| 00:00.05 | twisted[asteria] | it's actually, like, HD |
| 00:00.33 | file[laptop] | oh oh is house on... maybe |
| 00:00.41 | *** join/#asterisk nwhit (n=nwhit@ns1.whittrio.com) |
| 00:01.12 | nwhit | does anyone know if you can use sip subscriptions with call parking? |
| 00:01.31 | file[laptop] | dang nabbit where is it... |
| 00:01.40 | *** part/#asterisk bob_too (n=chris@rrcs-24-153-179-246.sw.biz.rr.com) |
| 00:01.40 | ender | nwhit: huh? |
| 00:01.57 | file[laptop] | maybe it's on at 10... |
| 00:02.00 | ender | n3u7: SuSE? what, are you a sadist? |
| 00:02.25 | nwhit | ender, you know ... exten => 701,hint,Park/701 something like that |
| 00:02.41 | ender | nwhit: oh sorry, no, I don't use hint stuff. |
| 00:02.54 | nwhit | so when you park a call, the light appears on phones that support subscriptions |
| 00:04.23 | hardwire | blah |
| 00:04.26 | hardwire | talk to me |
| 00:04.52 | hardwire | nwhit: yes.. |
| 00:04.56 | hardwire | using devstate w/ bristuff patches |
| 00:05.10 | hardwire | you can make a proactive manager to change the devstate of parked lines |
| 00:05.35 | hardwire | something that really needed to make it to CVS :) |
| 00:06.08 | nwhit | hardwire, from what i saw the bristuff and devstate was somewhat unstable |
| 00:06.24 | hardwire | you can't really pick and choose on this one. |
| 00:06.33 | nwhit | so... they are unstable |
| 00:08.22 | hardwire | yeh.. I somehow made it this far :) |
| 00:08.25 | hardwire | without using it |
| 00:08.34 | hardwire | trick is.. you train people to do things the smart way. |
| 00:08.35 | nwhit | some customers can get over not having "Hey John, pickup line 2" and they press the little button on the phone |
| 00:08.58 | hardwire | nwhit: of course.. |
| 00:09.23 | file | 'datz a key system |
| 00:09.31 | hardwire | and it can be emulated |
| 00:09.36 | file | to a certain extent |
| 00:09.42 | hardwire | if you can codezor |
| 00:10.04 | nwhit | codezor? |
| 00:10.15 | hardwire | file: you are weird. |
| 00:10.19 | hardwire | http://bugs.digium.com/view.php?id=5425 |
| 00:10.30 | hardwire | or.. are you not joshnet. |
| 00:10.33 | nwhit | yeah, but that is good target market for hosted pbx |
| 00:10.50 | hardwire | nwhit: yup |
| 00:10.55 | hardwire | and snom makes a product for it :) |
| 00:11.17 | nwhit | i played with 4s, yuck |
| 00:11.17 | hardwire | I need to like.. go meditate. |
| 00:11.19 | syle | i don;t care much for sip phones myself, to lazy to run ethernet to every phone in the house, what kind of analog phones support MWI? |
| 00:11.27 | hardwire | nwhit: they addressed what you want :) |
| 00:11.28 | Katty | file: comfy? |
| 00:11.36 | hardwire | syle: most adsi |
| 00:11.56 | hardwire | and you shouldn't sacrifice functionality for wite |
| 00:11.59 | hardwire | wite/wire |
| 00:12.04 | hardwire | thats just dumb. |
| 00:12.04 | file | yay comfy |
| 00:12.29 | hardwire | I stayed at a Motel 6 last night |
| 00:12.31 | hardwire | its 2 years old |
| 00:12.38 | hardwire | the window is already busted on the locking part |
| 00:12.43 | hardwire | the wind almost killed the window last night. |
| 00:12.51 | nwhit | hardwire, i should look into how to implement it natively |
| 00:12.56 | syle | hows a sip phone any better than a analog phone hooked up to your channel bank at home? |
| 00:13.00 | hardwire | and somebody was repairing their truck outside our window with air tools. |
| 00:13.09 | hardwire | syle: there are ways. |
| 00:14.04 | *** join/#asterisk techie (n=gus@adsl-068-209-242-072.sip.mia.bellsouth.net) |
| 00:14.04 | syle | ... |
| 00:14.37 | syle | maybe for a hotel room |
| 00:14.39 | hardwire | syle: there are ways! |
| 00:14.49 | ender | syle: for us, the sip phones were cheaper than the phones we needed for an Avaya PBX, they had much more capacity (no longer a key system), greater functionality (contact lists, distinctive ring, easy forwarding, dnd, etc...), and since we were doing a remodle of our office it was cheaper to just run 2 ethernet lines to each desk instead of an ethernet and analog line. |
| 00:14.50 | *** join/#asterisk dalfry (n=vaibhav@santa.vsharma.net) |
| 00:15.08 | ender | data closet is much easier to manage too. |
| 00:15.33 | ender | sound quality is better too |
| 00:15.56 | syle | still can do all that with asterisk+channel bank |
| 00:16.45 | ender | syle: all those functions were on the phone itself. |
| 00:17.02 | hardwire | syle: if I bought a bunch of analog phones for my house.. I would be stuck with analog equipment for 10+ years |
| 00:17.03 | syle | see thats the problem |
| 00:17.06 | ender | syle: easy to read LCD readout for forwarding stuff, for contact lists, for distinctive ring based on contact list, etc... |
| 00:17.13 | syle | now what if you want to share your call lists between all the phones |
| 00:17.28 | hardwire | syle: not that hard w/ provisioning |
| 00:17.38 | ender | syle: plus the configs can be held via ftp so if you have to replace a phone, just change the file name to match the new mac address and bam, exactly the way it was before. |
| 00:17.53 | ender | syle: copy them into the ftp file that all phones get. |
| 00:18.23 | syle | hehe |
| 00:18.32 | syle | i guess thats ok for the average voip person |
| 00:18.41 | syle | i hope developpers are actaully using channel banks though |
| 00:18.55 | ender | why? |
| 00:19.04 | syle | gives you more experience |
| 00:19.08 | syle | can run PRI lines with ease |
| 00:19.09 | syle | etc |
| 00:19.16 | ender | why would developers of VOIP be looking backward to older archaich hardware that is deprecated? |
| 00:19.38 | ender | syle: drop the PRI directly into the Asterisk box. Connect phones via ethernet. |
| 00:20.28 | ender | if I never had to deal w/ another channel bank and punchdown tool and silly little wires, I'd be a happy admin. |
| 00:20.48 | syle | i just think channel banks are kewler lol |
| 00:20.59 | ender | clean patch panels, patch ethernet cables, neatly routed to managed switches, etc... |
| 00:21.13 | *** join/#asterisk supaigtr (n=yurplsl@152.53.17.1) |
| 00:21.37 | syle | i;ve heard from many callcenters channel bank route is still better quality |
| 00:22.03 | syle | makes no sense to me either |
| 00:22.09 | supaigtr | sup ppl |
| 00:22.12 | ender | Our phone vendor tried to say that too, because they didn't know about * and how it worked. THey knew analog and they wanted to be paid to do analog. |
| 00:23.01 | *** join/#asterisk bugz (n=bugz@cpe-24-27-67-66.houston.res.rr.com) |
| 00:23.05 | ManxPower | Analog is simple, easy, has a horrible user interface. |
| 00:23.12 | bugz | i love asterisk <3 |
| 00:23.14 | ManxPower | oh, it's also reliable. |
| 00:23.18 | ender | supaigtr: channel bank vs voip |
| 00:23.20 | bugz | anyone have a bumper sticker out? |
| 00:23.38 | ender | ManxPower: sure, until one part of a cable gets pulled out and you have to scratch your head wondering wtf is going on. |
| 00:23.43 | syle | i just haven;t seen many sip cordless phones |
| 00:23.50 | supaigtr | channel bank = works 99.9998 VOIP = 98.00% + ppl aren't happy |
| 00:23.53 | ender | syle: voip-supply has a bunch |
| 00:23.58 | ManxPower | ender, Huh? You use a toner and probe and trace it back. |
| 00:24.19 | ManxPower | supaigtr, i've found that people are not happy with remembering and dialing *codec |
| 00:24.24 | ManxPower | *codes |
| 00:24.26 | ender | ManxPower: assuming my manager gave me budget for one. |
| 00:24.29 | syle | + zapbarge is alot better than chan_spy |
| 00:24.36 | ManxPower | ender, Less than $100. |
| 00:24.39 | syle | i;ve crashed box on chan_spy many times |
| 00:24.43 | ender | ManxPower: you don't know my manager (: |
| 00:24.50 | ManxPower | If you don't have an extra $100 in the budget, quit and go work for a real company. |
| 00:24.54 | supaigtr | ManxPOwer: We push PRI to a panasonic, sylantro, or avaya. |
| 00:24.58 | syle | suppose i should submit those bugs one of these days |
| 00:25.06 | *** part/#asterisk bugz (n=bugz@cpe-24-27-67-66.houston.res.rr.com) |
| 00:25.13 | lilneon | later everyone thnx for your help |
| 00:25.14 | ender | ManxPower: it's not a matter of 'in the budget' it's a matter of 'necessary' |
| 00:25.17 | supaigtr | We have a few * and pure IP deploys and most aren't happy. |
| 00:25.21 | ender | ManxPower: and I did quit that company and went to work for Real. |
| 00:25.26 | ManxPower | ender, if you go analog it's required. |
| 00:25.26 | *** part/#asterisk lilneon (n=tj_r3@cuscon12932.tstt.net.tt) |
| 00:25.51 | ManxPower | Just like if you go with VoIP then a 110 punch down tool is required. |
| 00:26.23 | ManxPower | ebay has lots of good stuff too. |
| 00:26.48 | *** join/#asterisk apardo (n=w0w0@238.Red-83-43-218.dynamicIP.rima-tde.net) |
| 00:27.06 | marc324 | whats a good sip ip phone? |
| 00:27.07 | *** join/#asterisk bugz (n=bugz@cpe-24-27-67-66.houston.res.rr.com) |
| 00:27.18 | ender | supaigtr: are you talking about VOIP to a DID providor? I'm talking about VOIP to peoples desks, and a PRI to * |
| 00:27.30 | ender | marc324: Polycom makes decent phones... |
| 00:27.32 | supaigtr | ender: Yea VOIP to the desk. |
| 00:27.38 | supaigtr | Polycom or SNOM. |
| 00:27.39 | bugz | polycoms phones are OK |
| 00:27.48 | bugz | NTP on them is not fun.. |
| 00:27.50 | ender | supaigtr: huh. My users are all happy, but it's a small deployment. |
| 00:28.05 | ender | bugz: huh? I've had 0 issues w/ time sync. |
| 00:28.06 | hardwire | hmm |
| 00:28.15 | bugz | ender: good for you ;) |
| 00:28.21 | supaigtr | ender: Seems like 8-12 phones works 20 -50 in a network that has internet access and things go all to hell. |
| 00:28.22 | bugz | ender: how do you deal with the dhcp issues? |
| 00:28.43 | bugz | ender: like other boxes stealing IP's or getting DHCP from the wrong server? |
| 00:28.44 | ender | supaigtr: we have 50~ phones on a dedicated network. |
| 00:28.54 | ManxPower | At $100/drop we decided not to install 2x as many ethernet ports in our new office. |
| 00:28.54 | ender | bugz: we put phones on their own network. |
| 00:28.57 | *** join/#asterisk gh0st (i=njzelp@your.vote.counts.votewithabullet.org) |
| 00:29.04 | ender | bugz: otherwise you can do mac address dhcp mapping. |
| 00:29.07 | supaigtr | ender: That works good but at that price in most buildings a digital key system is much cheaper. |
| 00:29.33 | syle | with the rhino analog side connects to a 25-pair amphenol, which is called RJ21 you supply the punch down block with a female mating connector |
| 00:29.35 | ender | supaigtr: we were remodeling our offices anyway, so it was same price for phone lines vs ethernet lines. |
| 00:29.58 | supaigtr | ender: Still right now you can't beat some of the modern key systems with *. |
| 00:30.21 | ender | price wise we could. |
| 00:30.42 | ender | and I'll be damned if I"m going to deal w/ another hardware PBX where I have no f'ing decent interface to it. |
| 00:30.50 | marc324 | what digital key system? |
| 00:30.58 | syle | if you are scared of punchdown blocks then run to a sip phone hehe |
| 00:31.17 | bugz | ender: so you put the phones on their own network but how do you keep them from being plugged in to the same network as all the PC's and still have a different dhcpc server |
| 00:31.18 | supaigtr | How? I mean I can have a full call center up in a day with 100 phones realiably. GUI interfaces proven setups. * seems to take weeks and tons of debugging and no GUI for configurations. |
| 00:31.31 | bugz | ender: without having to configure the DHCP for the PC's.. |
| 00:31.40 | ender | bugz: if the user plugs the phone into the wrong network, then the phone doesn't work. Pretty simple. |
| 00:31.54 | bugz | ender: ok so you are looking at alot of cable pulling |
| 00:32.04 | znoG | has anyone had experience unlocking sipura units? |
| 00:32.17 | SarahEmm | sipuras are locked? |
| 00:32.17 | bugz | ender: complete physical isolation from the PC network |
| 00:32.18 | ender | bugz: as I said, we were getting all new cable pulled anyway. |
| 00:32.21 | znoG | looks like my provider has password protected the factory reset option in the IVR |
| 00:32.26 | SarahEmm | ahh. |
| 00:32.33 | bugz | ender: ok, well, what if you are THE provider ;) |
| 00:32.48 | ender | bugz: if they were on the same physical network, then we'd do mac address matching for the phones, and do dhcp static IP addressing for the phones. |
| 00:32.54 | bugz | ender: if you really stop and think about it, it gets pretty complicated |
| 00:33.21 | supaigtr | ender: U using polys |
| 00:33.27 | ender | supaigtr: yes. |
| 00:33.29 | bugz | ender: in that case, how do you keep the phones from pulling an IP from the PC DHCP server? |
| 00:33.41 | ender | bugz: there is only ONE DHCP server. |
| 00:33.50 | ToR\L | or vlans |
| 00:33.50 | ToR\L | hehe |
| 00:33.55 | supaigtr | Got buddy lists working with that DHCP scheme or no DSS functions? |
| 00:33.56 | ender | bugz: that DHCP server matches mac addresses of the phones and assigns them different IPs than the PCs would get. |
| 00:34.09 | supaigtr | VLAN is much simpler. |
| 00:34.09 | bugz | ToR\L, ender , you are still looking at alot of overhead |
| 00:34.09 | ender | supaigtr: we don't support IM or buddy lists. |
| 00:34.15 | supaigtr | No DSS then? |
| 00:34.17 | syle | well i disagree on using non-asterisk...make your own GUI's...but channel banks better than sip phones yes i agree, faster setups using existing phone wire |
| 00:34.18 | bugz | ender: people calling you back saying they cant print... |
| 00:34.27 | ender | bugz: huh? can't print? |
| 00:34.53 | ender | supaigtr: can you vlan when you have a single ethernet port at a desk that has to run both the PC and the SIP phone? |
| 00:35.00 | supaigtr | syle: U talking about commercial system GUIs? |
| 00:35.06 | supaigtr | ender: yes. |
| 00:35.11 | bugz | ender: what im getting at is, there has to be a way to get dhcp to the phones so you dont have to configure each one - without having to administer their entire network for them |
| 00:35.15 | ender | supaigtr: based on mac address right? |
| 00:35.22 | syle | you stated you would rather use another PBX than asterisk correct? |
| 00:35.33 | supaigtr | ender: No just set vlan on phone and setup switch to notice tagged traffic. |
| 00:35.56 | ender | bugz: right. One DHCP server services the PCs as well as the phones. |
| 00:36.13 | supaigtr | syle: For production business systems yes. * isn't there yet. We have deployed it to a number of customers and 3 out of 50 are happy. |
| 00:36.14 | bugz | ender: even then you are looking at mac mapping |
| 00:36.18 | bugz | not fun |
| 00:36.29 | ender | syle: I think it depends on a location to location basis. New location rollouts I wouldn't install phone wire. Just ethernet. |
| 00:36.35 | syle | i;ve actually done some research on that issue, and results are...americans still feel usiong commercial PBX solutions are the best, while in european market, people are much more familiar with linux and use asterisk a great deal more |
| 00:36.40 | *** join/#asterisk alephcom (n=Miranda@207.34.97.130) |
| 00:36.53 | SarahEmm | supaigtr: you keep deploying it even though 47 of 50 customers aren't happy? |
| 00:36.57 | ender | bugz: I can think of a lot of things that are even less fun. |
| 00:37.04 | bugz | ender: i hear that |
| 00:37.20 | bugz | ender: we have a sip connect to moscow right now |
| 00:37.25 | bugz | ender: lots of fun... |
| 00:37.40 | supaigtr | syle: Its not a matter of comfort. Its a matter of what works. Its not just asterisk. When mitel rolled out systems on a PC it was a failure. Ppl aren't used to rebooting the phones or debugging core OS/software issues. |
| 00:37.44 | ender | bugz: mac mapping is fairly easy, especially since phones from a vendor tend to have a unique matching point. So you can do group and mask matches rather than individual entries for each phone. |
| 00:37.44 | *** join/#asterisk PocketIRC (n=pocketir@S01060050da6df072.sc.shawcable.net) |
| 00:38.28 | syle | i don;t see how that applies to analog phones and channel banks |
| 00:38.31 | supaigtr | SarahEmm: Most of those customers have multisite and they have evaluated the cost of tieing those sites together. * works for that and is cheap. Just cause they aren't happy doesn't mean they would pay more to be happy. |
| 00:38.36 | Renacor | anybody know where I can get the xml example for the polycom setup cfg's? |
| 00:38.40 | bugz | ender: i just need to learn a little more about it i guess - i still cant get over the whole physical aspect of it while keeping yourself isolated from "their network problems" |
| 00:38.48 | supaigtr | Renacor: PRC |
| 00:38.52 | syle | but level of comfort yes |
| 00:38.54 | bugz | know what i mean? |
| 00:38.54 | ender | bugz: right now, I have to go sip -> Asterisk -> IAX2 -> Asterisk -> Fujitsu PBX -> T1 for inbound/outbound calling. That was 'fun' to setup. |
| 00:39.05 | Renacor | supaigtr: PRC==? |
| 00:39.05 | supaigtr | bugz: you can setup a VLAN and have seperate DHCP for it. |
| 00:39.06 | bugz | ender: wow. |
| 00:39.13 | bugz | ender: can you even hear them? |
| 00:39.30 | ender | bugz: calls are crystal clear. |
| 00:39.39 | bugz | supaigtr: yeah thats a part of the .org im trying to get into their heads, to spend some money on a programmable switch |
| 00:39.44 | bugz | supaigtr: or lemme at a gentoo router |
| 00:39.45 | syle | asterisk in front of another pbx |
| 00:39.45 | supaigtr | Polycom resource center. |
| 00:39.46 | syle | why |
| 00:39.46 | ender | bugz: clearer than the avaya system our people use now. |
| 00:39.51 | Renacor | supaigtr: thanks |
| 00:40.03 | supaigtr | bugz: Dlink etc make cheap VLAN switches. |
| 00:40.08 | supaigtr | Np |
| 00:40.09 | bugz | ender: amazine |
| 00:40.25 | supaigtr | ender: Ulaw is very clear till you loose packets. |
| 00:40.35 | n3u7 | update:ububtu install failed on line 462 of netdb.h |
| 00:40.39 | bugz | .. my favorite is the dual and triple gateway systems |
| 00:40.44 | ender | bugz: the real problem was that the person we had to deal w/ on the Fujitsu side didn't understand enough about it to help me figure out the signaling for the T1 line between us and it. |
| 00:40.45 | n3u7 | sched.o error |
| 00:40.57 | ender | supaigtr: we just make sure we don't lose packets. |
| 00:40.57 | *** join/#asterisk USM (n=hengff@61.6.65.226) |
| 00:41.00 | bugz | ender: that made me laugh dude |
| 00:41.05 | bugz | ender: i know what you mean |
| 00:41.16 | n3u7 | SuSE9.3 install unsuccessful on account of Slang problem |
| 00:41.19 | n3u7 | great |
| 00:41.20 | USM | hi |
| 00:41.21 | syle | did you post config on voip-info.org ender? |
| 00:41.21 | *** join/#asterisk wolfson` (n=hehe@usr-kdh-208-6-58-26.beachlink.com) |
| 00:41.24 | ender | supaigtr: eventually we'll have sip phones -> asterisk -> PRI |
| 00:41.28 | n3u7 | another scripting language! |
| 00:41.29 | supaigtr | ender: possible on small network but very easy for something to happen and cause problems. |
| 00:41.36 | ender | syle: I found a posting there that helped me. |
| 00:41.43 | hardwire | yay |
| 00:41.46 | hardwire | my bug was merged |
| 00:41.50 | ender | supaigtr: yep, we're a small network. |
| 00:41.55 | hardwire | I feel happy |
| 00:42.12 | ender | n3u7: do you have a love for SuSE or are you just using it because somebody said (with agerman accent) that it was the best Linux? |
| 00:42.38 | USM | Could someone tell me the CVS HEad is for which version? 1.2X or 1.0X? |
| 00:42.39 | ender | syle: basically I just had to use em_wink signaling. |
| 00:42.44 | supaigtr | ender: We have a full lab of this stuff. The biggest problem we've had is deploying digital PRI to * in rural areas. Echo is a problem cause of the copper loops calling in. Echo can isn't quite there yet. |
| 00:43.07 | ender | supaigtr: even with the hardware cancellers? |
| 00:43.11 | n3u7 | ender: I'm using it because I used SuSE7.3 for a year with little or no problems |
| 00:43.15 | ender | supaigtr: luckilly we're in metro area. |
| 00:43.30 | ender | n3u7: hrm, good luck w/ that. I can't _stand_ suse. |
| 00:43.44 | n3u7 | ender: and because my ubuntu install is failing also |
| 00:43.49 | supaigtr | I have the digium ecan card there are problems with it. (no disrespect to digium) but there cards are problematic and the ecan doesn't do much at all. |
| 00:44.12 | syle | asterisk runs stable as shit on redhat 9.0, fedora seems not bad |
| 00:44.38 | supaigtr | syle: stable = 99.9999 in telecom |
| 00:45.09 | syle | well i watched a video cam at what OS core asterisk builders used |
| 00:45.10 | ender | n3u7: hahah. Well.... |
| 00:45.10 | syle | it was fedora |
| 00:45.19 | n3u7 | :/ |
| 00:45.40 | ender | We use CentOS |
| 00:45.47 | ender | Love that Red Hat koolaid. |
| 00:45.50 | syle | i;ll stick to what the developpers use |
| 00:45.55 | ender | but it's a bit more stable than Fedora. |
| 00:46.15 | ender | I like a 5 year lifespan vs a 12~18 month. |
| 00:46.20 | alephcom | USM: Head would be 1.2X I believe. |
| 00:46.28 | syle | yeah and when noone maintains centos anymore and it falls apart like the rest then what |
| 00:46.32 | ender | although I use Fedora on my desktop and run the Fedora Legacy project. |
| 00:46.40 | syle | think whitelinux died recently to |
| 00:46.52 | syle | dude felt he would rather work more at his dayjob or something |
| 00:47.04 | ender | syle: well, I'll cross that road when I come to it. HOwever I know a lot of the people involved w/ CentOS and I have a good feeling it will be around for quite a while. |
| 00:47.22 | syle | well i hope so for your sake :) |
| 00:47.23 | ender | syle: yeah, wbl was a one man operation and he wasn't very into community involvement. |
| 00:47.29 | Los415 | if your looking for the support and to be keeped up then just pay for rhe |
| 00:47.50 | ender | CentOS is much more of a community effort that is gaining the popularity of the Univerities, something that WBL never had. |
| 00:47.55 | syle | fedora is a whole shit load of developpers, its gonna stay so thats why i use it |
| 00:48.07 | syle | if it wasn;t fedora i;d run freebsd |
| 00:48.36 | ender | syle: don't get me wrong, I like Fedora. I just don't feel that it has the stability / longevity to run my core servers. |
| 00:48.44 | syle | although i don;t trust freebsd on multiprocessor machines quite yet |
| 00:49.01 | ender | I don't want to OS upgrade every 6~9 months, nor do I want to roll my own security updates after Fedora and Legacy drop it. |
| 00:49.09 | USM | alephcom: thanks |
| 00:49.17 | syle | you don;t feel? alot of universities i know of are backended on fedora |
| 00:49.34 | syle | tons of traffic |
| 00:49.54 | ManxPower | Note to self: check your junk mail folder more often -- there might be something important in it. |
| 00:49.55 | ender | syle: and even if CentOS goes away, the RHEL srpm updates are still there. it's a rebuild away from being on my CentOS system. |
| 00:50.14 | ender | syle: yep. Some universities use Gentoo as well. Nobody is perfect. |
| 00:50.27 | syle | lol that must be 1 or 2 |
| 00:50.40 | ender | unfortunately more than that :p |
| 00:51.32 | syle | they must have cut school funding |
| 00:51.40 | supaigtr | Anyone here insteresting in SER->*->MaxTNT config? |
| 00:51.41 | syle | used to be sparc solaris boxes and shit |
| 00:51.48 | ender | syle: there are a lot of uni's that are still using RHL9 for their backends. THey just gave up on security updates. |
| 00:52.09 | *** join/#asterisk azzie_ (i=az@cpe-24-168-17-173.si.res.rr.com) |
| 00:52.15 | alephcom | supaigtr: I have a client that wants to point his MaxTNT to *, got info on that? |
| 00:52.22 | ender | ah, rack anchors. Put a sun server at the bottom of your rack and that thing ain't going NOWHERE. |
| 00:52.29 | Los415 | supaigtr do you ever get a click click click sound from the tnt's when running with * |
| 00:52.32 | syle | i am still using rh9 on a production box |
| 00:52.39 | ender | syle: so am I |
| 00:52.54 | ender | syle: but then again, I run Fedora Legacy project so I know what security issues are out there. |
| 00:53.04 | *** part/#asterisk USM (n=hengff@61.6.65.226) |
| 00:53.10 | ender | actually, my system is RHL 7.3, not 9 |
| 00:53.23 | syle | i run fc3 and 4 at home for testing |
| 00:53.27 | Los415 | if it aint broke dont touch it |
| 00:53.35 | syle | actaully fc4 box is mostly dedicated to mythtv |
| 00:53.47 | ender | Los415: if you don't touch it for too long, somebody will touch it for you. |
| 00:54.02 | *** join/#asterisk dalfry (n=vaibhav@66.250.170.114) |
| 00:54.13 | ender | syle: hehe, yep, FC4 on my myth box. Fc4 on my shuttle, and on my ibook. CentOS on my other servers. |
| 00:54.29 | ender | I just havne't had the opportunity/energy to rebuild my web/mail system w/ CentOS |
| 00:54.40 | syle | i still have freebsd 4.x on my laptop |
| 00:54.42 | syle | eeks |
| 00:55.00 | syle | dual booting with XP of course |
| 00:55.18 | Los415 | yea i'm a freebsd 4.x fan my self got that running on a couple things |
| 00:55.25 | Los415 | web/mail related |
| 00:55.53 | syle | i have to upgrade to freebsd 5.x eventually 4.x don;t support 32 bit wireless cards |
| 00:56.01 | alephcom | Los415: How has your luck with FreeBSD and * been? Back when I started * was not working on FreeBSD so I had to do this linux stuff. |
| 00:56.19 | Los415 | i havent even tried * with freebsd |
| 00:56.27 | syle | me either |
| 00:56.36 | Los415 | i'm in the same boat as you |
| 00:56.47 | syle | i didn;t want to jump into freebsd with asterisk till i heard some great success stories |
| 00:56.57 | Los415 | yup |
| 00:57.05 | syle | devel goes on in linux so i just stick with that |
| 00:57.08 | alephcom | cool. Someday, I'll try it out. My boxes on running fedora. :-( That's me too. To far to the colo center. |
| 00:57.08 | Los415 | and i've been fairly happy with * on centos |
| 00:57.31 | syle | i;ve been happy with asterisk on fedora and rh9 |
| 00:57.41 | Los415 | yea i was doing everything on rh9 |
| 00:57.46 | Los415 | then started using centos |
| 00:58.04 | syle | centos use the 2.6 kernel? |
| 00:58.06 | ender | natural progerssion |
| 00:58.11 | ender | syle: CentOS 4 is yes. |
| 00:58.11 | alephcom | Is it any better? Notice any differences between centos and fc |
| 00:58.24 | Los415 | i'm still using centos 3.4 |
| 00:58.29 | Los415 | 2.4 kernel |
| 00:58.33 | syle | centos has less developpers :) |
| 00:58.36 | n3u7 | hmm |
| 00:58.47 | Los415 | at the time for my core server i was using gfs |
| 00:58.56 | Los415 | to connect to my fiber raid arrary |
| 00:59.02 | gaggaman | hi! |
| 00:59.05 | gaggaman | could maybe somebody help me with my bristuffed asterisk 1.09 and call Pickup? |
| 00:59.07 | Los415 | at the time rhe4 or centos 4 didnt support it |
| 00:59.14 | alephcom | gotcha |
| 00:59.19 | gaggaman | when I do a Pickup, i get: |
| 00:59.19 | syle | fiber array raid at home? |
| 00:59.26 | litage | is it safe to allow only tcp for rsync, or does rsync often use udp? |
| 00:59.26 | *** part/#asterisk scoates (n=sean@iconoclast.caedmon.net) |
| 00:59.27 | Los415 | hehe |
| 00:59.27 | gaggaman | <PROTECTED> |
| 00:59.28 | Los415 | no |
| 00:59.32 | Los415 | in our office |
| 00:59.33 | gaggaman | <PROTECTED> |
| 00:59.34 | syle | why not go raid 5 on sata |
| 00:59.37 | syle | ohhh |
| 00:59.39 | gaggaman | <PROTECTED> |
| 00:59.40 | ender | syle: um, |
| 00:59.47 | ender | syle: CentOS has as many developers as RHEL4. |
| 00:59.49 | Los415 | we have a full on raid array with fiber switches ect |
| 00:59.58 | ender | syle: which is slightly _more_ than Fedora on the Red Hat side. |
| 01:00.02 | Los415 | using gfs |
| 01:00.04 | Los415 | if you know what that is |
| 01:00.12 | syle | hmmm that is kewl |
| 01:00.14 | hardwire | opengfs |
| 01:00.18 | Los415 | and i know fedora at the time didnt support gfs |
| 01:00.22 | ender | syle: the CentOS folks just rebuild the RHEL packages and put it out as CentOS. |
| 01:00.22 | syle | i still like /etc/rc.d/init.d and /etc/sysconfig formats |
| 01:01.20 | Los415 | hardwire yea that be it |
| 01:01.27 | syle | well i never liked /etc/rc.d/init.d |
| 01:01.40 | Los415 | that stuff is really confussing if you never played with it |
| 01:01.45 | syle | much prefered one startup script in /usr/local/etc like fbd |
| 01:01.51 | syle | but /etc/sysconfig i like |
| 01:01.55 | Los415 | def not something i would recommend to someone who's never been able to use it |
| 01:02.07 | Los415 | it would be expensive just to learn to get all the gear to do it |
| 01:02.18 | ender | chkconfig makes init.d stuff very easy to work with. |
| 01:02.49 | syle | hmmm |
| 01:02.52 | ender | We teach people about chkconfig and not really about all the symlinks and such in /etc/rc.d/init.d/ and that works well. |
| 01:03.01 | syle | well i;ll have to checkout centos, seems very kewl |
| 01:03.05 | syle | they use yum for updates? |
| 01:03.11 | ender | yep |
| 01:03.27 | syle | garbage |
| 01:03.52 | syle | not up to quality to rhino;s now |
| 01:03.55 | syle | people toss them |
| 01:04.32 | ManxPower | LOL! |
| 01:04.42 | ManxPower | We buy like 1 total access per month off ebay |
| 01:04.54 | syle | that was a joke , quality wasn;t though |
| 01:05.21 | ManxPower | Now we are buying 1 Cisco 5500 series per month these days |
| 01:05.38 | supaigtr | syle: U have problems with the adtran? |
| 01:06.22 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 01:07.59 | *** join/#asterisk drumkilla (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 01:07.59 | *** mode/#asterisk [+o drumkilla] by ChanServ |
| 01:08.07 | ManxPower | supaigtr, We have standardized on TA750/TA850 |
| 01:08.21 | ManxPower | Mainly for integration with legacy PBXs |
| 01:08.24 | syle | As far as refurbished units go from non-rhino's you have to ask yourself, if those units will do the job for you, then why did someone give them up to eventually be called refurbished? You guessed it, they are inferior and is why most likely they sold that unit, and most likely why they have a Rhino. Lesson learned. |
| 01:08.30 | supaigtr | FXO or FXS? |
| 01:08.58 | ManxPower | supaigtr, FXS for connecting to Legacy PBX FXO/CO ports. |
| 01:09.15 | ManxPower | syle, mainly because people replaced their PBX and went CT1 or PRI. |
| 01:09.21 | supaigtr | Any problems with noise before the dial? |
| 01:09.32 | ManxPower | our CELC uses all Adtran TA750/TA850 and that's good enough for us. |
| 01:09.35 | ManxPower | supaigtr, no. |
| 01:09.53 | ManxPower | supaigtr, the few times we've seen that was traced down to a bad punchdown |
| 01:10.07 | supaigtr | Yep. or impedence |
| 01:10.16 | *** part/#asterisk Miggidy (n=Miggidy@203-59-9-189.perm.iinet.net.au) |
| 01:10.19 | supaigtr | We use adtran all over. |
| 01:10.36 | supaigtr | I like there support. Don't need much on the 750/850 |
| 01:11.05 | syle | rhino.s have excellent support, they'll even ssh into your system for you and fix anything |
| 01:11.49 | ManxPower | I'm sure they do. |
| 01:12.24 | syle | as far as i know those adtran models are no longer being made or supported right? |
| 01:12.33 | ManxPower | Are they also about $300 for the channel bank + power supply + battery backup + 24 FXS + T-1 port to telco, plus drop and insert DSX-1 port? |
| 01:12.49 | ManxPower | the TAs are cheap enough we can keep a couple of spares around. |
| 01:13.22 | syle | refurbished ones are |
| 01:13.23 | ManxPower | syle, It doesn't matter to us. they are cheap enough we can just replace it if it blows up, and we've not had any blow up. |
| 01:13.37 | ManxPower | Same for the Cat 550x switches. |
| 01:13.41 | syle | good to know |
| 01:13.46 | ManxPower | syle, What's the refurbished you are talking about? We just buy them used. |
| 01:13.53 | syle | can;t find them on ebay anymore though |
| 01:14.14 | supaigtr | 750 and 850 are still sold last time I ordered. |
| 01:14.59 | ManxPower | syle, This is the first time i've not been able to find TA750's for a good price on ebay. |
| 01:15.03 | ManxPower | supaigtr, they are. |
| 01:16.55 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
| 01:17.59 | syle | does anyone know if your account on paypal has to be verified in order for IPN to work? |
| 01:23.02 | *** join/#asterisk littleball (n=littleba@bb219-75-114-108.singnet.com.sg) |
| 01:23.40 | n3u7 | anyone kn ow how to compile asterisk without libnewt? |
| 01:23.46 | n3u7 | for astman |
| 01:36.37 | *** part/#asterisk techie (n=gus@adsl-068-209-242-072.sip.mia.bellsouth.net) |
| 01:37.48 | *** join/#asterisk nassy (n=nassy@207-38-252-103.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com) |
| 01:38.05 | *** join/#asterisk pressure_man (n=pressure@ip-202-37-228-1.internet.co.nz) |
| 01:38.33 | pressure_man | hi, can somebody please help me clarify the difference between peer and user? |
| 01:38.43 | pressure_man | i've read up on them, and am more confused than ever |
| 01:39.17 | syle | peer would be like your VOIP provider |
| 01:39.46 | pressure_man | ok, my set up has an asterisk box, and a cisco sip gateway loaded with 2 BRI WICs |
| 01:40.01 | pressure_man | for making outgoing calls from my network via the cisco, i need type=peer, right? |
| 01:40.21 | nassy | i just downloaded asterisk at home and cant wait to install it. i have one problem though. i dont have a monitor. is there anyway to connect to computer during the install process to answer any prompts. |
| 01:40.36 | syle | i;d set your gateway as a friend if you want anything comming in hehe |
| 01:40.56 | pressure_man | ok, but lets say i had a separate type=friend entry... |
| 01:41.14 | pressure_man | the type=peer would be for outgoing calls from my network to PSTN |
| 01:41.25 | pressure_man | oop i mean separate type=user |
| 01:41.40 | file[laptop] | user is only used for username/password authentication on inbound calls |
| 01:41.41 | pressure_man | the type=user would be for calls coming in from PSTN via the cisco to my asterisk box... right so far? |
| 01:41.43 | file[laptop] | most gateways don't do that |
| 01:42.04 | file[laptop] | they expect you to do IP based authentication |
| 01:42.19 | syle | yeah so peer is only thing that would work anyway |
| 01:42.29 | file[laptop] | a peer with insecure=very |
| 01:42.51 | pressure_man | ok... but am i right in my understanding that type=peer is for asterisk to place calls to gateway xyz, and type=user is for gateway xyz to palce a call to the asterisk dialplan? |
| 01:42.55 | *** join/#asterisk wunderkin (n=wunderki@VDSL-130-13-234-137.PHNX.QWEST.NET) |
| 01:43.05 | file[laptop] | it depends |
| 01:43.13 | *** part/#asterisk wunderkin (n=wunderki@VDSL-130-13-234-137.PHNX.QWEST.NET) |
| 01:43.15 | *** join/#asterisk wunderkin (n=wunderki@VDSL-130-13-234-137.PHNX.QWEST.NET) |
| 01:43.16 | timecop | i was always wondering |
| 01:43.24 | pressure_man | ok, now i'm confused |
| 01:43.30 | timecop | if youre always on laptop why should we see your retarded long nick with brackets on my irc window? |
| 01:43.40 | file[laptop] | a user is only used if it's doing username and password authentication, and stuff like Cisco 5300s or TNTs or providers, don't do username and password authentication when sending a call to you |
| 01:43.44 | file[laptop] | timecop: I'm not |
| 01:43.46 | timecop | should I add [furiously_masturbating] to my nick each time I do so? |
| 01:43.48 | file[laptop] | I'm at my desk during the day :P |
| 01:44.13 | pressure_man | hmm. |
| 01:44.43 | pressure_man | ok, the problem i have is that dtmfmode seems to be completely ignored, unless i get the gateway (in this case, a Voiceblue GSM gateway) to register |
| 01:45.08 | file[laptop] | match it as a peer with insecure=very, and set dtmfmode in there |
| 01:45.12 | pressure_man | and the only way i can get the GSM gateway to register is if i create a type=friend, with host=dynamic |
| 01:45.42 | MikeJ[Laptop] | I have a laptop too :P |
| 01:45.45 | pressure_man | i can't get it to register with a type=user and host=10.10.10.21 - instead it just arrives as a sip guest |
| 01:46.00 | pressure_man | (and uses the global setting for dtmfmode) |
| 01:46.05 | file[laptop] | just do what I said |
| 01:46.09 | file[laptop] | [voiceblue] |
| 01:46.10 | file[laptop] | type=peer |
| 01:46.13 | file[laptop] | host=it's IP |
| 01:46.13 | pressure_man | insecure=very sounds scary |
| 01:46.14 | file[laptop] | insecure=very |
| 01:46.16 | file[laptop] | dtmfmode=rfc2833 |
| 01:46.20 | file[laptop] | context=wherever |
| 01:46.33 | file[laptop] | and I'm allowed to do stuff like above, it's educational and I have seniority :P |
| 01:46.49 | supaigtr | seni.. who? |
| 01:46.51 | file[laptop] | pressure_man: it tells Asterisk to match based on the originating IP address of the SIP message |
| 01:46.59 | pressure_man | ok, erm, is there any disadvantage to using type=friend, for that kind of thing? |
| 01:47.09 | file[laptop] | well, your user entry will never be used |
| 01:47.14 | file[laptop] | so technically you're using extra memory for nothing |
| 01:47.17 | pressure_man | i mean, we place calls to this GSM gateway, and we also get calls from it. |
| 01:47.26 | file[laptop] | am I not getting through here? |
| 01:47.34 | pressure_man | type=friend combines both user and peer, so i drop the type=user record |
| 01:47.34 | file[laptop] | your GSM gateway probably does not send calls with a username and password |
| 01:47.40 | file[laptop] | the only thing that really does, is SIP phones |
| 01:47.47 | file[laptop] | so just use a peer. |
| 01:47.51 | file[laptop] | you can do all this in a single peer entry |
| 01:48.20 | pressure_man | the gsm gateway has the option to register |
| 01:48.43 | file[laptop] | registrations don't effect authentication or anything |
| 01:48.44 | pressure_man | it's basically working fine as type=friend... it's just that i've read that type=friend is evil, and i'm wondering why |
| 01:49.01 | file[laptop] | meh whatever, do what you wish |
| 01:49.26 | pressure_man | i'm just confused as hell by the documentation i've read so far - a lot of it offers conflicting information |
| 01:49.49 | file[laptop] | peers are used for sending calls out, or receiving calls (based on IP address matching) |
| 01:49.59 | file[laptop] | users are used for receiving calls (based on username+password) authentication |
| 01:50.10 | pressure_man | wow... that clarifies a lot just there |
| 01:50.15 | MikeJ[Laptop] | and fish are used for eating |
| 01:50.29 | MikeJ[Laptop] | spoons for digging |
| 01:50.30 | n3u7 | so much for SuSE |
| 01:50.41 | n3u7 | this will be my third os in week |
| 01:50.41 | MikeJ[Laptop] | and dogs for meeting girls |
| 01:50.41 | pressure_man | why hasn't somebody written that in the wiki? |
| 01:50.48 | file[laptop] | pressure_man: who knows |
| 01:50.50 | MikeJ[Laptop] | pressure_man, go for it... |
| 01:50.55 | file[laptop] | I'm a wealth of information though :P |
| 01:51.09 | MikeJ[Laptop] | yes, useful and useless |
| 01:51.13 | pressure_man | so ~is~ there anything wrong with just using type=friend? |
| 01:51.16 | *** join/#asterisk Koshatul (n=evangeli@ip157.net65.ipnetworks.net.au) |
| 01:51.29 | file[laptop] | it probably won't match for incoming calls from the Voiceblue unit |
| 01:51.38 | file[laptop] | thus why your dtmfmode isn't effective on inbound calls |
| 01:51.52 | pressure_man | no, it's in fact the ~only~ way it works correctly |
| 01:52.10 | pressure_man | (that i've found so far) |
| 01:52.51 | file[laptop] | ah so it does do username/password authentication? |
| 01:53.05 | file[laptop] | if you do a sip debug it'll verify it... |
| 01:53.08 | pressure_man | yes, if you tell the voiceblue to register |
| 01:53.10 | file[laptop] | Note: most gateways don't do that |
| 01:54.02 | pressure_man | it fails to auth if i don't config it to register |
| 01:54.14 | pressure_man | i'm using sip realtime, so i haven'trule out bugs with that |
| 01:54.29 | file[laptop] | does callerid work? ;) |
| 01:54.38 | pressure_man | yep |
| 01:55.09 | file[laptop] | do you have trustrpid=yes in sip.conf? |
| 01:55.29 | pressure_man | nope |
| 01:55.40 | file[laptop] | are you sure it's matching on the user? :) |
| 01:55.47 | pressure_man | yes |
| 01:55.54 | file[laptop] | well lemme tell you something |
| 01:55.58 | pressure_man | if i change its context, it ends up in a different context |
| 01:56.11 | file[laptop] | chan_sip figures out the username based on the username in the From header |
| 01:56.16 | file[laptop] | guess what that username in the From header is also used for |
| 01:56.28 | MikeJ[Laptop] | caller id! |
| 01:56.28 | file[laptop] | callerid! |
| 01:56.32 | MikeJ[Laptop] | woo hoo |
| 01:56.45 | MikeJ[Laptop] | mmmmmm |
| 01:57.01 | pressure_man | uhm |
| 01:57.03 | file[laptop] | pressure_man: interesting setup you have |
| 01:57.10 | file[laptop] | pressure_man: have fun with it :) |
| 01:57.15 | *** join/#asterisk dineshb (n=sekeksk@host-137-132-43-139.imcb.nus.edu.sg) |
| 01:57.42 | pressure_man | without register, i can see asterisk doing a "select * from sip where username='<callerid>'" |
| 01:58.06 | pressure_man | so it's using the callerid as the sip user |
| 01:58.26 | file[laptop] | which means it probably isn't matching on the user entry |
| 01:58.32 | dineshb | any call back experts here on asterisk? |
| 01:58.33 | pressure_man | no, it doesn't |
| 01:58.42 | pressure_man | which is why i get it to register, then it works. |
| 01:59.05 | file[laptop] | I'm just going to walk away now |
| 01:59.16 | dineshb | trying to put together something, but i want to be able to check the callerid agaist my cell phone, if its caller id = cell phone, hang up the call and dial my phone |
| 01:59.31 | file[laptop] | dineshb: dialplan logic and a call file |
| 01:59.44 | *** join/#asterisk |ynchmob (n=mag0o@adsl-066-156-092-028.sip.asm.bellsouth.net) |
| 01:59.52 | pressure_man | file, that's what i'm doing to asterisk... heading to openpbx |
| 02:00.18 | dineshb | file: i did it, but having some problem with the compare thing |
| 02:00.27 | file[laptop] | compare thing? |
| 02:00.28 | file[laptop] | it's not that hard |
| 02:00.34 | *** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 02:00.45 | dineshb | file: when i write the caller id to the call file |
| 02:00.45 | file[laptop] | exten => 8005551212/5068780147,1,Dowhatever |
| 02:00.53 | dineshb | tmp.call |
| 02:01.08 | file[laptop] | you could use a shell script or perl script to take whatever as an argument, and generate it |
| 02:02.11 | *** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net) |
| 02:02.46 | alephcom | I don't know what you mean by a callback expert but I do probably 200 a week. Not a big number but.... |
| 02:03.04 | dineshb | file: ok i will have a bash at it |
| 02:03.07 | dineshb | brb |
| 02:03.17 | *** part/#asterisk pressure_man (n=pressure@ip-202-37-228-1.internet.co.nz) |
| 02:04.23 | jake1932 | when did "make" become optional? |
| 02:04.29 | *** join/#asterisk schuylerdigium (n=Bosco@pcp03052091pcs.huntsv01.al.comcast.net) |
| 02:04.40 | MikeJ[Laptop] | ? |
| 02:04.51 | file[laptop] | MikeJ[Laptop]: 888 twat :P |
| 02:05.00 | jake1932 | used to be make clean; make; make install |
| 02:05.07 | jake1932 | now it's make clean; make install |
| 02:05.30 | jake1932 | according to here: http://www.asterisk.org/download |
| 02:06.52 | dineshb | alephcom: do u use callerid as ur phone? and then check it without accepting the call? |
| 02:07.02 | *** join/#asterisk Corydon76-home (i=gold@pdpc/supporter/sustaining/Corydon76-home) |
| 02:07.55 | *** join/#asterisk viLeR (i=1000@66.128.47.232) |
| 02:10.13 | *** join/#asterisk wunderkin (n=wunderki@12-219-165-109.client.mchsi.com) |
| 02:10.29 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@bzq-84-110-10-98.red.bezeqint.net) |
| 02:11.05 | Katty | mew. |
| 02:12.10 | jake1932 | or is make still required? |
| 02:12.34 | jake1932 | i'm getting this error now: chan_sip.c:555: error: syntax error before '<<' token |
| 02:12.37 | alephcom | What I do is designed for people to phone in and then they get a callback and get connected to a conferencing service. I do use the caller ids but I answer the phone because interaction is usually needed. That would be easy to cut out but.... |
| 02:12.49 | alephcom | The script is found at www.aleph-com.net/astpp |
| 02:13.00 | alephcom | There is a link on there somewhere I believe. I can find it yet |
| 02:13.30 | *** join/#asterisk N4SH (n=Bernardo@c-67-180-105-69.hsd1.ca.comcast.net) |
| 02:13.43 | PoWeRKiLL | hi |
| 02:13.51 | _DAW | hello |
| 02:13.52 | Katty | mew? |
| 02:13.59 | PoWeRKiLL | any ide why my zap channel answers before get a CONNECT on the PRI |
| 02:14.30 | Katty | :< |
| 02:14.30 | PoWeRKiLL | I got answered on the console as soon I get CALL PROCEEDING |
| 02:14.43 | N4SH | i have a question. i'm starting on my asterisk@home i installed CentOS 3.5 and i'm stuck from this point |
| 02:15.02 | Katty | silly humans. |
| 02:15.11 | _DAW | humans? |
| 02:15.41 | N4SH | how will i configure it? |
| 02:15.56 | jake1932 | sounds like a loaded question |
| 02:16.25 | jake1932 | N4SH: have you read any of the docs yet? |
| 02:17.05 | jake1932 | ~docs |
| 02:17.07 | jbot | from memory, docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
| 02:17.10 | N4SH | jake i'm reading the docs and tells me to access web gui |
| 02:17.16 | jake1932 | ah |
| 02:17.46 | jake1932 | for at home, you should be able to access AMP - a web portal |
| 02:17.58 | N4SH | how do i do that |
| 02:18.11 | jake1932 | when it starts up, i think it gives you an IP |
| 02:18.31 | N4SH | well it just gives me a local host |
| 02:19.20 | jake1932 | localhost from that machine, but does the machine has an IP reachable by a machine with a web browser? |
| 02:19.38 | N4SH | oh wait... let me see |
| 02:28.46 | N4SH | jake: i'm still having problems connecting to the machine |
| 02:29.03 | *** join/#asterisk Miggidy (i=user@dsl-202-72-180-171.wa.westnet.com.au) |
| 02:30.16 | *** join/#asterisk santiago (n=santiago@208.195.215.158) |
| 02:31.21 | *** join/#asterisk Beave (n=beave@vistech.org) |
| 02:32.06 | *** join/#asterisk greekman (n=alex@host254.209.113.199.conversent.net) |
| 02:33.01 | jake1932 | N4SH: do you have to use @home? are you familiar with linux at all? |
| 02:34.02 | jake1932 | reason being @home adds a lot of crap you may not need - and if you're new to asterisk - may be more confusing than just learning the standard configurations |
| 02:34.49 | N4SH | ok i'll try that instead |
| 02:34.56 | N4SH | tnx |
| 02:35.04 | jake1932 | try what? |
| 02:35.32 | *** join/#asterisk Sp3ciaL_K (n=alex@d141-139-99.home.cgocable.net) |
| 02:35.50 | Sp3ciaL_K | Hello |
| 02:36.31 | Sp3ciaL_K | has anyone got spandsp working with freebsd before? |
| 02:36.35 | jake1932 | N4SH - this is a lot closer to a standard config: http://www.xorcom.com/ |
| 02:39.01 | N4SH | oh ok. let me check |
| 02:40.42 | Sp3ciaL_K | well not sure if i need spandsp..i got a fax connect to a pap2 which then connects to my * box using SIP. when i try sending fax I notice a Unknown RTP codec 100, and fax failes. i can't start * with app_rx|txfax, gives me an error (undefine symbol lrint in libspandsp.so.0) |
| 02:43.20 | Sp3ciaL_K | so if anyone could shed some light that be cool. |
| 02:44.57 | Sp3ciaL_K | anyone went to Astricon? |
| 02:45.58 | *** join/#asterisk bugz (n=bugz@cpe-24-27-67-66.houston.res.rr.com) |
| 02:48.29 | pauldy | Sp3ciaL_K, did you check to see what codec 100 was |
| 02:48.36 | *** join/#asterisk paxr0 (i=jjhhjkh@216-155-91-143.bk2-dsl.surnet.cl) |
| 02:49.48 | *** join/#asterisk jdiskywlkr (n=kvirc@ip68-0-83-251.tu.ok.cox.net) |
| 02:50.53 | pauldy | maybe a combination of slin adpcm and ulaw |
| 02:50.53 | *** join/#asterisk brookshire[home] (n=mbrooks@esbrooks3.traveller.com) |
| 02:52.48 | Sp3ciaL_K | i tried google but couldnt see anywhere what codec 100 is |
| 02:53.45 | paxr0 | where i can find variable callerId? |
| 02:53.57 | wwalker | What does ATA stand for? I know what one is (802.3 -> POTS) but what's the acronym standfor? |
| 02:54.25 | crash3m__ | ATA == analog telphone adapter |
| 02:54.56 | wwalker | Thanks! |
| 02:56.18 | pauldy | Sp3ciaL_K, I think 100 is a mask but I'm not sure yet |
| 02:56.37 | *** join/#asterisk newmember (n=newmembe@S010600a0c93dce87.cg.shawcable.net) |
| 03:00.14 | pauldy | amazingly hard to find what that error really means |
| 03:00.28 | bugz | riing... |
| 03:00.30 | Sp3ciaL_K | i know |
| 03:00.32 | bugz | riiiiiing...... |
| 03:00.48 | bugz | all circuits are buys/congested. |
| 03:01.01 | brookshire[home] | buys? |
| 03:01.02 | brookshire[home] | :) |
| 03:01.02 | pauldy | I always was under the assumption it meant they could not agree on how to talk and the last number was the mask of the protocols that were useable but not accepted |
| 03:01.51 | Sp3ciaL_K | hmmm could be |
| 03:01.54 | n3u7 | anyone know how to compile asterisk without libnewt? |
| 03:02.28 | bugz | n3u7: USE=" -libnewt" emerge asterisk |
| 03:02.30 | bugz | ;D |
| 03:03.49 | Sp3ciaL_K | thanks anyways i'll stick to PSTN when faxing for now :P |
| 03:03.50 | pauldy | damb n3u7 I would have pulled my hair out by now and just installed libnewt and I consider myself a patient person |
| 03:04.47 | Sp3ciaL_K | blitzrage you therE? |
| 03:04.49 | pauldy | Sp3ciaL_K, I have only been able to get one fax to work perfectly voip but I used a hardware modem interfaces to an fxo port |
| 03:05.41 | Sp3ciaL_K | yea fax and voip dont mix too well yet |
| 03:07.23 | Sp3ciaL_K | okie i'll call it a night |
| 03:07.51 | Sp3ciaL_K | cya |
| 03:07.53 | *** part/#asterisk Sp3ciaL_K (n=alex@d141-139-99.home.cgocable.net) |
| 03:11.51 | nassy | im now learning about asterisk and it looks pretty interesting. hopefully one day it will replace our current PBX at work. a few questions about its capabilities: |
| 03:12.04 | nassy | 1, if the system is taxed can you (easily) expand it or integrate it with additional units (ie, if you need to add an additional T1, another asterisk box). we are always running out of something at work as we add more employees. |
| 03:12.25 | nassy | 2, is there the ability to set up call queues like when you call customer support and get placed in a line and it tells you there are 3 more people are ahead of you and you have 10 minutes. |
| 03:12.45 | nassy | 3, when the company takes over other companies can asterisk PBX's be integrated so that to customers it appears like if there are two companies but to the employees they can just dial extensions. employees for each company are not restricted to one location. |
| 03:13.02 | *** join/#asterisk Rowter (n=SilverDr@201.135.26.195) |
| 03:13.46 | brookshire[home] | 1 yes, sometimes |
| 03:13.47 | bugz | nassy: an asterisk ip-pbx on a p4 can power a medium sized city |
| 03:13.48 | brookshire[home] | 2 yes |
| 03:13.50 | brookshire[home] | 3 yes |
| 03:13.54 | bugz | scalability is NOT an issue |
| 03:14.19 | bugz | nassy: buying a DS-3 card.. ok you might have an issue with that but i hear they are in the making |
| 03:14.21 | nassy | sweet. now i just have to figure out how to replace the PBX after i get some asterisk experience |
| 03:14.32 | bugz | nassy: all you have to do is run alot of cat 5 |
| 03:14.42 | bugz | or buy some iaxys ;) |
| 03:15.07 | harryvv | is there a way to make a phone ring a light? |
| 03:15.08 | brookshire[home] | hey file |
| 03:15.13 | file[laptop] | hiiiiiii Matt |
| 03:15.16 | brookshire[home] | righ a light? |
| 03:15.21 | brookshire[home] | ring also |
| 03:15.23 | mtgh | haryvv: x10 |
| 03:15.32 | bugz | harryvv: hook it up to the fxo |
| 03:15.37 | bugz | pretty easy hack |
| 03:15.50 | brookshire[home] | oh.. like for the hearing impaired? |
| 03:16.02 | nassy | oh thats another thing we need for our noisy warehouse |
| 03:16.05 | *** join/#asterisk asterisk99 (n=dunoowhy@modemcable169.194-130-66.mc.videotron.ca) |
| 03:16.07 | bugz | or for titty bars :) |
| 03:16.13 | jake1932 | http://www.aaroncake.net/circuits/pflash.htm |
| 03:16.22 | *** join/#asterisk slak- (i=slak@us.fed-agent.us) |
| 03:16.29 | slak- | software phone that works under freebsd |
| 03:16.30 | slak- | hi |
| 03:16.34 | slak- | anyone know of one? |
| 03:16.41 | brookshire[home] | nassy: i'm sure you can extend agi do it :) |
| 03:17.07 | nassy | really looking forward to using asterisk |
| 03:17.16 | *** join/#asterisk syle2 (n=blah@unaffiliated/syle) |
| 03:17.32 | pauldy | someone should write a sip phone in flash that would be enat |
| 03:17.33 | asterisk99 | anyone know why audio from asterisk -> SIP phone doens't get sent? I can see playbacks() being played, but no audio |
| 03:17.42 | slak- | someone know of a softphone that works under freebsd |
| 03:17.46 | slak- | sip |
| 03:17.49 | slak- | or iaxc |
| 03:18.09 | brookshire[home] | asterisk99: firewall or codec problems |
| 03:18.26 | brookshire[home] | and tons of other possibilites it could be :) |
| 03:18.36 | pauldy | less common but I've had it be a resource issue not enough ram |
| 03:19.06 | brookshire[home] | slak-: you should just use linux.. it's better anyways ;) |
| 03:19.06 | slak- | ya |
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| 03:19.14 | nassy | are there any case studies of people replacing PBX's with asterisk. i would like to replace a toshiba strata system with asterisk but i dont have much time to learn it. we ran out of extensions and need to buy additional PBX equipment for the new employees. with asterisk id just run a cat 5 cable and im done. |
| 03:19.42 | znoG | nassy: well, that plus a channel bank, depending on your requirements |
| 03:19.53 | slak- | nassy i use it here |
| 03:19.56 | slak- | instead of a channelbank |
| 03:19.56 | brookshire[home] | nassy: you can check the http://www.digium.com website for some |
| 03:19.58 | slak- | we have ATA units |
| 03:20.01 | slak- | sipura |
| 03:20.05 | slak- | so far 20 extensions |
| 03:20.06 | nassy | znoG: oh, whats a channel bank. |
| 03:20.36 | brookshire[home] | znoG: you can use ipphones |
| 03:20.36 | znoG | i'm about to deploy 10 ATA's (to get 20 extensions on Asterisk). Far cheaper than a channel bank |
| 03:20.36 | znoG | brookshire[home]: yes, and they are expensive :) |
| 03:20.37 | slak- | znog thats what i dd |
| 03:20.37 | slak- | did |
| 03:20.55 | slak- | spa2000s |
| 03:20.55 | brookshire[home] | channel bank is for analog phones |
| 03:20.56 | znoG | slak-: thats them, or the PAP2-NAs, not sure which I'm gonna go for yet. |
| 03:21.00 | slak- | i use 2002 |
| 03:21.04 | slak- | and 2000 |
| 03:21.05 | slak- | sipura |
| 03:21.07 | slak- | works great man |
| 03:21.13 | znoG | slak-: i think the spa2000s can do remote provisioning, which might be handy for configuring them remotely instead of going in one-by-one |
| 03:21.14 | slak- | only one died so far its been a year |
| 03:21.15 | *** join/#asterisk denon (i=denon@synapse.subneural.net) [NETSPLIT VICTIM] |
| 03:21.15 | *** mode/#asterisk [+o denon] by ChanServ |
| 03:21.19 | slak- | i sent it back |
| 03:21.39 | slak- | i do it one by one |
| 03:21.45 | slak- | but if you know of a way i can config them all |
| 03:21.51 | slak- | please hook me up |
| 03:21.53 | znoG | i think remote provisioning might do it |
| 03:21.56 | znoG | look into it |
| 03:21.59 | znoG | google, etc |
| 03:22.15 | nassy | we use some toshiba ip phones. but the office phones have lcds on them and i cant replace them with regular phones and an ATA because the employees have grown to like some of the features on the 10 button phones. can you program the buttons on the phones that work with asterisk? |
| 03:22.30 | nassy | thanks. found the case studies. going to check them out |
| 03:22.31 | znoG | actually, to generate the configs for remote provisioning I think you need a special license to get the software to generate the config, from Sipura |
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| 03:22.50 | brookshire[home] | nassy: depends on the phones... |
| 03:23.03 | brookshire[home] | nassy: more than likely the toshiba phones will not work with asterisk |
| 03:23.15 | *** join/#asterisk Brijn (n=bas@S0106004063c0fa1f.vn.shawcable.net) |
| 03:23.23 | znoG | nassy: grow them out of liking those features, in favour of cost savings :) |
| 03:23.28 | znoG | nassy: (and more extensions) |
| 03:23.48 | brookshire[home] | with the polycom phones you can add extensions with xml.. |
| 03:23.54 | znoG | nassy: if you want a funky phone full of buttons and wicked features, you're going to spend a lot of money per phone |
| 03:23.57 | brookshire[home] | to the phones |
| 03:24.04 | Brijn | Does anybody here have an email address of the Asterisk at Home people (I don't see anything on the website).. Their current ISO's seem to be bad |
| 03:24.46 | nassy | based on our monthly bill i think we would make up the cost of buying new phones in a month |
| 03:24.46 | nassy | or two |
| 03:24.46 | znoG | maybe when the IP phones come down in price (ie. when they cost the same as one port on a sipura) then people will start to buy them |
| 03:25.09 | *** join/#asterisk morale (n=russell@secure.deadbolt.ca) |
| 03:25.09 | slak- | but |
| 03:25.10 | *** part/#asterisk litage (n=nick@203.220.55.70) |
| 03:25.10 | slak- | the sipura gives ya two ports |
| 03:25.10 | slak- | <PROTECTED> |
| 03:25.10 | znoG | but since you get 2 lines with a Sipura unit, most people get that instead of 2 Ip phones |
| 03:25.10 | slak- | we use the shittiest office phones |
| 03:25.11 | slak- | like 20$ panasonics |
| 03:25.38 | slak- | no one cares |
| 03:25.38 | slak- | and im happy |
| 03:25.38 | nassy | we spend about $200 i think for the toshiba ip phones |
| 03:25.38 | brookshire[home] | you just have to get use to transfering with the numeric pad |
| 03:25.55 | brookshire[home] | and remembering everyone else's extension |
| 03:25.55 | slak- | the vice-prez today dialed 911 by accident |
| 03:26.14 | slak- | and when they came and i checked the cdr who dialed it |
| 03:26.14 | slak- | and confronted him |
| 03:26.14 | slak- | he lied |
| 03:26.14 | slak- | ;/ |
| 03:26.14 | morale | heh |
| 03:26.29 | jake1932 | sure it was him and someone didn't use his phone? |
| 03:26.29 | slak- | im sure |
| 03:26.29 | slak- | noone uses his phone |
| 03:26.41 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 03:26.41 | slak- | he came out of his office lookin all stupid |
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| 03:27.06 | jake1932 | he's not signing your paycheck, right? |
| 03:27.17 | slak- | nah |
| 03:27.17 | slak- | he's just some fake |
| 03:27.19 | *** join/#asterisk _m_ (n=m@fbta199.fbta.uni-karlsruhe.de) |
| 03:27.28 | slak- | hired to work on the marketing side of things |
| 03:27.29 | jake1932 | yeah - cause that would suck |
| 03:27.30 | slak- | too bad he sucks |
| 03:28.01 | jake1932 | damn - asterisk is still compiling on my crappy p2 |
| 03:28.05 | slak- | ;) |
| 03:28.09 | jake1932 | been over an hour |
| 03:28.12 | slak- | jake need a soft phone for freebsd |
| 03:28.15 | slak- | anything come to mind? |
| 03:28.18 | jake1932 | don't have one |
| 03:28.32 | jake1932 | only know the win ones |
| 03:28.39 | jake1932 | and they all sucked |
| 03:28.40 | slak- | yes |
| 03:28.42 | slak- | diax is ok |
| 03:28.46 | jake1932 | oh |
| 03:28.50 | jake1932 | didn't try that one |
| 03:29.00 | slak- | diax is standalone its pretty nice |
| 03:29.04 | slak- | very lightweight |
| 03:29.10 | jake1932 | no choppiness? |
| 03:29.13 | slak- | nope |
| 03:29.32 | ManxPower | How does someone accidently dial 911/ |
| 03:29.39 | slak- | i dont know |
| 03:29.39 | jake1932 | ok - have to try that one day - been using hard phones and life is much easier |
| 03:30.14 | slak- | bottom right to top left diagonal |
| 03:30.20 | slak- | thats pretty hard to do unintentionally |
| 03:30.21 | slak- | ;/ |
| 03:30.25 | jake1932 | maybe the number was 201-456-0911 |
| 03:30.41 | jake1932 | and he accidentally hit the clicker |
| 03:31.09 | dineshb | how do i make the incomming calls from context from-pstn to activate a custom script instead of going to the local extenions |
| 03:31.13 | jake1932 | wonder if that would hold up in court |
| 03:31.25 | ManxPower | Trying to dial 9-1-212-555-1212 would require him to accidently dial 9 twice and 1 twice |
| 03:31.39 | slak- | i need something to display the cdr nice |
| 03:31.45 | slak- | web based |
| 03:31.47 | slak- | any ideas |
| 03:31.47 | jake1932 | dineshb: script, as in AGI? |
| 03:31.57 | nassy | i nearly dialed it my accident when i was trying to make an international call and didnt press the 0 properly. so i dialed 911 but because 9 is our outside access number it looked like 11 |
| 03:32.13 | slak- | i dont reequire 9 for 911 |
| 03:32.15 | slak- | just 911 works |
| 03:32.27 | ManxPower | slak-, we don't either |
| 03:32.29 | slak- | thanks for reminding me i haveto let people know |
| 03:32.34 | slak- | 411/911 and 611 |
| 03:32.35 | slak- | go right thru |
| 03:32.43 | dineshb | jake: using aah here, and it doesn't call the script |
| 03:32.47 | slak- | and drop a line if all busy |
| 03:32.57 | dineshb | it just goes to the normal pstn routines |
| 03:33.05 | timecop | <PROTECTED> |
| 03:33.08 | dineshb | i tried to include the context from-pstn-custom |
| 03:33.18 | Brijn | Does anybody here have an email address of the Asterisk at Home people (I don't see anything on the website).. Their current ISO's seem to be bad |
| 03:33.26 | ManxPower | dineshb, you really need to be on the A@H channel |
| 03:33.47 | dineshb | manxpower: which channel is that sorry? <-- lame question |
| 03:34.00 | jake1932 | seems to be a sudden influx of @home users today |
| 03:34.03 | marc324 | whats the recommended linux dist for asterisk? |
| 03:34.10 | marc324 | fedora/debian/slackware? |
| 03:34.15 | ManxPower | dineshb, I don't know, but not a lot of people use A@H here. |
| 03:34.16 | brookshire[home] | gentoo! |
| 03:34.18 | schuylerdigium | i like debian! |
| 03:34.18 | brookshire[home] | :D |
| 03:34.20 | schuylerdigium | no!!! |
| 03:34.20 | *** join/#asterisk wunderkin (n=wunderki@12-219-165-109.client.mchsi.com) |
| 03:34.24 | dineshb | marc: for starters use asterisk@home then move to debian when ur confident |
| 03:34.27 | brookshire[home] | debian or gentoo |
| 03:34.31 | brookshire[home] | or fedora |
| 03:34.31 | morale | whats asterisk@home? |
| 03:34.33 | brookshire[home] | or redhat ;) |
| 03:34.37 | jake1932 | @home will just confuse you |
| 03:34.37 | schuylerdigium | yes... |
| 03:34.41 | schuylerdigium | i was going to say redhat |
| 03:34.47 | schuylerdigium | i like CLI |
| 03:34.47 | dineshb | redhat is pricey |
| 03:34.48 | schuylerdigium | :) |
| 03:34.49 | marc324 | slackwar? |
| 03:34.51 | schuylerdigium | no |
| 03:34.52 | dineshb | talk about paying every year |
| 03:34.53 | brookshire[home] | or macosx |
| 03:34.53 | brookshire[home] | :D |
| 03:35.05 | jake1932 | i've been recommending rapid (xorcom.com) |
| 03:35.05 | ManxPower | morale, It's some silly distro with asterisk preinstalled and lots of custom scripts that try to make setting up a PBX a simple task. they fail. |
| 03:35.13 | morale | ahh. |
| 03:35.26 | jake1932 | rapid is debian based - and should get you started |
| 03:35.46 | brookshire[home] | man: it's good for learning.. but trying to do more advanced stuff breaks it |
| 03:35.48 | morale | is there a CVS HEAD changelog anywhere? |
| 03:36.05 | brookshire[home] | morale: i think they are working on one |
| 03:36.17 | morale | ah ok |
| 03:36.17 | brookshire[home] | it's something like 100 new changes |
| 03:36.23 | brookshire[home] | you can read the cvs logs though |
| 03:36.37 | morale | yeah |
| 03:36.46 | morale | i just rebuild the cvs head daily at 2:00am |
| 03:36.56 | brookshire[home] | hehe |
| 03:36.59 | brookshire[home] | on a cron? |
| 03:37.02 | jake1932 | i just tried to compile from CVS and it failed today |
| 03:37.06 | nassy | Brijn: i just burnt the asterisk at home iso. havent installed it yet because i dont havew a monitor. what is the problem you are having with it |
| 03:37.06 | morale | brookshire[home]: yeah |
| 03:37.17 | morale | jake1932: i just built it about 10 minutes ago and it worked |
| 03:37.19 | brookshire[home] | morale: cvs was broken for my yesterday :/ |
| 03:37.20 | jake1932 | CVS HEAD |
| 03:37.25 | brookshire[home] | at one point |
| 03:37.31 | brookshire[home] | it's fixed now |
| 03:37.33 | brookshire[home] | hehe :) |
| 03:37.45 | ManxPower | morale, it's call the asterisk-cvs mailing list |
| 03:37.48 | jake1932 | that explains it |
| 03:38.10 | jake1932 | i'm trying beta 1.2 now |
| 03:38.17 | *** join/#asterisk wunderkin (n=wunderki@VDSL-130-13-234-137.PHNX.QWEST.NET) |
| 03:38.20 | jake1932 | as soon as it finishes compiling |
| 03:38.42 | syle2 | got a problem, i finish up my code in h extension but its setting dst field to h instead of number called now |
| 03:38.53 | *** join/#asterisk drumkilla (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 03:38.53 | *** mode/#asterisk [+o drumkilla] by ChanServ |
| 03:39.02 | jake1932 | isn't h know to cause problems with the cdr |
| 03:39.07 | jake1932 | known |
| 03:39.53 | syle2 | idk |
| 03:40.13 | *** join/#asterisk file[laptop] (n=jcolp@mctnnbsa31w-142166095097.nb.aliant.net) |
| 03:40.39 | syle2 | is there a exit statement at all? |
| 03:40.52 | syle2 | i could try going back from h to original exten |
| 03:40.54 | syle2 | then exit |
| 03:41.01 | morale | is there anyway to make cvs login not prompt you for a password? |
| 03:41.45 | brookshire[home] | morale: cvs update |
| 03:41.48 | brookshire[home] | ?? |
| 03:41.49 | brookshire[home] | :D |
| 03:42.04 | morale | you still have to login first |
| 03:42.11 | brookshire[home] | intresting.. i never do |
| 03:43.55 | jake1932 | syle2: did you check out ResetCDR(w)? not sure if that's applicable... |
| 03:46.13 | syle2 | yep |
| 03:46.15 | syle2 | no docs on it |
| 03:46.24 | marc324 | in asterisk with db conf.... do you install the database on the asterisk server... or on another machine? |
| 03:46.45 | syle2 | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ResetCDR |
| 03:47.00 | brookshire[home] | marc324: i'm sure you can do either way |
| 03:47.20 | marc324 | performace factor... |
| 03:47.56 | syle2 | i think what i will do is save EXTEN into a variable |
| 03:48.21 | syle2 | then use a goto at end of h extension to go to a context with just nothing in it |
| 03:48.30 | *** join/#asterisk digime (n=drooth@ip68-111-235-172.sd.sd.cox.net) |
| 03:48.38 | syle2 | goto(mycontext,extvariable,1) |
| 03:48.47 | syle2 | about only hack i can think of |
| 03:48.53 | jake1932 | if it works... |
| 03:49.02 | syle2 | naw shit then i loose my context to |
| 03:49.04 | syle2 | fuck |
| 03:49.27 | syle2 | have to loop it in same context |
| 03:49.58 | syle2 | is there a exit command/ |
| 03:50.13 | syle2 | exten => blah, EXIT() |
| 03:50.16 | syle2 | or some crap |
| 03:50.44 | syle2 | actually i could just send it to end of the old context |
| 03:50.56 | syle2 | goto(mycontext,extvariable,15) |
| 03:50.59 | syle2 | i;ll try that |
| 03:51.10 | *** join/#asterisk bmg505 (n=leon@rndf-146-19-146.telkomadsl.co.za) |
| 03:55.06 | *** join/#asterisk hellagony (n=egutierr@irc.americatelnet.com.pe) |
| 03:57.50 | syle2 | it worked |
| 03:58.11 | jake1932 | the hacker prevails |
| 03:59.58 | *** join/#asterisk XTR (n=xtr@staff-nat.netnation.com) |
| 04:01.40 | syle2 | yep cdr is perfect |
| 04:01.49 | syle2 | fucking retarded i had to do that though |
| 04:06.34 | *** join/#asterisk aka-Debs (n=dbusch78@70.89.213.201) |
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| 04:15.51 | kb1_kanobe | evening all. |
| 04:16.15 | *** part/#asterisk schuylerdigium (n=Bosco@pcp03052091pcs.huntsv01.al.comcast.net) |
| 04:17.07 | *** join/#asterisk mmlj4 (n=looseduk@ip70-171-92-106.no.no.cox.net) |
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| 04:19.11 | *** join/#asterisk FuriousGeorge (n=brian@ool-44c5a9b8.dyn.optonline.net) |
| 04:19.12 | jayk- | i'm trying to set up a menu in asterisk. i want to let the caller press the extension of the person they want to transfer to after they press 1. for example, if i'm extension 100, they'd press 1 then 100. |
| 04:19.17 | jayk- | does anybody know how i could do this? |
| 04:24.54 | *** join/#asterisk Vco (n=Vco@S01060050da6df072.sc.shawcable.net) |
| 04:25.15 | *** join/#asterisk websae (n=websae@CPE-24-167-204-30.wi.res.rr.com) |
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| 04:28.11 | FuriousGeorge | <PROTECTED> |
| 04:28.50 | st3v | I want to put the asterisk box in a different room than the channel bank. Will it work if the channel bank and the T1 card are connected to the switch? |
| 04:29.35 | st3v | its probably a dumb question |
| 04:29.59 | kb1_kanobe | st3v: no, it won't work. The T1 isn't ethernet.. :-) |
| 04:30.41 | brookshire[home] | you can run a long t1 cable to the channel bank from the t1 card :) |
| 04:30.51 | kb1_kanobe | ... as long as it's under 1400 feet. |
| 04:31.01 | brookshire[home] | lol |
| 04:31.06 | brookshire[home] | it can be longer |
| 04:31.09 | brookshire[home] | if you have a repeater :) |
| 04:31.15 | kb1_kanobe | Ah yes, true... |
| 04:31.21 | jayk- | FuriousGeorge: is it ${exten}:1 or ${exten:1}? |
| 04:31.54 | brookshire[home] | and actually.. you only need 4 wires |
| 04:32.36 | st3v | can I use a cat 3 cable with RJ45 connectors |
| 04:33.11 | st3v | it will be less than 50 feet |
| 04:33.37 | kb1_kanobe | Yes. But stay away from flourescent light ballasts and stuff if possible 'cause they radiate electrical noise like crazy. |
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| 04:43.10 | *** part/#asterisk aka-Debs (n=dbusch78@70.89.213.201) |
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| 04:45.33 | brookshire[home] | kuku5: much better huh? |
| 04:45.34 | brookshire[home] | :) |
| 04:45.36 | kuku5 | how much cpu needed for ulaw <-> * <-> ulaw ? |
| 04:45.44 | kuku5 | yah |
| 04:45.53 | kuku5 | much better than 10 users on efnet |
| 04:46.35 | brookshire[home] | lol |
| 04:47.10 | drumkilla | not very much? |
| 04:47.20 | brookshire[home] | if you are only doing ulaw you don't need much at all |
| 04:47.26 | brookshire[home] | russell! |
| 04:48.28 | *** join/#asterisk spootnick (n=irc@50.118.233.220.exetel.com.au) |
| 04:48.57 | kuku5 | So according to my research. 15 calls/mbit can I get like 600 sim. callers on single cpu ? |
| 04:49.17 | spootnick | does anyone know a SIP softphone that supports video, apart from eyebeam and msn messenger? |
| 04:50.17 | timecop | yes |
| 04:50.17 | timecop | irc://bleach7@irc.irchighway.net/ |
| 04:50.21 | timecop | eh? |
| 04:50.49 | spootnick | was it for me? |
| 04:51.09 | kb1_kanobe | Anyone know of a CDR processor for Asterisk that can calculate observed channel utilisation in Erlangs? |
| 04:52.52 | timecop | no |
| 04:52.56 | timecop | that was in my cut buffer for some reason. |
| 04:53.52 | syle2 | i;m coding one for mysql kb1_kanobe |
| 04:54.01 | kb1_kanobe | ooo... how far off it is? |
| 04:54.03 | syle2 | that would be simple addin hehe |
| 04:54.07 | kb1_kanobe | , is it |
| 04:54.12 | syle2 | end of week |
| 04:54.25 | kb1_kanobe | will you announce on -dev? |
| 04:55.19 | syle2 | prob not |
| 04:55.31 | syle2 | those are more core asterisk dudes |
| 04:55.49 | *** join/#asterisk Eyecon (n=matt@jumala.plus.com) |
| 04:56.50 | kb1_kanobe | shame, I think there's interest. I know I'd appreciate being able to demonstrate my channel-sizing is working. |
| 04:56.52 | syle2 | idk if i will release it as i will be using it myself who knows |
| 04:57.03 | syle2 | its all in c though , no agi overhead |
| 04:58.28 | syle2 | i've spent countless hours on this already, if you want to bribe me that may work lol |
| 04:59.19 | Juggie | is there a sip client which supports messaging & pressence? |
| 04:59.25 | kb1_kanobe | Hmmm... The kb1 echo canceller owes me about half my hair, but let me think about it... :-) |
| 05:00.30 | syle2 | kb1? |
| 05:00.53 | syle2 | i lost alot of hair this weekend hehe |
| 05:00.58 | X-Rob | oooh |
| 05:01.14 | X-Rob | An echo cancellor that hardly sucks at all! |
| 05:01.37 | X-Rob | I had users complanging _every_ day. |
| 05:01.40 | X-Rob | now they don't complain at all |
| 05:01.44 | kb1_kanobe | it still doesn't work right, so no bowing please. |
| 05:01.52 | X-Rob | it works damn well. |
| 05:01.57 | syle2 | url? |
| 05:02.06 | X-Rob | syle2 - it's in CVS |
| 05:02.20 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 05:02.43 | kb1_kanobe | It's closer, but that was just code bugfixes. Now if someone could just explain the remaining problems - I'm no mathematician. |
| 05:03.04 | kb1_kanobe | you're welcome. :-) |
| 05:03.33 | X-Rob | kb1_kanobe - apparently, after speaking to a high-clue telco designed, doing 'good' echo can is extraordinarly hard, far harder than you (eg, me) expect. |
| 05:03.55 | kb1_kanobe | Yeah, I get that impression too. |
| 05:04.44 | kb1_kanobe | But I suspect the majority of issues being encountered are still bug-ish, not complex echos. Ie. the people who get unfettered echos on some calls. |
| 05:04.47 | *** join/#asterisk dasuberdavid (n=egg@pcp01534754pcs.huntsv01.al.comcast.net) |
| 05:06.02 | X-Rob | I had a look at the .h file, and it was so far above my head it wasn't funny 8-( |
| 05:06.48 | kb1_kanobe | Most of it shouldn't be an issue - the telco network is predictable. I suspect there is just an element missing, like a compander on the input to prevent the signal getting too hot, or some such. |
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| 05:07.10 | *** part/#asterisk tuppa (n=tuppa@eclipse.tuppa.org) |
| 05:12.58 | infinity1 | X-Rob: above or over your head |
| 05:12.59 | infinity1 | ? |
| 05:14.10 | *** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au) |
| 05:16.26 | *** join/#asterisk crash3m_ (i=crash3m@unaffiliated/crash3m) |
| 05:19.41 | syle2 | what kind of erlangs calculation? |
| 05:19.57 | kuku5 | syle2: what does the code do again ? |
| 05:20.03 | syle2 | one for each hour? |
| 05:20.31 | syle2 | 1pm 2.5 erlangs, 2pm 5 erlands |
| 05:20.34 | syle2 | kinda thing? |
| 05:21.10 | syle2 | well coding a billing application that will calculate ... |
| 05:21.17 | kb1_kanobe | I used to use Genesis (http://www.buygenesis.com/) many years ago. It was most useful because it had a grid report that had columns for the number of lines, rows for the time of day and gave a probablility of occurances of busies based on observed traffic. |
| 05:21.38 | syle2 | numbers based on definable rates you enter for each user |
| 05:21.38 | *** join/#asterisk spootnick (n=irc@50.118.233.220.exetel.com.au) |
| 05:21.48 | syle2 | user1 .02 |
| 05:21.55 | syle2 | user3 .006 |
| 05:21.57 | syle2 | etc |
| 05:22.11 | spootnick | hi all. i'm getting a "ocket_read: Rejected connect attempt from 204.13.233.114, who was trying to reach 's@'" message when i should be receiving calls from my iax account. any ideas on why that happens? |
| 05:22.13 | kb1_kanobe | Ie. over the last one month, based on observed traffic, you would have to have 16 trunks to have a 1 in 50 chance of an incoming call ringing busy, an so on. |
| 05:22.26 | syle2 | 2 sections, 1 for your providers you get DID etc from and one that deals with end users on your system |
| 05:23.05 | kb1_kanobe | Just trunk utilization for me - I use asterisk as PRI-IAX-PRI bridges, so I have lots of 23 channel PRIs for PBXes and lesser PRIs on the telco side. |
| 05:23.32 | *** join/#asterisk Jzalae (n=sk@216-220-248-35.midmaine.com) |
| 05:24.59 | kb1_kanobe | syle2: go to http://www.buygenesis.com/reports.htm#traf and download the 'all nortel/meridian reports' zip. Check out the 'Traffic Analysis' report. |
| 05:25.11 | syle2 | ok |
| 05:25.49 | kb1_kanobe | It's changed somewhat since I last used it, but you get the idea. Their example is based on trunks assigned to groups and then utilisation on each of those groups. |
| 05:27.14 | syle2 | are you a CLEC? |
| 05:27.24 | kb1_kanobe | Nope. Just an end user. |
| 05:27.47 | kb1_kanobe | (corporate end user) |
| 05:28.18 | syle2 | what reports in this zip file are you interested in seeing? |
| 05:28.55 | kb1_kanobe | just the 'Traffic Analysis', though if you were to implement others you'd be on the right track for other people. Genesis is very popular in the hotel switchboard market. |
| 05:29.32 | kb1_kanobe | It used to be (and probably still is) one of those 'recommended 3rd party' addons for Nortel systems. |
| 05:32.30 | pauldy | X-Rob, any news on grandstream fixing the handset echo they created when fixing the speaker phone echo |
| 05:32.58 | websae | what's a good GNU licensed program for billing with asterisk? |
| 05:32.59 | websae | anyone? |
| 05:33.10 | spootnick | quit |
| 05:33.22 | syle2 | none that i found |
| 05:33.25 | syle2 | why i am building my own |
| 05:34.05 | Juggie | is sip messaging supported in cvs head? |
| 05:34.25 | syle2 | i think i need a tutorial on erlangs |
| 05:34.48 | kb1_kanobe | http://www.erlang.com is reasonably good. |
| 05:35.00 | Dr_Ray | just use perl/ or sed awk |
| 05:35.10 | Dr_Ray | I built one in perl that works for us |
| 05:35.32 | syle2 | yeah i;d probably use perl for this run as a cronjob if i did it |
| 05:35.47 | Dr_Ray | ours runs as cgi webapp |
| 05:35.53 | Dr_Ray | it took all of 4 hours |
| 05:36.07 | Dr_Ray | I'm sure it is fugly |
| 05:36.15 | syle2 | no this is heavy processing of call logs |
| 05:36.19 | syle2 | cgi would timeout |
| 05:36.25 | syle2 | needs to be in c or perl backended |
| 05:37.24 | syle2 | its kind of like what webalizer does |
| 05:38.11 | kb1_kanobe | bbl, must go do play elsewhere. |
| 05:39.32 | blitzrage | woh, apparently I missed a light earthquake in California by only 3 hour |
| 05:39.34 | blitzrage | s |
| 05:41.27 | brookshire[home] | you can use the gsm codec :) |
| 05:41.33 | brookshire[home] | on your network! |
| 05:41.53 | syle2 | lol |
| 05:42.16 | syle2 | gsm ain;t that bad when i tried it |
| 05:42.35 | syle2 | but of course i;m talking about cell phone network |
| 05:43.21 | syle2 | hell i;d even settle for CLEC status |
| 05:43.45 | syle2 | would cost me about 1.4 million |
| 05:44.06 | kuku5 | syle2: what does the code do again ? |
| 05:44.10 | websae | good billing software--anyone? |
| 05:44.11 | kuku5 | ( ble ) |
| 05:45.26 | kuku5 | We have our crm - wondering if there is a method for a live cdr ( record for the start of the call, end of the call ) |
| 05:48.14 | syle2 | well whatever i want it to but right now its being programmed to be able to have definable rates for each of your users, so i can do say .03 for tollfree .02 for LD, .0whatever for incomming DIDs to a user(that part was a real bitch!), php backend it after for inserting into tables properly etc |
| 05:48.40 | syle2 | and code is taking care of updating the money left on account instantly after calls |
| 05:49.17 | syle2 | using nothing but custom asterisk modules and db triggers |
| 05:49.21 | syle2 | no agi bullshit |
| 05:49.59 | syle2 | every billing solution i looked at used agi calls |
| 05:51.21 | syle2 | why would you want to do that? |
| 05:51.48 | Igbothom_III | syle2; sounding quite nice :) |
| 05:52.20 | *** join/#asterisk pooh_ (n=hfwang@cust.15.241.adsl.cistron.nl) |
| 05:52.26 | Igbothom_III | g'day poo |
| 05:52.36 | pooh_ | Heya |
| 05:52.43 | Igbothom_III | (feck, forgot the <Tab> after that!!!) |
| 05:53.04 | syle2 | i don;t know why you would want a record for the start of a call |
| 05:53.13 | syle2 | give a reason why you would hehe |
| 05:53.49 | Igbothom_III | time/date stamp of the call? |
| 05:54.27 | syle2 | so you mean starttime , endtime fields for the cd record |
| 05:54.33 | kuku5 | yes |
| 05:54.46 | syle2 | hmm guess i could add that in |
| 05:54.46 | kuku5 | syle2: you have this up and running? |
| 05:55.39 | syle2 | still makes no practical sense you can just subtract date from cdr record from duration of the call |
| 05:55.45 | syle2 | or is this more a convenience |
| 05:56.10 | kuku5 | yeh - thats doable |
| 05:56.18 | kuku5 | syle2: you have this billing stuff done ? |
| 05:56.42 | syle2 | no i;m still running race conditions on it and more testing |
| 05:56.45 | syle2 | hoping friday |
| 05:56.47 | kuku5 | ok |
| 05:56.54 | kuku5 | you wanna sell it ? |
| 05:57.07 | syle2 | possibly |
| 05:57.10 | kuku5 | :) |
| 05:57.15 | kuku5 | how much |
| 05:57.21 | syle2 | no idea |
| 05:57.35 | syle2 | i;ll know when i;m finished |
| 05:57.43 | kuku5 | so friday ? |
| 05:58.21 | syle2 | next friday would be a more realistic release date |
| 05:58.26 | kuku5 | k |
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| 06:06.30 | *** join/#asterisk spootnick (n=irc@50.118.233.220.exetel.com.au) |
| 06:06.37 | *** join/#asterisk damned (n=vpol@prior.lanck.net) |
| 06:07.18 | spootnick | any goiax users out there? i'm getting a "rejected connect attempt" when receiving calls |
| 06:10.42 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:18.01 | *** join/#asterisk Delmar (n=Delmar@222-152-57-78.adsl.inspire.net.nz) |
| 06:18.48 | *** join/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1) |
| 06:18.50 | spootnick | :q |
| 06:32.35 | *** join/#asterisk darkskiez (n=darkskie@host86-138-169-183.range86-138.btcentralplus.com) |
| 06:33.49 | Igbothom_III | I'm reading in some places that ADSL is full duplex, yet others say that it is only half duplex. Anyone got any real data on this? |
| 06:37.36 | *** join/#asterisk niZon (n=ilt@S0106deadbeefbeef.wp.shawcable.net) |
| 06:41.02 | Igbothom_III | we have a client who can get an uncontended ADSL512/512 and an uncontended SHDSL 512/512 and the price difference is... noticeable |
| 06:42.29 | kuku5 | i have a problem with caller id - it shows uknown ... even though i set it |
| 06:48.34 | tessier | 512kb/s just seems so slow these days. |
| 06:48.47 | tessier | kuku5: What sort of line are you dialing out on? |
| 06:49.51 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-59-164.cybersurf.com) |
| 06:51.08 | *** part/#asterisk pooh_ (n=hfwang@cust.15.241.adsl.cistron.nl) |
| 06:54.56 | X-Rob | Igbothom_III - it's full duplex. |
| 06:56.54 | *** join/#asterisk ianrid (i=hidden-u@213-131-100-29.onyx.net) |
| 06:57.57 | websae | 512kb/s??? |
| 06:57.57 | websae | slow? |
| 06:58.07 | websae | can run a tun of g729 channels off that |
| 06:58.26 | websae | 64 to be exact |
| 06:58.31 | websae | *head now hurts |
| 06:58.53 | tessier | It's fine for running voip over, sure. |
| 06:59.02 | tessier | But it's slow as far as bandwidth goes these days. It's not even a T-1. |
| 06:59.19 | websae | T-1s are over rated |
| 06:59.30 | websae | can get residential cable packages faster than t1s |
| 06:59.32 | tessier | Actually...how do you figure you can run 64 g729 channels off of a 512kb/s connection? |
| 06:59.49 | tessier | Sure, and those residential cable packages are a whole lot faster than 512kb/s too. |
| 06:59.55 | tessier | Which is why I say 512 isn't much. |
| 06:59.57 | websae | haha |
| 07:00.01 | websae | what do you have? |
| 07:00.04 | kuku5 | tessier: iax |
| 07:00.09 | websae | for your broadband? |
| 07:00.23 | tessier | kuku5: Your provider might not be allowing you to set the CID then. |
| 07:00.32 | kuku5 | he does |
| 07:00.40 | kuku5 | it just works on some and doenst work on others |
| 07:00.42 | tessier | websae: I have cablemodem. 512kb/s up and 10Mb/s (they say) down. |
| 07:00.54 | websae | what can you download at? |
| 07:00.55 | tessier | kuku5: You mean on some calls it works and on others it does not and you don't change anything? |
| 07:00.58 | *** join/#asterisk Nukemizer (n=Nuke@160.7.249.15) |
| 07:01.05 | websae | what have you maxed your connection out at? |
| 07:01.20 | kuku5 | tessier: on some phones |
| 07:01.34 | kuku5 | ( recieving phones.... people out in the world :) ) |
| 07:01.55 | tessier | websae: I know I've gotten at least 160kb/s before (T-1 speed) but I've never downloaded from a nearby site to see how fast the cablemodem really is. |
| 07:02.14 | tessier | websae: You can only get 17 g729 calls over a 512kb line. |
| 07:02.25 | websae | i think i capped out at 600kb/s at debian |
| 07:02.33 | websae | on my roadrunner $29.99/month package |
| 07:02.43 | kuku5 | websae : whats your upload? |
| 07:02.52 | websae | that i haven't tried just yet--- |
| 07:02.57 | kuku5 | exactly |
| 07:03.05 | websae | got any good suggestions for trying that? |
| 07:03.07 | kuku5 | t1 guarantees your speed |
| 07:03.14 | kuku5 | yeh - upload via ftp |
| 07:03.20 | websae | so does speak easy dsl |
| 07:03.31 | websae | speak easy dsl has GREAT performance :) |
| 07:03.39 | websae | especially for asterisk |
| 07:04.29 | tessier | I tried to get speakeasy (had it once, about 5 years ago) but DSL is not available in my area. :( |
| 07:06.05 | websae | get a fractional t1 :) |
| 07:06.21 | tessier | Hate to think how much that might cost. |
| 07:06.32 | websae | $299/month |
| 07:06.38 | tessier | ouch |
| 07:06.44 | websae | hehe--yea |
| 07:06.49 | websae | what do you pay right now? |
| 07:06.56 | kuku5 | t1 = 300 |
| 07:07.14 | tessier | I have business cablemodem (faster upload, no blocked ports, two static IP's) for around $80/mo |
| 07:07.44 | websae | t1 = more than 300 |
| 07:07.55 | websae | that's not bad your biz cable |
| 07:07.57 | websae | whose your provider? |
| 07:08.05 | tessier | Cox |
| 07:08.08 | tessier | aka Cocksuckers |
| 07:08.20 | tessier | Actually, their business service isn't bad at all |
| 07:09.08 | websae | ha |
| 07:09.13 | websae | doesn't sound that bad |
| 07:09.23 | websae | time warner road runner---expensive for biz class |
| 07:09.33 | websae | biz class = full t1 |
| 07:09.35 | websae | pricing |
| 07:10.26 | kuku5 | tessier: stil prpblems withthe callerid |
| 07:11.38 | *** join/#asterisk Baph (n=Dave@dirobertson.plus.com) |
| 07:11.48 | *** join/#asterisk Poincare (n=jefffnod@dD577A88C.access.telenet.be) |
| 07:16.30 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-15-5.cybersurf.com) |
| 07:25.06 | *** join/#asterisk folsson (n=filip@h147n1fls32o985.telia.com) |
| 07:54.10 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 07:54.44 | puzzled | morning |
| 07:57.15 | PoWeRKiLL | any TE4XX owner ? |
| 07:58.16 | kb1_kanobe | syle2: start time, end time and duration in CDR reflect the fact that a call may be up, but not answered, for a period. |
| 08:02.15 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 08:08.04 | *** join/#asterisk jac]Z[oby (n=me@193.83.248.26) |
| 08:08.06 | jac]Z[oby | ahhoi |
| 08:08.18 | jac]Z[oby | can anyone help me with incoming number handling? |
| 08:09.04 | jac]Z[oby | anyone here? |
| 08:09.24 | PoWeRKiLL | jac]Z[oby ask you question |
| 08:09.49 | jac]Z[oby | i have an asterisk@home (sorry) with an isdn card |
| 08:10.04 | jac]Z[oby | and now id like to call an extension directly |
| 08:10.11 | jac]Z[oby | how does that work |
| 08:10.39 | jac]Z[oby | im alway coming into the waiting queue |
| 08:11.14 | jac]Z[oby | how do i check which number (incoming) has been dialled |
| 08:11.35 | PoWeRKiLL | I don't know about *@home but you should edit your extension.conf and put something like exten => yourincominnumber,1,Dial(SIP/yourdevice) |
| 08:11.40 | jac]Z[oby | because i dont know wether the phone provider is sending numbers after the actual phone number |
| 08:12.23 | jac]Z[oby | do i have to do it for every number? |
| 08:12.30 | jac]Z[oby | always? |
| 08:13.31 | PoWeRKiLL | no you can make a s exten so everything will come in or make something like exten => 718XXXXX.,1,Dial(SIP/toto) |
| 08:13.35 | jac]Z[oby | there is no such mechanism that directs it automatically to the extension number if its dialled afterwards |
| 08:14.03 | jac]Z[oby | that means that everyone is directed to one certain number |
| 08:14.38 | jac]Z[oby | btw |
| 08:14.52 | jac]Z[oby | when it comes to the incomingnumber |
| 08:15.02 | jac]Z[oby | which parts do i have to put in |
| 08:15.15 | jac]Z[oby | is that number only the number after the actual number |
| 08:15.31 | jac]Z[oby | or du o have to insert that plus the usual stuff |
| 08:15.44 | jac]Z[oby | 00country-zip-phone- extension... |
| 08:17.41 | jac]Z[oby | is there a poss. to see which number was dialled? |
| 08:17.47 | jac]Z[oby | that would be of great service |
| 08:19.21 | PoWeRKiLL | I don't know how your carrier send it |
| 08:19.43 | PoWeRKiLL | just go to asterisk console with verbosity asterisk -rvvvv make a call and see |
| 08:20.08 | jac]Z[oby | nope nothing in there |
| 08:20.25 | gaggaman | hi all! |
| 08:20.28 | gaggaman | could maybe somebody help me with my bristuffed asterisk 1.09 and call Pickup? |
| 08:20.38 | gaggaman | when I do a Pickup, i get: |
| 08:20.39 | jac]Z[oby | rofl |
| 08:20.41 | jac]Z[oby | same here |
| 08:20.45 | gaggaman | <PROTECTED> |
| 08:20.53 | gaggaman | <PROTECTED> |
| 08:20.58 | gaggaman | <PROTECTED> |
| 08:21.31 | gaggaman | so the line is picked up |
| 08:21.42 | gaggaman | but then something strange happens. |
| 08:23.42 | Igbothom_III | X-Rob; thanks (ADSL = Full Duplex) |
| 08:23.48 | Igbothom_III | sorry, long phone call |
| 08:27.53 | *** join/#asterisk Pazzo (n=thomas@host130-250.pool8172.interbusiness.it) |
| 08:36.01 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 08:40.51 | *** join/#asterisk linx (n=linx@debian.geek.nz) |
| 08:41.41 | *** part/#asterisk jac]Z[oby (n=me@193.83.248.26) |
| 08:41.42 | linx | when I call voicemail from a cisco ip phone (12SP+) it asks for the mailbox then the password |
| 08:42.14 | linx | when doing the same thing from an analog phone connected to fxs it only asks for the password |
| 08:42.29 | *** part/#asterisk lars-- (n=lars@lars.debian.us) |
| 08:48.40 | *** join/#asterisk sambal (n=sambal@213.148.236.189) |
| 08:51.56 | *** join/#asterisk iDunno (n=brettp@stef.sommitrealweird.co.uk) |
| 09:01.49 | *** join/#asterisk oej (n=Olle@apollo.webway.se) |
| 09:05.38 | *** join/#asterisk K-Bear (n=k-bear@h29n2fls32o815.telia.com) |
| 09:05.56 | K-Bear | Hello! |
| 09:06.40 | iDunno | morning |
| 09:07.23 | K-Bear | Sorry if I'm upsetting anyone by posting my question again, but since it was at least 10hrs ago and nobody had an answer then, maybe there are other ppl in here now who might |
| 09:07.27 | K-Bear | I'm having a pretty odd problem. I'm setting up an IAXy. I've manage to provision it and I've set up my Asterisk server to have it accepted. It's working to the point that I get a dial tone, I can dial all the extensions on the PBX. But when a call is connected, even when it's local in my own network, I don't get any sound from the device connected to the IAXy. I can hear the person at the extension I dialed, but they can't hear the IAXy. And |
| 09:07.28 | K-Bear | after 15s Asterisk hangs up. Anyone know what's wrong? |
| 09:07.53 | Dr_Ray | no nat? |
| 09:08.55 | K-Bear | Well, I have NAT. But since both my asterisk server, the extension I'm dialing and the IAXy is inside it, I figure that's not the problem |
| 09:09.26 | *** join/#asterisk CleanerX (n=nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 09:09.34 | K-Bear | They are all on the same subnet |
| 09:10.23 | kb1_kanobe | Presumably the IAXy doesn't see the call progressing beyond the dialing state, so eventually it hangs up. |
| 09:10.29 | kb1_kanobe | CVS or 1.0? |
| 09:10.37 | K-Bear | Old CVS |
| 09:11.21 | kb1_kanobe | There was a long standing issue with PRI and IAX not passing progress information properly which had similar symtoms, but I doubt it'd affect and IAXy |
| 09:11.45 | kb1_kanobe | this isn't dialing out of a PRI is it? |
| 09:12.11 | K-Bear | PRI? |
| 09:12.14 | kb1_kanobe | ISDN? |
| 09:12.18 | K-Bear | Ah, no |
| 09:12.50 | kb1_kanobe | Run the console with debugging and verbosity turned up and see what it has to say just as the call is dropped. |
| 09:12.54 | K-Bear | The other extension is a software IAX client |
| 09:13.01 | K-Bear | Ok |
| 09:15.02 | *** join/#asterisk fenlander (n=neils@82.152.81.57) |
| 09:15.51 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 09:20.02 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 09:20.04 | *** join/#asterisk MuppetMaster (n=MuppetMa@169.red-81-184-73.user.auna.net) |
| 09:20.47 | MuppetMaster | Hello |
| 09:20.50 | MuppetMaster | Anyone out there? |
| 09:21.26 | iDunno | nope. |
| 09:21.42 | MuppetMaster | Cool. |
| 09:22.01 | *** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62) |
| 09:22.11 | MuppetMaster | I have Asterisk v1.2beta and have a context that Answers, Waits (1) then does a Playback. But every message that gets played has the first 1 second or so cut. |
| 09:22.38 | MuppetMaster | Even if I move the Wait to (2), the same thing happens. Is there a way to solve this? Or do I just need to pad all first sound files with 1-2 seconds of silence? |
| 09:23.21 | K-Bear | kb1_kanobe: I can't find anything out of the ordinary. I'm trying to find a code snippet site to paste the output in... |
| 09:23.34 | kb1_kanobe | pastebin.ca |
| 09:24.20 | K-Bear | Ah, cheers |
| 09:25.25 | nfi|ermes | when i dial out from an internal, everything works fine but i can't listen the phone ring, (the tone, the noise to understand if it s free or busy) |
| 09:26.26 | K-Bear | kb1_kanobe: http://pastebin.ca/25940 |
| 09:27.16 | *** part/#asterisk MuppetMaster (n=MuppetMa@169.red-81-184-73.user.auna.net) |
| 09:28.36 | kb1_kanobe | K-Bear: that might be a little too deep. :-) |
| 09:28.44 | K-Bear | kb1_kanobe: :-) |
| 09:28.45 | kb1_kanobe | try 'set debug 255' and 'set verbose 255' |
| 09:30.02 | kb1_kanobe | however, I observe: IPV4 192.168.0.19:4569 vs. IPV4 192.168.2.2:4569. Was there only one call up during that log? |
| 09:30.02 | K-Bear | Only 'set verbose' is available, not 'set debug' |
| 09:30.30 | kb1_kanobe | Odd. *@home? |
| 09:30.45 | K-Bear | No, as I said. It's an old CVS version |
| 09:30.54 | kb1_kanobe | huh - must be very old. |
| 09:31.00 | K-Bear | And yes, only one call was made during that debug |
| 09:31.23 | kb1_kanobe | try 'show version' in the console |
| 09:32.02 | kb1_kanobe | Are both those IP addresses correct? |
| 09:32.07 | kb1_kanobe | (for the two endpoints)? |
| 09:33.00 | K-Bear | The version only shows that it's built by me on an i686, no version number |
| 09:33.12 | K-Bear | Yes, both IP's are correct |
| 09:33.30 | kb1_kanobe | Hmmm.... I would suggest updating to latest cvs as a first step. Sorry. |
| 09:33.42 | K-Bear | Ok |
| 09:34.12 | K-Bear | Well, thank you for helping me |
| 09:34.24 | kb1_kanobe | sorry it wasn't better. :-/ |
| 09:35.32 | K-Bear | No problem. I knew I was bound to upgrade sometime. Just hope it wasn't going to be anytime soon |
| 09:35.41 | K-Bear | *hoped |
| 09:36.33 | K-Bear | The thought of reconfiguring everything isn't so pleasant :-S |
| 09:36.58 | kb1_kanobe | you should be alright, just beware of the depreciation of the n+101 jumping behaviour |
| 09:38.31 | *** join/#asterisk Attila_Kovacs (n=kovacsat@dsl51B79178.pool.t-online.hu) |
| 09:38.33 | K-Bear | :-D |
| 09:39.46 | Attila_Kovacs | Hi All! Anybody can help me with chan_misdn? |
| 09:39.57 | *** join/#asterisk szer (n=Miranda@217.116.36.22) |
| 09:41.13 | szer | hi all |
| 09:43.30 | nfi|ermes | hi |
| 09:43.58 | nfi|ermes | when i dial out from an internal, everything works fine but i can't listen the phone tone (the noise to understand if it s free or busy) |
| 09:44.13 | *** join/#asterisk ard (n=ard@2001:7b8:32d:0:20c:6eff:fe18:d11f) |
| 09:44.48 | kb1_kanobe | nfi|ermes: it's called 'in-band signalling'. Ie. the busy sound, ringing sound and so on. |
| 09:44.57 | kb1_kanobe | What are you dialing from and what are you dialing to? |
| 09:47.22 | *** join/#asterisk kippi (n=chrisfro@untrust-gct.equinoxit.net) |
| 09:47.42 | *** join/#asterisk {zombie} (i=zombie@soulasylum.penguincare.com.au) |
| 09:48.04 | K-Bear | kb1_kanobe: H, sorry to bother you again. I came up on this ven I had upped the verbosity: "-- Executing Dial("IAX2/iaxy@iaxy/10", "IAX2/bjr|30") in new stack". ""IAX2/iaxy@iaxy/10" looks a bit wierd, don't you think? |
| 09:48.08 | K-Bear | *Hi |
| 09:49.14 | kb1_kanobe | user 'iaxy' on device 'iaxy' originated call using IAX2 protocol is I think what it's saying. |
| 09:49.59 | *** join/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8) |
| 09:50.01 | K-Bear | Ah, ok. So the device is in there too. I get it. Sorry to bother you |
| 09:50.10 | kb1_kanobe | no worries. :-) |
| 09:50.34 | K-Bear | First time I'm using any hardware. Only used software phones before... |
| 09:51.12 | Attila_Kovacs | No voice at all when there is incomming call on misdn channel. Outgoing calls are ok. |
| 09:52.23 | K-Bear | Sorry, no idea |
| 09:59.43 | *** join/#asterisk spiekey (n=spiekey@p549D18C1.dip0.t-ipconnect.de) |
| 09:59.45 | spiekey | hello |
| 10:00.06 | PoWeRKiLL | anyone using digium E1 card ? |
| 10:00.43 | spiekey | i am running sarge with a 2.6.8 kernel, and when i try to run "make linux26" in zaptel i get this error: You do not appear to have the sources for the 2.6.8-2-386 kernel installed. |
| 10:00.54 | spiekey | i have linked linux -> /usr/src/kernel-source-2.6.8 |
| 10:06.37 | spiekey | anyone? |
| 10:09.04 | kb1_kanobe | I'm running 2.6.x and I've always just run 'make' in zaptel. |
| 10:09.43 | spiekey | hmm...weird |
| 10:09.47 | gaggaman | spie maybe he is looking for /usr/src/linux-2.6.8-2-386 |
| 10:10.15 | gaggaman | try and make a link |
| 10:10.38 | spiekey | ln -s kernel-source-2.6.8 /usr/src/linux-2.6.8-2-386 - |
| 10:10.40 | spiekey | nope ;/ |
| 10:11.00 | gaggaman | ok :-) was just an idea :-) |
| 10:11.32 | gaggaman | then maybe just look into the Makefile |
| 10:12.16 | spiekey | <PROTECTED> |
| 10:12.42 | CBTCWwW | hmm interesting. I just saw a CD for sale on ebay.. the CD is asterisk software |
| 10:12.44 | spiekey | i dont have the "build" dir |
| 10:13.12 | iDunno | apt-get install kernel-headers-2.6.8-2-386 |
| 10:13.21 | iDunno | then try again. |
| 10:13.33 | iDunno | (IIRC, BICBW) |
| 10:14.34 | gaggaman | i have a problem with a bristuffed * 1.09 and call pickup |
| 10:14.38 | gaggaman | anybody? |
| 10:15.48 | CBTCWwW | not me, i'm here looking for a good manual for asterisk.. i'm about to dive into using it |
| 10:16.23 | spiekey | iDunno: thanks! :) |
| 10:16.26 | *** join/#asterisk [Airwolf] (n=airwolf@airwolf.xs4all.nl) |
| 10:16.42 | spiekey | i keep thinking that all i need is in the kernel source package :P |
| 10:17.37 | *** join/#asterisk jac]Z[oby (n=me@193.83.248.26) |
| 10:17.42 | jac]Z[oby | ahoi |
| 10:17.44 | oej | A good manual is to be found at http://www.asteriskdocs.org - the full O'reilly book |
| 10:17.54 | jac]Z[oby | anyone knows where i can find info about ENUM and asterisk? |
| 10:18.06 | jac]Z[oby | meaning using enum with asterisk |
| 10:18.35 | spiekey | oej: did you mean me? |
| 10:18.37 | oej | There is a README in CVS head |
| 10:18.42 | iDunno | spiekey: np :) |
| 10:18.53 | oej | for enum I mean |
| 10:19.04 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 10:19.07 | jac]Z[oby | what where? |
| 10:19.10 | jac]Z[oby | which cvs head? |
| 10:19.23 | oej | Asterisk CVS head of course... |
| 10:19.28 | spiekey | CBTCWwW: http://www.jaredsmith.net/AsteriskTFOT.zip |
| 10:19.37 | spiekey | its really good and quite simple to start with |
| 10:19.52 | gaggaman | What does <MASQ> in -- Hungup 'SIP/61-8140<MASQ>' mean? |
| 10:26.54 | kippi | has anyone had this error before Unable to connect to Asterisk Manager (111) when using AMP? |
| 10:30.24 | spiekey | how long will asterisk need to compile on a 233MHz mmx? |
| 10:31.01 | spiekey | 2h? |
| 10:31.08 | oej | gaggaman: It means it's a masqueraded channel - someone else took over, possibly in a transfer |
| 10:40.23 | MatsK | spiekey: it could take around 2 hour, on my C400 it took 1 hour and 35 minutes |
| 10:41.30 | MatsK | Any norwegians here ? |
| 10:41.48 | MatsK | Chack out www.asterisk.no |
| 10:42.03 | gaggaman | oej: strange. |
| 10:42.03 | MatsK | Chack = Check |
| 10:42.13 | spiekey | MatsK: thanks |
| 10:42.31 | gaggaman | I get: |
| 10:42.32 | gaggaman | <PROTECTED> |
| 10:42.32 | gaggaman | <PROTECTED> |
| 10:42.32 | gaggaman | <PROTECTED> |
| 10:42.42 | gaggaman | Thant |
| 10:42.48 | gaggaman | That's all. |
| 10:43.33 | gaggaman | SIP/61 was the extension who tried to Pickup the call. |
| 10:44.04 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 10:44.19 | gaggaman | so who took over? |
| 10:45.20 | gaggaman | Afterwards I see the channel I picked up as "ZOMBIE": |
| 10:45.22 | gaggaman | <PROTECTED> |
| 10:59.52 | macTijn | http://www.ngnsky.com/index.php |
| 11:06.57 | spiekey | i build asterisk from source and i installed capi with apt. now i get: /usr/lib/asterisk/modules/chan_capi.so: undefined symbol: ast_smoother_feed |
| 11:07.42 | *** join/#asterisk KriS83 (n=KriS@212.202.141.92) |
| 11:08.20 | KriS83 | Hi |
| 11:09.00 | *** join/#asterisk FITA1 (n=m_ahmed@202.5.145.50) |
| 11:09.05 | FITA1 | hi all |
| 11:09.38 | spiekey | hi |
| 11:09.40 | FITA1 | while compiling asterisk-1.0.9 i m getting this error /usr/local/bison/bin/bison ast_expr.y --name-prefix=ast_yy -o ast_expr.c |
| 11:09.40 | FITA1 | make: *** [ast_expr.c] Broken pipe |
| 11:09.55 | FITA1 | any help please ???? |
| 11:10.42 | *** join/#asterisk zotz (n=zotz@24.231.36.100) |
| 11:12.06 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 11:13.51 | KriS83 | When compiling the recent Beta-1 Version of chan-capi do I have to apply any patches so I can use something like: ;exten => 122,2,capiHOLD |
| 11:13.51 | KriS83 | ;exten => 122,3,capiECT,27:17 |
| 11:13.51 | KriS83 | <PROTECTED> |
| 11:14.09 | KriS83 | I have removed the ; infront obisously |
| 11:14.42 | *** join/#asterisk JimmyCarter (n=del@213083175015.sonofon.dk) |
| 11:14.56 | JimmyCarter | I have a question about channels |
| 11:15.05 | JimmyCarter | are 2 channels not in anyway associated while one is calling the other? |
| 11:15.21 | JimmyCarter | i.e. one channel is 'ring' the other is 'ringing'. |
| 11:16.00 | kb1_kanobe | FITA1: is bison installed, up to date and working properly? |
| 11:16.21 | FITA1 | yes bison is properly installed |
| 11:16.25 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net) |
| 11:17.37 | kb1_kanobe | fwiw: I'm using GNU Bison v1.875a on my system and all is well. What happens if you try to execute that command by hand in the relevant directory. |
| 11:18.07 | jac]Z[oby | hmm question |
| 11:18.35 | jac]Z[oby | do i have to do anything more than setting up an enum trunk to check wether an enum lookup is working? |
| 11:18.42 | jac]Z[oby | enum pros welcome |
| 11:18.47 | jac]Z[oby | pleez query me |
| 11:18.51 | nfi|ermes | is it possible to let the internal pc speaker ring when an incoming call arrives in xlite ? |
| 11:19.21 | jac]Z[oby | sure |
| 11:19.21 | jac]Z[oby | if you have 2 audio devices defned |
| 11:19.32 | *** join/#asterisk MatsK (n=mk@99.80-202-83.nextgentel.com) |
| 11:19.35 | jac]Z[oby | but i dont think that thats what you want |
| 11:19.56 | jac]Z[oby | buy a usb headset and everyxthing works fine |
| 11:20.10 | nfi|ermes | i have headset |
| 11:20.20 | jac]Z[oby | usb? |
| 11:20.29 | nfi|ermes | no |
| 11:20.31 | jac]Z[oby | see |
| 11:20.45 | jac]Z[oby | if u have a normal headset the internal ring gos through the headset |
| 11:20.50 | nfi|ermes | i don t want the phone ring in headset |
| 11:20.57 | jac]Z[oby | which means its all the same audio device |
| 11:21.02 | jac]Z[oby | i know |
| 11:21.12 | jac]Z[oby | like in skype vie the internal beeper |
| 11:21.17 | nfi|ermes | if i m far from my pc , i can t lsten the phone |
| 11:21.20 | jac]Z[oby | i dont think that works |
| 11:21.26 | KriS83 | btw I get the error: Oct 19 11:00:44 NOTICE[17148]: app_dial.c:1091 dial_exec_full: Unable to create channel of type 'CAPI' (cause 44 - Requested channel not available). Any help is appreciated |
| 11:22.01 | jac]Z[oby | but then you cant listen the internal beep eather |
| 11:22.08 | jac]Z[oby | that sux anyway?!?! |
| 11:22.16 | nfi|ermes | :| |
| 11:22.17 | spiekey | jac]Z[oby: suxx |
| 11:22.24 | nfi|ermes | other solution ? |
| 11:22.24 | JimmyCarter | anyone familar with asterisk-java ?? |
| 11:22.34 | jac]Z[oby | like boah ey?!?! |
| 11:22.37 | Ahrimanes | hardwire: awake? |
| 11:22.43 | jac]Z[oby | useles information @ spiekey |
| 11:22.53 | spiekey | jac]Z[oby: sorry |
| 11:22.59 | jac]Z[oby | enum support still welcome ;)) |
| 11:23.26 | jac]Z[oby | @ nfi|ermes like i said |
| 11:23.32 | jac]Z[oby | second sound card |
| 11:23.35 | jac]Z[oby | some old one |
| 11:23.36 | FITA1 | <kb1_kanobe>: I have also checked using "which bison" commmand and it is in the path |
| 11:23.48 | jac]Z[oby | than you can split the ringing and the audi signals |
| 11:24.03 | jac]Z[oby | damn i cant write anymore |
| 11:27.22 | *** join/#asterisk ful|work (n=fulgas@213.58.130.46) |
| 11:27.27 | ful|work | hey |
| 11:29.53 | *** join/#asterisk kippi (n=chrisfro@untrust-gct.equinoxit.net) |
| 11:30.12 | ful|work | got some issue when passing exten from oh323 to zap |
| 11:30.14 | ful|work | http://pastebin.ca/25947 |
| 11:30.18 | ful|work | it keeps crashing |
| 11:30.57 | *** join/#asterisk bweschke (n=bweschke@host-69-95-11-131.roc.choiceone.net) |
| 11:33.20 | *** join/#asterisk Abbas (n=Abbas@203.81.217.214) |
| 11:34.02 | Abbas | chan_sip.c:8059 sip_poke_noanswer: Peer '1002' is now UNREACHABLE! |
| 11:34.16 | Abbas | why this messag comes |
| 11:34.27 | JimmyCarter | Is there any way to make channels linked before the phone is picked up? |
| 11:34.36 | JimmyCarter | Abbas: try reload |
| 11:34.45 | Abbas | i have tried |
| 11:38.49 | jac]Z[oby | waaa damn you enum |
| 11:38.54 | jac]Z[oby | found tha error |
| 11:41.52 | KriS83 | I have a really suptid question I guess... where do I get app_capiHOLD.so and app_capiECT.so from? Which source provides them? Because the chan_capi-cm from Sourceforge doesn't. |
| 11:42.06 | KriS83 | stupid* |
| 11:46.17 | trym | It seems the expiry for outbound registration is too high.. because asterisk stops getting SIP requests when I call my sip number. If I call just after a registration it usually works |
| 11:48.09 | kippi | i am getting this error when i try to start asterisk |
| 11:48.10 | kippi | <PROTECTED> |
| 11:48.10 | kippi | Asterisk Dynamic Loader Starting: |
| 11:48.10 | kippi | <PROTECTED> |
| 11:48.42 | kippi | anyideas what this is? |
| 11:49.12 | psk | kippi: where's the error? |
| 11:50.21 | kippi | but if i do asterisk -r I can't connect to it and I don't get the asterisk commant |
| 11:50.22 | kippi | d |
| 11:51.13 | *** join/#asterisk fugitivo (n=ajf@209.13.241.249) |
| 11:51.16 | iDunno | how about asterisk -r -c |
| 11:51.19 | fugitivo | morming |
| 11:51.33 | kippi | iDunno: Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 11:51.44 | iDunno | well, does it? ;) |
| 11:51.57 | RoyK | hm.... |
| 11:51.59 | iDunno | and are you a user that has permission to access that file? |
| 11:52.10 | RoyK | app_rxfax doesn't seem to set its variables :( |
| 11:52.30 | kippi | yeah |
| 11:52.47 | kippi | ah |
| 11:53.17 | kippi | permission denied |
| 11:53.39 | iDunno | so, you have no permissions on the file ;) |
| 11:53.54 | kippi | iDunno: chmod the file? |
| 11:53.56 | iDunno | that's why you can't connect to it. not an error, just a simple permissions problem ;) |
| 11:54.34 | iDunno | depends on what you're planning on doing, I tend to leave it and become the real asterisk user when I want to connect to it. |
| 11:54.40 | ful|work | have asterisk started ? |
| 11:54.51 | ful|work | strace -eopen asterisk -vvvdg -c |
| 11:54.56 | ful|work | and check the permissions you got |
| 11:55.32 | kippi | i get |
| 11:55.32 | kippi | Asterisk Dynamic Loader Starting: |
| 11:55.33 | kippi | open("/etc/asterisk/modules.conf", O_RDONLY) = 11 |
| 11:55.33 | kippi | <PROTECTED> |
| 11:55.51 | ful|work | there you got |
| 11:55.55 | ful|work | no chan_modem.so |
| 11:56.46 | kippi | how can i install it? |
| 11:57.17 | ful|work | make; make install ? |
| 11:58.23 | *** join/#asterisk popvoxdave (i=user@dave2.toad.net) |
| 11:59.45 | *** part/#asterisk popvoxdave (i=user@dave2.toad.net) |
| 12:05.50 | *** join/#asterisk Blazint (n=blazin@222.164.203.225) |
| 12:06.45 | kippi | is there a how to? |
| 12:09.28 | ful|work | voip-info ? |
| 12:09.31 | gaggaman | I am still stuck with call pickup - could anybody help? |
| 12:09.50 | jac]Z[oby | so |
| 12:09.53 | jac]Z[oby | once again |
| 12:09.57 | jac]Z[oby | new question |
| 12:10.13 | jac]Z[oby | is there a possibility of securing asterisk behind a firewall |
| 12:10.22 | jac]Z[oby | closing some ports |
| 12:10.26 | jac]Z[oby | or whatever |
| 12:10.33 | jac]Z[oby | is there a list of some sort |
| 12:10.47 | gaggaman | depends what you are using. |
| 12:10.53 | gaggaman | SIP only? |
| 12:11.00 | *** join/#asterisk coppice (n=chatzill@123.192.17.210.dyn.pacific.net.hk) |
| 12:11.05 | jac]Z[oby | asterisk@home with sip and h323 |
| 12:11.25 | jac]Z[oby | but sib alone for the biginning woulb be ok 2 |
| 12:11.31 | gaggaman | don't know about h323 |
| 12:12.08 | gaggaman | for SIP, port 5060 UDP incoming and outgoing should be enough. |
| 12:12.17 | jac]Z[oby | really??? |
| 12:12.57 | gaggaman | as far as I know |
| 12:13.29 | gaggaman | if you changed your SIP port in SIP.conf, you have to use this, of course |
| 12:13.33 | jac]Z[oby | what bout the whole rtp shabang |
| 12:14.12 | *** join/#asterisk K-Bear (n=k-bear@h29n2fls32o815.telia.com) |
| 12:14.32 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:14.39 | gaggaman | udp teles. |
| 12:16.06 | gaggaman | jac: http://www.voip-info.org/tiki-index.php?page=Asterisk+firewall+rules |
| 12:16.54 | gaggaman | if both your phone and the * are on official IPs that should be all. |
| 12:18.26 | gaggaman | if the phone is NAT, you have to use the NAT settings in SIP.conf and make shure the NATting router is forwarding the SIP UDP packets to the phone. |
| 12:19.16 | *** join/#asterisk K-Bear (n=k-bear@h29n2fls32o815.telia.com) |
| 12:21.28 | JimmyCarter | Is there any way to make channels linked before the phone is picked up? |
| 12:23.23 | *** join/#asterisk MuppetMaster (n=MuppetMa@169.red-81-184-73.user.auna.net) |
| 12:23.26 | MuppetMaster | Hello |
| 12:23.51 | MuppetMaster | Is there an 'allowguest' option in the sip.conf for allowing guests access to your SIP channel on *? |
| 12:25.53 | drumkilla | MuppetMaster: yes ... and it is called ... "allowguest" ! |
| 12:25.56 | drumkilla | :) |
| 12:26.04 | drumkilla | in the [general] section of sip.conf |
| 12:26.13 | MuppetMaster | Is it anywhere on the Wiki? I can not find it in relation to sip.conf |
| 12:26.18 | drumkilla | no clue |
| 12:26.24 | drumkilla | it's a wiki ... |
| 12:26.25 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 12:26.30 | drumkilla | look in the sample configuration file |
| 12:26.33 | MuppetMaster | Indeed |
| 12:28.22 | tzafrir_laptop | Hi, when I "call" from an MusicOnHold extension to an Echo extension using a local channel and a call file, asterisk does not seem to do any work. |
| 12:28.38 | tzafrir_laptop | As if the channel is "on hold" |
| 12:29.17 | tzafrir_laptop | Any way to avoid that in order to properly load my asterisk server without using tons of real phones? |
| 12:36.37 | *** join/#asterisk heison (n=heison@ns.somanetworks.com) |
| 12:37.00 | JimmyCarter | anyone familar with asterisk-java ?? |
| 12:37.07 | MuppetMaster | A bit |
| 12:37.49 | *** join/#asterisk eivindtr (n=wingnut-@194.248.208.94) |
| 12:38.00 | JimmyCarter | have you ever used SIPPeersAction ? kinda like show sip peers |
| 12:38.11 | MuppetMaster | No, I have not. |
| 12:38.35 | JimmyCarter | ok then, its just that I get a wierd error everytime I use it. |
| 12:40.13 | kippi | can anyone help me with installing chan_modem.so as i am geting stuck |
| 12:45.16 | *** join/#asterisk igori (n=Igor@194.84.91.2) |
| 12:45.20 | *** part/#asterisk igori (n=Igor@194.84.91.2) |
| 12:45.26 | MuppetMaster | Well, put allowguest in the sip.conf section of the wiki. |
| 12:45.26 | *** join/#asterisk igori (n=Igor@194.84.91.2) |
| 12:45.33 | *** part/#asterisk igori (n=Igor@194.84.91.2) |
| 12:45.49 | MuppetMaster | Anyone know how to resolve the problem that is being experienced here? http://forums.digium.com/viewtopic.php?p=5961#5961 Specifically with FritzBoxes in Germany? |
| 12:46.15 | *** join/#asterisk inspired (i=mikael@213.197.167.61) |
| 12:47.04 | inspired | anyone using queues here with SIP agents? |
| 12:47.09 | *** join/#asterisk igori (n=Igor@194.84.91.2) |
| 12:47.10 | inspired | not Agent agents |
| 12:47.15 | *** part/#asterisk igori (n=Igor@194.84.91.2) |
| 12:47.42 | inspired | member => sip/1 instead of agent/1 |
| 12:47.44 | tzanger | damn |
| 12:47.52 | tzanger | I just had the wct4xxp driver take a big shit |
| 12:47.58 | tzanger | nothing useful in logs |
| 12:48.05 | tzanger | just zap/xx channels go crazy |
| 12:49.22 | *** join/#asterisk eivindtr (n=wingnut-@194.248.208.94) |
| 12:51.44 | *** join/#asterisk tehdely (n=delysiid@home.teambarry.org) |
| 12:52.28 | *** join/#asterisk AgentBlueUK (n=agentblu@premierit-66.premierit.com) |
| 12:52.49 | AgentBlueUK | anyone got a second for an iaxy query ? |
| 12:53.17 | *** join/#asterisk astoria (n=haydenth@66.235.201.217) |
| 12:53.20 | astoria | Good Morning, all. |
| 12:53.57 | K-Bear | AgentBlueUK: Asking the question instead will probably get you more attention |
| 12:54.15 | K-Bear | :-) |
| 12:55.04 | MuppetMaster | Someone has developed an Oracle native Realtime driver: http://forums.digium.com/viewtopic.php?t=1968 |
| 12:55.09 | Katty | mew |
| 12:55.46 | tzanger | morning |
| 12:55.47 | Katty | why do they sell baglets of peanuts at the gas station, and then put them in packages which are impossible to open with your barehands? |
| 12:55.51 | tzanger | <-- in Pittsburgh, PA |
| 12:56.07 | MuppetMaster | I am using Asteriskv1.2beta and I notice everytime I do a Playback of a sound file in a dialplan the first second or so gets cut. Even though I did answer and Wait (1) [even if I wait 2 it still happens]. How does one get this to stop? |
| 12:56.33 | Katty | i think it's a conspiracy, and they have the insurance company in on it, hoping that you'll have a car wreck! |
| 12:56.35 | tzanger | MuppetMaster: I had that problem with a VOIP provider but never without them |
| 12:56.47 | tzanger | Katty: you should use scissors |
| 12:56.54 | Katty | tzanger: but while driving? |
| 12:57.01 | tzanger | you're a woman, you should have a purse with every tool imaginable |
| 12:57.10 | Katty | tzanger: it's just the concept. if you buy something at a gas station, you should be able to open it! |
| 12:57.15 | Katty | tzanger: hardly. |
| 12:57.27 | Katty | tzanger: i do'nt carry scissors around. |
| 12:57.36 | Katty | it's simply illogical. |
| 12:58.29 | tzanger | illogical? if I had a pouch I'd have various tools in it |
| 12:58.38 | tzanger | leatherman especially |
| 12:58.42 | Katty | tzanger: i think you missed something. |
| 12:58.48 | Katty | tzanger: this is a girly rant. |
| 12:58.48 | AgentBlueUK | when provisioning the iaxy should you set your internal ip as the alt address as the inteernal and the global as the primary |
| 12:58.59 | Katty | tzanger: you're not supposed to be offering me solutions to fix my problem. |
| 12:59.21 | AgentBlueUK | i have the iaxy that way round an on our lan.but the box just keeps registering over and over |
| 12:59.37 | tzanger | Katty: I'm not? |
| 12:59.37 | tzanger | I'm a |
| 12:59.41 | tzanger | <PROTECTED> |
| 12:59.58 | Katty | tzanger: no, you're not. your a chronic fixit male. |
| 13:00.02 | Katty | tzanger: but you can /change/ |
| 13:00.08 | Katty | tzanger: primarily, by shutting up and listening ;) |
| 13:00.29 | tzanger | Katty: yeah my ex wife told me the exact, exact same thing |
| 13:00.44 | Katty | it's all in the psychology! |
| 13:02.11 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@mi.origenfinancial.com) |
| 13:03.07 | Katty | hi mike |
| 13:03.35 | MikeJ[Laptop] | hello |
| 13:05.07 | Katty | you know in algebra, where you're solving for x and you have to take the square root of each side of the equation, etc. is there an actual useful purpose? like a practical application? |
| 13:05.52 | MikeJ[Laptop] | of solving for a variable? |
| 13:05.58 | MikeJ[Laptop] | sure |
| 13:06.14 | iDunno | maths is all practical... |
| 13:06.14 | Katty | specifically the square root. |
| 13:06.28 | Katty | MikeJ[Laptop]: give me a senario in which it would apply. |
| 13:06.36 | MikeJ[Laptop] | yes, for being able to solve for a variable |
| 13:06.37 | Katty | MikeJ[Laptop]: not an equation, but a pratical aplication. |
| 13:06.43 | iDunno | I'd like to know why you're taking the square root of both sides... ;) |
| 13:07.01 | Katty | iDunno: because, obviously, one of the variables has a power of 2 on it. |
| 13:07.24 | iDunno | well, if it's a standard quadratic, then you just use the quadratic formula. |
| 13:07.34 | Katty | iDunno: you're missing my question. |
| 13:07.40 | Katty | iDunno: i know /how/ to do it. |
| 13:07.43 | Katty | iDunno: i'm very good at math. |
| 13:07.44 | `Sauron | Katty: Well, stuff like c^2 = a^2 + b^2 is useful |
| 13:07.46 | MikeJ[Laptop] | sure, for being able to equate a distance between 2 points, when all you know is the distance from the point you are at, to each of those points (when you are at a right angle to those points of course) |
| 13:08.02 | `Sauron | And you end up having to do a bunch of sqrt() |
| 13:08.12 | MikeJ[Laptop] | hey, that was my example :P |
| 13:08.16 | Katty | `Sauron: you're also missing the point. |
| 13:08.18 | iDunno | that's pythagoras ;) |
| 13:08.25 | Katty | MikeJ[Laptop]: that's..uhh.. |
| 13:08.29 | Katty | MikeJ[Laptop]: not quite parsing in my head. |
| 13:08.53 | `Sauron | Katty: I'm not missing your point. You're missing my answer. |
| 13:09.10 | `Sauron | You asked if there is an actual useful purpose. |
| 13:09.15 | `Sauron | The answer is "Yes" |
| 13:09.24 | Katty | `Sauron: i didn't ask for yes. |
| 13:09.36 | MikeJ[Laptop] | Katty, right triangle, you are at the right angle portion, you know the distance from you to the 2 other points, you want to know the distance between the points |
| 13:09.38 | Katty | `Sauron: nor did i ask for no. |
| 13:09.42 | Katty | `Sauron: mike's the only one who apparently understood my request. |
| 13:09.42 | spiekey | could anyone please give me a hin with this prolem? http://lists.digium.com/pipermail/asterisk-users/2005-October/130061.html |
| 13:09.53 | MikeJ[Laptop] | it is useful in surveying |
| 13:10.08 | Katty | MikeJ[Laptop]: will you draw me a picture? |
| 13:10.15 | MikeJ[Laptop] | and many other science\phisics\ME type applications |
| 13:10.30 | MikeJ[Laptop] | sure.. in your head.. you know what a right triangle looks like, right? |
| 13:10.52 | `Sauron | Katty: And even though Mike and I are talking about the exact same thing, I missed the point and he didn't? |
| 13:10.56 | `Sauron | Hehn. |
| 13:10.56 | Katty | MikeJ[Laptop]: of course. |
| 13:11.23 | MikeJ[Laptop] | say you are standing at the flagpole, due east 1 mile is the washington monument, due north one mile is the lincoln memorial (these numbers are made up) |
| 13:11.24 | Katty | MikeJ[Laptop]: top point is A, point at the 90 degree angle is B and the other one is C |
| 13:11.40 | iDunno | Katty: well, I understand the request, you want a practical *reason* and application for it... basically the practical application is to get the value of the unknown. |
| 13:11.40 | Katty | MikeJ[Laptop]: let's stick with a b and c |
| 13:11.49 | Katty | MikeJ[Laptop]: i take it i'm standing at C |
| 13:11.54 | Katty | MikeJ[Laptop]: I mean B |
| 13:12.04 | MikeJ[Laptop] | no, a |
| 13:12.05 | MikeJ[Laptop] | heh |
| 13:12.12 | `Sauron | Yawn. |
| 13:12.20 | MikeJ[Laptop] | you are standing at the right triangle side.. |
| 13:12.23 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 13:12.45 | jake1932 | all this for an analogy |
| 13:12.52 | MikeJ[Laptop] | so the one side is let's say 2 miles, and the other side is lets say 2 miles |
| 13:13.00 | `Sauron | jake: I would've just told her to stfw |
| 13:13.00 | Katty | yeah well at least i care (= |
| 13:13.05 | MikeJ[Laptop] | what is the distance between the other 2 points |
| 13:13.09 | Katty | not just a bunch of equations you stick in your head. |
| 13:13.12 | MikeJ[Laptop] | sqrt(8) |
| 13:13.14 | Katty | MikeJ[Laptop]: k |
| 13:13.38 | newl | who cares so long as the angles are congruent. 8) |
| 13:13.48 | `Sauron | Mike: I'm dissapointed you didn't use 5^2 = 3^2 + 4^2 |
| 13:13.53 | `Sauron | :p |
| 13:14.04 | *** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 13:14.05 | Katty | heh |
| 13:14.08 | Ariel_ | morning all |
| 13:14.10 | Katty | you guys should really grow up :) |
| 13:14.15 | Katty | seriously. |
| 13:14.19 | Katty | Ariel_: morning. |
| 13:14.23 | Katty | MikeJ[Laptop]: thanks for explination (= |
| 13:14.37 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:14.45 | MikeJ[Laptop] | np |
| 13:14.53 | `Sauron | Katty: surprisingly enough, that's the most practical application of Pythagoras' theorem, because it allows anyone to (easily) create 90deg corners... |
| 13:14.58 | *** join/#asterisk toddf (n=toddf@net-66-210-104-252.theshop.net) |
| 13:15.28 | `Sauron | Since a triangle with short sides of 3 and 4, has a long side of 5 |
| 13:15.36 | jake1932 | has anyone tried to apply the rpid patch to head recently? |
| 13:15.49 | jake1932 | i'm getting "4 out of 7 hunks FAILED" |
| 13:16.30 | `Sauron | shrug, whatever |
| 13:16.33 | *** part/#asterisk MuppetMaster (n=MuppetMa@169.red-81-184-73.user.auna.net) |
| 13:16.37 | newl | fix the four that are failing, recreate a new diff for everyone else. :) |
| 13:17.05 | jake1932 | newl: if I knew how to do that, I would |
| 13:19.05 | jake1932 | is this the correct way to call it: patch -p0<rpid.diff? |
| 13:19.20 | `Sauron | depends on the patch |
| 13:19.27 | `Sauron | sometimes you need a different value for -p |
| 13:19.32 | jake1932 | ok |
| 13:19.59 | *** join/#asterisk gambolputty (n=gambolpu@72.240.241.108) |
| 13:26.12 | jake1932 | `Sauron: it's got to be 0 - 1 and 2 don't even get that far |
| 13:27.31 | `Sauron | okay |
| 13:28.51 | crash3m | does anyone here know of a supplier of motorola vt1000s in the US? communitech has been dropped by motorola :/ |
| 13:29.40 | jake1932 | it doesn't work because it's already in there |
| 13:29.57 | Katty | crash3m: we sell motorola. |
| 13:30.02 | Katty | crash3m: but only locally. |
| 13:30.06 | crash3m | Katty: who is 'we'? |
| 13:30.11 | Katty | crash3m: my company. |
| 13:30.12 | crash3m | Katty: we can pay shipping... |
| 13:30.19 | Katty | crash3m: we only sell locally |
| 13:30.29 | Katty | crash3m: anywhere near st. louis? |
| 13:30.36 | Katty | crash3m: it support issues. |
| 13:30.37 | crash3m | Katty: just acrossed missouri! |
| 13:30.47 | Katty | crash3m: we don't sell anything we can't support. |
| 13:30.50 | crash3m | Katty: support? We're a VoIP provider, we dont need support |
| 13:30.57 | Katty | crash3m: so, we don't sell stuff halfway across the country |
| 13:31.02 | Katty | crash3m: that's beside the point |
| 13:31.06 | Katty | crash3m: company policy, you see. |
| 13:31.23 | crash3m | I can understand that, but again...we get support from motorola, just need a new supplier |
| 13:31.41 | `Sauron | crash3m: She's a girl. Don't argue with her. |
| 13:31.46 | `Sauron | s/girl/company |
| 13:31.52 | crash3m | lol |
| 13:33.02 | Katty | `Sauron: you mock me? |
| 13:33.53 | newl | Only if you're wearing your Buzz suit. |
| 13:34.30 | *** join/#asterisk Jzalae (n=sk@216-220-249-122.midmaine.com) |
| 13:35.05 | *** join/#asterisk slan (n=lba@user-12lml75.cable.mindspring.com) |
| 13:37.28 | Katty | i see. |
| 13:37.50 | Katty | `Sauron: there's no reason to be hateful |
| 13:38.14 | Beirdo | `Sauron: she's a girl, be nice to her. |
| 13:38.17 | `Sauron | I'm not being hateful, just flippant. |
| 13:38.26 | Beirdo | good morning, Katty :) |
| 13:38.46 | Katty | `Sauron: k |
| 13:38.50 | Katty | Beirdo: morning (= |
| 13:39.16 | *** join/#asterisk taec_ (n=phil@eventhorizon.hosting365.ie) |
| 13:40.08 | *** part/#asterisk slan (n=lba@user-12lml75.cable.mindspring.com) |
| 13:41.25 | taec_ | I have an asterisk server setup and incoming calls coming in over an E1-ISDN line. They go to the relevant extensions fine and everything is happy. Outgoing calls aren't happy though, I get an 'All circuits are busy' error message. Using the zaptel driver with a Digium card. I presume I'm trying to make Zap channels available to the extensions, but I can't find much documentation on doing this. |
| 13:41.56 | Katty | taec_: are they seperated into groups? like g1 |
| 13:42.27 | *** join/#asterisk bob_too (n=chris@rrcs-24-153-179-246.sw.biz.rr.com) |
| 13:42.42 | taec_ | There's one group currently, zap/g0 |
| 13:42.46 | Katty | taec_: with particular lines seperated into groups...and then dialout with a group... |
| 13:44.21 | taec_ | I've got one zap trunk - g0 ... does that sound right? |
| 13:45.19 | Katty | exten => _1xxxxxxxxxx,1,Dial,Zap/g1/ww${EXTEN} <- that's what mine looks like |
| 13:46.57 | *** join/#asterisk slan (n=lba@user-12lml75.cable.mindspring.com) |
| 13:47.03 | taec_ | ok, sorry to be a pain, but can you be specific as to where that should be located? |
| 13:47.16 | Katty | that's in extensions.conf |
| 13:47.45 | taec_ | *nods* in what section? |
| 13:48.12 | iDunno | the one that you're dialing out on ;) |
| 13:48.29 | iDunno | (and I think you mean in what context, but hey ;) |
| 13:48.54 | Katty | taec_: i have mine at the bottom, in a section i called [to-phone-company] |
| 13:50.00 | taec_ | Hmmm. OK, this might sound rather silly, but bear with me, I'm still finding my feet. I am aware of the different contexts in extensions.conf, but so far I haven't found a way to see how asterisk is entering them. Is it pre-defined? HOw does asterisk know to enter the context [to-phone-company] (for example) when dialing out? |
| 13:52.05 | *** part/#asterisk CoffeeIV_ (n=CoffeeIV@mail.airlinksystems.com) |
| 13:52.07 | *** join/#asterisk ennuyeux72 (n=ennuyeux@host-83-146-53-34.bulldogdsl.com) |
| 13:52.23 | jake1932 | taec_: your outgoing context is defined in your config files (sip.conf for sip and iax.conf for iax) |
| 13:52.45 | *** join/#asterisk michael1234 (i=michael3@escazu-a259.racsa.co.cr) |
| 13:52.50 | ennuyeux72 | has anyone got realtime with either resmysql or resodbc working on a loaded asterisk server |
| 13:52.51 | Katty | taec_: you have two sections, incoming and outgoing. my two contexts i've setup are [from-sip] and [from-zap] ...they predefined in other files like zapata.conf |
| 13:52.59 | michael1234 | I have a wierd problem if I put a call on hold I can get it back |
| 13:53.06 | michael1234 | and the other end hears music |
| 13:53.18 | Katty | taec_: in extensions.conf, you put in those two same contexts under the [from-zap] i have what to do with incoming calls. |
| 13:53.35 | michael1234 | has anyone seen this problem |
| 13:53.40 | Katty | taec_: under [from-sip] i have includes. include [to-phone-company] and include [extensions] etc |
| 13:54.17 | Katty | taec_: |
| 13:54.20 | Katty | [from-sip] |
| 13:54.20 | Katty | include => to-phone-company |
| 13:54.32 | Katty | taec_: there's lots of things under [from-sip] (= |
| 13:55.15 | taec_ | hehe :) ... ok quick one so, I can see [default] is in zapata.conf and [from-sip-external] is in sip.conf but there seem to be no other contexts defined. there's no [from-sip] or [from-sip-internal] |
| 13:56.05 | Katty | taec_: that's because i changed them |
| 13:56.16 | *** join/#asterisk kore (i=kore@mindwipe.org) |
| 13:56.19 | Katty | taec_: easier for me to read when i come back a few months later |
| 13:56.45 | Katty | zapata is coming /from/ zaptel |
| 13:56.50 | taec_ | *nods* |
| 13:56.51 | Katty | sip is coming /from/ sip phones |
| 13:56.53 | *** part/#asterisk kore (i=kore@mindwipe.org) |
| 13:57.14 | Katty | iDunno: i have iax outgoing too though. so i'd confuse myself. |
| 13:57.16 | taec_ | No no I get that, I was just wondering why there were no other contexts beside "unknown callers" in my sip.conf, but I missed the include files at the bottom |
| 13:57.30 | taec_ | [from-internal] is the context that's set for our extensions in here |
| 13:57.38 | Katty | k (= |
| 13:57.51 | *** join/#asterisk ic (n=ic@staff.rbi.speka.net) |
| 13:57.59 | iDunno | Katty: ahh - the outgoing handles both out via ISDN and sipgate. |
| 13:58.21 | iDunno | I haven't quite split those apart yet, mainly because I need it to fall over from one to the other. |
| 13:58.21 | Katty | iDunno: we have analog lines, actually. |
| 13:58.45 | Katty | iDunno: in include parked calls, extensions, internal from sip to sip (4 digit matching calls) and then to the phone company |
| 13:58.47 | *** join/#asterisk dalabera (n=Dalabera@pmr.pmrtechnologies.com) |
| 13:58.49 | iDunno | wow! cool! what are you using to talk to the analogue lines? |
| 13:58.58 | Katty | tdm400s |
| 13:59.06 | iDunno | makes sense. |
| 13:59.18 | Katty | we only have 8 lines in the building. |
| 13:59.19 | jake1932 | you have any echo issues at all? |
| 13:59.29 | Katty | jake1932: well it's digium hardware. |
| 13:59.34 | Katty | jake1932: not sangoma |
| 13:59.34 | jake1932 | heh |
| 13:59.38 | Katty | jake1932: so echo is a given. |
| 14:00.11 | Katty | jake1932: it's canceled out in a just a couple seconds. |
| 14:00.24 | tzanger | I need to interface * to an IWATSU omega-phone (ADIX) |
| 14:00.26 | jake1932 | Katty: yep - same thing i experienced |
| 14:00.30 | tzanger | T1 trunk card of course |
| 14:00.33 | Katty | jake1932: sangoma cards will, as i hear, cancel out the echo at the card. |
| 14:00.40 | Katty | jake1932: based on particular voltages, me thinks. |
| 14:00.48 | Katty | jake1932: or so 'joey' said. |
| 14:00.59 | *** join/#asterisk supa_thygar (i=thygar@tpr-165-255-171.telkomadsl.co.za) |
| 14:01.12 | tzanger | based on particulr voltages?? |
| 14:01.12 | supa_thygar | hi |
| 14:01.15 | Katty | he looked like joey from friends....worked at sangoma |
| 14:01.20 | jake1932 | Katty: tnx - i'll try to find a sangoma card |
| 14:01.29 | supa_thygar | i need to chat to some clever ppl any in here ? :) |
| 14:01.33 | Katty | jake1932: i'm not sure they've got those analog cards on the market yet |
| 14:01.47 | *** join/#asterisk synthetiq (n=roger@64.201.13.50) |
| 14:01.51 | jake1932 | but they work? |
| 14:01.55 | tzanger | I want to increase the taps on the KB1 canceller, give me 32ms or 64ms tail |
| 14:01.56 | jake1932 | :) |
| 14:02.08 | tzanger | 128 taps is only 16ms |
| 14:02.27 | *** join/#asterisk MatsK (n=mk@99.80-202-83.nextgentel.com) |
| 14:02.28 | coppice | or 64 bath tubs |
| 14:03.06 | Katty | jake1932: that's what 'joey' said. |
| 14:03.16 | taec_ | Right, excellent, thanks a million. Now to go find out how to break up the Zap channels into groups |
| 14:03.17 | Katty | jake1932: it's canceled right at the card, before it even goes to asterisk |
| 14:03.35 | Katty | taec_: i think that's in zaptel.conf ...not sure though |
| 14:03.38 | jake1932 | joey didn't mention a model number, did he? |
| 14:03.51 | Katty | jake1932: no, he only said they were still in the testing phase |
| 14:03.58 | mutilator | my birthday is friday!!!! send presents for me! |
| 14:04.03 | Katty | jake1932: bkw would know. |
| 14:04.09 | Katty | jake1932: and so would anthm probably |
| 14:04.10 | synthetiq | is there a module that allows you to hold registrations via real time? |
| 14:04.27 | jake1932 | k - tnx Katty |
| 14:04.33 | taec_ | Katty: zapata.conf, no? |
| 14:04.37 | taec_ | zaptel.conf is the driver configuration file? |
| 14:04.40 | Katty | taec_: i don't recal. |
| 14:04.43 | Katty | i mean recall. |
| 14:04.56 | Katty | taec_: i'd have to go digging through the conf files looking for it |
| 14:05.07 | Katty | taec_: somewhere you define which channels are for which groups. |
| 14:05.10 | taec_ | sure, ok. Seriously though, thanks for that. Really appreciate it, I am capable of understanding a lot more now! |
| 14:05.20 | Katty | taec_: yay |
| 14:05.36 | taec_ | The configuration file was hurting my head before :) |
| 14:05.42 | *** join/#asterisk zigman (n=zigman@irc.zigman.de) |
| 14:06.11 | Katty | anthm made mine all pretty....mine was a little messy and hard to read. |
| 14:06.32 | iDunno | zapata.conf ;) |
| 14:06.37 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 14:07.53 | synthetiq | is there a module that allows you to hold registrations via real time in mysql db? |
| 14:08.04 | taec_ | Katty, one more quick q .... the 'ww' in the exten => line for outbound dialing, what's that for? |
| 14:08.47 | taec_ | the Zap/g1/ww${EXTEN} ... what's the ww for? |
| 14:08.54 | synthetiq | wait |
| 14:08.58 | *** join/#asterisk gabb0 (n=gabb0@131.202.90.23) |
| 14:09.22 | synthetiq | youc an take it out |
| 14:09.54 | Katty | taec_: that's for waitwait |
| 14:10.05 | Katty | taec_: like when you're using a modem to dial up the internet, you have to wait for a dialtone |
| 14:10.10 | taec_ | ahh yeh |
| 14:10.22 | Katty | taec_: asterisk doesn't 'wait for a dialtone' instead we ww to make sure we have a dialtone first |
| 14:10.39 | Katty | taec_: otherwise, half of your calls will never go through. |
| 14:11.01 | Katty | taec_: that might only apply to analog lines, dunno |
| 14:12.25 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 14:13.29 | *** join/#asterisk docE (n=docE@66.237.242.41.ptr.us.xo.net) |
| 14:13.48 | docE | Does anyone know of any issues with Asterisk's H323 Module and Opterons? |
| 14:14.10 | *** join/#asterisk MuppetMaster (n=MuppetMa@169.red-81-184-73.user.auna.net) |
| 14:14.15 | MuppetMaster | Hello |
| 14:14.23 | docE | whadup |
| 14:14.30 | MuppetMaster | Does anyone know where to find PHPVoIPMail (http://www.kevinelliott.net/asterisk/AVC/about.php)? The links on that page do not work. |
| 14:14.44 | *** join/#asterisk fanguin (n=user@p548F6085.dip.t-dialin.net) |
| 14:14.47 | *** join/#asterisk wolfson (n=hehe@usr-kdh-208-6-58-26.beachlink.com) |
| 14:14.53 | docE | This isnt the place to be asking.. but I would imagine you could google it. |
| 14:15.31 | MuppetMaster | This is the Asterisk IRC, no? |
| 14:15.39 | docE | Yes |
| 14:15.40 | MuppetMaster | And this is a tool for Asterisk. And yes, I did Google, but no joy. |
| 14:15.41 | iDunno | yes. |
| 14:15.55 | iDunno | maybe it's gone then. |
| 14:16.02 | *** part/#asterisk dalabera (n=Dalabera@pmr.pmrtechnologies.com) |
| 14:16.04 | MuppetMaster | But the Java client is still there (http://www.kevinelliott.net/asterisk/AVC/about.php). |
| 14:16.06 | iDunno | maybe PHPVoIPMail was deemed evil. |
| 14:16.07 | docE | But asterisk related questions. Not 3rd party |
| 14:16.17 | MuppetMaster | Seems the PHP scripts were kept somewhere else, I emailed the guy. |
| 14:16.17 | docE | hold on I will find it. |
| 14:16.20 | *** join/#asterisk dalabera (n=Dalabera@pmr.pmrtechnologies.com) |
| 14:16.39 | iDunno | (or just didn't work, which would sum up most php applications, really) |
| 14:17.07 | MuppetMaster | docE First time I have heard the IRC is not for asking questions related to Asterisk, which of course this is. Asterisk has all sorts of plug-ins and add-ons, if it were limited to core Asterisk this would really stifle the discussion. |
| 14:17.15 | docE | funny.. I do tons in PHP and they work better than most MS compiled |
| 14:17.18 | MuppetMaster | I have seen things much further off topic spoken about here. ;) |
| 14:17.27 | MuppetMaster | docE Yes, I like PHP as well. |
| 14:17.37 | MuppetMaster | vmail.cgi is kind of ugly. |
| 14:17.42 | MuppetMaster | No offense intended. |
| 14:18.37 | iDunno | docE: erm - MS really suck though... |
| 14:18.48 | iDunno | (and perl) |
| 14:18.52 | docE | This is true.. its usually Twisted's fault.. |
| 14:18.54 | iDunno | PHP is buggy as hell. |
| 14:18.58 | docE | naa.. |
| 14:19.02 | MuppetMaster | iDunno Really? |
| 14:19.10 | docE | anywho.. dude.. your project is missing.. |
| 14:19.19 | MuppetMaster | docE Yes, missing in action. |
| 14:19.22 | docE | Contact your guy and tell him to fix his links.. |
| 14:19.29 | MuppetMaster | Well, I emailed the guy, and we will see if he comes back. |
| 14:19.30 | MuppetMaster | Thanks. |
| 14:19.35 | iDunno | MuppetMaster: watch security alerts about PHP sometime ;) |
| 14:19.54 | docE | Anywho.. Anyone doing H323 and Asterisk on a Opteron? |
| 14:19.57 | MuppetMaster | An awful lot of LAMP out there.... |
| 14:20.06 | docE | ME ME! |
| 14:20.09 | *** join/#asterisk lehel (n=asd@82.79.20.17) |
| 14:20.47 | lehel | hello |
| 14:20.58 | ful|work | hey |
| 14:21.00 | MuppetMaster | Hi |
| 14:27.34 | *** join/#asterisk b0xii (i=b0xii@pool-70-110-88-241.dfw.dsl-w.verizon.net) |
| 14:27.36 | iDunno | MuppetMaster: just because there's a lot of it doesn't make it good. |
| 14:27.38 | *** part/#asterisk jac]Z[oby (n=me@193.83.248.26) |
| 14:27.56 | iDunno | MuppetMaster: there's a metric *fuckload* of windows workstations, are you trying to claim that that's good? |
| 14:28.10 | iDunno | ;) |
| 14:28.32 | MuppetMaster | Good point. |
| 14:28.37 | MuppetMaster | I stand corrected. |
| 14:28.39 | MuppetMaster | ;) |
| 14:30.37 | iDunno | :) |
| 14:30.44 | *** join/#asterisk abatista (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 14:32.49 | *** join/#asterisk _T3_ (n=rposada@53.228.uio.satnet.net) |
| 14:33.08 | _T3_ | morning |
| 14:33.12 | *** part/#asterisk MuppetMaster (n=MuppetMa@169.red-81-184-73.user.auna.net) |
| 14:36.25 | bob_too | does the * test connection actually route to people? |
| 14:37.19 | riksta | it tries to connect to an IAX server |
| 14:37.45 | *** join/#asterisk pindonga (n=ricardo@209.99.226.48) |
| 14:37.58 | bob_too | riksta: right, but if i press zero, does a peson answer. guess i could find out. :) |
| 14:41.22 | *** join/#asterisk AlexCTI (n=alex@weston-69.65.86.197.myacc.net) |
| 14:41.23 | *** join/#asterisk clyrrad (n=ddd@CPE001195f553c7-CM0011aea484a4.cpe.net.cable.rogers.com) |
| 14:41.30 | *** join/#asterisk santiago (n=santiago@208.195.215.158) |
| 14:42.10 | clyrrad | How can I specify ${ACCOUNTCODE} for an IAX account in iax.conf? |
| 14:42.18 | AlexCTI | Hi, Anyone can help to convert MP3 files to raw format? |
| 14:43.12 | jake1932 | AlexCTI: did you try here: http://www.voip-info.org/wiki-Asterisk+mpg123+faking+it |
| 14:43.47 | riksta | bob_too: its recorded |
| 14:43.53 | AlexCTI | I'll try |
| 14:43.54 | tzafrir_laptop | works great for us. reduces CPU usage and the quality is the same |
| 14:44.21 | clyrrad | Does anyone know the answer to my question? |
| 14:44.40 | jake1932 | AlexCTI: sox supposedly handles mp3 with an add-in |
| 14:45.01 | tzafrir_laptop | I'll ask again: is there any reason that when I send a call through a Local channel, Asterisk seems to consider it as "muted"? |
| 14:45.20 | *** part/#asterisk bob_too (n=chris@rrcs-24-153-179-246.sw.biz.rr.com) |
| 14:45.27 | tzafrir_laptop | sox's built-in mp3 support isn't working well. The point is to do the transcoding beforehand |
| 14:45.38 | *** part/#asterisk pindonga (n=ricardo@209.99.226.48) |
| 14:45.50 | tzafrir_laptop | And play the raw file. |
| 14:46.00 | tzafrir_laptop | It actually may even not be larger |
| 14:46.37 | tzafrir_laptop | My question is: how can I create many calls on an Asterisk server for loading it |
| 14:47.20 | jake1932 | clyrrad: accountcode=xxx? |
| 14:47.24 | Ahrimanes | tzafrir_laptop: http://www.astertest.com/ |
| 14:47.56 | *** join/#asterisk mrtwister (n=mrtwiste@cable-9-42.cgates.lt) |
| 14:48.38 | clyrrad | jake1932 I have tried that I figured that would be the way to do it but does not work |
| 14:48.41 | *** join/#asterisk rob112 (n=robert@212.183.128.185) |
| 14:49.20 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 14:49.29 | rob112 | hi, can anyone help with a dsp.c issue? |
| 14:49.50 | *** join/#asterisk kippi (n=chrisfro@untrust-gct.equinoxit.net) |
| 14:49.52 | kippi | hey |
| 14:50.02 | rob112 | hi, can anyone help with a dsp.c issue? |
| 14:50.16 | kippi | I have to sip phones that are trying to connect, and i am getting this error Oct 19 15:47:25 NOTICE[9682]: chan_sip.c:9001 handle_request_register: Registration from '<sip:6696@10.69.69.20;user=phone>' failed for '10.69.69.88' |
| 14:54.38 | astoria | tzafrir_laptop: put your toll free number on USENET boards, advertising free phone sex.. :) |
| 14:54.50 | Ahrimanes | haha |
| 14:55.00 | Beirdo | "for a good time call..." |
| 14:55.07 | ManxPower | you do not have a [669] section of sip.conf |
| 14:55.15 | Ahrimanes | and do remember to turn off call forwarding |
| 14:55.16 | astoria | Yeah, I can scribble your phone number in bathroom stalls for you if you'd like.. |
| 14:55.24 | rob112 | hi can someone help me with a dsp.c problem? |
| 14:55.46 | ManxPower | rob112, Best of luck. Most of us consider dsp.c black magic. |
| 14:56.23 | rob112 | lol, oh. What are the chances of converting the US tone section in call progress to UK, has anyone seen it done? |
| 14:57.06 | ManxPower | rob112, I don't know. The US tone detection doesn't really work anyway. |
| 14:57.44 | *** join/#asterisk [Lamer] (i=Lamer@221.128.102.106) |
| 14:58.28 | rob112 | oh great, ive been working with NVLineDetect, trying to get the UK tones recognised. Any better ideas for call progress indication other than dsp? |
| 14:59.17 | ManxPower | rob112, *I* don't really have any suggestions. In the USA we don't need tone detection. The telco drops battery on the line for .5 second and Asterisk detects that just fine. |
| 14:59.39 | coppice | ManxPower: dsp.c isn't something black. its something brown and sticky :-) |
| 15:00.23 | tzanger | coppice: hahaha |
| 15:00.31 | rob112 | lol, thanks ManxPower. Banging my head against a brick wall on this one! Tried speaking to as many specialists as i can, getting no where fast. |
| 15:00.31 | ManxPower | Mostly it's non-USA/non-Canada that needs tone detection. rob112, your best bet might be to go thru the work to get ISDN BRI working with Asterisk and use an ISDN BRI line. That would eliminate all your disconnect and tone problems. |
| 15:01.00 | ManxPower | And in .EU ISDN BRI lines are reasonably priced. |
| 15:01.01 | *** join/#asterisk tecnico (n=tecnico@user-24-236-120-2.knology.net) |
| 15:01.17 | rob112 | Thought about that, ideally becuase of cost i would like the detection over IAX2. I just need something that well tell me if the channel picked up a tone. |
| 15:01.29 | rob112 | i could probably work the rest out... just not that simple i guess. |
| 15:01.32 | *** join/#asterisk tekati (n=captain@cpe-66-75-215-63.bak.res.rr.com) |
| 15:02.26 | rob112 | NVLine will detect on the fly over IAX2 but i keep falling over this dsp.c call progress prob. No matter what changes i make i seem to be getting further away. |
| 15:02.36 | ManxPower | rob112, you might want to check the asterisk-dev mailing lists and maybe ask on that mailing lists as well. |
| 15:02.38 | ManxPower | ~mailinglsit |
| 15:02.44 | [Lamer] | anyone knows of a digium card with more than 4 fxos? |
| 15:03.03 | rob112 | lol, done that today, hoping for a response, even thinking about putting a bounty out there to. |
| 15:03.16 | ManxPower | rob112, Unless you are using ulaw/alaw tones will be distorted by compressed codecs. |
| 15:03.55 | rob112 | See this is where i lack in experience, even though i hear the tone correctly are saying that the module wont? |
| 15:04.47 | ManxPower | rob112, The module will have to be more relaxed/liberal about variance in the tone. |
| 15:05.49 | rob112 | Re-thinking what i just said, i think the module is relying on the response it gets from dsp.c |
| 15:06.31 | rob112 | intresting, ive been using the GSM codec, maybe im going wrong here? gonna switch over to A/ULAW see what happens. Still gonna need those tone changes though? |
| 15:06.50 | devel | a question that somebody may be able to address quickly: we have a t1 (not PRI), and no callerid data. the calls come in with a text label of "asterisk", where does that come from? |
| 15:06.56 | ManxPower | rob112, As I said, asterisk-dev mailing list is your best chance. |
| 15:07.21 | rob112 | ok thanks Manx for your time, will drop in tomorrow let you know how i get on, and if so, seek more advice. |
| 15:07.30 | ManxPower | devel, CT1 (T-1, not PRI) doesn't provide Caller*ID Text. |
| 15:07.45 | *** part/#asterisk rob112 (n=robert@212.183.128.185) |
| 15:08.00 | ManxPower | devel, the callerid= line and usecallerid= line in /etc/asterisk/zapata.conf will be where you configure that stuff. |
| 15:08.25 | devel | ack, ManxPower. i'll just set them to "unknown", what with no cid data ever coming on those trunks. thanks. |
| 15:09.00 | *** join/#asterisk asterisk99 (n=dunoowhy@modemcable169.194-130-66.mc.videotron.ca) |
| 15:09.00 | devel | as a side note to anybody, my boss decided that "it was time" to cut to VOIP so he just forwarded the analog numbers over to our asterisk pbx. |
| 15:09.34 | asterisk99 | Anyone here using Asterisk on Gentoo? I am having probolems with zaptel not running on reboot |
| 15:10.51 | *** join/#asterisk philm (n=a@r43h15.res.gatech.edu) |
| 15:11.42 | philm | Anyone using zaptel on FreeBSD? |
| 15:13.46 | kippi | I am using AMP and it seems to be making the changes, but i am getting this error on asterisk chan_sip.c:9001 handle_request_register: Registration from '<sip:6696@10.69.69.20;user=phone>' failed for '10.69.69.88' |
| 15:13.47 | fulgas | anyone knows how to fix this -> http://pastebin.ca/25966 ? |
| 15:13.49 | kippi | any ideas? |
| 15:14.28 | *** join/#asterisk hanged (n=mefisto@upe.venta.lv) |
| 15:14.50 | fulgas | 10.69.69.88 != 10.69.69.20 |
| 15:15.16 | gaggaman | kippi: obviously the sip phone doesn't register. check credentials ! |
| 15:15.35 | kippi | so its a setting on the phone? |
| 15:16.42 | gaggaman | yes. "device" and "secret" must be the same as in amp. |
| 15:17.53 | *** join/#asterisk dinkdinkdink (n=d@pool-68-238-147-186.dllstx.fios.verizon.net) |
| 15:18.18 | gaggaman | anybody here who could help me with call pickup? |
| 15:19.44 | *** join/#asterisk dererk (n=dererk@unaffiliated/dererk) |
| 15:19.48 | dererk | Hi |
| 15:19.58 | dererk | Really newbie question |
| 15:20.14 | *** join/#asterisk ian_k (n=ian@gateway.digium.com) |
| 15:20.27 | ian_k | does anyone know where a vmail.cgi demo site is setup? |
| 15:20.36 | dererk | I've already ocnfigure extensions, iax, etc, and it says this "No such command 'dial' (type 'help' for help)" |
| 15:21.01 | dererk | Could you give me a tip? |
| 15:21.13 | dererk | I know is a little thing, but I really don't remmember |
| 15:21.42 | dererk | ian_k Could you? |
| 15:22.33 | ian_k | dererk - what's up? |
| 15:23.52 | dererk | I've already ocnfigure extensions, iax, etc, and it says this "No such command 'dial' (type 'help' for help)" |
| 15:23.52 | *** join/#asterisk TrevorSHarrison (n=trevorsh@24.49.36.218) |
| 15:23.57 | *** join/#asterisk myiagy (n=myiagy@200.138.215.78) |
| 15:23.58 | ian_k | type "show application dial" and see if it exists |
| 15:24.07 | ian_k | (from CLI) |
| 15:24.54 | dererk | ian_k It throughs a lot of text |
| 15:25.09 | dinkdinkdink | newb question - I'm getting a console message "INIT: Id "S0" respawning too fast: disabled for 5 minutes - no idea what it's referring to. |
| 15:25.53 | ian_k | dinkdinkdink -- edit /etc/inittab and see what s0 is... then figure out why it isn't working, or comment it out. |
| 15:26.23 | ian_k | dererk - Do you have any extensions defined in your [local] context in extensions.conf? |
| 15:26.50 | *** join/#asterisk logicalonline (n=Ken@209.242.52.25) |
| 15:27.02 | dererk | ian_k Yes, yes I do |
| 15:27.33 | devel | dinkdinkdink, that's usually the serial getty |
| 15:28.02 | ian_k | dererk - gimme a sec.. |
| 15:28.10 | dinkdinkdink | s0:12345:respawn:/sbin/agetty -i -h -L 9600 ttyS0 vt100 - one of the serial ports? |
| 15:28.35 | ian_k | dererk - what is the dial command you are trying to do? (the whole thing) |
| 15:28.53 | ian_k | dinkdinkdink - just comment it out and then type "kill -1 1" |
| 15:29.03 | dinkdinkdink | going to bios to see if it |
| 15:29.06 | dererk | Ummm, I thinks I forgot something now I remember |
| 15:29.08 | ian_k | don't go to bios |
| 15:29.09 | dinkdinkdink | is disabled |
| 15:29.52 | dererk | ian_k I forgot a configuration... i'll try it and then I'll be back |
| 15:30.24 | dinkdinkdink | doh - serial port disabled in bios.. that may do it |
| 15:33.25 | dererk | brb |
| 15:33.42 | *** join/#asterisk paryl (n=paryl@209.236.78.59) |
| 15:33.44 | *** part/#asterisk dererk (n=dererk@unaffiliated/dererk) |
| 15:34.23 | paryl | i've got 3 analog lines coming into my asterisk box and it just occured to me, i don't know how to forward them. |
| 15:34.32 | *** join/#asterisk [Airwolf] (n=airwolf@airwolf.xs4all.nl) |
| 15:34.39 | paryl | does that have to be done through the telco, or is there a method in asterisk? |
| 15:36.59 | copantl | some body knows howto interconnect 2 asterisk with sip trunk? |
| 15:37.12 | copantl | i just like iax trunk? |
| 15:37.20 | astoria | paryl: like a remote-call-forward? |
| 15:37.34 | copantl | yes |
| 15:38.03 | paryl | astoria: yeah. when everyone is out of the office, they forward all of thir lines |
| 15:38.09 | paryl | their* |
| 15:38.23 | copantl | like this: did----asteriskA----siptrunk10channels-----asteriskB |
| 15:38.56 | kippi | is there away to tell what is failing? |
| 15:38.58 | paryl | i guess it would be preferrable to use the telco's forwarding codes, eh? |
| 15:39.15 | kippi | SIP is there away to tell what is failing? |
| 15:40.00 | dinkdinkdink | ian - devel - thx - I ended up remarking the line out.. bios change didn't fix it. I am not planning on using a modem anyway. |
| 15:41.14 | *** join/#asterisk swK[work] (n=SwK@border0hsv.asterisksgi.com) |
| 15:43.20 | paryl | silly question, but can you use the * and # characters in the Dial() commands in extensions.conf? |
| 15:43.54 | astoria | paryl: well, you can do that yourself if you have multiple lines, just have it dial out to the remote line.. |
| 15:44.04 | astoria | paryl: i'm not sure if there is an easy way to get the telco to do that for you. |
| 15:44.48 | paryl | astoria: only issue is that an incoming call would then take up 2 lines, and at that point the customers would get a busy signal |
| 15:45.11 | paryl | the telco would be able to forward an unlimited number of calls to the same number automatically |
| 15:45.15 | *** join/#asterisk yxa (i=empty@cm121.gamma228.maxonline.com.sg) |
| 15:45.52 | paryl | so i'm thinking, maybe i could set up a way for asterisk to automatically dial the forward code (*71 i think) for the specific line and unforward the same way |
| 15:46.07 | astoria | paryl: well, that's remote call forward, but i don't think you'd be able to get them to do that frequently.. |
| 15:46.20 | astoria | paryl: yeah, if there is a forward code, you could do that automatically.. |
| 15:46.39 | astoria | paryl: i'm not too sure about services offered on POTS lines, I usually deal with PRIs |
| 15:46.49 | clyrrad | can anyone help me out with this error: NOTICE[6313]: chan_iax2.c:4968 register_verify: Peer 'xxxxxxxxxx' is not dynamic (from x.x.x.x), I can not use host=dynamic i need host to point to the VOIP provider |
| 15:48.01 | paryl | astoria: yeah, i think we have to add it as an extra service... shouldn't be an issue there |
| 15:48.15 | paryl | astoria: but can i using the * character in a dial string? |
| 15:48.30 | astoria | paryl: yeah, you should be able too.. |
| 15:49.40 | astoria | paryl: i'm sure there are a handful of emails on the mailing list about that. |
| 15:49.43 | *** join/#asterisk lehel (n=asd@82.79.20.17) |
| 15:50.20 | *** join/#asterisk konfuzed (n=konfuzed@H129.C72.B0.tor.eicat.ca) |
| 15:50.48 | clyrrad | Is there a way to set ougoing host only to my voip provider? |
| 15:51.14 | clyrrad | something like host-out= |
| 15:52.16 | fulgas | fromdomain= |
| 15:54.10 | clyrrad | fulgas will that specify to use my IAX host for outgoing calls? |
| 15:54.49 | *** part/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 15:56.21 | clyrrad | becase this is what I get now: Oct 19 11:56:16 WARNING[6313]: chan_iax2.c:6885 socket_read: Call rejected by 192.168.1.24: We are busy! |
| 15:59.22 | fulgas | check http://www.voip-info.org/wiki-Asterisk+config+iax.conf |
| 15:59.31 | fulgas | some providers require fromdomain |
| 15:59.32 | *** join/#asterisk brasco (n=root@83.137.128.7) |
| 15:59.48 | copantl | help me please? |
| 16:00.57 | clyrrad | fulgas... this provider does not require that. What I need to do is have an IAX phone connect to my Asterisk box, then connect out over IAX to my providers Asterisk Box |
| 16:01.09 | devel | copantl, we use sip between our asterisk boxes, but don't bother trunking it |
| 16:01.35 | fulgas | what's the auth type ? |
| 16:01.41 | fulgas | md5, rsa or plaintext? |
| 16:01.41 | clyrrad | md5 |
| 16:02.22 | astoria | i heart md5 |
| 16:03.44 | fulgas | are you using host=dynamic and defaultip ? |
| 16:04.05 | clyrrad | host=dynamic i have, but i do not have defaultip set |
| 16:04.47 | copantl | devel: you use iax for trunk right? |
| 16:05.31 | copantl | i need to interconnect my asterisk with a OCN switch |
| 16:05.45 | copantl | and the only way is SIp trunk |
| 16:06.27 | devel | copantl, actually, we have moved all of our iax "trunks" back to just using sip |
| 16:06.58 | copantl | do you now a howsto for dummys like me |
| 16:07.10 | devel | copantl, do they reference a sip trunking specification? |
| 16:07.41 | copantl | let me understand you just send sip normal connections? |
| 16:07.46 | copantl | or sip trunks ? |
| 16:07.48 | devel | affirmative, copantl |
| 16:07.53 | devel | normal connections |
| 16:08.07 | *** join/#asterisk jpm_SD (n=jpm@207-40-115-38.sugardog.com) |
| 16:08.46 | copantl | but i need to send 10 sips.... i just have to create 10 sip extensions? |
| 16:09.06 | yxa | what is the best way to let users manage their own extensions (ie edit extension.conf) securely? |
| 16:10.01 | iDunno | set it all up via DBput and DBget? |
| 16:10.17 | copantl | devel: but i need to send 10 sips.... i just have to create 10 sip extensions? |
| 16:10.20 | devel | copantl, what do you mean you need to send 10 sips? ten simultaneous calls? |
| 16:10.26 | copantl | yes |
| 16:10.48 | yxa | iDunno i'm quite new to that. can you point me to a url or something? |
| 16:11.00 | devel | normally, that's just "number@host" then the receiving host routes the call. |
| 16:11.52 | iDunno | yxa: I'm fairly new to it, I've not done it yet ;) |
| 16:12.06 | copantl | devel: yes |
| 16:12.08 | *** join/#asterisk frenzy (n=frenzy@196.41.54.233) |
| 16:12.58 | mishehu | alright, the question at hand is... 1 proc dual core opteron, or 2 proc single core opteron for an asterisk + mysql server... |
| 16:13.34 | *** join/#asterisk spiekey (n=spiekey@p549D18C1.dip0.t-ipconnect.de) |
| 16:13.37 | spiekey | hi |
| 16:13.47 | *** part/#asterisk santiago (n=santiago@208.195.215.158) |
| 16:14.03 | spiekey | i get the error: chan_capi.so: undefined symbol: |
| 16:14.04 | spiekey | ast_smoother_feed |
| 16:14.11 | spiekey | any idea why? |
| 16:14.46 | mishehu | because you didn't make the virgin sacrifice to the gods of unix. |
| 16:15.10 | jpm_SD | mishehu, good answer. |
| 16:15.12 | *** join/#asterisk _T3_ (n=rposada@53.228.uio.satnet.net) |
| 16:15.14 | mishehu | if you updated recently, perhaps you didn't clear out the old modules. |
| 16:15.50 | *** join/#asterisk nomazda (i=nyyankee@user-0c6tnqf.cable.mindspring.com) |
| 16:15.56 | *** join/#asterisk Skyhawk_1 (n=info@a62-216-22-13.adsl.cistron.nl) |
| 16:16.12 | *** join/#asterisk core-ix (n=ivo@2001:618:400:16fe:3:0:0:3) |
| 16:16.15 | Skyhawk_1 | stupid question .. does digium have any 4 x bri cards ? |
| 16:16.26 | zigman | nope |
| 16:16.31 | zigman | junghanns.net |
| 16:17.25 | spiekey | mishehu: huh? |
| 16:17.51 | *** join/#asterisk veto (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com) |
| 16:18.16 | jpm_SD | mishehu, to you hardware question, I personally would rather have 2 seperate single core systems for system redunacy. As it applies to asterisk, I'm not sure you'd see a tangle difference between using a dual core processor or a dual processor SMP. |
| 16:19.04 | jpm_SD | Then again I'm not an expert on SMP systems. |
| 16:19.28 | synthetiq | how about 2 dual proc systems =] |
| 16:19.42 | jpm_SD | synthetiq, Sounds good to me. |
| 16:20.02 | iDunno | 2 dual proc dual core systems ;) |
| 16:20.06 | iDunno | that's the way forwards. |
| 16:20.25 | synthetiq | i have 3 redundat dual procs |
| 16:20.30 | jpm_SD | Lets, just assume not everyone has buckets of cash lying around to build with. |
| 16:20.49 | synthetiq | wiritng load balancer now for it, got the iax2 part down |
| 16:21.11 | synthetiq | sip is the hard part |
| 16:21.36 | jpm_SD | IBM had a good presentation on build high avalibility systems. |
| 16:21.43 | jpm_SD | building rather. |
| 16:22.23 | paryl | in the dial string, is there a way to pause... to wait for the other end to pick up? |
| 16:22.34 | jpm_SD | Of course I just think they want to sell some bladecenters. |
| 16:22.46 | paryl | kinda like the comma in the dial string for a modem |
| 16:23.22 | jpm_SD | in 1.0.9 you can use 'w' for a 500ms pause. |
| 16:23.28 | *** join/#asterisk cpatry (n=grepmoo@65.39.228.5) |
| 16:24.00 | jpm_SD | I think that is still valid in 1.2 as well. but I've not had a chance to look at 1.2 yet. |
| 16:24.02 | paryl | so... 123wwww345? |
| 16:24.11 | jpm_SD | Yes |
| 16:24.15 | paryl | awesome, thanks! |
| 16:27.29 | yxa | if i scripted * to call 2 external numbers using 2 fxo, what is the command to bridge the 2 calls? |
| 16:28.08 | *** join/#asterisk |cleric| (n=dacleric@p54829744.dip0.t-ipconnect.de) |
| 16:29.05 | jpm_SD | yxa, there might be a way to do that with the dialplan, but I think you'd have to use an AGI and interface with the AMI to move the calls the way you want. |
| 16:29.52 | yxa | that sounds very complicated |
| 16:29.56 | spiekey | any idea how to solve the error error: chan_capi.so: undefined symbol: ast_smoother_feed ? |
| 16:31.54 | yxa | has anyone did something like that before? |
| 16:32.08 | *** join/#asterisk myiagy (n=myiagy@200.138.215.78) |
| 16:32.25 | jpm_SD | yxa, don't how complicated it would be, but you'd need to know perl or something along those lines. I imagine the logic would be easy. Conceptually it's simple .. unfortunately I don't know perl. |
| 16:32.46 | Inv_arp | spiekey: is this a fresh compilation? |
| 16:33.44 | yxa | jpm_SD there's also the polarity. ear-ear or ear-mouth |
| 16:33.54 | jpm_SD | yxa, you might do a search on call files... they are used to excute to call commands, they might be able to be able to do what you want.. |
| 16:35.40 | jpm_SD | yxa, I'd be looking to transfer the the two calls into a meetme conference room and use that to bridge the calls. |
| 16:36.28 | yxa | jpm_SD thats actually what i'm trying to avoid |
| 16:39.36 | *** join/#asterisk toddf (n=toddf@net-66-210-104-252.theshop.net) |
| 16:40.23 | *** join/#asterisk bronc (i=bronc@phalse.2600.COM) |
| 16:46.45 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 16:48.52 | *** join/#asterisk Caede (n=caede@sentry.zoom.com) |
| 16:49.43 | *** join/#asterisk mcf3782 (n=mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net) |
| 16:52.16 | paryl | i've tried opening ports 5600 and 10000-20000, but my SIP clients can't connect. if i set it in the DMZ, they work fine. which ports am i missing? |
| 16:52.53 | hypa7ia | might want to try 5060 instead of 5600 |
| 16:52.56 | hypa7ia | :-) |
| 16:54.14 | spiekey | Inv_arp: yes |
| 16:54.16 | *** join/#asterisk oej (n=Olle@apollo.webway.se) |
| 16:54.20 | spiekey | Inv_arp: asterisk compiled from source |
| 16:54.29 | paryl | my bad, yes, i meant 5060 |
| 16:54.37 | spiekey | Inv_arp: capi libs installed with apt on debian sarge |
| 16:54.48 | spiekey | Inv_arp: is my capi module too old? |
| 16:57.15 | *** join/#asterisk tainted_ (n=somewher@mail.k2usa.com) |
| 16:57.39 | tainted_ | has anyone here dealt with a Preston Garrison? (ptg123) |
| 16:58.22 | ender | paryl: you're opening UDP ports and not TCP ports right? |
| 16:58.31 | paryl | ender: right |
| 16:59.08 | paryl | what ports are actually *required*? the only common one among everything i've read is port 5060 |
| 16:59.18 | tainted_ | !seen ptg123 |
| 16:59.20 | ender | thats all I needed for registration. 5060 |
| 16:59.24 | tainted_ | ~seen ptg123 |
| 16:59.30 | jbot | ptg123 <~PTG123@ip68-106-24-139.ph.ph.cox.net> was last seen on IRC in channel #asterisk, 152d 10h 28m 59s ago, saying: '"replace your phone lines with cheaper voip with better features, oh shit wait, actually scratch that, you need to keep your phone line to have a phone line with us, so forget saving money, pay more :)"'. |
| 16:59.30 | hypa7ia | paryl: 5060 carries the ISP control traffic |
| 16:59.30 | ender | paryl: things get iffy though when you use nat |
| 16:59.37 | hypa7ia | the 10K-20K UDP ports carry the voice stream |
| 16:59.59 | paryl | everythign is udp though, no tcp, right? |
| 17:00.20 | hypa7ia | negatory |
| 17:00.23 | hypa7ia | 5060 is tcp |
| 17:00.33 | hypa7ia | i'm pretty sure |
| 17:02.11 | paryl | oh, d'oh... i think when i opend up 10k-20k before i left it on tcp |
| 17:02.16 | paryl | now it looks like it's working |
| 17:02.33 | paryl | and it looks like 5060 is udp only |
| 17:04.04 | *** join/#asterisk wunderkin (n=wunderki@12-219-165-109.client.mchsi.com) |
| 17:05.50 | clyrrad | I need to connect a remote IAX phone into my PBX for voice mail and VOIP access. Currently my PBX will register the IAX DID with the voip provider. And the Phone will connect to my Asterisk Box, how do I link the phone and voip service? |
| 17:05.53 | *** join/#asterisk loick (n=loick@APuteaux-151-1-30-110.w82-124.abo.wanadoo.fr) |
| 17:07.48 | mmlj4 | clyrrad: you treat the phone as an extension |
| 17:08.15 | mmlj4 | just like a phone on your LAN |
| 17:08.44 | Juggie | whats wrong with grep -R -H -a 'meh' *.c |
| 17:08.48 | Juggie | why wont it recurse |
| 17:08.59 | Juggie | it only seems to want to do the local dir, even though -R = recursive |
| 17:09.03 | Jzalae | because you have no dirs named *.c |
| 17:09.10 | iDunno | because it's only looking at *.c files/dirs |
| 17:09.16 | iDunno | try */*.c |
| 17:09.27 | Jzalae | erm, no |
| 17:09.39 | Jzalae | */*.c will only look at files one level down named *.c |
| 17:09.50 | iDunno | or use: find . -name '*.c' | xargs grep -H -a 'meh' |
| 17:09.56 | Jzalae | either grep the grep or use find and xargs |
| 17:10.05 | Jzalae | heh, like what he said :) |
| 17:10.37 | Beirdo | and change it to xargs -n 100 grep... |
| 17:10.43 | *** join/#asterisk myiagy (n=myiagy@200.138.215.78) |
| 17:10.43 | *** join/#asterisk fugitivo (n=ajf@209.13.241.249) |
| 17:10.49 | Beirdo | otherwise you may overflow the command line to grep |
| 17:11.17 | Jzalae | erm, i would think that xargs is smart enough not to do that |
| 17:11.19 | Beirdo | depending on how many files find gets |
| 17:11.20 | Beirdo | nope |
| 17:11.41 | Beirdo | xargs dumps them ALL on one line unless you tell it to do a max of BLAH per command |
| 17:11.46 | Beirdo | which is what -n 100 does |
| 17:11.50 | bronc | i luv asterisk |
| 17:11.52 | Beirdo | max 100 arguments |
| 17:12.08 | *** join/#asterisk damned (n=vpol@damned.vpol.org.ru) |
| 17:12.28 | iDunno | and use -print0 on find |
| 17:12.34 | iDunno | and -0 on xargs |
| 17:12.49 | iDunno | unless it's BSD, in which case, there's not an equiv that I know of ;) |
| 17:13.00 | Jzalae | and -- after the grep :) |
| 17:13.09 | Jzalae | er, after grep args |
| 17:13.19 | Jzalae | but all this is a bit overkill for interactive use |
| 17:14.27 | Jzalae | and modern bsd knows about -print0 and -0 |
| 17:16.07 | clyrrad | mmlj4, makes sense, how about making the md5 password? |
| 17:17.24 | Katty | mew. |
| 17:18.46 | *** join/#asterisk BladeRunner05 (n=feelme@adsl-ull-203-68.44-151.net24.it) |
| 17:21.43 | mmlj4 | clyrrad: you only need that when * interfaces registers with your VoIP provider... though I guess you could also set up md5 for the connecting between your remote phone and * (I haven't messed with any of that, sorry) |
| 17:21.51 | *** join/#asterisk Abbas (n=Abbas@gw3-fiberclient-148.brain.net.pk) |
| 17:22.21 | clyrrad | Yah i would like to use md5 for the remote IAX phone plantext passwords not very secure |
| 17:23.00 | Juggie | ping |
| 17:23.08 | Juggie | 5038 needs testers please! |
| 17:23.14 | Juggie | er, 5083 |
| 17:23.15 | Juggie | i mean |
| 17:23.36 | mmlj4 | i doubt you'll get more than one or two |
| 17:25.00 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.93) |
| 17:26.22 | *** join/#asterisk wunderkin (n=wunderki@VDSL-130-13-234-137.PHNX.QWEST.NET) |
| 17:27.40 | Nexis | anyone have a site that has the context for files going into spool/asterisk/outgoing |
| 17:32.38 | *** join/#asterisk jsmith (n=jsmith@38.119.177.48) |
| 17:32.46 | *** join/#asterisk darkskiez (n=darkskie@host86-138-169-183.range86-138.btcentralplus.com) |
| 17:33.13 | *** part/#asterisk logicalonline (n=Ken@209.242.52.25) |
| 17:34.36 | *** join/#asterisk |cleric| (n=dacleric@p5482B62E.dip0.t-ipconnect.de) |
| 17:35.11 | *** part/#asterisk jsmith (n=jsmith@38.119.177.48) |
| 17:35.25 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@bzq-82-80-162-23.red.bezeqint.net) |
| 17:37.02 | *** join/#asterisk WeezeyD (i=Weezey@vaio2.tsnetworks.ca) |
| 17:37.31 | WeezeyD | where can I get the Cisco 79XX firmware? |
| 17:37.38 | WeezeyD | err SIP firmware |
| 17:37.42 | *** join/#asterisk power1 (n=marktren@rndf-146-4-251.telkomadsl.co.za) |
| 17:42.06 | Juggie | cisco? |
| 17:42.20 | blitzrage | for $5! |
| 17:43.55 | *** join/#asterisk eivindtr (n=wingnut-@062016241059.customer.alfanett.no) |
| 17:44.02 | *** join/#asterisk zeedo (n=zeedo@80.68.92.188) |
| 17:44.43 | ManxPower | sometimes I hate tech. I bought an MP3 player at walmart yesterday and it's considered a "legacy device". |
| 17:45.08 | sigterm | doh |
| 17:45.30 | sigterm | that sucks |
| 17:46.20 | Katty | hmm. |
| 17:46.53 | tainted_ | why would u buy an mp3 player from walmart |
| 17:48.14 | ManxPower | Because it was cheap and it was there 8-) |
| 17:51.55 | ManxPower | WeezeyD, You can purchis the software from cisco |
| 17:52.04 | ManxPower | I think it's about $120. |
| 17:52.11 | ManxPower | This is one of the reasons we don't use cisco phones. |
| 17:52.21 | *** join/#asterisk CoffeeIV_ (n=CoffeeIV@mail.airlinksystems.com) |
| 17:53.36 | *** join/#asterisk CrazyYoss (n=nobody@adsl-69-236-44-222.dsl.pltn13.pacbell.net) |
| 17:54.18 | *** join/#asterisk konfuzed (n=konfuzed@H129.C72.B0.tor.eicat.ca) |
| 17:54.30 | CrazyYoss | Ive been trying to find a decent Authorize.net reseller, looking through the pages I cant help but get a cold feeling in my stomach that most of these companies are out to screw you. Anyone have a good experience with an authorize.net reseller? |
| 17:55.25 | Dr_Ray | why not just use paypal? |
| 17:55.48 | Dr_Ray | if you can't buy merchant account stuff on your own, you are going to get bent over |
| 17:56.07 | CrazyYoss | Dr_Ray: Ive read of some horror stories with paypal, freezing acounts, |
| 17:56.29 | Dr_Ray | Crazy - that's not a problem if you follow their rules |
| 17:56.43 | CrazyYoss | Dr_Ray: im asking for personal experiences here Im not asking for you to hold my hand |
| 17:56.48 | *** join/#asterisk diclophis (n=diclophi@adsl-69-238-124-226.dsl.pltn13.pacbell.net) |
| 17:56.53 | diclophis | hello |
| 17:56.59 | diclophis | anyone using realtime extensions? |
| 17:57.00 | Dr_Ray | any merchant account will freeze your account with fraud |
| 17:57.14 | diclophis | and or know howto support multiple contexts using realtime extensions? |
| 17:57.40 | Dr_Ray | Crazy - I meant, authorize net resellers exist to fill a market, for people who can't get merchant accounts |
| 17:58.01 | Dr_Ray | Crazy - it's going to be predatory |
| 17:58.19 | CrazyYoss | Dr_Ray: ahhh...but you cant deal directly with Authorize.net, unless you ARE a reseller |
| 17:58.53 | Dr_Ray | Crazy - we are brick and mortar business with a merchant account and we get buttraped on credit card transactions |
| 17:59.24 | diclophis | no realtime users eh... |
| 18:00.25 | CrazyYoss | Dr_Ray: Brick and mortar is better than ecommerce...usually the discount rate is 1% lower plus there are a ton of gateway fees |
| 18:00.49 | Dr_Ray | Crazy - we pay 3% on transactions, $1000 pays them $30 |
| 18:00.52 | *** join/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1) |
| 18:01.05 | *** join/#asterisk Xen^ (i=linux@202.5.131.159) |
| 18:01.33 | CrazyYoss | Dr_Ray: Ouch....are you deemed high risk? At least most the resellers Ive been looking at only charge 1.3 for swiped cards |
| 18:01.45 | *** join/#asterisk jimmy_deanPB (n=jhodapp@72.244.232.226) |
| 18:01.58 | Dr_Ray | Crazy - we are a hotel |
| 18:02.02 | *** part/#asterisk greekman (n=alex@host254.209.113.199.conversent.net) |
| 18:02.08 | Dr_Ray | Crazy - we see some charge backs |
| 18:02.46 | power1 | hey all, just added another fxo and 1 fxs to my digium card on an asterisk at home box. Does any1 know how to assign a extension to the analog phone plugged into the fxs module? |
| 18:03.08 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 18:03.13 | Dr_Ray | power - I know how to do that with asterisk |
| 18:03.17 | CrazyYoss | Dr_Ray: bummer, well thank you for your time. Question though: do you ever get guys coming in and spend 500 bucks in one night on sex numbers? |
| 18:03.18 | Dr_Ray | not asterisk@ home |
| 18:03.26 | *** part/#asterisk mcf3782 (n=mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net) |
| 18:03.41 | Dr_Ray | Crazy - no we block those, but we did have some 809 phone sex calls |
| 18:03.55 | power1 | Dr_Ray, how do i do it with asterisk? |
| 18:04.08 | Dr_Ray | power1 / extension.conf |
| 18:05.22 | *** part/#asterisk CrazyYoss (n=nobody@adsl-69-236-44-222.dsl.pltn13.pacbell.net) |
| 18:06.13 | *** join/#asterisk Xen^ (i=linux@202.5.131.149) |
| 18:06.48 | Xen^ | hello all can some one help me with macro ? |
| 18:06.56 | power1 | Dr_Ray, what do I add to the file. |
| 18:07.07 | Xen^ | http://pastebin.ca/25987 |
| 18:07.09 | blitzrage | Xen^: http://www.asteriskdocs.org <-- Read the book online |
| 18:07.18 | jpm_SD | Power1, after you do a 'genzaptelconf -s' can can just add a ZAP extension with AMP |
| 18:07.28 | power1 | Also any opinions on asterisk@home vs Asterisk....and what distro to put asterisk on top of? |
| 18:08.04 | Xen^ | i am trying to create a macro for local extensions like if i have 1000 of extensions then it will be hard to add them in extensions.conf :) |
| 18:08.13 | Xen^ | so i am trying to write script which will do this :) |
| 18:08.19 | Xen^ | http://pastebin.ca/25987 |
| 18:08.41 | power1 | jpm_SD, thanks |
| 18:09.14 | Xen^ | can any one :( |
| 18:09.22 | power1 | jpm_SD, ive allready done a " genzaptelconf -s -d " does that count? |
| 18:10.40 | jpm_SD | power1, yes. |
| 18:11.42 | power1 | jpm_SD, so when im in amp and i choose add extension there is no option to associate the extension number with the fxs channel. |
| 18:11.47 | Xen^ | any one please check this http://pastebin.ca/25987 |
| 18:12.05 | jpm_SD | should be SIP / IAX / ZAP |
| 18:12.20 | jpm_SD | you'd want to use the ZAP channel. |
| 18:13.03 | morale | how do i pull the cvs checkin log off of cvs.digium.com ? |
| 18:13.07 | power1 | jpm_SD, are you talking about the "trunk" section and not the extension section? |
| 18:13.20 | jpm_SD | power1, nope.. under extensions. |
| 18:13.33 | jpm_SD | power1, "phone protocol" |
| 18:13.52 | CoffeeIV_ | I'm trying to put my cdr records in a mysql db. As far as I can see everything (conf files and mysql) is set up correctly (same as another * server I have that works) -- but at *CLI> I see "cdr_addon_mysql.c: cdr_mysql: cannot connect to database server localhost" |
| 18:13.57 | power1 | jpm_SD, hmmmm lemme go and look, thanks for helping...much appreciated. |
| 18:14.05 | CoffeeIV_ | anything else I should check ? |
| 18:14.07 | jpm_SD | power1, looking at mine right now I can chose "SIP" "IAX" "ZAP" |
| 18:14.45 | power1 | jpm_SD, thanks..u r right..just never even saw it the 1st time <grin> <blush> |
| 18:16.53 | power1 | jpm_SD, thanks.....works 100%.....u up for another question? |
| 18:17.41 | jpm_SD | power1, I'm here... just a bit busy with work.. one minute. |
| 18:20.12 | power1 | ok.. |
| 18:20.56 | jpm_SD | ok whats the question power1 |
| 18:22.41 | power1 | ummm, ive got 2 fxo modules on my card, when both outgoing lines are available i want a certain channel to be used 1st , but it keeps using the 1st fxo module as the outgoing channel and only uses the second fxo if the 1st is busy ..how do i change this? |
| 18:24.17 | jpm_SD | power1, couple ways to skin that cat... |
| 18:24.36 | power1 | jpm_SD, yeah....? |
| 18:24.48 | *** join/#asterisk dinkdinkdink (n=d@pool-68-238-147-186.dllstx.fios.verizon.net) |
| 18:24.57 | snitt | drinkdrinkdrink |
| 18:25.23 | jpm_SD | power1, you can either make two new ZAP trunks ZAP/1 ZAP/2 (or whatever it valid for you) and then assign them in the order you want in your outbound routing. |
| 18:25.24 | brad_mssw | err, anyone know if iaxtel is down ? |
| 18:25.30 | *** join/#asterisk zeedo (n=zeedo@80.68.92.188) |
| 18:25.48 | jpm_SD | power1, or you can edit the defualt ZAP/g0 to be ZAP/G0 |
| 18:25.52 | *** join/#asterisk Xen^ (n=linux@202.63.195.81) |
| 18:26.22 | power1 | jpm_SD, or i suppose i can just swop the phone jacks at the back of the card..that would also work. |
| 18:26.34 | diclophis | anyone know howto configure multiuple contexts with realtime mysql configuration/ |
| 18:26.35 | diclophis | ? |
| 18:26.53 | ian_k | brad_mssw: yes |
| 18:27.09 | Nugget | nutty linux users. can't spell "how to" and always want to use mysql. :) |
| 18:27.15 | jpm_SD | power1, hehe.. yes but that "bad form" :P |
| 18:27.19 | ian_k | brad_mssw: use fwd or some other one. There is no ETA on the return of iaxtel |
| 18:27.29 | power1 | <grin> quick solutions often are! |
| 18:27.34 | brad_mssw | ian_k: ahh, interesting |
| 18:27.43 | brad_mssw | ian_k: yeah, just tried to sign up, and can't register ;) |
| 18:28.07 | power1 | jpm_SD, sory im kinda a newb just installed aterisk yesterday......could you give me a quick overview of what a trunk is? |
| 18:28.31 | ian_k | brad_mssw: www.freeworlddialup.com |
| 18:28.39 | *** join/#asterisk folsson (n=filip@lund-meje-sr0-vl101-249.perspektivbredband.net) |
| 18:28.43 | brad_mssw | ian_k: yeah, already looked at that one too :) |
| 18:29.08 | jpm_SD | power1, well.. without a lot of explaination there is a section in A@H named "Trunks" |
| 18:29.21 | jpm_SD | power1, Might look in there... |
| 18:29.50 | power1 | jpm_SD, ok..I think I need to do some reading.... |
| 18:30.15 | power1 | but thanks for the help.......what distro do u usually run asterisk on top of....ie: whats worked best for you? |
| 18:33.04 | ian_k | power1: the hardware platform matters much more than the distro |
| 18:33.11 | ian_k | any distro will run it |
| 18:33.30 | jpm_SD | power1, gentoo.. but ian_k is correct ... hardware is more important. |
| 18:33.41 | power1 | ian_k, care to elaborate on the hardware platform? u mean like intel chipset boards etc? |
| 18:34.49 | power1 | jpm_SD, ian_k ok so if u were going to put together a asterisk box for a 50 user ip phone network.....what would u use? |
| 18:34.49 | ian_k | power1: if you have digium hardware, compatibility is vital. Otherwise, you need a system that can devote the resources and low-latency to supoprt a realtime audio system |
| 18:35.04 | ian_k | what codec? |
| 18:35.24 | ian_k | power1: it makes a big difference if you have 50 sip phones with g.729 vs ulaw. |
| 18:35.25 | jpm_SD | Are you going to be transcoding a lot? |
| 18:35.29 | ender | power1: like PC, PPC, Sparc, RISC, or subplatforms, like ia32, ia64, x86_64, or sparc32, sparc64, or ppc, ppc64 |
| 18:35.30 | power1 | what , hardware platform, motherboard, ram, cpu etc? |
| 18:35.47 | jpm_SD | 50 user, I'd build more than one server. |
| 18:35.54 | jpm_SD | and load balance it. |
| 18:35.57 | ian_k | power1: with, or without zaptel hardware? |
| 18:36.00 | ender | for just 50?! |
| 18:36.05 | ender | that seems WAY overkill. |
| 18:36.12 | jpm_SD | I would yes |
| 18:36.18 | power1 | ian_k, with zaptel |
| 18:36.21 | ender | single server w/ SMP and ram, plus redundant storage seems PLENTY for me. |
| 18:36.25 | jpm_SD | Reduntacy ... |
| 18:36.37 | ender | jpm_SD: we have a cold spare. |
| 18:36.40 | jpm_SD | if one server craps out .. I don't want my call center down. |
| 18:36.44 | ian_k | power1: T1s or analog lines? |
| 18:36.49 | ender | well, I'd call it a 'warm' spare. |
| 18:36.57 | power1 | ian_k, analog |
| 18:37.49 | jpm_SD | power1, channel bank or are you running like 8 pots lines? |
| 18:37.54 | ian_k | power1: in that case, you won't have compatibility problems. Just get a decent (~2GHz) proc, and disable all the useless crap (like usb controllers, etc) on the box. |
| 18:38.25 | power1 | jpm_SD, 7 pots lines |
| 18:39.02 | ian_k | power1: if you will be transcoding, you'll want to up your processor speed. |
| 18:39.52 | power1 | hmmm, and handset wise what do u reccomend? |
| 18:40.02 | jpm_SD | Polycom 501. |
| 18:40.28 | jpm_SD | Even the 301s are nice... but the 501 is worth the extra. |
| 18:40.51 | power1 | jpm_SD, what about grandstreams? |
| 18:41.10 | jpm_SD | power1, people have mixed experiances with them. |
| 18:41.28 | power1 | ian_k, when would i need transcoding? |
| 18:41.31 | Katty | power1: we use polycom 500s here (= |
| 18:41.31 | jpm_SD | power1, some people swear by them ... others have them crash on them every 5 minutes. |
| 18:42.12 | ender | second recommentation for Polycoms. |
| 18:42.21 | ender | We have 301s and 501s. Nary a problem. |
| 18:42.50 | Katty | polycoms are quite nice, though slightly on the pricey side. |
| 18:42.59 | jpm_SD | power1, transcoding happens anytime your codec changes ... if all your internel use high bandwidth codecs like ulaw/alaw and you make a sip call to a ITSP with GSM .. thats a transcode. |
| 18:43.00 | Katty | and they support ulaw 8 mono .wav format for ringtones |
| 18:43.07 | Katty | how neat is that! (ok, so that's just a fluffy feature) |
| 18:43.18 | power1 | at the moment ive got a couple of phoned from a company called "act" very high quality and works very well. |
| 18:44.01 | power1 | jpm_SD, aha..thanks , so its like on the fly conversion. |
| 18:44.04 | jpm_SD | power1, if you have remote users chances are you'll be transcoding those calls too... but that just depends. |
| 18:44.11 | jpm_SD | power1, yes. |
| 18:45.12 | power1 | jpm_SD, hmmm, I need to do some reading on codecs? |
| 18:45.51 | jpm_SD | power1, 50 calls transcoding to ilbc on a conferance bridge will bring most any rig to it's knees. Not that you'd be in the situation... but it's something to consider. |
| 18:46.01 | power1 | i think these phones priorotize g.711 u-law |
| 18:48.09 | jpm_SD | power1, mainly you have to think about want codec will using with your VoIP provider. I would think that is where most of your transcoding will occure and then how many concurent calls you expect your system to be handling. |
| 18:48.45 | power1 | is ther a checklist to follow. to confirm that a system is performing properly...how can i benchmark audio quality etc...is there a procedure to follow to optomize asterisk? |
| 18:49.00 | jpm_SD | power1, I wish. |
| 18:49.31 | power1 | jpm_SD, so if a company is using only internal call and outgoing pots..then is it best to just stick to ulaw? |
| 18:49.49 | jpm_SD | power1, yup. no reason not too. |
| 18:49.52 | ender | power1: it's best to stick w/ the highest quality your phones and Asterisk natively support. |
| 18:50.05 | ender | power1: provided that you have enough local bandwidth for the userbase. |
| 18:50.45 | power1 | ender, hmmm, does having a switch that supports voip make a difference? |
| 18:51.06 | jpm_SD | having a switch that does QOS does. |
| 18:51.58 | power1 | jpm_SD, why do you use it to optomize voice traffic......how .....via port number and udp? |
| 18:53.17 | jpm_SD | well.. the polycoms I use will add QoS tags. Then you use the switch to give those tags higher priority. |
| 18:54.29 | jpm_SD | I think a lot of the SIP endpoints out there do some type of QoS or ToS tagging. |
| 18:55.15 | power1 | what field in the phone interface do i need to look for for ? |
| 18:55.16 | ender | power1: how a switch 'supports' one form of traffic is a mystery to me. Traffic is traffic. THats like saying your switch doesn't support http. |
| 18:55.24 | clyrrad | How does the IAX Registration and MD5 work? Is there some link between username and secret? I get an error about host failed MD5 Authentication. Can some clarify this for me? |
| 18:55.24 | *** join/#asterisk shido6 (n=curtis@d221-68-210.commercial.cgocable.net) |
| 18:56.13 | power1 | ender, layer 3 switches u can tell to priorotize certain types of traffic etc... |
| 18:57.07 | power1 | jpm_SD, ive got a field on these phones under qos that says " Voice TOS " with a numerical value.....does that sound fmiliar? |
| 18:58.35 | ender | power1: sure, thats completely different than 'supporting' a traffic type. |
| 18:58.51 | power1 | ender, yeah agreed! |
| 18:59.04 | clyrrad | any takers? |
| 19:03.20 | *** join/#asterisk jtodd (n=jtodd@ti.fox-den.com) |
| 19:03.25 | Delta34 | sup all, my sip show peers shows bad status times for cisco 7960 sip phones and all phones are in internal lan, anybody know how to improve those times |
| 19:03.29 | *** join/#asterisk Morpheus68 (n=chatzill@a20030.upc-a.chello.nl) |
| 19:03.39 | ManxPower | Delta34, define "bad status" |
| 19:03.49 | Morpheus68 | hi all |
| 19:03.56 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 19:04.31 | Morpheus68 | how do i give a busy signal to an external caller when an intarnal phone is busy |
| 19:04.44 | Delta34 | 70ms using version 7.4 cisco sip version, 140ms using version 7.5 cisco sip version |
| 19:04.50 | Morpheus68 | intarnal=internal |
| 19:04.57 | Delta34 | xten clients show like 3-5ms |
| 19:05.18 | Delta34 | is this normal? |
| 19:05.20 | ender | Morpheus68: that would be priorty +100 |
| 19:05.27 | clyrrad | Can anyone tell me how the MD5 authentication works for IAX? I am trying to get a remote IAX phone connected to my Asterisk Box, and im getting MD5 Authenticaion error |
| 19:06.02 | ender | Morpheus68: so if it's exten => foo,1,Dial(xxx) then you need exten => foo,102,BUSY although I don't know off the top of my head what application does the busy signal. You'll have to find that in the wiki |
| 19:06.44 | Morpheus68 | i tested that n+101 bud it did not work |
| 19:06.51 | Morpheus68 | so i'm going to test it again |
| 19:06.58 | *** join/#asterisk dudes (n=dudes@12-215-32-62.client.mchsi.com) |
| 19:07.08 | ender | Morpheus68: works like a charm here, we send them to voicemail w/ a b prefix for busy |
| 19:07.48 | Delta34 | any thoughts on how to make the sip status times better for my cisco phones? |
| 19:08.49 | *** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 19:09.05 | ender | Morpheus68: in my conf I have exten => _5XXX,2,Dial(SIP/${EXTEN},20); and exten => _5XXX,103,Voicemail(b${EXTEN}); |
| 19:14.42 | skyen | with the generic sipfriends from mysql-feature, i shouldn't be able to list peers using 'show sip peers' in the CLI, right? |
| 19:15.04 | skyen | my question is; would i be able to do this with some of the other database-bindings? |
| 19:15.20 | skyen | like realtime, ast-data or whatever they're called |
| 19:15.47 | fugitivo | mysql is evil |
| 19:17.05 | *** join/#asterisk docE (n=docE@66.237.242.41.ptr.us.xo.net) |
| 19:17.33 | *** join/#asterisk Meaty (n=cp_simbu@office.abi.ca) |
| 19:17.35 | docE | Oct 19 15:19:02 WARNING[21046]: channel.c:2435 ast_channel_make_compatible: No path to translate from H323/18139015182-289e(6) to SIP/8135696291-c1cd(256) |
| 19:17.35 | docE | Anyone know what this means? |
| 19:18.41 | docE | ohh nevermind.. |
| 19:18.47 | docE | damn codecs.. ARGH! |
| 19:19.23 | *** join/#asterisk fulgas (n=fulgas@a81-84-116-219.cpe.netcabo.pt) |
| 19:19.44 | skyen | fulgas: good answer ;) |
| 19:20.01 | fulgas | hey |
| 19:20.05 | fulgas | skyen ?? |
| 19:20.34 | skyen | oh, nickcompletion ;) |
| 19:20.41 | skyen | fugitivo: good answer ;) |
| 19:20.50 | fulgas | :) |
| 19:21.24 | asterisk99 | anyone using Asterisk on Gentoo ? |
| 19:21.30 | fugitivo | i do |
| 19:22.07 | asterisk99 | fugitivo: does ztcfg run ever bootup for u? |
| 19:22.31 | asterisk99 | fugitivo: "ever" = "every" |
| 19:22.36 | fugitivo | yes |
| 19:22.44 | fugitivo | rc-update add zaptel default |
| 19:22.51 | CoffeeIV_ | I installed asterisk from scratch (source is a month or so old, from CVS HEAD). I did everything I thought I was supposed to do to receive faxes; *CLI> show application rxfax works, and faxdetect is set in zapata.conf; but when I get a fax, it never starts receiving the fax, and the other end eventually times out. |
| 19:23.41 | asterisk99 | fugitivo: Intersting... that's not in any docs that I found |
| 19:23.41 | CoffeeIV_ | the log shows it hanging in rxfax until the timeout |
| 19:23.53 | asterisk99 | fugitivo: is that zaptel or ztcfg??? of does it make a diff? |
| 19:24.14 | fugitivo | asterisk99: zaptel is the init script in your /etc/init.d, it runs ztcfg and some other stuff |
| 19:24.49 | asterisk99 | fugitivo: Okeee. I added it and now for reboot to see if it works |
| 19:24.54 | fugitivo | ok |
| 19:27.06 | *** join/#asterisk HuNTER_SC (i=Junior@201-3-196-66.fnsce7004.dsl.brasiltelecom.net.br) |
| 19:27.14 | HuNTER_SC | ola alguem fala portugues? |
| 19:27.21 | fugitivo | eu falo |
| 19:27.43 | HuNTER_SC | fugitivo -> podes me da um help com *? |
| 19:27.56 | HuNTER_SC | fugitivo -> to precisando interligar 2 * |
| 19:28.02 | HuNTER_SC | mais nao sei como faz |
| 19:28.15 | HuNTER_SC | me falaram q da pra fazer sem registrar |
| 19:29.01 | asterisk99 | fugitivo: Aye carumba... that didn't work ... or ZTCFG did run but blew up during reboot --- any idea as to where the log is? |
| 19:29.25 | fugitivo | HuNTER_SC: IAX2 |
| 19:29.27 | dudes | I setup zaptel in local |
| 19:29.40 | dudes | works better than using the bootscript (at least from my experience) |
| 19:29.45 | fugitivo | asterisk99: what error are you getting? |
| 19:30.00 | HuNTER_SC | fugitivo mais tipo se eu tirar tirar o telefone do gancho e digitar um numero q ta la no outro * vai tocar? |
| 19:30.33 | HuNTER_SC | fugitivo digito 1234 q eh um ht q ta autenticado no outro * |
| 19:30.43 | *** join/#asterisk Meaty (n=cp_simbu@office.abi.ca) |
| 19:31.04 | fugitivo | HuNTER_SC: vc precisa interligar e configurar extensions.conf |
| 19:31.05 | asterisk99 | fugitivo: chanzap.sop failed ---- if I manually run ztcfg -vv and then start Asterisk, it works fine (but not automatically after reboot) |
| 19:31.30 | asterisk99 | fugitivo: make that chanzap.so |
| 19:31.49 | HuNTER_SC | fugitivo so vlw |
| 19:32.40 | asterisk99 | fugitivo: Actually, that's the last error msg... there are a bunch... all complaining about channel 1 (my Digium card) |
| 19:33.26 | Morpheus68 | ok, solved it |
| 19:34.10 | *** join/#asterisk paryl (n=paryl@209.236.78.59) |
| 19:34.13 | Morpheus68 | i have a php script that looksup ldap contact info, does some loging, and sends a jabber msg |
| 19:34.20 | Morpheus68 | and the ldap server was hanging |
| 19:34.36 | Katty | anyone framilier with photons, light bending, black holes, and a few of einstein's theories? |
| 19:34.46 | Katty | i have questions. |
| 19:34.47 | paryl | i keep getting "Removed default indication country 'us'" whenever i reload... what does that mean? |
| 19:35.03 | fugitivo | Katty: i have a photon light |
| 19:35.04 | Morpheus68 | so the whole call does not camethroug |
| 19:35.10 | Katty | fugitivo: light is made of photons. |
| 19:35.11 | paryl | Katty: yes :) |
| 19:35.21 | Katty | fugitivo: no surprise there. |
| 19:35.25 | Katty | paryl: excellent. how much do you know? |
| 19:35.53 | fugitivo | Katty: http://www.thinkgeek.com/gadgets/lights/38d4/ |
| 19:36.36 | jpm_SD | asterisk99, if I were guessing (and I am) it sounds like ztcfg is not getting run by the init script. I'd look at /etc/init.d/zapwhatever and see what's in it. again.. just guessing. |
| 19:37.23 | *** join/#asterisk cjk (n=cjk@80.92.64.103) |
| 19:37.39 | cjk | hello, anyone here who got the playback application working in features.conf ? |
| 19:41.47 | jarrod | im trying to incorporate mandatory authentication into SER before forwarding to asterisk.. anyone have any examples |
| 19:42.32 | *** join/#asterisk wunderkin (n=wunderki@VDSL-130-13-234-137.PHNX.QWEST.NET) |
| 19:45.35 | *** join/#asterisk wunderkin (n=wunderki@12-219-165-109.client.mchsi.com) |
| 19:46.41 | docE | What is the correct syntax for ooh323 addon for HEAD? |
| 19:46.53 | *** join/#asterisk rob314 (n=rob314@207.58.194.2) |
| 19:47.30 | docE | I am doing DIAL(OOH323/PEER/# |
| 19:47.44 | docE | I am doing DIAL(OOH323/PEER/${EXTEN},23) |
| 19:47.49 | docE | is this right? |
| 19:49.57 | *** join/#asterisk viLeR (i=1000@66.128.47.232) |
| 19:50.10 | jpm_SD | DocE I'm not familiar with 323 .. but I would think it would be technolofy@host/extension,options, |
| 19:50.50 | docE | acutally.. its OOH323/${EXTEN}@HOST,Options |
| 19:50.57 | *** join/#asterisk clive- (n=pirch@ndn-165-140-56.telkomadsl.co.za) |
| 19:50.58 | docE | I just did it.. Im such a dumb ass sometimes.. |
| 19:51.08 | jpm_SD | Interesting. |
| 19:51.23 | jpm_SD | is that specific to ooh323? |
| 19:51.36 | docE | I am building a 80 seat PBX for my company with H323 as our primary outbound |
| 19:51.39 | asterisk99 | jpm_SD: I am looking at /etc/init.d/zaptel... according to zaptel docs, I'm supposed to comment out if $system = redhat.... I can't that line at all |
| 19:51.41 | docE | I believe so |
| 19:53.09 | PupenoL | Does anybody know about packages for asterisk 1.2.0... for Debian Sarge ? |
| 19:53.48 | jpm_SD | asterisk99, unfortunately I can't answer the question for your specific distro... I wrote my own rc.local script to luanch asterisk... |
| 19:54.35 | asterisk99 | jpm_SD: is there a log I could look at that might contain errors during the boot-up process? |
| 19:55.13 | jpm_SD | dmesg ? |
| 19:55.32 | jpm_SD | <PROTECTED> |
| 19:55.44 | jpm_SD | those might shed some light. |
| 19:59.56 | infinity1 | PupenoL: ufortunately ...packages don't really exist. its a real bummer |
| 20:00.11 | PupenoL | infinity1: ok, I'll kinda fix that. |
| 20:00.36 | infinity1 | PupenoL: it migt be possible to convert the voip project on one of the deb servers |
| 20:00.42 | infinity1 | let me find a link |
| 20:02.45 | infinity1 | PupenoL: http://svn.debian.org/wsvn/pkg-voip/ ...i think they have a script to convert to packages. i haven't tried it. |
| 20:02.54 | infinity1 | PupenoL: let me know if you get anywhere. |
| 20:04.31 | morale | can someone help me figure out why i cannot dial 1800 numbers with my voip provider, they are not getting back to me and it is reporting "503 Service Unavailable" |
| 20:06.23 | infinity1 | morale: if you extension.conf is correct and you can dial other numbres, it must be them |
| 20:06.46 | morale | yeah i can dial other numbers except for 18xx style numbers |
| 20:07.22 | vader-wrk | quick survey, what linux os have you guys had the most luck with asterisk? |
| 20:07.31 | PupenoL | infinity1: is that repository available as anonymous with svn ? |
| 20:07.55 | PupenoL | vader-wrk: so far, gentoo. |
| 20:08.41 | jpm_SD | Gentoo here too. |
| 20:08.43 | clive- | vader redhat is very popular |
| 20:08.51 | morale | would "503: Service Unavailable" be returned from my VoIP uplink/provider? |
| 20:08.52 | jpm_SD | and CentOS 3.5 |
| 20:09.05 | *** join/#asterisk wolfson (n=hehe@usr-kdh-208-6-58-26.beachlink.com) |
| 20:09.08 | jpm_SD | morale ya |
| 20:09.26 | PupenoL | I'd consider Debian as well. |
| 20:09.30 | morale | then my voip provider must have something messed up |
| 20:09.56 | morale | exten => _1NXXNXXXXXX,1,Dial,IAX2/russellmcconnachie@FreeWorldTel/${EXTEN} - as straightforward as it comes |
| 20:10.28 | jpm_SD | It's no surprise that FWD would be having issues. |
| 20:11.21 | morale | i can't dial any numbers in the 403462xxxx prefix with FWD they don't have a route for that prefix or something |
| 20:12.30 | infinity1 | PupenoL: i'm not sure. i see they have a readme. |
| 20:15.15 | clyrrad | is there a way to parse your iax.conf file for syntax errors? |
| 20:16.44 | morale | what is a decent voip provider which provides a DID in alberta? or the 403 area code? |
| 20:18.40 | *** part/#asterisk paryl (n=paryl@209.236.78.59) |
| 20:19.08 | WeezeyD | how come when I do sip show peers my new 7940s show unspecified? |
| 20:19.55 | *** join/#asterisk nain (n=nain@202.154.245.18) |
| 20:20.18 | nain | Hello Everybody |
| 20:21.17 | *** part/#asterisk gabb0 (n=gabb0@131.202.90.23) |
| 20:23.18 | *** join/#asterisk nwhit (n=chatzill@wsip-24-234-120-72.lv.lv.cox.net) |
| 20:23.27 | nwhit | anyone here familiar with vicidial? |
| 20:24.28 | dudes | Your best bet is to ask a question |
| 20:25.39 | nwhit | i can get agents to login to vicidial, it has a lead list, but it does not send any calls to the agents |
| 20:26.04 | nain | can any one tell me how to use cvs for particular date specific version of asterisk download ? |
| 20:26.04 | nwhit | anything i can check out why it isn't calling out |
| 20:26.13 | Nexis | is it possable to have a outgoing call hang up after x seconds? |
| 20:26.43 | nain | as i like to download http://www.asterisk.org (CVS v1-0, 2005-09-08) so how to using cvs ? |
| 20:26.51 | jpm_SD | WeezeyD, I think "unspecified" indicates that the phones are not registered. |
| 20:27.58 | jpm_SD | Nexis, yes.. |
| 20:28.09 | jpm_SD | Nexis, with the L option |
| 20:28.36 | jpm_SD | Nexis, Dial(SIP/foo|20|L(2100000)) would hang up the call after 2100 seconds. |
| 20:29.27 | nain | Any one can help me with how to use cvs for asterisk http://www.asterisk.org (CVS v1-0, 2005-09-08) |
| 20:29.29 | *** join/#asterisk _santiago_ (n=santiago@208.195.215.158) |
| 20:30.08 | jpm_SD | Nexis, at the moment there is a problem with the syspoll() call and any timeout 260000 and over will fail to disconnect. |
| 20:30.14 | *** join/#asterisk nagl (n=nagl@213.235.241.6) |
| 20:32.01 | jpm_SD | 2600000 and over .. I forgot a 0.. |
| 20:33.01 | *** join/#asterisk wunderkin (n=wunderki@VDSL-130-13-234-137.PHNX.QWEST.NET) |
| 20:33.08 | *** part/#asterisk _santiago_ (n=santiago@208.195.215.158) |
| 20:35.53 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net) |
| 20:38.54 | infinity1 | tzafrir_laptop: didn't you say something about debian packages once? |
| 20:39.30 | *** join/#asterisk wunderkin (n=wunderki@12-219-165-109.client.mchsi.com) |
| 20:40.31 | PupenoL | nain: your best bet might be reading the cvs manual or some tutorial on cvs. |
| 20:40.47 | *** join/#asterisk Error_X (n=Error_X@80-86-211-195.ipc21.adsl.hesbynett.no) |
| 20:40.52 | PupenoL | nain: checking out of a branch at a given date shouldn't be too hard to find. |
| 20:41.28 | Error_X | Do I need ztdummy or a zaptel card for hearing the receptionist? |
| 20:41.30 | spiekey | has someone got * between telco and a PBX? |
| 20:41.46 | spiekey | how do i tell asterisk to forward all the calls to my PBX? |
| 20:43.36 | jpm_SD | spiekey, how do you have * connected to your your PBX? |
| 20:46.41 | spiekey | jpm_SD: with an Fritz! ISDN card |
| 20:47.49 | vader-wrk | are there any free windows based softphones you guys recommened? |
| 20:48.21 | jpm_SD | spiekey, hrmm.. I've never configured an ISDN card in Asterisk but if such a thing can be done you'd just send the calls you want to go to the pbx down that trunk. |
| 20:48.50 | Nugget | einfo "This ebuild now uses a heavily stripped down version of your CFLAGS" |
| 20:48.50 | Nugget | einfo "Don't complain because your -momfg-fast-speed CFLAG is being stripped" |
| 20:48.52 | Nugget | heh |
| 20:49.04 | spiekey | jpm_SD: i do that by configuring my zapta.conf file, right? |
| 20:49.23 | *** join/#asterisk RoyK (n=roy@55.80-202-161.nextgentel.com) |
| 20:50.06 | *** join/#asterisk stevek (n=stevek@slim-eth0.horizonlive.net) |
| 20:50.36 | jpm_SD | vader-wrk, sokol-associates having an IAX softphone that's suppose to be good. |
| 20:50.44 | *** join/#asterisk ManxPower (n=ewieling@adsl-67-65-233-194.dsl.lgvwtx.swbell.net) |
| 20:51.04 | jpm_SD | spiekey, honestly, I don't know the answer to that, is the ISDN card zaptel compatible? |
| 20:51.08 | malverian[work] | WTF?? |
| 20:51.11 | malverian[work] | Is Asterisk HEAD broken? |
| 20:51.29 | jpm_SD | malverian[work], possibly. |
| 20:51.32 | malverian[work] | Playback() and Background() no longer appear to work for me... |
| 20:51.35 | Nugget | If it is, why be shocked? That's the nature of development branches. |
| 20:51.37 | malverian[work] | Among other things... |
| 20:51.38 | ManxPower | malcolmd, CVS-HEAD is like my 85 yr old Aunt Betty - It has it's good days and it's bad days. |
| 20:51.54 | jpm_SD | ManxPower, haha |
| 20:52.05 | diclophis | is it possible to have a switch stateent in extensions.conf be for ALL contexts? |
| 20:52.16 | ManxPower | malverian[work], CVS-HEAD *is* the development code for Asterisk. |
| 20:52.37 | malverian[work] | ManxPower, I know.. it's also the only code that has many features I require. |
| 20:52.52 | malverian[work] | I suppose I could use the beta1 tag. |
| 20:52.54 | ManxPower | malverian[work], Why did you update, if you had a working system/ |
| 20:53.11 | ManxPower | It's like running Windows Vista on a production server. |
| 20:53.14 | malverian[work] | ManxPower, I don't have a perfectly working system. I get segfaults in chan_sip frequently. |
| 20:53.36 | malverian[work] | And am trying to determine the cause.. so I updated to latest CVS to debug the problem. |
| 20:53.51 | ManxPower | malverian[work], you, of course, are subscribed to the asterisk-cvs mailing list so you know what each change to the source code is, right? |
| 20:54.17 | Nexis | jpm_SD, ok, that helps some, next question, any idea how i would put those dial options into a file, that would then be moved into /var/spool/asterisk/outgoing |
| 20:54.30 | malverian[work] | Hmm.. looks like it's not sending any RTP packets... |
| 20:54.32 | malverian[work] | Interesting. |
| 20:54.49 | *** join/#asterisk zotz (n=zotz@24.231.36.100) |
| 20:54.50 | ManxPower | Log Message: |
| 20:54.50 | ManxPower | Massive cleanups |
| 20:54.57 | ManxPower | I won't be updating for a while. |
| 20:55.16 | jpm_SD | Nexis, funny you should ask about call files... I was just reading about them. |
| 20:55.18 | malverian[work] | Heh... |
| 20:55.43 | malverian[work] | app_echo still appears to be working oddly enough :-P |
| 20:55.49 | spiekey | jpm_SD: can i find this out with the zap tools? (if my isdn card is zaptel compatible) |
| 20:55.54 | spiekey | cause i am not really sure. |
| 20:56.27 | spiekey | its a really famous card here in germany...but that doesnt really help us right now :-/ |
| 20:56.30 | Nexis | yea, its hard to find good documentation on asterisk, voip-info can be hard due to the google cache of the wiki |
| 20:56.43 | ManxPower | "Cisco 3600 3640 Router w/32MFlash/128MDRAM" Whoo! Whoo! It will ship tomorrow |
| 20:56.51 | *** join/#asterisk auslandr (n=auslande@24.112.32.126) |
| 20:57.21 | ManxPower | spiekey, there are no ISDN BRI cards that are native zaptel. there are several that have 3rd party drivers, called ZapBRI |
| 20:58.10 | ManxPower | ~mailinglist |
| 20:58.13 | jbot | [mailinglist] Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 20:58.36 | tzafrir_laptop | zap_hfc and other drivers are part of the bristuff patch |
| 20:58.43 | malverian[work] | ManxPower, What date was that from? |
| 20:58.58 | tzafrir_laptop | availble, e.g, in the debian package of asterisk |
| 20:59.08 | ManxPower | malverian[work], today, but there have been MANY updates this week. guess how I know this. |
| 20:59.26 | malverian[work] | ManxPower, What date are you using successfully? |
| 20:59.49 | *** join/#asterisk dasuberdavid (n=david@gateway.digium.com) |
| 20:59.54 | ManxPower | Updating CVS-HEAD without being on the asterisk-cvs mailing list is like street car racing without wearing a seatbelt |
| 21:00.21 | *** join/#asterisk auslandr (n=auslande@24.112.32.126) |
| 21:00.22 | ManxPower | malverian[work], some time in late september, I think. |
| 21:00.36 | ManxPower | I really should drop back to 1.0.x |
| 21:01.55 | tzafrir_laptop | infinity1, you wanted to ask something about the debs? |
| 21:02.10 | auslandr | Silly Question: When an analog telco line is connected to a zaptel card & ringing, is there a way to have it ring a SIP extension without picking up the line? (If a certain line rings, i want to have a specific extension ring without answering the line, so LD callers won't get billed) |
| 21:02.34 | tzafrir_laptop | spiekey, IIRC the Fritz AVM card is not supported by zaphfc or any other zap-bri driver |
| 21:03.04 | vader-wrk | Are there any softphones that don't require a service and i can use with my asterisk system im going to be setting up? |
| 21:03.19 | vader-wrk | all the ones i seem to find are ones that want you to use them with their specific service |
| 21:03.25 | auslandr | vader-wrk: xten's x-lite is free and works well. |
| 21:03.25 | vader-wrk | ie. gizmo, firefly |
| 21:03.34 | jpm_SD | vader-wrk, x10 makes a softphone. |
| 21:03.44 | jpm_SD | gnomephone |
| 21:03.54 | vader-wrk | do they work with windows? |
| 21:04.00 | auslandr | xlite does |
| 21:04.02 | auslandr | it's quite nice |
| 21:04.09 | jpm_SD | Xlite works well. |
| 21:04.12 | Ariel_ | that is xten |
| 21:04.21 | Ariel_ | no x10 which makes remote devices |
| 21:04.48 | jpm_SD | Ariel_, is correct... sorry for the mis label. |
| 21:04.54 | auslandr | well, they're counterpath now. |
| 21:05.42 | auslandr | http://www.xten.com/index.php?menu=download |
| 21:05.51 | infinity1 | tzafrir_laptop: didn't you know where deb packages are or something once? |
| 21:06.24 | spiekey | tzafrir_laptop: how do i have to plug it in then? i have a octo BRI card here. |
| 21:06.34 | spiekey | does this one go towards telco? |
| 21:07.18 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
| 21:07.28 | ManxPower | I should write a review of this MP3 player I just bought. It has some...issues. |
| 21:07.30 | tzafrir_laptop | infinity1, which debs exactly? |
| 21:07.43 | infinity1 | tzafrir_laptop: asterisk head |
| 21:08.09 | tzafrir_laptop | deb http://rapid.dotsrc.org/ experimental/ |
| 21:08.11 | infinity1 | ManxPower: which one is it |
| 21:08.27 | tzafrir_laptop | spiekey, what card? |
| 21:08.40 | infinity1 | tzafrir_laptop: how long as that been up? |
| 21:08.42 | spiekey | http://www.junghanns.net/en/octoBRI_produkt.html |
| 21:09.19 | *** part/#asterisk oej (n=Olle@apollo.webway.se) |
| 21:09.28 | tzafrir_laptop | this one is naturally supported by drivers fron junghanns . IIRC it is qozap in the bristuff patch |
| 21:09.49 | spiekey | tzafrir_laptop: so i put that card towards telco? |
| 21:09.57 | *** join/#asterisk [Outcast] (i=outcast@222-153-56-150.jetstream.xtra.co.nz) |
| 21:10.09 | Nexis | jpm_SD, if you mentioned the answer to the question i missed it, but can you set the call limit time in a call file? |
| 21:10.09 | [Outcast] | does anyone have an example for sf_w |
| 21:10.13 | ManxPower | infinity1, "Wiren 512" |
| 21:10.19 | ManxPower | ..er... |
| 21:10.21 | [Outcast] | i need the zaptel.conf example |
| 21:10.26 | ManxPower | "Siren 512" |
| 21:10.29 | infinity1 | ManxPower: hm. i had an iriver h10. crap. |
| 21:10.31 | tzafrir_laptop | spiekey, I'm no ISDN expert. However I believe that this is the card's default mode |
| 21:10.38 | infinity1 | ManxPower: now i have ipod :) |
| 21:10.38 | *** join/#asterisk g__ (n=g@itd01fw-fibre.itdepartment.com) |
| 21:10.51 | spiekey | tzafrir_laptop: cool. and where do i need to plug my PBX into then? |
| 21:10.54 | ender | I'm having some problems w/ asterisk -rx |
| 21:10.56 | tzafrir_laptop | [Outcast], the zaptel source comes with one |
| 21:10.56 | ManxPower | It's a nice little device, but it does not have the features I want for podcasts. |
| 21:11.07 | ManxPower | infinity1, I spent $60 on this one. |
| 21:11.11 | ender | I'm trying to get ti to do 'show channels' but all I get is that 'show' isn't a command. |
| 21:11.15 | tzafrir_laptop | spiekey, Again, me no IDSN expert |
| 21:11.19 | ender | even though at the cli I can type 'show channels' and it works. |
| 21:11.39 | spiekey | tzafrir_laptop: but you must have some sort of a "feeling" :P |
| 21:11.44 | ManxPower | I want to be able to delete tracks without hooking up to a computer, and I want it to remember where it last was when I turn it back on. |
| 21:12.17 | ManxPower | ender, asterisk -rx "show channels" |
| 21:12.23 | ManxPower | make sure nothing is eating the quotes |
| 21:15.56 | *** part/#asterisk diclophis (n=diclophi@adsl-69-238-124-226.dsl.pltn13.pacbell.net) |
| 21:16.54 | ender | ah |
| 21:17.21 | *** join/#asterisk Chuji (i=Chuji@pcp09930052pcs.tulipgrove.tn.nash.comcast.net) |
| 21:17.41 | Chuji | cvs head is basically 1.2 beta right? |
| 21:17.59 | Chuji | Like, there aren't any new features going into head right now? |
| 21:18.30 | X-Rob | well. There's not _meant_ to be. |
| 21:18.37 | X-Rob | but new features are. |
| 21:19.44 | Chuji | But those featurs will ultimately make it to 1.2 on the first release? |
| 21:19.54 | Chuji | I'm just wondering, are they running in parallel |
| 21:20.08 | X-Rob | no. HEAD == 1.2 |
| 21:20.14 | drumkilla | malverian[work]: something wrong? |
| 21:20.25 | drumkilla | i hope not, i just changed a ton of stuff in the last couple days |
| 21:20.26 | malverian[work] | drumkilla, Yeah.. a lot of strangeness with RTP. |
| 21:20.33 | drumkilla | oh, well i didn't touch that |
| 21:20.38 | malverian[work] | Actually.. not just RTP it seems.. even with a Zap channel. |
| 21:20.50 | drumkilla | could be the new bridge timeout code |
| 21:20.50 | malverian[work] | Playback() and Background() both appear to not produce audio anymore. |
| 21:21.04 | drumkilla | if you find the date where stuff stops working, let me know |
| 21:21.06 | ender | I was really hoping for 1.2 at astricon. oh well. |
| 21:21.07 | malverian[work] | I reverted to yesterday, but it's still broken. |
| 21:21.19 | malverian[work] | drumkilla, Alright, will do so. |
| 21:21.21 | g__ | Question about AgentCallbackLogin(): does anyone know where this path went? http://bugs.digium.com/bug_view_page.php?bug_id=0001693 |
| 21:21.45 | g__ | (Apparently it was applied to CVS a year ago, but I can't find any reference to it.) |
| 21:21.57 | vader-wrk | whats a good web interface to control asterisk? |
| 21:22.15 | infinity1 | vader-wrk: vi ? :) |
| 21:22.37 | g__ | kill? |
| 21:22.38 | vader-wrk | isn't there web gui's to control asterisk? |
| 21:22.51 | infinity1 | vader-wrk: yea. i've heard of AMP |
| 21:22.54 | infinity1 | maybe thats it |
| 21:22.54 | Ahrimanes | lots |
| 21:23.02 | malverian[work] | Hmm.. app_page.so, interesting. |
| 21:23.03 | DaPrivateer | Ok I have a strange problem, if anyone might be able to offer some insight. After my PBX has been up for a few days it starts ignoring ringing from the second FXO card. I can still dial out on it, but it doesnt detect when the lines ring (this is on 2 out of 3 ports on the second card). When I reboot the box it works again. Any ideas? |
| 21:23.09 | Chuji | sheesh... so I'm much safer goign with 1.2 beta rather than head then? |
| 21:23.12 | vader-wrk | amp? |
| 21:23.27 | infinity1 | vader-wrk: you now know as much as me ... |
| 21:24.03 | Ahrimanes | vader-wrk: http://coalescentsystems.ca/index.php?option=com_content&task=view&id=31&Itemid=57 |
| 21:24.06 | X-Rob | Chuji - no, go with HEAD. |
| 21:24.55 | *** join/#asterisk greekman (n=alex@host254.209.113.199.conversent.net) |
| 21:25.14 | *** join/#asterisk fiber0pti (n=johndoe@207.114.199.98) |
| 21:26.34 | malverian[work] | Anyone else having major issues with CVS head or is it just me? |
| 21:26.38 | malverian[work] | Bloody hell.... |
| 21:27.51 | *** join/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1) |
| 21:30.50 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 21:30.51 | Katty | s/aound/around/ |
| 21:32.17 | gaggaman | a question to asterisk languages: |
| 21:32.51 | gaggaman | I know I can set the language with exten => SetLanguage() |
| 21:33.32 | gaggaman | But is there a way to set the language default to something else than us? |
| 21:35.02 | blitzrage | Katty: nerd-a-licious :) |
| 21:35.44 | Ahrimanes | Katty: hehe |
| 21:37.28 | Katty | what happens when you accelerate a human to almost the speed of light with electromagnetic fields? |
| 21:37.59 | pauldy | Katty baring any extra particle interaction they should sjrink up |
| 21:38.05 | Ahrimanes | Katty: you waer many rings since you man use electromagnetism for that? |
| 21:38.26 | malverian[work] | What the hell is happening here.. this makes no sense.. |
| 21:38.29 | Katty | pauldy: shrink? |
| 21:38.39 | Katty | pauldy: you mean convert partially to energy? |
| 21:38.45 | pauldy | yea sorry typing while sitting pretty much under the keyboard today back issue |
| 21:39.02 | Katty | pauldy: or rearrange and become more dense? |
| 21:39.14 | pauldy | become mroe dense |
| 21:39.31 | Katty | right. |
| 21:39.37 | Katty | hmm. |
| 21:39.42 | Katty | that could present a slight problem. |
| 21:39.58 | *** join/#asterisk w0w0 (n=w0w0@114.Red-83-41-6.dynamicIP.rima-tde.net) |
| 21:40.17 | pauldy | only if this shrinkage creates some type of endothermic reaction |
| 21:40.46 | Katty | would they revert when done going fast? |
| 21:41.23 | pauldy | I would assume so |
| 21:41.36 | CoffeeIV_ | Is there anyway to get access to the uniqueid that is the CDR from a dialplan ? |
| 21:41.41 | pauldy | at lest if you follow special realativity |
| 21:42.36 | pauldy | I would be currious to know if brain functions slow down at those speeds |
| 21:42.56 | CoffeeIV_ | ok I see it referred to as ${UNIQUEID} in the .conf files -- never mind |
| 21:43.24 | *** join/#asterisk mogorman (n=mogorman@gateway.digium.com) |
| 21:45.16 | *** join/#asterisk JASON-0 (n=jason@jason.unitz.ca) |
| 21:46.06 | auslandr | When a line connected to a zaptel card is rining, is there a way to ring through to a SIP extension without picking up the line? |
| 21:46.09 | Error_X | I have installed ztdummy and it is in use in lsmod.. But I can't hear anything when I'm calling my voicemail, echo-test and other codes |
| 21:46.21 | JASON-0 | Hello, I've installed the AMP portal but now when I modify the config files manually I don't know how to refresh the config.. the AMP portal doesn't see my changes.. |
| 21:48.04 | jpm_SD | JASON-0, That's the problem with AMP.. it doesn't allow you to easily make manual edits to your confs. |
| 21:49.15 | JASON-0 | is there a way to reload it through the command prompt ? |
| 21:50.16 | *** join/#asterisk wunderkin (n=wunderki@VDSL-130-13-234-137.PHNX.QWEST.NET) |
| 21:50.26 | pauldy | JASON-0, when using amp the only config files you can edit manualy are the _custom files |
| 21:50.42 | pauldy | thorugh the cli you can always run reload now |
| 21:50.52 | JASON-0 | ok thanks, I will try it out |
| 21:50.53 | JASON-0 | :) |
| 21:51.27 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
| 21:55.38 | *** join/#asterisk BladeRunner05 (n=gianni@adsl-221-213.37-151.net24.it) |
| 21:58.11 | *** join/#asterisk Al_Berto (i=Al_Berto@bandsal.at) |
| 21:58.53 | *** part/#asterisk core-ix (n=ivo@2001:618:400:16fe:3:0:0:3) |
| 22:00.16 | *** join/#asterisk zdrodek (n=zdrodek@tn-pf114-proxy.office.twins.net.pl) |
| 22:01.24 | *** part/#asterisk g__ (n=g@itd01fw-fibre.itdepartment.com) |
| 22:08.26 | malverian[work] | Asterisk Ready. |
| 22:08.26 | malverian[work] | socket: Connection refused |
| 22:08.32 | malverian[work] | ^ Anyone have an idea what would cause that? |
| 22:08.44 | Juggie | on the console? |
| 22:08.51 | malverian[work] | Yes. |
| 22:08.55 | Juggie | odd |
| 22:08.56 | malverian[work] | As I'm starting asterisk up. |
| 22:08.59 | Juggie | never saw that before :P |
| 22:09.01 | malverian[work] | I think this is the root of my problem. |
| 22:09.13 | Juggie | whats your problem |
| 22:11.54 | malverian[work] | For some reason no matter what version of asterisk I checkout, Playback() and Background() do not work. |
| 22:11.57 | malverian[work] | But using Echo() does. |
| 22:12.33 | DaPrivateer | Hrm.... so once a call is made on any of the zap channels the channel state remains offhook... |
| 22:13.40 | Igbothom_III | DaPrivateer; try changing the hangup detection method from Kewl Start (default) to Loop Start |
| 22:14.25 | DaPrivateer | signalling=fxs_ls ? |
| 22:14.30 | Igbothom_III | yup |
| 22:14.55 | Igbothom_III | in both /etc/zaptel.conf and also /etc/asterisk/zapata-channels.conf |
| 22:15.13 | DaPrivateer | hrm |
| 22:15.21 | DaPrivateer | i have a zapata.conf but no zapata-channels.conf |
| 22:15.24 | Igbothom_III | we definitely need this in Australyamate |
| 22:15.28 | infinity1 | does HEAD have a callerid problem? |
| 22:16.00 | infinity1 | exten => 2,n,Set(CALLERID(number)=14082702599) |
| 22:16.01 | infinity1 | exten => 2,n,Dial(${IAXCO1}/14086663884,15) |
| 22:16.06 | infinity1 | and i get unavailable number |
| 22:16.24 | infinity1 | when the phone rings |
| 22:16.55 | malverian[work] | It actually just doesn't send any RTP packets... |
| 22:17.34 | DaPrivateer | lol now it shows on hook while i am using the trunk |
| 22:17.46 | infinity1 | malverian[work]: i'm updated my head on the weekend. my problems aren't that severe. |
| 22:17.56 | Igbothom_III | heh |
| 22:18.16 | Igbothom_III | what country u in? |
| 22:18.40 | infinity1 | me? us |
| 22:18.42 | infinity1 | a |
| 22:18.54 | Igbothom_III | nope, DaPrivateer |
| 22:19.02 | DaPrivateer | US |
| 22:19.20 | Igbothom_III | k, then I'm oot of ideas - I've only just started with * :) |
| 22:20.42 | *** join/#asterisk dinkdinkdink (n=d@pool-68-238-147-186.dllstx.fios.verizon.net) |
| 22:23.04 | *** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 22:24.43 | infinity1 | i kinda doubt my voip provider is broke |
| 22:26.39 | X-Rob | infinity1 - it's CALLERID(num) |
| 22:26.48 | X-Rob | 'show function CALLERID' |
| 22:27.47 | infinity1 | *CLI> show application callerid |
| 22:27.48 | infinity1 | Your application(s) is (are) not registered |
| 22:28.09 | *** join/#asterisk loick (n=loick@APuteaux-151-1-34-29.w82-120.abo.wanadoo.fr) |
| 22:28.26 | *** join/#asterisk loud (n=ariel@cypher.punk.net) |
| 22:29.41 | infinity1 | [Syntax] |
| 22:29.41 | infinity1 | CALLERID(datatype) |
| 22:29.50 | *** join/#asterisk darenw (n=daren@rrcs-67-79-20-162.sw.biz.rr.com) |
| 22:29.50 | infinity1 | looks like i have it right |
| 22:30.05 | infinity1 | ahhh |
| 22:30.12 | infinity1 | they changed it from number to num? |
| 22:31.19 | *** join/#asterisk viLeR (i=1000@66.128.47.232) |
| 22:31.56 | malverian[work] | infinity1, For some reason a reboot fixed it... |
| 22:32.02 | darenw | what would be better to connect to asterisk with from outside my firewall iax or sip? |
| 22:32.20 | malverian[work] | infinity1, Though the only things running prior to reboot were sendmail, apache, metalog, bash, and som ekernel processes.. |
| 22:32.22 | *** join/#asterisk RoyK (n=roy@55.80-202-161.nextgentel.com) |
| 22:33.19 | infinity1 | malverian[work]: strange. maybe an ldconfig would have fixed it? |
| 22:33.37 | infinity1 | X-Rob: nope. num/number doesn't make a difference |
| 22:36.36 | *** join/#asterisk Bigs (n=ST@83.98.237.158) |
| 22:44.33 | *** join/#asterisk rene- (n=rene@201.137.156.205) |
| 22:44.41 | rene- | hello |
| 22:45.32 | rene- | there was some link for a chinese made wctdm card clone, i cant seem to find it, anyone has it at hand? |
| 22:48.02 | *** join/#asterisk hcir (n=hcir@rdbck-static-532.palmer.mtaonline.net) |
| 22:49.22 | *** part/#asterisk ic (n=ic@staff.rbi.speka.net) |
| 22:49.35 | *** part/#asterisk Bigs (n=ST@83.98.237.158) |
| 22:49.36 | X-Rob | rene- - it's by, uh |
| 22:49.40 | X-Rob | openpbx mentions it |
| 22:49.43 | X-Rob | *goes looking*( |
| 22:50.05 | X-Rob | OpenVOX |
| 22:50.06 | X-Rob | <PROTECTED> |
| 22:52.36 | darenw | Any one know a quick solution to this error when starting asterisk... chan_oss.c:257 sound_thread: Read error on sound device: Resource temporarily unavailable |
| 22:52.55 | Katty | mew. |
| 22:54.45 | malverian[work] | Hm... |
| 22:54.55 | malverian[work] | nxserver really should use a newer version of X libraries :-P |
| 22:55.44 | iDunno | darenw: disable the oss channel by putting noload => chan_oss.so in modules.conf? |
| 22:56.18 | iDunno | alternatively, make sure that asterisk is in a group that can access the sound card and that nothing else has grabbed it (like, say, arts or esd) |
| 22:57.51 | DaPrivateer | Ok, still no progress... For some reason after asterisk has been up for a day or so its stops answering 2nd and 3rd slot on my FXO card. The slots all show offhook after receiving their first call, and won't reset. Additionally, I get a polarity warning on all three lines no matter which way I plug the line in. Anyone have any ideas? |
| 22:57.56 | DaPrivateer | (im in the US) |
| 22:58.23 | Igbothom_III | damn, not good :( |
| 23:00.03 | Igbothom_III | exten => _0.,1,Dial(SIP/freecall/${EXTEN:1}) <--- does that look right for extensions.conf? I have the [astrasip-out] context, but don't see any activity at all on the * console (verbose = 50) when I dial a call with a "0" prefix - like it isn't even getting to the * box. I can ring internal SIP phones fine, just not outbound |
| 23:00.15 | *** join/#asterisk wundaboy (n=asdf@c-67-164-107-68.hsd1.or.comcast.net) |
| 23:00.49 | wundaboy | i need help configuring my polycom ip500 to talk to junction networks. i dont have an asterisk box but i figured this would be the place to go. |
| 23:00.51 | DaPrivateer | you have an extension called freecall? |
| 23:01.09 | Igbothom_III | yeah - thatv was a mis-type :) |
| 23:01.21 | Igbothom_III | it was meant to be astrasip-out |
| 23:01.34 | wundaboy | i think i have it configured correctly but its behind nat |
| 23:02.07 | Igbothom_III | it rings engaged, but nothing on the * console |
| 23:02.25 | DaPrivateer | ok wait |
| 23:02.32 | Igbothom_III | wundaboy; one phone, directly to their server thru your firewall? |
| 23:02.49 | *** join/#asterisk kiwnix (n=egarcia@120.red-82-158-158.user.auna.net) |
| 23:02.51 | DaPrivateer | so when you dial 0 and something else, you want it to change to something in the astrasip-out context? |
| 23:02.55 | wundaboy | Igbothom_III: yes, i want it to be able to send and recieve calls |
| 23:03.12 | Igbothom_III | (as you can tell, I'm just learning * and not doing wonderfully well at this point with the docs that are, well, less than useful) |
| 23:03.21 | Igbothom_III | what's the setup you entered into the phone? |
| 23:03.49 | DaPrivateer | Igbothom_III - was what i just said correct? |
| 23:04.21 | Igbothom_III | sorry, when I dial 088888888 (for example) I want it to dial out using the astrasip provider |
| 23:05.20 | wundaboy | Igbothom_III: i have it setup just like the jnctn networks page shows it |
| 23:05.36 | Igbothom_III | ~pb |
| 23:05.37 | jbot | well, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
| 23:06.53 | wundaboy | is there a problem having a phone dial out through nat? |
| 23:07.01 | Igbothom_III | and can you also mention what's happening? Obviously something is going wrong, but you've so far not mentioned that at all! |
| 23:07.27 | iDunno | you want Dial(SIP/${EXTEN}@freecall) |
| 23:07.40 | Igbothom_III | ta, will change it... |
| 23:07.49 | DaPrivateer | lol i was just typing that |
| 23:08.03 | iDunno | assuming that you've got a register => blahwiththefreecallstuff in the top of sip.conf |
| 23:08.23 | iDunno | :) |
| 23:08.43 | Igbothom_III | yup, and sip show registry says I'm registered |
| 23:08.47 | Igbothom_III | hehe |
| 23:09.10 | DaPrivateer | iDunno any idea for my issue? |
| 23:09.22 | Igbothom_III | still nothing on the * console, still rings engaged |
| 23:09.37 | Igbothom_III | sip03.astrasip.com.au:5060 88880426 105 Registered |
| 23:11.26 | rene- | X-Rob: thx |
| 23:11.48 | iDunno | DaPrivateer: nope, sorry, only using a HFC-PCI card in our setup :/ |
| 23:11.50 | wundaboy | does anyone have an asterisk box i can send and recieve calls through? |
| 23:14.29 | dudes | wundaboy - that depends |
| 23:14.38 | dudes | What do you need it for |
| 23:14.48 | Igbothom_III | http://pastebin.ca/26010 is my config |
| 23:15.19 | wundaboy | dudes: i have a polycom ip500 and a junction networks account. |
| 23:15.33 | dudes | So you want a server to run it from |
| 23:15.33 | wundaboy | dudes: i need to talk to the junction networks account and have voicemail. |
| 23:15.34 | Igbothom_III | wundaboy; so far you've not said anything at all about what your actual problem is apart from "its broke". Explaining your situation will likely elicit helpful responses |
| 23:16.24 | wundaboy | Igbothom_III: im not sure what my problem is. all i know is when i pickup the handset and dial i get a busy signal. i believe everything is configured correctly on the phone. |
| 23:16.37 | Igbothom_III | aaahhhh, now some information! |
| 23:16.59 | Igbothom_III | are you dialing numbers on their network? |
| 23:17.06 | wundaboy | no im dialing pstn numbers |
| 23:17.18 | Igbothom_III | have you tried dialing numbers on their network first? |
| 23:17.29 | dudes | so checked sip debug to see why it fails |
| 23:17.30 | wundaboy | like what number? |
| 23:17.41 | wundaboy | sip debug? |
| 23:17.44 | wundaboy | im so confuzed |
| 23:17.53 | dudes | sip debug on the asterisk CLI and make a call |
| 23:17.57 | Igbothom_III | I don't know - YOU are the one subscribed to their network - have a read of their website and see what they have available for troubleshooting |
| 23:18.19 | Igbothom_III | he has no * server - just direct to his ITSP from his 501 |
| 23:18.19 | wundaboy | Igbothom_III: i have followed all steps on their site |
| 23:18.32 | wundaboy | it says to enable STUN but i dont know what that is, or how to do it |
| 23:18.42 | *** join/#asterisk marc324 (n=marc3234@206-248-135-84.dsl.teksavvy.com) |
| 23:18.52 | marc324 | what filesystem to use for *? |
| 23:19.12 | Igbothom_III | and its not working. So, have you tried dialing a number on their network (and as YOU are on their network, we are not) look at their website and see what other numbers are on their network - like support numbers, for instance |
| 23:19.21 | wundaboy | dudes: i have no * box |
| 23:19.34 | Igbothom_III | marc324; a journalling one |
| 23:19.43 | Igbothom_III | most "* distros" use ext3 |
| 23:20.03 | dudes | I can provide you with a server if you give me some config info |
| 23:20.34 | *** join/#asterisk HuNTER_SC (i=Junior@201.34.131.145) |
| 23:20.42 | wundaboy | sure |
| 23:20.45 | Igbothom_III | wb HuNTER_SC |
| 23:20.46 | wundaboy | can i query you? |
| 23:22.17 | *** join/#asterisk dinkdinkdink (n=d@pool-68-238-147-186.dllstx.fios.verizon.net) |
| 23:23.50 | HuNTER_SC | Good Night, I have two ASterisK and you not knowing as to make to establish connection the two. Then can give an aid to me? |
| 23:25.47 | X-Rob | HuNTER_SC - IAX trunking |
| 23:26.36 | HuNTER_SC | X-Rob - which option that I have that to mark in iax.conf? |
| 23:26.44 | X-Rob | HuNTER_SC - lots. Read voip-info.org |
| 23:27.59 | HuNTER_SC | X-Rob I ja read more I did not find therefore that I am here asking. |
| 23:29.24 | marc324 | ext3 or resiserfs? |
| 23:29.30 | marc324 | reiserfs |
| 23:34.15 | *** join/#asterisk dinkdinkdink (n=d@pool-68-238-147-186.dllstx.fios.verizon.net) |
| 23:36.24 | *** join/#asterisk kb1_kanobe (n=krisbout@h24-207-96-50.cst.dccnet.com) |
| 23:40.50 | nvrs | reiserfs! |
| 23:40.52 | kb1_kanobe | g'evening all. |
| 23:40.55 | wundaboy | how does sip work? |
| 23:41.04 | *** join/#asterisk czarex (n=warezxz@tor/session/x-d780a150cfff9fc0) |
| 23:41.48 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 23:44.03 | *** join/#asterisk spootnick (n=irc@50.118.233.220.exetel.com.au) |
| 23:45.04 | *** join/#asterisk n3u7 (n=neutrin0@CPE000d8802a707-CM0011e6c7edb1.cpe.net.cable.rogers.com) |
| 23:45.19 | clyrrad | is there anyting special that needs to be done to enable call waiting on remote IAX phones connected to my Asterisk Box? I have the feature enabled in the phone setup, but I never hear the call waiting beep, it goes directly to voicemail. Anyone know what im doing wrong? |
| 23:45.27 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 23:46.27 | Katty | mew. |
| 23:46.44 | Ariel_ | hello Katty |
| 23:46.50 | n3u7 | greetings |
| 23:46.51 | Igbothom_III | mewning |
| 23:46.53 | Katty | (= |
| 23:47.03 | Ariel_ | Hello everyone |
| 23:47.16 | clyrrad | Hi :) |
| 23:47.31 | *** join/#asterisk bugz (n=bugz@cpe-24-27-67-66.houston.res.rr.com) |
| 23:47.33 | clyrrad | is there anyting special that needs to be done to enable call waiting on remote IAX phones connected to my Asterisk Box? I have the feature enabled in the phone setup, but I never hear the call waiting beep, it goes directly to voicemail. Anyone know what im doing wrong? |
| 23:48.21 | spootnick | anybody using GoIAX? I'm getting a "chan_iax2.c:6629 socket_read: Rejected connect attempt from 204.13.233.114, who was trying to reach 's@'" message when receiving a call |
| 23:48.44 | pauldy | clyrrad, I know by default AMP comes with call waiting disabled |
| 23:48.48 | pauldy | *70 shoudl fix it |
| 23:49.12 | clyrrad | AMP? |
| 23:49.29 | *** join/#asterisk pr0m (n=pr0methe@24-75-196-70.chvlva.adelphia.net) |
| 23:49.43 | pauldy | asterisk management portal, it is used by asterisk at home for the web insterface and controls the asterisk config files |
| 23:50.03 | clyrrad | I am not using Asterisk@Home |
| 23:50.04 | Katty | Ariel_: mine head hurts afer quantum theories :< |
| 23:50.28 | pauldy | ok but are you using amp? |
| 23:50.41 | Ariel_ | Katty, sorry |
| 23:50.42 | clyrrad | Not that i am aware of :s |
| 23:50.44 | twisted[asteria] | software testing == dull |
| 23:50.47 | twisted[asteria] | just FYI |
| 23:51.00 | pauldy | Katty, graduate studies or just having a bit of fun |
| 23:51.01 | Ariel_ | clyrrad, you need to do *70 |
| 23:51.25 | Ariel_ | spootnick, I use it, it works your setup is not setup correctly |
| 23:51.33 | clyrrad | *70 does not do anyting except produce an error on the CLI that there is no *70 context |
| 23:51.54 | bugz | im trying to use the check_sip plugin but cant seem to get the checkcommand string right |
| 23:51.54 | asterisk99 | anyone know if the IAXy (S101i) has a gain control or settings somewehere I can tweak... device is coming in 'hot' |
| 23:51.57 | Katty | pauldy: fun, of course. |
| 23:52.00 | spootnick | Ariel_: any chance you can take a look at my confs in pastebin? I can't figure this one out |
| 23:52.04 | Ariel_ | clyrrad, wow what have you changed on it. AMP default *70 and *71 are for call waiting |
| 23:52.15 | Ariel_ | sure spootnick |
| 23:52.56 | clyrrad | Ariel_ have not changed anyting, perhaps i have to code that into the dial plan? |
| 23:53.16 | Ariel_ | clyrrad, no it is part of the default. |
| 23:53.29 | clyrrad | and call waiting is working on the other phones just by enabling it in the webadmin of the IP Phone |
| 23:53.31 | Ariel_ | if your phone or device is part of the correct context |
| 23:53.50 | spootnick | Ariel_: http://pastebin.ca/26012 |
| 23:53.52 | bugz | asterisk99: ive had problems with them, like some network devices, routers, smart switches, dhcp servers, etc, like to store the mac address and ip mapping of the iaxy, causing problems with it being flashed and put back on the network with different information |
| 23:54.00 | spootnick | Ariel_: i'm trying to dial in using my DID |
| 23:54.05 | Ariel_ | spootnick, will do give me a minute |
| 23:54.10 | clyrrad | I have the phone included in the incomming, outgoing, parkedcalls and voicemail context, is there one i have missed? |
| 23:54.45 | groogs | clyrrad: phones should be in the from-internal context, that includes all those |
| 23:54.51 | groogs | you can't put a device in more than one context .... |
| 23:55.15 | kuku5 | Is there a way to view parked calls on a phone ? |
| 23:55.25 | clyrrad | groogs, yes i have a context called 'my_phones' which includes all the other contexts |
| 23:56.00 | clyrrad | and in sip.conf or iax.conf in the phones context=my_phones |
| 23:56.02 | groogs | why? |
| 23:56.05 | spootnick | kuku5: to *view* parked calls... not that I'm aware of. but you can probably fetch a list and have it said back to you using text to speech |
| 23:57.07 | groogs | clyrrad: what is wrong with from-internal ..? either way if you do have a reason to use your own, look at from-internal in extensions.conf.. you probably don't have the app-callwaiting etc lines |
| 23:57.35 | *** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 23:57.52 | Igbothom_III | hhmmm, http://pastebin.ca/26013 is my config, but I cannot make outbound calls (not tried receiving calls yet) thru sip. Any clues? I followed the info in voip-info.org, but the config this gave doesn't work - no * console output at all (verbose=50) and an engaged signal upon calling. internal calls work fine. |
| 23:58.06 | clyrrad | groogs, yes you are correct, i do not have a from-internal or call waiting lines, do you have an online reference i can see? |
| 23:58.26 | groogs | no, look at from-internal |
| 23:58.43 | groogs | though i'm still curious why you're not just using that |
| 23:58.56 | marc324 | is the tcl/tk package needed for asterisk? |
| 23:59.38 | clyrrad | groogs, i never knew about it, its alway worked with out it |
| 23:59.41 | *** join/#asterisk supaigtr (n=yurplsl@152.53.17.1) |
| 23:59.47 | supaigtr | Hello |