| 00:01.15 | vpp | but i can push my glosly over priced cards on people but act like i'm fighting the machine |
| 00:01.15 | vpp | lol |
| 00:01.15 | shepherd | so what's your problem? |
| 00:01.15 | shepherd | over priced? |
| 00:01.30 | shepherd | omg.. have you seen what others charge for a pbx solution? |
| 00:01.30 | vpp | but u miss the point |
| 00:01.30 | shepherd | go research.. you're full of crap |
| 00:01.30 | vpp | others dont say oh look its free |
| 00:01.30 | shepherd | asterisk is free :) |
| 00:01.31 | shepherd | in fact! |
| 00:01.31 | vpp | we're killing ourselves to keep it free |
| 00:01.31 | vpp | yes it is |
| 00:01.31 | shepherd | you can use it 100% free |
| 00:01.31 | shepherd | with softphones, etc :) |
| 00:01.37 | shepherd | so what's your point? |
| 00:01.41 | vpp | but digium make their money off it by pushing over priced hardware |
| 00:01.48 | Qwell | over priced? |
| 00:01.49 | shepherd | now.. if you want something more advanced.. then you need an adapter, etc |
| 00:01.59 | Qwell | looked at ANY other PBXs lately? |
| 00:02.03 | shepherd | and don't expect digium to just give you that free as well |
| 00:02.04 | gbdrbob | of course you and make money on open source software! Publishing code under the GPL doesn't commit you to communinism1 |
| 00:02.13 | vpp | wait a second |
| 00:02.13 | jebba | vpp, uh, show me /anything/ that does what digium hardware does for less... troll.... |
| 00:02.13 | vpp | asterisk is free |
| 00:02.19 | vpp | ok |
| 00:02.27 | vpp | so pull that out the equation |
| 00:02.35 | shepherd | and you're full of crap.. |
| 00:02.35 | vpp | so now we have digium cards |
| 00:02.37 | shepherd | move on |
| 00:02.37 | shepherd | :) |
| 00:02.40 | shepherd | NEXT! |
| 00:02.41 | vpp | ok never mind |
| 00:03.05 | gbdrbob | you don't have to buy them! You've already said you could make cheaper fxo modules - go do that and sell them. I'm sure there would be a market |
| 00:03.29 | shepherd | yup.. that's the whole point of asterisk |
| 00:03.41 | vpp | my point is simply this.. digium + asterisk preach how they are so great because they are cheaper/free compared to the rest |
| 00:03.49 | Qwell | they are |
| 00:03.57 | vpp | except they take the free asterisk + add their overpriced cards = < than the fully comercial |
| 00:04.05 | jebba | vpp, uh, show me /anything/ that does what digium hardware does for less... troll.... |
| 00:04.09 | shepherd | wtf, how so? |
| 00:04.29 | jebba | in fact, got a URL? I want one :) |
| 00:04.39 | shepherd | yes.. find us a cheaper solution please |
| 00:04.41 | Qwell | I'd LOVE to see you get an avaya PBX (with hardware support) for anywhere NEAR the price of a * setup |
| 00:04.43 | vpp | the point is how can u say oh everyone else charges u over the odd |
| 00:04.47 | shepherd | or cisco |
| 00:04.52 | vpp | but still charge over the odd yourself? |
| 00:04.57 | vpp | less over the odds but still way over |
| 00:05.20 | Qwell | go away troll, nobody cares what you have to say |
| 00:05.26 | vpp | and tied to free asterisk it pushes your product because overall its cheaper |
| 00:05.34 | gbdrbob | * doesn't need a pstn interface to work either - you can go fuly if and rent your pstn termination from one of the many providers out there. |
| 00:05.39 | hypa7ia | vpp do you know anything about the open source dev model, and the various business models around it? |
| 00:05.44 | vpp | yes |
| 00:06.32 | Juggie | has anyone tried yak voip? |
| 00:06.41 | Deryl | why do you guys tolerate this penis drip? |
| 00:06.49 | Juggie | they use the xten sip phone so i'm thinking its sip |
| 00:07.03 | gbdrbob | vpp: we get back to supply and demand - things are priced according to what the market can bear in this business - digium still dramatically undercut the competition |
| 00:07.06 | coldfeet | guys anyone here worked with realtime and asterisk, I just installed the cvs-head and have setup my tables etc etc...but on looking at extconfig file, it doesnt say where I can put in IP address or user/pass details to connect to my db, just asks for db name |
| 00:07.14 | hypa7ia | vpp: asterisk cards are around $1500 for a t1 card. a basic cisco router with callmanager express is like $20000. ever heard of the term "order of magnitude"? |
| 00:07.32 | vpp | and sangoma is 350$ for a t1 card |
| 00:07.40 | vpp | yet they are rubished |
| 00:07.49 | vpp | i'm not her promoting them |
| 00:07.50 | Juggie | a digium t1 card is not 1500 |
| 00:07.52 | shepherd | $1500 for a quad t1 |
| 00:07.54 | vpp | just to be clear |
| 00:08.00 | Juggie | a quad is yes |
| 00:08.03 | Juggie | not a single span |
| 00:08.10 | vpp | digium single T1 is about 600$ |
| 00:08.27 | vpp | which is rediculous |
| 00:08.27 | shepherd | and where can you get a sangoma card for $350? |
| 00:08.38 | Qwell | 600/24=25 |
| 00:08.39 | Deryl | now you're changing the story? are digium cards 1500 or 600? |
| 00:08.41 | vpp | b2b |
| 00:08.43 | Deryl | you can't flip flop in the middle |
| 00:08.43 | vpp | easily |
| 00:08.47 | Qwell | TWENTY-FIVE DOLLARS PER PORT |
| 00:08.49 | Qwell | piss off |
| 00:08.56 | Qwell | You're so fucking wrong, and stupid to boot. |
| 00:09.00 | shepherd | Deryl: both |
| 00:09.07 | shepherd | Deryl: $600 for a single t1 |
| 00:09.11 | coldfeet | seems like a chilled group :-) |
| 00:09.12 | shepherd | $1500 for 4 t1 |
| 00:09.21 | hypa7ia | coldfeet: normally is :-/ |
| 00:09.28 | vpp | 589$ for a single port digium t1 last i checked |
| 00:09.35 | vpp | but i dunno i dont buy their stuff |
| 00:09.40 | Qwell | vpp: Do you have ANY idea how much we pay at my work for an avaya PBX *EACH MONTH*? |
| 00:09.49 | Deryl | shepherd: but he said a sangoma SINGLE port T1 card was 350 so therefore that means he's comparing their single card as well (digium's) |
| 00:09.52 | rabelais | and 550 for sangoma a101, what's the difference? |
| 00:09.53 | vpp | Qwell: probably too much |
| 00:09.54 | Mavvie | Deryl: I have a digium card here for $125 |
| 00:10.08 | Deryl | no one is THAT dumb to compare a quad port with a single port and expect it to be the same price |
| 00:10.14 | shepherd | Mavvie: t1 card? |
| 00:10.19 | vpp | a101 does dchannel in hardware to start |
| 00:10.22 | vpp | also its 350$ b2b |
| 00:10.28 | Deryl | so I'm wondering how much more he's going to troll before he finally stops adjusting to your argument |
| 00:10.31 | vpp | digium laugh at u if u ask for any qty price |
| 00:10.39 | Mavvie | it's pretty useless saying "this <brand> name card costs so and so much" and then getting upset about it because you didn't specify properly which type of card it was. |
| 00:10.39 | vpp | tell u to get in line to buy 1 off price |
| 00:11.34 | vpp | lol oh forget it |
| 00:11.36 | *** part/#asterisk vpp (~noone@host-83-146-50-131.bulldogdsl.com) |
| 00:11.47 | Deryl | vpp: did you fax digium your business license to prove that you ARE in fact a b2b and not some kid with a bit of money wanting the discount? |
| 00:11.52 | Deryl | hehe |
| 00:12.54 | *** join/#asterisk szw2001 (~vip@218.1.218.187) |
| 00:14.39 | *** part/#asterisk coldfeet (~c@213.78.240.109) |
| 00:15.35 | rabelais | I'm fed up with broadvoice and want to switch over to someone else, but am debating whether I should keep my number, does anyone know what happens at the routing level if I do a number transfer? will a dialed call go to broadvoice's pbx/routers and _then_ be transferred over to the transferred provider, or will it go straight to my new provider, never seeing broadvoice's system ever...cause if it still has to go to broadv |
| 00:15.35 | rabelais | oice, doesn't that mean I may still face the same downtime as before? |
| 00:15.41 | darwin35 | whats up with aserisk being limited to 10 #include lines |
| 00:15.53 | *** join/#asterisk mitcheloc (~mitcheloc@lgb-cust-208.57.66.122.mpowercom.net) |
| 00:16.48 | mitcheloc | anyone here have experience with the dialtones being lost via zaptel channel? i.e. i call into the menu in asterisk and i dial 106, about 1out of 2 times it'll think i dialed 16 instead of 106 |
| 00:17.33 | MikeJ[Laptop] | dtmf being lost, not dialtones |
| 00:17.42 | mitcheloc | right |
| 00:17.57 | mitcheloc | sorry, yea i confused the two |
| 00:18.51 | *** join/#asterisk |Vulture| (~V@199.227.253.212) |
| 00:19.29 | *** join/#asterisk Turulo (~weed@80.26.237.112) |
| 00:19.39 | mitcheloc | anyone on that? it's pretty important, and driving me crazy |
| 00:19.44 | mitcheloc | maybe "relaxdtmf"? |
| 00:19.45 | |Vulture| | how do you exclude a directory in a tar, say I wanted to "tar -cf /mnt/backup/backup.tar /" but didn't want it to backup /mnt/backup |
| 00:20.23 | mtgh | man tar |
| 00:20.31 | mtgh | search for exclude |
| 00:20.39 | |Vulture| | yea but its so much easier just to ask from someone who knows |
| 00:21.04 | rabelais | that would be an instant ban in my book if I was an op |
| 00:21.04 | mitcheloc | i already had that enabled though |
| 00:21.32 | Silik0n | any iLand customers here? |
| 00:21.39 | |Vulture| | hahaha I answer questions in here all the time cut me some slack... so I didn't want to look up -X |
| 00:21.57 | mitcheloc | good point |
| 00:22.04 | mitcheloc | share the love ;) |
| 00:22.14 | |Vulture| | ;) |
| 00:22.23 | Silik0n | more like share the HATE! |
| 00:22.56 | |Vulture| | nah I only share hate when the conversation changes to AMP or *@Home |
| 00:23.02 | Silik0n | hah |
| 00:23.23 | mitcheloc | why do you hate those? |
| 00:23.53 | file | it all goes down hill from there |
| 00:23.57 | mitcheloc | please share, i was thinking of making a similar product |
| 00:24.02 | |Vulture| | it takes the learning curve and just pisses all over it |
| 00:24.03 | mitcheloc | i'd like to know the downfalls of those |
| 00:24.21 | |Vulture| | the problem is that it makes * easy... but hard at the same time |
| 00:24.25 | |Vulture| | it takes away from learning of a DP |
| 00:24.47 | |Vulture| | the DP in it is only variables, and for someone learning *... good luck |
| 00:24.58 | *** join/#asterisk lattice (~lattice@zz193144.cipherkey.net) |
| 00:25.01 | Silik0n | mitcheloc: heres the problem those are tailored for a very specific use... like the name states "at home" and when it comes down to it people usually want something thats not in there so you have to figure out exactly how to mod your system to work with it |
| 00:25.16 | *** join/#asterisk newsmafia (~newsmafia@wsip-68-15-19-142.sd.sd.cox.net) |
| 00:25.19 | |Vulture| | and its limited... for a home install.. works great, but for anyone wanted to have it work well... config it yourself |
| 00:25.23 | mitcheloc | but seriously, do you want to spend 2 hours configuring a server everytime you get a new install? it takes frigging forever! |
| 00:25.28 | mitcheloc | same commands...over and over |
| 00:25.47 | jets | mitcheloc: that's why i have a shell script to apply my patches and wanted libraries and compile everything |
| 00:25.49 | mitcheloc | true, i see your points |
| 00:25.50 | |Vulture| | mitcheloc: well I have a DP that I use over and over but its mine I know how to change every aspect of it |
| 00:25.51 | jets | and nfs copy my default configs |
| 00:25.55 | mitcheloc | heh i bought www.asteriskathome.com lol |
| 00:25.58 | Silik0n | mitcheloc: dood... if you are going to be building a ton of them do 1 dialplan you will use on all of them and then work from that |
| 00:26.04 | |Vulture| | making your DPs are what makes you a true * user |
| 00:26.17 | |Vulture| | Silik0n: lol were saying the same thing |
| 00:26.26 | mitcheloc | dp = ? |
| 00:26.31 | |Vulture| | dialplan |
| 00:26.37 | Silik0n | dp == double penetration? |
| 00:26.43 | x9net | lol |
| 00:26.46 | mitcheloc | heh, oh yea...dial plan is easy part |
| 00:26.49 | jets | loves my double penetration |
| 00:26.50 | |Vulture| | DVDA! |
| 00:27.03 | mitcheloc | i'm talking about the tedious linux install/upgrade/installing mysql/apache/asterisk, etc everything you need |
| 00:27.03 | Silik0n | hah |
| 00:27.05 | |Vulture| | quad penetration :P |
| 00:27.17 | |Vulture| | lol |
| 00:27.18 | *** join/#asterisk redhate (~redhate@200.233.68.41) |
| 00:27.26 | mitcheloc | theres only 2-3 holes, depending on who your looking at heh |
| 00:27.28 | jets | mitcheloc: then image a box. |
| 00:27.40 | file | ooh penetration |
| 00:27.43 | Silik0n | mitcheloc: dood thats where things like ${FAVORITE_DISTRO} and scripted installs come in to play |
| 00:27.51 | mitcheloc | true |
| 00:27.53 | |Vulture| | mitcheloc: thats what I do image... but these new SATA drives don't let me |
| 00:27.57 | mitcheloc | i just wrote my script today lol |
| 00:28.01 | |Vulture| | so I am gunna try this tar backup/restore |
| 00:28.14 | mitcheloc | but yea..i see |
| 00:28.18 | mitcheloc | anyone put firewalls on their boxes? |
| 00:28.22 | mitcheloc | to only allow ssh + voip ports? |
| 00:28.22 | |Vulture| | I do |
| 00:28.25 | Silik0n | |Vulture| : dood... google around for ghost replacements there are some that are quite good |
| 00:28.30 | |Vulture| | block almost everything |
| 00:28.35 | x9net | i got a hardware firewall runing |
| 00:28.41 | mitcheloc | yea, i've got one i was thinking of putting on it |
| 00:28.42 | Silik0n | mitcheloc: only idiots leave SQL servers open to the public ;) |
| 00:28.52 | jets | mitcheloc: ya I use a pretty retentive IPtables script and change my ssh port plus use port knocking for the ssh port |
| 00:28.55 | |Vulture| | Silik0n: I tried a few all-in-ones for linux but none worked for my SATA even though it said it would (DELL SATA mind you) |
| 00:29.06 | mitcheloc | only idiots right software that comes with default blank passwords, as part of the config script it should make you change it! |
| 00:29.19 | Silik0n | |Vulture| we use tons of dell boxen... we have one that works just fine (with 3ware cards even) |
| 00:29.37 | mitcheloc | 400sc = the perfect asterisk hardware |
| 00:29.53 | |Vulture| | I use a lot of 400SC but they replaced the 400SC with SC420 |
| 00:29.56 | |Vulture| | thats what i use now |
| 00:29.58 | |Vulture| | they are SATA |
| 00:30.06 | *** part/#asterisk redhate (~redhate@200.233.68.41) |
| 00:30.20 | Jason357 | hi |
| 00:31.44 | jets | hi. |
| 00:32.48 | file | stroke it baby! |
| 00:32.52 | file | make it shoot sparks! |
| 00:33.15 | |Vulture| | file: thats not the only thing it shoops :O |
| 00:33.17 | |Vulture| | shoots |
| 00:33.21 | |Vulture| | DAMNIT I messed that up |
| 00:33.27 | *** join/#asterisk Pete_Largo (~Pete_Larg@adsl-65-71-225-121.dsl.rcsntx.swbell.net) |
| 00:34.11 | file | yeah you did |
| 00:35.06 | Silik0n | smokin! |
| 00:35.25 | *** join/#asterisk newbien (~e@147.241.33.65.cfl.res.rr.com) |
| 00:37.23 | jets | hrm |
| 00:41.31 | jets | Gorgeous! |
| 00:43.18 | *** join/#asterisk DFT (~dft@CPE0040050149d6-CM00080d77ae83.cpe.net.cable.rogers.com) |
| 00:43.35 | DFT | evenin |
| 00:46.15 | *** join/#asterisk elric (~kavit@ppp114-10.static.internode.on.net) |
| 00:47.12 | *** join/#asterisk remmo (~rem@smack.isp.net.au) |
| 00:47.15 | *** join/#asterisk Pete_Largo (~Pete_Larg@adsl-65-71-225-121.dsl.rcsntx.swbell.net) |
| 00:47.54 | mitcheloc | is chan_alsa deprecated? |
| 00:47.59 | mitcheloc | i can't find info on it anywhere |
| 00:48.20 | mitcheloc | any how tos on it or something |
| 00:51.15 | mitcheloc | damn, what happened to that info? it must have been wiped off of voip-info |
| 01:15.11 | *** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc) |
| 01:15.11 | *** topic/#asterisk is Asterisk: The Open Source PBX || Astricon Europe -- Madrid, Spain -- June 15-17 || PBX Developers Conference in Chicago Aug 3-5 Register Now! http://www.asterlink.com/cluecon.htm |
| 01:16.01 | DFT | sorry dude, just trying to get my asterisk started |
| 01:18.42 | *** join/#asterisk TheEmperor (~TheEmpero@203.114.48.47) |
| 01:18.52 | TheEmperor | can someone tell me is this is correct? |
| 01:18.54 | TheEmperor | exten => s,1,Dial(IAX2/${ARG1},60,tT) |
| 01:21.32 | bjohnson | TheEmperor: yes |
| 01:21.39 | jets | on a queue timeout where does the call go? |
| 01:21.45 | jets | t,1? |
| 01:21.55 | *** join/#asterisk scubasteve (~steve@nc-67-77-54-135.dyn.sprint-hsd.net) |
| 01:21.55 | jets | or +1 |
| 01:22.03 | TheEmperor | bjohnson: but when i press #, the call does not get transferred? |
| 01:22.06 | bjohnson | TheEmperor: assuming ${ARG1} contains the right info including the number to dial |
| 01:22.21 | bjohnson | TheEmperor: then I guess it doesn't include the right info |
| 01:22.32 | TheEmperor | bjohnson: yes, when i call another iax extension it works it's just the transfer that doesn't work.. |
| 01:22.34 | bjohnson | jets: I don't know |
| 01:22.58 | mmlj4 | any issues with * running on AMD 64bit? |
| 01:23.04 | scubasteve | Can someone spare a few minutes to help out with what I think is an RTP/NAT problem? Took a 7960 from home, am camping and using wireless hooked up.. My * network is 192.168.1.x and my network here at the campsite is the same. I have NAT=yes in sip.conf but it looks like RTP is still going to the 192.168.1.x network, never to hit the internet on my * side. |
| 01:23.12 | bjohnson | mmlj4: I think bkw_ does that |
| 01:23.24 | mmlj4 | thanks |
| 01:24.38 | scubasteve | Anyone? |
| 01:25.46 | file | we don't do it *yet* |
| 01:25.49 | file | but it works fine on 64-bit |
| 01:26.27 | mmlj4 | for what values of "we"? |
| 01:26.34 | mitcheloc | has anyone heard anything about intercom documentation for asterisk (chan_alsa/oss)? |
| 01:26.36 | file | the company bkw_ and I work for |
| 01:26.40 | mmlj4 | ah, ok |
| 01:27.02 | file | if David is in a good mood, we should have a testing rig up for 64-bit in a day |
| 01:27.15 | *** join/#asterisk techie (gus@antibala.com) |
| 01:27.35 | file | mmm dual opteron box |
| 01:27.49 | mmlj4 | i'm actually looking at using a P4 (oh, i hate intel), due to AMD_64 nervousness.... if you guys think it's stable, I'm willing to give it a shot |
| 01:27.51 | mitcheloc | is say opteron 2.2 faster then xeon 3.2? |
| 01:28.02 | file | I have a 3.4GHz P4 with 64-bit ;) |
| 01:28.05 | mmlj4 | mitcheloc: by a mile |
| 01:28.07 | file | that's why I said it works fine |
| 01:28.15 | mitcheloc | mmlj4: oh really? |
| 01:28.27 | mitcheloc | heh i just bought a new server (1u/ibm/3.2ghz xeon/1gb ram) |
| 01:28.29 | mmlj4 | check the benchmarks sites |
| 01:28.39 | mitcheloc | but i was eyeing those opterons...i know nothin about them though |
| 01:28.40 | DFT | <PROTECTED> |
| 01:28.42 | DFT | here = 0, tmp->channel = 2, channel = 2 |
| 01:29.04 | DFT | is this telling me that I should have chan 2 specified as chan 0 in zaptel.conf? |
| 01:29.34 | mmlj4 | file: do you have any digium cards in that box? I'm sure SIP works fine, but the hardware is what bothers me |
| 01:30.28 | file | mmlj4: no, but I remember reading mailing list posts about them working fine |
| 01:30.35 | mmlj4 | fair enough |
| 01:30.37 | file | cause someone asked about 64-bit |
| 01:33.55 | TheEmperor | i still can't get call transfer to work pressing # .. |
| 01:33.58 | Silik0n | anyone here use iLand for colo? |
| 01:34.21 | TheEmperor | can someone please have a look to see what I did wrong? http://pastebin.ca/11845 |
| 01:34.23 | TheEmperor | thanks :) |
| 01:35.45 | DFT | nm, last message, zaptel.conf didn't like ; comment delimiters |
| 01:38.09 | *** join/#asterisk shmaltz (~chatzilla@ool-43551098.dyn.optonline.net) |
| 01:45.18 | bjohnson | TheEmperor: you've watched the cli to see if the dial command used is the one you think is being used? |
| 01:45.39 | bjohnson | TheEmperor: ie .. you don't have a _. or something grabbing the call? |
| 01:46.21 | *** join/#asterisk Deryl (pgpkeys@static-141-149-128-140.buff.east.verizon.net) |
| 01:46.29 | TheEmperor | bjohnson:a ? |
| 01:46.54 | TheEmperor | bjohnson: i've gone and changed the whole thing to exten => 2006,1,Dial(IAX2/2006,60,tT) |
| 01:46.59 | TheEmperor | still doesn't work :( |
| 01:47.31 | TheEmperor | also, in my iax.conf under [general] there is dtmfmode=rfc2883 |
| 01:48.55 | bjohnson | watch the cli |
| 01:49.02 | bjohnson | during a call |
| 01:49.10 | bjohnson | see what the system sees |
| 01:50.26 | mitcheloc | anyone here have an ibm laptop? |
| 01:50.56 | bjohnson | TheEmperor: you're pressing the exten to be transferred to as well right? not just '#' |
| 01:51.13 | bjohnson | mitcheloc: yes. but a coworker is using it |
| 01:51.25 | TheEmperor | bjohnson: all i need is to press # right? or did i do something wrong... |
| 01:51.36 | mitcheloc | i'm just bummed that the keys show so much wear and tear, my dell i had before for 2 years didn't show it |
| 01:51.40 | bjohnson | what exten are you trying to transfer to? |
| 01:51.50 | TheEmperor | bjohnson:http://pastebin.ca/11846 |
| 01:52.03 | TheEmperor | bjohnson:i use 2007 to call 2009 then transfer the call to 2010 |
| 01:52.04 | PatrickDK | hmm, I love my ibm laptop |
| 01:52.13 | bjohnson | mitcheloc: stop eating pancakes with maple syrup and using irc before washing hands |
| 01:52.16 | mitcheloc | me too, but don't you notice that? |
| 01:52.23 | PatrickDK | hmm, not really |
| 01:52.31 | bjohnson | TheEmperor: so you would type #2010 |
| 01:52.37 | PatrickDK | the dell my wife has, the keys keep breaking |
| 01:52.47 | mitcheloc | i don't do that =p, i take good care of it (2.0/1gb ram/80gb/fingerprint reader/bluetooth/abg) hehe |
| 01:52.53 | mitcheloc | i got it for $2200 ;) |
| 01:52.59 | bjohnson | I've got Dells, IBMs, and Compaqs .. all about the same |
| 01:53.04 | mitcheloc | (take home, with taxes included) |
| 01:53.13 | mitcheloc | ibms got the best kb |
| 01:53.14 | TheEmperor | bjohnson: tried doing #2010 doesn't work :( |
| 01:53.18 | PatrickDK | I love the ibm little finger mouse pointer |
| 01:53.20 | bjohnson | I would never buy Sony or HP (unless it was a hell of a deal) |
| 01:53.26 | tzanger | PatrickDK: the wiggle dick? |
| 01:53.29 | tzanger | I HATE those things |
| 01:53.29 | PatrickDK | ya |
| 01:53.33 | tzanger | gimme a trackpad anyday |
| 01:53.34 | mitcheloc | yea i've been learning to use it |
| 01:53.35 | PatrickDK | I hate touchpads |
| 01:53.44 | mitcheloc | i only used touchpads before my ibm laptop |
| 01:53.50 | mitcheloc | but they did something write in implementing it |
| 01:53.50 | PatrickDK | trackballs are my fav though |
| 01:54.12 | bjohnson | I hate touchpads AND pointer sticks. Mouse kicks them. |
| 01:54.15 | shmaltz | anybody here read the article in 2600 about cisco ip phone security? |
| 01:54.20 | TheEmperor | bjohnson: any ideas? |
| 01:54.23 | bjohnson | TheEmperor: don't know then |
| 01:54.28 | bjohnson | TheEmperor: watch the cli |
| 01:54.32 | mmlj4 | shmaltz: URL? |
| 01:55.22 | TheEmperor | bjohnson: yeah, nothing happens when i press # and all.. |
| 01:58.39 | mitcheloc | shmaltz: hook it up |
| 01:59.28 | *** join/#asterisk Moc[Toronto] (~mochouina@142.131.66.185) |
| 01:59.34 | shmaltz | no url |
| 01:59.37 | Moc[Toronto] | Hi all |
| 01:59.39 | shmaltz | you got to buy it |
| 01:59.46 | mitcheloc | bah |
| 01:59.47 | mitcheloc | ok |
| 01:59.48 | shmaltz | will see if I can find anything |
| 02:00.20 | DFT | 2600 used to be online |
| 02:00.50 | DFT | okay, so I've got asterisk running vanilla out of the box..wtf do I do know:) |
| 02:00.55 | MikeJ[Laptop] | hey moc |
| 02:02.01 | DFT | please excuse the noobness, I have no telephony background |
| 02:02.38 | shmaltz | it goes on to tell you how insecure tftp and dhcp is |
| 02:02.44 | shmaltz | with the plain text passwords |
| 02:03.20 | shmaltz | and how to sniff the mac out, download the sipxxxxxxx.cnf files |
| 02:03.37 | shmaltz | and upload your own by connecting to the telnet server on the cisco phone |
| 02:03.46 | Moc[Toronto] | hi mike |
| 02:03.54 | shmaltz | the passsword can be accessed using the tftp files |
| 02:04.11 | shmaltz | anybody that ever configured a cisco phone thru dhcp/tftp knows about this |
| 02:05.19 | TheEmperor | bjohnson:weird problem, on softphones it works but it doesn't work on hard phones, the call transer i mean |
| 02:07.04 | *** join/#asterisk jskcr|lappy (~jskcr@jskcr.user) |
| 02:08.33 | *** join/#asterisk dev2005 (~dev2003@222.33.36.205) |
| 02:09.54 | *** join/#asterisk kks (~kks@202.73.10.8) |
| 02:13.48 | TheEmperor | bjohnson:i don't think my hard phones are sending the right tones? |
| 02:16.01 | *** join/#asterisk lunchbox08 (~geoff@64.128.43.66) |
| 02:16.41 | *** part/#asterisk lunchbox08 (~geoff@64.128.43.66) |
| 02:17.19 | shido6 | :( |
| 02:17.32 | shido6 | yes, shmaltz |
| 02:19.22 | shmaltz | shido6, yep |
| 02:20.12 | *** join/#asterisk PBXtech (nik@63-226-102-79.slkc.qwest.net) |
| 02:22.22 | *** part/#asterisk szw2001 (~vip@218.1.218.187) |
| 02:23.25 | *** join/#asterisk BrentM (~brentmcgh@S0106000d6130c27e.ok.shawcable.net) |
| 02:23.26 | *** join/#asterisk syslod (~yurplsl@65.114.15.71) |
| 02:23.37 | BrentM | hello? |
| 02:24.15 | BrentM | im having lag issues with the asterisk anyone have any ideas |
| 02:25.03 | syslod | Hello. |
| 02:25.32 | *** join/#asterisk gte (~dylan@c-24-9-153-107.hsd1.co.comcast.net) |
| 02:25.53 | *** join/#asterisk shido6 (~shido@d57-87-253.home.cgocable.net) |
| 02:28.05 | *** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc) |
| 02:28.06 | *** topic/#asterisk is Asterisk: The Open Source PBX || Astricon Europe -- Madrid, Spain -- June 15-17 || PBX Developers Conference in Chicago Aug 3-5 Register Now! http://www.asterlink.com/cluecon.htm |
| 02:29.34 | *** part/#asterisk DFT (~dft@CPE0040050149d6-CM00080d77ae83.cpe.net.cable.rogers.com) |
| 02:32.20 | syslod | BrentM: What ur issues? |
| 02:32.41 | *** join/#asterisk orion88 (~orion@ool-4357e17a.dyn.optonline.net) |
| 02:33.42 | *** join/#asterisk pimpwell (~pimpwell@ool-44c6ab45.dyn.optonline.net) |
| 02:33.56 | pimpwell | anyone from New York (NYC/Westchester) Here? |
| 02:34.21 | dev2005 | who is from CHINA? |
| 02:34.22 | MikeJ[Laptop] | no, we don't let newyorkers in here, sorry |
| 02:34.27 | MikeJ[Laptop] | mao |
| 02:34.42 | pimpwell | anyone from Zimbabwe here |
| 02:34.54 | file | who here actually knows who I am? |
| 02:35.01 | MikeJ[Laptop] | ummmm |
| 02:35.02 | MikeJ[Laptop] | ME! |
| 02:35.11 | MikeJ[Laptop] | I know |
| 02:35.19 | pimpwell | you're rick james |
| 02:35.19 | MikeJ[Laptop] | your file! |
| 02:35.23 | MikeJ[Laptop] | mf |
| 02:35.28 | file | pimpwell: you lose |
| 02:35.45 | MikeJ[Laptop] | sucker |
| 02:35.57 | pimpwell | f^ck this game |
| 02:36.00 | pimpwell | Im not playing anymore |
| 02:36.19 | MikeJ[Laptop] | wah...crybaby :D |
| 02:36.30 | file | now now you two, play nice |
| 02:36.34 | pimpwell | Im watching superman |
| 02:36.41 | MikeJ[Laptop] | oh.. btw ^=u |
| 02:37.06 | MikeJ[Laptop] | ain't so super anymore is he.. |
| 02:37.18 | MikeJ[Laptop] | wow, did I jsut say that.... |
| 02:37.21 | pimpwell | ya |
| 02:37.26 | MikeJ[Laptop] | ouch... |
| 02:37.34 | pimpwell | you're going to hell |
| 02:37.45 | pimpwell | in a wheelchair |
| 02:38.10 | MikeJ[Laptop] | don't beleive in hell |
| 02:38.31 | MikeJ[Laptop] | if it does exist, all the cool people are down there anyway |
| 02:38.36 | file | spanish flea! |
| 02:38.43 | pimpwell | spanish fly? |
| 02:38.51 | file | no, flea |
| 02:38.59 | file | you lose again |
| 02:39.01 | pimpwell | I have a website to make |
| 02:39.08 | MikeJ[Laptop] | dude, if you have little bugs down there, they probably arn't fleas |
| 02:39.27 | BrentM | syslod: it is really laggy using eyebeam. it jitters and laggs but if we use x lite we have no problems with incomming calls. eyebeam seems to lag |
| 02:39.28 | file | :( |
| 02:40.11 | MikeJ[Laptop] | hehe |
| 02:40.31 | darwin35 | grrrrr |
| 02:40.44 | darwin35 | ok this system is pissing me offf |
| 02:40.52 | darwin35 | the inbound all works |
| 02:40.55 | MikeJ[Laptop] | better than pissing on you |
| 02:40.59 | darwin35 | the outbound is now |
| 02:41.12 | MikeJ[Laptop] | great... all set then |
| 02:41.15 | MikeJ[Laptop] | :D |
| 02:41.35 | darwin35 | no |
| 02:41.49 | darwin35 | I got 10 ciscos to load up |
| 02:42.03 | MikeJ[Laptop] | have fun |
| 02:42.05 | darwin35 | I can call them but they cant call the real world |
| 02:42.22 | MikeJ[Laptop] | can they call each other? |
| 02:42.27 | darwin35 | yes |
| 02:42.28 | file | the roof, the roof, the roof is on fire |
| 02:42.36 | *** join/#asterisk implicit (~implicit@ip68-7-149-247.sd.sd.cox.net) |
| 02:43.06 | MikeJ[Laptop] | ok, fix your dialplan |
| 02:43.11 | MikeJ[Laptop] | we don't need no water |
| 02:44.15 | gte | darwin35, check if your 'from-sip' context includes the pstn context that you're trying to use. |
| 02:46.04 | MikeJ[Laptop] | what if he does not call it from-sip |
| 02:47.41 | joaovianna | Hi guys! Anyone using astcc ? I installed in my * box, but I can't put it working... The problem is, the web-admin is working but it does not create my database (mysql). If I create manually my database, it seens that my perl cant open it. But I don't have any message. Only "Database unavailable -- please check configuration". There are any log that I can see whats happening ? |
| 02:47.42 | gte | MikeJ[Laptop], An assumption (10 Cisco's) on my part. Replace that with whatever SIP/Skinny context the phones are in. |
| 02:52.08 | BrentM | Can Anyone Help My Present Situation: it is really laggy using eyebeam. it jitters and laggs but if we use x lite we have no problems with incomming calls. eyebeam seems to lag |
| 02:52.19 | gte | Does anybody know if updating MWI via QSIG is possible with *? I'm trying to light up message waiting indicators on the remaining Nortel phones of an Asterisk/Meridian integration, using Comedian. |
| 02:53.47 | MikeJ[Laptop] | gte you can do it by dialing a set of codes to the meridian, it does not require qsig |
| 02:53.58 | MikeJ[Laptop] | I don;t have the codes handy atm |
| 02:54.24 | darwin35 | http://pastebin.ca/11856 |
| 02:54.45 | tzanger | MikeJ[Laptop]: *1[exten to ringback] |
| 02:55.17 | tzanger | er sorry |
| 02:55.25 | tzanger | *1[exten to indicate msg for] |
| 02:55.42 | darwin35 | ? |
| 02:55.51 | darwin35 | thats for mike |
| 02:55.52 | tzanger | *1243 will cause MWI on 243 to light up, and the "call" or "msg" softbutton will cause the phone to ring the extension that dialled *1243 |
| 02:57.05 | *** join/#asterisk rene- (~root@200.78.176.114) |
| 02:57.14 | rene- | hey |
| 02:57.18 | tzanger | that is only through an ATA though |
| 02:58.08 | Juggie | gte, i would say yes.... it should be no problem. |
| 02:58.13 | rene- | i wonder how does one goes to connect a v35 interface channel bank to a digium t1 interface? |
| 02:58.27 | Juggie | with an ATA, and a few scripts |
| 02:58.43 | *** join/#asterisk Kumbang (~ecvs@167.205.24.4) |
| 02:58.47 | gte | MikeJ[Laptop]: You wouldn't happen to have those codes? |
| 02:58.59 | joaovianna | Anyone using astcc ? |
| 02:59.37 | *** join/#asterisk Nukemizer (~Nuke@66.237.85.58.ptr.us.xo.net) |
| 03:02.34 | *** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc) |
| 03:02.34 | *** topic/#asterisk is Asterisk: The Open Source PBX || Astricon Europe -- Madrid, Spain -- June 15-17 || PBX Developers Conference in Chicago Aug 3-5 Register Now! http://www.asterlink.com/cluecon.htm |
| 03:03.02 | Kumbang | can * register to one sip proxy with more than one sip account? |
| 03:03.12 | *** join/#asterisk Himeko (~himeko@S01060040ca128fc3.ed.shawcable.net) |
| 03:03.27 | Juggie | Kumbang, can you put more then one register line in sip.conf? |
| 03:03.28 | Nukemizer | can anyone help me create a dial plan that will allow dialing *82 when using my fxs and fxo cards ? |
| 03:04.02 | Juggie | nuke, put *82,1,.... in your dialplan? |
| 03:04.27 | Juggie | exten=> *82,1,... |
| 03:04.35 | Nukemizer | as an extension ? |
| 03:04.45 | Nukemizer | ahh ok.. perfect will try that |
| 03:05.04 | Juggie | yes |
| 03:05.14 | Juggie | in the context which handles calls for the interface you want to put that function on |
| 03:05.27 | Juggie | on zap there are alot of built in * codes |
| 03:06.17 | Nukemizer | Juggie, thank you. you are right many * codes. testing now :) |
| 03:10.38 | SuPrSluG | joaovianna: u have to make tables in mysql that aren't there yet. check the nufone.pl script and add any missing tables |
| 03:14.32 | *** join/#asterisk forrestc{hm} (~forrestc@iMach.com) |
| 03:14.42 | forrestc{hm} | Hello |
| 03:15.21 | *** part/#asterisk forrestc{hm} (~forrestc@iMach.com) |
| 03:15.36 | *** join/#asterisk Legend (~legend@24.244.142.133) |
| 03:16.27 | SuPrSluG | joaovianna: u need: brands; cards;cdrs;iaxfriends;routes;sipfriends;trunks |
| 03:16.59 | rene- | what is the cheapest to connect a digium T1 card to a v35 channel bank? |
| 03:21.09 | joaovianna | SuPrSluG: Thanks, I checking ... I create manually my astcc database. |
| 03:21.52 | SuPrSluG | joaovianna: u need: just add the missing ones |
| 03:23.43 | SuPrSluG | joaovianna: don't forget to restart mysql |
| 03:23.48 | *** part/#asterisk rene- (~root@200.78.176.114) |
| 03:23.59 | joaovianna | SuPrSluG: nufone.pl ? Where I can get this script ? |
| 03:25.00 | SuPrSluG | joaovianna: sorry. just look at the 2nd reply. those are the tables u need |
| 03:25.07 | SuPrSluG | joaovianna: u need: brands; cards;cdrs;iaxfriends;routes;sipfriends;trunks |
| 03:25.37 | SuPrSluG | joaovianna: or it fails w/ the message u posted |
| 03:26.04 | *** join/#asterisk coppice (~chatzilla@155.199.17.210.dyn.pacific.net.hk) |
| 03:26.46 | *** join/#asterisk dev2005 (~dev2003@222.33.36.205) |
| 03:26.52 | *** join/#asterisk TheEmperor (~TheEmpero@203.114.48.47) |
| 03:27.04 | TheEmperor | does anyone know where i can find the dtmf tones set out in firefly? |
| 03:27.20 | joaovianna | SuPrSluG: Thanks. I create manually those tables. My cgi seens not working with my mysql. |
| 03:28.29 | *** join/#asterisk aminorex (~tony@c66.191.69.132.dul.mn.charter.com) |
| 03:28.58 | joaovianna | SuPrSluG: Database unavailable -- please check configuration. How can I check if my perl is open the tables ? There are any log ? |
| 03:29.22 | SuPrSluG | joaovianna: did u restart mysql? |
| 03:30.09 | mmlj4 | can someone please try connecting to my box? sip:87@sip.joeykelly.net |
| 03:30.24 | joaovianna | SuPrSluG: yes. How can I check if my perl is working with mysql ? |
| 03:30.53 | SuPrSluG | joaovianna: not sure |
| 03:31.20 | dev2005 | who has made his own Tormenta 2 PCI Card? |
| 03:31.23 | dev2005 | thanks a lot |
| 03:31.33 | SuPrSluG | joaovianna: when u go to the web page. does it display? |
| 03:33.18 | joaovianna | Yes, It read my configuration from ".conf", but when it tries to create or update data I get message "Database unavailable -- please check configuration" |
| 03:33.49 | coppice | dev2005: digium? govarian? :-) |
| 03:35.46 | Nukemizer | I am having trouble locating what is causing *8 to return a fast busy from my dail plan structure. Is there a feature or blocked function that would prevent me from creating a *82 dial plan ? |
| 03:35.57 | SuPrSluG | joaovianna: what do u have in /var/lib/astcc/astcc-config.conf? |
| 03:37.03 | SuPrSluG | joaovianna: all look correct? |
| 03:38.45 | *** join/#asterisk Kaos76k (~chatzilla@CPE-24-163-140-163.new.res.rr.com) |
| 03:38.49 | marlowe | <PROTECTED> |
| 03:39.08 | Kaos76k | Is anyone familiar with AMP and enum trunks? |
| 03:41.06 | Kaos76k | Anyone here? |
| 03:41.19 | Juggie | amp sucks |
| 03:41.28 | Juggie | it is the epiphany of lazyness |
| 03:41.31 | joaovianna | SuPrSluG: I past the configuration file for you. |
| 03:41.35 | Juggie | learn to write your own dialplan |
| 03:41.55 | Kaos76k | Well Juggie... thanks for the help. |
| 03:42.18 | Juggie | you'll thank me later when you discover amp is a pain |
| 03:42.21 | Juggie | and you cant do what you want |
| 03:42.51 | Kaos76k | So far the only thing I have had a problem with is setting up an enum trunk. |
| 03:43.15 | Kaos76k | Goodnight. |
| 03:43.20 | *** part/#asterisk Kaos76k (~chatzilla@CPE-24-163-140-163.new.res.rr.com) |
| 03:44.20 | SuPrSluG | joaovianna: sounds like an issue of cgi not having permission to read the DB. |
| 03:45.09 | joaovianna | SuPrSluG: Yes, I'm trying to get a simple perl cgi to test my connection ... Thanks. |
| 03:45.36 | SuPrSluG | joaovianna: is the owner of astcc DB mysql or root? |
| 03:47.44 | joaovianna | SuPrSluG: I'm using root as user. |
| 03:49.40 | *** join/#asterisk drumkilla (~russell@user-24-214-77-225.knology.net) |
| 03:49.40 | *** mode/#asterisk [+o drumkilla] by ChanServ |
| 03:49.40 | *** join/#asterisk TonyAlmeida (~tonyalmei@61.33.161.6) |
| 03:55.47 | SuPrSluG | joaovianna: is the /var/www/cgi-bin/astcc-admin.cgi owner also root |
| 03:56.26 | *** join/#asterisk newmember (user@S010600d0b76b1f36.cg.shawcable.net) |
| 04:01.15 | joaovianna | SuPrSluG: Yes, all perms... |
| 04:01.18 | mmlj4 | can someone please try connecting to my box? sip:87@sip.joeykelly.net |
| 04:01.42 | *** join/#asterisk Newbie___ (~me@218.111.224.205) |
| 04:01.50 | Newbie___ | hi all |
| 04:03.07 | Newbie___ | if i successfully register using X-Ten, can i do the same thing with * ? |
| 04:03.28 | Newbie___ | i mean register with a provider |
| 04:04.32 | Sato1 | you mean, register your Xten with asterisk? |
| 04:04.33 | SuPrSluG | joaovianna: go to the web page and try only filling in the first 5 items leave the rest blank. |
| 04:05.01 | Newbie___ | Sato1: i meant register Xten with another provider |
| 04:05.05 | joaovianna | SuPrSluG: Lets do it... |
| 04:05.15 | Newbie___ | and register * with the same provider |
| 04:05.17 | SuPrSluG | joaovianna:just use the save button and not create db |
| 04:05.23 | Sato1 | Newbie, depends on the provider, if they has sip port |
| 04:05.49 | Sato1 | if the provider has also IAX port, then you can use your asterisk instead of your xten |
| 04:06.30 | Newbie___ | they gave me a SIP server domain, i assume is sip registeration |
| 04:06.59 | Sato1 | ask them if they have IAX enabled |
| 04:07.17 | Sato1 | otherwise, you can connect your * using SIP protocol, see the sip.conf |
| 04:07.24 | SuPrSluG | joaovianna:also can u log into the db at the cmd line? |
| 04:07.45 | Newbie___ | i did use sip.conf to connect , keep getting unable to register |
| 04:08.02 | Newbie___ | they also gave me, SIP server domain and Server IP address |
| 04:08.13 | Newbie___ | which fit in nicely in xten and i can make call |
| 04:08.36 | Sato1 | you can specify domain=something before "register" line |
| 04:08.42 | Newbie___ | but in *, i know host=server ip, but SIP domain ? |
| 04:10.01 | joaovianna | SuPrSluG: OK, My configuration was not saved in the config file. |
| 04:10.23 | Sato1 | Newbie___: http://www.voip-info.org/wiki-Asterisk+config+sip.conf |
| 04:10.40 | Newbie___ | i just did domain=xxx before register line, * return as registeration timeout |
| 04:10.42 | SuPrSluG | joaovianna:got to be a permissions problem |
| 04:10.55 | Newbie___ | Sato1: thanks, i read them |
| 04:12.10 | joaovianna | SuPrSluG: Sorry, my changes were saved... |
| 04:12.47 | SuPrSluG | joaovianna:can u get to the brands on the web page now? |
| 04:12.57 | joaovianna | SuPrSluG: I can open my database and tables from phpmyadmin, but I don't know about Perl. |
| 04:13.36 | SuPrSluG | joaovianna:i don't think the issue is perl. |
| 04:14.52 | joaovianna | SuPrSluG: When I go to brands I have database problem... I'm pasting you the web link... |
| 04:16.26 | *** join/#asterisk t0p (t0p@tech-mgr.chatri.com) |
| 04:17.09 | t0p | G' Morning |
| 04:19.30 | *** join/#asterisk neowillis (~Willis@211.144.49.185) |
| 04:21.01 | SuPrSluG | joaovianna: i think u need to add grant privledges to mysql. do u have3 webmin? |
| 04:22.13 | *** join/#asterisk TheEmperor (~TheEmpero@203.114.48.47) |
| 04:22.40 | joaovianna | SuPrSluG: Yes, my root has privitedges to all accounts. |
| 04:28.52 | *** join/#asterisk neowillis (~neowillis@211.144.49.185) |
| 04:28.58 | SuPrSluG | joaovianna: /etc/init.d/mysql restart |
| 04:30.18 | SuPrSluG | joaovianna: it says your mysql up 16 days u must restart mysql daemon |
| 04:30.20 | joaovianna | SuPrSluG: restarted ! |
| 04:31.19 | t0p | joaovianna: what's your problem? |
| 04:33.23 | joaovianna | t0p: astcc is not working... It seems perl is not open the database right. |
| 04:33.56 | *** join/#asterisk Nukemizer (~Nuke@66.237.85.58.ptr.us.xo.net) |
| 04:33.59 | t0p | joaovianna: are you upgrading from 3.x to 4.x or something? |
| 04:34.54 | joaovianna | t0p: I'm using fedora (core 3), asterisk (1.0.7), perl (5.8.6). |
| 04:37.00 | t0p | joaovianna: I am on exactly the same environment |
| 04:37.23 | t0p | joaovianna: and had a problem with mysql |
| 04:37.27 | SuPrSluG | joaovianna: do u have perl DBI installed |
| 04:37.37 | joaovianna | t0p: I had a lot of problems putting perl running. |
| 04:38.07 | joaovianna | SuPrSluG: Yes, DBI is isntalled. |
| 04:38.23 | t0p | joaovianna: I found about the issue from some websites recommending an upgrade to 4.x |
| 04:38.23 | joaovianna | SuPrSluG: Yes, DBI,DBD:mysql |
| 04:38.54 | joaovianna | t0p: What problem with mysql ? |
| 04:39.12 | t0p | joaovianna: what's your exact problem? what are the errors? |
| 04:39.46 | joaovianna | t0p: perl seems not connect to my mysql database. |
| 04:40.04 | t0p | joaovianna: from *? |
| 04:40.16 | *** join/#asterisk newsmafia (~newsmafia@wsip-68-15-19-142.sd.sd.cox.net) |
| 04:40.28 | t0p | joaovianna: are you using realtime? |
| 04:41.14 | joaovianna | t0p: I think my perl is the problem. What is realtime ? |
| 04:44.52 | t0p | joaovianna: https://bugzilla.redhat.com/bugzilla/show_bug.cgi?id=141062 |
| 04:45.33 | t0p | joaovianna: have a look at that one and see if it's your case |
| 04:46.13 | joaovianna | t0p: I checked, but my mysql is working fine. I have other program in php using without problem. |
| 04:46.27 | joaovianna | SuPrSluG: What do you think ? |
| 04:47.56 | Nukemizer | Is there a way to make * dial a phone number once to Zap channels are bridged ? using fxs/fxo ports |
| 04:47.57 | SuPrSluG | joaovianna: seeing those messages in the top left corner. i think u may be right. i may be a mysql/perl conflict |
| 04:48.14 | *** join/#asterisk NewSole (dave@i216-58-44-245.avalonworks.net) |
| 04:48.25 | *** join/#asterisk NewSole (dave@i216-58-44-245.avalonworks.net) |
| 04:48.29 | SuPrSluG | joaovianna: those should not be there |
| 04:49.00 | joaovianna | SuPrSluG: Those messages are some flags that I put to show the value of variables database and user, etc... |
| 04:50.19 | NewSole | hello |
| 04:55.29 | SuPrSluG | joaovianna: do an ls -l /var/lib/mysql/astcc |
| 05:00.44 | *** join/#asterisk joaovianna (naturalvoi@node-40247a6a.ewr.onnet.us.uu.net) |
| 05:04.04 | joaovianna | SuPrSluG: I think my perl is not working... |
| 05:05.32 | *** join/#asterisk santiago (~santiago@63.245.86.248) |
| 05:05.40 | *** join/#asterisk x9net (9xnet@dsl017-096-140.lax1.dsl.speakeasy.net) |
| 05:07.47 | *** join/#asterisk Moc[Toronto] (~mochouina@142.131.66.185) |
| 05:08.04 | Moc[Toronto] | damn night.. |
| 05:08.22 | NewSole | lol |
| 05:08.36 | NewSole | hot dam... lol |
| 05:09.54 | NewSole | what up Moc |
| 05:10.12 | NewSole | What are u doing in TO |
| 05:10.17 | *** part/#asterisk newsmafia (~newsmafia@wsip-68-15-19-142.sd.sd.cox.net) |
| 05:13.30 | Moc[Toronto] | work, gota redo an intranet in 1 week |
| 05:13.55 | Moc[Toronto] | over 20gig of intranet split in 4 different region to put into 1 page |
| 05:13.59 | Moc[Toronto] | 1 site I mean |
| 05:14.34 | NewSole | ack |
| 05:14.38 | NewSole | sounds fun |
| 05:15.41 | NewSole | I am messing around tring to get website up.... |
| 05:15.41 | x9net | lol |
| 05:16.07 | x9net | haha |
| 05:16.31 | x9net | i should be doing more asp stuff |
| 05:16.38 | Moc[Toronto] | I hate doing ASP .... |
| 05:16.46 | NewSole | hey it looks good sofar |
| 05:16.51 | Moc[Toronto] | it feel alot better to do PHP |
| 05:17.28 | x9net | i know asp better but i have 2 linux servers and one windows box and only linux is setup at the datacenter rigt now :( |
| 05:17.32 | x9net | so its php for now |
| 05:17.59 | t0p | coppice: are you there? |
| 05:18.14 | NewSole | got most of the shell done... tomorrow is DATA.... |
| 05:18.14 | coppice | no |
| 05:18.36 | t0p | coppice: :-) Sorry to disturb you again |
| 05:19.26 | t0p | coppice: is your library designed to initiate the "Answer Back" signal also? |
| 05:19.28 | Moc[Toronto] | gota sleep, waking up in 6.5 hours |
| 05:19.36 | Moc[Toronto] | 5.5 hours I mean |
| 05:19.49 | NewSole | lol same here.... |
| 05:19.50 | x9net | have fun |
| 05:20.02 | coppice | t0p: I am not sure what yuo mean |
| 05:20.10 | NewSole | got to have most of the data up by 5pm tomorrow before meeting |
| 05:20.34 | t0p | coppice: it's a pulse of about 120 ms |
| 05:20.52 | coppice | t0p: to do what? |
| 05:20.55 | t0p | coppice: use it to indicate charging process |
| 05:21.44 | coppice | t0p: oh, you mean charge pulses. i haven't implemented that, because I am not sure what to do with them. |
| 05:22.24 | t0p | coppice: I mean after a free anouncement prompt, there is usually a pulse to initiate taxation |
| 05:23.15 | coppice | t0p: probably, but I am not sure how it is handled. I have never worked with a version of R2 which does this |
| 05:24.59 | coppice | t0p: so what happens? the call is treated as a free call, and then after the pulse charging begins? |
| 05:25.07 | *** join/#asterisk JamesDotCom (~james@sweep.bur.st) |
| 05:27.35 | t0p | coppice: Offer->Accept->Connect, I usually send a pulse of 120 ms before I do GC Connect with dialogic |
| 05:28.35 | t0p | coppice: after the pulse I and telco start to count on how long the caller will stay online |
| 05:29.04 | coppice | t0p: please don't use offensive words like dialogic in public forums :-) |
| 05:29.18 | t0p | coppice: Sorry :-( |
| 05:30.03 | coppice | t0p: their R2 is really bad. it crashes when anything isn't exactly as expected |
| 05:30.26 | t0p | coppice: i see |
| 05:31.14 | t0p | coppice: I really appreciate your library, and may have a chance to use it one day |
| 05:31.56 | t0p | coppice: but don't know how long R2 will survive here |
| 05:32.45 | coppice | R2 is still strong in a huge number of countries. I wrote the library with south america and china in mind. however, I have people using it all over the place now |
| 05:34.38 | Himeko | hehe http://www.sparkfun.com/shop/index.php?shop=1&itemid=416& |
| 05:35.03 | t0p | coppice: how long have you made them? I see from the name it's like "pre15", is it still in a beta version? |
| 05:36.35 | t0p | coppice: some people here changed to ss7 as they say R2 is slow |
| 05:36.53 | coppice | t0p: some of the variants are in heavy use and some are just getting their first users. so, some are well tested and some are not. the pre15 thing is because I keep doing things to debug areas new people are trying. The first users have been using it for about 9 months |
| 05:37.14 | coppice | R2 is slow, clumsy, lacks features, and should have died 30 years ago. |
| 05:38.24 | t0p | coppice: but it still stays fresh :-) |
| 05:38.51 | coppice | its like zombie that won't lie down in its grave |
| 05:41.52 | coppice | you'd be amazed how many clumsy old obsolete protocols are still in heavy use |
| 05:42.41 | t0p | coppice: I don't know about in other countries but they are still in use here |
| 05:43.25 | coppice | they are in use almost everywhere. I think only small places like HK and Singapore have clean networks |
| 05:46.55 | *** join/#asterisk Wi_Fi (~OUT@c-24-127-12-85.hsd1.ca.comcast.net) |
| 05:47.21 | t0p | coppice: Nice to talk to you, I better go for lunch now. |
| 05:49.11 | Silik0n | in the states if you were in need of a big ass pipe (say 500mb) who would you talk to? |
| 05:51.28 | *** join/#asterisk newmedian (~crowlther@Quebec-HSE-ppp230300.qc.sympatico.ca) |
| 05:52.41 | newmedian | I'm looking over a quote re: PRI lines from Bell/Megalink. yikes. |
| 05:53.12 | PTG123 | anyone use an ata186 for faxing? |
| 05:55.07 | Silik0n | PTG123: last time I tried that i could only get 711 passthru working but that was a couple years ago |
| 05:55.19 | newmedian | Does anyone have a unit cost per PSTN they can throw me to compare? (Yes, I know Bell/Megalink is likely the most expensive provider) |
| 05:55.25 | PTG123 | i can't figure out how to get that even working |
| 05:57.33 | Silik0n | newmedian: PRIs? |
| 05:57.39 | Silik0n | that all depends on the market you are getting them in |
| 05:58.03 | Silik0n | and if you are looking in .ca or .us or .mx |
| 05:58.31 | Silik0n | and in the states it varies from city to city (hell sometimes block to block to block) |
| 05:58.31 | newmedian | Silik0n: Yes, PRIs, in Toronto/Canada. For example, aside from all the claptrap re-worded terminology services and blarney, the unit price per PSTN is (in Canadian dollars): no contract: $29.13, 1 year: $26.13, 2 years: $25.13, 5 years: $22.63. |
| 05:58.47 | PTG123 | new: where do you need a pri in? |
| 05:59.32 | newmedian | Silik0N & PTG123: likely Toronto/Canada to start, then throughout Canada. |
| 05:59.47 | PTG123 | what kind of quotes are you getting, and why not a sip handoff? |
| 06:00.09 | coppice | PTG123: you probably won't get an ATA186 to FAX |
| 06:00.28 | newmedian | The quotes are just starting to come in, so I've not enough info to compare, but Megalink looks mighty expensive, and seems like they nickel and dime you for everything. |
| 06:00.29 | PTG123 | coppice: why not they work with vonage |
| 06:00.56 | coppice | PTG123: intermittently. they don't do T.38 |
| 06:01.00 | PTG123 | new: why not a sip handoff? |
| 06:01.10 | PTG123 | coppice: well ulaw SHOULD work ok, on a decient connection |
| 06:01.25 | coppice | PTG123: what makes you think that? |
| 06:01.37 | newmedian | I was looking at setting up some basic DID and termination, as a provider, and also handle some SOHO remotely hosted scenarios etc. I would prefer to be the provider, rather than subcontract out. |
| 06:02.12 | PTG123 | new: yah but many disadvantages to a pri instead of sip./. you can still get sip from a wholesaler, and be the provdier |
| 06:03.29 | newmedian | There seems to be a lot of volatility in the market with regard to reliability & availability of providers; Putting the key component in the hands of others also puts your reputation in their hands. |
| 06:03.46 | coppice | PTG123: read http://www.soft-switch.org/foip.html and see if you still expect it to work well |
| 06:03.56 | jeffik | new: you looking for Toronto DID term? |
| 06:04.15 | PTG123 | you do know around 35% of all normal phone calls are converted to ulaw and routed across the us :) |
| 06:04.32 | *** join/#asterisk DFT (~dft@CPE0040050149d6-CM00080d77ae83.cpe.net.cable.rogers.com) |
| 06:04.57 | coppice | PTG123: was that addressed to me? |
| 06:05.05 | orion88 | what is ulaw? |
| 06:05.06 | PTG123 | yes |
| 06:05.15 | newmedian | jeffik: on what basis? Crunching numbers to see what's the best way to go. |
| 06:05.25 | coppice | PTG123: the figure is more like 99%, but that is irrelevant |
| 06:05.57 | newmedian | Judge Dredd is the ulaw |
| 06:06.01 | PTG123 | well llocal calls aren't :) and thats more then 60% of calls |
| 06:06.08 | PTG123 | so in theory it should work fine |
| 06:06.17 | DFT | newmedian: ha |
| 06:06.25 | coppice | PTG123: all local calls are converted to ulaw inthe US |
| 06:06.37 | PTG123 | no they aren't |
| 06:06.57 | coppice | do you still have an analogue switch, then? |
| 06:07.31 | PTG123 | anyhow thats neither here nor there i would just like a fax to go through 1 out of 2 times even :) |
| 06:08.10 | PTG123 | there must be some setting i have wrong on this ata adapter |
| 06:08.48 | coppice | look at the URL I gave, and see my explanation of why ulaw is irrelevant. A few ATAs have a fax mode where they try to work around some of the issues, but not usually that successfully |
| 06:09.17 | Silik0n | PTG123 theres a setting in there somewhere from fax passthru |
| 06:09.17 | newmedian | Which reminds me, slightly OT, who are the puppet masters behind J2? (www.j2.com www.j2corporate.com etc.) |
| 06:09.36 | *** join/#asterisk shidan (~shidan@CPE000625dbadc2-CM014280007905.cpe.net.cable.rogers.com) |
| 06:09.36 | Silik0n | its been too long tho.. check out cco for the manual on that thing |
| 06:09.41 | PTG123 | Silik0n, i think i did that one right |
| 06:10.07 | PTG123 | what is the lbrcodec any idea? |
| 06:12.45 | Silik0n | no my wife blew up my ata186 over a year ago heh |
| 06:13.42 | newmedian | anger or hottiness? |
| 06:13.51 | Silik0n | some how while "cleaning" my desk she plugged the wrong power adapter back into it |
| 06:14.34 | coppice | power adaptors are the work of satan. he gave most of them the same plug :-) |
| 06:14.42 | PTG123 | i am suprised that would break it |
| 06:16.10 | Silik0n | smoked it |
| 06:16.37 | coppice | PTG123: hey, you're even surprised that local calls use ulaw :-) |
| 06:16.49 | PTG123 | the jury is still out on that one :) |
| 06:16.59 | PTG123 | i couldn't argue that one way or another |
| 06:17.11 | coppice | yet you try |
| 06:17.14 | PTG123 | my calls probably do, since i am on fibreoptics |
| 06:17.54 | coppice | practically everyone is on a digital exchange, and the first thing they do with any analogue signal is digitise it to ulaw or alaw |
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| 06:18.18 | PTG123 | well actually i am all voip now, hence my faxing issue |
| 06:18.26 | PTG123 | you know i am keeping a vonage account just for faxing |
| 06:18.30 | PTG123 | how sad is that :) |
| 06:20.22 | *** join/#asterisk oej (~oej@213.204.186.40) |
| 06:22.00 | newmedian | does anyone have a link to network map(s) that show the undersea routes out of New Zealand? (e.g. to Australia, Hawaii, etc.) |
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| 06:22.35 | *** join/#asterisk RestLessGemini (~umairbari@202.142.189.86) |
| 06:23.12 | bsunil | hello can anybody tell me how to ste the callerid |
| 06:23.24 | Qwell | setcallerid |
| 06:25.09 | PTG123 | you do it in the sip.conf or iax.conf for the device |
| 06:26.20 | Qwell | Don't message me...ever |
| 06:29.22 | Silik0n | hmmmmmm |
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| 06:37.51 | jeffik | newmedian: i use Unlimitel in Canada very good, i know the owner |
| 06:40.52 | *** join/#asterisk TheEmperor (~TheEmpero@203.114.48.47) |
| 06:45.40 | TheEmperor | can someone tell me, when i add a |
| 06:45.40 | TheEmperor | t |
| 06:45.54 | *** join/#asterisk clive- (~pirch@rndf-146-37-54.telkomadsl.co.za) |
| 06:45.54 | TheEmperor | to the dialplan, do I need to do define # in order for it to work? |
| 06:49.27 | *** join/#asterisk neowillis (~neowillis@211.144.49.185) |
| 06:51.08 | Zeeek | TheEmperor isn't it easy to try that and see? |
| 06:51.30 | TheEmperor | Zeeek: how can I check that? |
| 06:52.00 | Zeeek | well, I'd assume if it worked without "defining #" you'd be done? |
| 06:52.16 | TheEmperor | well that's the thing, it doesn't work.. |
| 06:52.30 | Zeeek | the question to ask then is why what you did doesn't work |
| 06:53.08 | Zeeek | first of all, what 't' are you talking about? |
| 06:53.15 | TheEmperor | to transfer the call |
| 06:53.29 | TheEmperor | i am wondering, does it matter what codecs are used? |
| 06:53.31 | Zeeek | so, "using 't' in the dial options..." |
| 06:53.47 | TheEmperor | yes, sorry or not explaining clearly |
| 06:53.55 | Zeeek | and it makes it real hard to answer |
| 06:54.02 | Zeeek | start with one thing at a time |
| 06:54.14 | Zeeek | so, "using 't' in the dial options..." what is the ret |
| 06:54.18 | TheEmperor | ok, when i put t into the dial plan, call transfer does not work |
| 06:54.20 | Zeeek | rest of the question? |
| 06:54.27 | Zeeek | explain transfer |
| 06:54.37 | Zeeek | who called whom? |
| 06:55.04 | TheEmperor | say i am extension 2005, i call extension 2006, then 2006 wants to transfer me to 2007, 2006 needs to press #, but in my case it doesn't work.. |
| 06:55.52 | Zeeek | what does your dial command look like? |
| 06:56.09 | Zeeek | Dial(SIP/2006,20,t) ? |
| 06:56.23 | TheEmperor | correct |
| 06:56.35 | Zeeek | should work, so must be the other end client |
| 06:56.42 | TheEmperor | exten => 2006,1,Dial(IAX2/2006,60,t) |
| 06:56.53 | Zeeek | what client is IAX2/2006 ? |
| 06:57.04 | TheEmperor | it's a voip hardphone |
| 06:57.16 | Zeeek | ah I know why it isn't working then |
| 06:57.20 | Zeeek | I have one of those |
| 06:57.29 | TheEmperor | ? |
| 06:57.33 | TheEmperor | codec problem? |
| 06:57.35 | Zeeek | The # is defined as a call key |
| 06:57.39 | Zeeek | right? |
| 06:57.41 | TheEmperor | yes |
| 06:57.52 | TheEmperor | so how can I fix this? |
| 06:57.54 | Zeeek | when you call you press # to send the dial string on those phones |
| 06:58.03 | Zeeek | Tell me I'd be interested |
| 06:58.18 | Zeeek | I haven't worked with it enough to know but that is irritating |
| 06:58.22 | TheEmperor | i got it working once, but now I can't get it working anymore |
| 06:58.28 | Zeeek | especially for retrieving vmail! |
| 06:58.31 | TheEmperor | another thing is, if I use SIP, it's no problem |
| 06:58.39 | Zeeek | that's interesting |
| 06:58.45 | Zeeek | what phone? |
| 06:58.47 | TheEmperor | but I find IAX2 voice quality is better than using SIP |
| 06:59.09 | TheEmperor | Any reason for that? |
| 06:59.12 | Zeeek | what phone? |
| 06:59.17 | TheEmperor | PA168S |
| 06:59.20 | TheEmperor | Using that chip |
| 06:59.23 | Zeeek | that a chip |
| 06:59.34 | Zeeek | what phone? it will vary by phone |
| 06:59.59 | Zeeek | ezeephone or netweb? |
| 07:00.01 | TheEmperor | oh, the at320ed |
| 07:00.43 | TheEmperor | yeah |
| 07:01.03 | TheEmperor | using SIP has all the features but I find with IAX2 protocol, voice quality is better |
| 07:01.21 | Zeeek | have you read this? http://www.voip-info.org/wiki-ATCOM+AT-320 |
| 07:01.22 | TheEmperor | but lacks all these features :( |
| 07:01.47 | Zeeek | you got the latest firmware and all that? |
| 07:02.12 | TheEmperor | yes |
| 07:02.24 | Zeeek | http://www.aredfox.com/edownloads.htm |
| 07:02.28 | Zeeek | for reference |
| 07:02.47 | oej | TheEmperor: When we get the new SIP jitterbuffer, SIP will kick ass with IAX2 :-) |
| 07:02.51 | Zeeek | I like that Windows config program, it's a nice touch |
| 07:03.12 | Zeeek | I tried SIP on my "IAXphones" but didnt like it |
| 07:03.13 | niZon | SIP and IAX are control protocols, they have nothing to do with eachother |
| 07:03.15 | TheEmperor | but why is it that when i use iax2 over sip, the quality is better? |
| 07:03.42 | Zeeek | yes I don't know why, but that's just the implimentation in hos phones |
| 07:03.43 | Zeeek | those |
| 07:03.54 | Zeeek | SIP isn't fabulous for some reason |
| 07:03.55 | oej | The emperor: Most IAX2 implementations has a good jitter buffer |
| 07:04.13 | oej | Asterisk doesn't have a SIP jitterbuffer... yet |
| 07:04.30 | Zeeek | IAX hardphones are, like MArk Twain once said, "Like watching a dog walk on its hind legs" |
| 07:04.31 | riksta | it does i just finished it |
| 07:04.51 | Zeeek | "It isn't done well, but you are always surprised to see it done at all" |
| 07:04.58 | Zeeek | Mark Twain rocks |
| 07:04.59 | TheEmperor | so now the problem is I want to use IAX2 but I don't have call transfer :( |
| 07:05.13 | Zeeek | and the # is hard coded in STABLE AFAIK |
| 07:05.19 | oej | IAX2 has native call transfers |
| 07:05.30 | *** join/#asterisk remmo (~rem@smack.isp.net.au) |
| 07:05.34 | *** join/#asterisk n4y (~frodo7@host-ip237-209.crowley.pl) |
| 07:05.37 | TheEmperor | oej:i can't seem to get call transfer to work when I press # on the hard phones |
| 07:05.44 | Zeeek | oej so how to park a call without using the # key? |
| 07:05.48 | oej | Strange hard phone I guess |
| 07:05.51 | Zeeek | ya |
| 07:06.15 | oej | Zeeek: With stable, I guess you are forced to use the # key |
| 07:06.28 | Zeeek | plus we have one message service that calls us and says please press pound key to hear the message |
| 07:06.41 | oej | But with an IAX client, like a hard phone or a soft phone, the coder could implement a "transfer" button like we have on some SIP phones |
| 07:06.45 | Zeeek | ya but the # key on those hardphones is the dial key! :) |
| 07:06.57 | Zeeek | agreed, "couild" being the key |
| 07:07.09 | Zeeek | coulda .= shoulda |
| 07:07.53 | Zeeek | I haven't played enough, there may even be a transfer button - I don't have those phones here |
| 07:08.17 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 07:09.02 | Zeeek | Oddly enough, I like SIP phopnes with send buttons. It lets you dial very long numbers with bifocals |
| 07:09.24 | Zeeek | without timing out while you hopelessly try to read the last 10 digits |
| 07:11.30 | *** join/#asterisk Inv_arp (junya@adsl-3-237-164.mia.bellsouth.net) |
| 07:14.19 | *** join/#asterisk szw2001 (~vip@218.1.218.187) |
| 07:20.17 | TheEmperor | Zeek: this is weird |
| 07:20.52 | TheEmperor | Zeek: if I use my sip soft phone to call my iax2 hard phone, pressing # on the hard phone works |
| 07:23.14 | Zeeek | interesting |
| 07:23.42 | TheEmperor | ain't that weird?? |
| 07:23.54 | TheEmperor | wonder if anyone knows why |
| 07:23.55 | Zeeek | I know on my phone, # is call and no # functions will work. |
| 07:24.04 | Zeeek | did you read that wiki page I posted |
| 07:24.05 | Zeeek | ? |
| 07:24.07 | TheEmperor | yes |
| 07:24.09 | TheEmperor | nothing there |
| 07:24.13 | Zeeek | nothing about it there? |
| 07:24.15 | Zeeek | ok |
| 07:24.23 | Zeeek | up to you then to find it and post the answer |
| 07:24.33 | TheEmperor | i'll try to find out the answer.. |
| 07:26.40 | *** join/#asterisk ilium007 (~ilium@203.94.178.214) |
| 07:27.09 | ilium007 | i am testing my asterinisk setup from work, and i need to open up my Netscreen firewall to allow incomming SIP traffic |
| 07:27.19 | ilium007 | from what i can gather it is UDP 5060 traffic ??? |
| 07:27.46 | ilium007 | can someone tell me if this is correct ? |
| 07:29.45 | ilium007 | +? |
| 07:30.03 | *** join/#asterisk rabelais (~blank@ca-stmnca-cuda4-gen2c1-171.vnnyca.adelphia.net) |
| 07:33.07 | *** join/#asterisk ptblank (~MURDER1@24-49-13-142.lmdaca.adelphia.net) |
| 07:36.18 | Zeeek | astrisk is behind NAT? |
| 07:36.22 | Zeeek | or client? |
| 07:37.34 | ilium007 | yes it is |
| 07:37.51 | ilium007 | asterisk server nehind NAT firewall and client is also behond NAT forewall |
| 07:37.56 | Zeeek | I'd recommend TCP/UDP and when it works, remove TCP and see |
| 07:38.08 | Zeeek | because I've never been sure |
| 07:38.25 | Zeeek | I think you may need to open 10000-n for the asterisk side too |
| 07:38.53 | Zeeek | <PROTECTED> |
| 07:38.54 | ilium007 | the protocol is listed as UDP, but my Netscreen 5GT only allows a VIP ( sort of like port forwarding ) to TCP ports |
| 07:38.58 | ilium007 | SIP is UDP |
| 07:39.16 | pimpwell | gnight yall |
| 07:39.17 | pimpwell | http://www.tshirthell.com/images/contestpics/a249_003.jpg |
| 07:40.01 | Zeeek | you could try putting * in the DMZ |
| 07:40.37 | ilium007 | yeah i suppose i can....hmmmmmm |
| 07:41.35 | ilium007 | thanks |
| 07:42.52 | Zeeek | never has worked for me though |
| 07:42.59 | TheEmperor | given up, gone back to SIP |
| 07:43.03 | TheEmperor | working good now :) |
| 07:43.08 | Zeeek | a defector - horrible |
| 07:43.14 | TheEmperor | what to do? |
| 07:43.32 | TheEmperor | how can I deliver a pbx system without call transfer to a client? |
| 07:43.35 | Zeeek | find and publish a solution |
| 07:43.40 | TheEmperor | they wouldn't accept it... |
| 07:43.55 | Zeeek | you think that phone is ready for prime time? |
| 07:44.30 | TheEmperor | with SIP I think it is acceptable |
| 07:44.48 | Zeeek | I honestly prefer my BT102 to those phones |
| 07:44.58 | Zeeek | but that is very subjective obviously |
| 07:45.28 | Zeeek | I also like pockets on my T-shirts |
| 07:45.38 | Zeeek | but no pen protectors |
| 07:46.06 | TheEmperor | haha |
| 07:46.10 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 07:46.17 | Zeeek | join #asterisk-waynes_world |
| 07:46.28 | *** join/#asterisk clive- (~pirch@rndf-146-37-54.telkomadsl.co.za) |
| 07:47.09 | Zeeek | I own three IAX phones now |
| 07:47.36 | Zeeek | One doesn't work worth a shit, one works "ok" and one works "fairly well" |
| 07:48.10 | Zeeek | God I've wasted a lot of money to voIP experiments |
| 07:48.55 | Zeeek | "I did not say I was rich! I said, if I had all the money I have spent, I would *be* rich!" |
| 07:49.24 | Zeeek | I seem to be profound on world history today |
| 07:49.37 | Zeeek | Mark Twain, Chicago bluesmen... |
| 07:50.00 | Zeeek | somebody better break in or I'll be kicked for flooding! |
| 07:50.24 | citats | Zeeek: which iax phone is which? |
| 07:50.36 | Zeeek | You can guess but I will be polite and not say |
| 07:50.54 | Zeeek | in the name of community solidarity |
| 07:51.35 | *** join/#asterisk jason357 (~m00@67.159.26.120) |
| 07:51.38 | Zeeek | my latest adventure with voip expenditures isn't bad either |
| 07:52.22 | Zeeek | I bought a polycom for about $200 and brought it home. It would cost at least 50% more here |
| 07:52.52 | Zeeek | unfortunately I had to send it back. Sending it FedEx or UPS would cost $110 ONE WAY! |
| 07:53.21 | Zeeek | What would you do? (rhetorical question) pay $200 shipping both ways or gamble? |
| 07:54.16 | Zeeek | it was a tough decision, but it got resolved by the fact that I didn't have enough cash to pay and had to send it via the P.O. for $45 |
| 07:54.27 | Zeeek | but now we all know, it may never arrive... |
| 07:55.36 | Zeeek | Lesson: bite the bullet and find a local distributor who sells the thing for 2x the US price but will exchange it if it's bad |
| 07:57.02 | *** join/#asterisk implicit (~implicit@ip68-5-148-1.oc.oc.cox.net) |
| 07:57.53 | citats | MALLOC_DEBUG = -include $(PWD)/include/asterisk/astmm.h |
| 07:58.02 | citats | bah, stupid mouse |
| 07:59.07 | Zeeek | shit happens |
| 08:05.14 | *** join/#asterisk prh (~paul@212.13.203.80) |
| 08:15.31 | *** join/#asterisk shido6 (~shido@d57-87-253.home.cgocable.net) |
| 08:17.19 | *** join/#asterisk Delvar (~irc@83.146.53.34) |
| 08:22.32 | *** join/#asterisk ]expic (~Inferna@217.22.176.122) |
| 08:22.35 | *** join/#asterisk darby_t (~tom@host-ip237-209.crowley.pl) |
| 08:22.59 | ]expic | i have BRI card and i see errors like May 16 09:41:05 NOTICE[31353]: PRI got event: HDLC Abort (6) on Primary D-channe |
| 08:22.59 | ]expic | l of span 4 |
| 08:23.03 | *** join/#asterisk ellvis (~ellvis@adsl-flat-basic-105.84-47-116.telecom.sk) |
| 08:23.08 | ellvis | hi people |
| 08:23.10 | *** join/#asterisk fenlander (~neils@82.152.81.57) |
| 08:23.14 | ]expic | looks like ISDN layer 2 is down, can anybody heelp me? |
| 08:24.57 | shido6 | whats up? |
| 08:25.33 | ]expic | i have BRI card and i see errors like May 16 09:41:05 NOTICE[31353]: PRI got event: HDLC Abort (6) on Primary D-channe |
| 08:25.33 | ]expic | -=10:22:43=- <]expic> l of span 4 |
| 08:25.41 | ellvis | is there a way how to reduce traffic during phonecalls? i am getting 10 times higher values then online calculators are telling me (http://www.asteriskguru.com/bandwidth_calculator.php). i am using g729 (over iax2 channel) |
| 08:26.54 | ]expic | ellvis: payload |
| 08:29.43 | ]expic | can it be the problem that i missmatched with linecoding? |
| 08:30.59 | t0p | err, can mp3 be used with ControlPlayback? |
| 08:32.45 | ellvis | ]expic: so, then it's all right, right? |
| 08:33.30 | ]expic | ellvis: whar is all right? |
| 08:33.43 | ]expic | ellvis: i see green light on BRI port |
| 08:33.51 | ]expic | ellvis: but i cannot get incoming calls |
| 08:34.08 | ]expic | ellvis: i see these incoming HDLC errors |
| 08:34.16 | ellvis | ]expic: ah, sorry, haven't seen your question |
| 08:34.32 | ellvis | ]expic: it was reaction on your "payload" |
| 08:34.47 | ]expic | t0p: yes you can play mp3, with MP3Player or mpg123 |
| 08:35.03 | ellvis | ]expic: and when you call that line from outside, you're getting non-existent number? |
| 08:35.27 | ]expic | ellvis: i think yes |
| 08:35.39 | ]expic | ellvis: i also have TA built-in into NT1 |
| 08:36.07 | ellvis | ]expic: check the type of line. if it's point - to - point or if it's point - to - multipoint |
| 08:37.15 | ]expic | ellvis: is it normal that asterisks says PRI signalling was activated on BRI line? |
| 08:37.24 | *** join/#asterisk joepurl (~joepurl@61.172.25.30) |
| 08:38.05 | ellvis | ]expic: i don't know, i don't think so. but my experience is very poor... |
| 08:38.28 | ]expic | ellvis: it's my first BRI card , so i am also not very convinced |
| 08:40.41 | ellvis | ]expic: i had once problem with BRI and it was that the line was configured as point-to-point and i had to ask provider to change it to point-to-multipoint |
| 08:40.45 | t0p | ]expic: Yeah, but is it possible to use it for ControlPlayback directly? |
| 08:40.48 | *** join/#asterisk pids (~sockpuppe@dsl081-072-084.sfo1.dsl.speakeasy.net) |
| 08:41.15 | ]expic | ellvis: only ISP can provide me this info? |
| 08:41.29 | ellvis | ]expic: yes |
| 08:42.38 | ]expic | ellvis; what's the difference between multipoint and point2point? |
| 08:42.50 | pids | Anyone have any idea why an IAX call would get disconnected after about 1.5 minutes while someone is leaving a message. It only happens when someone is calling in from one of our UK numbers. Calling into out US numbers works fine. |
| 08:43.50 | ellvis | ]expic: when the line is point-to-point, you can't receive calls untill they're not comming from one place |
| 08:44.01 | ellvis | ]expic: at least, that's how i understand it |
| 08:44.40 | ellvis | ]expic: that was the problem i had - none was abble to call me, i changed it and things went right |
| 08:46.36 | *** join/#asterisk newmember (~newmember@S010600a0c93dce87.cg.shawcable.net) |
| 08:54.49 | *** join/#asterisk wt (~wt@242678hfc216.tampabay.res.rr.com) |
| 08:55.10 | wt | anyone here know the dial application well? |
| 08:58.20 | wt | By default (no options), what happens if you Dial an extension that is in use? |
| 08:58.57 | wt | I thought that it went to the next priority, but that doesn't seem to be right |
| 08:59.27 | ]expic | May 16 10:57:55 NOTICE[31827]: PRI got event: HDLC Abort (6) on Primary D-channe |
| 08:59.27 | ]expic | l of span 1 |
| 08:59.30 | ]expic | :( the same |
| 09:06.47 | ellvis | pri got event... it's shouldn't be there if you're using bri, no? |
| 09:07.24 | ellvis | i don't have much experiences with CAPI at all, i used (still use) only HiSax |
| 09:09.22 | *** join/#asterisk Hali_303 (~Hali_303@224.235-182-adsl-pool.axelero.hu) |
| 09:09.33 | Hali_303 | hi! |
| 09:09.42 | Hali_303 | i'm installing an X100P card |
| 09:09.51 | Hali_303 | downloaded zaptel-1.0.7.tar.gz |
| 09:09.55 | Hali_303 | and compiled, installed |
| 09:10.06 | Hali_303 | however, when doing modprobe wcfxo: |
| 09:10.27 | Hali_303 | /lib/modules/2.4.26/misc/wcfxo.o: init_module: No such device |
| 09:10.27 | Hali_303 | Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. |
| 09:10.27 | Hali_303 | <PROTECTED> |
| 09:10.27 | Hali_303 | /lib/modules/2.4.26/misc/wcfxo.o: insmod /lib/modules/2.4.26/misc/wcfxo.o failed/lib/modules/2.4.26/misc/wcfxo.o: insmod wcfxo failed |
| 09:10.50 | Hali_303 | what could be the cause? the card is installed of course |
| 09:11.06 | Hali_303 | lspci: |
| 09:11.08 | Hali_303 | 0000:00:14.0 Communication controller: Individual Computers - Jens Schoenfeld Intel 537 |
| 09:11.48 | Hali_303 | (and this appears 7 times, with numbers like 0000:00:14.1 .. 0000:00:14.7) |
| 09:15.49 | *** join/#asterisk Aze` (~aze@80.105.162.229) |
| 09:20.25 | *** join/#asterisk TheEmperor (~TheEmpero@203.114.48.47) |
| 09:20.44 | *** join/#asterisk flynux (pefwp2r@217.145.32.104) |
| 09:22.30 | *** join/#asterisk RoyK (~roy@179.80-203-29.nextgentel.com) |
| 09:27.11 | Zeeek | what's the IRQ table look like? |
| 09:29.01 | *** join/#asterisk gaggaman (~ga@host-82-135-28-39.customer.m-online.net) |
| 09:29.29 | TheEmperor | anyone implemented bandwidth shapers in their * boxes? |
| 09:39.47 | Hali_303 | Zeeek, you mean /proc/interrupts? |
| 09:40.52 | Zeeek | ya |
| 09:41.00 | *** join/#asterisk brc__ (~brian@brc.base.supporter.pdpc) |
| 09:41.15 | Hali_303 | <PROTECTED> |
| 09:41.15 | Hali_303 | <PROTECTED> |
| 09:41.15 | Hali_303 | <PROTECTED> |
| 09:41.15 | Hali_303 | <PROTECTED> |
| 09:41.15 | Hali_303 | <PROTECTED> |
| 09:41.16 | Hali_303 | <PROTECTED> |
| 09:41.18 | Hali_303 | <PROTECTED> |
| 09:41.22 | Hali_303 | <PROTECTED> |
| 09:41.24 | Hali_303 | <PROTECTED> |
| 09:41.26 | Hali_303 | <PROTECTED> |
| 09:41.50 | Zeeek | looks good (don't flood the channel, use http://pastebin.ca) |
| 09:42.05 | Zeeek | check on /etc/zapata.conf |
| 09:42.22 | Hali_303 | Zeeek, ok, sorry |
| 09:42.29 | Zeeek | no big deal at them moment |
| 09:42.39 | Zeeek | but you'll get called on that later :) |
| 09:43.16 | Hali_303 | http://pastebin.ca/11873 |
| 09:43.34 | Hali_303 | I didnt touch zapata.conf yet.. maybe this is the problem? |
| 09:44.04 | Zeeek | yes it is |
| 09:44.11 | eper-werk | anybody bought hardware from voiptalk? (there the Uk distro for the wildcard) they seem to not be very good at replying to emails etc as in i've ordeed something voer a week ago and there not replying to when/if it will be shipped |
| 09:44.13 | Zeeek | what docs have you been working from? |
| 09:44.42 | Hali_303 | Zeeek, I just googled and found some mailing list posts.. do you have a good doc on installing the X100P? |
| 09:45.02 | Zeeek | Hali_303 you'll need something like this: fxsks = 1 |
| 09:45.08 | Zeeek | for an FXO |
| 09:45.22 | Zeeek | Starter tutorial: |
| 09:45.22 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
| 09:45.22 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
| 09:45.22 | Zeeek | http://www.automated.it/guidetoasterisk.htm |
| 09:45.23 | Zeeek | THE reference of the moment: |
| 09:45.23 | Zeeek | http://www.asteriskdocs.org |
| 09:45.38 | Hali_303 | ok thx, I'll take a look at those |
| 09:45.41 | Zeeek | look at the automated.it site real quick and find the zaptel part |
| 09:46.04 | Zeeek | in fact:http://www.automated.it/guidetoasterisk.htm#_Toc49248763 |
| 09:46.59 | Zeeek | eper-werk don't they give a phone number? |
| 09:47.14 | Zeeek | I have an account but never ordered any hardware from them |
| 09:48.54 | Hali_303 | Zeeek, from this doc, it seems that I'll have to do modprobe wcfxo before editing /etc/zapata.conf |
| 09:49.11 | Hali_303 | imean zaptel.conf |
| 09:49.34 | Hali_303 | anyways I've added fxsks=1 |
| 09:49.49 | Hali_303 | but the error message remains ;( |
| 09:51.03 | Hali_303 | shouldnt the X100P show up in the interrupt table? maybe there are too many cards in the machine (3 ethernet cards+1 VGA cards) and they take up too many irqs? |
| 09:51.11 | Zeeek | yes it should |
| 09:51.29 | Zeeek | wow! I didn't look at the crads, just the numbers |
| 09:51.32 | Hali_303 | but as you can see, it doesnt show up in mine |
| 09:51.37 | Zeeek | IRQ numbers. Your card isn't seen! |
| 09:51.49 | Hali_303 | yes. then this will be the problem :( |
| 09:51.55 | Zeeek | <PROTECTED> |
| 09:52.19 | Hali_303 | hmm strange.. how could I debug what is causing the problem? |
| 09:52.24 | Zeeek | isn't there a message on boot about the card? |
| 09:52.32 | Hali_303 | in /var/log/messages? |
| 09:53.05 | Zeeek | Found a Wildcard FXO: Wildcard X101P |
| 09:53.05 | Zeeek | PCI: Found IRQ 7 for device 02:0a.0 |
| 09:53.05 | Zeeek | wcfxo: DAA mode is 'FCC' |
| 09:53.10 | Zeeek | dmesg |
| 09:53.33 | *** join/#asterisk ]expic (~Inferna@217.22.176.122) |
| 09:54.17 | Hali_303 | Zeeek, this should be the output of the wcfxo driver, right? |
| 09:54.28 | Zeeek | ya sumptin like that |
| 09:54.37 | Hali_303 | but since it is a kernel module, it doesnt get loaded on boot |
| 09:54.51 | Hali_303 | only when I load it with modprobe wcfxo |
| 09:54.54 | Zeeek | only if you tell it to |
| 09:54.54 | Hali_303 | manually |
| 09:54.57 | Hali_303 | yes |
| 09:55.01 | Zeeek | agreed |
| 09:55.07 | Hali_303 | but then it gives the above errors on the irq ;( |
| 09:55.44 | Zeeek | painful solution, but you'd have to remove all non-essential PCI cards and see if you see it in theIRQ table |
| 09:55.56 | Hali_303 | yes, I'm afraid :( |
| 09:55.59 | Zeeek | or switch em around |
| 09:56.12 | Zeeek | you have 2 eth ifaces? |
| 09:56.16 | Hali_303 | 3 :) |
| 09:56.18 | Zeeek | what else is in this box? |
| 09:56.23 | Zeeek | o 3... |
| 09:56.31 | Hali_303 | 3 PCI ethernet cards |
| 09:56.43 | Zeeek | looks like that's it. Do you need USB? |
| 09:56.44 | Hali_303 | and 1 VGA cards |
| 09:56.50 | Hali_303 | no.. |
| 09:56.59 | Hali_303 | hm good idea I turn off USB in the bios, right? |
| 09:57.00 | Zeeek | maybe try to disable that then |
| 09:57.08 | Zeeek | hopefully |
| 09:57.08 | Hali_303 | and maybe the 2nd IDE too |
| 09:57.15 | Zeeek | if there isn't any |
| 09:57.34 | Zeeek | you *could* pull two of the eth just for grins... |
| 09:57.44 | Hali_303 | ok thx! |
| 09:57.47 | Zeeek | np |
| 09:57.54 | Hali_303 | I'll try and report if it works :) |
| 09:57.57 | Hali_303 | bye |
| 09:58.00 | *** part/#asterisk Hali_303 (~Hali_303@224.235-182-adsl-pool.axelero.hu) |
| 09:58.02 | Zeeek | of course the quality of support is proportional to the price |
| 09:59.16 | TheEmperor | does anyone have an example of gotoiftime config? |
| 09:59.46 | Zeeek | there are a few here: http://asteriskdocs.org |
| 10:00.10 | TheEmperor | Zeeek: will check it out thanks |
| 10:00.39 | Zeeek | download the PDF and read it. It'll save you a lot of IRC time :) |
| 10:01.06 | Zeeek | it's 12 noon here - do you know where your interrupt routines are? |
| 10:08.59 | RoyK | ~lart Zeeek |
| 10:10.15 | *** join/#asterisk trimi` (~da@62.162.232.175) |
| 10:17.38 | RoyK | anyone that knows how i can find out which processes are using the most i/o time? |
| 10:18.57 | wt | anyone here know anything about ChanIsAvail? |
| 10:19.27 | wt | I am looking for a little info |
| 10:20.33 | wt | such as, why, when the channel is busy, does "exten => s,2,ChanIsAvail(Zap/2)" not jump to priority 103 when Zap/2 is busy |
| 10:23.12 | RoyK | er |
| 10:23.15 | RoyK | why? |
| 10:23.45 | wt | why what? |
| 10:24.09 | wt | I am trying to implement 911 as suggested in the voip-info wiki |
| 10:24.10 | tzafrir | wt, do you use ${CHANAVAILORIG} ? |
| 10:24.16 | *** join/#asterisk r0d3nt (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 10:24.31 | wt | tzafrir: I have never heard of that |
| 10:24.49 | tzafrir | ChanIsAvail does not jusmp. It simply sets some channel variables |
| 10:25.00 | RoyK | wt: don't bother with tzafrir. he's prolly drunk |
| 10:25.01 | tzafrir | show application ChanIsAvail |
| 10:25.24 | wt | Why does the wiki say "If none of the requested channels are available the new priority will be n+101 (unless such a priority does not exist, in which case ChanIsAvail will return -1)." then? |
| 10:25.36 | wt | that must be incorrect |
| 10:26.08 | wt | that says the same thing tzafrir |
| 10:26.22 | tzafrir | wt, well it WorksForMe (with Zap) |
| 10:26.30 | tzafrir | on -stable |
| 10:26.32 | wt | I have Zap, too |
| 10:26.40 | wt | tzafrir: how do you do it? |
| 10:26.45 | wt | I am on 1.0.7 |
| 10:26.50 | wt | with Zap devices |
| 10:26.57 | tzafrir | RoyK: me ? drunk? nah |
| 10:27.03 | RoyK | :) |
| 10:27.19 | wt | ${CHANAVAIL} is Zap/2-busy-34432432 after ChanIsAvail |
| 10:27.30 | wt | it just doesn't seem to jump as expected |
| 10:27.50 | wt | and I don't know how to work around it or make it work |
| 10:28.02 | wt | exten => s,1,SetVar(SET_EMERG_FLAG=0) |
| 10:28.02 | wt | exten => s,2,ChanIsAvail(${EMERGENCY_TRUNK}) |
| 10:28.02 | wt | exten => s,3,SetGlobalVar(EMERGENCY=1) |
| 10:28.02 | wt | exten => s,4,SetVar(SET_EMERG_FLAG=1) |
| 10:28.07 | wt | that's what I have |
| 10:28.12 | tzafrir | wt, pastebin |
| 10:28.16 | RoyK | ~pastebin> |
| 10:28.16 | tzafrir | ~pastebin |
| 10:28.17 | jbot | i heard pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 10:28.24 | wt | sorry |
| 10:28.36 | wt | I thought 4 lines wouldn't be too bad with the low traffic on the channel |
| 10:28.36 | RoyK | ~lart wt |
| 10:28.42 | RoyK | :) |
| 10:29.44 | RoyK | kill doesn't work :( |
| 10:30.33 | wt | http://pastebin.ca/11876 |
| 10:30.46 | wt | that is the whole section |
| 10:31.19 | wt | the 102 and 104 were testing if I was off by one |
| 10:31.42 | tzafrir | wt, try testing ${CHANAVAILORIG}? |
| 10:31.48 | *** join/#asterisk Hali_303 (~Hali_303@a84-0-150-92.adsl-pool.axelero.hu) |
| 10:32.04 | Hali_303 | Zeeek, wow ;) now it works!!! I just changed the order of the cards! |
| 10:32.07 | tzafrir | or CHANAVAIL |
| 10:32.28 | Hali_303 | Zeeek, I mean modprobe now works, on the asterisk side, I have to do a lot of config I guess |
| 10:32.35 | Zeeek | ok great news! |
| 10:33.00 | wt | is there a way to do a regex match? |
| 10:33.08 | *** join/#asterisk zoa (~zoa@pirus.securax.be) |
| 10:33.17 | *** join/#asterisk TheEmperor (~TheEmpero@203.114.48.47) |
| 10:35.33 | *** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc) |
| 10:35.33 | *** topic/#asterisk is Asterisk: The Open Source PBX || Astricon Europe -- Madrid, Spain -- June 15-17 || PBX Developers Conference in Chicago Aug 3-5 Register Now! http://www.asterlink.com/cluecon.htm |
| 10:35.42 | RoyK | ~lart hiself |
| 10:35.50 | RoyK | ~lart himself |
| 10:35.54 | wt | ~lart jbot |
| 10:36.22 | Zeeek | Hali_303 just a few lines in /etc/zaptel.conf |
| 10:36.35 | Zeeek | and a few in zapata.conf |
| 10:38.21 | *** join/#asterisk newbien (~e@147.241.33.65.cfl.res.rr.com) |
| 10:38.36 | Hali_303 | Zeeek, yes, it seems that zaptel.conf and zapata.conf now works okay, because ztcfg -vv: |
| 10:38.42 | Hali_303 | Channel 01: FXS Kewlstart (Default) (Slaves: 01) |
| 10:38.42 | Hali_303 | 1 channels configured. |
| 10:38.48 | *** join/#asterisk kdayn (~kdayn@codeine.svnets.lv) |
| 10:38.50 | Zeeek | there you are |
| 10:39.00 | Hali_303 | now what to do next? the tutorial on the web ends with this.. |
| 10:39.02 | Zeeek | start calling your friends now |
| 10:39.13 | *** join/#asterisk fantomax1 (~fanto@81.208.114.250) |
| 10:39.14 | Hali_303 | I need to set up an extension I guess |
| 10:39.20 | Hali_303 | which uses this card |
| 10:39.23 | Zeeek | you need to read this: |
| 10:39.39 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
| 10:39.49 | Hali_303 | ok, thx! |
| 10:40.31 | Zeeek | or if you feel really lucky add exten => 2000,1,Dial(ZAP/1/18005551212) |
| 10:40.45 | Zeeek | extensions reload and dial 2000 |
| 10:40.59 | Zeeek | (after asterisk running, obviously) |
| 10:41.56 | Hali_303 | 18005551212 is the phone number to dial when extension 2000 dialed right? |
| 10:42.02 | wt | setgroup can be used for lines you are dialing to? |
| 10:42.05 | Zeeek | yes, dumb example |
| 10:42.13 | Hali_303 | Zeeek, ok I'll try, thx!! |
| 10:42.28 | Zeeek | first read that link - just a few pages |
| 10:42.47 | Zeeek | "Handing Off Calls to the Analog Phone Network via SIP" |
| 10:42.52 | wt | can I use SetGroup to seize an extension for 911 calls? |
| 10:43.25 | Zeeek | in an office setting? |
| 10:45.04 | Zeeek | I think the only good 911 solution for fixed phones is to have an ordinary phone that can be switched on to an ordinary line to call 911, think of it as a fire extinguisher. Any voip solution will fall short by that measure |
| 10:46.23 | Zeeek | I'm always game to bet money on voip hardware, but less excited about betting someone's life on voIP itself :) |
| 10:47.54 | wt | I am using POTS |
| 10:48.08 | wt | I just want to be able to force anyone else off when 911 is dialed |
| 10:48.46 | Zeeek | my system is failsafe |
| 10:49.16 | wt | that way these people don't have to go down 2 floors to get to a phone that 911 works on |
| 10:49.19 | Zeeek | a phone with the ringer removed would be great |
| 10:49.30 | wt | I am not looking for worst case scenario |
| 10:49.31 | Zeeek | put one on every floor |
| 10:49.55 | TheEmperor | is this a correct command? exten => 4000,1,Playback(weareclosed) |
| 10:49.55 | Zeeek | again, betting lives on zapata and asterisk... nah |
| 10:50.45 | fantomax1 | hi all |
| 10:51.13 | Zeeek | wt I expect though that to do what you describe you'd need to write code to destroy or at least hangup a channel and then make the call |
| 10:51.17 | wt | Well, even supposing I did that, I would like 911 to work from any phone if the asterisk box is up |
| 10:51.24 | fantomax1 | maximum retries exceeded on calls .. is a matter of connection or system lack of resorces ? |
| 10:51.45 | Zeeek | fantomax1 it means asterisk can't reach whatever it's tring to reach |
| 10:51.47 | wt | Zeeek: that sounds correct |
| 10:52.11 | Zeeek | maybe softhangup |
| 10:52.22 | fantomax1 | is a matter of connection ? or too many calls ? |
| 10:52.34 | wt | Zeeek: its in my dialplan already |
| 10:52.35 | fantomax1 | i have 370 channels on a double Xeon |
| 10:52.40 | Zeeek | fantomax1 I think it's the device not being reachable |
| 10:52.53 | fantomax1 | the termination you mean ? |
| 10:53.07 | fantomax1 | maybe it's got the timeout |
| 10:53.14 | Zeeek | wt why screw with seeing if a channel is available at all? If someone calls 911 hang em all up! |
| 10:53.14 | wt | Zeeek: I just need to know how to detect if a channel is used and then I can jump to the softhangup |
| 10:53.48 | wt | Zeeek: I guess that curiosity comes from when I get to do a bigger installation |
| 10:53.57 | Zeeek | you could do something like this: |
| 10:54.23 | Zeeek | Playback(ernie in sales is calling 911 again) |
| 10:54.29 | Zeeek | Hangug(everyone) |
| 10:54.36 | Zeeek | Dial(ZAP/1/911) |
| 10:54.41 | wt | not to mention, if someone jumps in at just the right time, it would get the line before I do, and then I need to softhangup again |
| 10:54.46 | Zeeek | pseudocode obviously |
| 10:55.34 | wt | I wish that there was a semaphore or mutex so that I could synchronize the access to the region |
| 10:55.39 | wt | in the dialplan, that is |
| 10:56.01 | wt | I think that makes sense |
| 10:56.04 | Zeeek | if wishes were horses... |
| 10:56.10 | Zeeek | beggars would ride |
| 10:56.10 | wt | Zeeek: thanks :-P |
| 10:56.19 | Zeeek | and if a frog had wings... |
| 10:56.26 | Zeeek | he would bump his ass when he jumps |
| 10:56.30 | wt | I was hoping you'd say, "but there is |
| 10:56.32 | wt | " |
| 10:56.34 | Zeeek | ooops - wouldn't |
| 10:56.45 | *** join/#asterisk csg (foobar@i-195-137-6-228.freedom2surf.net) |
| 10:57.06 | Zeeek | I guess you'll have to write ome code to access the semaphores |
| 10:57.21 | wt | or just synchronize access to a channel |
| 10:57.38 | *** join/#asterisk easimon (~easimon@localhorst.kawo2.RWTH-Aachen.DE) |
| 10:57.40 | wt | so that I could just say hangup the channel and then dial atomically |
| 10:57.41 | Zeeek | I was surprised to learn I couldn't just barge in on a channel (like picking up a phone) |
| 10:58.19 | wt | for Zap channels, it looks like you can |
| 10:58.32 | Zeeek | yes you can in a way |
| 10:58.38 | wt | not that you have zap channels |
| 10:58.41 | Zeeek | and it's cool because they can't hear you |
| 10:58.50 | Zeeek | oh but we do |
| 10:58.57 | wt | oh |
| 10:59.14 | Zeeek | ture I refered to voIP above in my tirade about 911 but I should have said pbx |
| 10:59.25 | wt | I wish I could display globals from the console also; that'd be neat |
| 10:59.36 | Zeeek | can anyone seriously say "I trust my life to asterisk"? |
| 10:59.41 | Zeeek | honestly? |
| 10:59.58 | wt | I have to trust my life to NorTel when I am at FedEx |
| 11:00.01 | Zeeek | "my life and the potential loss of lives and property of hundreds" |
| 11:00.04 | wt | what's the difference? |
| 11:00.17 | Zeeek | come on, you know what the difference is |
| 11:00.25 | Zeeek | not that that's so great either :) |
| 11:00.40 | wt | there's not much difference when it comes down to it |
| 11:01.23 | Zeeek | of course, if you're confident (you generic not YOU) with your asterisk install, then you have to never update without very, very extensive testing |
| 11:02.00 | Zeeek | I don't know, do pbx like nortel drop calls for no visible reason from time tot ime and stuff like that? |
| 11:02.16 | Zeeek | we had a pbx here once. It worked as reliably as a phone |
| 11:02.40 | Zeeek | not much intelligence there, but it did what it had to do |
| 11:02.57 | Zeeek | asterisk is great, but it is a work in progress |
| 11:03.29 | Zeeek | and everyone that uses asterisk should listen in to at least ONE developer conference call |
| 11:05.18 | Zeeek | wt wrt dispalying globals, ya it would be nice |
| 11:09.19 | *** join/#asterisk hellop (~LeeHarvey@cpe-70-93-40-171.hawaii.res.rr.com) |
| 11:09.37 | hellop | hi |
| 11:11.08 | Zeeek | hello.p |
| 11:14.21 | Pj386 | Hey, anyone got any info on GAPS (the centralized configuration server from grandstream) ? |
| 11:14.50 | Pj386 | I can't find any info, even on grandstream's website (only reference in manuals to "Please refer to GAPS documentation") |
| 11:15.26 | Zeeek | you sure it wasn't "please refer to the many gaps in our documentation" ? |
| 11:15.38 | *** join/#asterisk RestLessGemini (~umairbari@202.142.189.86) |
| 11:16.10 | *** join/#asterisk TheEmperor (~TheEmpero@203.114.48.47) |
| 11:16.10 | RoyK | grandstream sucks.... |
| 11:16.11 | Pj386 | I'm not talking about asterisk ;) |
| 11:16.14 | Zeeek | haha |
| 11:16.33 | TheEmperor | can someone please tell me why this command doesnt work? i have already recorded the sound file...n someone tell me why this command doesn work? i have already recorded the sound file exten => 4000,2,Playback(weareclosed) |
| 11:16.33 | TheEmperor | * Disconnected |
| 11:16.34 | Zeeek | never hear of the GAPS from them |
| 11:16.36 | *** join/#asterisk jeffgus (~jeffgus@alpha.zimage.com) |
| 11:16.48 | Pj386 | though this one was low :( |
| 11:16.49 | Zeeek | what's on 1 ? |
| 11:17.00 | TheEmperor | Answer |
| 11:17.16 | Zeeek | what does CLI say? |
| 11:17.18 | RoyK | TheEmperor: background, not playback |
| 11:17.28 | Pj386 | should be a meta tftp server wich allow you to store phone configs and distribute them |
| 11:17.38 | TheEmperor | RoyK:oh, i think you are right... |
| 11:17.45 | Zeeek | Pj386 ya, I guessed that |
| 11:17.49 | TheEmperor | Zeeek: when i dial 4000, nothing happens, just hangs up my phone.. |
| 11:18.00 | Zeeek | no CLI output? |
| 11:18.14 | TheEmperor | Zeeek:none.. |
| 11:18.23 | Zeeek | and calling another extension works? |
| 11:18.36 | TheEmperor | yes |
| 11:18.46 | Zeeek | maybe the phone is eating some numbers ? |
| 11:18.52 | Zeeek | or adding some? |
| 11:18.52 | RoyK | set verbose 99 |
| 11:18.57 | TheEmperor | hmm, i will check |
| 11:21.19 | *** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc) |
| 11:21.20 | *** topic/#asterisk is Asterisk: The Open Source PBX || Astricon Europe -- Madrid, Spain -- June 15-17 || PBX Developers Conference in Chicago Aug 3-5 Register Now! http://www.asterlink.com/cluecon.htm |
| 11:21.21 | TheEmperor | how do I check if the sound file I recorded is still there? |
| 11:21.27 | Zeeek | ls ? |
| 11:21.46 | Zeeek | but be logical, if the file isn't found asterisk will complain |
| 11:22.04 | Zeeek | dial 4111 |
| 11:22.10 | TheEmperor | hmm, i changed the number from 4000 2700, still doesn't work |
| 11:22.11 | Zeeek | what happens? |
| 11:22.37 | Zeeek | you reloading extensions? |
| 11:22.45 | TheEmperor | same thing.. |
| 11:22.47 | TheEmperor | nothing happens |
| 11:23.08 | Zeeek | put the 4000 ext in pastebin |
| 11:23.34 | TheEmperor | just checked the sounds folder, the sound file i recorded is there.. |
| 11:24.41 | TheEmperor | Zeeek:http://pastebin.ca/11881 |
| 11:24.44 | Zeeek | use logic: create an exten => 4000,1?NoOp(Yo!) and see what happens |
| 11:25.35 | Zeeek | is closed included in the context your phone is dialing from? |
| 11:25.53 | TheEmperor | Zeeek: oh... |
| 11:25.57 | Zeeek | hmmmm |
| 11:26.31 | TheEmperor | Zeeek: something I missed... :) |
| 11:31.58 | *** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc) |
| 11:31.58 | *** topic/#asterisk is Asterisk: The Open Source PBX || Astricon Europe -- Madrid, Spain -- June 15-17 || PBX Developers Conference in Chicago Aug 3-5 Register Now! http://www.asterlink.com/cluecon.htm |
| 11:33.44 | *** join/#asterisk bkw__ (~brian@adsl-69-154-0-136.dsl.tulsok.swbell.net) |
| 11:38.11 | *** join/#asterisk bublbobl (~chatzilla@nat-80-64-224-17.e-qual.fr) |
| 11:40.00 | *** join/#asterisk bkw__ (~brian@adsl-69-154-0-136.dsl.tulsok.swbell.net) |
| 11:46.44 | *** part/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca) |
| 11:49.26 | *** join/#asterisk meppl (mephisto@p54AAF17E.dip.t-dialin.net) |
| 11:57.01 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 11:59.18 | *** join/#asterisk cjk (~cjk@80.92.75.120) |
| 11:59.50 | cjk | hi, is there any way to save registration information (astdb) into a mysql db without using realtime module? |
| 12:05.14 | TheEmperor | does festival eat a lot of cpu/memory while running and/or running in the background? |
| 12:07.33 | *** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc) |
| 12:07.33 | *** topic/#asterisk is Asterisk: The Open Source PBX || Astricon Europe -- Madrid, Spain -- June 15-17 || PBX Developers Conference in Chicago Aug 3-5 Register Now! http://www.asterlink.com/cluecon.htm |
| 12:09.49 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 12:10.01 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-221-140.dsl.scarlet.be) |
| 12:12.18 | *** join/#asterisk makhtar (~ageller@mail.bulletinnews.com) |
| 12:13.18 | *** join/#asterisk lehel (~lehel@82.79.20.17) |
| 12:13.22 | lehel | hello |
| 12:14.22 | lehel | pls.. somebody help me to configure zaptel drivers for asterisk |
| 12:14.53 | lehel | i have a TDM400P |
| 12:15.11 | lehel | 4FXO-s |
| 12:15.34 | lehel | chan_zap.c:764 zt_open: Unable to open '/dev/zap/channel': No such device |
| 12:16.35 | tzafrir | jbot, genzaptelconf is a script to general zaptel.conf and zapata.conf snippet for TDM cards, availble from http://tzafrir.org.il/genzaptelconf or from the Rapid zaptel packaged. Ask tzafrir about it. |
| 12:16.36 | jbot | okay, tzafrir |
| 12:16.42 | tzafrir | ~genzaptelconf |
| 12:16.44 | jbot | i heard genzaptelconf is a script to general zaptel.conf and zapata.conf snippet for TDM cards, availble from http://tzafrir.org.il/genzaptelconf or from the Rapid zaptel packaged. Ask tzafrir about it. |
| 12:16.56 | lehel | tzafrir: |
| 12:17.04 | lehel | line 13: Unable to open master device '/dev/zap/ctl' |
| 12:17.04 | lehel | ./genzaptelconf: line 556: ast-cmd: command not found |
| 12:17.47 | tzafrir | jbot, genzaptelconf is also ignore warning about missing ast-cmd. |
| 12:17.48 | jbot | tzafrir: okay |
| 12:17.51 | tzafrir | ~genzaptelconf |
| 12:17.52 | jbot | from memory, genzaptelconf is a script to general zaptel.conf and zapata.conf snippet for TDM cards, availble from http://tzafrir.org.il/genzaptelconf or from the Rapid zaptel packaged. Ask tzafrir about it. ignore warning about missing ast-cmd. |
| 12:18.19 | lehel | May 16 15:17:49 WARNING[20841]: chan_zap.c:764 zt_open: Unable to open '/dev/zap/channel': No such device |
| 12:18.30 | tzafrir | ls -l /dev/zap/channel |
| 12:18.46 | tzafrir | and also: is the module zaptel loaded? |
| 12:19.05 | lehel | crw-r--r-- 1 root root 196, 254 2005-05-16 15:07 /dev/zap/channel |
| 12:19.11 | lehel | the zaptel module isn't loaded |
| 12:19.20 | tzafrir | is asterisk running as root? |
| 12:19.39 | lehel | the lights aren't green |
| 12:19.41 | tzafrir | If not, you probably need a chown/chmod to allow the asterisk user to write there. |
| 12:20.05 | tzafrir | On a standard debian system, /dev/zap should be owned by the group dialout |
| 12:20.15 | lehel | asterisk is running as root.. i think |
| 12:20.49 | tzafrir | hmm: no such device: probably module not loaded |
| 12:21.05 | lehel | May 16 15:17:49 WARNING[20841]: loader.c:440 load_modules: Loading module chan_zap.so failed! |
| 12:21.05 | lehel | yes |
| 12:21.22 | tzafrir | lehel, I meant the kernel module |
| 12:21.29 | tzafrir | /sbin/lsmod |grep zaptel |
| 12:21.56 | lehel | nothing |
| 12:22.07 | tzafrir | zaptel module is not loaded |
| 12:22.13 | lehel | i said that |
| 12:22.18 | *** join/#asterisk coppice (~chatzilla@155.199.17.210.dyn.pacific.net.hk) |
| 12:22.35 | tzafrir | That was an asterisk module in your message before. |
| 12:22.43 | tzafrir | Do you have zxaptel modules built? |
| 12:22.55 | tzafrir | zaptel, that is |
| 12:23.16 | lehel | zaptel.conf is not configured |
| 12:23.40 | tzafrir | without the module zaptel loaded, zaptel.conf has no use |
| 12:23.51 | lehel | ./etc/zaptel.conf : loadzone and defaultzone.. missing something |
| 12:23.56 | tzafrir | zaptel.conf is not used by the kernel space |
| 12:24.06 | lehel | how can i load the zaptel module? |
| 12:24.13 | tzafrir | It is only used by ztcfg |
| 12:24.22 | tzafrir | modprobe zaptel |
| 12:24.34 | tzafrir | or simpler: modprobe wcfxs |
| 12:24.44 | lehel | ohh,, i forgot to tell you: |
| 12:24.54 | lehel | when i compiled zaptel: |
| 12:25.01 | lehel | some.. PPP ... ?? |
| 12:25.30 | lehel | "depmod: ..Unresolved symbols... |
| 12:26.01 | lehel | i read: i should compile zaptel without PPP support |
| 12:26.14 | lehel | to prevent this error |
| 12:26.32 | *** join/#asterisk _omer (dfsdf@202.147.167.213) |
| 12:26.33 | lehel | modprobe: /lib/modules/2.4.27-1-386/misc/zaptel.o: insmod zaptel failed |
| 12:26.35 | tzafrir | what source did you use for zaptel? |
| 12:26.44 | lehel | there it is |
| 12:26.48 | tzafrir | 2.4.27-*1*-386? |
| 12:26.52 | lehel | yap |
| 12:27.00 | lehel | not good? |
| 12:27.12 | tzafrir | you should upgrade your kernel to 2.4.27-2-386 |
| 12:27.14 | ]expic | May 16 12:41:16 NOTICE[31827]: PRI got event: HDLC Abort (6) on Primary D-channe |
| 12:27.14 | ]expic | l of span 1 |
| 12:27.21 | ]expic | can anybody tell me what is wrong? |
| 12:27.36 | tzafrir | When you get the time |
| 12:27.49 | ]expic | it's BRI junghanns card |
| 12:28.06 | tzafrir | It was a kernel fix that broke the ABI of many modules, and hence the name change |
| 12:29.10 | *** join/#asterisk OzJames79 (~opera@CPE20320889-1842-1.gex.ncable.net.au) |
| 12:29.12 | tzafrir | (And for 2.4.27-2-386 or 686 or whatever you'll be ableto use my pre-compiled modules, naturally) |
| 12:30.01 | lehel | ok tzafrir ! |
| 12:30.09 | lehel | i'll do now that upgrade |
| 12:31.22 | lehel | but anyway this PPP support ruins zaptel?.. |
| 12:34.19 | *** join/#asterisk Wi_Fi (~OUT@c-24-127-12-85.hsd1.ca.comcast.net) |
| 12:36.08 | *** join/#asterisk darby_t (~tom@host-ip237-209.crowley.pl) |
| 12:40.14 | tzafrir | what version of asterisk do you use? from where? |
| 12:43.33 | Zeeek | aïe caramba |
| 12:43.45 | RoyK | ehlo |
| 12:43.53 | lehel | tzafrir: from cvs |
| 12:44.32 | tzafrir | oh, then my pre-compiled 1.0.7 won't help you much, I figure :-( |
| 12:44.51 | *** part/#asterisk OzJames79 (~opera@CPE20320889-1842-1.gex.ncable.net.au) |
| 12:45.18 | lehel | and if i remove.. and download the 1.0.7 ?and then compile? |
| 12:45.44 | Zeeek | jbot, learning asterisk using a GUI is like learning sex through masturbation. You'll never get the good stuff. |
| 12:45.45 | jbot | Zeeek: okay |
| 12:45.46 | lehel | but i think the problem is with the zaptel! |
| 12:46.00 | Zeeek | ~learning asterisk |
| 12:46.30 | RoyK | Zeeek: most people start off with masturbation... |
| 12:46.39 | Zeeek | absolutely |
| 12:47.01 | Zeeek | that's what I'm doihng with jbot right now |
| 12:47.10 | RoyK | lol |
| 12:47.27 | tzafrir | ~bot abuse |
| 12:47.28 | jbot | Leave me alone.. I feel abused and molested. |
| 12:47.34 | Zeeek | jbot, RoyK is a good audience |
| 12:47.35 | jbot | ...but royk is already something else... |
| 12:47.40 | Zeeek | ah. |
| 12:47.47 | Zeeek | ~RoyK |
| 12:47.48 | jbot | somebody said royk was someone who hasn't learned that drinking and driving isn't cool |
| 12:47.54 | Zeeek | eewwww |
| 12:48.04 | Zeeek | ~Zeeek |
| 12:48.21 | *** join/#asterisk easimon (~easimon@localhorst.kawo2.RWTH-Aachen.DE) |
| 12:48.37 | RoyK | jbot: no, RoyK is that viking asterisk guru |
| 12:48.38 | jbot | okay, RoyK |
| 12:48.39 | Zeeek | jbot, 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." |
| 12:48.40 | jbot | Zeeek: okay |
| 12:48.46 | Zeeek | ~Zeeek |
| 12:48.47 | jbot | extra, extra, read all about it, 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." |
| 12:49.53 | iCEBrkr | hahah |
| 12:50.34 | coppice | woody allen said masturbation is people who believe sex is much too personal to share with another human being |
| 12:50.45 | Zeeek | he's right, damn it |
| 12:51.01 | ]expic | ca anybody help me with BRI card? |
| 12:51.21 | coppice | why? are they that heavy? :-) |
| 12:51.21 | Zeeek | he laso said death is exaclty like sex except you don't feel nauseus after |
| 12:51.41 | *** join/#asterisk docelm0 (~docelm0@67.106.194.90.ptr.us.xo.net) |
| 12:52.14 | *** join/#asterisk MattH (~matth@noc-wireless.chilitech.net) |
| 12:52.18 | *** join/#asterisk sheez (~scott@66.219.228.65) |
| 12:52.43 | MattH | Hi.. does anyone know of any example anywhere to do this: dial into my * box... enter an access code... and then get prompted for the number to dial and have * dial out... and connect the two calls? |
| 12:53.06 | Zeeek | didn't you ask this yesterday? |
| 12:53.11 | RoyK | coppice: hi |
| 12:53.17 | MattH | Zeeek: no |
| 12:53.18 | coppice | ho |
| 12:53.39 | MattH | Zeeek: I don't believe I've ever asked this question before |
| 12:53.40 | Zeeek | well someone did, too bad you mised it :) |
| 12:53.42 | MattH | lol |
| 12:53.52 | MattH | I know it can be done.. just wondered if there was an example before I go re-inventing the wheel :) |
| 12:53.58 | Zeeek | but thanswer was something like "look at DISA and .call files" |
| 12:54.18 | Zeeek | hint: look at "wakeme" AGI |
| 12:54.45 | *** part/#asterisk sheez (~scott@66.219.228.65) |
| 12:54.53 | Zeeek | I remember someone actually writing a callback from a hotel room at 3AM |
| 12:55.12 | MattH | lol |
| 12:55.22 | Zeeek | the only problem was if they answer "Hotel Paradise, please hold" while you recordeing is asking for your room number |
| 12:55.44 | MattH | ahh disa looks like what I want |
| 12:55.53 | Zeeek | would the callback number vary or be the same? |
| 12:56.15 | Zeeek | for one person or several? |
| 12:56.23 | Zeeek | red or blus? |
| 12:56.23 | MattH | it would vary.. which is why it would be better if the user could call INTO the system and then call out... as all the calls TO the system will be local |
| 12:57.27 | Zeeek | well if the number is limited they could enter a password and get a callback |
| 12:57.36 | MattH | yeah that could work as well |
| 12:57.41 | Zeeek | or even just call and hang up |
| 12:59.21 | Zeeek | wait, "have asterisk connect the two calls" you mean it would call one number, then the other and then connect those two? |
| 12:59.28 | MattH | no sorry |
| 12:59.34 | MattH | like I'm local to the asterisk system but not voip or in the office |
| 12:59.44 | Zeeek | you want a callback with dialtone so asterisk is billed for the call |
| 12:59.49 | MattH | I can call local access number 601-blah... asterisk prompts me to call.. i enter the number... and asterisk connects us |
| 12:59.51 | *** join/#asterisk sheez (~scott@66.219.228.65) |
| 12:59.54 | MattH | basically yeah |
| 13:00.00 | MattH | or DISA... either look like they will work well |
| 13:00.19 | sheez | has anyone in here had any luck with mitel ip sets with sip? |
| 13:00.26 | Zeeek | you want to use POTS from home to use POTS or voip from office? |
| 13:01.33 | Zeeek | definitely, DISA , PIN and you'll have dialtone and can dial any number including vmail |
| 13:02.01 | Zeeek | or add functions before DISA for vmail, intercom, whatever |
| 13:02.17 | MattH | yup that's what I'm looking for =) |
| 13:02.58 | ]expic | CRC error for HDLC frame on card 1 |
| 13:03.03 | ]expic | can anybody helop me what it is? |
| 13:03.10 | ]expic | layer2 is down |
| 13:05.24 | tzanger | it means there was a CRC error in the HDLC frame |
| 13:05.34 | tzanger | it means that the line quality is subpar |
| 13:05.42 | ]expic | tzafrir: yes, what things i should look first? |
| 13:05.45 | tzanger | or there is something causing the driver to miss bits |
| 13:07.34 | ]expic | shared interrupt? |
| 13:08.43 | ]expic | <PROTECTED> |
| 13:09.13 | *** join/#asterisk Blu3 (~david@ip24-250-18-88.ri.ri.cox.net) |
| 13:09.14 | ]expic | strage |
| 13:09.19 | ]expic | i set bit to 1 |
| 13:09.23 | ]expic | to clock from line |
| 13:09.42 | ]expic | span=1,1,3,ccs,ami |
| 13:09.50 | ]expic | span=1,1,3,ccs,ami |
| 13:09.56 | ]expic | span=1,1,3,ccs,ami |
| 13:10.02 | ]expic | sorry |
| 13:10.39 | *** join/#asterisk darth-timeus (darth@200.105.128.61) |
| 13:10.43 | darth-timeus | Hi |
| 13:11.14 | darth-timeus | i still have the one way audio problem, so i try to compile the asterisk with the gcc-3.2.2 |
| 13:11.40 | darth-timeus | but i can't make the zaptel work, because it compile, install, but it don't create |
| 13:11.50 | darth-timeus | the dev/zap directories |
| 13:11.50 | tzanger | ]expic: PLEASE try to understand wtf you're doing |
| 13:12.01 | tzanger | why are you defining the same span 4 times |
| 13:12.20 | Sato1 | darth-timeus, do you have udev? |
| 13:12.27 | Sato1 | darth-timeus, or normal /dev? |
| 13:13.24 | tzanger | and why do you have the LBO set? |
| 13:13.53 | darth-timeus | Sato 1: yes i have udev |
| 13:14.33 | Sato1 | darth-timeus, go to your zaptel directory, and see the README.udev |
| 13:14.53 | Sato1 | ...file |
| 13:16.03 | darth-timeus | i have readed it, and i have included everything in my files, but i can't make it work |
| 13:16.50 | tzanger | ]expic: hello? |
| 13:17.33 | Sato1 | darth-timeus, what does ztcfg -vvv says? |
| 13:18.14 | ]expic | <tzanger>it's nistake |
| 13:18.21 | ]expic | <tzanger>i just paste 4 times |
| 13:18.25 | ]expic | <tzanger>i do it once |
| 13:18.52 | ]expic | <tzanger>that's why i said sorry, cause pasted it 4 times |
| 13:19.08 | darth-timeus | Sato 1: line 0: Unable to open master device '/dev/zap/ctl' |
| 13:20.00 | Sato1 | well, not sure if thats the solution, after i had that, i rebooted my compyter and everything works |
| 13:20.13 | *** join/#asterisk jeffik (~jeffik@Toronto-HSE-ppp3685698.sympatico.ca) |
| 13:21.36 | darth-timeus | Sato 1: ok, i'll give it a try |
| 13:23.24 | *** join/#asterisk SuPrSluG (~SuPrSluG@pool-70-16-33-249.buff.east.verizon.net) |
| 13:24.38 | *** join/#asterisk LoRez (lorez@lorez.staff.freenode) |
| 13:25.03 | *** join/#asterisk lunchbox08 (~geoff@64.128.43.66) |
| 13:26.54 | *** join/#asterisk makkia (~pippo@nat.xsec.it) |
| 13:27.00 | lunchbox08 | Anyone know a way to restart | stop | start the zaptel module as a non root user? |
| 13:27.16 | Sato1 | using manger |
| 13:27.45 | Hali_303 | I would like to download the latest zaptel driver, however, digium's CVS doesnt seem to work! does it work for you? |
| 13:28.17 | Hali_303 | export CVSROOT=:pserver:anoncvs@cvs.digium.com:/usr/cvsroot |
| 13:28.20 | Hali_303 | cvs login |
| 13:28.27 | bublbobl | hi all, I need to record a call for quality testing purpose, is it hard to set * for this or shall I better plug a recorder on my IP-Phone ? |
| 13:28.28 | Hali_303 | and then it waits for 3-4 minutes then timeouts |
| 13:29.03 | *** join/#asterisk lehel (~lehel@82.79.20.17) |
| 13:29.03 | lunchbox08 | how would you restart zaptel via manager? |
| 13:29.09 | lehel | back |
| 13:29.25 | lehel | tzafrir: Linux dev01 2.4.27-2-686 < ok? |
| 13:29.57 | Sato1 | Hali_303, digium.com cvs is working, at least for me |
| 13:30.08 | Hali_303 | satol, with the above lines? |
| 13:30.25 | Sato1 | lunchbox08, see the info about manager on http://www.voip-info.org |
| 13:30.27 | Hali_303 | Sato1, I've just tried it a minute ago an it hangs |
| 13:30.53 | Sato1 | Hali_303, with those lines, actually, just checked a couple minutes ago |
| 13:31.17 | lehel | tzafrir: ?? |
| 13:31.17 | Hali_303 | hmm then what could cause cvs stop working.. |
| 13:31.22 | lunchbox08 | I am looking at it right now... I do not see anything in regarding restarting zaptel... I can do a reload but that does nothing in regards to zaptel... |
| 13:31.50 | *** join/#asterisk SexyKen (~ksandell@66.238.198.222.ptr.us.xo.net) |
| 13:31.55 | SexyKen | Hey guys -- quick question... |
| 13:31.59 | *** join/#asterisk ariel_ (~Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 13:32.06 | Sato1 | lunchbox08, "Action: command" "command: unload chan_zap.so"..... |
| 13:32.12 | SexyKen | ...when some of my phones try to make an outgoing call it says 'No authority found'....any ideas why? |
| 13:32.47 | lunchbox08 | Sato1, Thank you very much... |
| 13:32.48 | Sato1 | SexyKen, could be lot of things |
| 13:33.06 | Sato1 | lunchbox08, no problem |
| 13:33.21 | SexyKen | May 16 07:32:17 WARNING[2513]: chan_iax2.c:6075 socket_read: Call rejected by 64.201.119.146: No authority found |
| 13:33.33 | SexyKen | Sato -- hrm. |
| 13:34.07 | Sato1 | that mainly happends when Dial command does not send the proper user/pass, or the right extension |
| 13:34.43 | SexyKen | Any ideas on what I should do to trouble shoot it? |
| 13:36.18 | Sato1 | not with that information, that just the result of the problem, i think you could paste the Dial command that is causing that |
| 13:36.51 | SexyKen | It doesn't show the actual dial command...just that error. |
| 13:37.18 | *** join/#asterisk yxa (empty@cm162.gamma226.maxonline.com.sg) |
| 13:37.27 | Sato1 | the dial command in your dialplan, to see how you are making the call |
| 13:39.28 | Sato1 | lunchbox08, http://www.voip-info.org/wiki-Asterisk+manager+api |
| 13:42.15 | ariel_ | hello everyone |
| 13:42.37 | Sato1 | morning ariel_ |
| 13:44.02 | darwin35 | why would a inbound exten loop and dial itself back out |
| 13:44.49 | Sato1 | because it is not right configured? |
| 13:44.50 | darwin35 | http://pastebin.ca/11889 |
| 13:46.51 | darwin35 | exten => 8667871709,1,Goto(in-pstn) |
| 13:47.02 | darwin35 | it goes to a macro |
| 13:47.23 | Sato1 | hehe |
| 13:47.31 | darwin35 | yeah it loops |
| 13:48.14 | darwin35 | also I cant seem to make outbound calls |
| 13:48.33 | darwin35 | all the nmbrs other wise work inbound |
| 13:49.06 | Sato1 | i think you could paste the actual dialplan |
| 13:49.16 | newl | valgrind it to death hehe |
| 13:49.22 | darwin35 | its huge |
| 13:49.43 | tzanger | bkw_: I'm signing up for an asterlink acct right now |
| 13:50.28 | RoyK | darby_t: patebin it |
| 13:50.34 | RoyK | pastebin, even |
| 13:51.13 | *** join/#asterisk bkw__ (~brian@adsl-69-154-0-136.dsl.tulsok.swbell.net) |
| 13:51.26 | RoyK | hm |
| 13:51.34 | RoyK | from show memory summary chan_sip.c |
| 13:51.35 | RoyK | <PROTECTED> |
| 13:51.49 | RoyK | that just climbes and climbes and climbes |
| 13:51.56 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 13:52.28 | RoyK | <PROTECTED> |
| 13:52.30 | RoyK | there |
| 13:52.34 | tzanger | wtf |
| 13:52.43 | RoyK | but I have no idea why it's not freed |
| 13:52.44 | tzanger | bkw_: why the fuck do I need a numeric-only password for an account?! |
| 13:52.53 | RoyK | ? |
| 13:52.59 | bkw_ | tzanger, because you can't enter them over da phone |
| 13:53.11 | tzanger | bkw_: blech. ok |
| 13:53.36 | bkw_ | tzanger, soon you'll have the ability to use your asterlink account from any PSTN line too |
| 13:53.42 | tzanger | I see |
| 13:53.49 | bkw_ | oh now he gets it |
| 13:53.50 | bkw_ | hehe |
| 13:53.50 | tzanger | so how do I enter my username then? :-) |
| 13:53.57 | bkw_ | your user_id is 6 digits |
| 13:54.10 | tzanger | do I *need* a tollfree number? |
| 13:54.18 | bkw_ | you don't have to take one |
| 13:54.34 | tzanger | I don't see that option |
| 13:54.39 | bkw_ | RoyK, were are you getting these numbers? |
| 13:54.49 | tzanger | please assign one, use one from below, I want a custom one |
| 13:54.58 | newl | 1-888-tza-nger :) |
| 13:55.02 | bkw_ | tzanger, i'll have the guys fix that then |
| 13:55.07 | tzanger | hahahaha |
| 13:55.13 | *** join/#asterisk OzJames79 (~opera@CPE20320889-1842-1.gex.ncable.net.au) |
| 13:55.18 | bkw_ | it was there a few weeks ago when I tested it.. |
| 13:55.39 | *** part/#asterisk OzJames79 (~opera@CPE20320889-1842-1.gex.ncable.net.au) |
| 13:55.49 | bkw_ | we went to lowes at freakin 7am today |
| 13:55.56 | bkw_ | greg woke me up at 5:45am to go |
| 13:55.57 | *** part/#asterisk lunchbox08 (~geoff@64.128.43.66) |
| 13:56.08 | bkw_ | we spent 678 bucks |
| 13:56.27 | bkw_ | we bought a new front door... |
| 13:56.41 | riksta | that's an expensive front door |
| 13:56.44 | tzanger | bkw_: usually we see the CVS # as a CCV # |
| 13:56.53 | newl | riksta: I was thinking the same. |
| 13:56.59 | *** join/#asterisk Milligan (~support@wkstn6.gnwd-noc.valuelinx.net) |
| 13:57.02 | bkw_ | tzanger, you're kidding me right? |
| 13:57.18 | tzanger | bkw_: no, CCV is what I always have heard |
| 13:57.24 | tzanger | the 3 digit code that is not embossed |
| 13:57.32 | bkw_ | ya but I do not recall what we called it |
| 13:57.33 | bkw_ | hehe |
| 13:57.38 | tzanger | you called it the CVS # |
| 13:57.56 | tzanger | ugh you pop up my info too, damn you |
| 13:57.57 | tzanger | just email it |
| 13:57.57 | ]expic | pplz according BRI problem i solved it |
| 13:58.01 | ]expic | thank you |
| 13:58.03 | ]expic | :) |
| 13:58.55 | bkw_ | hahaha |
| 13:58.59 | bkw_ | stupid i'll fix that too |
| 13:59.05 | bkw_ | it is CVV2 |
| 13:59.16 | tzanger | yeah CVV/CVV2 |
| 13:59.26 | NewSole | morning folks |
| 13:59.33 | bkw_ | foooooooolks |
| 13:59.46 | NewSole | lol |
| 14:00.06 | NewSole | well I could have said morning blolks |
| 14:00.07 | *** join/#asterisk mAsH` (~mAsH@ppp-217-133-150-46.cust-adsl.tiscali.it) |
| 14:00.54 | mAsH` | hi |
| 14:01.50 | tzanger | bkw_: how the fuck is my login incorrect?! |
| 14:02.02 | ]expic | Extension '80075599' in context 'from-pstn' from '21553554' does not exist |
| 14:02.06 | tzanger | I logged in, now it's on a second page with just two form entries: email and password |
| 14:02.11 | ]expic | can anybody explain me why it doesn;'t go to "s"? |
| 14:02.16 | tzanger | and (again, dumbass popup) "login incorrect" |
| 14:04.41 | bkw_ | tzanger, its your user@asterlink.com |
| 14:04.45 | ]expic | > "s" is run when a call comes in and Asterisk does not know the dialed |
| 14:04.45 | ]expic | > number. It does NOT mean "meaning anything coming in through that context |
| 14:04.45 | ]expic | > will start there" |
| 14:04.49 | ]expic | is that true? |
| 14:05.03 | bkw_ | ]expic, it can mean both |
| 14:05.09 | bkw_ | "can" being the keyword |
| 14:05.24 | ]expic | bkw: so should i make "i"? |
| 14:05.35 | ]expic | bkw: cause i moved from fxo to bri |
| 14:05.45 | ]expic | bkw: and now i get unknown extension |
| 14:06.03 | tzanger | bkw_: you might want to make a note of that somewhere, because it's asking for an email addy and I don't have an asterilink.com email addy |
| 14:07.35 | tzanger | bkw_: benshaw@asterlink.com login incorrect |
| 14:09.03 | bkw_ | tzanger, let me see |
| 14:09.12 | bkw_ | are you using the form on the frontpage? |
| 14:09.22 | tzanger | I might be doing something really stupid but it's asking for my email addy so that's what I had been putting |
| 14:09.34 | tzanger | bkw_: no, I entered my user/pass on that and it took me to this other screen |
| 14:09.36 | Sato1 | how accurated would be iptables accounting to measure trafic generated by iax in a call? |
| 14:09.41 | tzanger | bkw_: with "email" and "password" |
| 14:09.47 | bkw_ | let me see |
| 14:10.01 | bkw_ | it takes 5-10 min to get your account to active state |
| 14:10.13 | tzanger | hmm okay it's been that long NOW but I'll wait a little bit more |
| 14:10.31 | tzanger | and the second screen "email" really is "username@asterlink.com" and not my signup email addy? |
| 14:11.01 | bkw_ | well you get an email account with us too |
| 14:11.02 | bkw_ | ;) |
| 14:11.24 | bkw_ | it still shows pending |
| 14:11.27 | bkw_ | let me kick it |
| 14:11.37 | *** join/#asterisk sivana (~sivana@mixdown.ca) |
| 14:11.39 | *** join/#asterisk Seyr (~Seyr@rrcs-24-227-133-226.sw.biz.rr.com) |
| 14:11.45 | Seyr | mornin |
| 14:11.54 | bkw_ | well let me switch computers first |
| 14:11.55 | bkw_ | brb |
| 14:13.09 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 14:13.09 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 14:14.41 | darwin35 | http://pastebin.ca/11893 here is the dial plan |
| 14:15.09 | darwin35 | to go with the loop I was having |
| 14:16.36 | bjohnson | http://bigjohnson.ca |
| 14:16.40 | bjohnson | hehe |
| 14:16.48 | *** join/#asterisk n4y (~frodo7@host-ip237-209.crowley.pl) |
| 14:16.54 | Seyr | anyone here ever do any clustering with *? |
| 14:17.14 | *** join/#asterisk loick (~loick@APuteaux-151-1-52-222.w82-124.abo.wanadoo.fr) |
| 14:17.49 | RoyK | nope |
| 14:18.49 | Sato1 | darwin35, and the loop is when you dial to a toll-free number? |
| 14:19.32 | Zeeek | bjohnson hey, uh notch... |
| 14:20.42 | darwin35 | yes |
| 14:21.05 | *** join/#asterisk bonez39 (~aint@drjones.dsl.xmission.com) |
| 14:21.11 | darwin35 | when you dial the 800 to call in it loops and does not goto the autoattendant like it should |
| 14:21.43 | mAsH` | i have a hfc card in nt mode, i used bri_net as signalling, but when i start * i got this error: |
| 14:22.01 | mAsH` | <PROTECTED> |
| 14:22.06 | mAsH` | anyone can help me ? |
| 14:22.19 | docelm0 | What's the name of the company doing E911 Services? Intrado? |
| 14:22.24 | Sato1 | darwin35, an intoming call? |
| 14:22.31 | darwin35 | yes |
| 14:22.31 | bkw_ | bri_net isn't valid |
| 14:22.36 | bkw_ | pri_net is |
| 14:22.40 | bkw_ | if you're using bristuff |
| 14:22.42 | mAsH` | in zaptel 1.0.6 i use it |
| 14:22.44 | bkw_ | you need to contact those people |
| 14:22.48 | bkw_ | no you don't |
| 14:22.53 | bkw_ | zaptel HAS NEVER done BRI |
| 14:22.59 | bkw_ | unless you use bristuff |
| 14:23.11 | mAsH` | i use bristuff |
| 14:23.14 | bkw_ | in which case you'll have to contact the author of that software |
| 14:23.31 | darwin35 | BKW come to the conf |
| 14:23.36 | darwin35 | I need input |
| 14:24.56 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com) |
| 14:25.19 | Sato1 | darwin35, you got 2 errors |
| 14:25.30 | darwin35 | ? |
| 14:25.34 | mAsH` | bkw_: who? |
| 14:25.41 | darwin35 | what lines |
| 14:25.45 | Sato1 | line 24 and 25, and 27-8 |
| 14:25.47 | Sato1 | 27-28 |
| 14:25.52 | *** join/#asterisk ronn (ronn@host217-46-199-162.in-addr.btopenworld.com) |
| 14:26.08 | Sato1 | dont know really if a repeated priority makes a loop |
| 14:26.23 | Sato1 | but that does not correct anyway |
| 14:28.34 | Sato1 | got it? |
| 14:28.40 | darwin35 | exten => 8667871709,1,Goto(in-pstn) |
| 14:28.48 | darwin35 | thats the line on the server coonf |
| 14:29.11 | RoyK | darby_t: goto(in-pstn,s,1) might work even better :) |
| 14:29.24 | darwin35 | ok |
| 14:29.33 | Sato1 | and much better if the priority is fixed |
| 14:29.47 | RoyK | show application goto |
| 14:29.48 | Sato1 | you got to exten with the same "1" |
| 14:29.49 | RoyK | tamtitam |
| 14:30.14 | *** join/#asterisk bobx (~bobx@lowfreq.trancemitter.org) |
| 14:31.27 | Sato1 | btw, line 28 shows "exten => 8667871709,1,Goto(autoattend)" instead of what you posted here |
| 14:31.42 | Sato1 | it should be "exten => 8667871709,1,Goto(autoattend)" |
| 14:31.45 | Sato1 | oops |
| 14:31.50 | Sato1 | it should be "exten => 8667871709,2,Goto(autoattend)" |
| 14:32.35 | *** join/#asterisk loick (~loick@APuteaux-151-1-52-222.w82-124.abo.wanadoo.fr) |
| 14:32.52 | Sato1 | lines 24 and 25 shows the same mistake |
| 14:32.55 | darwin35 | http://pastebin.ca/11895 |
| 14:34.42 | *** join/#asterisk Nuxi (~nuxi@cust-sdsl-204-250-82-202.bzn-co-i1000-01.bridgeband.net) |
| 14:35.23 | Sato1 | did you fix that priority? |
| 14:35.34 | darwin35 | yes |
| 14:35.46 | Sato1 | now it does not loop |
| 14:36.14 | RoyK | http://bugs.digium.com/view.php?id=4287 |
| 14:36.15 | RoyK | bah |
| 14:38.38 | darwin35 | 9,2,Goto(autoattend,s,1) |
| 14:38.53 | darwin35 | <darwin35> exten => 18667871709,1,dbput(Last/Caller=${CALLERIDNUM}) |
| 14:38.54 | darwin35 | <darwin35> exten => 18667871709,2,Goto(autoattend,s,1) |
| 14:39.34 | Sato1 | that looks better |
| 14:41.23 | darwin35 | it still loops back out |
| 14:41.29 | darwin35 | and dials itself |
| 14:41.59 | *** join/#asterisk makhtar (~ageller@mail.bulletinnews.com) |
| 14:42.17 | ronn | i have been trying to use the peer option in sip.conf to authenticate by host name ... but haven't managed so far |
| 14:42.59 | ronn | even if i specified type=peer .. asterisk keeps on asking for authentication |
| 14:43.10 | docelm0 | Does anyone know a provider of E911 in Canada? |
| 14:43.42 | *** part/#asterisk szw2001 (~vip@218.1.218.187) |
| 14:43.53 | ronn | do i need to use insecure=yes as well ? |
| 14:44.06 | Sato1 | =very |
| 14:45.40 | *** join/#asterisk makhtar (~ageller@mail.bulletinnews.com) |
| 14:45.53 | ronn | Sato1: thanks |
| 14:46.21 | ronn | Sato1: that means any call from that IP is allowed? |
| 14:49.05 | *** join/#asterisk aminorex (~tony@c66.191.69.132.dul.mn.charter.com) |
| 14:50.01 | *** join/#asterisk mikes2277 (~mike@wireless-206.222.58.99.omnilec.com) |
| 14:50.23 | *** join/#asterisk Blackvel (~blackvel@dsl-084-057-126-008.arcor-ip.net) |
| 14:51.40 | mikes2277 | I am using the latest CVS but my * voicemail always cuts off messages after 30 seconds... I have maxmessage=250 in voicemail.conf but it doesn't seem to help, any ideas? |
| 14:51.55 | *** join/#asterisk jefrey (~tmnut@203.115.193.176) |
| 14:52.03 | jefrey | i set autocreeper=no |
| 14:52.10 | *** join/#asterisk mutilator (~animenodv@65.111.201.79) |
| 14:52.18 | mikes2277 | whats that? |
| 14:52.18 | jefrey | but it seems like external peers still can send in traffic without authentication |
| 14:52.28 | jefrey | was there such bug in the older version? |
| 14:52.45 | mikes2277 | on the vm? not that I know of |
| 14:52.54 | mikes2277 | but i never tried |
| 14:53.26 | jefrey | no |
| 14:53.29 | jefrey | chan_sip |
| 14:53.37 | mikes2277 | oops sorry |
| 14:54.19 | mikes2277 | my irc client didnt start sending messages to me until after I sent my first one |
| 14:54.50 | mikes2277 | does anyone know if maxmessage is the correct var to set? |
| 14:55.22 | darwin35 | btw why is asterisk set to a limit of 10 #includelines in extensions.conf |
| 14:55.40 | *** join/#asterisk focks (~craig@nsc66.147.95-93.newsouth.net) |
| 14:56.14 | tzanger | darwin35: it better not have that limitation |
| 14:57.19 | *** join/#asterisk _scat (~scott@private.harmonyip.com) |
| 14:58.54 | *** join/#asterisk tessier_ (~treed@203.210.212.17) |
| 14:59.45 | focks | Can someone take a look at my extensions.conf http://pastebin.com/285131 and show me where I need to put a pause in to give my telco time to take digits correctly? |
| 15:02.10 | *** join/#asterisk grolloj (~chatzilla@slim-eth0.horizonlive.net) |
| 15:02.26 | *** join/#asterisk Gand_DJ (~gandalf@ptr-207-54-104-24.ptr.terago.ca) |
| 15:03.43 | *** join/#asterisk argtor123 (~argtor@206.223.136.211) |
| 15:03.52 | darwin35 | tz ? |
| 15:04.17 | darwin35 | what limitation |
| 15:04.22 | argtor123 | Need some help from anyone who as experience with the EiCon DIVA 4BRI server cards and asterisk |
| 15:04.27 | *** join/#asterisk MacDeath1 (davidn@196.202.248.34) |
| 15:04.45 | MacDeath1 | hi there |
| 15:04.53 | Seyr | anyone here setup * in a cluster? |
| 15:04.54 | argtor123 | anyone who would be able to help? |
| 15:05.01 | darwin35 | tzanger ? |
| 15:05.01 | *** join/#asterisk HA (HeadachesA@wsip-68-99-73-32.tu.ok.cox.net) |
| 15:05.13 | tzanger | what |
| 15:05.27 | *** join/#asterisk Trickyphillips (~Trickyphi@adsl-68-121-21-99.dsl.irvnca.pacbell.net) |
| 15:05.30 | darwin35 | it does |
| 15:05.40 | docelm0 | Seyr, asterisk to my knowlegde cant be setup in a "TRUE" cluster. It can be "clustered" behind a load balancer |
| 15:05.45 | darwin35 | the #includelines |
| 15:06.04 | darwin35 | 1.0.7 has a set amount of #include lines of 10 |
| 15:06.24 | darwin35 | pissed me off |
| 15:06.31 | Seyr | docelm0: so mostly like heartbeat with ldirector or something? |
| 15:06.35 | MacDeath | does anyone here use voipfone.co.uk? |
| 15:06.45 | darwin35 | any more input on my issue |
| 15:06.47 | mutilator | whats a "TRUE" cluster? |
| 15:06.49 | docelm0 | yes check out www.ultramonkey.org |
| 15:06.52 | *** join/#asterisk lehel (~lehel@82.79.20.17) |
| 15:06.56 | lehel | hello |
| 15:07.02 | Seyr | docelmo: i noticed the wiki has a link to OpenSSI as well |
| 15:07.09 | Seyr | docelm0: im familiar with UM |
| 15:07.14 | lehel | zaptel module isn't loading |
| 15:07.18 | focks | signate.com has a clustered product |
| 15:07.27 | docelm0 | Seyr, then you should be good to go.. |
| 15:07.29 | MacDeath | I get the following error message when I try call |
| 15:07.30 | MacDeath | <PROTECTED> |
| 15:07.30 | MacDeath | <PROTECTED> |
| 15:07.41 | MacDeath | lehel : what errors are you getting? |
| 15:07.42 | argtor123 | any Eicon Diva 4BRI card guru's around??? |
| 15:08.07 | Seyr | what about SER? anyone use * behind SER? |
| 15:08.12 | lehel | MacDeath: modprobe zaptel: /lib/modules/2.4.27-2-686/misc/zaptel.o: insmod zaptel failed |
| 15:08.24 | RoyK | argtor123: that must be kapejod |
| 15:08.32 | docelm0 | I have looked into that but found it wouldnt do the job for me |
| 15:08.42 | lehel | MacDeath: ztcfg: line 143: Unable to open master device '/dev/zap/ctl' |
| 15:08.46 | RoyK | argtor123: he wrote chan_capi |
| 15:08.59 | RoyK | or chan_crappy as it is often referred to as |
| 15:09.21 | argtor123 | RoyK: Many thanks |
| 15:09.27 | argtor123 | grin |
| 15:09.33 | *** join/#asterisk bofh42 (~bofh42@pD9EC07D2.dip0.t-ipconnect.de) |
| 15:09.59 | *** join/#asterisk dwmw2 (~dwmw2@nat-pool-stn.redhat.com) |
| 15:10.13 | MacDeath | lehel : does your card appear when booting up |
| 15:10.18 | MacDeath | cat /proc/pci |
| 15:10.45 | MacDeath | <PROTECTED> |
| 15:11.03 | *** join/#asterisk facek_ (faceoff@devel.acdbddh.eu.org) |
| 15:11.05 | facek_ | buba |
| 15:11.12 | lehel | MacDeath: no ;( |
| 15:11.23 | Zeeek | MacDeath on one server only? |
| 15:12.01 | *** join/#asterisk wasabi_ (~wasabi@207.55.180.100) |
| 15:12.20 | *** join/#asterisk [Lamer] (Lamer@221.128.88.75) |
| 15:12.34 | *** join/#asterisk doolph (doolph@200.46.148.35) |
| 15:12.36 | lehel | MacDeath: the lights are off |
| 15:12.52 | MacDeath | Zeeek : yeah |
| 15:12.54 | darwin35 | http://pastebin.ca/11897 ok current extensions.conf |
| 15:12.59 | MacDeath | voipfone.co.uk |
| 15:13.10 | MacDeath | the only SIP server i attach to |
| 15:13.18 | darwin35 | still having issues with 1800 inbound and all outbound |
| 15:13.19 | Zeeek | get a free FWD account |
| 15:13.19 | doolph | hSegmentation fault |
| 15:13.19 | doolph | [root@www asterisk]# Warning, flexibel rate not heavily tested! |
| 15:13.20 | doolph | Ouch ... error while writing audio data: : Broken pipe |
| 15:13.20 | *** join/#asterisk dalabera (~Dalabera@mail.pmrtechnologies.com) |
| 15:13.24 | dalabera | hello guys |
| 15:13.58 | doolph | how can I disable music on hold |
| 15:14.21 | Zeeek | MacDeath a free SIP acct somewhere else would allow you to test and prove it's not from your end |
| 15:14.38 | MacDeath | Zeeek : is fwd SIP or IAX? |
| 15:14.43 | Zeeek | either or both |
| 15:14.57 | Zeeek | didn't you say you use SIP? |
| 15:15.15 | MacDeath | i use IAX for FWD |
| 15:15.17 | MacDeath | and that is working |
| 15:15.29 | Zeeek | doesn't mean much if your problem is with SIP |
| 15:15.35 | MacDeath | yeah |
| 15:15.40 | Zeeek | get e sipgate acct or something |
| 15:15.55 | Zeeek | there's no lack of free SIP accounts I must have 20 of them |
| 15:17.22 | Zeeek | iptel.org, sipgate.de, inphonex.com, like2foe.com, mycom.it, e164.org... to name a few |
| 15:17.55 | Zeeek | MacDeath could just be that they are circuit busy |
| 15:18.05 | Zeeek | no room for your call |
| 15:18.12 | MacDeath | Zeeek : its possible |
| 15:18.22 | lehel | depmod: *** Unresolved symbols in /lib/modules/2.4.27-2-686/misc/zaptel.o |
| 15:18.23 | MacDeath | thats why i was asking if anyone has used them |
| 15:18.30 | Zeeek | unless the condition lasts for like three weeks |
| 15:19.14 | Zeeek | hey, by the way: 42 |
| 15:19.15 | lehel | somebody nows about PPP support?.. |
| 15:19.47 | Zeeek | do you guys know that Hitchiker's Guide is on the BBC now and you can listen on the web anytime? |
| 15:19.47 | lehel | i'm compiling zaptel without PPP support.. but still 'have problems |
| 15:21.00 | bublbobl | <PROTECTED> |
| 15:21.08 | *** join/#asterisk christo (~chris@office.enovi.com) |
| 15:21.25 | bublbobl | Hitchiker's Guide to galaxy, or to * ;-) |
| 15:21.30 | doolph | does chan_sip.so is important? |
| 15:21.35 | Zeeek | galaxy |
| 15:21.47 | Zeeek | doolph depends on whether you use sip |
| 15:22.33 | doolph | yes i want to use sip |
| 15:24.59 | Zeeek | maybe you can avoid loading moh |
| 15:25.19 | Zeeek | are you sure you have the right version of moh? |
| 15:25.34 | doolph | nope |
| 15:25.43 | lehel | 0 |
| 15:25.48 | Zeeek | mp3player I meant |
| 15:25.49 | *** join/#asterisk sean (~sean@iconoclast.caedmon.net) |
| 15:25.57 | Zeeek | have you read yhis page? http://www.voip-info.org/wiki-Asterisk+cmd+MP3Player |
| 15:26.29 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 15:26.29 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:27.03 | Zeeek | <PROTECTED> |
| 15:29.49 | RoyK | ~lart Zeeek |
| 15:29.55 | *** join/#asterisk Trickyphillips (~Trickyphi@adsl-68-121-21-99.dsl.irvnca.pacbell.net) |
| 15:30.01 | doolph | if I dont have any hardware, ztdummy, usb, i dont need zaptel right? |
| 15:30.30 | mmlj4 | you're just doing SIP, doolph? then no. |
| 15:30.40 | doolph | cool |
| 15:30.52 | doolph | I only need SIP |
| 15:31.06 | doolph | and SIP or H323 trunks |
| 15:31.27 | docelm0 | ~google my_butt |
| 15:33.11 | Zeeek | ~seen AnyGoodMovies |
| 15:33.15 | jbot | i haven't seen 'anygoodmovies', Zeeek |
| 15:33.21 | Zeeek | me neither |
| 15:33.36 | mmlj4 | can someone please try connecting to my box? sip:87@sip.joeykelly.net |
| 15:44.07 | *** join/#asterisk glock (~glock@wbs-196-2-117-219.wbs.co.za) |
| 15:46.03 | *** join/#asterisk lehel (~lehel@82.79.20.17) |
| 15:46.04 | glock | Anyone know if AMP support ZAP channels yet? |
| 15:46.34 | *** join/#asterisk syle (~blah@wnpgmb02dc1-61-40.dynamic.mts.net) |
| 15:46.41 | wasabi_ | So how do you all manage SIP addresses? They are email form... do you unify them with email addresses? username@domain.com... and address them internally with ext@box.domain.com? |
| 15:46.49 | wasabi_ | Or do you all just use ext@box.domain.com for everything |
| 15:48.01 | MacDeath | Zeeek |
| 15:48.12 | MacDeath | it seems to work now |
| 15:48.29 | MacDeath | for out going calls |
| 15:48.51 | MacDeath | but incoming calls give me this |
| 15:49.05 | MacDeath | May 16 17:39:46 NOTICE[409]: chan_sip.c:7288 handle_request: Failed to authenticate user |
| 15:51.03 | [Lamer] | Hi, Zeeek how do you split large extensions.conf? I will try not to use realtime |
| 15:52.16 | darwin35 | ok mine is not working on outbound |
| 15:54.10 | bjohnson | [Lamer]: include |
| 15:57.10 | bublbobl | bybye all |
| 15:59.25 | Zeeek | as I was saying earlier : http://www.bbc.co.uk/radio4/hitchhikers/ |
| 15:59.42 | Zeeek | you can listen (or record) the episodes of the series |
| 16:03.17 | ellvis | Zeeek:) to you have a towel with you?:) |
| 16:03.23 | ellvis | to=do |
| 16:03.23 | *** join/#asterisk oej (~oej@192.36.80.8) |
| 16:04.08 | Zeeek | of course I never go anywhere without a towel |
| 16:05.00 | *** join/#asterisk ManxPower (~eric@dsl-209-205-172-111.i-55.com) |
| 16:05.53 | HA | is there anything better than festival for tts that is also free? |
| 16:06.02 | Zeeek | no |
| 16:06.32 | Zeeek | there is something that costs $29/voice or something like that |
| 16:06.32 | HA | Think the Vogons might have something? |
| 16:07.00 | Zeeek | "it's the drizzle that makes me morose" |
| 16:07.01 | *** join/#asterisk jets (~brian@guardian.pmt.org) |
| 16:07.02 | Qwell | You'll need to connect a babelfish to your trunk |
| 16:07.02 | darwin35 | ok think I fixed the outbound calls now back to the inbound 866 issue |
| 16:07.48 | Seyr | HA: Cepstral has a real good one, but its like $30.00US per voice |
| 16:08.12 | Zeeek | that's the one |
| 16:08.24 | Zeeek | I have that, it's ok but tedious after a few tests |
| 16:08.32 | MacDeath | May 16 18:07:26 NOTICE[651]: chan_sip.c:7288 handle_request: Failed to authenticate user |
| 16:08.40 | Zeeek | A really good one would costs thousands |
| 16:08.44 | Seyr | I've heard Festival is a LOT better if you take the time to tweak it |
| 16:09.21 | Zeeek | MacDeath that's fairly clear, the area of the problem |
| 16:09.22 | MacDeath | Zeeek : I have insecure=very |
| 16:09.37 | HA | well, i've tweaked it to use a multisyn voice but it only works if i pre-record the files because of how long it takes to load the multi-syn voice. |
| 16:09.39 | MacDeath | thats for incoming calls |
| 16:09.46 | Zeeek | oh you need "insecure=very, very, verys, no SOOO insecure" |
| 16:10.38 | MacDeath | lol |
| 16:10.43 | MacDeath | ummm |
| 16:10.49 | MacDeath | it registers fine |
| 16:10.50 | *** join/#asterisk tzanger (~tzanger@mixdown.ca) |
| 16:10.52 | MacDeath | i can call out now |
| 16:11.40 | Zeeek | so call them and ask! |
| 16:11.46 | Zeeek | they have a great support team I'm sure |
| 16:11.52 | MacDeath | lol |
| 16:11.54 | MacDeath | hardly |
| 16:11.57 | Zeeek | not? |
| 16:12.03 | MacDeath | they reply to emails |
| 16:12.08 | MacDeath | in about 24 hours time |
| 16:12.15 | Zeeek | that's something novel in this biz |
| 16:12.30 | basilioM | hi.. |
| 16:12.47 | basilioM | does any one have good experience with H323 and asterisk? |
| 16:12.50 | Zeeek | pastebin your sip setup MacDeath |
| 16:12.51 | HA | whats the syntax for text2wave? i swear i just did this and now its not working. |
| 16:13.12 | *** join/#asterisk _omer (dfsdf@202.147.167.213) |
| 16:13.57 | *** join/#asterisk lehel (~lehel@82.79.20.17) |
| 16:14.11 | _omer | I am using SIP, how I can check what digits my carrier is sending ? |
| 16:14.15 | lehel | when i'm "make"ing asterisk: |
| 16:14.17 | lehel | make[1]: *** [chan_zap.o] Error 1 |
| 16:14.47 | MacDeath | i will do that now |
| 16:14.50 | Zeeek | <PROTECTED> |
| 16:14.58 | Zeeek | text2wave^^^^^^ |
| 16:15.33 | *** join/#asterisk felipeao (~felipeao@200.146.100.6) |
| 16:16.51 | _omer | I am using SIP, how I can check what digits my carrier is sending ? |
| 16:17.14 | blitzrage | from the CLI? |
| 16:17.20 | _omer | yes |
| 16:17.41 | *** join/#asterisk Inv_arp (junya@adsl-3-237-164.mia.bellsouth.net) |
| 16:18.06 | felipeao | Boards with the INTEL 537 (Ambient MD3200) chipset can be used as an FXO card. Does any1 know of any other cheap board with a similar chipset that can also be used as an FXO card??????? |
| 16:18.53 | blitzrage | cd /etc/asterisk ; add 'debug' without quotes to the console => line in logger.conf ; perform a 'logger reload' at the CLI ; sip debug peer <carrier_peer> ; view the DTMF digits that get sent |
| 16:19.11 | blitzrage | felipeao: no, its dependent on the chipset, and those are the ones you need |
| 16:19.34 | darwin35 | got outbound workking |
| 16:19.39 | glock | Anyone know if AMP works with asterisk cvs? |
| 16:19.48 | darwin35 | got directline in working |
| 16:19.53 | _omer | thanks blitzrage ....I do it and let you know.. |
| 16:20.04 | felipeao | blitzrage, I know it depends on the chipset, but I would like to know if I may use any other chipsets besides INTEL`s 537 |
| 16:20.12 | blitzrage | _omer: you don't have to let me know :) |
| 16:20.36 | felipeao | blitzrage, I ask that cause Ive heard of people using Motorolas chipset |
| 16:20.46 | _omer | :) ... |
| 16:20.58 | felipeao | to be precise, Morotola`s 62802-51 |
| 16:21.05 | felipeao | Mrotola=Motorola |
| 16:21.07 | blitzrage | felipeao: nope |
| 16:21.43 | blitzrage | felipeao: Ambient/Intel MD3200 chipset - thats it |
| 16:21.59 | felipeao | blitzrage, ok, thanks! |
| 16:22.13 | *** join/#asterisk Syncros (~sysop@noc.routermonkey.net) |
| 16:22.17 | blitzrage | np |
| 16:22.26 | MacDeath | Zeeek : http://pastebin.com/285173 |
| 16:24.02 | Zeeek | here's a suggestion of the top of what passes for "my head", use the name of the provider in the peer entry unless you expect to only have one |
| 16:24.34 | Zeeek | I have never used that register syntax but I guess it works? |
| 16:24.46 | *** join/#asterisk Uther_P (~uther_p@66.180.120.83) |
| 16:25.02 | Zeeek | you may need or want to use two pers, one for incoming and one for outgoing |
| 16:25.05 | Zeeek | peers |
| 16:25.35 | Zeeek | some people insist that type=friend is BAD in this situation |
| 16:25.53 | MacDeath | kay, thanks |
| 16:25.57 | MacDeath | i will change that |
| 16:26.10 | Zeeek | fromuser may not be 'welcome' here |
| 16:26.30 | Zeeek | don't they give a config example? or it that it? |
| 16:27.23 | *** join/#asterisk CoffeeIV (~rristroph@mail.airlinksystems.com) |
| 16:29.00 | lehel | line 0: Unable to open master device '/dev/zap/ctl' |
| 16:29.23 | dalabera | Guys, With have one Avaya IP office and 1 asterisk box, the problem I have is that All calls are received from the avaya and transfer to *, but this calls more than 50% are being received with echo and haven't a solution for this, any ideas or clues?? |
| 16:29.58 | Zeeek | MacDeath here is an example I founf that works with the hardest one I have had to get working |
| 16:30.01 | Zeeek | http://pastebin.ca/11904 |
| 16:30.40 | Zeeek | this was one of the few times I had to screw around for a long time to get this working |
| 16:30.49 | Zeeek | the rest of my peers entries are very short |
| 16:30.54 | MacDeath | im trying that now |
| 16:31.05 | MacDeath | is the insecure option per entry? |
| 16:31.13 | Zeeek | not the two username and fromuser that I said you shouldn't use... |
| 16:31.20 | Zeeek | yes I'd do that |
| 16:31.33 | Zeeek | otherwise everything is insecure ..... |
| 16:31.37 | shido6 | MacDeath, http://pastebin.com/285180 |
| 16:31.46 | shido6 | Zeeek, http://pastebin.com/285180 |
| 16:32.00 | Zeeek | except he wants to be called |
| 16:32.07 | shido6 | if you want to be called |
| 16:32.08 | shido6 | then |
| 16:32.39 | *** join/#asterisk outtolunc (outtolunc@adsl-66-218-53-170.dslextreme.com) |
| 16:34.02 | shido6 | http://pastebin.com/285182 |
| 16:34.08 | shido6 | MacDeath, http://pastebin.com/285182 |
| 16:34.15 | shido6 | Zeeek, http://pastebin.com/285182 |
| 16:34.33 | *** join/#asterisk bannerman (~bannerman@209.216.176.43) |
| 16:34.36 | doolph | hey |
| 16:34.40 | doolph | i am getting this error |
| 16:34.41 | doolph | Warning, flexibel rate not heavily tested! |
| 16:34.47 | doolph | Ouch ... error while writing audio data: : Broken pipe |
| 16:34.47 | doolph | Segmentation fault |
| 16:34.52 | doolph | what is it? |
| 16:34.56 | Qwell | an error |
| 16:35.03 | MacDeath | shido6 : thanks, trying that now |
| 16:36.18 | MacDeath | shido6 : you dont have a username at all there? |
| 16:37.24 | Zeeek | doolph there are several messages about this on the mailing list |
| 16:37.36 | *** part/#asterisk DFT (~dft@CPE0040050149d6-CM00080d77ae83.cpe.net.cable.rogers.com) |
| 16:38.26 | *** join/#asterisk braniff (~44r@braniff.user) |
| 16:38.30 | Zeeek | or look here: http://www.voip-info.org/tiki-index.php?page=Asterisk+config+musiconhold.conf |
| 16:38.41 | CoffeeIV | This is kind of OT and a long shot but I've asked everywhere else: I have an ADIT 600, setting it up for first time. I get nothing on RS-232 and it does't DHCP or use a default IP. is there a trick ? better docs than on CarrierAccess's site ? |
| 16:40.27 | braniff | about how long does it take to set up asterisk using a normal USA phone line with 3 extensions on the inside ? |
| 16:41.21 | Zeeek | need more info |
| 16:41.41 | HA | braniff: 5 to 10 minutes maybe more if you want more. |
| 16:41.42 | *** join/#asterisk ManxPwr (~eric@dsl-209-205-172-111.i-55.com) |
| 16:42.03 | _omer | exten => 444111,1,GOTO(incoming-language,s,1) <------ how do I know that my carrier is really sending 444111? |
| 16:42.08 | *** join/#asterisk [Lamer] (Lamer@221.128.97.85) |
| 16:42.15 | *** join/#asterisk pashah (~root@194.67.183.7) |
| 16:42.30 | *** part/#asterisk pashah (~root@194.67.183.7) |
| 16:42.46 | Nuxi | Actually, it can take hours to figure out how to get people in this room to do it for you. |
| 16:42.52 | Zeeek | if it doesn't branch yiou'll know |
| 16:42.56 | braniff | HA, that's much less time (and work) than i would have expected ! |
| 16:43.01 | Zeeek | Nuxi heh |
| 16:43.14 | *** part/#asterisk Seyr (~Seyr@rrcs-24-227-133-226.sw.biz.rr.com) |
| 16:43.34 | [Lamer] | Zeeek: my man, can you show me a simple math calculation in dialplans |
| 16:43.35 | HA | braniff: i setup a test box from scratch, having never used * before, in less than 5 minutes, and had it working. |
| 16:43.51 | Zeeek | you mean like 2+2? |
| 16:43.55 | [Lamer] | Zeeek: the Math thing is funny to me |
| 16:44.07 | _omer | and for me its chemistry :) |
| 16:44.14 | Zeeek | never used it |
| 16:44.20 | *** join/#asterisk eKo1 (~bernd@metrored-gw.tropicohn.com) |
| 16:44.22 | doolph | Zeeek how can I disable MusicOnHold? |
| 16:44.23 | focks | what issues arrise with analog lines and asterisk? is it slow dialing? is the quality noticeably worse than PRI? |
| 16:44.33 | doolph | since I dont have ztdummy or zap hardwares |
| 16:44.34 | [Lamer] | Zeeek: say SetVar(FOO=5) |
| 16:44.35 | eKo1 | analog sucks |
| 16:44.50 | [Lamer] | Zeeek: and I want to increment the FOO by 3 |
| 16:44.56 | HA | analog is anal? |
| 16:45.02 | focks | eKo1 can you be a little more specific |
| 16:47.09 | *** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc) |
| 16:47.09 | *** topic/#asterisk is Asterisk: The Open Source PBX || Astricon Europe -- Madrid, Spain -- June 15-17 || PBX Developers Conference in Chicago Aug 3-5 Register Now! http://www.asterlink.com/cluecon.htm |
| 16:47.17 | doolph | Zeeek now the problem is just Segmentation fault |
| 16:47.26 | doolph | what can cause it |
| 16:47.28 | braniff | how much processing power does * require for the simple setup i described ? does it require a dedicated box ? |
| 16:47.30 | Zeeek | Lamer it has to do with Setvar, ${foo} and + 3 |
| 16:47.55 | focks | eKo1 i have TDM400 with FXS/FXO modules. is that the crappy equipment you refer to? |
| 16:48.07 | eKo1 | Yes, I have one of those. It sucks. |
| 16:48.20 | focks | that doesn't sound promising |
| 16:48.27 | HA | dedicated box would be good, but if you are only doing 3 internal extensions, and using SIP internal Extensions then you could get away with something small. our test box might be a p3 400 if that. |
| 16:48.40 | eKo1 | Well, analog sucks in general. |
| 16:48.43 | eKo1 | Stick to digital |
| 16:48.45 | [Lamer] | Zeeek: yeah but I didn't get it done right I tried Math(FOO,${FOO}+3) |
| 16:48.51 | eKo1 | like PRI |
| 16:49.03 | focks | eKo1 as i said, we do have a PRI but i have to demo it with analog before he'll let me switch it |
| 16:49.30 | Zeeek | I havent' used math yet but setvar(foo=${foo}+3) sounds reasonable |
| 16:49.31 | eKo1 | I guess for demo reasons, it should be OK. |
| 16:49.41 | eKo1 | But for production, you'll suffer. |
| 16:50.02 | focks | that's why i won't use analog in production |
| 16:50.14 | focks | i'm just asking what to look for in analog that will go away with digital |
| 16:50.18 | Zeeek | analog has such a warm sound, like with vinyl |
| 16:50.36 | Zeeek | and the echo is so reassuring to know that you're alive |
| 16:50.58 | Zeeek | and when the canceller kicks in you can tell and all :) |
| 16:51.19 | shido6 | braniff, what are you trying to do? |
| 16:51.34 | focks | Zeeek is dialing slow with analog? |
| 16:52.04 | Zeeek | braniff I have three regular phones and two lines. You could do 3+1 phont line with one TDM400 card and 3xFXS and 1 FXO |
| 16:52.11 | Zeeek | focks not here |
| 16:52.22 | Zeeek | dialing used to be slow but it's instant now |
| 16:52.32 | Zeeek | Answering can be slow though |
| 16:52.36 | focks | Zeeek must you send a wait or anything like that before your digits? |
| 16:52.44 | *** join/#asterisk Bentley (~Bentley@S01060080c8135e6a.cg.shawcable.net) |
| 16:52.52 | Zeeek | never |
| 16:52.54 | *** join/#asterisk Hmmhesays (negative3k@66.173.103.108) |
| 16:53.05 | Zeeek | but that would depend on your telco, no? |
| 16:53.13 | focks | i imagine so |
| 16:53.30 | Zeeek | well our equip is relatively modern here now |
| 16:53.36 | [Lamer] | Zeeek: err it seems to work now |
| 16:53.38 | focks | Zeeek in my initial test with *@home, it was about 50/50 whether you'd get a successful call or receive "if you'd like to make a call, please hang up..." |
| 16:53.39 | Zeeek | a few years ago it was hopelessly old |
| 16:53.52 | Zeeek | drop @home |
| 16:54.08 | Zeeek | ~Zeeek |
| 16:54.09 | jbot | i heard 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." |
| 16:54.09 | focks | i have |
| 16:54.19 | braniff | Zeeek, where is the best place to buy the equipment you mentioned ?? |
| 16:54.24 | Zeeek | where are you? |
| 16:54.27 | braniff | usa |
| 16:54.28 | _omer | lol & jbot |
| 16:54.32 | focks | thevoipconnection |
| 16:54.42 | Zeeek | first check digium site for "retail" price |
| 16:54.53 | Bentley | hello. I'm trunking 2 * boxes via iax2. As soon as I add trunk=yes to iax.conf at both ends, i can no longer make calls (iax2 debug shows a "Subclass: INVAL"). Both boxes have Digium cards. Any ideas? |
| 16:54.59 | Zeeek | I have ordered from voipsupply.com |
| 16:55.05 | focks | them too |
| 16:55.08 | _omer | any one from india? |
| 16:56.31 | *** join/#asterisk pashah (~pashah@ns.itconnection.ru) |
| 16:56.42 | pashah | hello all |
| 16:57.56 | _omer | exten => 444111,1,GOTO(incoming-language,s,1) <------ how do I know that my carrier is really sending 444111? |
| 16:58.40 | Uther_P | _omer: if it goes into incoming-language |
| 16:59.05 | pashah | what is the best way to start if I need to register the .start. of the call (ie when the call have) arrived in cdr? agi? |
| 16:59.19 | pashah | start reading i mean |
| 16:59.32 | *** join/#asterisk abefroman (~abefroman@h-67-103-145-2.chcgilgb.covad.net) |
| 16:59.53 | _omer | Uther_P : yes , If I send 444111 through my IP Phones...it works.... |
| 16:59.55 | docelm0 | omer you could debug your messages |
| 17:00.48 | Zeeek | Bentley you have zaptel hw ? |
| 17:00.59 | _omer | console => debug . . . .now what? I dont see |
| 17:01.12 | _omer | :( |
| 17:01.13 | Bentley | Both boxes have digium cards, and modules are loaded, yes |
| 17:01.18 | abefroman | Im looking at Asterisk for a possible replacement phone system. I have a number of remote offices and telacommuters. Can tellacommuters take a sip phone home with them and use it as if they were in the office ? |
| 17:01.36 | docelm0 | no sip debug or iax debug |
| 17:01.39 | *** join/#asterisk harryvv (~leonardo@S010600a0c93f6f7e.vs.shawcable.net) |
| 17:01.41 | Zeeek | Bentley i shot my wad then |
| 17:01.49 | Uther_P | _omer: what do you mean by sending... you mean as the number dialed, or after the call is answered, as digits entered? |
| 17:01.56 | Bentley | Zeeek: thx - i'll try -users |
| 17:02.05 | mishehu | bah. |
| 17:02.06 | Zeeek | figure of speech, hehe |
| 17:02.12 | docelm0 | abe yes |
| 17:02.13 | mishehu | finally finalized the PRI order. |
| 17:02.35 | mishehu | now I won't be wasting $600 on the t110p |
| 17:02.37 | mishehu | heh |
| 17:02.42 | _omer | <Uther_P> : before answering the call....... |
| 17:02.50 | _omer | <Uther_P> : I mean.....before answering the call....... |
| 17:03.00 | HA | anybody know how to make festival load a voice and keep it loaded? |
| 17:03.13 | Zeeek | keep me loaded, please |
| 17:03.20 | Zeeek | I'll sing for you |
| 17:03.24 | Uther_P | _omer: 444111 is the did? |
| 17:03.38 | abefroman | docelm0: What type of call quality can I expect and will all the features work? Will I have to expose astersisk to the internet or are you talking VPN ? |
| 17:03.57 | HA | Zeeek: if you can stay loaded into festival and provide realtime tts with no delay then you've got a deal. |
| 17:04.15 | pashah | what is the proper way to grab asterisk NOTICE messages with agi? |
| 17:04.15 | docelm0 | you can use VPN or expose to the net. Your choice. I recommend using g729 |
| 17:04.26 | docelm0 | I have it setup like this for my office where I work now |
| 17:04.43 | _omer | <Uther_P> and <docelm0> ..thanks I got it :D ... |
| 17:04.52 | Zeeek | realtime tts with no delay.... muhahahah |
| 17:05.51 | harryvv | Anyone know of a downloadable script that would probe warning and error message and triger a sound file? |
| 17:06.04 | Nuxi | pashah, I believe you need to use the manager api to get NOTICE messages. |
| 17:06.15 | Uther_P | harryvv: wouldn't be that hard to write one |
| 17:06.38 | harryvv | in what Uther_P :) |
| 17:06.41 | Uther_P | perl |
| 17:06.46 | harryvv | yea |
| 17:06.46 | docelm0 | VB! |
| 17:06.52 | Uther_P | haha |
| 17:06.54 | Zeeek | grep |
| 17:06.56 | Uther_P | don't write in vb |
| 17:07.00 | docelm0 | php! |
| 17:07.27 | Nuxi | cobol! |
| 17:07.52 | pashah | Nuxi: that means there is not way I can do that with agi? I think if I will be using manager api heavyly it will cause pain, am i not right? |
| 17:08.10 | HA | harryvv: c#.net using mono and something else painful would be good to use. |
| 17:08.28 | harryvv | on linux? that is a windows product |
| 17:08.31 | Uther_P | harryvv: while (<STDIN>) { $blah .= $_; } if ($blah =~ /message you are looking for/i) { `mpg123 soundfile`; } |
| 17:08.33 | Nuxi | pashah, I use the manager api from my agis without a problem. |
| 17:08.43 | Uther_P | C# blows nutz |
| 17:08.48 | HA | harryvv: c#.net can be used on linux with mono. |
| 17:09.17 | abefroman | Anyone here use Cisco 7940 phones with asterisk? How well do they work? What features are available ? |
| 17:09.28 | harryvv | where are the cli messages stored anyway /var/log/asterisk |
| 17:09.34 | Uther_P | they aren't |
| 17:09.35 | Uther_P | heh |
| 17:09.37 | pashah | Nuxi: wount i be to rude if I will ask you an example of agi for that? |
| 17:09.46 | pashah | s/to/too/ |
| 17:10.00 | harryvv | i hear lots of good things about 7940s |
| 17:10.06 | harryvv | but thay are pricy |
| 17:10.15 | shido6 | if you can afford the 40 |
| 17:10.17 | shido6 | get the 60 |
| 17:10.30 | shido6 | I have a crisco 7960 here and I love it |
| 17:10.35 | abefroman | harryvv: we have an existing system with about 40 stations and want to use them |
| 17:10.36 | Uther_P | crisco, heheh |
| 17:10.50 | shido6 | you need 40 crisco's, abefroman ? |
| 17:10.53 | shido6 | they work great |
| 17:11.05 | shido6 | 6 lines per phone |
| 17:11.08 | shido6 | great speakerphone |
| 17:11.10 | Nuxi | pashah, I believe there are examples in phpagi version 2. |
| 17:11.10 | Uther_P | are you saying crisco on purpose? :P |
| 17:11.15 | shido6 | yes |
| 17:11.23 | Uther_P | heh |
| 17:11.23 | pashah | Nuxi: thanks |
| 17:11.38 | shido6 | I had a barbietone which i used and tried to break to get the crisco |
| 17:11.43 | shido6 | but the barbietone just wouldnt break |
| 17:12.19 | rob- | try unplugging it during a firmware upgraxde:-) |
| 17:12.24 | rob- | upgrade |
| 17:12.28 | abefroman | shido6: we have 40 now we may buy more but I wanted to know if this is an option if we go with asterisk .. sounds like they work fine ! |
| 17:12.36 | rob- | that broke my ata286 |
| 17:13.15 | *** join/#asterisk Patrick^ (~patrickm@birch4.mountaincable.net) |
| 17:13.25 | harryvv | abo, any problems with those phones? |
| 17:14.04 | Nuxi | pashah, there are no good examples of exactly what you want, but you should be able to use add_event_handler to do that. |
| 17:14.41 | pashah | Nuxi: thanks for the pointer |
| 17:14.54 | Nuxi | Let me know if you get stuck in the process somewhere. |
| 17:15.44 | *** join/#asterisk juanjoc (~juanjoc@200.73.189.82) |
| 17:15.45 | pashah | Nuxi: ok |
| 17:15.51 | harryvv | Anyone here recomend a voip supply reseller on the west coast? |
| 17:16.19 | *** join/#asterisk lehel (~lehel@82.79.20.17) |
| 17:16.24 | RaYmAn-Bx | anyone aware of any recent asterisk CVS-HEAD changes that could cause <TAB>-complete to stop working in the cli? |
| 17:16.27 | lehel | hey |
| 17:16.28 | Qwell | its a shame newegg doesn't sell voip hardware |
| 17:16.47 | doolph | how can I reinstall * from scratch |
| 17:16.48 | lehel | <PROTECTED> |
| 17:16.57 | harryvv | newegg is excelent |
| 17:16.59 | lehel | <when trying to run asterisk |
| 17:17.09 | harryvv | negegg is Very fast at shipping parts. |
| 17:17.13 | lehel | i think i have problems with zaptel;(( |
| 17:17.14 | harryvv | newegg :) |
| 17:17.17 | Qwell | harryvv: If I pay the $2.99 rush processing, with ground shipping, I have my shit the next day |
| 17:17.29 | Qwell | as long as I make the order by about noon |
| 17:17.33 | lehel | zaptel module isn't loading!.. why? |
| 17:17.40 | HA | Digium should use Newegg for selling the cards. |
| 17:17.48 | HA | When is Digium gonna have a DS3 card? |
| 17:17.54 | Qwell | soon |
| 17:18.03 | HA | how soon? |
| 17:18.10 | Qwell | sales@digium.com |
| 17:18.13 | Qwell | ask away |
| 17:18.40 | *** join/#asterisk DrWho17 (~MIKE@mike-new.tc3net.com) |
| 17:18.44 | lehel | i have a TDM400P .. 4 fxo's.. need help configuring my Wildcard! |
| 17:19.10 | shido6 | lehel, whats up? |
| 17:19.14 | lehel | line 0: Unable to open master device '/dev/zap/ctl' |
| 17:19.20 | Qwell | udev? |
| 17:19.30 | Qwell | README.udev |
| 17:20.01 | harryvv | lehel, go back and read the howtos. |
| 17:20.10 | docelm0 | I dunno DS3 in a PC? The TDM 4 T1/E1 card has enough trouble with its resources.. Im kinda hesistant to even think about a DS3 card |
| 17:20.35 | harryvv | doc, what problems do you encounter? |
| 17:20.45 | Qwell | "It's slow on my 233" |
| 17:21.07 | docelm0 | When all 4 are in use the linux crashes cause of memory interrupts.. I sent the card back and bought a cisco gateway |
| 17:21.31 | HA | what kind of machine were you using docelm0? |
| 17:22.07 | harryvv | what card doc |
| 17:23.34 | *** join/#asterisk jets (~jets@guardian.pmt.org) |
| 17:24.21 | jets | Anyone ever seen a lot of CRC errors on your transmit pair to your * box? |
| 17:24.29 | doolph | how can I uninstall asterisk |
| 17:24.33 | shido6 | docelm0, theres a trick to it :) |
| 17:24.36 | bkw_ | WHY OH WHY DO PEOPLE NOT LISTEN TO ME |
| 17:24.44 | bkw_ | I do know what i'm talking about most of the time |
| 17:24.45 | denon | you dont yell LOUD enough :) |
| 17:24.46 | bkw_ | :P |
| 17:24.48 | Zeeek | I heard you: netx! |
| 17:24.52 | Connor- | bkw_ Most? |
| 17:25.03 | jets | bkw_: because people have self serving biases and don't want the truth from you, they want there opinion reaffirmed. people are stupid like that. |
| 17:25.08 | bkw_ | if i'm not right i look up the answer and correct myself. |
| 17:25.21 | bkw_ | well get this |
| 17:25.26 | bkw_ | meetme does have a design flaw |
| 17:25.27 | shido6 | docelm0, you bought a cisco gateway because you couldnt configure your quad card? |
| 17:25.30 | bkw_ | its not ztdummy thats at fault |
| 17:25.32 | bkw_ | its meetme |
| 17:25.35 | bkw_ | meetme SUCKS |
| 17:25.39 | bkw_ | it needs to be fixed |
| 17:25.47 | denon | "submit a patch" |
| 17:25.51 | denon | :P~ |
| 17:25.51 | bkw_ | we tried |
| 17:25.55 | bkw_ | we said screw it |
| 17:25.57 | bkw_ | we wrote our own |
| 17:25.59 | shido6 | hehe |
| 17:26.07 | bkw_ | nobody wants to listen to us |
| 17:26.08 | denon | mark didnt want the patch? |
| 17:26.15 | bkw_ | and HEAVEN forbid we create an extra thread |
| 17:26.23 | CoaxD | denon: Just got a quote from AT&T for a Burstable T1. $427/mo |
| 17:26.25 | bkw_ | well we created an extra thread to do the sound playing |
| 17:26.28 | bkw_ | and other stuffs |
| 17:26.34 | CoaxD | denon: Not oversold, SLA, the whole works |
| 17:26.37 | Connor- | who's not listing to you bkw? |
| 17:26.37 | bkw_ | because meetme will lag X ms of the enter exit tones |
| 17:26.43 | CoaxD | denon: Crazy shit, man |
| 17:26.51 | bkw_ | so if you have people coming and going in a meetme it will lag more an dmore |
| 17:26.54 | bkw_ | over time |
| 17:27.01 | bkw_ | then get this if you go into the user/admin menu |
| 17:27.08 | bkw_ | you lag the amount of time you were in that menu |
| 17:27.15 | bkw_ | or you did last I checked |
| 17:27.30 | W1thdraw | how do i get asterisk to auto start when i start linux |
| 17:27.54 | doolph | I have running asterisk fine, then I installed asterisk-addons and zaptel (i dont have hardware) now i get segmentation fault, how can i reinstall it all correctly? |
| 17:27.58 | bkw_ | um you make an init script |
| 17:28.11 | bkw_ | doolph, if you don't have hardware don't install zaptel |
| 17:28.22 | bkw_ | what does it say when it segfaults? |
| 17:28.27 | bkw_ | that might be most helpful |
| 17:28.31 | doolph | only that |
| 17:28.33 | bkw_ | you don't need everything from asterisk addons |
| 17:28.38 | doolph | segmentation fault |
| 17:28.43 | HA | got a question for ya bkw_...is it possible to make festival load a voice and keep the voice loaded so that it doesn't reload the voice everytime it tries to encode? |
| 17:28.44 | bkw_ | asterisk -vvvgc |
| 17:28.51 | kajtzu | whoa. x-ten finally released their x-lite client for linux. no betas anymore :) |
| 17:28.54 | bkw_ | HA don't use festivle |
| 17:29.00 | bkw_ | its a pile of dog shit |
| 17:29.09 | denon | cepstral rocks |
| 17:29.11 | denon | it really does |
| 17:29.13 | bkw_ | yep |
| 17:29.18 | W1thdraw | auto start asterisk? |
| 17:29.27 | bkw_ | W1thdraw, thats a standard *nix admin question |
| 17:29.29 | tessier_ | Anyone know how many layers of include => asterisk will parse? |
| 17:29.30 | HA | what would you recommend then? i need something that is free. I'm presently using text2wave to do the encoding, but it reloads the voice each time. |
| 17:29.33 | doolph | bkw_ well i just removed /etc/asterisk now |
| 17:29.42 | bkw_ | doolph, asterisk -vvgc |
| 17:29.48 | doolph | ok |
| 17:29.48 | bkw_ | would have told you what module was causing the segfault |
| 17:29.50 | W1thdraw | bkw_ any sugestions? |
| 17:29.53 | doolph | for what is -g? |
| 17:29.59 | bkw_ | drops core |
| 17:30.30 | bkw_ | W1thdraw, well you create/write/invent/use/searc for an init script to go in /etc/rc.d or /etc/init.d on your distro |
| 17:30.36 | bkw_ | then you add it to the default run level |
| 17:30.48 | Zeeek | W1thdraw try looking here: it's old but I think it says http://www.automated.it/guidetoasterisk.htm |
| 17:30.52 | bkw_ | and if you're evil |
| 17:30.56 | bkw_ | you can use inittab to start asterisk |
| 17:30.58 | bkw_ | :P |
| 17:31.02 | W1thdraw | sorry if this is a noob question but how do i know what run level to set it to? |
| 17:31.14 | bkw_ | these are basic system admin concepts |
| 17:31.17 | bkw_ | you should know these things |
| 17:31.29 | Zeeek | http://www.automated.it/guidetoasterisk.htm |
| 17:31.33 | Zeeek | oops |
| 17:31.37 | *** join/#asterisk imagmo (~imagmo@c-24-20-249-117.hsd1.or.comcast.net) |
| 17:31.40 | Zeeek | http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x389.html |
| 17:31.44 | bkw_ | see this is like your doctor coming in and saying "where is the heart located?" |
| 17:31.46 | bkw_ | scary shit |
| 17:31.58 | Zeeek | W1thdraw is working on his high school project |
| 17:32.02 | bkw_ | ah |
| 17:32.08 | bkw_ | then I give you "some" slack |
| 17:32.16 | Zeeek | look at http://asteriskdocs.org and save a lot of time |
| 17:32.18 | doolph | bkw_ if I dont have zaptel i cant use music on hold right |
| 17:32.19 | bkw_ | the contrib directory has the init scripts in it |
| 17:32.21 | Zeeek | our time! |
| 17:32.26 | bkw_ | doolph, you sure as hell can |
| 17:32.37 | W1thdraw | ok thanks guys |
| 17:32.43 | bkw_ | ok find me the moron putting stupid shit in the wiki.. i'm gonna thump him |
| 17:32.44 | Zeeek | doolph is another one who refuses to read |
| 17:32.44 | bkw_ | or her |
| 17:32.46 | *** join/#asterisk Uther_P (~uther_p@66.180.120.83) |
| 17:32.58 | doolph | Ouch ... error while writing audio data: : Broken pipe |
| 17:33.01 | doolph | i get that message |
| 17:33.11 | doolph | then it exits |
| 17:33.39 | Zeeek | W1thdraw you could search for safe_asterisk as well |
| 17:33.51 | Zeeek | that'll prolly lead you where you want to be |
| 17:34.35 | bkw_ | doolph, um thats not really anything to worry about |
| 17:34.37 | bkw_ | you can ignore tht |
| 17:34.44 | bkw_ | but knowing where it segfaults would be most helpful |
| 17:34.54 | bkw_ | why not go in and NOT install asterisk-addons |
| 17:35.00 | bkw_ | because it has some shit in there that can break shit |
| 17:35.09 | bkw_ | like res_config_mysql for example |
| 17:35.16 | bkw_ | go to /usr/lib/asterisk/modules |
| 17:35.17 | bkw_ | rm -rf * |
| 17:35.24 | bkw_ | then |
| 17:35.24 | bkw_ | cd /usr/src/asterisk |
| 17:35.31 | bkw_ | make clean install |
| 17:35.31 | bkw_ | and try again |
| 17:35.55 | bkw_ | you're one of those "lets install everything just because we can... I don't know what this does.. but lets install it!" |
| 17:36.09 | _scat | anyone have a good like to howto's for asterisk? |
| 17:36.09 | doolph | ok |
| 17:36.09 | doolph | i have removed everything now |
| 17:36.24 | Zeeek | http://www.automated.it/guidetoasterisk.htm |
| 17:36.24 | Gand_DJ | Question (unrelated to * sorta..lol)... for someone who's only played with java programming.. what would be best to learn... Visual Basic, C or C++? |
| 17:36.27 | bkw_ | I strip my asterisk installs down when I get them working.. no need having crap you don't use. |
| 17:36.41 | bkw_ | Zeeek, I don't think doolph is paying attention |
| 17:36.43 | doolph | bkw_ I was trying to install AMP from scratch copying the bash script from asterisk@home |
| 17:36.45 | Zeeek | _scat the above link is old but good |
| 17:36.50 | bkw_ | doolph, AMP ewww |
| 17:36.51 | bkw_ | total ass |
| 17:36.51 | *** join/#asterisk newmedian (~crowlther@Quebec-HSE-ppp230300.qc.sympatico.ca) |
| 17:36.55 | Zeeek | je's been on my ignore list for hours |
| 17:36.56 | _scat | thanks |
| 17:37.02 | *** join/#asterisk [TK]D-Fender (~joe@4.67.252.216.dsl1.colba.net) |
| 17:37.08 | doolph | then what do you suggest |
| 17:37.14 | bkw_ | if you can install AMP you don't need it |
| 17:37.21 | Zeeek | _scat my advice is also to read the whole PDF at http://asteriskdocs.org |
| 17:37.30 | doolph | i dont need what |
| 17:37.36 | newmedian | bkw: ha! |
| 17:37.50 | bkw_ | you have the clue level to install AMP you can use emacs/vi/nano/pico/$EDITOR to manage asterisk. |
| 17:38.11 | doolph | yes |
| 17:38.13 | bkw_ | hell I can't even get AMP to install |
| 17:38.18 | bkw_ | and i'm not fucking newbie |
| 17:38.27 | doolph | I have asterisk working with manual installation |
| 17:38.29 | Zeeek | bkw_ your base advise is good for any project: install what you need first and get it working. Then try addons |
| 17:38.35 | newmedian | bkw: you just need to change your crufty installation for a nice and clicky Asterisk@Home and you'll be all set. ;) |
| 17:38.39 | Gand_DJ | appears asterisk@home website is down |
| 17:38.42 | _scat | k, will do... |
| 17:38.42 | doolph | and I have edited sip.conf, voicemail.conf... |
| 17:38.46 | doolph | etc |
| 17:38.51 | Gand_DJ | I got AMP installed manually last year, and I was a newbie :) |
| 17:39.00 | Gand_DJ | I still am kinda |
| 17:39.00 | Gand_DJ | lol |
| 17:39.01 | bkw_ | newmedian, apparently you don't know me... or you wouldn't be saying that. |
| 17:39.14 | doolph | but now I want something more easy to add sip.channels or trunks |
| 17:40.39 | doolph | well |
| 17:40.43 | doolph | now i need to eat something |
| 17:40.53 | *** join/#asterisk heison (~heison@209.205.25.50) |
| 17:41.11 | darwin35 | BKW come to conf |
| 17:41.14 | darwin35 | pls |
| 17:41.57 | harryvv | anyone know of any voip phone suppliers that are located on the west coast? |
| 17:44.02 | kFuQ | <PROTECTED> |
| 17:44.02 | kFuQ | Cisco Systems, Inc. |
| 17:44.02 | kFuQ | 170 West Tasman Dr. |
| 17:44.02 | kFuQ | San Jose, CA 95134 |
| 17:44.02 | kFuQ | USA |
| 17:44.05 | kFuQ | (408)526-4000 |
| 17:44.05 | kFuQ | (800)553-NETS or |
| 17:44.05 | *** join/#asterisk Defraz (~t0tal@65.103.222.4) |
| 17:44.07 | kFuQ | (800)553-6387 |
| 17:44.11 | kFuQ | there ya go harryvv |
| 17:44.11 | kFuQ | lol |
| 17:46.04 | *** join/#asterisk elpcns (~cpacheco@cpe-70-115-51-207.elp.res.rr.com) |
| 17:46.57 | harryvv | Fk Suppliers not Manufactors |
| 17:47.07 | harryvv | like voip supply or attacom |
| 17:47.31 | harryvv | dont worry about it |
| 17:49.01 | HA | how well does cepstral work with asterisk and how difficult is it to setup? |
| 17:49.27 | Zeeek | easy to set up, works fine but generates wav file first |
| 17:50.15 | Zeeek | or maybe I didn't set that part up. I just needed a conversion of text files |
| 17:51.39 | harryvv | what is it |
| 17:51.54 | Zeeek | tts |
| 17:52.07 | elpcns | hello everybody. first time on this channel. I got here because I have a problem with a TDM400p I recently got. Can someone help? |
| 17:52.21 | Zeeek | spit out the problem |
| 17:52.25 | elpcns | ok |
| 17:52.28 | *** join/#asterisk L|NUX (~linux@202.5.145.54) |
| 17:53.36 | elpcns | I have installed this card on a LInux (CentOS 3.4) Asterisk (v1.07) server. Every time I reboot the server kudzu tells me that the card has been removed |
| 17:54.12 | Zeeek | and you do what afetr that? |
| 17:54.24 | elpcns | and offers me to configure it again after this wctdm driver does not load |
| 17:54.42 | Zeeek | so modprobe, ztcfg are done on each reboot? |
| 17:54.57 | elpcns | I just disable kudzu to avoid this |
| 17:55.05 | Zeeek | possibly a sleep(1) between |
| 17:55.37 | Zeeek | how are the drivers loaded if there is no modpobe? |
| 17:55.57 | tessier_ | <PROTECTED> |
| 17:55.59 | elpcns | modprobe zaptel is all the time loaded |
| 17:56.10 | tessier_ | Can anyone help me make sense of that Dial log message? |
| 17:56.14 | tessier_ | IAX2 calling to SIP? huh? |
| 17:56.29 | Zeeek | why not? |
| 17:56.43 | tessier_ | oh..the inbound leg is IAX2 and * is converting to SIP? |
| 17:57.15 | Zeeek | someone who is using IAX2 happenbs to be calling someone else who uses SIP |
| 17:57.38 | Zeeek | neither one knows or cares |
| 18:03.47 | jets | Hey which matt at digium is tech support today :P |
| 18:04.53 | *** part/#asterisk Gand_DJ (~gandalf@ptr-207-54-104-24.ptr.terago.ca) |
| 18:11.05 | *** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc) |
| 18:11.05 | *** topic/#asterisk is Asterisk: The Open Source PBX || Astricon Europe -- Madrid, Spain -- June 15-17 || PBX Developers Conference in Chicago Aug 3-5 Register Now! http://www.asterlink.com/cluecon.htm |
| 18:16.35 | newmedian | Do you think there's sufficient enough thrust (perhaps of http://rfc.net/rfc1925.html quality) that Asterisk@Home should get it's own #asteriskathome, and/or an #asteriskbeginner style channel? At some point the irritation level may boil over. :) |
| 18:17.01 | *** join/#asterisk mrunix (~bwann@69.30.204.34) |
| 18:24.42 | Zeeek | that got a unianimous accolade |
| 18:25.42 | *** part/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 18:26.31 | jets | hey file are you at your office |
| 18:27.23 | *** join/#asterisk Spooch (~rath@p549A0A72.dip0.t-ipconnect.de) |
| 18:27.33 | *** join/#asterisk jmacz (~jmacz@63.245.86.146) |
| 18:31.22 | Qwell | wtf |
| 18:31.36 | Qwell | I have this analog phone...its got a mute button...that doesn't do shit |
| 18:33.07 | *** join/#asterisk rcam (~rcammobil@adsl-218-151-77.jax.bellsouth.net) |
| 18:34.44 | syle | congratulations |
| 18:35.29 | *** join/#asterisk bjkmad (~prometheu@85.100.237.140) |
| 18:36.53 | bjkmad | need help about g729 codec |
| 18:37.17 | *** join/#asterisk alt_phil (~alt_phil@abgtr1.abgnetwork.net) |
| 18:37.31 | alt_phil | Hey all |
| 18:37.40 | RoyK | bjkmad: just buy it :) |
| 18:37.53 | bjkmad | impossible |
| 18:38.05 | bjkmad | i am student |
| 18:38.12 | Qwell | Then don't use it |
| 18:38.27 | Micc | Is there a known problem with iaxclient and DTMF tones? |
| 18:39.08 | bjkmad | is there any open source or free g279 codec |
| 18:39.09 | Micc | The app I'm developing using iaxclient is sending dtmf tones all garbled. Then I tried iaxcomm and it has the same problem. |
| 18:39.46 | alt_phil | I had a quickie question... How can I stop an agent's voicemail from answering a queue? If an agent's phone doesn't ring (like on DND), but they're logged on, the queue call goes strait to their voicemail. |
| 18:40.03 | Qwell | So, can I mute a call from CLI? |
| 18:40.05 | alt_phil | Anyway to stop that from happening? I just don't want an agent's voicemail to ever answer a queue. |
| 18:43.16 | *** join/#asterisk mutilator (~animenodv@65.111.201.79) |
| 18:43.50 | jets | Mormon Matt |
| 18:43.51 | bjkmad | there are about 300 people here and nobody is talking,everybody is so busy? |
| 18:43.52 | jets | Matt matt matt |
| 18:44.05 | doolph | what you need bjkmad |
| 18:44.24 | Qwell | bjkmad: If you can't afford g729, you don't need it. |
| 18:44.35 | bjkmad | i have to use g729 codec |
| 18:44.42 | Qwell | So buy it |
| 18:44.50 | jets | alt_phil: What I do is have a "queue" extension and a "internal" extension. |
| 18:44.56 | doolph | bjkmad there's a trial |
| 18:44.59 | bjkmad | i am student:) |
| 18:45.04 | jets | When they log in log them in on there queue extension, e.g. 1000 --- that doesn't have voicemail in its extension dialplan |
| 18:45.06 | doolph | check http://www.readytechnology.co.uk/open/g729/ |
| 18:45.07 | Qwell | bjkmad: Then you don't need it |
| 18:45.37 | doolph | but it will expire if you dont buy it |
| 18:45.41 | bjkmad | this is about my senior project |
| 18:45.48 | alt_phil | Ok, that's a cool way of doing it - thanks jets. |
| 18:45.50 | doolph | it only cost about $20 |
| 18:46.10 | bjkmad | per chanell,isn't it? |
| 18:46.18 | Qwell | $10 per channel from Digium |
| 18:46.24 | Micc | bjkmad, implement your own. |
| 18:46.30 | *** join/#asterisk christo (~chris@195.82.111.57) |
| 18:46.31 | doolph | yeah |
| 18:46.53 | bjkmad | good suggestion:) |
| 18:47.17 | Qwell | I don't think you can... |
| 18:47.23 | Qwell | not legally |
| 18:47.54 | syle | what about SIP and g729? |
| 18:48.02 | doolph | syle? |
| 18:48.30 | syle | what codec do you use for SIP? |
| 18:48.44 | doolph | any |
| 18:48.44 | Micc | bjkmad, improve on the patent 10% and repatent your improvments. |
| 18:48.52 | RoyK | doolph: expire? i mean - it's open source..... |
| 18:49.14 | doolph | well it is not free |
| 18:49.43 | Qwell | any way you look at it, if you're going to use g729 legally in the US, you MUST buy a license |
| 18:49.52 | implicit | or you must license your implementation |
| 18:49.56 | RoyK | i know |
| 18:50.01 | RoyK | but who cares? :) |
| 18:50.08 | implicit | you can get the reference sources from ITU-T |
| 18:50.09 | RoyK | noone will know if you don't |
| 18:50.20 | RaYmAn-Bx | Qwell: the interesting question is which other countries you also have to buy a license |
| 18:50.20 | implicit | RoyK, not true, failures won't be noticed |
| 18:50.23 | Qwell | RoyK: Thats true, but we don't support it. |
| 18:50.37 | Nuxi | Or, find some way of invalidating the patent, which (if possible) would cost much, much, much more than the license. |
| 18:50.38 | Qwell | support or condone |
| 18:50.39 | implicit | RoyK, but if you are successful in any sort of VoIP business people will find out |
| 18:50.44 | RoyK | i buy my g.729 licenses for production |
| 18:50.56 | RoyK | just ordered another 50 |
| 18:50.57 | RoyK | :P |
| 18:50.58 | implicit | for non-commercial use i don't think it is illegal |
| 18:51.04 | implicit | RoyK, who's licenses do u use? |
| 18:51.11 | RoyK | digium's |
| 18:51.16 | implicit | ah |
| 18:51.17 | implicit | ok |
| 18:51.21 | implicit | they are $$$ |
| 18:51.22 | implicit | :) |
| 18:51.23 | RoyK | they're ok |
| 18:51.28 | RoyK | well |
| 18:51.32 | RoyK | boss pays |
| 18:51.38 | RoyK | i don't |
| 18:51.45 | syle | what you mean per channel? you mean if you were running a channel bank? |
| 18:52.00 | RoyK | they should have differenciated prices on those |
| 18:52.01 | implicit | syle, in simultaneous use |
| 18:52.02 | Qwell | !google g729 license howto |
| 18:52.11 | Qwell | ~google g729 license howto |
| 18:52.13 | Qwell | wrong channel :D |
| 18:52.26 | RoyK | ~lart Qwell |
| 18:52.28 | implicit | if someone has a couple DS3s or more they should just license the code for themselves |
| 18:52.47 | mmlj4 | any idea why someone would be trying to connect on UDP ports 1026-1029? that's not related to SIP, or FWD by any chance? |
| 18:53.05 | bjkmad | i have seen a lot of people who buy licence by they still in trouble |
| 18:53.13 | RoyK | mmlj4: perhaps they want to test if you're running an old portmapper or something |
| 18:53.18 | RoyK | :) |
| 18:53.32 | RoyK | always a lot of nice people on the net probing your boxes |
| 18:55.43 | *** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc) |
| 18:55.44 | *** topic/#asterisk is Asterisk: The Open Source PBX || Astricon Europe -- Madrid, Spain -- June 15-17 || PBX Developers Conference in Chicago Aug 3-5 Register Now! http://www.asterlink.com/cluecon.htm |
| 18:55.50 | Qwell | ahh |
| 18:55.53 | darwin35 | ~sex |
| 18:55.55 | jbot | updatedb; locate; talk; date; cd; strip; look; touch; finger; unzip; uptime; gawk; head; apt-get install condom; mount; fsck; gasp; more; yes; yes; yes; more; umount; apt-get remove --purge condom; make clean; sleep |
| 18:56.22 | RoyK | rotfl |
| 18:56.29 | HA | ~more sex |
| 18:56.57 | HA | jbot must be tired. |
| 18:56.59 | darwin35 | ~moresex |
| 18:57.12 | darwin35 | lol |
| 18:57.16 | Corydon-w | Or lagged |
| 18:57.24 | darwin35 | this inbound issue is killing me |
| 18:57.44 | Nuxi | ~google sex |
| 18:57.53 | darwin35 | all the direct inbound work |
| 18:57.58 | Qwell | Nuxi: You're going to break google |
| 18:58.00 | implicit | darwin35, to tell you the truth, it doesn't seem like a big issue :) |
| 18:58.06 | Qwell | Nuxi: Too many results |
| 18:58.09 | RoyK | ~fsck Nuxi |
| 18:58.10 | jbot | e2fsck /dev/Nuxi : warning! filesystem contains morons! |
| 18:58.20 | RoyK | :) |
| 18:58.24 | darwin35 | but the 800 and main nmbr dont point right |
| 18:58.51 | Qwell | darwin35: Are you working in the right context? |
| 19:00.39 | [TK]D-Fender | I'm about to make some final proposal's the big cheese here about an * solution and am trying to pin down which phone models I'm best served with. My top picks right now are Uniden UIP-200, Polycom 500/600, and Cisco 7940/7960. Any specific caveats that I should be aware of in this? |
| 19:02.03 | *** join/#asterisk cmk (~cmk_@p54A3F055.dip.t-dialin.net) |
| 19:04.35 | darwin35 | poly or cisco |
| 19:05.03 | *** join/#asterisk Duante (~test@rrcs-67-78-88-146.sw.biz.rr.com) |
| 19:05.27 | darwin35 | hold a min |
| 19:06.00 | *** join/#asterisk flynux (0p23wpu@pingou.in) |
| 19:06.44 | Duante | yawn any body here/ |
| 19:06.46 | Duante | ? |
| 19:06.46 | xkev | fender, I went with polycom after review of snom, polycom and cisco |
| 19:07.07 | Qwell | I would go with the ciscos, but...thats just me |
| 19:07.09 | xkev | they _all_ have minor bugs :) |
| 19:07.23 | xkev | the polycom 600s are dreamy. 7960 can't touch it. :) |
| 19:07.52 | Qwell | yeah, because, if the 7960 touches the 600, the 600 cries and runs away |
| 19:07.54 | jets | Should I be concerned about CRC errors on a t1 |
| 19:08.04 | implicit | Qwell, not an untrue statement |
| 19:08.07 | Qwell | :p |
| 19:08.08 | *** part/#asterisk hellop (~LeeHarvey@cpe-70-93-40-171.hawaii.res.rr.com) |
| 19:08.12 | xkev | my decision was based primarily on the microbrowser, the interface, and the cost. |
| 19:08.28 | Qwell | I can't say I've used any IP phones, really |
| 19:08.32 | xkev | ..oh and not having to use 48V or cisco proprietary PoE |
| 19:08.33 | Duante | hmm anybody know how to on incoming calls ... example: if the person dialed 838-2002 it routes it to extension 400. but if they dial 838-2003 it routes it to extension 700 |
| 19:08.35 | Micc | I have a question about how asterisk deals with DTMF and iax2 to sip in-band dtmf |
| 19:08.42 | Qwell | If somebody is willing to ship me some, I'll be glad to make an informed opinion |
| 19:08.45 | *** join/#asterisk darwin35 (~darwin35@24.3.226.147) |
| 19:08.49 | darwin35 | oooops |
| 19:09.00 | Qwell | Duante: How are they coming in? |
| 19:09.02 | xkev | duante, depends on what tech is bringing you the incoming calls |
| 19:09.18 | Duante | iax2 im using voicepulse... |
| 19:09.37 | Qwell | Does it send it to exten => 8382003,1 ? |
| 19:09.44 | xkev | should be via extensions in the context your incoming calls hit |
| 19:09.56 | Micc | Does anyone want to help me understand how DTMF works with these protocols and asterisk? Maybe the best way to do DTMF is to play the sounds over the line as samples instead of having IAX/asterisk handle it. |
| 19:10.00 | Qwell | if so, thats VERY simple |
| 19:10.24 | xkev | for PRI (same thing) I do _NXXNXXXXXX,1,DBget(jump=DID/${EXTEN}); then Goto(${jump}); |
| 19:10.26 | Qwell | Micc: There are generally two ways to send DTMF. inbound, and rfc..something |
| 19:10.28 | tzanger | Micc: you can't send continuous tones over compressed codecs |
| 19:10.31 | xkev | ..but I have lots of DID |
| 19:10.34 | tzanger | rfc2833 |
| 19:10.36 | Qwell | don't msg me |
| 19:10.45 | Qwell | erm, inband rather |
| 19:11.44 | darwin35 | ok who was I chatting |
| 19:12.11 | Duante | not sure qwell... im using voicepulse with iax2... the call comes in and hits the main greeting in extensions.conf.. but how can i test what number triggered it? |
| 19:12.35 | Qwell | So it uses s, or what? |
| 19:12.42 | Qwell | RTFVPM |
| 19:13.04 | Duante | yes |
| 19:13.04 | xkev | try it |
| 19:13.27 | Qwell | NuFone doesn't suck. They send it to the exten matching the DID |
| 19:13.31 | xkev | if VP calls a Dial(IAX2/you@there/${DID}) then you should get the DNID |
| 19:13.58 | Duante | how do i test the dnid? |
| 19:14.03 | Duante | in the extensions.conf |
| 19:14.05 | xkev | do you have two register lines for your two DIDs? |
| 19:14.19 | xkev | NoOp(${DNID}) iirc |
| 19:14.41 | xkev | ..if so, then make two contexts and the stupid 's' will work that way |
| 19:14.57 | Qwell | Maybe its just defaulting to s? |
| 19:15.03 | xkev | I suspect it would |
| 19:15.05 | Qwell | Have you tried making an exten that matches your DID? |
| 19:15.11 | Duante | nope |
| 19:15.13 | xkev | (xkev) try it |
| 19:15.15 | rcam | The server for Nufone is switch-1.nufone.net right? |
| 19:15.22 | Qwell | rcam: or switch-2 |
| 19:15.34 | Duante | asterisk by default routes to matching extension? |
| 19:15.35 | Duante | jw |
| 19:16.37 | Qwell | depends on the provider |
| 19:16.43 | Qwell | I guess |
| 19:17.16 | marlowe | Does anyone know how to set on a polycom ip500 via the web based interface to only use specific rtp ports? |
| 19:17.25 | marlowe | i see forced port and port range start |
| 19:17.31 | marlowe | i dont see a port range end though |
| 19:19.49 | *** join/#asterisk _THEEND_ (~DrEaM@80.18.184.226) |
| 19:19.50 | *** join/#asterisk vinko (~vinkoval@63.170.64.37) |
| 19:20.00 | _THEEND_ | hi! |
| 19:20.10 | _THEEND_ | someone could help me? i have this error... |
| 19:20.11 | _THEEND_ | insmod: error inserting 'zaphfc.ko': -1 Invalid module format |
| 19:21.56 | shido6 | 2.6? |
| 19:22.00 | darwin35 | http://pastebin.ca/11914 this is what I get when some one dial in on the main nmbr |
| 19:22.07 | *** join/#asterisk Micc (~mic@c-24-18-35-120.hsd1.wa.comcast.net) |
| 19:22.14 | _THEEND_ | yes |
| 19:22.15 | _THEEND_ | 2.6 |
| 19:22.38 | Micc | Grrr. My linux box decided it wanted to eat all available memory and start swapping like crazy and not let me switch workspaces. Fun fun. |
| 19:22.59 | Micc | Its only been running for a few weeks. |
| 19:23.50 | Micc | Anyways, I wasn't able to get the response to my question about dtmf because I couldn't switch workspaces. Does someone want to talk to me about dtmf stuff real quick? |
| 19:24.24 | Micc | Is the best way to just play the sound down the pipe or let IAX/Asterisk handle the dtmf. |
| 19:24.26 | DrWho17 | Micc: sounds like you should diagnose your linux problem first |
| 19:24.34 | rcam | jbot Seen jerjer |
| 19:24.46 | jbot | jerjer <~JerJer@DSL-226.206-rt-bras.che.centurytel.net> was last seen on IRC in channel #asterisk, 1d 6h 12m 32s ago, saying: 'ManxPower: sounds interesting, but i've still never used it'. |
| 19:24.46 | Micc | DrWho, this isn't a production machine. |
| 19:24.53 | _THEEND_ | noone coul help me? insmod: error inserting 'zaphfc.ko': -1 Invalid module format |
| 19:24.57 | _THEEND_ | could |
| 19:24.57 | Micc | DrWho, we run BSD for our production asterisk server. |
| 19:25.13 | DrWho17 | ouch |
| 19:26.38 | [TK]D-Fender | xkev : So you've had a pretty decent experience with your IP 600's? Any unresolved issues? |
| 19:27.06 | cjk | is there a way to use h323 and g729 in asterisk? |
| 19:27.09 | Duante | hmm still trying to figure out my question if anybody has some input |
| 19:27.23 | doolph | cjk yes |
| 19:27.28 | doolph | just you need to find out |
| 19:27.41 | elpcns | Sorry, I'm back again. I have two problems with a TDM400P (4FXO modules rev. H) 1. Kudzu tells me that the card has been removed every time I reboot the server (only when the server is rebooted) 2. The card is not dialing correctly a given phone number. I always get the message from the TELCO sayin that the phone number dialed is incorrect. I'm using asterisk 1.07 and CentOS 3.4. Does any one has ever have a problem like this? |
| 19:28.27 | darwin35 | Moc |
| 19:28.33 | doolph | elpcns your need to fix your dialplan |
| 19:28.48 | *** join/#asterisk bah (048830696@AC8046BC.ipt.aol.com) |
| 19:29.06 | elpcns | but I'm executing a simple dial(zap/1/7805800) |
| 19:29.43 | cjk | doolph: so its not just basic configuration? |
| 19:30.06 | elpcns | no I don't think so |
| 19:30.20 | Duante | to use g729 you need a liscence for each conversion... but if your phones are all g729 you dont need anything |
| 19:30.23 | Duante | just turn the codec on |
| 19:30.49 | Duante | but if your phones(g729) -> service(not g729) its 1 conversion per open line |
| 19:31.55 | *** join/#asterisk Johnsie (~john@acs-24-154-32-12.zoominternet.net) |
| 19:32.04 | Duante | also if you have any other codec allowed besides g729 |
| 19:32.06 | RoyK | Duante: you need license for transcoding |
| 19:32.07 | Duante | asterisk will choose it |
| 19:32.16 | RoyK | not for doing passthrough |
| 19:32.19 | Duante | thats what i said just above |
| 19:32.24 | Duante | liscence for each conversion. |
| 19:32.32 | RoyK | er. yes. |
| 19:32.39 | RoyK | i read conversation..... :P |
| 19:32.58 | RoyK | ever heard of the dyslexic satanist that sold his soul to santa? |
| 19:33.06 | Duante | um no |
| 19:34.16 | Duante | anyone else have some ideas about my problem? i dont think you guys explained your thoughts very well so i wasnt sure what you suggested |
| 19:34.18 | focks | and the guy who worships dog |
| 19:35.44 | Duante | heres a question... if i didnt want to use default s... how would i change it? so incoming call would use something besides default |
| 19:36.22 | elpcns | Can someone help me to figure out what might be the problem with this TDM400p card? |
| 19:36.43 | tessier_ | elpcns: I have sworn off tdm400p cards. |
| 19:36.56 | tessier_ | Nothing but cheap cisco routers with fxo ports for me. They cost about the same anyhow. |
| 19:37.11 | elpcns | ok I see |
| 19:37.20 | tessier_ | And actually the more I learn about SER the less I find for asterisk to do in my system. |
| 19:37.27 | tessier_ | Need to investigate SEMS and see if it is ready for production yet |
| 19:38.13 | elpcns | are you using sip express together with Cisco ? |
| 19:38.26 | doolph | OMG |
| 19:38.29 | doolph | lol |
| 19:38.33 | doolph | sorry im happy |
| 19:38.37 | tessier_ | elpcns: yes |
| 19:38.42 | doolph | i got installed * |
| 19:38.43 | doolph | lol |
| 19:39.01 | tessier_ | lollerskatezomfg!!!!11!!11!!oneone |
| 19:39.08 | elpcns | how does it work? tessier |
| 19:43.01 | *** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc) |
| 19:43.02 | *** topic/#asterisk is Asterisk: The Open Source PBX || Astricon Europe -- Madrid, Spain -- June 15-17 || PBX Developers Conference in Chicago Aug 3-5 Register Now! http://www.asterlink.com/cluecon.htm |
| 19:43.11 | *** join/#asterisk cursor (~kevin@andromeda.office.cursor.biz) |
| 19:44.35 | elpcns | Does anyone know how to deal with the TDM400P card from Digium? |
| 19:45.00 | cursor | How to deal with it? |
| 19:45.27 | elpcns | Yes I've been having some problems with it |
| 19:45.30 | cursor | It's a card - if you have enough of them, you can deal a hand |
| 19:45.52 | cursor | Did you try Digium support? |
| 19:46.09 | elpcns | not yet |
| 19:46.15 | cursor | ok - what's the problem |
| 19:47.04 | elpcns | I got this card from ebay and I have two problems with a TDM400P (4FXO modules rev. H) 1. Kudzu tells me that the card has been removed every time I reboot the server (only when the server is rebooted) 2. The card is not dialing correctly a given phone number. I always get the message from the TELCO sayin that the phone number dialed is incorrect. I'm using asterisk 1.07 and CentOS 3.4. Does any one has ever have a problem like this? |
| 19:47.10 | elpcns | I got this card from ebay and I have two problems with a TDM400P (4FXO modules rev. H) 1. Kudzu tells me that the card has been removed every time I reboot the server (only when the server is rebooted) 2. The card is not dialing correctly a given phone number. I always get the message from the TELCO sayin that the phone number dialed is incorrect. I'm using asterisk 1.07 and CentOS 3.4. Does any one has ever have a problem like this? |
| 19:47.13 | elpcns | I got this card from ebay and I have two problems with a TDM400P (4FXO modules rev. H) 1. Kudzu tells me that the card has been removed every time I reboot the server (only when the server is rebooted) 2. The card is not dialing correctly a given phone number. I always get the message from the TELCO sayin that the phone number dialed is incorrect. I'm using asterisk 1.07 and CentOS 3.4. Does any one has ever have a problem like this? |
| 19:47.16 | focks | ahh |
| 19:47.19 | cursor | echo |
| 19:47.20 | cursor | :-) |
| 19:47.57 | elpcns | sorry |
| 19:48.09 | darwin35 | http://pastebin.ca/11914 need hlp with this |
| 19:48.12 | *** part/#asterisk darth-timeus (darth@200.105.128.61) |
| 19:48.17 | CoaxD | um |
| 19:48.25 | cursor | elpcns - sorry, I don't know |
| 19:48.50 | shido6 | darwin35, hhehehe |
| 19:48.53 | shido6 | what are you doing? |
| 19:48.54 | shido6 | :) |
| 19:49.03 | DrWho17 | elpcns: disable kudzu, and dial the correct number |
| 19:49.45 | elpcns | I did disable kudzu, but is this correct? |
| 19:49.53 | cursor | darwin35: How are you calling Dial() ? |
| 19:50.42 | marlowe | Does anyone know how to set on a polycom ip500 via the web based interface to only use specific rtp ports? |
| 19:50.46 | marlowe | i see forced port and port range start |
| 19:50.47 | marlowe | i dont see a port range end though |
| 19:50.53 | elpcns | exten => _9NXXXXXX,2,Dial(Zap/1,{EXTEN:1}) |
| 19:51.15 | Qwell | elpcns: Do you need to dial an areacode? |
| 19:51.17 | cursor | marlow: It probably doesn't use a lot of ports |
| 19:51.37 | elpcns | it's a local number |
| 19:52.01 | elpcns | with prefix 9 |
| 19:52.19 | Qwell | ${, not { |
| 19:53.02 | elpcns | you mean I have to add the area code? |
| 19:53.10 | cursor | elpcns: Dial(Zap/1/${EXTEN:1}) |
| 19:53.19 | Qwell | no, use the var, don't try to send crap chars |
| 19:53.34 | cursor | Or, probably, Dial(Zap/g1/${EXTEN:1}) |
| 19:54.47 | elpcns | something like this: exten => _9.,2,Dial(Zap/1/${EXTEN:1}) |
| 19:54.53 | cursor | g1 |
| 19:54.57 | cursor | instead of 1 |
| 19:55.03 | Qwell | 9.? that would be silly |
| 19:55.10 | Qwell | 9011..etc |
| 19:55.32 | cursor | yes - you probably don't want to send everything over a zap link |
| 19:55.44 | Uther_P | you would want either _9NXXNXXXXXX and/or _9NXXXXXX |
| 19:55.47 | cursor | You might want to consider VoIP - especially for international calls |
| 19:56.07 | DrWho17 | elpcns: no, adding the $ sign in front of {EXTEN:1} would have fixed it already |
| 19:56.36 | DrWho17 | I always disable kudzu, I usually know when I'm adding hardware or not, I don't need kudzu to tell me |
| 19:56.45 | elpcns | yes I missed it ($) sorry but it's in my dial plan |
| 19:57.31 | eKo1 | DrWho17: That's OK if you're the only admin. screwing around in the server room. |
| 19:57.34 | elpcns | ok Kudzu has been disable |
| 19:57.35 | DrWho17 | well, what does debug say when you try to dial it? |
| 19:57.38 | doolph | hi |
| 19:57.54 | DrWho17 | eKo1: why do you want them to run into kudzu asking questions? |
| 19:58.21 | eKo1 | Actually, the kudzu is for me. I need to know what the hell happened. |
| 19:58.32 | DrWho17 | I disable everything that is non-essential, less things to remember and screw you over later |
| 19:59.01 | doolph | hello |
| 19:59.06 | doolph | who do need help |
| 19:59.12 | elpcns | I do |
| 19:59.31 | doolph | about what |
| 19:59.39 | DrWho17 | elpcns: is the card any good? Have you tried any of the other ports beside the one you are testing with? |
| 19:59.54 | DrWho17 | sometimes people put those on ebay, because of lightning strike issues |
| 19:59.54 | elpcns | TDM400P misbehavior |
| 20:00.29 | elpcns | yes I even tried with diferent hardware |
| 20:00.41 | doolph | elpcns |
| 20:00.44 | elpcns | yes |
| 20:00.50 | doolph | what's the problem |
| 20:02.06 | elpcns | everytime when I reboot the server kudzu rediscover the card. I have already disabled kudzu but I'm not sure if this workaround is appropiate |
| 20:02.24 | doolph | kudzu?¿ |
| 20:02.32 | elpcns | yes |
| 20:02.43 | Qwell | elpcns: There are times, where you should be selective about who you get help from... |
| 20:02.46 | Qwell | This is one of those times. |
| 20:02.49 | doolph | try chkconfig --remove kudzu |
| 20:03.00 | elpcns | ok |
| 20:03.28 | *** join/#asterisk krw (~kenwiesne@border.logicalonline.com) |
| 20:03.57 | doolph | anyways what is kudzu |
| 20:03.58 | doolph | lol |
| 20:04.02 | focks | hardware detection |
| 20:04.09 | focks | it's annoying |
| 20:04.11 | doolph | ah |
| 20:04.16 | doolph | that shitty |
| 20:04.22 | cursor | As I understand it, krudzu just probes hardware and loads any missing kernel modules |
| 20:04.25 | *** join/#asterisk Pete_Largo (~Pete_Larg@adsl-65-71-225-121.dsl.rcsntx.swbell.net) |
| 20:04.31 | cursor | why not just set up the machine to load the modules by default? |
| 20:04.32 | *** join/#asterisk ariel_ (~Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 20:04.37 | ariel_ | hello everyone |
| 20:04.44 | cursor | hello |
| 20:04.52 | focks | honestly, who yanks cards in and out of their machines between boots |
| 20:05.16 | focks | and doesn't know what modules to load/unload in the event that they like to yank cards |
| 20:05.17 | cursor | honestly, who boots their server? |
| 20:05.18 | cursor | :-) |
| 20:05.25 | focks | right on |
| 20:05.31 | elpcns | ok I'm listening |
| 20:05.39 | eKo1 | focks: admins. who tell the dumb technicians to do so. |
| 20:05.47 | cursor | Which kernel version do you have? |
| 20:05.52 | focks | uname -a |
| 20:05.57 | ariel_ | I just booted 2 servers.. (Had to change there names). |
| 20:06.05 | Qwell | real men hotswap pci |
| 20:06.14 | focks | ariel_ hostnames? |
| 20:06.18 | focks | no reason to reboot |
| 20:06.30 | Qwell | unless its windows |
| 20:06.32 | eKo1 | Qwell: that doesn't look right. |
| 20:06.32 | focks | Qwell lol, i used to do that with a Packard Bell and ISA modems |
| 20:06.36 | ariel_ | focks, yes host name.. (Windows servers) |
| 20:06.42 | doolph | lol |
| 20:07.07 | focks | Qwell shorted out the LEDs on my NIC but it worked fine otherwise |
| 20:07.18 | Qwell | focks: I hotswapped RAM once...on accident |
| 20:07.27 | focks | how did that go? ;) |
| 20:07.36 | cursor | :-) |
| 20:07.38 | Qwell | it hung...thats about it |
| 20:07.39 | cursor | very hot seap |
| 20:07.41 | cursor | swap |
| 20:07.46 | focks | i hotswapped a WD IDE drive and RMAed it |
| 20:07.59 | jets | i like things that are hung. |
| 20:08.27 | Qwell | psh, I hotswap IDE all the time. :p |
| 20:08.27 | Qwell | it doesn't WORK, but... |
| 20:09.24 | *** join/#asterisk R3DB0x (nobody@66.142.28.36) |
| 20:09.34 | *** join/#asterisk mango_man (~mango_man@pc-62-30-33-29-pr.blueyonder.co.uk) |
| 20:09.55 | focks | what should my sup file look like to get stable instead of latest CVS? |
| 20:09.55 | focks | would i use '*default release=stable tag=.' ? |
| 20:09.55 | cursor | You'd do well in the middle ages then |
| 20:10.57 | _THEEND_ | insmod: error inserting './zaphfc.ko': -1 Invalid module format |
| 20:11.00 | _THEEND_ | noone could help me? |
| 20:11.31 | cursor | Did you build that module on the machine you're using now? |
| 20:13.30 | cursor | airplane noises? that's a new excuse |
| 20:14.47 | sivana | heh |
| 20:14.48 | Qwell | cursor: You know, like "Vroom! zipp!" |
| 20:14.49 | Qwell | etc |
| 20:14.50 | Pete_Largo | what's wrong with airplane noises? |
| 20:15.07 | cursor | When I make "airplane noises", everyone else runs away |
| 20:15.28 | *** join/#asterisk klasstek (~nunyobiz@sta-206-168-218-206.rockynet.com) |
| 20:15.35 | cursor | Although the silent ones are worse |
| 20:15.37 | cursor | :-) |
| 20:15.39 | Pete_Largo | that's half the fun, then you can chase them! |
| 20:15.50 | _THEEND_ | cursor yes |
| 20:15.52 | cursor | stealth bombers |
| 20:16.11 | x9net | lol |
| 20:17.36 | slePP | anyone remember who western canada internet is? |
| 20:17.43 | focks | anyone had trouble with this? |
| 20:17.44 | focks | chan_sip.c: In function `build_user': |
| 20:17.44 | focks | chan_sip.c:10007: parse error before `struct' |
| 20:18.10 | *** join/#asterisk Exstatica (Exstatica@jumping.on.the.bed.are.not.umpteenmonkeys.com) |
| 20:19.50 | doolph | where does *77 records save to? |
| 20:19.52 | pimpwell | rogers.com canada? |
| 20:22.10 | *** join/#asterisk HoopyCat (user@nocrtucker.netaccnt.net) |
| 20:22.31 | *** part/#asterisk Bentley (~Bentley@S01060080c8135e6a.cg.shawcable.net) |
| 20:22.34 | doolph | hey |
| 20:22.38 | doolph | who is onoline |
| 20:24.04 | HoopyCat | onoline is a close relative to gasoline; it's primarily used in molotov cocktails. it got its name from the sound uttered by the recipients of said object, e.g. "oh no!" |
| 20:24.21 | Pete_Largo | lol |
| 20:25.13 | cursor | Is that similar to the aaaarghline? |
| 20:25.59 | *** join/#asterisk bonez39 (~aint@drjones.dsl.xmission.com) |
| 20:26.04 | *** join/#asterisk mikewho2 (~asdf@nts-221.3-185-64-static.nts-online.net) |
| 20:26.10 | mikewho2 | Hello guys, anyone around? |
| 20:26.12 | *** join/#asterisk bjk_mad (~prometheu@85.100.235.0) |
| 20:26.25 | Uther_P | I'm more of a cylinder |
| 20:26.30 | clint_ | No. |
| 20:26.33 | cursor | mikewho2: no - there's nobody |
| 20:26.34 | x9net | I got a budgetone 101 and when i call out at about 14 seconds the call drops, im thinking its with the codecs anyone have any ideas? xlite works perfect calling out, so * is set up right. |
| 20:26.49 | mikewho2 | blah :p |
| 20:26.53 | mikewho2 | i knew id get that answer on irc |
| 20:27.03 | Uther_P | then why ask? |
| 20:27.05 | Uther_P | what a boring game |
| 20:27.20 | mikewho2 | so is asterisk the most popular voip pbx out there? |
| 20:27.23 | darwin35 | man this is killing me not getting this figured out |
| 20:27.29 | darwin35 | and no one helping |
| 20:27.31 | Uther_P | x9net: is it always the same time? |
| 20:27.55 | darwin35 | http://pastebin.ca/11926 |
| 20:27.59 | x9net | yea like 14 sec some times 10 sec |
| 20:28.08 | mikewho2 | im kinda new to voip |
| 20:28.10 | x9net | the call will go out but just drip |
| 20:28.26 | darwin35 | when I dial in on the company 866-XXX_XXX nmbr it loops and fails |
| 20:28.27 | Uther_P | x9net: when using xlite, is there accually a mic pluged in and sending sound, or were you just using it to test connection? |
| 20:28.45 | x9net | i got a head set hooked up |
| 20:28.53 | doolph | lol |
| 20:29.05 | Uther_P | ahh, wondering if the problem might have been callprogress |
| 20:29.09 | Uther_P | I had a problem with that |
| 20:29.38 | x9net | so * is configured right, i think it has something to do with g711u i just cant seem to see how to set that in the bugetone101 |
| 20:29.40 | mikewho2 | what kind of hardware would i need at the client lvl if all my calls are routed through a pbx server |
| 20:29.43 | mikewho2 | ipphone? |
| 20:29.51 | Uther_P | heh, what? |
| 20:29.58 | mikewho2 | i have a central server |
| 20:30.01 | Uther_P | thats gotta be the vaugest question ever |
| 20:30.05 | Uther_P | vaguest |
| 20:30.12 | mikewho2 | What im driving at here is this |
| 20:30.22 | mikewho2 | I want to have a server running this software |
| 20:30.25 | x9net | you mean a hardware phone? |
| 20:30.29 | mikewho2 | correct |
| 20:30.44 | x9net | you need a sip phone, |
| 20:30.47 | darwin35 | you have to setup the device |
| 20:30.51 | Uther_P | any ip phone that supports sip or aix will generally work |
| 20:30.54 | doolph | you have to setup the server |
| 20:30.59 | doolph | heh |
| 20:31.01 | darwin35 | in sip.conf and threw its webpage |
| 20:31.19 | Uther_P | x9net: bugetone uses pcmu by default |
| 20:31.29 | mikewho2 | is there something in the protocol that automatically finds IP phones if its on the same network |
| 20:31.35 | Uther_P | heh, no |
| 20:31.46 | Uther_P | the phones have to register |
| 20:31.48 | doolph | you need to register all ip phones |
| 20:31.56 | doolph | cool Uther_P |
| 20:31.59 | mikewho2 | does this happen during DHCP? |
| 20:32.06 | darwin35 | any input on the pastbin |
| 20:32.11 | mikewho2 | or is it better to have static IPs and just register the phones |
| 20:32.27 | doolph | mikewho2 it doesnt care, you need to add the server in your ip phone |
| 20:32.39 | mikewho2 | i see. |
| 20:32.47 | *** part/#asterisk darwin35 (~darwin35@24.3.226.147) |
| 20:33.39 | x9net | is pcmu g711u? |
| 20:34.25 | *** part/#asterisk bjk_mad (~prometheu@85.100.235.0) |
| 20:34.51 | *** join/#asterisk kb1_kanobe (~krisbouti@h24-207-80-55.cst.dccnet.com) |
| 20:35.10 | eKo1 | yes |
| 20:35.21 | *** join/#asterisk jeffik (~Jeff@69.158.19.117) |
| 20:35.24 | x9net | hmm |
| 20:36.01 | x9net | i just shot a email to grandstream so ill see what they say. |
| 20:36.48 | *** join/#asterisk Seyr (~Seyr@rrcs-24-227-133-226.sw.biz.rr.com) |
| 20:36.51 | Seyr | heyas people |
| 20:37.18 | *** join/#asterisk Granamear (~none@ddsl-216-68-219-94.fuse.net) |
| 20:37.37 | x9net | Any hardware sip phones you would recomend? im thinking i want to get a better phone. |
| 20:37.38 | Seyr | i have an IAX peer and if I add another one, my first one drops. anyone have a guess? |
| 20:40.11 | *** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc) |
| 20:40.12 | *** topic/#asterisk is Asterisk: The Open Source PBX || Astricon Europe -- Madrid, Spain -- June 15-17 || PBX Developers Conference in Chicago Aug 3-5 Register Now! http://www.asterlink.com/cluecon.htm |
| 20:40.18 | harryvv | focks its not nice |
| 20:40.26 | harryvv | I dont like its looks. |
| 20:40.30 | syle | sipura is kewl but doesn;t the cisco 7960 support more like 8 lines? |
| 20:40.45 | harryvv | for 129 you can get a really good phone by buying a polycom ip300 |
| 20:40.55 | focks | the grandstream 2000s are more like an executive type phone than the 841 |
| 20:41.12 | x9net | http://voipstore.atacomm.com/shops/ViewItem.aspx/27934028032-36931158528.htm |
| 20:41.17 | x9net | that the one your talking about? |
| 20:41.35 | harryvv | 841 is a cheap ip home phone :) even my uniden looks alot nicer then that phone. |
| 20:41.46 | syle | omg |
| 20:41.53 | *** join/#asterisk Cassador (cass@81.193.137.71) |
| 20:41.53 | syle | you got the uniden cordless phone! |
| 20:41.57 | syle | tell me about it please |
| 20:41.58 | Cassador | Salute |
| 20:42.15 | harryvv | I found a polycom supplier here that can sell me the same priiced phone as in the states and its a local reseller. |
| 20:45.20 | syle | hmm reading some thread on the internet, ip phone is same as SIP phone i thought |
| 20:45.44 | *** part/#asterisk braniff (~44r@braniff.user) |
| 20:45.53 | syle | yeah dude i think the cisco 7960 is the way to go |
| 20:45.53 | harryvv | ugg syle, its just a standard analog phone running off a ata. But thay do make nice looking phone. |
| 20:46.17 | Qwell | Is it difficult to mod phones? |
| 20:46.35 | Qwell | This mute button only works when depressed...I want to make it like a switch |
| 20:46.36 | cursor | no - just paint it green and attach a spoiler |
| 20:46.41 | focks | like to run Tetris on them> |
| 20:46.42 | syle | i am thinking the cisco 7960 and the uniden cordless phone for my house, since wireless is just unreliable half the time |
| 20:46.57 | Qwell | lame, no screws |
| 20:47.00 | cursor | The 7960 mute is a toggle |
| 20:47.07 | Qwell | cursor: This is a POS analog |
| 20:47.24 | syle | cursor : you american or canadian? |
| 20:47.27 | cursor | put the caller on hold instead of mute |
| 20:47.29 | Qwell | momentary switches for mute is rediculous |
| 20:47.41 | cursor | that'll latch, and it can play music too |
| 20:47.43 | Qwell | cursor: conference call, I need to hear, but not always talk |
| 20:47.55 | *** join/#asterisk NewSole (dave@i216-58-44-245.avalonworks.net) |
| 20:48.03 | Qwell | is it possible to (un)mute from the * CLI? |
| 20:48.11 | syle | yeah that is gay hmmmm |
| 20:48.19 | syle | mute has to work on conferance calls for sure |
| 20:48.19 | focks | to make wireless reliable, simply add wires ;) |
| 20:48.27 | cursor | haha |
| 20:48.27 | Qwell | syle: indeed |
| 20:48.46 | Micc | I'm having problems with DTMF tones generated from asterisk. I assume asterisk is suppose to generate it because its getting sent as an IAX2 frame then has to convert to inband for my dialout SIP line. |
| 20:48.47 | ManxPwr | syle, No! There are many VoIP protocols. SIP is just one of them. |
| 20:48.50 | syle | i am taking a shit in the bathroom half the time on conferance calls |
| 20:48.56 | syle | noone wants to hear that lol |
| 20:48.57 | cursor | to increase wireless security - simply add wires |
| 20:48.59 | focks | yikes |
| 20:49.26 | cursor | You should have a toilet room then |
| 20:49.28 | harryvv | btw, is there a phone that can be setup with a button to "make bussy" ? say a really hated goverment agency that deals with the public can have this feature configured and if a cutomer service agent is getting nasty calls from a clinet say..he owes taxes he/she can put him on busy by activating that button? |
| 20:49.34 | cursor | Nobody wants to see a turd in the bath |
| 20:50.11 | syle | harryvv i;ve done that with asterisk |
| 20:50.17 | focks | i usually take reserve all shit-taking to video conference calls |
| 20:50.37 | *** part/#asterisk Seyr (~Seyr@rrcs-24-227-133-226.sw.biz.rr.com) |
| 20:50.40 | harryvv | syle but with a botton or say a dtmf ? |
| 20:50.41 | syle | callerid match him then answer ; congestion |
| 20:50.59 | *** join/#asterisk Trakk (~Trakk@adsl-10-248-231.mia.bellsouth.net) |
| 20:50.59 | harryvv | what if its one uniqe call every week? |
| 20:51.05 | cursor | and wait for the overdue tax fines |
| 20:51.05 | syle | you mean realtime? |
| 20:51.10 | harryvv | yes |
| 20:51.12 | syle | or permanent block his number |
| 20:51.40 | harryvv | syle, I did some work at a tax office ..thought it was interesting and a neet feature. |
| 20:51.47 | eKo1 | The client will just call you from a payphone and screw you over. |
| 20:51.51 | harryvv | syle..dealing with 5 million customers? |
| 20:51.54 | syle | i am not sure how you could do it as call is comming in since to even use the phone at that point would just answer his call |
| 20:52.00 | syle | hmmmm |
| 20:52.07 | syle | i think i could code something that would do that though |
| 20:52.20 | harryvv | ek01, canadians are not that radical |
| 20:52.21 | harryvv | :) |
| 20:52.40 | eKo1 | not yet... |
| 20:52.41 | syle | interface with agi and perl i probabably could |
| 20:52.45 | harryvv | I would call the button the "just simmer down button" :) |
| 20:53.10 | cursor | Or the "wind that customer right up" button |
| 20:53.11 | niZon | how can you get ANI info in asterisk? |
| 20:53.20 | syle | i don;t think there is anyway way to do it on the phone though while a call is comming in |
| 20:53.23 | syle | from your computer you could |
| 20:53.56 | syle | just click something on your computer and that interfaces with asterisk and you could block it |
| 20:54.19 | syle | unless.... |
| 20:54.23 | syle | you had 2 ports |
| 20:54.55 | syle | its a real bitch to do harry lol |
| 20:54.56 | eKo1 | niZon: I thought that's just the callerid. |
| 20:55.08 | syle | easiest way is just to block the number from further calling you hehe |
| 20:55.29 | ManxPwr | Many IP phones have a Do=Not-Disturb option |
| 20:55.49 | *** part/#asterisk extremis (~extremis@cpe-24-175-55-177.houston.res.rr.com) |
| 20:55.56 | harryvv | syle, say the reciving party saw the caller id..then pressed the button. |
| 20:56.08 | harryvv | syle..but thay cannot..thay owe taxes ;) |
| 20:56.40 | syle | yes but the problem is after they have called , a dial plan already is executing for the call |
| 20:56.42 | harryvv | Manx, including the ip300 |
| 20:57.01 | doolph | anyone here know how to fix chan_oh323.so issues |
| 20:57.02 | harryvv | syle, true. |
| 20:57.14 | bannerman | man you could just pay your taxes :-P |
| 20:57.42 | Uther_P | heh, most people don't bother trying to use h323 |
| 20:57.43 | syle | i think only way would be to modify the c source |
| 20:57.49 | syle | of asterisk |
| 20:57.53 | harryvv | bann...people who owe taxes dont have access to these agencies like I do. |
| 20:57.55 | syle | before the bridge takes place |
| 20:58.04 | bannerman | ah |
| 20:59.10 | mikewho2 | what IPphone do you guys recommend |
| 20:59.12 | mikewho2 | the cisco one? |
| 20:59.22 | Uther_P | yea |
| 20:59.27 | Uther_P | the $500+ cisco phones |
| 20:59.28 | kajtzu | is it possible to pass other parameters than just the conference number to Application: Meetme through the manager interface? |
| 20:59.31 | Uther_P | heh |
| 20:59.35 | bannerman | I like the Polycom IP500 |
| 20:59.42 | mikewho2 | really, how much did that run ya bannerman? |
| 20:59.46 | bannerman | $200 |
| 20:59.47 | syle | well i don;t think hes talking about taxes, unless he doesn;t own his own house or he'd be screwed hehe, thinking maybe collection agency |
| 20:59.47 | kajtzu | setting Data: 1234,,1234 doesn't seem to work |
| 21:00.05 | Qwell | After a dial, there isn't any way my dialplan could catch DTMF from me, is there? |
| 21:02.30 | *** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc) |
| 21:02.31 | *** topic/#asterisk is Asterisk: The Open Source PBX || Astricon Europe -- Madrid, Spain -- June 15-17 || PBX Developers Conference in Chicago Aug 3-5 Register Now! http://www.asterlink.com/cluecon.htm |
| 21:02.32 | syle | naw |
| 21:02.32 | cursor | or drill a hole and add a new switch |
| 21:02.39 | Qwell | There are no screws...I'd have to bust it |
| 21:02.41 | syle | this is software level |
| 21:02.47 | cursor | cut the mic line and wire it through a switch |
| 21:03.17 | cursor | you could firewall the RTP packets being sent to/from your phone :-) |
| 21:03.22 | Qwell | heh |
| 21:03.33 | syle | i would inform cisco to update the IOS software |
| 21:04.02 | syle | they need an if condition if the phone is not currently muted, mute=off |
| 21:04.27 | Qwell | why would cisco care about my $5 radio shack analog phone? |
| 21:04.46 | cursor | They'll try to sell you a 7960 |
| 21:04.51 | syle | talking about cursor's 7960 cisco phone hehe |
| 21:04.55 | cursor | and one of their PBXs :-) |
| 21:04.57 | Qwell | Whats wrong with his phone? |
| 21:05.13 | Qwell | He was describing the "proper" solution. |
| 21:05.18 | cursor | My phone is great |
| 21:05.31 | Qwell | I should just splurge, and get a $10 phone |
| 21:05.33 | syle | how much you pay for it? |
| 21:05.47 | syle | its about 300 bucks right? |
| 21:05.59 | cursor | The phone on my desk was $100 from eBay |
| 21:06.08 | Qwell | a 7960? |
| 21:06.11 | cursor | yes |
| 21:06.14 | Qwell | not bad at all |
| 21:06.15 | syle | nice |
| 21:06.19 | cursor | It was a good deal |
| 21:06.22 | Qwell | defective repaired? |
| 21:06.27 | cursor | no |
| 21:06.30 | Qwell | hmm |
| 21:06.30 | cursor | just second-hand |
| 21:06.31 | Qwell | rare |
| 21:06.42 | Qwell | used ones are almost the price of a new one usually |
| 21:06.56 | cursor | I see them cheap every now and again |
| 21:07.07 | syle | http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=16164&item=5195438989&rd=1 |
| 21:07.08 | cursor | It's best to jump in and get one cheap - even if you don't need it right now |
| 21:07.17 | Qwell | pick me up one for < $150, and I'll pay you for it later :p |
| 21:07.27 | cursor | :-) |
| 21:07.35 | krw | hello all, i'm getting a seg fault when I call a url with a querystring from CURL. Anyone know what might cause this? |
| 21:07.38 | PTG123 | anyoen know of a decient firewall distribution based on bsd? |
| 21:07.45 | cursor | $150 is like £0.0000002 |
| 21:07.50 | cursor | British Pounds |
| 21:07.55 | syle | cursor |
| 21:08.01 | cursor | That'll be me |
| 21:08.02 | syle | what version IOS you got the phone up to now? |
| 21:08.08 | syle | 7.4? |
| 21:08.10 | cursor | 7.1, I think |
| 21:08.14 | krw | ptg123: check out http://www.m0n0.ch/wall |
| 21:08.23 | syle | only 7.1 ? you have cisco login? |
| 21:08.30 | cursor | nope |
| 21:08.41 | syle | hmm that sucks |
| 21:09.00 | cursor | I'll update it when Cisco fix their XML stuff |
| 21:09.05 | cursor | so I can code softkeys etc. |
| 21:09.19 | syle | http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=61840&item=5774800814&rd=1 |
| 21:09.20 | PTG123 | krw: there a web based setup for this? |
| 21:09.22 | syle | 100 bucks |
| 21:09.39 | syle | so if anyone wants a good phone there you go hehe |
| 21:09.40 | krw | ptg123: yep, works very well. there are screenshots on the web site |
| 21:10.22 | PTG123 | krw: looks good.. any recommendations for micro pc's to run it on? |
| 21:10.25 | syle | cursor |
| 21:10.28 | syle | you use POE? |
| 21:10.36 | PTG123 | krw: does it do QOS? |
| 21:10.44 | loud | i wonder if 7.0.3 supports g726 |
| 21:10.59 | cursor | no |
| 21:11.06 | cursor | I have a power "brick" |
| 21:11.23 | cursor | POE would be nice, but it's not worth the price |
| 21:11.34 | harryvv | syle 100 bucks times 2 |
| 21:11.48 | harryvv | its not going to be 100 dollars I am cirtain of that. |
| 21:12.10 | harryvv | cursor, for you or customers |
| 21:12.32 | cursor | both |
| 21:12.49 | cursor | POE isn't really worth the price |
| 21:12.55 | syle | i wonder if my gigabit switch supports POE |
| 21:12.55 | cursor | unless you happen to have it already |
| 21:12.59 | *** part/#asterisk Uther_P (~uther_p@66.180.120.83) |
| 21:13.01 | cursor | or your switch comes with it already |
| 21:13.02 | Qwell | You can't do gbe and poe |
| 21:13.09 | Qwell | both use 4 pairs |
| 21:13.28 | focks | which module is used for TDM400? |
| 21:13.32 | cursor | brb... |
| 21:13.34 | Qwell | focks: wctdm |
| 21:13.47 | wt | syle: is it over $2k? |
| 21:13.47 | focks | that's right |
| 21:14.00 | Cassador | srry m8s..., for use sip only , how many conf should I config? |
| 21:14.26 | focks | Qwell in additon to zaptel right? |
| 21:14.31 | Qwell | focks: yes |
| 21:14.34 | Cassador | Extensions.conf, Sip.conf ? |
| 21:14.49 | focks | Qwell hmm, just hangs when i modprobe it |
| 21:15.47 | *** part/#asterisk Patrick^ (~patrickm@birch4.mountaincable.net) |
| 21:17.31 | blitzrage | Cassador: and rtp.conf if you need to limit the number of ports Asterisk listens on for RTP |
| 21:18.24 | bannerman | Anyone use Nufone? |
| 21:18.31 | bannerman | Major problems today, more than any other day |
| 21:18.39 | Qwell | bannerman: works fine here. What problems? |
| 21:19.00 | bannerman | Qwell: Calls fade out or drop completely |
| 21:19.02 | Cassador | thanx blitzrage :) ..., Its the first time that Im trying to do anything in it..., I just want to make some kind of test with xlite |
| 21:19.19 | *** join/#asterisk p1tst0p (~Will@82-38-104-189.cable.ubr03.donc.blueyonder.co.uk) |
| 21:19.20 | Qwell | bannerman: sure its not your connection? It's been fine for me all day |
| 21:19.21 | bannerman | Qwell: I've had problems for weeks, but just today it's been particularly bad |
| 21:19.40 | bannerman | Qwell: Connection is lovely, and I talked to another fellow today who has the same exact problems. |
| 21:20.03 | bannerman | qwell: what's your ping like? |
| 21:20.36 | Qwell | 73 to switch-1, 90 to switch-2 |
| 21:20.43 | bannerman | hm, worse than mine |
| 21:20.48 | bannerman | I get 62 |
| 21:20.50 | bannerman | to switch-1 |
| 21:21.56 | bannerman | I was going to ask you about your setup, but there really isn't much that I haven't tried |
| 21:22.07 | bannerman | 2.4 and 2.6 kernels, 1.0.7 and CVS-HEAD on both kernels |
| 21:22.21 | bannerman | with firewall + qos, without, disconnected the computer network completely for a day to ensure that it was a clean connection |
| 21:22.37 | bannerman | I use sip phones.. polycom ip500s and some cheapoes |
| 21:24.57 | syle | i have a good question: ata adapters obviously can;t do this but lets say routers: where your dsl or cable modem plugs in...anyone offer QOS on them or traffic shaping, Ie: if your downloading stuff, it will cut that download speed and give more bandwidth to your call if you pickup line |
| 21:26.27 | syle | currently i think only more high end cisco routers do this |
| 21:26.45 | Qwell | my $50 router does QoS |
| 21:27.01 | RaYmAn-Bx | syle: Sipura SPA 2100 can actually do that..It works as a router though..you plug in internet in one end and all your computer in the other |
| 21:27.18 | Gh0sty | buy linksys, load some hacked firmware and you got a wonderfull router :) |
| 21:27.48 | syle | so the sipura is doing NAT as well? |
| 21:28.00 | RaYmAn-Bx | I'm not actually sure, but I would assume so |
| 21:28.32 | HoopyCat | g'nite folx |
| 21:28.39 | syle | is it actualy a router? ie: you can log into it, map ports, have switch plugged into it fine, do firewalling and support xbox live? |
| 21:29.38 | RaYmAn-Bx | check the site :) I'm not entirely sure..I just remember that it's supposed to be able to do that :) I don't have one myself |
| 21:29.57 | blitzrage | QoS isn't traffic shaping though |
| 21:30.01 | Micc | So I'm going to assume I've got to generate my own DTMF tones when using the IAX protocol because FRAME_DTMF doesn't seem to work. |
| 21:30.05 | *** join/#asterisk laotan (~jesse@H38.C18.B96.tor.eicat.ca) |
| 21:30.42 | Micc | Asterisk sends a garbled tone that sounds almost like two or three shorter tones. |
| 21:30.53 | Micc | Could this have something to do with timing? |
| 21:31.37 | syle | yeah what we need is to traffic shape fxs ports |
| 21:32.36 | Micc | hmmm... I think I might have found my problem is with voicepulse and timing. |
| 21:33.35 | RaYmAn-Bx | the SPA 2100 obviously sends it's own traffic first and then whatever traffic other people needs..so essentially it does do traffic shaping. But it's simple and only prioritizes it's own traffic..I don't think it does anything else (and only upstream traffic) |
| 21:36.11 | Micc | nope, false alarm. voicepulse is something else. |
| 21:37.07 | syle | not sure |
| 21:37.26 | syle | anyone charging over 1.2 cents a minute is gay though hehe |
| 21:39.14 | Beirdo | I always though being gay had to do with sexual preference, not how much you charge for VoIP |
| 21:39.20 | syle | ray: i think sipura is selling great things, but i am not explaining to end users how they have to go buy a switch cause sipura only supports 1 ethernet port hehe |
| 21:39.35 | Qwell | Beirdo: What you charge for service is based on sexual preference, I think is what he's implying |
| 21:39.41 | Beirdo | hehe |
| 21:39.44 | syle | i guess that is why everyone is going with linksys, dlink etc |
| 21:40.58 | syle | for the tech savy it don;t matter, they only want 1 port and link up with their 10 port gigabit switch, but most people don;t do that hehe |
| 21:42.12 | juanjoc | Does anyone know if spandsp can be used on 64 bit Linux running on an AMD64 machine? |
| 21:42.40 | juanjoc | I'm mainly interested in the stable version of spandsp (0.0.1k) |
| 21:43.15 | syle | 64 bit linux damn, you guy unnecessarily expensive hardware :) |
| 21:43.19 | syle | buy |
| 21:43.41 | *** join/#asterisk Cresl1n (~matt@216.207.245.23) |
| 21:43.41 | Qwell | Whats wrong with 64bit? |
| 21:44.03 | jontow | too many bytes! |
| 21:44.06 | jontow | *cough* |
| 21:44.07 | juanjoc | spandsp is not compiling correctly. |
| 21:44.18 | *** join/#asterisk christo (~chris@195.82.111.57) |
| 21:44.27 | *** join/#asterisk xai (~pasta@cpe-70-112-17-10.austin.res.rr.com) |
| 21:44.44 | *** join/#asterisk EdwinL (edwin@phoenix.officegeneral.com) |
| 21:44.45 | syle | considering i could probably outdo your 64 bit machine for same price with 2 32 bit machines and a load balancer you tell me :) |
| 21:44.45 | xai | Is there a standard cdr tracking app for billing? |
| 21:45.27 | *** join/#asterisk wolfson (~hehe@68-187-185-225.dhcp.mant.nc.charter.com) |
| 21:46.06 | syle | xai i had same question actually |
| 21:46.25 | EdwinL | greetings |
| 21:46.38 | syle | whats everyone using for account billing with asterisk? |
| 21:46.50 | Qwell | app_cdr? |
| 21:47.04 | *** join/#asterisk bjohnson (~bjohnson@66.11.188.235) |
| 21:47.15 | syle | qwell: where can i find documents on that function? |
| 21:47.22 | Qwell | google? |
| 21:47.26 | xai | I just am curious what most people consider stable/standard/etc.. |
| 21:47.28 | Qwell | I don't know. I probably made it up |
| 21:48.20 | bannerman | Qwell: While you're enjoying newbie questions, do you have any idea what might be wrong on my side to get these calls to drop so often? |
| 21:48.33 | Qwell | bannerman: not sure... |
| 21:48.46 | Qwell | tried calling them? |
| 21:48.49 | bannerman | Qwell: hahaha. |
| 21:48.54 | bannerman | Qwell: Good one. |
| 21:49.05 | Qwell | what? |
| 21:49.10 | bannerman | Qwell: Multiple voicemails and emails, no answer. |
| 21:49.42 | bannerman | Qwell: I think I'm too little for thme. |
| 21:50.16 | *** join/#asterisk darwin35 (~darwin35@24.3.226.147) |
| 21:50.31 | bannerman | Qwell: The best response I've had yet has been here. Usually I Greg or Jer will give me a thing or two to try, then stop responding. |
| 21:50.39 | *** part/#asterisk darwin35 (~darwin35@24.3.226.147) |
| 21:50.53 | *** join/#asterisk darwin35 (~darwin35@24.3.226.147) |
| 21:51.01 | darwin35 | http://pastebin.ca/11926 |
| 21:51.51 | kb1_kanobe | bannerman: what is the configuration that's dropping calls? also, which code base - stable or head? |
| 21:52.44 | syle | qwell so where are the app_cdr files stored? |
| 21:53.11 | bannerman | kb1_kanobe: I've used two systems. One is a single P3, used 2.6 kernel there, tried both 1.0.7 and CVS-HEAD. Other platform is 2.4, dual athlon, had 1.0.7, upgraded to CVS-HEAD at Greg's request which made no difference. |
| 21:53.20 | *** join/#asterisk sleepy_one (~chatzilla@cpe-24-166-32-45.neo.res.rr.com) |
| 21:54.11 | kb1_kanobe | how are the calls travelling? Ie. PRI->IAX, SIP->IAX...? |
| 21:54.18 | sleepy_one | hello all :) is there a way I can extract the number called to reach asterisk when I get calls on my PRI? I need to route calls based on the number dialed to reach the system. tia |
| 21:54.47 | sleepy_one | I have a T1 PRI with a Digium T100p wct1xxp card |
| 21:55.01 | syle | why? |
| 21:55.31 | EdwinL | i have trouble configuring the zaptel driver on a server with both TE405P & TDM400P cards. it kept giving me ZT_CHANCONFIG fail for various reasons no matter how i change the span/channel# around. |
| 21:55.43 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 21:55.54 | kb1_kanobe | sleepy_one: calls arriving PRI implicitly contain the dialed number - all PRI trunks are effectively DID lines. Go study the wiki for a while... |
| 21:56.04 | sleepy_one | have you executed lspci to see if the cards are detected? |
| 21:56.05 | Cassador | I have this message: "Registrations from 'Sipphone <sip:teste@10.0.02> failed for '10.0.0.1' |
| 21:56.15 | Cassador | What this means?? Or what I need to check? |
| 21:56.53 | xai | syle: i don't see anything that is exceptional yet. Do you? |
| 21:57.00 | darwin35 | your sip.conf and the config in the phone |
| 21:57.07 | syle | when you do show modules....use count=1 or 0 does that mean if its enabled or not? |
| 21:57.19 | darwin35 | make sure the username and passwd match |
| 21:57.44 | Cassador | Darwin35: ok..., I will check |
| 21:57.58 | EdwinL | if i put only the config for 1 card and load only the driver for that card. they |
| 21:58.09 | syle | no buy i am wondering if app_cdr = 0 on show modules means its not being used |
| 21:58.12 | EdwinL | they'll work nut not both at teh same time. |
| 21:58.27 | bannerman | kb1_kanobe: SIP -> IAX |
| 21:59.16 | bannerman | I've used every combination of protocols on both sides, used some cheap SIP phones, some polycom ip500 and some softphone.. exact same behavior |
| 21:59.45 | focks | is there anything I can do for echo cancellation and hollow-sounding analog calls with a TDM400? |
| 22:01.51 | *** join/#asterisk santiago (~santiago@63.245.86.146) |
| 22:04.06 | *** part/#asterisk anderiv (~anderiv@207-67-87-34.gen.twtelecom.net) |
| 22:06.20 | *** join/#asterisk Sedorox (brandon@sedorox.staff.smartserv) |
| 22:09.59 | *** join/#asterisk Rawplayer (kevin@oom-killer.org) |
| 22:10.34 | *** part/#asterisk eKo1 (~bernd@metrored-gw.tropicohn.com) |
| 22:12.08 | *** join/#asterisk shmaltz (~chatzilla@ool-43551098.dyn.optonline.net) |
| 22:12.39 | *** join/#asterisk Manipura (~chatzilla@dsl-hunt-66-18-224-38-cgy.nucleus.com) |
| 22:14.44 | *** part/#asterisk Rawplayer (kevin@oom-killer.org) |
| 22:15.31 | Manipura | Does anyone have any comments on Link2Voip.. I'm thinking about switching to them for DID's & termination. |
| 22:15.41 | syle | xai |
| 22:15.42 | *** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 22:15.45 | syle | i think i found something |
| 22:16.00 | syle | ---/var/log/asterisk/cdr-csv/Master.csv |
| 22:16.12 | syle | ---/var/log/asterisk/cdr-custom/Master.csv |
| 22:16.23 | syle | just need something to get these into mysql |
| 22:16.51 | outtolunc | syle cvs co asterisk-addons |
| 22:17.43 | syle | kewl |
| 22:17.54 | syle | then how do i tell it to log to mysql for app_cdr? |
| 22:18.23 | xai | syle: yea. |
| 22:18.31 | *** join/#asterisk zyke (~zakforeve@84.45.132.117) |
| 22:18.41 | xai | syle: sorry been too busy,. |
| 22:18.57 | *** join/#asterisk lattice (~lattice@d64-180-160-215.bchsia.telus.net) |
| 22:19.39 | *** join/#asterisk marky (~emes@216.206.77.241) |
| 22:19.53 | marky | anybody willing to lend me some thoughts |
| 22:20.02 | marky | my sip debug says i can't authorize |
| 22:20.20 | marky | and i'm pretty sure i've got all my nat taken care of |
| 22:20.31 | eivindtr | marky: Then it's probably right... |
| 22:20.42 | Nuxi | marky, when are you going to return that thought? |
| 22:21.07 | marky | tomorrow |
| 22:21.09 | marky | high noon |
| 22:21.17 | Nuxi | that'll work. |
| 22:21.48 | zyke | what is a Zap/pseudo channel? is that what's used for meetme? |
| 22:21.52 | syle | xai |
| 22:21.55 | syle | i think i get idea |
| 22:22.20 | syle | install addons first , then include module in modules file, then modify cdr_custom with database info |
| 22:22.33 | xai | syle: http://www.voip-info.org/wiki-Asterisk+billing and |
| 22:23.29 | eivindtr | Anyone have any tips on how to control timing? I have a Sangoma PRI card, but the sound is bad. I want to use ztdummy instead. |
| 22:23.58 | xai | syle: and http://www.voip-info.org/wiki-Open+Source+Billing+Systems |
| 22:24.08 | eivindtr | marky: Got some more info to go with that thought? |
| 22:24.19 | syle | ahh good research dude |
| 22:24.42 | syle | i was searching unix server itself hehe |
| 22:25.45 | syle | perfect |
| 22:25.49 | syle | http://www.voip-info.org/wiki-Asterisk+config+cdr_mysql.conf |
| 22:25.53 | syle | thats what i was looking for |
| 22:25.59 | syle | talk to you later xai got to go |
| 22:26.19 | zyke | what is a Zap/pseudo channel? is that what's used for meetme? |
| 22:28.30 | marky | eivindtr, i'm lookin at some stuff |
| 22:28.44 | marky | just going over nat with * |
| 22:28.57 | marky | i'm just trying to get a call going out |
| 22:29.02 | marky | such a small mission. |
| 22:29.32 | eivindtr | well, helping you is not such a small mission with info like that. What client do you use? |
| 22:29.50 | marky | client? |
| 22:29.53 | marky | sip device? |
| 22:29.56 | zyke | anyone knows how to stop a Zap/pseudo ? |
| 22:30.27 | eivindtr | yeah, any of those.. |
| 22:30.57 | marky | handytone 486 |
| 22:31.01 | sleepy_one | cya all l8r |
| 22:31.25 | marky | sorry, i'm mid-call |
| 22:31.40 | marky | handytone, but a@h still won't auth with my ISP |
| 22:32.27 | eivindtr | marky: never used a@h. Make sure you don't mess up user name with auth name. SIP is a bit wierd about those. Safe bet is to keep them the same... |
| 22:32.55 | zyke | anyone knows how to stop a Zap/pseudo channel? |
| 22:33.08 | outtolunc | stop now <G> |
| 22:34.00 | eivindtr | Anyone knows how to fine-select timing source in the *-config? I want to use ztdummy even though I have a zaptel device... |
| 22:34.01 | zyke | outtolunc: stop now Zap/pseudo ? |
| 22:34.33 | *** part/#asterisk makhtar (~ageller@mail.bulletinnews.com) |
| 22:34.35 | outtolunc | no, 'stop now' from the CLI |
| 22:35.09 | outtolunc | there is NO WAY to kill it.. zap destroy will skip right over it (unless a new way has been added in the last couple weeks) |
| 22:35.15 | zyke | outtolunc: but stop now would shut down asterisk .. |
| 22:36.00 | outtolunc | yeah it would |
| 22:36.44 | zyke | outtolunc: it's causing the box to run at 99% |
| 22:36.56 | *** join/#asterisk likwid-- (likwid@nc-65-173-73-231.dyn.sprint-hsd.net) |
| 22:37.11 | outtolunc | i've already told you what you need to do, if you somehow can't get that.. then live with it <G> |
| 22:38.06 | *** join/#asterisk DannyF (~dannyf@h197n2fls32o865.telia.com) |
| 22:41.54 | outtolunc | chan_local.c line 321 local_destroy() has been commented out |
| 22:45.13 | outtolunc | and the only thing that ever gets near local_hangup is using the 'glaredetect' option |
| 23:10.18 | *** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc) |
| 23:10.18 | *** topic/#asterisk is Asterisk: The Open Source PBX || Astricon Europe -- Madrid, Spain -- June 15-17 || PBX Developers Conference in Chicago Aug 3-5 Register Now! http://www.asterlink.com/cluecon.htm |
| 23:10.30 | marky | hmmm |
| 23:10.35 | marky | that's weird |
| 23:10.41 | niZon | ~seen boris |
| 23:10.53 | jbot | boris <boris@S01060040ca1e5b54.wp.shawcable.net> was last seen on IRC in channel #asterisk, 3d 1h 44m 49s ago, saying: 'what?'. |
| 23:10.53 | marky | guess i haven't irc'd for a short while |
| 23:11.05 | niZon | laggy bot |
| 23:11.30 | *** part/#asterisk juanjoc (~juanjoc@200.73.189.82) |
| 23:13.32 | *** join/#asterisk bah (048830696@ACAD2286.ipt.aol.com) |
| 23:20.35 | *** join/#asterisk outtolunc (~me@adsl-69-110-50-162.dsl.pltn13.pacbell.net) |
| 23:29.38 | *** join/#asterisk terrapen (~cjs@fw-01.satx.bikeworld.net) |
| 23:30.06 | *** part/#asterisk terrapen (~cjs@fw-01.satx.bikeworld.net) |
| 23:31.46 | *** join/#asterisk MikeJ[Laptop] (~jayden@pcp02795302pcs.roylok01.mi.comcast.net) |
| 23:33.26 | zyke | anyone knows what a Zap/pseudo channel is? |
| 23:35.08 | marcus5 | its a fake zap channel |
| 23:35.14 | marcus5 | used as a timing source for meetme/moh and such |
| 23:35.27 | marcus5 | if an asterisk server doesnt have real zap channels |
| 23:36.36 | outtolunc | .. /usr/src/asterisk/doc/localchannel.txt as a description |
| 23:40.36 | *** join/#asterisk Syncros (~sysop@noc.routermonkey.net) |
| 23:42.06 | zyke | thanks marcus5 |
| 23:42.18 | zyke | and outtolunc |
| 23:42.27 | *** join/#asterisk tsp (~tyler@S01060080c825173c.vc.shawcable.net) |
| 23:42.30 | tsp | My alsa problems are solved! |
| 23:42.41 | tsp | had to set both input and output to hw:0,0 - they were commented out before |
| 23:42.57 | tsp | now - I need a good free iax provider - fwd keeps canceling my cals |
| 23:43.00 | tsp | calls |
| 23:44.28 | tsp | anyone got a free iax provider? I'm behind a nat firewall |
| 23:44.51 | zyke | tsp: how do u mean free provider? |
| 23:45.20 | tsp | well - I only want to call other users of sip, not outbound pstn - maybe toll free though |
| 23:46.22 | zyke | i think virbiage got such service |
| 23:46.33 | zyke | www.virbiage.com |
| 23:46.43 | *** join/#asterisk JoshuaTree (~not@100.mw.wgl.dcsi.net.au) |
| 23:46.50 | *** join/#asterisk harryvv (~leonardo@S010600a0c93f6f7e.vs.shawcable.net) |
| 23:47.54 | tsp | not them - I can't use firefly |
| 23:47.57 | JoshuaTree | Has anyone used SIPPS with Asterisk? |
| 23:48.49 | doolph | sipps? |
| 23:49.12 | JoshuaTree | yeah it's made by ahead its a Windows SIP Client |
| 23:50.16 | doolph | if it is sip client it should be supported |
| 23:51.00 | JoshuaTree | well i can get it to work local but from the outside world it fails |
| 23:52.09 | JoshuaTree | and asterisk goes crazy |
| 23:52.20 | JoshuaTree | or maybe its the software |
| 23:52.24 | JoshuaTree | i don't know |
| 23:53.11 | PTG123 | define crazy |
| 23:53.26 | JoshuaTree | i logs in the account |
| 23:53.31 | JoshuaTree | then it logs it out |
| 23:53.52 | JoshuaTree | i don't know if its SIPPS logging it out or Asterisk |
| 23:54.36 | JoshuaTree | on the SIPPS client the logged IP address it is getting is the asterisk box, then it switches to 127.0.0.1 |
| 23:56.10 | JoshuaTree | as for example it gets sip:USERNAME@10.0.0.1:5060 then switches to sip:USERNAME@127.0.0.1:5060 |
