 | knarfly | saftsack: Digium is * |
| 23:01.22 | knarfly | saftsack: unless like M$ they use FreeBSD for their mail servers |
| 23:01.55 | mog | nope |
| 23:01.58 | mog | we use asterisk |
| 23:02.16 | wunderkin | no they really use meridian |
| 23:02.17 | nettie | knarfly damn you discovered their secret!! they'll kill u now :) |
| 23:02.18 | nettie | eheheh |
| 23:02.34 | SS7 | meridian?! ha! we use Western Electric 1A2 ;) |
| 23:02.44 | knarfly | 8-) |
| 23:02.46 | mog | centrix SS7 |
| 23:02.52 | symlink | nope, cans with string |
| 23:03.03 | SS7 | yeah, the 1A2s come with centrex service |
| 23:03.28 | *** part/#asterisk troytop (n=troyt@remote.activestate.com) |
| 23:04.02 | *** join/#asterisk setti (n=setti@191-79-112.adsl.terra.cl) |
| 23:04.21 | SimoAmi | is anyone familiar with verizon lines? |
| 23:04.27 | SS7 | SimoAmi: sure |
| 23:04.34 | SS7 | SimoAmi: whats yer question |
| 23:04.40 | *** join/#asterisk bananapc (n=bdf@nvg1-01.yvr.sxip.com) |
| 23:04.42 | saftsack | knarfly, yes i know but maybe they use anything else ;) |
| 23:05.11 | SimoAmi | what service do I need from them to replace existing POTS lines? |
| 23:05.21 | SS7 | SimoAmi: POTS lines :) |
| 23:05.27 | knarfly | saftsack: couldn't say...but I would vebture the answer is no...they developed * |
| 23:05.28 | SS7 | what are you trying to do |
| 23:06.16 | RageMax | anyone at a university? |
| 23:06.22 | MiniDevil | Does anyone in here use the Xorcom Rapid bundled Asterisk? |
| 23:06.22 | SimoAmi | I'm trying to build an asterisk based pbx that reduces the phone bill of individual analog lines |
| 23:06.47 | SS7 | SimoAmi: well, you could cut off all the CLASS and custom calling feature stuff |
| 23:07.06 | SS7 | SimoAmi: but an analog line is an analog line, regardless of what CPE you have sitting on the end of it |
| 23:07.17 | SimoAmi | this is for a travel agency that receives and make long-distance calls |
| 23:07.30 | SimoAmi | but only inside the us |
| 23:07.36 | SS7 | SimoAmi: how many stations? |
| 23:07.37 | MiniDevil | SimoAmi: Why not just go with BroadVoice.com like I am? 24.95 unlimited world to 35 countries |
| 23:07.42 | *** part/#asterisk bananapc (n=bdf@nvg1-01.yvr.sxip.com) |
| 23:07.46 | SS7 | how many simultaneous calls? |
| 23:07.53 | *** part/#asterisk Egonis (n=Egonis@207.245.14.10) |
| 23:08.06 | SimoAmi | 3 simul with 4 stations |
| 23:08.32 | SimoAmi | these are accessed through a toll free and 2 local lines |
| 23:08.42 | *** join/#asterisk zerst (n=zerst@66.172.109.250.nwinternet.com) |
| 23:08.43 | SimoAmi | 2 local numbers sorry |
| 23:08.46 | SS7 | SimoAmi: you could probably get away with using an ITSP and a fast DSL connection |
| 23:09.15 | SimoAmi | can I make outgoing calls with DSL? |
| 23:09.23 | SS7 | SimoAmi: yes, hence the ITSP |
| 23:09.45 | SimoAmi | what's ITSP |
| 23:09.50 | SS7 | ~itsp |
| 23:09.51 | jbot | itsp is probably Internet Telephony Service Provider. An ITSP is a "VoIP Phone Company" |
| 23:10.16 | SimoAmi | ok, something similar to ISDN,T1? |
| 23:10.21 | SS7 | no |
| 23:10.29 | SS7 | ISDN and T1 are transport an signaling photocols |
| 23:10.31 | SS7 | er |
| 23:10.33 | SS7 | protocols |
| 23:10.52 | SS7 | an ITSP is a company |
| 23:11.10 | SimoAmi | ah, so |
| 23:11.24 | MiniDevil | Voice over IP runs through your current active internet connection. Using Fast DSL or Cable connection would allow you to use the internet, and have simultaneous telephone conversations through Voice over IP (VoIP) |
| 23:11.27 | SimoAmi | i should approach this company for their phone service? |
| 23:11.45 | SS7 | SimoAmi: you should do research or hire a consultant at this point |
| 23:11.50 | knarfly | SS7: or it could be a hack with a bunch of bandwidth in his garage? You can never be sure these days. |
| 23:11.56 | SS7 | knarfly: haha true |
| 23:12.47 | knarfly | give me a few * servers and a couple of t1's fract into a channel mixer and I'm my Ma Bell |
| 23:12.47 | SimoAmi | I want to do a research, but some tech advice is definitely helpful |
| 23:12.54 | SS7 | ~book |
| 23:12.55 | jbot | [book] a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 23:12.57 | SS7 | ~docs |
| 23:12.58 | jbot | hmm... docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 23:13.39 | SimoAmi | wow thanks guys |
| 23:14.12 | SS7 | So3kris: I'm also a consultant |
| 23:14.13 | SS7 | er |
| 23:14.14 | SS7 | SimoAmi: |
| 23:14.21 | SS7 | SimoAmi: where are you located? |
| 23:14.23 | SimoAmi | now I just need help with the specific stuff. the fxo thing |
| 23:14.28 | SimoAmi | new york |
| 23:14.31 | SS7 | SimoAmi: ah ok |
| 23:14.41 | SS7 | SimoAmi: fxo port connects to a phone line |
| 23:15.07 | SimoAmi | ok,one question... |
| 23:15.14 | SS7 | sure |
| 23:16.00 | knarfly | ~book |
| 23:16.01 | jbot | book is probably a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 23:16.07 | SimoAmi | I had the impression that I could get one digital line that handles multiple incoming/outgoing calls and that's more affordable than the analog lines |
| 23:16.22 | SS7 | SimoAmi: yes, but not for an installation as small as 3 lines |
| 23:16.23 | MiniDevil | You can |
| 23:16.41 | SS7 | SimoAmi: the cost savings come once you start replacing upwards of about eight analog lines |
| 23:16.45 | *** join/#asterisk d-tech (n=dtc@72.245.233.107) |
| 23:17.31 | SimoAmi | SS7: very interesting |
| 23:17.54 | SS7 | SimoAmi: because the digital line consists of 23 voice channels |
| 23:17.57 | SimoAmi | so you recommand leaving the analog lines and using FXO's to wire them |
| 23:18.11 | SS7 | SimoAmi: sure, or look into using an ITSP |
| 23:18.31 | MiniDevil | SS7: Can't you just get a VoIP provider, and have additional lines put in at 1.99 a month extra, using the PBX to deal with all that, saving a ton of money more? |
| 23:18.44 | MiniDevil | SS7: Or was I wrong with going with VoIP |
| 23:18.58 | knarfly | SimoAmi: What is the current internet connection at the location you're going to install *? |
| 23:19.02 | SS7 | MiniDevil: I'm still very iffy on running a business entirely on voip |
| 23:19.17 | SS7 | MiniDevil: the current selection of ITSPs are too flaky for my liking |
| 23:19.23 | SimoAmi | knarfly: RCN cable internet |
| 23:19.24 | knarfly | SS7: It is dangerous in mission critical situations. |
| 23:19.33 | SimoAmi | 5MB |
| 23:19.43 | SS7 | SimoAmi: what kind of uplink speed? |
| 23:19.50 | MiniDevil | Dangerous how so? |
| 23:20.13 | knarfly | SimoAmi: Then you should try some experimenting like I did with the ISTP's they work fine and can let you know what your getting before you blow a bundle |
| 23:20.21 | SimoAmi | I don't know that at this moment |
| 23:20.32 | SS7 | SimoAmi: your uplink speed is important to know |
| 23:20.42 | knarfly | MiniDevil: Ever been in a storm and the lights flicker or go out? |
| 23:21.01 | SS7 | SimoAmi: if you're sharing your internet connection with voice and data, you will also need to take traffic shaping and queueing into account |
| 23:21.18 | SimoAmi | it's similar to my internet, let me test my upload speed |
| 23:21.20 | knarfly | MiniDevil: The analog phone from Bell South keep working while all the other appliances are off until the power comes back on. |
| 23:21.24 | MiniDevil | Indeed, however I run a Cyberpower 1500 UPS... between 60-120 minutes battery backup time |
| 23:21.49 | *** join/#asterisk xbmodder_newlapp (i=nobody@atarack/staff/xbmodder) |
| 23:21.52 | knarfly | SimoAmi: I run cable and can easilt handle 10 calls without a problem |
| 23:22.20 | xbmodder_newlapp | Would anyone be interesting in a VPS with low-cost sip termination/origination, and ztdummy support? |
| 23:22.28 | knarfly | MiniDevil: In Miami a hurrican can knock out the power for days...something I worry about here in paradise |
| 23:22.49 | MiniDevil | ahhh yes, knarfly. I can see your point. Generator time then. |
| 23:23.15 | MiniDevil | Luckily I dont have that up here. And I figure if its that bad, it can take out the landlines too so not much you can do there. |
| 23:23.56 | SimoAmi | wow 10 calls would make the client so happy |
| 23:24.03 | *** join/#asterisk ben_d (n=ben@cpe-66-67-129-4.rochester.res.rr.com) |
| 23:24.10 | *** join/#asterisk imdabest (n=imdabest@202.147.186.58) |
| 23:24.18 | imdabest | hello room |
| 23:24.30 | MiniDevil | SimoAmi: Will you look in PM's? I wish to send you a link |
| 23:24.39 | SS7 | room? what is this, AOL in 1997? |
| 23:24.50 | MiniDevil | you mean AOHell. |
| 23:25.16 | SimoAmi | ok, will |
| 23:25.18 | knarfly | Moire like AO-Stay the Hell Away from |
| 23:26.04 | imdabest | hey can any one help me with local extension |
| 23:26.05 | MiniDevil | Indeed, knarfly. |
| 23:26.25 | SS7 | imdabest: ask your question and someone should answer it |
| 23:27.14 | imdabest | i have made local extension which works fine if i use IAX2 but when i use it with SIP it doesnt |
| 23:27.44 | knarfly | imdabest: so lets see it... |
| 23:27.45 | SimoAmi | sorry for the delay |
| 23:27.57 | SimoAmi | here's the test results |
| 23:28.02 | SimoAmi | Your download speed : 837 kbps or 104.7 KB/sec. |
| 23:28.02 | SimoAmi | Your upload speed : 359 kbps or 44.9 KB/sec. |
| 23:28.05 | SS7 | imdabest: are they both coming into the same context? |
| 23:28.15 | imdabest | yeah |
| 23:28.46 | imdabest | if i use SIP in extension.conf it dosent work but if i change it to IAX it start working |
| 23:28.51 | SS7 | SimoAmi: using ulaw companding, you should be able to handle four concurrent calls |
| 23:28.52 | *** join/#asterisk postel_ (n=jp@unaffiliated/postel) |
| 23:28.55 | knarfly | MiniDevil: What was that link, I'd like to test it. |
| 23:29.16 | SS7 | imdabest: use pastebin and show us your sip.conf, iax.conf, and extensions.conf |
| 23:29.20 | SS7 | ~pb |
| 23:29.21 | jbot | rumour has it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
| 23:29.39 | lucaferr | Is anyone using an ata188 or ata186 device with asterisk? I need some help with upgrading the firmware to support SIP.. |
| 23:29.47 | imdabest | i am new here could you tell me how to paste in pastebin thing? |
| 23:29.58 | SS7 | imdabest: oh hell, just go to pastebin.ca |
| 23:30.20 | SimoAmi | four simultaneous is just great even 3 is |
| 23:30.26 | knarfly | imdabest: he's right just click on a link and it will tell you what to do |
| 23:30.27 | *** join/#asterisk anthm (n=anthm@000-442-412.area4.spcsdns.net) |
| 23:30.27 | *** mode/#asterisk [+o anthm] by ChanServ |
| 23:30.37 | lucaferr | is anyone willing to help me obtain the latest firmware image from cisco.. I don't have a cco account.. :( |
| 23:30.39 | SS7 | SimoAmi: this also assumes you arent using it for data |
| 23:31.27 | SimoAmi | I still think I should go for asterisk because it has features like call waiting "Currently assisting other customers...please wait" |
| 23:31.28 | lucaferr | it's just ridiculous in my opinion that they require a contract for firmware updates.. |
| 23:31.33 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 23:31.33 | *** mode/#asterisk [+o denon] by ChanServ |
| 23:31.53 | SimoAmi | with the current system it just keeps ringing |
| 23:31.56 | zerst | Anyone ever hook up a Merlin Legend to an Asterisk box using digium hardware? |
| 23:32.16 | SS7 | zerst: yow |
| 23:32.21 | knarfly | Merlin - Bwhahahahaha |
| 23:32.28 | zerst | yeah, sad |
| 23:32.51 | SS7 | id set up a merlin as a museum piece :) |
| 23:32.53 | knarfly | I'm sure merlin is a about as propritary as it gets... |
| 23:32.56 | *** join/#asterisk skraelings001 (n=skraelin@201.230.239.132) |
| 23:33.06 | zerst | but I have then hooked up.... just getting HDLC Abort (6) errors across the span |
| 23:33.19 | zerst | dropped in a new TE110P card and same thing |
| 23:33.22 | nettie | lucaferr Juniper does the same.. they have ppl working to code/test updates .. this cost money, it's normal they charge |
| 23:33.22 | SS7 | I had a Merlin 410 that I gave to a friend a year ago |
| 23:33.53 | *** join/#asterisk queso (n=josh@nr13-216-68-211-77.fuse.net) |
| 23:33.56 | SS7 | zerst: you got the framing and coding set up correctly? |
| 23:34.18 | zerst | I thunked so... |
| 23:34.25 | imdabest | done |
| 23:34.28 | zerst | esf and b8zs |
| 23:34.36 | zerst | running the card as the clock source |
| 23:34.42 | zerst | set the merlin to the read from the span. |
| 23:34.49 | wunderkin | zerst, are you getting them constantly? |
| 23:34.51 | SS7 | imdabest: uh...ok...so what are the URLS? |
| 23:35.00 | zerst | like 4 per second |
| 23:35.21 | queso | I need help if someone has a second... I setup my first IAX2 extension and I got loudhush (a macosx iax2 program) logged in, but no matter what number it says "Phone 200 is currently unavailable" (200 is the extension I setup) |
| 23:35.21 | wunderkin | i had that happen too, i had to force the slot to 33mhz or something like that |
| 23:35.28 | imdabest | http://pastebin.ca/86524 |
| 23:35.44 | zerst | wunderkin: on the lucent brick? |
| 23:35.53 | wunderkin | ... no on your pc |
| 23:36.54 | wunderkin | as long as everything else is setup right |
| 23:37.02 | imdabest | is it enough or you want me to put whole sip,iax,extensions.confs there? |
| 23:37.15 | SS7 | imdabest: pastebin console output for the failed call |
| 23:37.52 | imdabest | i didnt faild dials and just shows calling |
| 23:37.55 | imdabest | hang on |
| 23:38.02 | SS7 | what? |
| 23:38.18 | SS7 | that wins the "unintelligible sentence of the afternoon" award |
| 23:38.39 | imdabest | ? |
| 23:38.40 | imdabest | me |
| 23:38.44 | SS7 | yes you |
| 23:38.52 | *** join/#asterisk zerst (n=zerst@66.172.109.250.nwinternet.com) |
| 23:38.53 | imdabest | :P |
| 23:38.56 | SS7 | what is "i didnt faild dials and just shows calling" supposed to mean? |
| 23:39.21 | *** join/#asterisk AsteVen (n=Asterisk@201.243.154.210) |
| 23:39.23 | AsteVen | Hi |
| 23:39.49 | skraelings001 | i'm trying to make a prompt when one dial and it doesn't match none ? but so far, when i dial a non-matchable number it connects and got silence. ? i'd like something like exten => i,n,Playback() ??? any idea |
| 23:40.27 | SS7 | ok, i take that back, imdabest. skraelings001 wins the award by a nose |
| 23:40.46 | zerst | :0 |
| 23:42.14 | AsteVen | Greetings, first time on the channel, sorry for asking before giving but i hope I can help somebody soon. I have a problem registern an IAX client, it says Restricting Registration for peer 'clientname' to 60 seconds. Im using IAX as an alternative to sip (Im in a private network using Eyebeam where Asterisk has a public ip) |
| 23:42.20 | [TK]D-Fender | skraelings001 : Pastebin your entire defective context |
| 23:42.25 | skraelings001 | SS7 : sorry i'm not a native-english speaker |
| 23:42.52 | zerst | Thats ok, neither are the enlish! |
| 23:42.53 | skraelings001 | SS7: i didn't explain well myself |
| 23:44.05 | [TK]D-Fender | AsteVen : Ok, that means the the server being registered to forces frequent reregistrations to maintain the link. |
| 23:44.53 | imdabest | http://pastebin.ca/86531 |
| 23:45.15 | skraelings001 | [TK]D-Fender : it's not part of any context yet. i was wondering how i could do such thing, i pretend to announce users when dial a wrong number or they don't have privileges to make some calls. |
| 23:45.21 | *** join/#asterisk bjohnson_ (n=bjohnson@i216-58-51-153.cybersurf.com) |
| 23:45.32 | mitcheloc | rofl, teliax has a forum and it's filled with posts about anal sex and cheap drugs and supermegagirls! yay! |
| 23:46.24 | AsteVen | The server is forcing frequent registrations, do I disable that then? |
| 23:46.30 | imdabest | no phone rings but if i change it to IAX2 in extension.conf it works smoothly |
| 23:47.22 | [TK]D-Fender | skraelings001 : yes it is if you are trying it and its not working... |
| 23:48.39 | [TK]D-Fender | imdabest :your dial is not formated right. You don't need the IP in there. Thats what registering is for. |
| 23:49.23 | [TK]D-Fender | imdabest : should be in the form of : Dial(TECH/peername,timeout,options) |
| 23:49.35 | AsteVen | Do you guys have problem using Eyebeam from a pvt Network with a public ip asterisk server¿? |
| 23:49.48 | [TK]D-Fender | imdabest : the peer's you configured are set as dynamic and * will know the IP when it registers |
| 23:50.06 | [TK]D-Fender | AsteVen : nope. Works fine |
| 23:50.44 | imdabest | ok |
| 23:51.04 | imdabest | let me check i am coming in few mins |
| 23:51.13 | knarfly | AsteVen: I'd be careful...whose * server on public network are you using...? |
| 23:51.26 | AsteVen | I solved the IAX2 problem using register => username:password@serveraddress and it works fine the IAX2 |
| 23:52.31 | AsteVen | My Eyebeam get stuck in Registering, Eyebeam ip is private, ASterisk is a public one |
| 23:54.32 | knarfly | AsteVen: the * server will need certain ports open...I assume since it's public it's firewalled |
| 23:54.41 | *** part/#asterisk MonkeyHugs (n=MonkeyHu@63.149.122.93) |
| 23:55.12 | skraelings001 | [TK]D-Fender: i thought i could use "i" extension, but didn't work out |
| 23:55.19 | knarfly | AsteVen: and you need a few tweaks in the X-Lite setting...hold on I know a good link on this |
| 23:56.00 | *** part/#asterisk ingOmar (n=chatzill@remote.activestate.com) |
| 23:56.35 | AsteVen | cool I tried using the STUN |
| 23:58.09 | *** join/#asterisk brut- (n=brut@66.173.4.254) |
| 23:58.27 | knarfly | AsteVen: http://forums.digium.com/viewtopic.php?t=7854&highlight=xlite |
| 23:58.45 | *** join/#asterisk stubert (i=stu@techtools.actusa.net) |
| 23:58.48 | AsteVen | Ty |
| 23:59.16 | [TK]D-Fender | skraelings001 : Depends from where. In an IVR you can use "i". From SIP/IAX you can't |
| 23:59.53 | knarfly | http://forums.digium.com/viewtopic.php?t=7851&highlight=xlite |