| 00:03.01 | The_DoC^ | J4k3 : that like you posted will work for a fxo |
| 00:03.17 | The_DoC^ | s/like/link |
| 00:03.27 | The_DoC^ | link even |
| 00:10.48 | *** join/#asterisk slima (i=slima@unaffiliated/slima) |
| 00:15.26 | *** join/#asterisk james2703 (n=jsharryp@58.69.37.13) |
| 00:19.54 | james2703 | good day guys, can someone has an insight on how many g.711 calls can accomodate on E1 Channels ? my setup is * -> SIP -> PSTN |
| 00:20.17 | JT | e1 data or voice? |
| 00:20.22 | *** join/#asterisk orkid_ (n=orkid@bas1-barrie18-1242382227.dsl.bell.ca) |
| 00:20.27 | james2703 | data |
| 00:20.29 | *** join/#asterisk ez` (n=ez@c66.203.210-59.clta.globetrotter.net) |
| 00:20.45 | JT | does it go over the Internet? |
| 00:21.17 | james2703 | yes, my latency with my SIP GW provider is 5-7 miliseconds |
| 00:21.32 | james2703 | but i don't know after my ITSP where or how did they route the PSTN |
| 00:22.09 | JT | 23-24 calls if the link is dedicated to the purpose |
| 00:23.08 | james2703 | oic, i'm having a 12 simultaneous call only with 2mbps max my own mrtg says i'm only getting about 1.6 mbps but i'm having a problem with the quality |
| 00:24.00 | JT | what sort of quality problems? |
| 00:24.23 | james2703 | on the agent side the voice is clear, but on the PSTN side (cal |
| 00:24.35 | james2703 | callee according to them it's choppy |
| 00:25.13 | james2703 | i'm suspecting that my ITSP is routing the call via internet |
| 00:25.37 | JT | choppy, you'll have to examine it more carefully |
| 00:25.45 | JT | it could just be RTP silence supression |
| 00:25.52 | *** join/#asterisk bkw__ (n=brian@adsl-70-142-59-250.dsl.tul2ok.sbcglobal.net) |
| 00:26.04 | james2703 | actually i ask my ITSP to disable this VAD thing on their side |
| 00:26.43 | JT | but did they? |
| 00:27.00 | james2703 | as far as i know yes, because the console stop bugging me already :) |
| 00:27.08 | james2703 | but to be honest i don't really know |
| 00:27.34 | *** join/#asterisk booray (n=booray@static-66-15-145-196.bdsl.verizon.net) |
| 00:27.38 | james2703 | my peer equipment according to them is Quintum Tenor |
| 00:27.55 | james2703 | i'm using recent asterisk 1.2 branch |
| 00:28.38 | JT | maybe the silence supression is upstream to them |
| 00:28.48 | JT | but have you actually examined the choppyness for yourself? |
| 00:29.24 | james2703 | JT: yes i did, for me it's acceptable but for the agent it's not :) |
| 00:29.53 | james2703 | JT: i suspect this could be a softphone issue also |
| 00:30.12 | james2703 | they are using Counterpath I-Beam softphone |
| 00:30.38 | JT | hrm maybe, softphones tend to suck |
| 00:31.40 | james2703 | JT: how about using a g.729a codec ? |
| 00:32.11 | james2703 | JT: but i suspect this is not a bandwidth issue since my max out is 2mbps while i'm only getting 1.6 |
| 00:32.51 | james2703 | JT: on a g.729a pass thru how many channels do i need to license ? any idea on 15 seater agent |
| 00:33.07 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 00:33.07 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 00:33.24 | james2703 | do you think 30 channels license would be enough? |
| 00:34.28 | JT | you should get a hell of a lot more channels on g.729 |
| 00:34.43 | james2703 | hehehe |
| 00:35.23 | james2703 | actually, last time when i test the g.729 intel version it accumulate upto 56 channels on 12 simultaneous call :p |
| 00:37.17 | JT | not sure what you mean |
| 00:37.21 | *** join/#asterisk clintc (n=clintc@128.227.123.25) |
| 00:37.27 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 00:38.41 | james2703 | are you familiar with g.729 (free version/development version)? |
| 00:38.47 | *** join/#asterisk J4k3- (i=jsuter@dhcp-12-197-128-58.intrastar.net) |
| 00:38.53 | james2703 | this codec doesn't come from digium |
| 00:39.46 | JT | yes but what you you mean 56 channels on 12 calls? |
| 00:39.56 | *** join/#asterisk dj-fu (n=deejay@203-167-190-166.dsl.clear.net.nz) |
| 00:40.36 | SwK | the free g729 version should not be used |
| 00:40.51 | james2703 | SwK: yes i know |
| 00:41.00 | SwK | you are violating the patent and opening yourself to a large claim (for 10K channels) |
| 00:41.07 | clintc | we have a 4 port digium t1/e1 card and are trying to use it with our existing t1s that use e&m wink signaling, it mostly works but drops calls at random times and leaves "strange state 6" in the asterisk message log, has anyone gotten e&m wink to work reliably? did you change any defaults in the zapata.conf file for the e&m settings? |
| 00:41.15 | JT | SwK: he said testing |
| 00:41.30 | james2703 | SwK: not only that, it really sucks compare to g.729a comes from digium :p |
| 00:41.59 | dj-fu | Hi. I've currently got an a@h server running with 4 sip Linksys phones. I'm using pstn lines, I do not know anything about what kind of signalling they are on. Can anyone point me in the right direction, I want to install asterisk by hand instead of with a@h |
| 00:42.29 | JT | use fxsks signalling usually on fxo lines |
| 00:42.33 | JT | otherwise fxsls |
| 00:42.47 | *** join/#asterisk orkid_ (n=orkid@74.13.35.98) |
| 00:42.53 | dj-fu | hell, I don't even know what kind of card is in the a@h box, it was installed before I got here. problem is it stops working all the time, the quickdial extensions don't work at all |
| 00:42.54 | james2703 | JT: on every call i make when i type on the console show channel i can see 1 active call and 2 channels up right? |
| 00:42.56 | dj-fu | other nonsense |
| 00:43.10 | james2703 | show channels |
| 00:43.35 | james2703 | 20 active channels |
| 00:43.35 | james2703 | 10 active calls |
| 00:44.11 | JT | james2703: sure, the g.729 usage is dependant upon the number of concurrent transcoding sessions |
| 00:44.44 | james2703 | so you mean 10 active calls = 20 g.729a license? |
| 00:45.00 | *** join/#asterisk saftsack (n=oliver@pD9E053E4.dip.t-dialin.net) |
| 00:45.17 | clintc | dj-fu: sounds like an old setup, the fastest way to go might be to use the new asterisk-gui, try that first as it sounds like you have a dedicated box |
| 00:45.33 | JT | no, 10 active transcoding sessions, so if your host is g.729, and your phones are, and the media goes direct, that's 0 transcoding sessions |
| 00:45.42 | dj-fu | clintc, I have a dedicated box, but I wouldn' tlike to anymore. I'm trying to move it from that old box to this new machine I've put it |
| 00:45.52 | dj-fu | it's higher specced and has the pppoe connection running straight into it |
| 00:46.37 | JT | yeah, straight asterisk is a good idea |
| 00:46.47 | clintc | dj-fu: will the new box only be running asterisk? if so, one of the complete iso setups will get you going really quickly |
| 00:47.24 | dj-fu | clintc, no, currently it's running pppd/iptables/squid/samba/http/ssh |
| 00:47.27 | *** join/#asterisk hoowa (n=chatzill@210.83.203.100) |
| 00:47.34 | dj-fu | if possible, I'd like to have it running on that machine. |
| 00:47.47 | *** join/#asterisk J4k3^ (i=jsuter@dhcp-12-197-128-58.intrastar.net) |
| 00:47.48 | hoowa | :) |
| 00:47.58 | JT | clintc: why advise him to go asterisk-gui, he's already made the better choice of going with asterisk? :) |
| 00:48.03 | james2703 | JT: if my softphone is g.729a then my SIP Trunk to my provider is G.729a then do i need a g.729 license? |
| 00:48.50 | clintc | JT: I had the impression he is starting at the beginning of the learning curve... and the current phone system is unstable |
| 00:48.53 | james2703 | JT: sorry man, i don't really get it :) could you elaborate more? |
| 00:48.59 | JT | james2703: not if the media is passed through, but if asterisk ever has to manipulate the audio, like play a sound file, etc, then you will need transcoding licences |
| 00:49.37 | dj-fu | clintc, I am at the beginning of the learning curve, but only for asterisk - i'm quite competent in everything else |
| 00:49.43 | JT | clintc: better to learn asterisk itself, sounds like his inhereted a@h system was unstable too |
| 00:49.54 | JT | james2703: are you using freepbx? |
| 00:50.11 | james2703 | JT: no |
| 00:50.19 | james2703 | JT: plain asterisk |
| 00:50.54 | JT | ah ok, sip trunk is a freepbx term |
| 00:51.06 | JT | as the concept doesn't really make much sense |
| 00:51.18 | JT | you have sip connections |
| 00:51.30 | JT | that are created and destroyed as needed, but no trunks |
| 00:51.33 | clintc | dj-fu: learning straight asterisk is the best, you can have it do just what you want and have a much less complex dialplan, but it will take longer to get it set up... if you can take the time then yes |
| 00:51.52 | dj-fu | I'd like to learn it so I can roll it out in the future by hand |
| 00:51.55 | J4k3 | JT: I think the idea is a "virtual trunk" |
| 00:52.04 | james2703 | JT: ok, then |
| 00:52.08 | dj-fu | The a@h box has a 4port phone thing in it. is this an ISDN card? or? |
| 00:52.23 | JT | J4k3: a freepbx idea for newbiews |
| 00:52.25 | JT | :) |
| 00:52.26 | mitcheloc | isn't aah really old? |
| 00:52.46 | dj-fu | I assume so, that may explain why it's terribly unstable |
| 00:52.47 | JT | dj-fu: are the lines analogue? it's probably a TDM400P |
| 00:52.53 | dj-fu | JT, yes, analogue lines |
| 00:53.01 | JT | isdn is only for digital |
| 00:53.08 | JT | you cannot intermix the hardware |
| 00:53.13 | clintc | dj-fu: but I would always run the phone system on it's own box with no other jobs, especially for a work environment |
| 00:53.26 | dj-fu | it's not going to be utilized much |
| 00:53.33 | J4k3 | JT: well, thats what freepbx's target audience is, isn't it? |
| 00:53.39 | J4k3 | n00bz |
| 00:53.59 | JT | J4k3: yes, but all the same, it's a freepbx specific term really :) |
| 00:54.02 | dj-fu | says the guy with numbers in his name |
| 00:54.25 | dj-fu | clintc, is cpu usage a problem, or? |
| 00:54.49 | dj-fu | well I think that asterisk by hand on this system is probably better than this current unstable pos |
| 00:55.29 | dj-fu | Is there a handy tutorial anywhere |
| 00:56.05 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 00:56.07 | dj-fu | JT, providing it's a TDM400P, are options for this card in the kernel configuration? |
| 00:56.18 | dj-fu | the tutorials I've seen have stuff about zaptel and whatnot |
| 00:56.36 | clintc | dj-fu: no 4 lines will not tax it much... I like to set up a phone system, lock it down, and then not touch it unless absolutely necessary, you can't do that if the box does lots of other things |
| 00:57.15 | dj-fu | there's only two actual physical lines, the card is a 4port though :\ |
| 00:57.32 | JT | dj-fu: not so much the kernel, you need to configure /etc/zaptel.conf and /etc/asterisk/zapata.conf |
| 00:57.48 | dj-fu | I see |
| 00:57.48 | *** join/#asterisk Grnd-Wire (n=groundwi@71-217-127-175.tukw.qwest.net) |
| 00:57.51 | JT | dj-fu: is it populated with 4 modules? |
| 00:57.55 | dj-fu | there's a net-misc/zaptel package in my servers package manager |
| 00:57.59 | dj-fu | JT, how do you mean? |
| 00:58.09 | clintc | dj-fu: the setup won't be that hard... it's the dialplan features |
| 00:58.31 | Grnd-Wire | greetings! |
| 00:58.31 | JT | a tdm400p can have between 0 and 4 pluggable module boards on it (mind you it's pretty useless with 0) |
| 00:58.39 | clintc | dj-fu: for a system that small why are they using asterisk, just curious? |
| 00:58.54 | *** join/#asterisk sahafeez (n=sahafeez@ip68-7-112-53.sd.sd.cox.net) |
| 00:59.01 | JT | a module can be FXS or FXO, there are different types |
| 00:59.10 | Grnd-Wire | dj-fu: Is there a system "too small" to use Asterisk for? |
| 00:59.44 | booray | quick question, if I change a voicemail password, yet it doesn't update in voicemail.conf, where did it go? |
| 00:59.54 | orkid_ | /dev/null ? :) |
| 00:59.55 | JT | Grnd-Wire: shouldn't you be asking clintc? |
| 01:00.11 | booray | orkid_: you'd think so, except it actually worked |
| 01:00.18 | booray | s/orkid_/orkid |
| 01:00.33 | Grnd-Wire | oh oops.. |
| 01:00.42 | Grnd-Wire | Yes, yes I should be :) |
| 01:01.20 | clintc | Grnd-Wire: I would say it depends on how reliable you need your phone system to be... a commodity pc without proper support does not make for a very reliable phone system |
| 01:02.18 | orkid | booray, weird then |
| 01:02.26 | Grnd-Wire | clintc: Yeah - that's why you buy a "commodity Dell" with a 4hour or NBD support agreement (still around $800).. I think that takes care of your proper support issue.. |
| 01:02.54 | dj-fu | clintc, I believe the reason behind going with asterisk was so that we could have a sip connection to Auckland, the major city in New Zealand, where most of our suppliers are |
| 01:03.00 | Grnd-Wire | clintc: And I can actually get parts for a true commodity PC (that uses standard hardware) |
| 01:03.04 | *** join/#asterisk bkw__ (n=brian@adsl-70-142-59-250.dsl.tul2ok.sbcglobal.net) |
| 01:03.06 | dj-fu | and therefore have free calling to there |
| 01:03.08 | JT | takes care of one aspect, and 4 hours+ is a lot of downtime |
| 01:03.35 | JT | dj-fu: ah, local calls are free, aren't they? |
| 01:03.48 | dj-fu | JT, it's 3 hours away, different area code, not free |
| 01:03.52 | dj-fu | we're in Rotorua, not Auckland :) |
| 01:04.03 | JT | dj-fu: yes, but i asked if local calls were free |
| 01:04.04 | Grnd-Wire | JT: I don't know of any phone vendors here in town that would be able to respond any faster.. |
| 01:04.12 | *** join/#asterisk i3inary (i=i3inary@ip68-8-91-87.sd.sd.cox.net) |
| 01:04.20 | JT | Grnd-Wire: most pabxes are less likely to fail |
| 01:04.42 | shido6 | ? |
| 01:04.43 | dj-fu | it also lets us use the sip connection to dial mobile phones, for a cheaper rate |
| 01:05.04 | Grnd-Wire | JT: This whole discussion came about from "how small is the office".. I would counter that argument with, smaller offices are WILLING to accept some level of risk/downtime, because they are NOT willing to buy really good maintenance contracts.. |
| 01:05.07 | JT | true |
| 01:05.21 | JT | dj-fu: was it a commercial sip provider? |
| 01:05.43 | clintc | Grnd-Wire: yes, that is most true |
| 01:05.46 | dj-fu | JT, yes, something like $10/month for free calling in Auckland (where the sip server is located), and cheap calling to other stuff |
| 01:05.48 | JT | Grnd-Wire: most even small pbx boxes are unlikely to fail |
| 01:05.55 | JT | dj-fu: ah ok |
| 01:06.21 | Grnd-Wire | JT: I had a Merlin Messaging R3 fail on Tuesday afternoon.. and I couldn't re-install it until Friday afternoon when it came back from repair. |
| 01:06.25 | dj-fu | I'd much rather have a PBX, but asterisk seems like a nice thing to learn, something I can add to my repetoir |
| 01:06.30 | clintc | Grnd-Wire: which why I said it *depends* on how reliable things need to be |
| 01:06.46 | orkid | clintc, what. you can't build a reliable asterisk box?! |
| 01:06.57 | JT | orkid: not cheaply |
| 01:07.02 | orkid | why not? |
| 01:07.08 | orkid | what do you need? good fxo/fxs? |
| 01:07.13 | JT | good computer hardware is not cheap |
| 01:07.19 | Grnd-Wire | orkid: That's kinda what I'm trying to figure out.. |
| 01:07.20 | JT | cheapshit hardware blows up |
| 01:07.24 | orkid | cpu speed?! |
| 01:07.34 | Grnd-Wire | JT: Are you familiar with Dell's business product line? (Optiplex) |
| 01:07.34 | JT | no, redunadt power supplies, raid 1 |
| 01:07.36 | orkid | what do you need. |
| 01:07.37 | JT | server grade mobo |
| 01:07.38 | orkid | oh COME OOON |
| 01:07.48 | orkid | RAID1 for a PBX? i haven't had an HD failure in years |
| 01:07.49 | JT | Grnd-Wire: it's a business desktop line |
| 01:07.51 | JT | not server |
| 01:08.12 | JT | orkid: yes, pbxes are highly critial, i don't run any servers not on RAID, let alone a pbx of all things |
| 01:08.12 | Grnd-Wire | JT: So buy a server - for the features you get, it still beats the price of most phone systems.. |
| 01:08.21 | JT | i've had a number of power supplies blow up |
| 01:08.25 | JT | Grnd-Wire: indeed |
| 01:08.35 | JT | i'm just saying it's not /that/ cheap to do it right |
| 01:08.44 | Grnd-Wire | JT: Except for the Vodavi Starplus STS, and maybe the Partner.. |
| 01:08.57 | orkid | if for business yes, but for home i don't see a reason |
| 01:09.04 | JT | i've seen plenty of hard drive failures too |
| 01:09.13 | Grnd-Wire | but you factor in the fact it can do off-premise extensions, and IP Trunking (something every office can use :P) - and it becomes a class all of its own. |
| 01:09.16 | clintc | orkid: we use, raid arrays, dual redundant power, server grade parts, and redundant backhard for everything... but we also have 80 people using this phone system... it just can not go down |
| 01:09.23 | dj-fu | one more question - is there anything I can backup from my a@h machine to make my manual install easier? individual phone configuration perhaps? |
| 01:09.28 | JT | sure, but i run my home servers like a mission critical enterprise too :P |
| 01:09.28 | orkid | clintc, oh i see. then obviously. |
| 01:09.38 | JT | my gateway hasn't had any system downtime in over 400 days |
| 01:09.56 | orkid | so that's the only problem, your computer hardware reliability? how bout dropped calls/line noise/etc? |
| 01:10.22 | JT | orkid: yeah, software reliability is a big thing at the moment |
| 01:10.35 | JT | things like asterisk let you do a lot of things cheaply |
| 01:10.43 | JT | they also have a lot of bugs as a result |
| 01:10.46 | orkid | what do you mean by 'big thing' ... . difficult to get? |
| 01:10.50 | orkid | oh is ee |
| 01:10.51 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 01:10.56 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 01:11.11 | JT | i mean, some people have boxes that asterisk runs on for months without a restart |
| 01:11.22 | JT | other people restart their asterisk daemon nightly |
| 01:11.45 | JT | due to difficult to track down problems, often relating to hardware or hardware drivers |
| 01:11.57 | Grnd-Wire | JT: So to change the subject a little bit.. What would be the best way to configure a 1-to-1 relationship between a Zap channel, and an extension.. I want to configure it so the phone will always generate calls on a specific Zap port when it dials out.. It's for a bridge, before you start telling me that's "old thinking".. |
| 01:12.13 | JT | Grnd-Wire: that's easy |
| 01:12.28 | JT | just get the extension to dial Zap/<num> |
| 01:12.30 | Grnd-Wire | JT: Is it just gotoif logic in the dialplan? |
| 01:12.37 | JT | why would you need that? |
| 01:12.40 | JT | i'm confused |
| 01:12.43 | clintc | we are having a problem with e&m wink signalling... we have pri circuits on order... but if anyone has changed the e&m defaults to prevent dropped calls, let me know |
| 01:13.02 | JT | clintc: haven't had e&m experience myself |
| 01:13.09 | [TK]D-Fender | JT : You're not alone... |
| 01:13.21 | Grnd-Wire | JT: Because every Zap port is plugged into an extension on their main phone system.. I'm extending four phones to a remote office, using Asterisk as a bridge on either side.. |
| 01:13.30 | clintc | JT: yes, almost no one has... hence the need for the PRI |
| 01:13.37 | dj-fu | How do I know if I have pri or bri |
| 01:13.55 | Grnd-Wire | JT: So when they dial out, they need to use their port - and when they call other people in the office, it'll actually look like it's coming from them.. :P |
| 01:14.03 | [TK]D-Fender | dj-fu : Is this a normal home circuit where you are? |
| 01:14.08 | JT | dj-fu: i thought you said you had analogue |
| 01:14.31 | JT | Grnd-Wire: umm, a context for every extension? |
| 01:14.38 | JT | that ought to do it |
| 01:14.39 | dj-fu | JT, yeah, I do - it's an option in installing asterisk for my distro, gentoo - USE="pri" etc. |
| 01:14.49 | dj-fu | do I not need pri/bri? |
| 01:14.52 | JT | no |
| 01:14.55 | JT | they're only for digital |
| 01:14.57 | Grnd-Wire | JT: Is that the best way, or is some gotoif logic in the main context going to be better? |
| 01:15.00 | dj-fu | ah, super - thanks |
| 01:15.00 | JT | not applicable for analogue |
| 01:15.01 | [TK]D-Fender | Oh go... PORTS.... |
| 01:15.50 | JT | Grnd-Wire: what's the problem you're seeing, repeated code? |
| 01:15.52 | saftsack | why did you stop the reliability discussion? xD |
| 01:16.11 | JT | saftsack: seemed that people sort of agreed! |
| 01:16.15 | Grnd-Wire | JT: Yeah, actually - I guess that's it.. |
| 01:16.32 | Grnd-Wire | JT: Cause I can assign a variable based on some conditionals, then trigger the dial command.. |
| 01:16.45 | Grnd-Wire | (using that variable as part of the dial string, of course) |
| 01:16.46 | JT | Grnd-Wire: use a macro or something to dial, it could look at the calling extension and you could look up a reference of global vars or astdb |
| 01:16.47 | saftsack | we use, raid arrays, dual redundant power, server grade parts, and redundant backhard for everything |
| 01:16.54 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.149) |
| 01:17.26 | JT | saftsack: cool |
| 01:17.26 | saftsack | this was said before and my opinion is, why dont you use 2 cheap server connected with heartbeat= |
| 01:17.30 | Grnd-Wire | JT: hmm.. yeah - I think that sounds even better.. :D |
| 01:18.03 | clintc | saftsack: we connect the servers with shared block devices |
| 01:18.11 | dj-fu | mm, heartbeat is great |
| 01:18.14 | JT | there is no asterisk solution for failover of media gatewaying that doesn't drop all current calls :/ |
| 01:18.20 | saftsack | you can get a cheap desktop system for about 250eu. two times this would be 500eu. then you buy all connections to the outside as embedded gateways and then you have a cheap and reliable phone system |
| 01:18.31 | saftsack | clintc, what are shared block devices? |
| 01:18.54 | saftsack | wait i google for it |
| 01:19.02 | dj-fu | if I pop open the side of my box and take a look at the zaptel card in it, will it have a stamp on a chip so I can work out what card it is? |
| 01:19.18 | JT | dj-fu: yeah, and you can see what modules are in it |
| 01:19.29 | JT | it's likely a tdm400p, that takes up to 4 |
| 01:19.30 | dj-fu | I see |
| 01:19.45 | clintc | dj-fu: the modules can be any mix of fxo/fxs |
| 01:19.55 | dj-fu | sorry, what is fxo/fxs. |
| 01:19.56 | saftsack | tdm 400p with echo guaranteeeee ;) |
| 01:20.11 | JT | ~fxofxs |
| 01:20.22 | jbot | well, fxofxs is An FXO port expects to receive dialtone and receive ring voltage, you plug a phone LINE into this. An FXS port expects to provide dialtone and provide ring voltage, you plug a PHONE into this. |
| 01:20.22 | clintc | but you probably have 2 fxo and 2 fxs |
| 01:20.27 | Grnd-Wire | dj-fu: Also, if you can identify what COLOR they are - that's helpful.. |
| 01:20.37 | dj-fu | righto |
| 01:20.41 | *** join/#asterisk vvortex3 (i=Lanniste@71.193.3.146) |
| 01:20.43 | vvortex3 | hi |
| 01:20.46 | Grnd-Wire | dj-fu: If they are RED, you plug LINES into them (FXO).. if they are GREEN, you plug PHONES into them (FXS) |
| 01:20.47 | saftsack | clintc, do you mean network-raid1 with shared block device? |
| 01:20.59 | clintc | saftsack: yes, that's it |
| 01:21.09 | saftsack | dont swap them. you can damage your card then |
| 01:21.30 | saftsack | clintc, but there must be a main computer then, which executes all commands .... |
| 01:21.51 | vvortex3 | so...i am the typical confused noob with asterisk, i have asterisk@home going...got SIP in and out from broadvoice, im not understanding how inbound calls get to my asterisk@home box...does the box itself LOG INTO broadvoice so that calls can come in to me, or does broadvoice initiate connections to my asterisk@home box |
| 01:22.02 | shido6 | media failover is easier to realise if you are using IAX |
| 01:22.09 | JT | shido6: howso? |
| 01:22.15 | dj-fu | Could anyone take a quick look at this? I need to work out which use flags to enable (the things which are like -stuff) http://rafb.net/p/gAW4pg70.html |
| 01:22.27 | clintc | saftsack: we have a primary, secondary, and test ... with scripts that setup the secondary if the primary fails |
| 01:22.33 | saftsack | i think that the hard disks are not the most critical parts because you can use flash disks for the initial functions of the pbx. |
| 01:22.40 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 01:22.43 | shido6 | well, that is better answered if we know or if the app watching thet box knows when a failure is about to occur then transfer all calls to another IAX server |
| 01:22.49 | saftsack | clintc, ok then it is safe :) but why do you use server grade hardware? |
| 01:23.00 | [TK]D-Fender | vvortex3 : ... |
| 01:23.04 | [TK]D-Fender | ~freepbx |
| 01:23.10 | jbot | from memory, freepbx is unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 01:23.11 | [TK]D-Fender | ~trixbox |
| 01:23.12 | jbot | extra, extra, read all about it, trixbox is unable to be supported here. Try joining #freepbx and asking there, or use the trixbox forums at http://www.trixbox.org/modules/newbb/ |
| 01:23.13 | vvortex3 | im using that |
| 01:23.23 | shido6 | A calling B to call C then dropping B , B being the "ailure about to happen |
| 01:23.29 | clintc | saftsack: 80 person call center... we record every concurrent call.... |
| 01:23.35 | vvortex3 | but the question itslef is asterisk oriented |
| 01:23.45 | saftsack | is this allowed? :) |
| 01:23.55 | shido6 | lets wrap our brains around this for a moment... |
| 01:24.08 | shido6 | what happens when A is about to take a crap. |
| 01:24.10 | shido6 | or C |
| 01:24.20 | saftsack | clintc, what has this to do with server grade hardware? i mean a cheaper desktop board will bring the same performances, just without the reliability |
| 01:24.22 | shido6 | you will need a matrix of servers |
| 01:24.34 | clintc | saftsack: "this call may be recorded for quality assurance.... :) |
| 01:24.44 | shido6 | one for each box or three for each 1 box... hrmm.. |
| 01:25.01 | saftsack | clintc, so that you can throw out people after every month? xD :) |
| 01:25.13 | *** join/#asterisk J4k3 (i=jsuter@12.197.128.58) |
| 01:25.28 | [TK]D-Fender | vvortex3 ; there is little to be understood when it comes to configuring via FreePBX. you fill in the blanks, and thats about it. There are plenty of guides out there specifically with it & Broadvoice in mind. |
| 01:25.29 | JT | saftsack: cheaper desktop boards don't have the same performance either |
| 01:25.35 | vvortex3 | i guess a more appropriate question would be, does asterisk log into my outbound SIP trunks |
| 01:25.40 | JT | shido6: i fail to see how that resolves media gatewaying |
| 01:25.49 | *** join/#asterisk h0 (i=fakhir@unaffiliated/fakhir) |
| 01:25.50 | vvortex3 | or do the sip providers initiate a connection to my asterisk box for each call |
| 01:25.53 | clintc | saftsack: we also have a project wherein we have to transcribe every call for about 1000 calls |
| 01:25.56 | shido6 | media to where? |
| 01:26.05 | JT | a physical interface |
| 01:26.11 | *** join/#asterisk quidpro (n=quid246@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 01:26.19 | DrukenLPY | vvortex3: it depends on your configuration with your provider |
| 01:26.25 | DrukenLPY | it can go both ways |
| 01:26.48 | *** join/#asterisk J4k3 (i=jsuter@dhcp-12-197-128-58.intrastar.net) |
| 01:27.25 | shido6 | the easy way is to reinitiate the calls on another box when the calls fail... which means all calls going through the box will need a reliable path or URI to reach them even behind their respective nats... |
| 01:27.31 | shido6 | so im talking to sally |
| 01:27.32 | vvortex3 | damn, i dont know which broadvoice does by default |
| 01:27.36 | shido6 | boxes die |
| 01:27.36 | JT | the correct answer is that the provider initialises an inbound sip call to you |
| 01:27.40 | shido6 | then seconds later |
| 01:27.41 | [TK]D-Fender | vvortex3 : Both of those statemens mean jsut about the same thing |
| 01:27.43 | shido6 | my phone rings |
| 01:27.53 | shido6 | and im connected again to whoever I was talking to with no intervention from me |
| 01:27.58 | shido6 | which isnt what you're looking for |
| 01:27.58 | JT | and your box if on a dynamix ip, must regularly register with the provider, so it knows where you are |
| 01:28.00 | shido6 | ....... |
| 01:28.06 | shido6 | hrmm... |
| 01:28.21 | shido6 | how could we use the IAX transfer to NOT drop the call.... |
| 01:28.36 | shido6 | without buying tons of boxes.. |
| 01:28.41 | JT | shido6: it's not a transfer problem |
| 01:28.59 | JT | it's a problem where you actually require the one task to be handled in parallel by 2 boxes |
| 01:29.07 | saftsack | clintc, ok ... what telephones do you use? |
| 01:29.15 | shido6 | can u explain that? |
| 01:29.17 | JT | telco hardware does this all the time |
| 01:29.40 | JT | shido6: 2 cpus/systems perform the same task so if one fails, the other seamlessly continues |
| 01:29.50 | shido6 | hrmm... |
| 01:30.15 | vvortex3 | btw, brodcom confirms that I initiate a connection to them to receive inbound calls |
| 01:30.31 | clintc | saftsack: we are working on our own java based softphone that we hope will be of high enough quality to release to the community, polycom 501s for the vips, linksys phones for staff |
| 01:30.33 | shido6 | so if im using a T1 with 23 ports and 2 calls are present on the T ive got two boxes using those channels simultaneously - 1 "listening" and one talking on the channel.... ? |
| 01:31.03 | JT | vvortex3: read what i said, i explained exactly how it works, their explanation is faulty. you REGISTER with a sip provider so they know how to call you, if there's a call to send to you, they initiate it |
| 01:31.20 | [TK]D-Fender | vvortex3 : Here's the short version : You "register" to them to inform them of your current IP. they then know to send calls to you to that IP. This has no impact on your ability to SEND calls. Only to tell them where tos end the calls coming in. |
| 01:31.29 | shido6 | lets use 1 channel for example. |
| 01:31.36 | shido6 | 2 boxes for 1 channel on the T |
| 01:31.43 | JT | shido6: something like that, it comes down to having shared hardware at the end anyway |
| 01:31.54 | [TK]D-Fender | vvortex3 : THIS IS AN INTERMITTENT ACT WHICH IS NOT A CONSTANT "CONNECTION". |
| 01:32.03 | saftsack | clintc, polycom rocks :) i have a 430 here, but one thing is hard to understand for me .... |
| 01:32.31 | shido6 | do you have a whiteboard app JT? |
| 01:32.35 | saftsack | a pots telephone is much more cheaper. are the more functions of an ip phone worth to pay the 10x price? |
| 01:32.37 | JT | nah |
| 01:32.56 | JT | saftsack: in a business environment, yes |
| 01:32.57 | shido6 | got vnc and a little "paint" like program? |
| 01:33.06 | JT | most businesses have pabx or key system phones anyway |
| 01:33.07 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 01:33.07 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 01:33.19 | DrukenLPY | JT: in ANY environment, YES! :) |
| 01:33.46 | JT | shido6: not really either, do you really need it? :) |
| 01:33.46 | JT | DrukenLPY: hmm? |
| 01:33.53 | saftsack | can someone tell me, how much money you paid for one polycom? :) |
| 01:34.01 | puzzled | shido6: iirc this box uses this shared hardware: http://www.bicomsystems.com/products/C/P/319/255_2797/ |
| 01:34.06 | clintc | saftsack: the asterisk setup is at least 65 percent less expensive then the analog pbx it will replace |
| 01:34.24 | clintc | even with the polycom 501s :) |
| 01:34.47 | [TK]D-Fender | saftsack : For most common business use, no. Analog FXS can be just fine. For higher volume places, less so. non-analog sets have serverbased hold w/ MOH, and all the other goodies and an interface to useful functionality that is actually USEFUL. |
| 01:35.20 | puzzled | JT: for box2 taking over gracefully the ongoing calls on box1 it must be one very fast switchover |
| 01:35.56 | vvortex3 | d-fender, you rule |
| 01:36.16 | JT | puzzled: yes, a lot of telco setups run parallel parity hardware with failure detection |
| 01:36.28 | saftsack | [TK]D-Fender, ok |
| 01:36.32 | JT | or distributed processors with a message passing system |
| 01:36.42 | saftsack | clintc, this sounds reliable |
| 01:36.59 | quidpro | Hmm, is there a way to test if an extensions is sending valid caller ID (ie. just numbers)? |
| 01:37.17 | JT | quidpro: extension ==? |
| 01:37.20 | saftsack | but if you buy a new office .... so there are no cables atm. you can pay 300$ for a polycom or 20$ for an pots telephne + 60$ for a channel. |
| 01:37.26 | vvortex3 | ok so here is what i want to accomplish, maybe you can point me in the right direction... my site will create voicemails by script for a user identified by phone number, when that user calls the number, he can do various things with each message (save,delete,skip) and each should initiate a script... |
| 01:37.45 | J4k3 | good god that server machine looks like a nightmare |
| 01:37.51 | puzzled | JT: ah like the stratus and tandem non-stop stuff. when I was at Lucent I sold stratus based solutions to wcom |
| 01:37.53 | J4k3 | its complexity would be its most likely failure point. |
| 01:38.03 | DrukenLPY | where you getting a good pots telephone for 20? |
| 01:38.05 | quidpro | JT: Hmm.... sorry extension. |
| 01:38.09 | JT | puzzled: yeah |
| 01:38.20 | J4k3 | with a 90 day purchase-to-turnup schedule. |
| 01:38.23 | DrukenLPY | maybe one of those useless clam phones... |
| 01:38.28 | puzzled | JT: at the time never got into the tech details too much which shows :) |
| 01:38.34 | shido6 | how much is a box like this? |
| 01:38.47 | puzzled | shido6: many, many zero's |
| 01:39.04 | puzzled | the test box we shipped to wcom was $1M+ |
| 01:39.15 | *** join/#asterisk flenders (n=fserto@unaffiliated/flenders) |
| 01:39.24 | shido6 | for 128 calls? |
| 01:39.25 | clintc | saftsack: what is the $60 for a channel? |
| 01:39.42 | flenders | hey, has anyone used those SPA-921 phones? |
| 01:39.42 | puzzled | shido6: lol, nah, tons of them |
| 01:39.50 | saftsack | asterisk + channel(for example tdm2400p) + pots telephone |
| 01:39.53 | [TK]D-Fender | saftsack : 70$USD = 2 channels |
| 01:40.00 | puzzled | flenders: have one here. not used it in quite some time though |
| 01:40.05 | JT | quidpro: what type of extension |
| 01:40.07 | shido6 | why dont they call it "EOTS 1000 E)lbows O)n the T)able S)ir 1000" |
| 01:40.10 | quidpro | JT: SIP |
| 01:40.23 | flenders | puzzled: how do you resume calls that are on hold? |
| 01:40.30 | JT | J4k3: yeah, that's great for a small setup, telco exchanges just canNOT have the whole thing go down, ever, pretty much |
| 01:40.52 | [TK]D-Fender | vvortex3 : You are starting very backward with FreePBX. it is not meant for real custom stuff. |
| 01:40.53 | puzzled | flenders: think you can jusr press the bliking light on the right top side |
| 01:40.58 | saftsack | [TK]D-Fender, so it would be 35 + 20 -> 55USD for one person. compared to the expensive voip phones this is an alternative :) |
| 01:41.17 | flenders | puzzled: that's on the SPA-941 |
| 01:41.17 | JT | they work on a very modular architecture, if one card blows up, it affects one little bit, and can be hot swapped |
| 01:41.28 | quidpro | JT: Tried "exten => _1NXXNXXXXXX/_XXXXX.,1,NoOp" where next line sets CID if there is nothing and command 2 dials-out... but doesnt' seem to work |
| 01:41.45 | puzzled | flenders: oh sorry. then that's the one I have here. dunno about the 921 |
| 01:42.04 | flenders | puzzled: the 921 is a single line, and when I put calls on hold, I can make other calls, answer other calls, but it never shows an option to resume calls |
| 01:42.11 | flenders | puzzled: no worries then |
| 01:42.12 | [TK]D-Fender | saftsack : Yes, it is entirely an option. Then again, you have all the limitations of POTS. 2 calls max, waiting for CID, waiting for CWCIW, having to locally power the phone, lack of practical "hold", less intuitive interface. |
| 01:42.39 | JT | quidpro: why don't you do NoOp(${CALLERID(num)})? |
| 01:42.41 | saftsack | yes this is true, but it is an option ;) |
| 01:43.26 | quidpro | JT: Let me give that s hot. |
| 01:43.28 | quidpro | a shot |
| 01:43.35 | saftsack | i go to bed now. gn8 |
| 01:43.38 | vvortex3 | when you edit exentsions.conf do you restart asterisk? |
| 01:43.39 | clintc | saftsack: most of our staff will use zero cost softphones, with inexpensive headsets, but in business setting you get extensions that move with the phone |
| 01:43.42 | [TK]D-Fender | saftsack : Like I said, entirely so. I would sooner put a company on SPA-2002's and analog phones than GXP-2000's :) |
| 01:43.47 | JT | going NoOp with no argument prettu much does nothing :P |
| 01:43.52 | puzzled | shido6: a diagram of (afaict) a non-stateful failover is here: http://integrics.com/products/enswitch/guides/latest/en/system/architecture/ |
| 01:43.54 | clintc | saftsack: I was going to list more but won't keep you |
| 01:44.00 | clintc | :) |
| 01:44.09 | saftsack | ^^ |
| 01:44.28 | JT | vvortex3: restart or 'extensions reload' in the console |
| 01:44.28 | saftsack | clintc, ok but if you want to move you can get BRI phones :) |
| 01:44.34 | vvortex3 | thanks |
| 01:44.39 | saftsack | they are cheap too :) |
| 01:44.50 | clintc | call logs, shared directories, information displays, tetris! :) |
| 01:45.24 | saftsack | tetris, lol |
| 01:45.33 | quidpro | JT: Won't that just print the CallerID number... ie. exten => _1NXXNXXXXXX,1,NoOp(${CALLERID(num)}) ? |
| 01:45.37 | saftsack | i go to bed now, gn8 |
| 01:45.38 | JT | yes |
| 01:45.49 | JT | isn't that what you want to check quidpro |
| 01:46.24 | quidpro | JT: Okay, sorry I should have been more clear... I want * to put in valid callerID info if an extension sends something invalid (ie. 911). |
| 01:46.30 | *** join/#asterisk mog_home (n=rah@12-214-191-64.client.mchsi.com) |
| 01:46.30 | *** mode/#asterisk [+o mog_home] by ChanServ |
| 01:46.50 | quidpro | Ie. if invalid info is received, sub another number (ie. 12125551212) |
| 01:47.07 | JT | where would it be receiving invalid info from |
| 01:47.11 | JT | pstn? |
| 01:47.26 | shido6 | I would think the redfone box would be best for a media failure situation |
| 01:47.43 | quidpro | No, just extensions from users... they can set Caller ID.... |
| 01:47.58 | JT | yeah it's quite a good solutions, shido6, i think active calls would still be lost |
| 01:48.27 | shido6 | i'll figure it out.. |
| 01:48.43 | JT | quidpro: you'd need some pattern to determine if a number is "invalid", if so, it would then Set the callerid |
| 01:49.11 | quidpro | JT: Yeah... I was thining.. if it is less than 7 digits... then sub the example number |
| 01:50.21 | quidpro | I tried the 1NXXNXXXXXX,/_XXXXXXX trick but doesn't seem to work |
| 01:50.36 | quidpro | err, minus that comma |
| 01:51.14 | JT | _X. may |
| 01:51.25 | JT | <PROTECTED> |
| 01:51.37 | JT | have a pattern for valid callerid too |
| 01:51.46 | quidpro | hmmm, let me try |
| 01:52.51 | orkid | so umm, can i get a DID phone number in canada and hook it up to an asterisk box on the net?! |
| 01:52.51 | orkid | is htat possible? |
| 01:53.22 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 01:53.40 | JT | yes |
| 01:53.55 | orkid | awesome! know of any providers? |
| 01:54.40 | *** join/#asterisk J4k3 (i=jsuter@dhcp-12-197-128-58.intrastar.net) |
| 01:54.46 | shido6 | nufone |
| 01:54.48 | orkid | nvm, voip-info.org is helping :) |
| 01:54.50 | orkid | thx |
| 01:55.19 | shido6 | what province? |
| 01:56.56 | orkid | ON |
| 01:57.22 | orkid | seems like all places offer some minutes with their plans.... can't you just get a number and unlimited incoming calls? |
| 01:57.39 | shido6 | yes. |
| 01:57.49 | orkid | really. where? |
| 01:57.53 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 01:58.12 | orkid | hey putzz ! |
| 01:58.23 | putzz | sup? |
| 01:58.31 | orkid | nm |
| 01:58.41 | orkid | reading the * book |
| 01:58.47 | putzz | right on |
| 01:58.52 | putzz | o'reilly? |
| 01:59.42 | orkid | yup |
| 01:59.56 | orkid | still looking for a cheap fxo/fxs :) |
| 02:00.05 | orkid | will install * before i get hardware prolly though |
| 02:00.05 | putzz | ebay! |
| 02:00.18 | orkid | yeah. ebay doesn't have many. how does hsipping to canada look anyway (from the states) |
| 02:00.29 | putzz | u havent tried * yet? |
| 02:00.39 | orkid | no i haven't. want to read some docs first |
| 02:03.01 | The_DoC^ | I collected all of my pap2's from unhappy vonage customers |
| 02:03.25 | marc\cba | why were they unhappy? |
| 02:03.36 | JT | because it was vonage |
| 02:03.42 | putzz | lol |
| 02:03.48 | *** join/#asterisk fall0ut (i=tim@realfuckingnews.com) |
| 02:03.56 | marc\cba | i've never used them |
| 02:04.04 | The_DoC^ | cost and the Cable internut provider her sucks do they were getting dropped calls |
| 02:04.18 | The_DoC^ | her = here |
| 02:04.26 | marc\cba | /where is here? |
| 02:04.30 | The_DoC^ | do = so |
| 02:04.58 | orkid | want to send one over The_DoC^ ? |
| 02:04.58 | orkid | :) |
| 02:05.44 | The_DoC^ | orkid, all in service, well testing |
| 02:06.13 | orkid | :( for me |
| 02:06.17 | orkid | :| for you |
| 02:06.57 | *** join/#asterisk s0lid (n=jlq@210.213.199.9) |
| 02:08.52 | shido6 | can you crack them The_DoC^ ? |
| 02:09.48 | The_DoC^ | shido6 : you mean unlock? |
| 02:09.53 | shido6 | yes |
| 02:10.03 | The_DoC^ | yea |
| 02:10.11 | shido6 | how long does it take? |
| 02:10.35 | putzz | unlockig it is easy |
| 02:10.41 | The_DoC^ | took me less than 5 minutes after I spent 2 hours unlocking a rt31p2 |
| 02:10.44 | orkid | anyone have hardware sent from US to Canada? |
| 02:10.58 | orkid | say from ebay? how much are the border costs? |
| 02:11.12 | putzz | orkid I ordered all my stuff from ebay shipped thru ups wich charges u fees |
| 02:11.22 | putzz | I paid 12 dollars in duties |
| 02:11.31 | orkid | plus brokerage? |
| 02:11.33 | J4k3 | haha, you guys have to pay to be canadian |
| 02:11.36 | J4k3 | ;) |
| 02:11.39 | putzz | heh |
| 02:11.44 | putzz | only ups does that |
| 02:11.48 | orkid | at least we're kinda free :) |
| 02:11.51 | *** part/#asterisk clintc (n=clintc@128.227.123.25) |
| 02:11.55 | putzz | if u order with fedex and ...u dont pay |
| 02:12.05 | orkid | really, how about usps? |
| 02:12.59 | putzz | ups charges u duties and.... |
| 02:13.26 | J4k3 | heh |
| 02:13.31 | orkid | how about us postal service though. |
| 02:13.34 | J4k3 | ups charges extra to deliver the box without beating the crap out of it |
| 02:13.38 | J4k3 | its called "package hockey tax" |
| 02:13.39 | orkid | lol |
| 02:13.45 | putzz | lol |
| 02:14.08 | orkid | yeah, the consensus generally is that ups sucks, and fedex is expensive but good |
| 02:14.16 | orkid | usps might not have brokerage fees, but not sure. |
| 02:14.19 | J4k3 | the winners of the package hockey championship every quarter gets to take any new/available routes. |
| 02:14.24 | JT | usps is best for overseas |
| 02:14.25 | J4k3 | so they can move up from loader to drive. |
| 02:14.27 | J4k3 | er driver. |
| 02:14.32 | putzz | UPS charges u for everything |
| 02:14.50 | JT | ups is amazingly mind blowingly expensive to send outside us/canada |
| 02:15.35 | J4k3 | its getting to the point I prefer usps to ups... at least ups manages to get packages delivered before 7pm |
| 02:15.43 | J4k3 | our driver's route is like 140 miles... totally insanely long |
| 02:16.02 | *** join/#asterisk emiquelito (n=evandro@c934ebc2.virtua.com.br) |
| 02:16.14 | J4k3 | err usps manages to get packages delivered before 7pm |
| 02:16.14 | *** join/#asterisk flashnet (i=flashnet@17.15.185.213.dk-hvi.res.sta.perspektivbredband.net) |
| 02:16.56 | *** join/#asterisk dj-fu (n=deejay@203-167-190-166.dsl.clear.net.nz) |
| 02:18.14 | dj-fu | alright - got the card installed in my machine. Stamed on the chips and the daughterboards: TDM400 rev g, tiger 320, two green, two red |
| 02:18.20 | dj-fu | any ideas what module that'd be? |
| 02:18.38 | The_DoC^ | 2 fxo and 2 fxs |
| 02:19.07 | orkid | how much did you pay dj-fu |
| 02:19.08 | orkid | ? |
| 02:21.27 | *** join/#asterisk orlock (i=jwr@202.44.174.4.static.nexnet.net.au) |
| 02:21.43 | orlock | Does anybody here have any experience in debugging voip quality issues? |
| 02:21.54 | orlock | using pure SIP/RTP |
| 02:22.34 | JT | you've been asking for months now :P |
| 02:22.55 | dj-fu | orkid, no idea - it was here when I got here. I'm just trying to fix it |
| 02:23.06 | dj-fu | The_DoC^, I meant kernel module |
| 02:23.10 | orkid | :) |
| 02:23.10 | orlock | JT: yes, i know, and havent been able to actually get a reasonable response from anybody |
| 02:23.15 | orkid | work system dj-fu ? |
| 02:23.18 | JT | orlock: is it voip to your provider? |
| 02:23.23 | orlock | JT: yup |
| 02:23.30 | JT | orlock: the answer might be "pray" for most people |
| 02:23.31 | dj-fu | orkid, yeah |
| 02:23.34 | JT | get them to look into it |
| 02:23.35 | orkid | lol |
| 02:23.39 | orkid | lol, orkid |
| 02:23.44 | orkid | err, lol JT |
| 02:23.49 | dj-fu | strange - udev managed to pick up the card and modprobed zaptel and wctdm |
| 02:23.55 | orlock | yeah, they have been a great help |
| 02:24.15 | orkid | i sense sarcasm! (am i righrt?) |
| 02:24.28 | orlock | no actually |
| 02:24.31 | orlock | they have been |
| 02:24.37 | *** join/#asterisk olsen (n=diego@200.61.236.33) |
| 02:24.37 | J4k3 | JT: suggestion... buy a good hardphone, set it up to bypass your * and talk directly to the provider. Does the problem still occur? If yes, contact the provider, figure out where the problem is... if no, then its your * config or something |
| 02:24.40 | dj-fu | doh, forgot to plug in the power. /me goes back into the closet |
| 02:24.43 | J4k3 | errr |
| 02:24.45 | J4k3 | orlock: |
| 02:24.54 | orlock | J4k3: yeah |
| 02:25.00 | orlock | J4k3: best suggestion i have heard so far |
| 02:25.06 | orlock | thats probably going to be my next step |
| 02:25.16 | *** join/#asterisk erickperez (n=lolo@200.75.246.173) |
| 02:25.38 | erickperez | Hi there, quick question. What does the module zttranscode do? |
| 02:26.54 | dj-fu | brb, again. |
| 02:29.54 | orlock | J4k3: they are on ADSL, and its incoming audio they are having problems with |
| 02:30.07 | orlock | its a pain.. without QoS, you get told to implement QoS |
| 02:30.18 | ChrisN | I have a context which answers, runs an AGI, and hangs up. I have an h ext which runs a deadagi on hangup. The AGI calls the meetme application to connect the user to a conference. If the user hangs up from within the meetme, control is not returned to the AGI and the AGI exits with status -1. The channel hangs up and the h ext is not called. Any way to either get the AGI to continue or cause the h ext to be called? |
| 02:30.25 | orlock | with QoS, you dont know if the QoS is interfering with the latency on your ICMP packets for monitoring... |
| 02:31.38 | erickperez | Hi there, quick question. What does the module zttranscode do? |
| 02:32.18 | JT | i swear i saw that question 5min ago |
| 02:33.11 | Grnd-Wire | JT: Did you see it get answered though? :P |
| 02:33.11 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 02:33.11 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 02:33.23 | JT | Grnd-Wire: no :) |
| 02:33.57 | erickperez | JT: I just saw that binding to wctdm module...just asking... |
| 02:34.16 | JT | erickperez: i have no idea, never seen it before, and i guess no one else active has either |
| 02:35.55 | erickperez | oh, ok...thanks |
| 02:39.14 | orkid | erickperez, transcoding on a zt device? :) |
| 02:40.55 | orkid | BASTA! asterisk is masked in gentoo-x86_64 |
| 02:41.46 | toresbe | I have the solution, orkid |
| 02:41.58 | toresbe | Don't use gentoo. |
| 02:42.16 | orkid | gentoo is a good distro though toresbe |
| 02:42.56 | toresbe | no it isn't... it's awful in pretty much every way |
| 02:43.11 | orkid | echo "net-misc/asterisk" >> /etc/portage/package.keywords fixed that problem |
| 02:43.15 | orkid | gentoo is great! i love it. |
| 02:43.25 | litage | how long (cm, inches?) is the TDM2400? |
| 02:43.35 | toresbe | orkid: May I ask you something? |
| 02:43.44 | orkid | sure |
| 02:43.45 | toresbe | orkid: How long did it take you to install asterisk? |
| 02:43.53 | orkid | i haven't installed it yet. |
| 02:43.59 | toresbe | Heh. |
| 02:44.07 | *** join/#asterisk hohum (n=dcorbe@c-71-62-76-68.hsd1.va.comcast.net) |
| 02:45.03 | *** join/#asterisk oej (n=olle@nat.sigmasoft.com) |
| 02:45.31 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 02:48.46 | *** join/#asterisk ping2921 (n=marc3234@206-248-130-67.dsl.teksavvy.com) |
| 02:49.51 | *** join/#asterisk dj-fu (n=deejay@203-167-190-166.dsl.clear.net.nz) |
| 02:50.10 | Juggie | oej! |
| 02:50.54 | oej | Yay |
| 02:53.35 | Juggie | not watching the superbowl? :P |
| 02:53.53 | oej | I am, but not very interested. Friends forcing me |
| 02:54.09 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 02:54.54 | orkid | toresbe, just installed asterisk. it took me about 5 minutes to compile. |
| 02:55.11 | orkid | toresbe, and it's running now using /usr/sbin/safe_asterisk |
| 02:55.16 | orkid | toresbe, i havne't configured it yet |
| 02:55.30 | JT | litage: not sure, but it's basically a full length pci card |
| 02:55.36 | JT | and you can look up the length of that |
| 02:55.53 | JT | litage: will only fit in a server case or large desktop case |
| 02:56.12 | JT | maybe 10inches? |
| 02:57.09 | erickperez | Some ideas/pointers as to what the cause of this warning? |
| 02:57.11 | erickperez | WARNING[27249]: chan_zap.c:6299 ss_thread: CallerID returned with error on channel 'Zap/4-1' |
| 02:57.43 | erickperez | my tdm400p (fxo) is connected to an avaya system, via loopstart. callerid=asreceived. |
| 03:00.49 | Juggie | oej, i dont blame you... football sucks :) |
| 03:01.20 | Corydon76-home | erickperez: generally that means that the Caller*ID signal was not readable |
| 03:01.43 | bkruse_home | Corydon76-home: when did you leave? |
| 03:01.48 | *** join/#asterisk Putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 03:01.54 | Corydon76-home | bkruse_home: yesterday afternoon |
| 03:02.01 | bkruse_home | Corydon76-home: :( |
| 03:02.03 | erickperez | Corydon76: Understood. Can it be related to the avaya side not passing CID properly? |
| 03:02.26 | Corydon76-home | bkruse_home: sorry that you didn't have a chance to make out with me. We can discuss it after you turn 18. :-P |
| 03:02.36 | bkruse_home | Corydon76-home: LOL |
| 03:02.44 | Corydon76-home | erickperez: could be any number of things |
| 03:02.48 | booray | woah, woah |
| 03:03.52 | ChrisN | ls |
| 03:03.58 | erickperez | ok |
| 03:05.04 | Corydon76-home | booray: something wrong? |
| 03:07.59 | *** join/#asterisk infernix (n=nix@spirit.infernix.net) |
| 03:11.47 | orkid | where can one obtain the digium dev-lite kit? i can't find any info on it on digium.com |
| 03:11.49 | dj-fu | Anyone tell me how I can tell which one of my ports on the back of my TDM400P are the fxo's? |
| 03:12.03 | dj-fu | it's got two fxo's, two of the other ones |
| 03:12.13 | dj-fu | actually forget that. Which side is 0 |
| 03:12.28 | dj-fu | I know which one are the fxo's by dmesg - just need to work out what whole corresponds to what number xD |
| 03:12.50 | *** part/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 03:13.04 | orkid | dj-fu, not written on the PCB anywhere? |
| 03:14.38 | catpants | i believe 0 starts on the side opposite the pci connector |
| 03:15.02 | dj-fu | so 0 is topside, 4 is bottom? |
| 03:15.05 | dj-fu | err, 3 is bottom |
| 03:15.44 | catpants | depends on how it is mounted :P but that is probably right |
| 03:16.29 | dj-fu | bottom being closest to motherboard |
| 03:16.42 | catpants | yeah |
| 03:17.05 | Putzz | hmm |
| 03:23.19 | flenders | is there a way to lower the time asterisk takes to learn the echo cancel thing? |
| 03:27.30 | *** join/#asterisk infernix (n=nix@spirit.infernix.net) |
| 03:27.33 | dj-fu | any idea what's the best codec for call quality? |
| 03:27.41 | sbingner | flenders, enable echo training? |
| 03:27.42 | dj-fu | my sip phones are going straight out to PSTN |
| 03:27.52 | sbingner | dj-fu, ulaw or alaw |
| 03:28.01 | dj-fu | ok |
| 03:28.16 | dj-fu | what about g711u? |
| 03:28.22 | dj-fu | I can't see ulaw or alaw in these phones config |
| 03:28.45 | sbingner | g711u = ulaw |
| 03:29.17 | dj-fu | ah, wicked! |
| 03:30.39 | orlock | gddamn |
| 03:30.50 | De_Mon | I'm tryin to get asterisk talking to exchange 2007 using openser but people are usualy dead in #openser... http://pastebin.ca/338775 shows 2 calls, one that works and one thats going thru openser that doesnt. any openser.cfg suggestions? |
| 03:31.46 | *** join/#asterisk VivekKhera (n=vivek@c-69-140-110-248.hsd1.md.comcast.net) |
| 03:32.18 | Snake-Eyes | I'm not dead ! |
| 03:32.53 | De_Mon | you're not in #openser either |
| 03:32.56 | De_Mon | <_< |
| 03:33.06 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 03:33.07 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 03:33.32 | orkid | what;'s the diff b/w sipv1 and SPIv2? |
| 03:33.36 | orkid | SIPv2 that is |
| 03:33.55 | De_Mon | a whitepaper |
| 03:33.57 | Snake-Eyes | De_Mon, mis-read thought you said #ser :P |
| 03:34.27 | stealth | an FX0 is what expects a dial tone? and a FXS is what creates a dial tone |
| 03:34.42 | JT | dj-fu: use alaw, you live in an alaw country |
| 03:34.48 | De_Mon | isn't ser kinda dead? no more active development and all |
| 03:35.10 | JT | stealth: yes |
| 03:35.24 | Snake-Eyes | De_Mon, no, version 0.10 coming soon |
| 03:35.42 | JT | damn americans always recommending everyone use ulaw ;) |
| 03:35.53 | JT | how does openser compare to sipX? |
| 03:35.56 | De_Mon | hehe |
| 03:36.02 | Snake-Eyes | lol JT |
| 03:36.03 | De_Mon | sipx is a pbx openser is a proxy |
| 03:36.14 | JT | sipx is a proxy |
| 03:36.37 | JT | well |
| 03:36.42 | JT | it does have more pbx functions |
| 03:36.44 | JT | but still |
| 03:36.44 | De_Mon | not according to sipfoundry |
| 03:36.59 | De_Mon | We think sipX is the leading open source IP PBX in terms of scalability, robustness and ease of use. The sipX IP PBX has been success.... |
| 03:37.19 | JT | was reading the sipX site, it's designed to be a high performance proxy, but also has some sip pbx functionality |
| 03:37.29 | J4k3 | yay... isn't this #asterisk? :o |
| 03:37.30 | JT | it's definately more a sip proxy than asterisk |
| 03:37.44 | JT | yeah, it is |
| 03:38.04 | JT | they don't do the same thing as asterisk anyway |
| 03:38.11 | JT | sipx is sip ONLY |
| 03:38.13 | dj-fu | JT, sorry, what is the g711asdfladksjfl thing for alaw? |
| 03:38.21 | JT | G.711a |
| 03:38.36 | dj-fu | ah super |
| 03:38.56 | orkid | wow, all this hardware is so expensive. |
| 03:39.11 | orkid | not so widespread as regular computer hardware maybe that's why |
| 03:39.12 | JT | aparently making proper telephony hardware isn't cheap :P |
| 03:39.16 | JT | atas are the cheapest |
| 03:39.32 | J4k3 | define "expensive" |
| 03:39.35 | J4k3 | it all seems rather cheap to me |
| 03:40.29 | dj-fu | JT, is ulaw illegal or something, what do you mean by I'm in an alaw country |
| 03:40.38 | Snake-Eyes | lol |
| 03:40.58 | Snake-Eyes | you had that coming, JT :P |
| 03:41.01 | J4k3 | woop woop its the alaw police |
| 03:41.08 | Qwell | dj-fu: They only really use ulaw in North America |
| 03:41.14 | JT | dj-fu: it's just non optimal, as the phone networks are all running one standard |
| 03:41.21 | dj-fu | ah, I see :) |
| 03:41.22 | JT | and japan :) |
| 03:41.38 | Qwell | They probably use jlaw |
| 03:41.49 | Qwell | just different enough to require another standard :p |
| 03:42.01 | JT | he he, nah it is ulaw, but yeah weird pri signalling :) |
| 03:47.41 | weazahl | a requirement for a feature on asterisk is NBS, i cannont find this anywhere. where can i find it? |
| 03:47.59 | J4k3 | and china uses g.communist |
| 03:48.12 | Grnd-Wire | J4k3: haha |
| 03:48.32 | Putzz | can someone help me I want to make a small script that when I get the message "Trunk is reporting BUSY" in console it will email me ? |
| 03:49.22 | [TK]D-Fender | weazahl : Explain "NBS" .... |
| 03:49.47 | weazahl | that is all its says, requires NBS |
| 03:49.59 | JT | weazahl: i guess you'll have to find out what it is |
| 03:50.00 | [TK]D-Fender | Putzz : Just do a System call in your dialplan based on the dialstatus |
| 03:50.12 | [TK]D-Fender | weazahl : And NBS stands for..... |
| 03:50.39 | file | [TK]D-Fender: hello be to you |
| 03:50.44 | weazahl | ahh Network Broadcast Sound Support |
| 03:51.02 | Qwell | there was a chan_nbs... |
| 03:51.03 | [TK]D-Fender | file : Bah weep grah na weep nini bong! |
| 03:51.17 | Qwell | s/was/is/ |
| 03:51.54 | litage | is it problematic to put 2 or more TDM400Ps in the same box? |
| 03:52.15 | [TK]D-Fender | litage : Passing 2 is considered quite unrecommended. |
| 03:52.32 | litage | [TK]D-Fender: "passing"? |
| 03:52.39 | [TK]D-Fender | litage : And using more than 8 FXO is generally considered silly in lieu of partial PRI |
| 03:52.55 | [TK]D-Fender | litage : 2 = OK, 3 = think again... |
| 03:53.06 | litage | ah |
| 03:53.16 | Grnd-Wire | [TK]D-Fender: Why does Rhino make an 8 port device then? |
| 03:53.21 | [TK]D-Fender | litage : digium writes as much in all their data sheets. You should have read them all by now :) |
| 03:53.50 | litage | [TK]D-Fender: i haven't found the blurb on their site discussing multiple TDM cards |
| 03:53.50 | [TK]D-Fender | Grnd-Wire : Is 8 > 8? |
| 03:54.18 | [TK]D-Fender | litage : Digium reccommends no more than 2 TDM/digital cards or any type in a system |
| 03:55.48 | [TK]D-Fender | Grnd-Wire : and thats 8 ports on *1* card. The question is that PRI offers much greater functionality over analog, and the costs become nearly identical aroun 10 channels. |
| 03:55.48 | litage | [TK]D-Fender: thanks. do you by chance have the url to digium's page discussing that? if not, i'll root around and find it |
| 03:55.48 | Grnd-Wire | [TK]D-Fender: I'm referring to the 3=think again part.. Digium makes a 4port card, and Rhino makes an 8.. Are there performance reasons why you don't want to use them like that? |
| 03:55.48 | orkid | if you're still trying to figure out with ports are for which doughtercards dj-fu check out the asterisk book acrobat reader page 79 (there's a nice pic there) |
| 03:55.48 | [TK]D-Fender | litage : No, you'll have to hunt around for it, but its mentioned all over the plce. |
| 03:55.48 | litage | thanks |
| 03:55.48 | [TK]D-Fender | Grnd-Wire : You are thinking way out of context. |
| 03:56.20 | vvortex3 | hmm...asterisk@home i have calls coming in finally but by default it goes to the macro-dial context which always just says "Busy"...well, if i put my own stuff in macro-dial it works but i dont see why its even going to that context |
| 03:56.20 | *** join/#asterisk rue_mohr (n=rue@h24-207-91-121.cst.dccnet.com) |
| 03:56.20 | Grnd-Wire | [TK]D-Fender: You're saying that I should be analyzing why I'v got that sort of port count, and I'm NOT using a T1 interface.. right? |
| 03:56.20 | [TK]D-Fender | Grnd-Wire : I'm saying why would you want to HAVE 12 analog lines when a partial PRI may cost about the same and offer so much more and have LESS wiring? |
| 03:56.22 | Grnd-Wire | [TK]D-Fender: Agreed! |
| 03:56.35 | [TK]D-Fender | Grnd-Wire : Good... |
| 03:56.56 | Grnd-Wire | [TK]D-Fender: I was combining two conversations.. heh.. |
| 03:56.59 | [TK]D-Fender | And Rhino's cards are 1-dimensional crap which should be avoided. |
| 03:57.02 | Grnd-Wire | ok - I'm outta here.. |
| 03:57.08 | [TK]D-Fender | Grnd-Wire : And getting neither quite right :) |
| 03:57.13 | Grnd-Wire | Duly noted.. |
| 03:57.26 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 03:57.26 | *** mode/#asterisk [+o russellb] by ChanServ |
| 03:57.48 | file | omg it's russellb |
| 03:57.49 | dj-fu | w00t! |
| 03:57.53 | dj-fu | I just called 705 from 701! |
| 03:57.59 | dj-fu | I _am_ the man! |
| 03:58.04 | russellb | yay phones |
| 03:58.11 | file | russellb: ltns! |
| 03:58.17 | russellb | i like making phones ring ... as long as it's not fo rme |
| 03:58.46 | vvortex3 | yay, my asterisk@home reads numbers to me |
| 03:58.55 | dj-fu | a@h sucks! |
| 03:59.12 | vvortex3 | yay! |
| 03:59.36 | dj-fu | ok - I can't find this anymore |
| 03:59.43 | dj-fu | I've got extensions set up so I can call one another in the offices |
| 03:59.44 | vvortex3 | mmm...gf stopped by just to bring me carls junior |
| 03:59.46 | dj-fu | but how the hell can I call out |
| 04:00.01 | file | I'm having CAT5 issues :( |
| 04:00.12 | russellb | stupid broken connector |
| 04:00.17 | vvortex3 | broken cat? |
| 04:00.19 | russellb | why is your wifi being lame? |
| 04:00.23 | [TK]D-Fender | Mew. |
| 04:00.44 | vvortex3 | asterisk can make my website do fun things |
| 04:01.04 | russellb | file: are you still alive? |
| 04:01.19 | russellb | uh oh. new cable was a failure. |
| 04:01.30 | [TK]D-Fender | vvortex3 : * used to make me coffe, and let in my guests |
| 04:01.38 | vvortex3 | hahaha that owns |
| 04:01.41 | shido6 | extension 666 is for the remote toilet flush |
| 04:02.02 | [TK]D-Fender | shido6 : Thats the one thing i swore I'd never be so lazy as to implement :) |
| 04:02.05 | vvortex3 | extension 666 activates my dog's shock collar |
| 04:02.11 | shido6 | had to do it. |
| 04:02.22 | file | russellb: unknown |
| 04:02.26 | [TK]D-Fender | shido6 : And on my consulted systems I typically leave them "666" as a way to dial me direct via SIP :) |
| 04:02.36 | file | russellb: I should steal er I mean borrow kpfleming's atheros minipci card ^_^ |
| 04:03.18 | *** join/#asterisk topping (n=topping@204.152.96.238) |
| 04:03.19 | rue_mohr | I seem to be having firefox issues, can someone tell me min sys requirements for an asterisk box with a T1 card? |
| 04:03.29 | *** join/#asterisk thoughtpolice (n=austin@ip70-185-140-61.lu.dl.cox.net) |
| 04:03.31 | Qwell | rue_mohr: it all depends |
| 04:03.41 | rue_mohr | ok, give me a ballpark |
| 04:04.27 | rue_mohr | hmm 0% free space on / maybe its not a firefox issue... |
| 04:07.29 | *** join/#asterisk wnfaknd (n=bleh@cpe-24-30-183-141.socal.res.rr.com) |
| 04:08.00 | orkid | sooo, i hear that the old x101p was an intel/alliant chip (md3200?) and that the x100p was a motorola? or something like that |
| 04:09.06 | *** join/#asterisk jcast2 (n=jcast@adsl-76-211-176-145.dsl.wlfrct.sbcglobal.net) |
| 04:09.18 | rue_mohr | Qwell, ok, well, how would a PII-300 fair do you think? |
| 04:09.37 | Qwell | rue_mohr: should be okay for a PRI, if it's doing nothing else |
| 04:09.52 | Qwell | maybe |
| 04:09.54 | rue_mohr | 1 line |
| 04:10.01 | rue_mohr | ~4 phones |
| 04:10.16 | dj-fu | thanks for your help today guys, I may be back with more stupid questions later - hometime now |
| 04:10.37 | rue_mohr | newbridge 3624 |
| 04:11.14 | rue_mohr | need to do ivr and voicemail |
| 04:13.57 | wnfaknd | anyone familiar with vicidial? |
| 04:16.36 | jcast2 | Hello all! Seems like a good spot to "interrupt." I have been reading up and getting pretty excited about getting my home PBX up and running. I am trying to figure out the most cost-effective way to go about it. |
| 04:17.09 | jcast2 | I'll only have one or two connections to outside landlines, but I want to be able to hook up several analog phones within the house. |
| 04:17.50 | *** join/#asterisk Putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 04:17.59 | JT | on one or independant extensions, jcast2 ? |
| 04:18.03 | jcast2 | I've looked at the TDM400P/2400P cards... |
| 04:18.17 | Qwell | jcast2: the TDM800P should be selling quite soon |
| 04:18.35 | jcast2 | I'd like each phone to have its own, independent extension. (It would be fun.) |
| 04:18.37 | JT | potentially overkill for home |
| 04:18.48 | JT | the tdm800p fully populated would not be cheap |
| 04:20.13 | *** join/#asterisk k-man_ (n=jason@unaffiliated/k-man) |
| 04:20.14 | jcast2 | I willing to spend some money, but wondered if it was "better" to use a couple of TDM400 cards or one 2400 card. |
| 04:20.29 | flenders | sbinger: sorry mate, had to step out for a bit... yes, echo training |
| 04:20.40 | JT | tdm2400p would be "better" |
| 04:20.49 | JT | tdm800p would be an option too |
| 04:20.58 | JT | you could also use a pile of ATAs |
| 04:20.58 | JT | also |
| 04:21.13 | JT | is there any reason why you want analogue extensions instead of sip? |
| 04:21.46 | orlock | Hmm. |
| 04:21.50 | *** join/#asterisk hito (n=angelsz@202.59.70.162) |
| 04:22.08 | jcast2 | JT, this will surely demonstrate my lack of knowledge in the area, but is it possible to use the t1 card (TE110p I think) to get 24 channels internally? |
| 04:22.24 | hito | checking for ptlib-config... no |
| 04:22.24 | hito | Cannot find ptlib-config - please install and try again |
| 04:22.25 | hito | help? |
| 04:22.40 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 04:23.02 | JT | jcast2: yes, if coupled with a channel bank |
| 04:23.33 | jcast2 | JT, I want analog phones simply becase I have so many of them (I collect old phones). I'll eventually get some IP phones too. |
| 04:23.39 | orlock | JT: reckon that having some servies running on an asterisk system could cause call quality issues, even if its not actually utilising them? |
| 04:23.39 | *** part/#asterisk a1fa (n=a1fa@unaffiliated/a1fa) |
| 04:23.41 | JT | ah right |
| 04:23.53 | JT | i have a t1 card and channel bank at home |
| 04:23.59 | JT | they were cheap :) |
| 04:24.22 | JT | orlock: possibly, but are there any load spikes? |
| 04:25.43 | *** join/#asterisk Putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 04:26.30 | *** part/#asterisk Rusty1 (n=Rusty1@cpe-72-226-96-74.nycap.res.rr.com) |
| 04:26.38 | jcast2 | JT, you do? Neat. I was thinking it might be cheaper this way (using T1 and channel bank) than using the TDM400/2400p cards with modules, but wasn't sure. I saw the T1 card while browsing the web, but not the channel bank so couldn't make an accurate guess as to the cost. |
| 04:27.34 | JT | jcast2: ah right, well t1 channel bank prices vary wildly |
| 04:27.41 | JT | second hand is obviously cheaper |
| 04:27.48 | Qwell | moreso if you want a modular channelbank |
| 04:27.56 | JT | adtran and adit are good brands |
| 04:28.00 | JT | but cost a bit |
| 04:28.28 | JT | i've got a little channel bank that's no longer made but can be got somtimes new for USD$120 called a Zhone Zplex 10 |
| 04:28.36 | jcast2 | So I'm guessing, at this point, that nothing is quite cheap. :-) |
| 04:28.38 | litage | does the TDM2400P come with a breakout box and connecting cable? |
| 04:28.38 | JT | it does the job, but fxo callerid doesn't work |
| 04:28.55 | Qwell | litage: I don't think so, but it would be wise to call and ask sales |
| 04:29.11 | Qwell | I've never bought one, so... |
| 04:29.30 | litage | thanks Qwell |
| 04:29.41 | orlock | JT: nope, none to explain the constant issues |
| 04:29.53 | orlock | JT: they are all 1.6Ghz+ machines, gig of ram, not hitting swap at all |
| 04:30.04 | orlock | and it doesnt matter how manu concurrent calls |
| 04:30.11 | orlock | just about to take Asterisk out the equation |
| 04:30.46 | JT | hmm |
| 04:30.53 | JT | what does it sound like? |
| 04:31.12 | orlock | JT: dropouts aparently |
| 04:32.44 | jcast2 | Thanks, Qwell and JT for your assistance... |
| 04:33.06 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 04:33.07 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 04:34.14 | hito | i'm missing ptlib-config on rh9, any suggestions? |
| 04:38.47 | hito | oops my bad |
| 04:40.56 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 04:42.11 | *** join/#asterisk dseeb_ (n=dcb@58.165.245.205) |
| 04:43.51 | vvortex3 | i did a yum install festival and then ran it ./festival & ...and whenever i use Festival in extensions i get "unbound variable : tts_textasterisk" |
| 04:46.10 | *** join/#asterisk flying_Luck (n=melifaro@ppp85-141-152-200.pppoe.mtu-net.ru) |
| 04:46.29 | *** join/#asterisk a1fa (n=a1fa@unaffiliated/a1fa) |
| 04:48.47 | weazahl | why wont configure find the libs for odbc or sqlite on fc6? they are installed. do i need to delete a file tomake configure redetect |
| 04:48.56 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 04:49.41 | weazahl | i guess a make clean would be a god idea |
| 04:49.44 | weazahl | good |
| 04:50.59 | *** join/#asterisk sudhir492 (n=sudhir@c-71-63-59-45.hsd1.va.comcast.net) |
| 04:51.10 | sudhir492 | Hi all |
| 04:51.46 | Putzz | . |
| 04:55.26 | Putzz | how can I make a dialplan that will read "1*CLI> Everyone is busy/congested at this time |
| 04:55.33 | Putzz | and email me? |
| 04:56.09 | Putzz | is there a easy way other then grepping stuff from log? |
| 04:56.35 | sudhir492 | what are you trying to do? |
| 04:56.54 | weazahl | could it be that sqlite3 is installed not sqlite causing a problem? also openodbc is installed and it wont find it. |
| 04:56.56 | Putzz | I want so when I dial a # if the trunk gives me busy/congested I get a report on my email |
| 04:57.59 | sudhir492 | exit |
| 04:58.16 | *** join/#asterisk osiris (n=osiris@c-71-205-27-131.hsd1.mi.comcast.net) |
| 04:59.18 | *** join/#asterisk osas (n=nnnnnnnn@CABLE-72-53-75-252.cia.com) |
| 04:59.35 | Putzz | anybody got any ideas for me? |
| 05:01.22 | *** join/#asterisk dj-fu (n=deejay@202-89-157-198.dyn-dsl.ubs.qsi.net.nz) |
| 05:01.48 | Putzz | nobody got any ideas? |
| 05:07.01 | *** join/#asterisk juice (n=juice@mo-76-0-47-34.dhcp.embarqhsd.net) |
| 05:10.32 | orkid | hmmm, is rockwell a supported chipset? : http://www.voip-info.org/wiki/view/Asterisk+Slimming |
| 05:10.35 | orkid | and search for rockwell |
| 05:12.22 | Putzz | chan_modem_aopen is the A/Open (Rockwell) analog voice modem driver |
| 05:12.22 | Putzz | (yuck!) |
| 05:12.24 | Putzz | lol |
| 05:14.42 | orkid | yeah. |
| 05:15.02 | orkid | i'm looking through asterisk.user lists now, seems like people have gotten rockwell modems to work on PSTN but half-duplex |
| 05:15.10 | orkid | or 'one way' rather |
| 05:15.15 | [TK]D-Fender | orkid : Stop asking for pain using no longer supported equipment. |
| 05:15.25 | [TK]D-Fender | orkid : Chan_modem was dropped in 1.4 IIRC. |
| 05:15.42 | [TK]D-Fender | orkid : Realize that * & VoIP isn't all about fetting everything for free |
| 05:15.47 | orkid | :) |
| 05:16.30 | [TK]D-Fender | Putzz : Do it in your dialplan. make a System call to whatever script you want once DIALSTATUS returns congestion. |
| 05:20.41 | orkid | [TK]D-Fender, yes I know. |
| 05:26.17 | aptura | TK, I understand that * now interfaces with cell phones with its code. So there is some kind of RF pci hardware cards that plugs into the slot? |
| 05:27.16 | aptura | okay disregard. |
| 05:27.28 | aptura | not what I consider a chice product with celliax |
| 05:28.07 | JT | yes you can buy gsm cards |
| 05:28.48 | aptura | I see |
| 05:28.55 | dj-fu | Hi, I've managed to get my extensions going to call between the phones in my office, I can't seem to work out how to call out |
| 05:29.00 | dj-fu | there's nothing in the tutorial I wa sfollowing, either |
| 05:29.03 | JT | junghanns.net for example |
| 05:29.06 | dj-fu | can anyone point me in the right direction? |
| 05:29.52 | *** join/#asterisk dj-fu (n=deejay@202-89-157-198.dyn-dsl.ubs.qsi.net.nz) |
| 05:30.13 | JT | dj-fu: yes, you could make it require a prefix to dial out, something like: _0XXXXXXXX,1,Dial(Zap/g1/${EXTEN:1}) |
| 05:30.23 | k-man_ | what does the "trunk name" setting do in sip.conf? |
| 05:30.26 | JT | that would use a prefix of 0 to dial out |
| 05:30.35 | JT | and call over zap group 1 |
| 05:30.38 | dj-fu | JT, I'll probably have to start by configuring a Zap first I suppose |
| 05:30.44 | JT | dj-fu: indeed |
| 05:30.45 | aptura | sweet |
| 05:30.51 | dj-fu | and where would I do that? |
| 05:31.01 | JT | dj-fu: what is this tutorial, is is the book? |
| 05:31.07 | JT | dialplan, extensions.conf |
| 05:31.09 | aptura | I can see this being used in a remote location. |
| 05:31.22 | dj-fu | JT, a tutorial on gentoo-wiki.com |
| 05:32.08 | JT | oh |
| 05:32.12 | JT | dj-fu: look at the book |
| 05:32.13 | JT | ~thebook |
| 05:32.26 | jbot | i heard thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 05:32.29 | dj-fu | ;o |
| 05:32.35 | dj-fu | thanks dude :) you've been a real help today |
| 05:32.44 | JT | no probs |
| 05:33.07 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 05:33.07 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 05:38.18 | orkid | the book is good, i'm reading it now :) |
| 05:43.09 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 05:43.23 | Putzz | u told me u were reading hours ago |
| 05:43.25 | Putzz | heh |
| 05:44.24 | orkid | i'm still reading! |
| 05:44.33 | *** join/#asterisk mattwj2006 (n=Matt@c-76-17-131-68.hsd1.mn.comcast.net) |
| 05:44.34 | orkid | and playing CS and doing other things :) |
| 05:44.44 | mattwj2006 | hey guys |
| 05:44.54 | mattwj2006 | I am going to work on some asterisk stuff tonight for fun :) |
| 05:45.03 | orkid | Putzz, you have a motorola modem? ;) lol |
| 05:49.05 | mattwj2006 | I am going to add conference rooms |
| 05:54.39 | k-man_ | what does one put in the users.conf file |
| 05:55.08 | [TK]D-Fender | k-man_ : One ignores it because it is another wasted venture |
| 05:55.18 | k-man_ | [TK]D-Fender, oh |
| 05:55.19 | k-man_ | ok |
| 05:55.20 | k-man_ | thanks |
| 05:55.32 | [TK]D-Fender | k-man_ : its a sad attempt to combin sip.conf, voicemail.con,e tc into one file for some theory of simplification. |
| 05:55.33 | k-man_ | [TK]D-Fender, what about sip_notify.conf |
| 05:55.43 | k-man_ | [TK]D-Fender, ok |
| 05:55.53 | [TK]D-Fender | k-man_ : that isn't a real file. thats FreePBX leftover crap, IIRC |
| 05:55.59 | k-man_ | oh |
| 05:56.00 | k-man_ | ok |
| 05:56.11 | [TK]D-Fender | k-man_ : Were you previously running FreePBX on your system? |
| 05:56.30 | [TK]D-Fender | k-man_ : Or have ported over configs from it? |
| 05:56.51 | k-man_ | [TK]D-Fender, no, i stupidly did make sample or something |
| 05:56.59 | k-man_ | that filled /etc/asterisk with tons of crap |
| 05:57.03 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 05:57.08 | [TK]D-Fender | Perhaps it IS a real file now.. |
| 05:57.11 | k-man_ | i'm tempted to delete the whole lot |
| 05:57.47 | bkruse_home | k-man_: rm -rf /* ? |
| 05:57.47 | [TK]D-Fender | Could be for sending reboots to phones, etc... largely irrelevent. |
| 05:57.51 | k-man_ | bkruse_home, really, that is an old joke, and in fact, its not very funny either |
| 05:58.27 | k-man_ | bkruse_home, touch /'-rf /*' |
| 05:58.57 | k-man_ | yay, stupid things to do in unix |
| 05:58.58 | orkid | yeah, that's not funny dude :| |
| 05:59.24 | k-man_ | try this one for kicks.... man woman |
| 06:00.16 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
| 06:00.18 | k-man_ | so can you run asterisk with only sip.conf and extensions.conf files? and no other config files |
| 06:00.19 | k-man_ | ? |
| 06:00.24 | [TK]D-Fender | k-man_ : I wasn't sure about extra parameters and did "man touch" today... I felt ... *dirty* |
| 06:00.36 | k-man_ | [TK]D-Fender, hehe |
| 06:00.52 | [TK]D-Fender | k-man_ : users.conf = waste of time. |
| 06:00.57 | [TK]D-Fender | k-man_ : move along :) |
| 06:01.09 | k-man_ | ie, can i just delete everything in /etc/asterisk and leave only sip.conf and extensions.conf? |
| 06:01.54 | [TK]D-Fender | k-man_ : No, many things would complein. |
| 06:02.01 | k-man_ | oh |
| 06:02.19 | k-man_ | so whats the bare minimum of files i need in there? |
| 06:02.29 | [TK]D-Fender | k-man_ : asterisk.conf, modules.conf, zapata.conf are rather essential and will bomb if only for their absence. |
| 06:02.35 | [TK]D-Fender | k-man_ : Don't worry so much! |
| 06:02.47 | [TK]D-Fender | k-man_ : Learn how to use what you need and ignore the rest for now. |
| 06:02.58 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 06:03.05 | orlock | arfg fuck |
| 06:03.12 | [TK]D-Fender | k-man_ : Leave the super-byte-level-optimizations to those Gentoo Ricers ;) |
| 06:03.13 | orlock | how important is ztdummy? |
| 06:03.26 | orlock | if you are using pure SIP |
| 06:03.33 | [TK]D-Fender | orlock : Only as important as your need for a software zaptel timing source :) |
| 06:03.48 | k-man_ | [TK]D-Fender, yeah do i need the zapata.conf file if im only doing sip? |
| 06:04.05 | [TK]D-Fender | k-man_ : If you are still loading chan_zap.so you do.... |
| 06:04.19 | [TK]D-Fender | k-man_ : Even if you arne't suing zaptel channels for anything productive. |
| 06:04.45 | k-man_ | ok |
| 06:07.22 | k-man_ | what is cdr.conf? |
| 06:07.48 | k-man_ | nm, i worked it out |
| 06:08.15 | orlock | [TK]D-Fender: f i was having intermittant audio dropouts that were not due to network issues, could that be cause dby a lack of ztdummy? |
| 06:10.01 | [TK]D-Fender | orlock : noe. |
| 06:10.11 | [TK]D-Fender | nope* |
| 06:11.08 | orlock | nope? |
| 06:11.14 | orlock | whats it needed for? |
| 06:12.13 | [TK]D-Fender | orlock : Do you seriously not know what features use Zaptel timing sources? |
| 06:12.54 | orlock | [TK]D-Fender: i _did_ know, but then went on holidays for 6 weeks and drunk myself stupid |
| 06:13.23 | [TK]D-Fender | orlock : MeetMe, IAX2 Trunking. |
| 06:14.03 | orlock | yeah, nothing i use |
| 06:16.28 | orlock | hmm |
| 06:16.38 | orlock | setting up an asterisk server here for them to use |
| 06:16.45 | orlock | see if that resolves it |
| 06:19.48 | wnfaknd | i understand that after modifying any conf file, asterisk must be restarted. what is the easiest way to restart it? |
| 06:20.23 | JT | not correct |
| 06:20.35 | JT | only the relevant module must be restarted |
| 06:20.41 | wnfaknd | ic |
| 06:20.46 | JT | unless you change your zaptel channel configuration |
| 06:21.23 | wnfaknd | ic |
| 06:21.49 | wnfaknd | so just changing values to say adm.conf would not require a restart of asterisk? |
| 06:21.58 | wnfaknd | amd |
| 06:22.08 | [TK]D-Fender | wnfaknd : Even more incorrect. |
| 06:22.24 | wnfaknd | oh boy |
| 06:22.51 | wnfaknd | care to explain? |
| 06:23.09 | [TK]D-Fender | wnfaknd : Extensions.conf, and voip channel reiver config, and a few other only require a "reload" from CLI which doesn't even mean dropping channels or reloading a module. |
| 06:23.15 | [TK]D-Fender | driver* |
| 06:23.38 | wnfaknd | ok, so just reload from the CLI |
| 06:23.39 | wnfaknd | cool |
| 06:23.57 | JT | yes, reload is what i meant |
| 06:24.04 | JT | i used the wrong term |
| 06:24.05 | [TK]D-Fender | Ones requiring a module reload : chan_zap (zapata / zaptel), musiconhold, and Voicemail are the most commonly manually reloaded ones |
| 06:24.09 | wnfaknd | thx |
| 06:24.32 | [TK]D-Fender | eg: reload res_musiconhold.so |
| 06:24.39 | kaldemar | hello. does anyone know if there's a separate irc channel for dundi? |
| 06:25.09 | rue_mohr | seems there isn't |
| 06:25.41 | [TK]D-Fender | kaldemar : not really. Nobody seriously cares enough about it. |
| 06:25.48 | kaldemar | do'h |
| 06:28.22 | kaldemar | would someone happen to know the behaviour in case there are two different contacts in a dundi network with the same priority? |
| 06:29.24 | kaldemar | uh. he got depressed. |
| 06:30.42 | rue_mohr | kaldemar, teach me all about dundi and I'll see if I can answer your question |
| 06:31.05 | rue_mohr | start with "once upon a time" |
| 06:31.14 | rue_mohr | and end with "happily every after" |
| 06:32.27 | kaldemar | one upon a time, there was mark. then he came up with dundi. nobody seriously cared enough about it and everyone lived happily ever after. |
| 06:32.43 | kaldemar | satisfied? |
| 06:32.50 | rue_mohr | hmmm |
| 06:33.07 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 06:33.07 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 06:33.27 | rue_mohr | I'v thought about it hard, I cant answer your question, sorry. |
| 06:33.38 | orlock | ngargh |
| 06:34.30 | rue_mohr | but if I were coding it, both of them would get contacted |
| 06:34.58 | rue_mohr | if there were two replies, the second one would be disposed of |
| 06:36.55 | kaldemar | yea. guess i'll just have to setup a network with n nodes and try it or use the source. :P |
| 06:37.54 | *** join/#asterisk vader-- (n=me@c-71-226-201-15.hsd1.nj.comcast.net) |
| 06:37.55 | vader-- | hola |
| 06:38.27 | vader-- | anyone running the 8.6 sip firmware for the cisco 7940G phones? |
| 06:42.50 | orkid | hola hovito no |
| 06:45.02 | vvortex3 | anyone know why festival gives unbound variable errors? |
| 06:47.54 | *** join/#asterisk inspired (n=mikael@cl-330.sto-01.se.sixxs.net) |
| 06:53.28 | *** join/#asterisk yxa (n=lonari@58.185.90.101) |
| 06:55.01 | *** join/#asterisk mog_home (n=rah@c-71-207-215-93.hsd1.al.comcast.net) |
| 06:55.01 | *** mode/#asterisk [+o mog_home] by ChanServ |
| 06:58.01 | *** part/#asterisk mattwj2006 (n=Matt@c-76-17-131-68.hsd1.mn.comcast.net) |
| 07:01.39 | *** join/#asterisk Strom_M (n=strom@24.86.0.236) |
| 07:01.48 | CunningPike | Hey, Strom_M |
| 07:01.53 | Strom_M | hey there |
| 07:01.55 | CunningPike | Strom_M: All settled in? |
| 07:01.57 | CunningPike | :) |
| 07:02.00 | Strom_M | i just arrived at the hotel |
| 07:02.04 | CunningPike | Perfect |
| 07:02.21 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 07:03.22 | Strom_M | walked down robson and had dinner :) |
| 07:04.55 | rudholm | Strom_M: I forgot to give you the Canadian change I have |
| 07:05.29 | rudholm | I have a bunch of canadian change I need to get rid of |
| 07:05.42 | CunningPike | rudholm: Send it to me :) |
| 07:05.50 | rudholm | heh |
| 07:05.50 | rudholm | I |
| 07:05.57 | rudholm | I'm not sure the postage would be worth it :) |
| 07:06.02 | CunningPike | :) |
| 07:06.36 | aptura | hiya cunning |
| 07:06.44 | orlock | melt it down |
| 07:06.48 | Strom_M | rudholm: heheh, no worries |
| 07:06.49 | orlock | probably worth more as scrap |
| 07:06.55 | Strom_M | i have tons of canadian change somewhere too |
| 07:07.49 | Strom_M | and i will probably come back with even more |
| 07:08.09 | Strom_M | i wonder how much I get raped for using a scotiabank atm |
| 07:08.32 | rudholm | you know you liked it |
| 07:08.33 | *** join/#asterisk Rhizome (n=Rhizome@81.191.147.145) |
| 07:08.45 | CunningPike | Hey, aptura - been a while |
| 07:08.48 | aptura | i used scotiabank at one time. |
| 07:09.08 | aptura | cunning yea it has. how well have you delt with last months weather on your scooter? |
| 07:09.20 | CunningPike | Very well thanks |
| 07:09.22 | aptura | It was a very crappy winter |
| 07:09.29 | CunningPike | Drove my truck instead lol |
| 07:09.34 | aptura | Ohh i bet |
| 07:09.35 | aptura | ;) |
| 07:10.01 | aptura | Btw, got my truck out of the body shop. The truck looks better then before it went in. |
| 07:10.31 | JT | isn't that the idea? :P |
| 07:10.57 | aptura | If you need body repairs and the shop or you cannot find replacment sheet metal take it to corvetts unlimited in surrey. One of there bodyman knows sheet metal bending in an out and made a new rocker panel for my truck. |
| 07:11.08 | aptura | The shop is right next to the scottroad sky train. |
| 07:11.19 | JT | in the uk? |
| 07:11.23 | aptura | no bc |
| 07:11.48 | aptura | BC has city names simular to uk. BC was formerly owned by the common wealth. |
| 07:12.03 | Strom_M | hence "BRITISH columbia" |
| 07:12.12 | rudholm | yeah, kinda intuitive like that :)\ |
| 07:12.13 | aptura | china,india,falken islands, who knows what else. |
| 07:12.38 | JT | yeah |
| 07:12.43 | rudholm | well, they managed to hold on to the Falklands |
| 07:12.52 | JT | we have some place names same as in england too |
| 07:13.02 | JT | none quite as british as surrey though :P |
| 07:13.08 | aptura | Downside with this is the massive influx of immigrants from these countries are allowed to move here. There should be some kind of immigration regs based on housing neeeds. |
| 07:13.10 | orlock | surrey hills? |
| 07:13.12 | rudholm | they had US too at one point, iirc :) |
| 07:13.14 | orlock | Doncaster? |
| 07:13.18 | Strom_M | very little in southern california has british names |
| 07:13.18 | JT | surry hills, orlock |
| 07:13.34 | Strom_M | Pasadena, San Luis Obispo, Pollo del Mar, etc |
| 07:13.44 | orkid | man, i wanna go to socal so bad right now. toronto is freezing cold! |
| 07:13.47 | rudholm | yeah, here in socal we have just some street names, but those are more fanciful than historic |
| 07:13.56 | rudholm | dude, it was 85 here today |
| 07:14.04 | Strom_M | and sunny |
| 07:14.06 | aptura | Vancouver BC is pretty much tapped out for realestate. Thay have run out of room |
| 07:14.09 | rudholm | and clear! |
| 07:14.17 | Strom_M | <3 los angeles |
| 07:14.22 | Strom_M | i miss the sun already |
| 07:14.25 | rudholm | it's still 71 |
| 07:14.27 | aptura | rudholm where you at? |
| 07:14.36 | rudholm | Los Angeles |
| 07:14.39 | aptura | ahh |
| 07:15.07 | rudholm | supposed to be 80 tomorow |
| 07:15.09 | aptura | CunningPike didnt it feel kind of warm today for a febuary day? |
| 07:15.22 | aptura | rudholm is that above seasonal highs? |
| 07:15.47 | Strom_M | well, I was able to walk down Robson tonight and not turn into an icicle |
| 07:15.57 | Strom_M | so that's a plus |
| 07:16.01 | rudholm | that's a bit warmer than average, yes |
| 07:16.12 | rudholm | average high for this time of year is about 71 |
| 07:16.45 | aptura | thats not good |
| 07:16.47 | orlock | it was 34 deg today |
| 07:16.54 | orlock | but i think we are talking different scales |
| 07:16.59 | aptura | has it been creaping up over the years? |
| 07:17.17 | CunningPike | aptura: It was very pleasant - meant to be even milder tomorrow |
| 07:18.09 | aptura | I know what thing, If I get into a position to buy property its 1. going to be away from major cities and 2. away from areas that will experaince major drought. I have one city in mind and it meets both those with a huge mountain behind it. |
| 07:18.31 | aptura | global warming will be a serios issue with extreem water restrictions in the future. |
| 07:19.40 | aptura | nice thing is the mountain supplys a good chunch of its water ONLY to a town of 30k |
| 07:19.47 | aptura | chunk :) |
| 07:22.24 | *** join/#asterisk shinux__ (n=shinux@196.220.25.102) |
| 07:22.28 | james_ | heh, this city meets none of those requirements |
| 07:22.56 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 07:23.05 | CunningPike | aptura: Where's that, then? |
| 07:23.07 | aptura | CunningPike just wanted to mention of interest there is going to be a huge wind farm built on the extreem end of north vancouver island or at least in that aproximate area. That will help in reducing water flows out of the dams. |
| 07:23.22 | aptura | CunningPike what I will be doing is in the future ;) |
| 07:24.06 | aptura | but the sweet thing about this area is extreem low crime and way out of the way of major highways which usually draws in urban sprawl and crime. |
| 07:26.30 | james_ | crime's great |
| 07:27.22 | rudholm | so where is this utopia, aptura? |
| 07:27.37 | aptura | heheh |
| 07:27.42 | aptura | Im not saying :) |
| 07:27.53 | rudholm | what country is it in? |
| 07:28.39 | james_ | gee, dont say, we might all move there at once... |
| 07:28.44 | aptura | hehehe |
| 07:28.45 | aptura | :) |
| 07:28.46 | rudholm | yeah, serously |
| 07:29.04 | rudholm | I plan on leaving my house and job and friends to move to someone else's idea of the ideal city |
| 07:29.11 | aptura | Once I buy my land in this city then will mention it. |
| 07:29.19 | rudholm | oh please |
| 07:29.20 | james_ | you're insane |
| 07:29.27 | rudholm | don't you think you're taking yourself a bit seriously? |
| 07:29.30 | aptura | I like it small :) |
| 07:29.31 | james_ | you use an open source application |
| 07:29.39 | james_ | that reveals it sources |
| 07:31.15 | Strom_M | i like software that's free as in stfu |
| 07:31.22 | aptura | My idea of a utopia is buying all the land I can so it can be controled one what is built on it. I hate the idea of big box stores. I detest walmart so dont go there. |
| 07:31.41 | aptura | Strom_M yea, do you get paid for your free software work? |
| 07:31.46 | Strom_M | there is no walmart in the city of los angeles AFAIK |
| 07:31.52 | Strom_M | aptura: I'm not a coder |
| 07:32.13 | aptura | Thats okay. |
| 07:32.31 | Strom_M | but I do charge handsomely for consulting |
| 07:32.48 | rudholm | yes, his invoices are very handsome, I've seen them. |
| 07:32.58 | rudholm | all orderly and itemized and everything |
| 07:33.01 | Strom_M | :D |
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| 07:33.20 | Strom_M | PHONE WORK 40 H |
| 07:33.30 | Strom_M | PL2PAYKTHX |
| 07:33.40 | aptura | Strom over 100? |
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| 07:33.46 | Strom_M | yes |
| 07:33.55 | aptura | just in your city? |
| 07:34.01 | aptura | and how long |
| 07:34.06 | Strom_M | in los angeles, yes |
| 07:34.13 | Strom_M | what do you mean "and how long" |
| 07:34.14 | aptura | big ass city |
| 07:34.15 | aptura | :) |
| 07:34.15 | rudholm | and that's real dollars, not Canadian |
| 07:34.23 | Strom_M | bahahaha |
| 07:34.23 | aptura | long as when you first started. |
| 07:34.34 | Strom_M | year and a half? ish? |
| 07:34.36 | aptura | Canadians are cheap |
| 07:34.49 | aptura | Thats good. |
| 07:35.10 | aptura | When you started did you drag a pbx and phone with you or what did you use? |
| 07:35.31 | Strom_M | I dont understand the question |
| 07:35.55 | aptura | I have my phone logging in thought my external ip in the event I need to take it with me to demo it. This is of course as long as the customers firewall permits it. |
| 07:36.27 | Strom_M | i always hate demos |
| 07:36.30 | Strom_M | so i never do them |
| 07:36.41 | Strom_M | the customer is smart enough to know what a phone looks like |
| 07:36.48 | aptura | Sure |
| 07:36.55 | rudholm | yeah, that sounds like sales :) |
| 07:36.57 | aptura | So how do you aproach the customer? |
| 07:37.07 | rudholm | the customers approach him |
| 07:37.14 | uwe | hello, ive been trying to setup an IAX2 trunk for the last couple of days, with no luck, my system stops functioning correctly (i dont recive calls from pstn and can dial out anymore) if i set the type to user or friend! does any one know what could possibly cause similar symptoms ? |
| 07:37.17 | aptura | not in the begining |
| 07:37.21 | rudholm | they have or want voip deployments |
| 07:37.26 | rudholm | yes, in the beginning |
| 07:37.46 | aptura | rudholm not if he was first starting out unless he advertises :) |
| 07:37.58 | aptura | besides let Strom_M speak :) |
| 07:38.08 | Grnd-Wire | uwe: I'll bet you're using FreePBX/TrixBox?.. You need to make sure you have a "context=" line in your configuration! |
| 07:38.09 | Strom_M | no, rudholm is accurate on this one L) |
| 07:38.10 | Strom_M | er :) |
| 07:38.18 | rudholm | well, he has a reputation, and he is listed as a consultant in a couple of places at least |
| 07:38.25 | aptura | I see |
| 07:38.41 | aptura | I could have started three years ago. |
| 07:39.42 | rudholm | ...but you didn't because you were plotting your escape to Utopia Anonyma? |
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| 07:39.44 | uwe | well, Grnd-Wire , is context=from-pstn fine? these are two ast boxes |
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| 07:40.13 | phpboy | hey all, anybody ever played around with processing cell phone calls through asterisk PBX? |
| 07:40.23 | phpboy | mostly for the recording of cell phone calls |
| 07:40.36 | uwe | Grnd-Wire, yes, im using freepbx, but im taling a close look at the generated files |
| 07:40.43 | phpboy | be it directly or indirectly through asterisk |
| 07:40.54 | Grnd-Wire | uwe: If you're trying to link them together, so they can call each others extensions - you want to use "from-internal".. |
| 07:41.26 | uwe | Grnd-Wire, thank you very much, this is a great help |
| 07:41.29 | aptura | A utopia is a place to relax in a mountain setting but still not to far away from basic nessesities. I plan on building my home and it will be very energy efficient with a soth facing wall to heat it. Part of the house will be sunk into a hillside "I hope" to keep it warm with minimal warming. |
| 07:42.33 | vvortex3 | god damn it, now festival has no errors but just says nothing |
| 07:42.39 | aptura | The ideal is to not depend on external energy sources. |
| 07:42.53 | aptura | gosh darn it what you have issues with festival? |
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| 07:43.26 | vvortex3 | yes |
| 07:45.56 | vvortex3 | where do i even get flite |
| 07:46.19 | uwe | well, i wonder, why would this break asterisk functunality anyway? its not used in any routes!! |
| 07:49.04 | Grnd-Wire | uwe: If you don't have it assigned to a context, bad things will happen.. |
| 07:49.16 | mmbl13 | hi, my system "stopped" last night i had to reboot trough "hard reset" - where could i find hints for the reason of that crash?! |
| 07:49.47 | Grnd-Wire | Trust me, I learned that the hard way.. How do you think I knew you were using FreePBX? :P |
| 07:49.47 | mmbl13 | checking harddisk reports PASSED (smartctl) |
| 07:49.49 | uwe | Grnd-Wire, heh ! sounds like doing something wrong while trying out voodoo ;) |
| 07:49.59 | mmbl13 | i remember that there is a -- MARK -- in logfiles on my other systems, does centos do that too? |
| 07:50.10 | uwe | bad things happen (period) |
| 07:50.31 | vvortex3 | screw fetival, found flite instructions |
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| 07:50.54 | Grnd-Wire | mmbl13: That is a configuration option you can specify for syslogd. it does that at configurable intervals, so you can see in your logs that syslogd has NOT crashed, and is indeed still running. |
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| 07:51.16 | Grnd-Wire | mmbl13: Try a "man syslogd" and you should be able to find out what parameters you need to change.. |
| 07:52.33 | aptura | I have a question about void in c. Why are there different ways to write void at the start of a function. Or main? examples int main() void func (void) etc? |
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| 08:11.05 | mmbl13 | Grnd-Wire: thx |
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| 08:30.54 | uwe | Grnd-Wire, all, may i tell you what im setting up maybe youll know what im doing incorrectly? i have this setting currently (which is simply not breaking anything, i know its awfully incorrect) http://pastebin.com/875692 , what do i need to do (other than making the type=user for the incomint (no?) ? |
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| 09:03.59 | jeremy_g | <PROTECTED> |
| 09:03.59 | jeremy_g | <jeremy_g> like i could route incoming calls on the gsm interface |
| 09:03.59 | jeremy_g | <jeremy_g> over to its wifi |
| 09:03.59 | jeremy_g | <jeremy_g> i wonder what drivers have been written for asterisk for the gsm interfaces |
| 09:04.19 | *** join/#asterisk luchshiy (n=anonymou@d212-53-104-193.cust.tele2.ch) |
| 09:04.27 | J4k3 | theres a really sexy GSM interface card |
| 09:04.32 | J4k3 | drop your SIM in and go. |
| 09:04.47 | J4k3 | supports up to 4 radios concurrently. |
| 09:04.55 | J4k3 | prepare to spend a small fortune on it |
| 09:05.08 | J4k3 | theres bluetooth |
| 09:05.12 | J4k3 | that works |
| 09:05.28 | J4k3 | you just need to dedicate a handset to your * box. |
| 09:07.12 | dj-fu | I'm up to the enabling dialing out section, I want to set it up so I don't have to dial anything to get an outside line (using wildcard matching). I've got that sussed, but 'the book' only shows an example on dialing 9 to get out |
| 09:07.14 | dj-fu | and stripping the nine |
| 09:07.25 | dj-fu | so how can I make it dial the number instead of trying to strip something |
| 09:09.49 | JT | do not put :1 in it |
| 09:10.15 | dj-fu | oh |
| 09:10.16 | dj-fu | doh ;P |
| 09:10.18 | dj-fu | thanks |
| 09:12.42 | Grnd-Wire | So umm.. I'm having an odd problem with this TDM04B.. |
| 09:12.47 | jeremy_g | uhhh! |
| 09:13.01 | jeremy_g | J4k3: name that bitch :P |
| 09:13.08 | Grnd-Wire | I've got it set to route incoming calls on channel 1 to an extension that lives across an IAX2 trunk.. and it's actually WORKING! |
| 09:13.35 | Grnd-Wire | The only problem is, when I answer the phone.. it doesn't stop ringing! (the Digium never actually connects the call) |
| 09:14.31 | jeremy_g | Gr1ncheux: :D |
| 09:14.37 | jeremy_g | err |
| 09:14.48 | jeremy_g | Grnd-Wire::D |
| 09:14.58 | Grnd-Wire | jeremy_g: Yea? |
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| 09:16.41 | dj-fu | are priority 102+ special priorities or something? I don't understand, wouldn't using the next number do the same? |
| 09:17.20 | Grnd-Wire | dj-fu: It's just a way for you to catch those events.. They figure you're not going to have 100 lines of dialplan code.. |
| 09:17.55 | Grnd-Wire | Here's the weird part.. In the dialplan, it doesn't look like FreePBX is calling an "Answer" anywhere! |
| 09:19.05 | dj-fu | Grnd-Wire, so it's just a large number to give you room to add stuff in between? |
| 09:19.13 | Grnd-Wire | dj-fu: Essentially.. |
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| 09:36.56 | dj-fu | Say if I have two extensions per phone, I'm configuring them as like 701, 702 / 703, 704 |
| 09:37.04 | dj-fu | is there anyway I can get around having to configure voicemail for each extension |
| 09:37.09 | dj-fu | and incoming routes |
| 09:37.18 | dj-fu | like, can I make it right both of the extensions instead of just the one |
| 09:37.27 | dj-fu | currently it just rings 701 instead of 701 and 702 |
| 09:37.32 | jeremy_g | Would it work if i fetch data from a remote sql db and insert it into the local db, and program asterisk real time to read from the db? |
| 09:38.01 | jeremy_g | is the asterisk real time db manually alterable? (offcourse, it gotta be but lemme ask anyway) |
| 09:38.25 | jeremy_g | secondly, if the remote database is OOP based. would it support sql? |
| 09:38.26 | *** part/#asterisk [Airwolf] (n=airwolf@smurf.comsenso.com) |
| 09:38.45 | J4k3 | dj-fu: I have an idea... why not use extensions like "7011" and "7012" up to "7019" |
| 09:39.13 | J4k3 | so when 701 is rung... you know it actually belongs to any/all of those #'s |
| 09:39.17 | J4k3 | seems like it'd be easier to deal with |
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| 09:41.14 | dj-fu | J4k3, that's not the issue, I want it to ring both extensions instead of just ringing 701 |
| 09:42.01 | dj-fu | am I thinking about this the right way? If I have two "lines" per phone, whenever someone rings I want it to ring both, for example, if someone is on line 701, 702 will ring and you'll be able to hold 701 and answer 702 |
| 09:42.09 | dj-fu | I'm sure it's possible, just have no idea how to do it :) |
| 09:42.12 | J4k3 | well, you could make 701 the ring group, and it'd ring 7011, 7012. 7013, 7014, etc. |
| 09:42.35 | dj-fu | ooh. ring groups. |
| 09:42.38 | dj-fu | haven't got to that in the book yet |
| 09:42.40 | dj-fu | nevermind :) |
| 09:42.49 | J4k3 | yeah |
| 09:42.51 | J4k3 | but |
| 09:42.51 | dj-fu | i've just been configuring my entire system for each extension - lol |
| 09:43.03 | Grnd-Wire | Does Digium's support work 24/7 ? |
| 09:47.45 | jeremy_g | yo guys, i have heard about this open source pbx called asterisk? does it run on linux? |
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| 09:51.08 | JT | dj-fu: "lines"? what sort of phones are there? |
| 09:52.51 | uwe | um, when i add an iax2 trunk, things go wrong, and i cant get in or out from the box, all report busy, Grnd-Wire suggested it might the context missing, so i added it, still, when i set the type to user, everything collapses again, looking at the logs, i see something like "Executing NoOp("IAX2/Z-peer-20", "Returned from dialparties with no extensions to call") in new stack" which seems to me causing the problem, the sequence later shows Executing congest |
| 09:52.51 | uwe | ion! looking at the dial plan, i dont see anything unusual, does anybody have any idea how this problem is happening?! |
| 09:53.37 | uwe | the message mentioned above shows how newly incoming calls (from an already functional iax2 trunk) are treated |
| 09:56.17 | JT | freepbx? |
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| 10:03.28 | misc-- | hi... I have a quick question |
| 10:04.43 | misc-- | I have an asterisk server behing a linux box. This box is forwarding *everything* for a certain public IP using 1 to 1 nat to the private IP of the asterisk box. Is there anything else that needs to be done for it work? Any modules that need to be loaded? |
| 10:08.45 | uwe | JT, well , i am using freepbx, but ive been testing directly from the files now, not applying using freepbx |
| 10:09.01 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 10:09.06 | JT | err all those files are completely freepbx derived |
| 10:09.14 | dj-fu | JT, I'm using the term lines not literally. I have Linksys phones and they have four configurable sip "lines" (someone tell me the right word). |
| 10:09.23 | uwe | so shouldnt be relevent, but yes, all basic config genereated by freepbx |
| 10:09.28 | JT | i'd just call them a connection |
| 10:09.38 | dj-fu | right, because it's sip - very true. |
| 10:09.46 | JT | but i don't understand why you need multiple extension lines, dj-fu |
| 10:09.52 | JT | extension numbers |
| 10:09.56 | JT | for the line buttons |
| 10:10.00 | Grnd-Wire | dj-fu: That's actually what they're called! Everyone calls them lines.. If you wanted to avoid confusion, you could call them "line appearances" |
| 10:10.21 | dj-fu | sorry, I'm new to this - first day of manual asterisking. |
| 10:10.29 | dj-fu | JT, well, I guess I don't |
| 10:10.55 | dj-fu | JT, I just need it so that when you press 3, it rings both connections to the technical phone out the back, and that when I transfer from technical to reception, it does the same, rings both |
| 10:11.14 | dj-fu | I'd rather not have two extension numbers, that's just how it was configured previously so I thought it was normal |
| 10:11.29 | JT | err |
| 10:11.49 | JT | you mean it rang 2 phones, or it rang 2 "lines" on the one phone |
| 10:12.12 | dj-fu | 2 line apperances on the one phone |
| 10:12.24 | JT | what on earth is the point? |
| 10:12.39 | dj-fu | so you can hook switch or put someone on hold and then talk to the other person? |
| 10:12.44 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 10:12.58 | dj-fu | or is what I'm trying to do the wrong way to go about what I'm setting out to achieve |
| 10:13.16 | JT | i don't see why you can't do that without ringing both |
| 10:13.20 | JT | i'd say so |
| 10:13.33 | JT | you can have multiple sip connects between identical phone numbers |
| 10:13.43 | dj-fu | oh. |
| 10:14.15 | dj-fu | So once the phonecall is connected to that phone, it becomes independant of the physical PSTN line? I can hook switch merrily with just one sip connection? |
| 10:14.33 | JT | why do you need hook switching if it's a sip phone? |
| 10:14.53 | dj-fu | um |
| 10:15.11 | dj-fu | so I can put someone on hold and then talk to the other person trying to ring my phone? |
| 10:15.37 | JT | you press the hold button? |
| 10:15.40 | dj-fu | yea |
| 10:15.50 | JT | that is not hook switching |
| 10:15.55 | dj-fu | doh |
| 10:16.07 | dj-fu | ok, what is what I'm trying to do called? |
| 10:16.13 | JT | hold :) |
| 10:16.40 | dj-fu | argh |
| 10:17.06 | dj-fu | so you're telling me I can have two "line apperances" on my SIP phone with a single extension number, and "hold" between them? |
| 10:17.41 | JT | sure |
| 10:17.48 | dj-fu | I guess that makes sense, I never tried it, but there's nothing on the phone manual saying you can't register ext2 to the same PEER as ext1 |
| 10:18.09 | JT | i'm not an expert in that model of phone |
| 10:18.21 | dj-fu | Nor am I, unfortunately |
| 10:18.27 | JT | but it should work, but it would be good to speak with someone who's dealed with that phone before |
| 10:18.50 | dj-fu | I'll test it when I'm back in work on wednesday. I'm going to edit all of my extensions and voicemails and whatnot now |
| 10:18.53 | dj-fu | ridiculous ;( |
| 10:18.57 | dj-fu | I'm such a newb. |
| 10:19.12 | JT | ah well |
| 10:20.51 | dj-fu | let me ssh forward one of the phones web interface and try it |
| 10:20.54 | dj-fu | you've got me all excited |
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| 10:23.30 | dj-fu | JT, well, both of the "extensions" on the phone registered to my extension #701 fine |
| 10:23.44 | dj-fu | which is cool, I guess |
| 10:26.51 | dj-fu | heh. that makes my life a whoole lot easier. |
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| 10:27.43 | dj-fu | sorry to nag you dude :) |
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| 10:33.07 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 10:33.07 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 10:35.08 | *** join/#asterisk Igbothom_3rd (n=Hilton@office.quarkit.com.au) |
| 10:35.14 | *** join/#asterisk darviria (n=darviria@194-105-181-29.ifb.co.uk) |
| 10:35.27 | dj-fu | damn, he got pissed at me xD |
| 10:35.56 | *** join/#asterisk s1gny|wrk (n=s1gny@p54917C09.dip.t-dialin.net) |
| 10:36.31 | *** part/#asterisk s1gny|wrk (n=s1gny@p54917C09.dip.t-dialin.net) |
| 10:40.56 | BrokenNoze_ | Anyone used AddSipHeader? how would I use it from the manager API? |
| 10:42.17 | JT | just afk |
| 10:42.37 | *** join/#asterisk luisjose (n=ljd@unaffiliated/luisjose) |
| 10:46.01 | *** join/#asterisk pjz (n=pj@zachs.place.org) |
| 10:46.33 | *** join/#asterisk Beirdo_ (n=gjhurlbu@unaffiliated/beirdo) |
| 10:48.14 | dj-fu | JT, does http://rafb.net/p/niojqA21.html look alright for what we were talking about? all the phones are registering OK |
| 10:50.17 | *** join/#asterisk Aurs (n=Aurs@81.191.112.104) |
| 10:55.24 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 10:55.24 | *** join/#asterisk orkid (n=orkid@bas1-barrie18-1242375010.dsl.bell.ca) |
| 10:56.27 | JT | dj-fu: umm looks ok |
| 10:56.33 | JT | nothing but to test stuff i guess |
| 10:57.04 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 10:59.08 | mafkees | ello |
| 10:59.52 | *** join/#asterisk llirk (i=krill@snorkel.rtfm.net.au) |
| 11:00.12 | orlock | snorkel... |
| 11:00.16 | orlock | i know that name |
| 11:00.33 | llirk | yeah? |
| 11:00.34 | orlock | llirk: 2600? |
| 11:00.45 | llirk | not in recent years |
| 11:00.49 | llirk | but yes |
| 11:01.13 | orlock | i went to school with the snorkel admin |
| 11:01.23 | llirk | fletch? |
| 11:01.26 | orlock | yeah |
| 11:01.33 | orlock | few years below him though |
| 11:01.40 | llirk | ahh right |
| 11:01.40 | orlock | met him at a 2600 meet last year |
| 11:02.01 | llirk | i dont usualy go |
| 11:02.15 | orlock | me neither for a while, only been 2 or 3 times |
| 11:02.33 | orlock | 10pm and i am doing work |
| 11:02.44 | llirk | yeah? what you working on |
| 11:03.25 | orlock | setting up a new server |
| 11:03.37 | orlock | got some weird weird quality issues at a site with an on-site asterisk server |
| 11:03.49 | orlock | so setting one up in the office |
| 11:03.56 | orlock | boss is busting my balls |
| 11:04.08 | phpboy | http://www.pastebin.ca/341148 <--- the south african dream |
| 11:04.10 | orlock | bloody salesguy that sold 9 asterisk sites quit |
| 11:06.58 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 11:07.25 | *** join/#asterisk tzanger (n=tzanger@208.68.91.47) |
| 11:09.03 | *** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net) |
| 11:09.18 | *** join/#asterisk tdi (n=tdi@reykin.pozman.pl) |
| 11:12.56 | orkid | detroit's the most dangerous world in the city (for murders per capita) |
| 11:13.15 | orkid | mexico city is pretty rough i hear (for americans and people with any money) |
| 11:13.38 | tdi | orkid: lol |
| 11:13.39 | JT | most dangerous world in the city? |
| 11:13.51 | orkid | err whatever ;) |
| 11:13.53 | Ahrimanes | hehe |
| 11:13.54 | tdi | orkid: apparently you have not been to east europe :) |
| 11:13.58 | phpboy | HAEhaHAe |
| 11:14.00 | orkid | it's 613am :) |
| 11:14.48 | phpboy | if you live in johannesburg suburbia, you're kinda screwed |
| 11:14.54 | phpboy | they're saying it in that sense |
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| 11:25.07 | *** join/#asterisk dexter_speed (n=dexter@dsl-242-48-89.telkomadsl.co.za) |
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| 11:27.58 | *** part/#asterisk s1gny (n=s1gny@p54914BF1.dip.t-dialin.net) |
| 11:28.18 | dexter_speed | has anyone had problems with TDM2400P cards not hanging up properly ? |
| 11:28.46 | dexter_speed | I have 3x TDM2400P in a box here, and I get weird issues when people hangup, where the get connect to other people using the * pabx. |
| 11:29.10 | dexter_speed | or customers get connected to each other, and then the lines lock up, and sit there indefinetly. |
| 11:30.23 | *** join/#asterisk jm|work (n=jm@sentry.flags.co.uk) |
| 11:30.37 | *** join/#asterisk coppice (n=chatzill@55.157.17.210.dyn.pacific.net.hk) |
| 11:32.55 | dj-fu | hey if I have like a fax connected to my fxs do I have to configure anything special in asterisk to let it dial faxes? |
| 11:33.07 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 11:33.08 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 11:33.08 | dj-fu | it isn't sip.. so I wouldn't guess so, but I don' tknow |
| 11:34.32 | *** join/#asterisk in-pt (n=lokesh@estrela.nortenet.pt) |
| 11:35.38 | tzafrir | dexter_speed, what type of hangup detection do you have? where is "here"? |
| 11:35.50 | tzafrir | What coutry? |
| 11:36.03 | *** join/#asterisk martineyles_ (n=martiney@adsl-w-234.as15758.net) |
| 11:36.14 | martineyles_ | Hello |
| 11:36.32 | martineyles_ | I have a problem talking between two x-lite clients |
| 11:36.43 | martineyles_ | the voice is one-way |
| 11:36.45 | dexter_speed | tzafrir: south africa |
| 11:37.02 | dexter_speed | tzafrir: I have hordes of other asterisk boxes, that work fine. just this one screwing around |
| 11:37.03 | martineyles_ | however, if they both dial the asterisk conference no. it works fine |
| 11:37.16 | martineyles_ | voice both ways |
| 11:37.33 | dexter_speed | dj-fu: I also have a problem with fax machines dialing out over my fxs/fxo cards |
| 11:37.39 | martineyles_ | a diagram of my setup is at http://www.bytronic.com/tests/VoIP-NAT.png |
| 11:37.43 | dexter_speed | dj-fu: it gets to the other side, but then the fax fails ? |
| 11:38.40 | tzafrir | dexter_speed, IIRC you'll have to rely on busydetect |
| 11:38.56 | tzafrir | but there may be some tweaks with the busypattern |
| 11:39.15 | dexter_speed | tzafrir: I have busydetect, and busycount=4/2/6 tried all those. and I have hangup on reverse poliarty thingy enabled too |
| 11:39.45 | dexter_speed | tzafrir: I think it might be my cards, I just wanted to know if anyone has experienced this aswell |
| 11:40.43 | drako | martineyles_, i don't recommend you xlite |
| 11:41.05 | martineyles_ | what alternative would you reccomend? |
| 11:41.17 | drako | martineyles_, OS? |
| 11:41.52 | tdi | hey xlite rox |
| 11:41.52 | drako | tdi, not really |
| 11:41.52 | martineyles_ | drako, Windows 2000 for some, Windows XP for others |
| 11:41.58 | drako | martineyles_, Cubix |
| 11:42.07 | *** join/#asterisk angryuser (n=Miranda@i03v-213-44-169-43.d4.club-internet.fr) |
| 11:43.08 | tdi | drako: for linux, xlite is the best solution |
| 11:43.16 | drako | tdi, are you crazy? |
| 11:43.19 | tdi | twinkle is also nice, but for me it hangs every BYE |
| 11:43.28 | drako | tdi, is te worse solution |
| 11:43.29 | tdi | drako: no i am not, works best for me |
| 11:43.35 | martineyles_ | drako, url for cubix? |
| 11:43.38 | drako | Twinkle is. |
| 11:43.58 | tdi | Twinkle is what ? |
| 11:44.01 | drako | tdi, is unstable, bad sound, is bloated |
| 11:44.08 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 11:44.18 | tdi | drako: twink or xlite ? |
| 11:44.19 | drako | Twingle is a good choice, to be honest, the only that works ok. |
| 11:44.21 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 11:44.27 | angryuser | good day, in features.conf i have *1 to start the record into file, in dial i have TtWw flags, but when i press *1 nothing happens, any ideas? |
| 11:44.52 | drako | tdi, i've been testing lots of softphone for linux and the only one working decent is twinkle so far |
| 11:45.03 | puzzled | morning |
| 11:45.14 | tdi | drako: ok, that i can agree |
| 11:45.32 | tdi | drako: because twinkle does not work on my linux installation |
| 11:45.39 | tdi | i canont say a bad word |
| 11:45.47 | tdi | that is why i am using xlite |
| 11:45.54 | drako | tdi, oh really? what it does says? |
| 11:46.05 | drako | tdi, xlite does not even support alsa sound |
| 11:46.08 | tdi | drako: hangs on BYE |
| 11:46.11 | tdi | just |
| 11:46.21 | drako | oh really? distro? |
| 11:46.22 | tdi | drako: yes, agree, that is crap |
| 11:46.26 | tdi | drako: gentoo unstable |
| 11:46.30 | drako | oh |
| 11:46.34 | drako | what do you expect |
| 11:46.34 | drako | hehe |
| 11:46.39 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 11:46.42 | tdi | but only on BYE send |
| 11:46.45 | tdi | not receive |
| 11:47.02 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 11:47.16 | tdi | drako: i expect it to work :) everything works fine, gentoo unstable is more stable than stable one :) |
| 11:47.32 | drako | I've been using xlite for a while on my workstation, is a decent pc, and it works just normal, it lacks of some features such ans transfer |
| 11:47.39 | tdi | drako: how much does cubix cost ? |
| 11:48.05 | drako | tdi, not sure, you can use it without pay under some terms i belieave |
| 11:48.18 | drako | But when i tried xlite on my laptop |
| 11:48.19 | tdi | yes. but cannot download it |
| 11:48.22 | drako | everyting was a mess |
| 11:48.52 | drako | its the hell bloated, sounds is terrible, gave me lots of headaches at work |
| 11:49.02 | drako | good thing i found twingle |
| 11:49.55 | dj-fu | the best softphone I've found is Linphone. |
| 11:50.53 | tdi | drako: do you know sth reasonable for macosx? |
| 11:51.14 | drako | tdi, yes well a friend needed one |
| 11:51.16 | drako | and it works ok |
| 11:51.20 | drako | its iax based |
| 11:51.26 | tdi | nee i nees sip |
| 11:51.29 | tdi | need |
| 11:51.34 | drako | dj-fu, not really, is simple and it DOES crash my asterisk ... |
| 11:51.45 | drako | tdi, well i have no idea :/ |
| 11:51.51 | drako | tdi, that was a problem to find one |
| 11:52.05 | dj-fu | question, If I have a fax connected to my fxs, do I need to make extensions.conf recognize it so it can dial out? create it's own custom context perhaps? |
| 11:52.08 | tdi | drako: none, i will use xlite :) |
| 11:52.10 | drako | tdi, and i came with this iax client cuz xlite is the hell unstable on OSX |
| 11:52.20 | drako | it does crash every 20min |
| 11:52.22 | tdi | aha |
| 11:52.25 | drako | with luck. |
| 11:52.34 | dj-fu | ekiga is another sip compatible softphone which was nice, i couldn't get it going though |
| 11:52.44 | tdi | dj-fu: i you love gnome |
| 11:52.51 | drako | ekiga is not nice. |
| 11:52.53 | tdi | and want to have it and alllll his libs |
| 11:52.55 | drako | ekiga crash |
| 11:52.57 | drako | aswell |
| 11:53.02 | drako | and sound is really bad. |
| 11:53.14 | dj-fu | strange? I thought it used the same codecs as everything else. |
| 11:53.18 | dj-fu | I think you're really bad, you big liar. |
| 11:53.22 | drako | i've been into softphone a lot lately |
| 11:53.41 | drako | dj-fu, no at all, been testing lots of clients diferents computers |
| 11:53.43 | dj-fu | tdi, http://203.167.190.166/Screenshot.png |
| 11:53.59 | drako | but i have to admit that xlite works ok on Windows. |
| 11:54.13 | dj-fu | anyone know the answer to my FXS question? |
| 11:54.17 | Ahrimanes | xlite config is a mess.. |
| 11:54.59 | tdi | dj-fu: yes but you to have gnome-libs and i do not want them |
| 11:55.31 | tdi | dj-fu: if you want to dial you probably have to :) |
| 11:57.05 | dj-fu | cool |
| 11:57.06 | dj-fu | thought so, I just didn't know |
| 11:57.37 | tdi | drako: heheh lol |
| 11:57.43 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 11:57.45 | tdi | i did not know that twinkle has a cli |
| 11:57.49 | tdi | clie does not hang |
| 11:57.49 | dj-fu | another little logistics question, I have two PSTN lines, zap1 and 2. If I use zap1 as my incoming calls, should all my outgoing calls be put over zap2? or will they happily travel over zap1 |
| 11:57.57 | drako | cli? wtf? really? |
| 11:59.06 | tdi | <PROTECTED> |
| 11:59.45 | tdi | dj-fu: if it is occupied, then they will not travel |
| 12:00.22 | martineyles_ | drako, tdi - I want something that I don't have to pay for - for basic voice calling |
| 12:00.39 | martineyles_ | that is why I chose x-lite |
| 12:00.49 | tdi | martineyles_: i am also using xlite |
| 12:01.01 | tdi | what is the problem |
| 12:01.14 | martineyles_ | well, I can call asterisk in the network |
| 12:01.28 | martineyles_ | and someone on another network can call asterisk |
| 12:01.35 | dj-fu | tdi, I see- I thought so |
| 12:01.40 | martineyles_ | and we can thus do a conference call |
| 12:02.09 | martineyles_ | BUT when we try to call direct - x-lite to x-lite, the audio is one way |
| 12:02.41 | tdi | martineyles_: propbably sb is behing nat |
| 12:02.59 | martineyles_ | so I gather that x-lite is sending audio direct in this case, rather than going through asterisk |
| 12:03.13 | martineyles_ | tdi: what is sb? |
| 12:03.14 | tdi | if you call direct then yes |
| 12:03.20 | tdi | martineyles_: sb is somebody |
| 12:03.29 | tdi | sorry for abbreviation |
| 12:03.34 | martineyles_ | ok - we are both behind nat |
| 12:03.50 | drako | martineyles_, well that sound nat problem and with Cubix you can use iax so you wont have any problem. |
| 12:04.03 | martineyles_ | I'm not up on all my etlas (extended three letter abbreviations) |
| 12:04.26 | martineyles_ | drako - is cubix free? |
| 12:04.55 | martineyles_ | and if so, where can I download it? |
| 12:05.13 | tdi | i cannot download it |
| 12:05.23 | tdi | i am also searching for it |
| 12:05.40 | ThoMe | a good cable less (NOT wlan) DECT VOIP phone? |
| 12:05.46 | ThoMe | like gigaset.. siemens |
| 12:05.53 | ThoMe | ideas? |
| 12:05.59 | drako | martineyles_, not really , read the license, its free under some terms and freely downloadable |
| 12:06.24 | drako | I installed it on my brother computer and is really good. |
| 12:08.40 | martineyles_ | searching for cubix didn't work - but now I search for cubix iax and find |
| 12:08.41 | martineyles_ | http://www.virbiage.com/cubix.php |
| 12:11.10 | martineyles_ | there is no download link |
| 12:11.11 | drako | Anyone here with experience with sipura devices? |
| 12:11.24 | drako | martineyles_, keep looking |
| 12:12.36 | BrokenNoze_ | anyone had problems with the Redirect API not finding the right channel? |
| 12:12.40 | coppice | martineyles_: they keep changing the name of that softphone. those guys provide VoIP service in the UK, and maybe elsewhere |
| 12:16.29 | martineyles_ | coppice - I found a "branded" version - but I think it is like the branded x-lites - ie. it is set to plug and play with a set voip provider and no more. |
| 12:16.50 | zoa | im not sure virbiage is still active |
| 12:17.08 | martineyles_ | but I will check |
| 12:17.14 | zoa | the website seems to be from 2005 |
| 12:17.28 | drako | Anyone here with experience with SIPURAs devices? |
| 12:17.29 | zoa | i know the programmer of the phone had a bike accident or so |
| 12:17.37 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 12:17.50 | zoa | or at least the original programmer |
| 12:18.03 | *** join/#asterisk ai-a (n=jake@cpc5-oxfd1-0-0-cust523.oxfd.cable.ntl.com) |
| 12:18.04 | martineyles_ | that |
| 12:18.11 | martineyles_ | that's a shame |
| 12:18.17 | Grnd-Wire | Any of you folks familiar with the Digium TDM04b? |
| 12:18.51 | coppice | virbiage is freshtel, and they are a VoIP provider in some places. In the UK they provide the VoIP service sold as Tesco (at least, I think its Tesco's one) |
| 12:19.00 | martineyles_ | yes - firefly is only for use with the firefly voip network |
| 12:19.08 | ai-a | Im in the UK, looking for a LAN phone adapter that supports SIP over my existing lan, to my PBX which is located elsewhere on the internet. Whats a good box to buy ? |
| 12:19.24 | coppice | firefly was the original name for the softphone |
| 12:19.39 | martineyles_ | and no doubt tesco internet phone will just be for tesco etc. |
| 12:19.54 | martineyles_ | I need a soft-phone that works with MY asterisk server |
| 12:20.00 | ai-a | coppice: im using Firefly with an existing SIP service for testing, and the caller is getting echo, why is that ? |
| 12:20.24 | ai-a | martineyles_: firefly supports IAX and SIP too. |
| 12:20.33 | ai-a | i use it to connect to Asterisk. |
| 12:21.11 | tdi | ai-a: buy linksys pap2t |
| 12:22.00 | dj-fu | drako, all of my phones are Sipura phones |
| 12:22.06 | dj-fu | rebranded Linksys |
| 12:22.07 | *** join/#asterisk toot (n=toot@84.19.255.123) |
| 12:22.18 | ai-a | tdi: searching,.. aprix £40 i see. and that will work over existing router? regardless of DSL or ADSL? i have netgear, im looking at taking it to work and just jacking it in :) |
| 12:22.42 | drako | dj-fu, i have 2 problems with sipura devices, sometimes on the FXS (Sipura 2002) it does not work the ## transfer |
| 12:23.04 | drako | dj-fu, and with Sipura-3000 sometimes it does not hang up the call so the line remains busy |
| 12:23.15 | tdi | drako: you only have problems |
| 12:23.33 | tdi | we have tousands of linksys (that is sipura) and work fine |
| 12:23.55 | *** join/#asterisk Navman (n=Navman@62.108.206.82) |
| 12:24.05 | dj-fu | all of mine work perfectly :\ |
| 12:24.08 | tdi | ai-a: yes, it should work ok. |
| 12:24.22 | tdi | they are cool indeed |
| 12:24.27 | dj-fu | I find that sometimes if I don't put the handset down correctly it doesn't hang up properly, the switch is a bit dicky in the hook I think |
| 12:24.50 | martineyles_ | "Firefly uses Speex under the Xiph.org variant of the BSD license." |
| 12:24.59 | martineyles_ | is speex a voip phone? |
| 12:25.06 | dj-fu | drako, I've also seen the transfer issue.. you usually have to wait for the phone to start calling and then hit xfer, it's a bit strange |
| 12:25.08 | dj-fu | speex is a codec |
| 12:25.10 | dj-fu | vbr |
| 12:25.11 | Ahrimanes | it's an audio codec |
| 12:26.06 | ai-a | martineyles_: google.com search speex |
| 12:26.16 | Grnd-Wire | So why can't I get my TDM04B to ANSWER the line? |
| 12:26.28 | Grnd-Wire | When I execute the answer command, it just ignores it.. |
| 12:26.52 | martineyles_ | firefly only has options for firefly |
| 12:27.39 | *** join/#asterisk leandroaugusto (n=leandroa@mx.canaltelecom.com.br) |
| 12:29.47 | ai-a | martineyles_: READ - http://www.toyz.org/mrblog/archives/00000194.html |
| 12:30.09 | ai-a | http://www.asteriskguru.com/tutorials/firefly_softphone.html |
| 12:30.16 | *** join/#asterisk littleball (n=littleba@bb220-255-152-63.singnet.com.sg) |
| 12:30.37 | ai-a | martineyles_: they all state SIP and IAX2.. where do you only see Firefly network ? |
| 12:30.56 | littleball | hello, who can recommend a device for me to connect to my mac or ibm notebook so that i can call through fix line? |
| 12:31.20 | ai-a | lilalinux: using asterisk ? |
| 12:31.54 | littleball | ues |
| 12:31.59 | littleball | uyes |
| 12:32.15 | littleball | linux or mac os |
| 12:32.21 | lilalinux | ai-a: why else would I be here? |
| 12:32.30 | ai-a | well, just checking :) |
| 12:32.49 | *** join/#asterisk nighty- (n=nighty-@66-163-28-100.ip.tor.radiant.net) |
| 12:33.06 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 12:33.07 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 12:33.08 | nighty- | hi has anyone ever succeeded in contacting moimstone ? |
| 12:33.51 | ai-a | littleball: http://www.voip-info.org/wiki-Asterisk+phones |
| 12:33.56 | ai-a | lilalinux: wrong person sorry. |
| 12:34.04 | tzafrir | Grnd-Wire, give us some information about your configuration |
| 12:34.26 | tzafrir | Do you see the channel in 'zap show channels' in the CLI? |
| 12:36.36 | *** join/#asterisk orkid (n=orkid@bas1-barrie18-1242375010.dsl.bell.ca) |
| 12:36.52 | *** join/#asterisk luchshiy (n=anonymou@d212-53-104-193.cust.tele2.ch) |
| 12:37.14 | martineyles_ | http://www.freshtel.net/firefly/download/ |
| 12:38.16 | martineyles_ | is where I got firefly from |
| 12:38.28 | UlbabraB | hi, I've set a Verbose(${DIALSTATUS}) in my dialplan after a Dial() command and it does'nt reports CANCEL and NOANSWER statuses |
| 12:38.43 | UlbabraB | all other statuses are reported correctly |
| 12:45.34 | BrokenNoze_ | Wireless IP Phones. are they any good? surely lack of bandwidth makes them useless in an office environment? |
| 12:45.56 | dj-fu | lack of bandwidth |
| 12:45.57 | dj-fu | lol |
| 12:46.15 | Grnd-Wire | dj-fu: He does have a valid point you know.. |
| 12:46.42 | BrokenNoze_ | no? am I being thick here? I have a client that wants to use 15 handsets across a site |
| 12:47.07 | BrokenNoze_ | is bandwidth not going to cause a problem? |
| 12:47.36 | dj-fu | alaw/ulaw is the highest bandwidth codec @ 64kbps |
| 12:48.54 | BrokenNoze_ | Um, I don't suppose thats gonna cause an issue on a 54MBps network for 15 handsets then |
| 12:49.14 | dj-fu | one would hope so |
| 12:49.21 | dj-fu | providing you use good stable wireless gear anyway |
| 12:49.27 | Grnd-Wire | BrokenNoze_: no, but are you sure you will get 54mbps in all locations? |
| 12:49.31 | BrokenNoze_ | anyone recommend a handset |
| 12:49.31 | dj-fu | not crappy SoHo access point |
| 12:49.52 | Grnd-Wire | and you know that the more devices you have - you start dividing up your 54mbps.. |
| 12:50.10 | Grnd-Wire | and yeah - Use Cisco wireless AP's, or suffer the consequences.. ;P |
| 12:50.11 | *** join/#asterisk Tebi (n=rantis@gw.aller.fi) |
| 12:50.59 | tdi | or a mikrotik base |
| 12:51.04 | tdi | <PROTECTED> |
| 12:51.22 | BrokenNoze_ | Yep, using Cisco. |
| 12:52.15 | dj-fu | well - you could have 54 wifi voip handsets WITH a reasonable AP and still have enough bandwidth for calls |
| 12:52.23 | BrokenNoze_ | Have to check Qos though |
| 12:52.56 | BrokenNoze_ | as they will be using other devices on the network |
| 12:53.02 | dj-fu | ah |
| 12:53.38 | BrokenNoze_ | yeah, could cause an issue |
| 12:53.46 | toot | hey folks just wondering what people use for load generation with asterisk/sip? |
| 12:54.10 | BrokenNoze_ | can u get QoS wireless access points to priorities voice traffic? |
| 12:54.24 | dj-fu | BrokenNoze_, Cisco AP's should handle that. |
| 12:54.48 | BrokenNoze_ | dj-fu : OK, thanks. not much of a network guy |
| 12:55.04 | dj-fu | I'd recommend hiring a CCNA/CCIE then, if you're looking at doing cisco stuff |
| 12:55.27 | Grnd-Wire | BrokenNoze_: You will definately want to enlist the help of a competent network engineer - someone who is familiar with VoIP, and QoS!.. yeah - CCNA/CCNP |
| 12:55.40 | Grnd-Wire | You probably don't want to pay for a CCIE.. :P |
| 12:55.54 | BrokenNoze_ | done by the client apparently, but I don't wanna go in saying the hardwares good enough if it ain't |
| 12:57.13 | dj-fu | right, CCIE is a little higher up |
| 12:57.16 | BrokenNoze_ | However we're supplying the Telephony hardware and servers. Recommend a Wireless handset anyone? |
| 12:57.28 | Grnd-Wire | BrokenNoze_: Yeah! You can't warrantee something that is already there.. So get a disclaimer in writing in advance.. :P |
| 12:57.32 | dj-fu | linksys or cisco may make one |
| 12:57.43 | dj-fu | let me take a quick look at my cisco supplier |
| 12:57.48 | Grnd-Wire | Linksys has one.. There are several actually - I've never used any of them though.. |
| 12:57.52 | BrokenNoze_ | Oh we will! I've had enough problems in the past with networks no to! |
| 12:58.28 | BrokenNoze_ | Yeah, they all do them. probably go the Cisco route then if no one's used them |
| 12:58.40 | Grnd-Wire | BrokenNoze_: here.. Checkout the site.. http://www.telephonydepot.com/Wireless_VoIP_s/78.htm |
| 12:58.45 | dj-fu | I can't see any dedicated wifi Linksys phones here, just normal linksys phones w/ attachable wireless dongles |
| 12:59.09 | Grnd-Wire | Zyxel makes one, Linksys makes one, Dlink - and there's a couple of brands I've never heard of before. |
| 12:59.29 | BrokenNoze_ | Client was talking about the Zxyel x2000 |
| 12:59.34 | dj-fu | yuck, zyxel :o |
| 12:59.43 | BrokenNoze_ | not good? |
| 12:59.50 | dj-fu | not imo |
| 13:00.05 | dj-fu | only ever had issues with zyxel gear |
| 13:00.09 | Grnd-Wire | They make kickass modems :P |
| 13:00.11 | dj-fu | I tend to stick with linksys & cico |
| 13:00.14 | dj-fu | +s |
| 13:00.23 | dj-fu | but I'm ccna-inclined :) |
| 13:00.56 | BrokenNoze_ | Its a cost thing really. gotta win the contract |
| 13:01.53 | dj-fu | I hate contracts like that. |
| 13:01.58 | dj-fu | I'd rather quote for good, reliable gear |
| 13:02.01 | dj-fu | (cisco and linksys) |
| 13:02.11 | dj-fu | than try and beat other peoples quotes with budget arse d-link etc |
| 13:02.34 | Grnd-Wire | dj-fu: Yeah - I won't work with clients that don't let me use good equipment.. |
| 13:02.40 | BrokenNoze_ | Client is probably willing to go with the Cisco, so might be able to sway for the better gear |
| 13:02.59 | Grnd-Wire | dj-fu: If they're that cheap, they won't want to pay my rates to keep servicing the stuff. ;) |
| 13:03.08 | dj-fu | that's the thing |
| 13:03.16 | dj-fu | it's a double edged sword buying crap gear |
| 13:03.26 | dj-fu | cause you gotta pay for a tech to come out every time it stuffs up to fix it |
| 13:04.24 | BrokenNoze_ | yeah, one site in Germany on in Czech |
| 13:04.32 | BrokenNoze_ | 2 over here in UK |
| 13:04.44 | BrokenNoze_ | so Can't be doing with hardware issues |
| 13:06.38 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 13:06.41 | dj-fu | what kind of security would you run on the AP? |
| 13:06.50 | dj-fu | wpa2-psk enterprise? radius? |
| 13:06.55 | BrokenNoze_ | um. unsure |
| 13:07.08 | angryuser | i am unable to record a call by typing *1, in my diam WwTt flags are checked, any ideas? |
| 13:07.16 | angryuser | *dial |
| 13:08.09 | tzafrir | dj-fu, security is the things you don't run ;-) |
| 13:09.20 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 13:15.02 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 13:16.44 | toot | anyone doing load testing? i'm lost as to the right app to use to sip testing |
| 13:18.54 | *** join/#asterisk drako (n=ljd@unaffiliated/luisjose) |
| 13:19.38 | *** join/#asterisk reinhard (n=reinhard@80.120.117.86) |
| 13:19.42 | *** part/#asterisk reinhard (n=reinhard@80.120.117.86) |
| 13:20.30 | *** join/#asterisk jmls (n=asterisk@62.49.235.130) |
| 13:20.47 | jmls | what format is best for ripped a cd for moh ? |
| 13:20.49 | *** join/#asterisk Ethon (i=arne@Oldman.steinkamm.com) |
| 13:20.54 | jmls | (ripping) |
| 13:20.56 | Ethon | Hi |
| 13:21.27 | Grnd-Wire | jmls: MP3 |
| 13:23.24 | *** join/#asterisk Crescendo (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 13:23.38 | ai-a | martineyles_: still having issues with FireFly ? |
| 13:24.00 | ai-a | martineyles_: read this - - - - http://www.freshtel.net/help/firefly/tech/#protocolsupport - notice it says SUPPORTS SIP. |
| 13:25.41 | *** join/#asterisk DrukenLPY (n=jdumais@CPE000854ddcdb1-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 13:26.47 | *** join/#asterisk DrukenHME (n=jdumais@CPE000854ddcdb1-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 13:27.18 | jmls | Grnd-Wire: thanks. What about .gsm or .[u|a]law ? |
| 13:28.29 | Grnd-Wire | jmls: Sure, but MP3 is the easiest - and will sound great.. You asked for the best, and that was the answer I gave you. |
| 13:29.03 | *** join/#asterisk hal2385612 (n=chatzill@host217-44-172-12.range217-44.btcentralplus.com) |
| 13:29.19 | tzafrir | It won't sound better than slinear on Asterisk, as most audio passed around is 8kHz anyway |
| 13:29.35 | hal2385612 | hello all |
| 13:29.43 | tzafrir | Not to mention that mp3 is lossy compression |
| 13:29.45 | hal2385612 | hope everyone is in good spirits |
| 13:29.49 | hal2385612 | ;-) |
| 13:29.54 | hal2385612 | Can I ask a question please? |
| 13:30.04 | tzafrir | sure |
| 13:30.09 | hal2385612 | I am trying to install zaptel on centos 44 |
| 13:30.17 | hal2385612 | from |
| 13:30.18 | hal2385612 | source |
| 13:30.36 | hal2385612 | zaptel 1.4.0 |
| 13:31.12 | hal2385612 | there is a patch in the download directory at http://ftp.digium.com/pub/zaptel/ |
| 13:32.22 | hal2385612 | should I apply this patch, and how do I know what arguments to use with the patch commend (ie -p0 or -p1 or -p2 etc) |
| 13:32.27 | Grnd-Wire | tzafrir: yeah, well ulaw is bloated, considering all of those 8khz voice channels.. and I'm not aware of a turnkey CD ripper that creates GSM files... |
| 13:32.52 | dj-fu | hal2385612, you know by what directory you are in and what the patch is trying to patch |
| 13:33.07 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 13:33.07 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 13:33.42 | hal2385612 | yes, cd to the /tmp/zaptel-1.4.0 directory and run patch -p? < ../zaptel-patch |
| 13:33.48 | hal2385612 | I think this is correct, isn't it? |
| 13:34.11 | dj-fu | depends on what is in zaptel-patch like I said |
| 13:34.21 | dj-fu | mos tpatches are generated in the root of the source being patched, yes |
| 13:34.22 | dj-fu | but some are not |
| 13:34.30 | tzafrir | Grnd-Wire, if you'll rip it directly to mp3, you'll get CD-quality mp3. It will probably be bigger than the "uncompressed, bloated" ulaw file |
| 13:34.47 | tzafrir | hal2385612, why not build zaptel 1.4.0? |
| 13:35.23 | hal2385612 | tzafrir - yes, I am building zaptel. but I thought you had to patch it before you configure and make? |
| 13:35.37 | tzafrir | what patch, exactly? |
| 13:35.58 | Grnd-Wire | tzafrir: meh.. Then you didn't set your options correctly on your ripper. |
| 13:36.00 | hal2385612 | this one, tza - http://ftp.digium.com/pub/zaptel/zaptel-1.4.0-patch.gz |
| 13:36.07 | tzafrir | Grnd-Wire, and then you'll have to transcode it in runtime |
| 13:36.32 | Grnd-Wire | tzafrir: Now that I won't dispute.. |
| 13:36.41 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 13:36.50 | *** join/#asterisk Gr1ncheux (n=devine@unaffiliated/gr1ncheux) |
| 13:37.10 | tzafrir | hal2385612, no. That patch is an alternative to the tarball. |
| 13:37.11 | hal2385612 | Grndwire - I am sure you can convert rcp to gsm on the command line. I used Windows recorder to create my sound files and then converted them |
| 13:37.23 | hal2385612 | oh, really, tsa? |
| 13:37.25 | hal2385612 | tza? |
| 13:37.53 | hal2385612 | My problem is that I never know how to find the information that would tell me that - do you have any suggestions? |
| 13:38.31 | hal2385612 | Grndwire - I meant pcm not rcp |
| 13:39.12 | tzafrir | hal2385612, you simply don't need the patch. The rest of what you did is fine |
| 13:41.01 | hal2385612 | TZA - I really appreciate your advice - thank you. I would, though, like to be self suffiecient. What methods do you use in my situation, to find out what the correct procedure is? |
| 13:43.46 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-72-65-23-44.bflony.east.verizon.net) |
| 13:44.02 | uwe | um, im trying to use agents, can anyone point a good documentation about agents in asterisk, i managed so far to log it in and out to queues, but apparently, its not being called when a queue is called! and i cant find anything about editing queues to accept or not agents (so far) |
| 13:45.32 | *** join/#asterisk IPmonger (n=ipmonger@c-68-84-208-206.hsd1.pa.comcast.net) |
| 13:45.48 | hal2385612 | http://www.voip-info.org/wiki-Asterisk+agents |
| 13:46.08 | hal2385612 | http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/c1294.html |
| 13:46.35 | hal2385612 | uwe - if you explore these two sites, you will be able to find the answer |
| 13:47.03 | hal2385612 | I found them very informative when I last configured asterisk a could of years ago |
| 13:47.19 | uwe | thank you, maybe ill be lucky with asteriskdocs :) |
| 13:47.47 | hal2385612 | also, google for "asterisk howto agents" |
| 13:47.58 | tzafrir | hal2385612, generally voip-info.org is the best single place I can think of |
| 13:49.00 | hal2385612 | yes, me too, but sometimes people document their experiences and call them HOWTO or Tutorial |
| 13:49.06 | hal2385612 | which can be useful |
| 13:49.30 | hal2385612 | oh, do you mean, regarding my question, tza? |
| 13:49.46 | hal2385612 | but how did you know the purpose of the patch file? |
| 13:51.29 | hal2385612 | Grnd-wire - this is the command I used to covert sound files to GSM - |
| 13:51.31 | hal2385612 | sox -c1 $BASEFILENAME.wav -r 8000 $BASEFILENAME.gsm |
| 13:51.48 | hal2385612 | in case it is useful |
| 13:53.33 | *** join/#asterisk evgeni (i=evgeni@212-7-20-235.stv.ee) |
| 13:53.37 | hal2385612 | are you still there, tza? |
| 13:54.37 | hal2385612 | My main problem is that I recieve the following errors when installing zaptel |
| 13:54.47 | hal2385612 | file /usr/lib/gcc/i386-redhat-linux/3.4.6 from install of zaptel-1.4.0-1 conflicts with file from package libstdc++-devel-3.4.6-3 |
| 13:54.57 | hal2385612 | file /usr/libexec/gcc/i386-redhat-linux/3.4.6 from install of zaptel-1.4.0-1 conflicts with file from package cpp-3.4.6-3 |
| 13:55.00 | hal2385612 | etc |
| 13:55.24 | hal2385612 | the problem is that 3.4.6 is a directory |
| 13:55.37 | hal2385612 | and should not be overwritten |
| 13:55.51 | tzafrir | hal2385612, /usr/lib/gcc/i386-redhat-linux/3.4.6 from install of zaptel-1.4.0-1? |
| 13:56.01 | hal2385612 | yes, that is correct |
| 13:56.05 | tzafrir | What package is it, exactly? |
| 13:56.19 | hal2385612 | what do you mean, tza? |
| 13:56.26 | hal2385612 | do you mean the rpm? |
| 13:56.41 | evgeni | Hello, maybe i have a dumb question, but i cant find any information, about connecting asterisk 1.4 with cisco as5300 with radius authorization |
| 13:57.14 | evgeni | maybe someone know any good links exept google and cisco.com |
| 13:57.34 | hal2385612 | evgeni |
| 13:57.36 | hal2385612 | http://www.google.co.uk/search?q=asterisk+cisco+as5300&source-id=Mozilla%20Firefox&start=0 |
| 13:57.44 | hal2385612 | Surely there is something there? |
| 13:58.15 | hal2385612 | or here - http://www.google.co.uk/search?hl=en&q=%2Basterisk+cisco+%2Bas5300+%2Bradius&btnG=Search&meta= |
| 14:01.13 | *** join/#asterisk orkid (n=orkid@74.13.35.98) |
| 14:03.17 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 14:07.33 | *** join/#asterisk lowlevel (n=Stuart@CPE00145176d140-CM000f9f1e356e.cpe.net.cable.rogers.com) |
| 14:08.13 | hal2385612 | is anyone able to offer me any advice about the problem I have compiling and installing zaptel, please? |
| 14:08.59 | uwe | hmmm |
| 14:09.53 | hal2385612 | also does anyone know of the IRC command to search for a topic when the /list command is performed? When I do /list on its own, I get too many to read, so I would like to grep or filter them |
| 14:11.53 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) |
| 14:12.08 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 14:12.22 | dj-fu | depends on the network |
| 14:12.26 | dj-fu | usually /list keyword |
| 14:12.27 | dj-fu | suffices |
| 14:12.39 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 14:15.10 | *** join/#asterisk ttrygve (n=trelvik@207.54.140.109) |
| 14:18.20 | *** join/#asterisk Gr1ncheux (n=devine@unaffiliated/gr1ncheux) |
| 14:18.54 | ChkDigit | dj-fu: Which IRC client do you use? |
| 14:19.56 | *** join/#asterisk littleball (n=littleba@bb220-255-152-63.singnet.com.sg) |
| 14:19.56 | dj-fu | xchat |
| 14:20.04 | dj-fu | I gave up on irssi. I use way too many networks |
| 14:20.05 | hal2385612 | I use chatzilla with Firefox |
| 14:20.37 | hal2385612 | so there is no command to search or filter the channel list? |
| 14:21.02 | dj-fu | do /quote list KEYWORD |
| 14:21.11 | dj-fu | list is server specific, it's nothing to do with the client |
| 14:21.39 | ChkDigit | window->channel list... |
| 14:21.45 | ChkDigit | from the menu. |
| 14:22.08 | ChkDigit | There is a regex search that the client does on the server list. |
| 14:22.14 | dj-fu | cool |
| 14:22.17 | dj-fu | didn't even know about that |
| 14:22.22 | dj-fu | like I said, I jus tuse /list |
| 14:22.39 | ChkDigit | yup. |
| 14:23.09 | hal2385612 | Mmm, that doesn't work for me... /quote list '*rpm*' |
| 14:23.22 | hal2385612 | just gives me my previous list |
| 14:23.41 | dj-fu | I get nothing back from /list rpm |
| 14:24.09 | Chris-NB | hi |
| 14:24.15 | Chris-NB | anyone familiar with mISDN? |
| 14:24.35 | Chris-NB | I want to configure a Beronet BN8s0 to use port 1-6 as TE and 7 and 8 as NT |
| 14:24.39 | hal2385612 | no, those list commands don't work for me |
| 14:24.41 | Chris-NB | is this possible? |
| 14:24.59 | hal2385612 | could you give me an example of /quote list that returns a list of channels? |
| 14:25.53 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:26.22 | hal2385612 | Chris, how about Handbook for installaion of Digium and beroNet interface cards |
| 14:26.27 | hal2385612 | http://www.beronet.com/download/card_installation_guide.pdf |
| 14:26.57 | hal2385612 | and http://www.voip-info.org/wiki/view/Asterisk+mISDN+channels |
| 14:28.04 | hal2385612 | Chris - are you proficient at using IRC? Do you know how to filter or search the /list command for specific words or topics? |
| 14:28.33 | littleball | hello, i have installed asterisk in my notebook. (linux or mac os). is it possible to call through asterisk + fixline? |
| 14:29.08 | hal2385612 | littleball - you need a telephony interface |
| 14:30.12 | hal2385612 | I don't think there are any made by digium for laptops http://www.digium.com/en/products/hardware/analogcards.php |
| 14:32.31 | hal2385612 | You could connect to a VOIP telephone provider via AIX2 (or whatever the proprietry protocol is called, I can;t remember) and dial landlines in this way |
| 14:32.53 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:32.53 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:33.06 | littleball | ha12385612, i am testing something |
| 14:33.07 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 14:33.07 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 14:33.10 | Chris-NB | hal2385612, what do you mean with proficient at using IRC ? |
| 14:33.31 | hal2385612 | Chris, do you know how...to filter or search the /list command for specific words or topics? |
| 14:33.34 | littleball | before i connect to provider through e1 |
| 14:34.07 | hal2385612 | The protocol is IAX, sorry! |
| 14:34.36 | Chris-NB | hal2385612, to find a specific channel, concerning an interesting word or topic? |
| 14:34.49 | hal2385612 | yes, please |
| 14:34.59 | *** join/#asterisk Deep6 (n=jhenkel@208.38.35.162) |
| 14:35.02 | littleball | it seems that i need to add FXO interface to my notebook. who can recommend? |
| 14:35.05 | Deep6 | guys I have a problem with a cisco 7912 not staying registered with the server...if I power it off and back on again it works.. |
| 14:35.10 | littleball | one port is enough |
| 14:35.22 | Deep6 | it's a 7912 running sip |
| 14:35.24 | Deep6 | I should add |
| 14:35.36 | Deep6 | is there the concept of sip keep alives or something? |
| 14:35.46 | Chris-NB | hal2385612, ? what please? I don't understand |
| 14:35.47 | hal2385612 | littleball - I used to use http://www.voiptalk.org/products/index.html, but there may be a better one now |
| 14:36.27 | simplexio | has enyone used openpbx? whats the difference between asterisk 1.2 , 1.4 and openpbx ? |
| 14:36.37 | hal2385612 | Do you know the IRC command to search the channels for a specific word, as when running the command /list, it just lists everything and it is impossibel to find anything |
| 14:37.11 | hal2385612 | Littleball - what do you mean, one port is enough? |
| 14:37.30 | simplexio | <PROTECTED> |
| 14:38.02 | littleball | ha123, because it is just for testing. i only need to make a call through asterisk and my home fix line |
| 14:38.04 | hal2385612 | Littleball - I just told you that I do not believe that there are any analogue cards available for laptops |
| 14:38.24 | hal2385612 | did you check out the link I posted? |
| 14:38.40 | dexter_speed | has anyone had problems, connecting a fax machine to an fxs card, dialing out via fxo card ? |
| 14:38.52 | dexter_speed | where the fax machine gets to call the others side but fax communication fails ? |
| 14:39.38 | littleball | checking now |
| 14:39.48 | hal2385612 | Littleball - connect to a VOIP service provider, preferably over IAX2 and make calls that way |
| 14:40.11 | littleball | ha12385612, i see. but i am trying some special things |
| 14:41.12 | *** join/#asterisk yassine (n=yassine@dsl.voicint.com) |
| 14:41.24 | hal2385612 | littleball - see http://www.voiptalk.org/products/voip-comparison.html |
| 14:42.48 | littleball | ha12385612, in the last link, what is SMS mean? |
| 14:42.48 | hal2385612 | there may be cheaper US Providers. But this is the cool thing about VOIP, you can configure your PBX to use a provider that is local to the destination of your phonecall |
| 14:43.01 | hal2385612 | Mobile phone messaging |
| 14:43.07 | littleball | how does it work? |
| 14:43.18 | littleball | i am testing this . :-) |
| 14:43.39 | littleball | this is the reason i want to buy a fxo for testing |
| 14:43.51 | hal2385612 | well, you will have to read their documentation - I don't know what services they provide with regard to SMS |
| 14:44.19 | *** join/#asterisk apardo (n=apardo@87.217.144.104) |
| 14:44.22 | hal2385612 | well, just buy a cheap PC - any P4 or higher will do, and set up Asterisk on that |
| 14:45.08 | hal2385612 | see Mark's tutorial at http://www.youtube.com/watch?v=ONOxNJquatk to quickly be up and running |
| 14:45.38 | littleball | ok. maybe this is the only way... |
| 14:46.17 | *** join/#asterisk DaveCanoe (n=Dave@dhcp3022.nanog.merit.net) |
| 14:46.50 | hal2385612 | simplexo - sorry for the delay - /list #channelname would only be useful if you already know the channelname |
| 14:47.03 | hal2385612 | I would like to find channels that may meet my requirements |
| 14:47.26 | *** join/#asterisk Dovid (n=Dovid@85.159.160.207) |
| 14:48.54 | *** join/#asterisk mercestes (n=merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 14:49.00 | hal2385612 | I wish someone would help me now? |
| 14:49.08 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-221.ph.ph.cox.net) |
| 14:49.27 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 14:49.31 | Dovid | hal: whats the issue ? |
| 14:49.39 | hal2385612 | hi Dovid |
| 14:49.41 | mercestes | Ok, hal23856...err..man with an annoying username. What do you need? |
| 14:49.53 | hal2385612 | HAHA - thanks merc! |
| 14:49.57 | mercestes | :D |
| 14:50.08 | hal2385612 | Simply "Hal" will do |
| 14:50.30 | hal2385612 | I am trying to install the zaptel modules on Centos44 |
| 14:50.50 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 14:50.56 | mercestes | You know about "the bug" right? |
| 14:50.57 | hal2385612 | after I compile, I get conflicts |
| 14:51.21 | Dovid | ~centosbug |
| 14:51.28 | jbot | from memory, centosbug is a problem with the 2.6.9-42 kernels prior to 2.6.9-42.0.1. If you can't compile zaptel, do a 'yum update', you're running an old kernel. If you HAVE to run an old kernel, the fix is "sed -i s/rw_lock/rwlock/ /usr/src/kernels/*/include/linux/spinlock.h" |
| 14:51.28 | *** join/#asterisk CrashSys (n=kumba@208.177.233.66.ptr.us.xo.net) |
| 14:52.13 | hal2385612 | My kernel is 2.6.9-42.0.8.EL |
| 14:52.22 | rue_mohr | is my T1 card a tdm? |
| 14:52.35 | rue_mohr | it isn't is it |
| 14:53.24 | Dovid | wha error r u gettin ? |
| 14:53.30 | hal2385612 | I get the follow errors (about 15 of them) when installing the RPM I created using checkinstall: |
| 14:53.32 | hal2385612 | file /usr/lib/gcc/i386-redhat-linux/3.4.6 from install of zaptel-1.4.0-1 conflicts with file from package libstdc++-devel-3.4.6-3 |
| 14:53.40 | *** join/#asterisk ucionas (i=utis@diedas.soften.ktu.lt) |
| 14:53.42 | hal2385612 | file /usr/libexec/gcc/i386-redhat-linux/3.4.6 from install of zaptel-1.4.0-1 conflicts with file from package gcc-g77-3.4.6-3 |
| 14:53.46 | mercestes | omg |
| 14:53.48 | hal2385612 | file /usr/libexec/gcc/i386-redhat-linux/3.4.6 from install of zaptel-1.4.0-1 conflicts with file from package gcc-java-3.4.6-3 |
| 14:53.48 | mercestes | pastebin |
| 14:53.49 | hal2385612 | etc |
| 14:53.50 | mercestes | ~pastebin |
| 14:53.52 | jbot | well, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or http://rafb.net/paste/ |
| 14:54.24 | hal2385612 | was that for me, merc? Sorry if I have acted improperly |
| 14:54.40 | CrashSys | I'm having an issue with Asterisk using 160% of CPU (according to top) on a Dual-Core 2.6ghz system to ring CONSOLE/dsp and two Sip extensions. I'm using the Conole/dsp/answer and noanswer patch, and the OSS Wrapper for Alsa (My sound card isn't supported under OSS). |
| 14:54.43 | mercestes | If you wanna list your errors dump them to pastebin |
| 14:55.04 | mercestes | it looks like your gcc is incompatible with whatever zaptel you are trying to install |
| 14:55.37 | hal2385612 | http://pastebin.ca/341315 |
| 14:55.41 | mercestes | thanks |
| 14:55.42 | hal2385612 | that's cool. merc! |
| 14:57.10 | mercestes | first is /usr/libexec/gcc/i386-redhat-linux/3.4.6 a file or a directory? |
| 14:57.20 | hal2385612 | well, I am completely uptodate with my yum updates |
| 14:57.37 | mercestes | I can see that. I *think* that might be hurting you tho |
| 14:57.59 | hal2385612 | 3.4.6 is a directory |
| 14:57.59 | mercestes | did you update gcc recently? |
| 14:58.07 | mercestes | yea, I thought so. |
| 14:58.07 | *** join/#asterisk misc-- (n=misc@203.87.183.134) |
| 14:58.07 | hal2385612 | um, let me check |
| 14:59.22 | ucionas | any reference to random hunt group? |
| 14:59.31 | ucionas | thanks |
| 15:00.18 | hal2385612 | yes, merc, I can see that gcc was updated at the start of Jan - Jan 02 15:03:20 Updated: gcc.i386 3.4.6-3 |
| 15:01.08 | mercestes | Have you tried to install anything else? |
| 15:01.10 | hal2385612 | and libgcc - Jan 02 15:00:12 Updated: libgcc.i386 3.4.6-3 |
| 15:01.27 | hal2385612 | yes, my system has been working fine |
| 15:02.06 | mercestes | so you've been able to install other packages? |
| 15:02.28 | hal2385612 | yes, I have merc |
| 15:04.01 | hal2385612 | one thing to note is that I exlude the following from my yum updates: |
| 15:04.29 | hal2385612 | exclude=kernel kernel-devel kernel-doc kernel-smp* kernel-hugemem* kernel-largesmp* mozilla* seamonkey* devhelp* |
| 15:04.58 | mercestes | try rolling back gcc one step |
| 15:05.16 | mercestes | for some reason Zaptel is trying to place files onto your system that wish to overwrite or replace their gcc components |
| 15:05.54 | hal2385612 | Yes, I realised that, merc - why would it try to replace gcc files, do you think? |
| 15:06.02 | mercestes | Zaptel does patch and recompile your kernel source. |
| 15:06.23 | hal2385612 | maybe there is a ./configure option that would disable this behavious |
| 15:06.26 | hal2385612 | maybe there is a ./configure option that would disable this behaviour? |
| 15:07.18 | hal2385612 | I am sorry, but I have not yet fully investigated how to find out what all the configure options are and what they are for. Help on this would be appreciated! |
| 15:07.40 | mercestes | Well, it is possible that these "conflicts" are supposed to exist and are normally blind to you, me and everyone else. |
| 15:08.54 | hal2385612 | well, the RPM won't install with them. I don't think they are intended, as the problem is that zaptel want to replace a directory, rather than individual files |
| 15:09.31 | mercestes | well, I'm hearing that your RPM is hosed. |
| 15:10.18 | hal2385612 | really? the problem is with the RPM rather than the compiled files? |
| 15:10.36 | mercestes | hal2385612: Frmo what I get in #centos |
| 15:10.56 | hal2385612 | Sorry, merc, I don't follow you |
| 15:11.03 | hal2385612 | coudl you elaborate, please? |
| 15:11.09 | *** join/#asterisk J4k3 (i=jsuter@dhcp-12-197-128-58.intrastar.net) |
| 15:11.14 | mercestes | I asked in #CentOS and I'm hearing that your RPMs are hosed and Zaptel should not be trying to replace your gcc files. |
| 15:11.57 | hal2385612 | what does "RPMs are hosed" mean? |
| 15:12.04 | mercestes | are you installing from an RPM? |
| 15:12.10 | hal2385612 | sorry for being ignorant |
| 15:12.35 | mercestes | You don't have to apologize. I had to ask someone else. ;) |
| 15:12.45 | mercestes | Your RPMs are broken. |
| 15:12.52 | hal2385612 | well, I did a ./configure and then a checkinstall (which performs a make and then builds an RPM) |
| 15:12.54 | mercestes | download new RPMs or try installing from source instead of from RPMs |
| 15:13.02 | mercestes | it's possible that the RPMs you are using are not meant for yoru distro |
| 15:13.19 | mercestes | ok, well, that's broken. |
| 15:13.27 | *** part/#asterisk jmls (n=asterisk@62.49.235.130) |
| 15:13.34 | hal2385612 | I see - so if I go ahead and ./configure / make / make install then it won't corrupt my install of gcc? |
| 15:14.00 | hal2385612 | what happens when there is a conflict between make install and a file that already exists on the filesystem? |
| 15:14.10 | *** join/#asterisk Strom_M (n=strom@m0e5e36d0.tmodns.net) |
| 15:14.26 | hal2385612 | with RPM, it terminates with an error. Will make install stop rather than overwrite files? |
| 15:15.23 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:22.55 | *** join/#asterisk lenscape (n=lenscape@dale.gotadsl.co.uk) |
| 15:23.12 | lenscape | wow! busier channel than I was expecting |
| 15:23.44 | lenscape | the cost of the hardware is putting me off from using this for a phone system |
| 15:23.54 | *** join/#asterisk vvortex3 (i=Lanniste@71.193.3.146) |
| 15:24.03 | zoa | the hardware is pretty cheap |
| 15:24.32 | lenscape | the stuff referred to on the Asterisk web site? |
| 15:24.32 | vvortex3 | can someone point me in the right direction as to how to customize the greetings and options in VoiceMailMain() |
| 15:27.15 | *** join/#asterisk GiantPickle (n=GiantPic@S01060016b600537f.gv.shawcable.net) |
| 15:29.12 | uwe | argh! a little question, generally, when i have an agent, that joins a queue (1*, 1002#) , do i have to do something else to make it recive calls!! |
| 15:29.19 | tzafrir | lenscape, test with soft phones ;-) |
| 15:29.44 | tzafrir | hal2385612, the location of that file you mentioned doesn't make sense |
| 15:29.57 | tzafrir | Maybe it is a badly-packages rpm package? |
| 15:30.10 | lenscape | tzafrir: this is for a home phone system. 2 lines, 4 phones |
| 15:30.47 | CrashSys | http://www.pastebin.ca/341335 (I'm getting kernel panics, anyone got any suggestions from this debug?) |
| 15:31.01 | tzafrir | lenscape, use an x100p and deal with the echo :-( |
| 15:32.06 | hal2385612 | tza - you can use the tune utility to stop the echo, I think |
| 15:33.00 | SuPrSluG | is it possible queue a meetme room? |
| 15:33.06 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 15:33.07 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 15:34.06 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 15:34.06 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:34.13 | SuPrSluG | where the conf acts as the agent |
| 15:34.25 | *** join/#asterisk drako (n=ljd@unaffiliated/luisjose) |
| 15:34.28 | zoa | it is |
| 15:34.35 | zoa | with some voodoo |
| 15:34.50 | hal2385612 | Tzafrir - thank you for continuing with my issue - what file did you mean did not make sense? |
| 15:34.54 | file | zoa! |
| 15:35.00 | zoa | ah mr file |
| 15:35.31 | tzafrir | hal2385612, where is that rpm from? |
| 15:36.17 | lenscape | tzafrir: I'll read up. Thanks |
| 15:36.18 | misc-- | why would my external sip devices that are connecting to my asterisk server appear to come from my asterisk server's gateway rather than the the IP of the sip device? |
| 15:36.19 | SuPrSluG | zoa, i'll need chicken bones and a hair of a bat? |
| 15:36.26 | *** join/#asterisk amdtech (i=amdtech@nat/digium/x-7c8e902f1bb3c204) |
| 15:36.47 | hal2385612 | mercestes may have discussed this earlier. It is zaptel, telephony drivers from digium |
| 15:36.47 | *** part/#asterisk lenscape (n=lenscape@dale.gotadsl.co.uk) |
| 15:36.56 | *** join/#asterisk [Airwolf] (n=airwolf@84.241.216.237) |
| 15:37.07 | Ethon | <PROTECTED> |
| 15:37.13 | hal2385612 | I did a ./configure, and then checkinstall, and recieved the errors at http://pastebin.ca/341315 |
| 15:37.14 | Ethon | sorry, wrong window |
| 15:38.00 | hal2385612 | merc - that may be a setting in the softphone, I think |
| 15:38.30 | SuPrSluG | misc--,you may need to give external ip and lan info in sip.conf |
| 15:38.59 | *** join/#asterisk ctaloi (n=ctaloi@nat-66-218-1-41.usadatanet.com) |
| 15:39.00 | misc-- | SuPrSluG: I don't have a LAN IP, just a public IP |
| 15:39.13 | SuPrSluG | no nat? |
| 15:39.20 | misc-- | no nat at all |
| 15:39.31 | tzafrir | hal2385612, I really have no idea why the zaptel install target looks there |
| 15:39.38 | tzafrir | there: usr/lib/gcc/i386-redhat-linux/3.4.6 |
| 15:39.44 | tzafrir | Looks strange indeed |
| 15:39.55 | tzafrir | I agree that installing with checkinstall makes sense |
| 15:39.58 | misc-- | although... I did setup a bridge device on my linux gateway - this is where asterisk thinks the connections are coming from but there are literally no nat rules. And, when I connect to the public IP by ssh for example, it shows I'm coming from my real IP |
| 15:40.27 | tzafrir | (checkinstall is a wrapper to 'make install' that generates an rpm/deb/whatever from the installed files) |
| 15:40.27 | hal2385612 | tza - it may be because I am using checkinstall rahter than make install (as I want an rpm so that I can uninstall it in future). |
| 15:40.35 | SuPrSluG | shorewall? |
| 15:40.47 | misc-- | no.. just another Centos box |
| 15:41.01 | SuPrSluG | straigt iptables? |
| 15:41.04 | hal2385612 | yes, do you think, if I run make install, it may automatically overwrite the files, or will it terminate with an error, as rpm did? |
| 15:41.17 | misc-- | I have eth0 which is my public IP, then eth1 is internal and eth2 goes to a switch with asterisk connected - eth0 and eth2 are bridged into a logical subnet that hosts the public IP |
| 15:41.41 | tzafrir | hal2385612, at this point I would start debugging: bash -x, strace -f, debug flags, or whatever |
| 15:41.57 | misc-- | I am only natting the internal, definitly nothing else though |
| 15:42.03 | tzafrir | (because I have no specific clue) |
| 15:43.26 | SuPrSluG | we just did a setup similar. we use shorewall to give virtual interfaces and map public ip's to internal lan ip's |
| 15:43.38 | misc-- | oh right, and how did that pan out? |
| 15:43.41 | BrokenNoze_ | anyone used the Zxytel Prestige 2000W? any good? |
| 15:43.45 | misc-- | did you have external sip devices connecting to it? |
| 15:43.52 | SuPrSluG | yours sounds more like a dmz setup |
| 15:44.00 | misc-- | kind of |
| 15:44.52 | hal2385612 | ok, well, I appreciate your advice until now, thank you for your effort. i shall go to the centos room, to see if they know |
| 15:44.55 | SuPrSluG | yes. it was a challenge to get it recognize the correct public ip's from external sip devs. |
| 15:45.10 | *** join/#asterisk marv[work] (n=timr@24.214.206.254) |
| 15:45.15 | misc-- | how did you do it!? It works on one external phone but not another - exact same config |
| 15:45.22 | SuPrSluG | try setting external public ip in sip.conf. |
| 15:45.25 | misc-- | maybe sip.conf does need to specify the public IP |
| 15:45.35 | misc-- | ok, will give that a go |
| 15:46.14 | SuPrSluG | i guess you won't need to define a lan. as it's in a dmz |
| 15:46.27 | misc-- | yeah that's right |
| 15:46.35 | SuPrSluG | give it a whirl. won't hurt anything |
| 15:46.37 | misc-- | it has a route to our internal LAN but it only has one IP |
| 15:46.40 | misc-- | yep |
| 15:51.25 | misc-- | well, that didn't work :/ |
| 15:53.14 | uwe | can anyone help/advise what this means: rtp.c: Got RTCP report of 132 bytes |
| 15:53.35 | *** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com) |
| 15:54.07 | uwe | bad? normal? |
| 15:54.22 | *** join/#asterisk GiantPickle (n=GiantPic@S01060016b600537f.gv.shawcable.net) |
| 15:54.31 | jeremy_g | what is the current key lenght for the unbreakable security? is it 4k? |
| 15:54.39 | *** join/#asterisk umop3plsdn (n=da@24-38-58-10-st.lndnnh.adelphia.net) |
| 15:56.23 | *** join/#asterisk penguinFunk (n=penguin@87.224.86.46) |
| 15:57.54 | penguinFunk | hi guys, anyone had problems with using asterisk + FXO gateway digium card...The system is up and running beautifully, but when users try to access automated systems where you are offered a range of options and have to dial particular numbers to get particulr extensions then the numbers you dial are not recognised |
| 15:59.01 | penguinFunk | its as if any dialled numbers are ignored once the phone call has been established |
| 15:59.02 | penguinFunk | is this an asterisk thing? is it fixable ? |
| 15:59.02 | *** join/#asterisk Strom_M (n=strom@m055e36d0.tmodns.net) |
| 15:59.39 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
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| 16:00.58 | *** join/#asterisk b11d (n=no@234-200-29-134.hcc.mnscu.edu) |
| 16:01.01 | *** join/#asterisk Ebola (n=Ebola@host81-151-91-139.range81-151.btcentralplus.com) |
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| 16:03.25 | mog | penguinFunk, does this happen with all ivrs? and is asterisk getting your dtmf presses correctly, turn on dtmf logging in logger.conf |
| 16:05.24 | hal2385612 | perguin - are the volumes set correctly? |
| 16:05.31 | *** join/#asterisk russellb (i=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 16:05.31 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:05.40 | hal2385612 | try running fxotune before anything else |
| 16:06.17 | penguinFunk | we have fxo'tuned the lines |
| 16:06.33 | penguinFunk | i will do that thanx mog, i dont have another ivr to test with at the moment |
| 16:06.50 | penguinFunk | anyone know of any ivr's i can dial to test ? |
| 16:06.59 | mog | digiums ^_^ |
| 16:07.04 | mog | 1877-linux-me |
| 16:07.11 | penguinFunk | :] |
| 16:07.13 | penguinFunk | thx |
| 16:07.20 | penguinFunk | whats the calling code for america ? |
| 16:07.38 | De_Mon | Asterisk isn't setting the destination correctly when exchange 2007's call is proxied thru openSER. Is this an openSER issue or should asterisk be handing the call differently? http://pastebin.ca/341366 |
| 16:07.39 | Qwell[] | 1 is the country code |
| 16:08.49 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 16:11.21 | penguinFunk | well digiums ivr is working fine |
| 16:11.39 | penguinFunk | must be some sort of incompatability with this particular ivr |
| 16:13.39 | penguinFunk | incompatibility even |
| 16:14.09 | mog | well id reccomend calling it and have dtmf debug make sure that its sent |
| 16:14.29 | *** join/#asterisk GiantPickle (n=GiantPic@S01060016b600537f.gv.shawcable.net) |
| 16:15.02 | penguinFunk | okay thank you |
| 16:15.41 | mog | but it sounds like it is working, just is an odd ivr |
| 16:16.41 | *** join/#asterisk umop3plsdn (n=da@24-38-58-10-st.lndnnh.adelphia.net) |
| 16:17.45 | *** join/#asterisk lenne_dk (n=Miranda@83.72.129.7.ip.tele2adsl.dk) |
| 16:18.14 | misc-- | SuPrSluG: don't suppose you have any other suggestions? |
| 16:19.27 | *** join/#asterisk topping (n=topping@204.152.96.238) |
| 16:19.28 | SuPrSluG | misc--,what is your linux box using for a firewall? |
| 16:19.36 | misc-- | just iptables |
| 16:20.07 | misc-- | very basic setup - currently allowing all traffic, forwarding through the different subnets at will |
| 16:20.49 | CrashSys | Looking at this kernel dump http://www.pastebin.ca/341335 , would it be correct to say that the snd_pcm_plugin (which I believe is part of the kernel) is causing my kernel panic? |
| 16:21.08 | CrashSys | or is snd_pcm_plugin some portion of asterisk? |
| 16:21.10 | penguinFunk | dtmf log is showing the tones as being sent |
| 16:21.39 | misc-- | what's weird is that if I ssh into asterisk and check where I'm coming from, it's all expected - this to me sounds like an asterisk bug. Not only that, but I can't get an iax2 trunk working to my server on the real internal IP - I have to use the VPN IP. Of course, I can ssh and connect to the internal IP from any other network app. Asterisk has some bug in the networking I am convinced 100% |
| 16:23.09 | *** join/#asterisk J4k3 (i=jsuter@165.sub-70-216-188.myvzw.com) |
| 16:23.44 | SuPrSluG | misc--, you should have an entry like: ACCEPT net $FW udp 4569,5060,10000:20000 |
| 16:24.31 | *** join/#asterisk Lann (i=Lanniste@adsl-63-200-88-82.dsl.scrm01.pacbell.net) |
| 16:24.42 | Lann | asterisk owns |
| 16:24.52 | mercestes | it does. |
| 16:24.56 | mercestes | all our base r asterisk's. |
| 16:24.59 | Qwell[] | What does it own? |
| 16:25.02 | *** join/#asterisk GiantPickle (n=GiantPic@S01060016b600537f.gv.shawcable.net) |
| 16:25.03 | Lann | it owns my pbx |
| 16:25.22 | mercestes | CrashSys: It seems like an alsa thing to me. |
| 16:25.35 | lenne_dk | I'm missing an acknowledgement message when I have made a successful transfer. I find it a little negative just to get a busy tone. I'd like alison saying "Transfer succesful" or something. |
| 16:25.38 | misc-- | SuPrSluG: well, I am accepting everything on that IP. I'm not denying anything at all either |
| 16:25.44 | mercestes | CrashSys: snd being sound |
| 16:26.01 | Lann | coming back to the office, 2 people lost their offices...fun day of hearing bitching today |
| 16:27.40 | Qwell[] | they "lost" them? |
| 16:27.51 | Lann | so...how do you customize VoiceMailMain to have custom prompts and menus? |
| 16:28.02 | Qwell[] | Lann: hack the code |
| 16:28.02 | Lann | yeah they got put on the main floor |
| 16:28.10 | Lann | what code |
| 16:28.18 | Qwell[] | we've got plans to eventually make it all dialplan, but... |
| 16:28.33 | Lann | where is the voicemailmain code located |
| 16:28.38 | mercestes | Ooo, VMMain dialplan based? |
| 16:28.47 | mercestes | YAY! that would rock! Eewww....that would suck |
| 16:28.49 | Lann | that'd be cool |
| 16:29.03 | Lann | well in my case its absolutely necessary |
| 16:29.21 | mercestes | So they no longer want "press 3 for mail box options."? |
| 16:29.31 | Lann | there is no they |
| 16:29.35 | Qwell[] | Lann: apps/app_voicemail.c |
| 16:29.40 | *** join/#asterisk itfdude (n=carey@cpe-72-179-59-233.austin.res.rr.com) |
| 16:29.42 | Lann | hmm...irritating |
| 16:29.51 | Lann | but doable |
| 16:30.27 | mercestes | hrm. What has happened to us? |
| 16:30.33 | Lann | i need to change the prompts....i could probably get away with just that |
| 16:30.33 | mercestes | Why must we submit to this notion of choice? |
| 16:30.54 | Lann | wait no...i need to change the prompts and have delete message actually hit a script |
| 16:31.15 | *** join/#asterisk map7 (n=map7@teksup41.lnk.telstra.net) |
| 16:31.16 | mercestes | Why is it like that? Because that's the way it is, bitches! Deal with it! |
| 16:31.37 | mercestes | Gah, ppl are so lazy. "it's too hard to learn new stuff....I want it the *OLD* way." |
| 16:31.51 | Lann | hahaha |
| 16:31.53 | mercestes | and then I tell them, "Pay me your old bill, and I will make it exactly like the old way." |
| 16:32.00 | mercestes | then they shut up. |
| 16:32.11 | mercestes | and all is happiness. |
| 16:32.20 | *** join/#asterisk amdtech (i=amdtech@nat/digium/x-de2a85d9e2850d56) |
| 16:32.42 | Lann | i also need to figure out how i can generate voicemails by script ... :-( |
| 16:32.49 | CrashSys | no kidding... people pay me to do an * install and all, and want me to make it work exactly like their old AT&T partner system (yes it's that old) |
| 16:32.51 | Lann | by combining audio files |
| 16:33.07 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 16:33.07 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 16:33.16 | Lann | hahaha i think i have an at&t partner system here |
| 16:33.40 | CrashSys | well first they wanted it fully automated and all |
| 16:33.47 | CrashSys | 3 weeks later, they want it like their old system... |
| 16:34.03 | mercestes | and then they want it like their old system...but automated.. |
| 16:34.08 | CrashSys | cause evidently no one ever works at their desk, or checks voicemail... |
| 16:34.15 | mercestes | I wanna see AT&T deal with the crap we deal with. |
| 16:34.19 | CrashSys | mercestes: basically... |
| 16:34.38 | CrashSys | But now i've got a problem with chan_oss causing kernel panics |
| 16:34.45 | CrashSys | so now there's no shop ringer... heh |
| 16:35.08 | CrashSys | so i'm back to Chan_alsa and no shop ringer |
| 16:35.12 | CrashSys | which is fun |
| 16:35.14 | mercestes | Users: "my phones ring too much, too loud and too long, and it's hard to ignore. make it stop." End-Users: "No one answers their phone. I don't want menus, and voicemails, and crap! I want to dial *ONE* number and get exactly who I want immediately, with no ringing. Scratch that...I don't even want to have to dial a number." |
| 16:35.25 | CrashSys | nothing like trying to get some guy to hit the reset button on the server |
| 16:35.31 | mercestes | CrashSys: Alsa was causing your kernel panic btw. |
| 16:35.50 | CrashSys | alsa? |
| 16:35.54 | mercestes | yea |
| 16:35.58 | robl^ | mercestes: app_mindreader ;-) |
| 16:36.01 | mercestes | snd_pcm_plugin is an alsa thing. |
| 16:36.08 | mercestes | robl^: chan_telepathy! :D |
| 16:36.20 | CrashSys | I figured it was the OSS Plugin |
| 16:36.26 | mercestes | CrashSys: Maybe compile alsa into the kernel? |
| 16:36.29 | CrashSys | but I had jet to sift through the kernel makefiles to find out |
| 16:36.39 | CrashSys | Alsa is compiled, cause OSS only supports like a Soundblaster 16 |
| 16:36.46 | mercestes | wait wait. |
| 16:36.48 | mercestes | cvs.sourceforge.net/viewcvs.py/alsa/alsa-kernel/core/oss/pcm_plugin |
| 16:36.52 | mercestes | It is an oss thing. |
| 16:36.56 | *** join/#asterisk GiantPickle (n=GiantPic@S01060016b600537f.gv.shawcable.net) |
| 16:36.57 | *** join/#asterisk SwK[Work] (n=SwK@24.214.206.254) |
| 16:37.26 | *** part/#asterisk SwK[Work] (n=SwK@24.214.206.254) |
| 16:37.41 | mercestes | so yea, your right. =/ |
| 16:37.51 | Lann | you're |
| 16:38.05 | *** join/#asterisk SwK[Work] (n=SwK@24.214.206.254) |
| 16:38.14 | misc-- | so many people say your when they should say you're, I find it the most gramatical error |
| 16:38.15 | mercestes | thanks, Lann. |
| 16:38.28 | mercestes | misc--: The most gramatical error? |
| 16:38.30 | Qwell[] | grammatical |
| 16:38.36 | mercestes | quantify grammatical. |
| 16:38.37 | misc-- | just testing :o |
| 16:38.39 | CrashSys | The patch I got from Corydon to make CONSOLE/dsp/answer and CONSOLE/dsp/noanswer work applies only to OSS... |
| 16:39.32 | CrashSys | so now i'm back to scratchin' my head to make a shop ringer bell... |
| 16:39.44 | *** join/#asterisk Jason99 (n=jason@jason.unitz.ca) |
| 16:39.47 | Corydon-w | What's wrong with OSS? |
| 16:40.02 | CrashSys | The drivers for it only support like a Sound Blaster 16 |
| 16:40.19 | Corydon-w | but chan_oss isn't related to that |
| 16:40.21 | Lann | i wonder if i can find a walkthrough for diaplan based voicemail |
| 16:40.33 | Corydon-w | chan_oss will still work with chan_alsa, as long as you have /dev/dsp |
| 16:40.47 | mercestes | CrashSys: Just pastebin your kernel trace again..:) It'll save you some tmie. |
| 16:40.52 | Corydon-w | or rather, chan_oss will work with alsa |
| 16:40.54 | mercestes | it'll save you some time too. |
| 16:41.00 | CrashSys | http://www.pastebin.ca/341335 |
| 16:42.28 | Corydon-w | Ah |
| 16:42.56 | *** join/#asterisk zmef420 (n=zmef420@metarb3-pool3-130.mtco.com) |
| 16:42.57 | CrashSys | In order to use the chan_oss patch for console/dsp/noanswer (etc), I have to use the OSS API for Alsa, cause the native kernel OSS wont load my sound card... |
| 16:43.13 | Corydon-w | CrashSys: how do you figure? |
| 16:43.29 | CrashSys | Corydon: Cause it doesn't list AC97 or Intel Azalia hardware in the setup |
| 16:43.47 | Corydon-w | CrashSys: all chan_oss does is to access /dev/dsp. That's all. |
| 16:43.52 | *** join/#asterisk mut (n=ana@65.111.222.120) |
| 16:44.03 | mut | i have a sipura ata returning "busy here" when calling it |
| 16:44.12 | mut | but neither of the 2 ports are in use |
| 16:44.20 | mut | and there is no weird call forward settings in the ata or anything |
| 16:44.24 | mut | what else might it be? |
| 16:44.25 | CrashSys | corydon: well then i've got another problem then... |
| 16:44.40 | CrashSys | cause using chan_alsa doesn't bomb anything... but the patch chan_oss does... |
| 16:44.44 | SuPrSluG | CrashSys,try an overhead page |
| 16:44.47 | Corydon-w | CrashSys: what distro? |
| 16:44.51 | CrashSys | Slack 11.0 |
| 16:44.57 | Corydon-w | SuPrSluG: that's what he's trying to do |
| 16:45.05 | CrashSys | The paging i've got working |
| 16:45.15 | CrashSys | it ran for 3-weeks with chan_alsa and overhead paging fine... |
| 16:45.31 | CrashSys | it's when I tried to patch chan_oss and make a shop ringer that I get kernel panics |
| 16:45.46 | Corydon-w | CrashSys: you know that you can't load chan_alsa and chan_oss at the same time, right? |
| 16:45.57 | CrashSys | the dial plan is the same with the addition of 3-lines on my incoming context |
| 16:46.11 | CrashSys | corydon: yes, I explicitly load modules and only chan_oss was loaded at the time |
| 16:47.21 | CrashSys | so if chan_oss will with with an ALSA kernel without oss-specific extensions then maybe I should recompile the kernel without the OSS Support... |
| 16:47.31 | CrashSys | or is the oss support what makes /dev/dsp appear? |
| 16:47.45 | Corydon-w | No, just sound support, period |
| 16:48.15 | Corydon-w | /dev/dsp is simply the device used to access the soundcard |
| 16:48.26 | CrashSys | Right now the kernel only has ALSA compiled in it with OSS Compatibility API's for Alsa... |
| 16:48.39 | CrashSys | None of it is modular, all compiled in... |
| 16:49.02 | Corydon-w | Well, you could try modularizing |
| 16:49.19 | CrashSys | Hmmm... interesting idea... |
| 16:49.24 | Corydon-w | What was the point of making it monolithic? |
| 16:49.39 | CrashSys | incase of HD failure |
| 16:49.48 | CrashSys | or some portion of the OS became unreadable... |
| 16:49.48 | Corydon-w | Eh? |
| 16:49.54 | CrashSys | habit primarily |
| 16:50.02 | Lann | so im sure editing the .c file isnt the only way to get custom voicemail...wouldnt there be an AGI app? |
| 16:50.32 | robl^ | only if your write an AGI voicemail from scratch |
| 16:50.46 | Hmmhesays | and that would be a pain in the ass |
| 16:50.46 | Lann | damn damn damn |
| 16:50.46 | Corydon-w | In case of HD failure, you can't even be sure the kernel would load uncorrupted |
| 16:51.43 | Lann | all i want is the ability to record and retrieve messages, and then for each message the ability to delete,skip, or enter a value , and have each of those go to script |
| 16:51.56 | CrashSys | corydon: it's mostly habit... |
| 16:52.01 | Lann | not that that's simple |
| 16:52.23 | hal2385612 | if anyone is interested in the problem I posted earlier, I think this is the answer: "In order to compile the zaptel package on your system, you need to have the kernel source version matching the kernel version that is already running on your system, and you may also need to ensure that the version of gcc used to compile the kernel is the same version as what you have installed." |
| 16:54.04 | hal2385612 | My installation of gcc has been updated by yum, so this means that it now does not match the one that I used to compile the kernel |
| 16:55.55 | *** join/#asterisk GiantPickle (n=GiantPic@S01060016b600537f.gv.shawcable.net) |
| 16:56.40 | SuPrSluG | anyone using astlinux? |
| 16:57.29 | Lann | anyone else ever notice that they cannot afford another programmer's services if they ever did in fact need them, im like...20 hours of headache or pay someone...the headache is usually worth it |
| 16:57.48 | SuPrSluG | could i theoretically uncompress a kernel for use? |
| 16:58.11 | *** join/#asterisk PupenoR (n=pupeno@200.123.183.91) |
| 16:58.16 | mercestes | hal2385612: That should be true for any package. |
| 16:58.42 | CrashSys | Yum apparently = evul... like the 3rd or 4th guy i've seen that has problems cause of yum... |
| 16:58.50 | FuriousGeorge | i have a tdm400p with two fxo and 2 fxs. fxs had been used in the past. for some reason, everything appears to be working with the fxs, they show up with "zap show channels" and ztcfg, and when i call then the cli says they ring but in reality they dont |
| 16:59.02 | mercestes | CrashSys: Yea, it's pretty ebul. |
| 16:59.08 | mercestes | why I am !redhat. |
| 16:59.12 | mercestes | well, on eof the reasons |
| 16:59.18 | CrashSys | Corydon: So you suggest to modularize the sound-side of things and see if that eases my kernel panics? |
| 16:59.24 | FuriousGeorge | i made sure the molex was plugged in... cant figure out why this is going on |
| 17:00.17 | Jason99 | has anyone tested asterisk with 400+ peers? |
| 17:00.33 | *** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) |
| 17:00.33 | mercestes | FuriousGeorge: here's a dumb question. If you call it and it says "ringing" and you try to answer.....does it connect? |
| 17:00.59 | mercestes | Jason99: yes. why? |
| 17:01.27 | mercestes | well, I wouldn't say "tested." More, "dumped all our eggs in one giant basket and centered our company model around the basket." |
| 17:02.40 | Jason99 | I have 480 peers with qualify turned on all... everytime I reload i get 30-40 peers go UNREACHABLE.. and 10 seconds later come back as REACHABLE.. but this is causing problems because during that time those peers cannot receive calls |
| 17:03.18 | Jason99 | what are your thoughts? |
| 17:04.03 | mercestes | realtime? |
| 17:04.23 | Jason99 | what do you mean? |
| 17:04.31 | *** join/#asterisk J4k3 (n=jsuter@12.45.185.225) |
| 17:04.41 | Jason99 | I am realoding real time.. if I add/remove a peer, I need to reload |
| 17:04.44 | mercestes | Is it realtime asterisk? |
| 17:05.00 | Jason99 | hmm.. i dont know what that is |
| 17:05.14 | mercestes | are your peers defined in sip.conf or in a database? |
| 17:05.22 | Jason99 | sip.conf |
| 17:05.42 | mercestes | Then it's not RTA. These 30-40 peers, are they all in the same location(s) or are they randomly interspersed? |
| 17:06.05 | Jason99 | seems to be random peer lost each reload |
| 17:06.14 | mercestes | nice.. |
| 17:06.40 | Jason99 | from what I can see is that Asterisk isnt processing the responses quickly enough and it keeps retransmitting |
| 17:07.06 | mercestes | OK, this * box with 400 peers. Are the phones and * on internal or external IPs? Nat or not natted? |
| 17:07.44 | Jason99 | the * box is on the outside protected by a firewall, some phones are behind NAT and some arent.. depends on the location |
| 17:08.01 | Jason99 | It's a Dell 2950 |
| 17:09.19 | CrashSys | a 400-peer sip.conf |
| 17:09.28 | CrashSys | that's impressive... |
| 17:09.56 | Jason99 | ya, 8872 lines :P |
| 17:09.56 | CrashSys | I hope they were all the same phone |
| 17:10.06 | Jason99 | no all difefrent.. why? |
| 17:10.18 | CrashSys | one template for all phones |
| 17:10.30 | mercestes | Jason99: It could be any number of things then. On a reload it could be your router freaking out. |
| 17:10.48 | *** join/#asterisk joe_k (i=joe@adsl-65-67-81-98.dsl.ltrkar.swbell.net) |
| 17:10.57 | penguinFunk | how can i get hold of that addmailbox binary mentioned here in some docs ? |
| 17:11.10 | penguinFunk | it didnt come with the source code i downloaded from the asterisk site |
| 17:11.13 | mercestes | All the qualify does is check th ephone at it's last known address and if it sees it it's happy and if it does not then it starts pinging it. What yoru doing is missing the first ping and then hitting the second one. |
| 17:11.14 | Jason99 | mercestes: I see.. let me do more tests |
| 17:11.57 | *** join/#asterisk J4k3^ (n=jsuter@12.45.185.225) |
| 17:11.58 | mercestes | I'm betting on a reload that * does a broadcast to all phones and then all phones respond at once. Imagine what that looks like on 400 peers. |
| 17:12.35 | CrashSys | Prolly droppin' packets for either security or overload reasons |
| 17:12.39 | Jason99 | mercestes: that's exactly whats happening.. it sends a packet to all phones and expects a reply on all within 1 second otherwise it retransmits up to 5 times |
| 17:12.49 | mercestes | Right. |
| 17:13.01 | mercestes | We never did reloads in the middle of the day. We did sip prunes. |
| 17:13.08 | mercestes | but we also had RTA so we could prune an individual peer. |
| 17:13.15 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
| 17:13.19 | mercestes | I don't think sip.conf supports sip prune peer |
| 17:13.22 | mercestes | maybe it does. |
| 17:13.31 | Jason99 | all I need to do is load the new sip.conf.. is there another way? |
| 17:14.02 | mercestes | not that I know of. * scans the database on every new peer request so RTA might be a viable "fix." |
| 17:14.23 | mercestes | I just would not suggest a sip realtime reload ever during business hours. |
| 17:14.33 | mercestes | because peers won't seed until they A: Register, B: ring, C: make a call. |
| 17:14.50 | mercestes | so it won't be 10 seconds any more, it will be anywhere from instant to an hour. |
| 17:14.50 | Bobthehunter | anyone form sipfroundy here ? |
| 17:15.02 | Jason99 | ok |
| 17:15.05 | Jason99 | Thanks mercestes |
| 17:15.09 | mercestes | NP. |
| 17:15.18 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 17:15.18 | mercestes | Could also try adding some redundancy with anothre router.... |
| 17:15.23 | Jason99 | mercestes: if I go realtime I'd be ok? |
| 17:15.24 | mercestes | and maybe lighten up your firewall a bit. |
| 17:15.28 | *** part/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
| 17:15.45 | mercestes | Jason99: It would be worth a test. It would put sip.conf into a database (much easier) and allow you to do prunes to individual peers to do "updates." |
| 17:16.08 | penguinFunk | how can i get hold of that addmailbox binary mentioned here in some docs ? it didnt come with the asterisk source code i downloaded, how can i compile it etc? |
| 17:16.20 | mercestes | so instead of dumping the entire file into memory and doin ga shoutout to 400 peers it would actively scan the entire table everytime it needed new information to authenticate a peer. |
| 17:16.22 | wunderkin | penguinFunk, not needed |
| 17:16.31 | mercestes | the "scanning" part would be done on the database end. |
| 17:16.38 | *** join/#asterisk Crescendo (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 17:16.53 | Jason99 | ah i see.. does it support all same features as sip.conf? |
| 17:16.54 | mercestes | just use a dedicated database server to keep your * box clear. |
| 17:16.58 | mercestes | Yea. |
| 17:17.14 | mercestes | in fact the table structure is pretty insane because it includes *every* option in sip.conf with default values. |
| 17:17.25 | Jason99 | I will look into that.. thanks again |
| 17:17.32 | mercestes | =/ I'm not sure what the compatibility is for 1.4. 1.2.? is still supported for RTA sip. |
| 17:18.01 | mercestes | you can also move VM.conf in RTA as well, and your CDRs. I would suggest against putting extensions.conf into RTA. |
| 17:18.09 | *** join/#asterisk nahirean (i=nahirean@unaffiliated/nahirean) |
| 17:18.22 | *** join/#asterisk RoyK (n=roy@ti211310a080-9075.bb.online.no) |
| 17:18.37 | mercestes | and. NP.. :) Good luck. |
| 17:19.17 | CrashSys | welp... I guess i'll try removing the offending brance of code from my kernel, recompiling, and seeing if it still bombs... |
| 17:19.19 | *** join/#asterisk flashnet (i=flashnet@17.15.185.213.dk-hvi.res.sta.perspektivbredband.net) |
| 17:19.22 | *** join/#asterisk amdtech (i=amdtech@nat/digium/x-4ea69a4d21045676) |
| 17:20.33 | CrashSys | mercestes: You familiar with polycom's? |
| 17:20.55 | *** join/#asterisk a1fa (n=a1fa@unaffiliated/a1fa) |
| 17:21.05 | a1fa | does anybody know a cheap gsm to landline gateway? |
| 17:21.07 | penguinFunk | wunderkin: howcome ? |
| 17:21.13 | a1fa | that either has its own controler or can use an old nokia? |
| 17:21.18 | CrashSys | And if so, are you aware of how I can disable the "You have missed Xx Calls" by default on the PCom's? |
| 17:21.21 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 17:21.27 | penguinFunk | can i just add the appropriate entries to sip.conf,voicemail.conf and extensions.conf and it will work wunderkin? |
| 17:21.34 | wunderkin | yes |
| 17:21.37 | CrashSys | I'm tired of hearing "The phone system cant be working right, it says i've missed 34 calls! I've never gotten 34 calls!" |
| 17:21.38 | penguinFunk | ok thanks |
| 17:21.42 | mercestes | CrashSys: Yea a little. |
| 17:22.26 | mercestes | CrashSys: Yea...what they don't realize is that "missed calls" is *every* attempt at contact at that polycom that didn't result in them pikcing up their phone. |
| 17:22.37 | CrashSys | Plus evidently 1 line crackles now that never ever crackled before... but the other 3 work fine... (god I hate verizon) |
| 17:22.56 | CrashSys | mercestes: Yes, I know... and I explain such... but then they go "Well, I didn't get any voicemail!" |
| 17:23.24 | CrashSys | LCD Displays sucks :( |
| 17:23.31 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 17:23.43 | wunderkin | no, dumbass people suck |
| 17:23.44 | a1fa | such as Dock-N-Talk, but cheaper? |
| 17:23.55 | CrashSys | I cant fix a dumb-ass tho |
| 17:24.03 | CrashSys | I can possibly turn the message off on the phone tho |
| 17:24.46 | mercestes | You can modify it with serverMissedCalls |
| 17:24.51 | *** join/#asterisk duel (n=spam@mail.asaiatm.com) |
| 17:25.04 | mercestes | I think 1 will make it more accurate. |
| 17:26.31 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 17:26.42 | mercestes | yea, that's what I'm readin gnow. |
| 17:26.47 | mercestes | nothign on disabling. |
| 17:27.07 | CrashSys | well, even if it just doesn't display be default |
| 17:27.19 | CrashSys | it can still keep track, I just dont want it to show unless they go lookin' for it |
| 17:28.02 | a1fa | does anybody know of a good gsm to landline bridge? |
| 17:28.25 | *** join/#asterisk Qwell[] (i=qwell@pdpc/sponsor/digium/Qwell) |
| 17:28.25 | *** mode/#asterisk [+o Qwell[]] by ChanServ |
| 17:31.59 | *** join/#asterisk kannan (n=kannan@58.68.25.67) |
| 17:33.03 | putzz | can someone recommend a A-Z termination provider with DID (payasyougo), im going to kill sellvoip dudes took my money and they domain expires on the 14th of feb |
| 17:33.07 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 17:33.07 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 17:33.24 | a1fa | putzz : lol |
| 17:33.33 | a1fa | did they give you a service? |
| 17:34.13 | putzz | no |
| 17:34.26 | putzz | I knew they were hacks but u know they were cheap and all |
| 17:34.36 | b11d | haha thats what you get |
| 17:34.40 | putzz | so they give me my money or service or I put a reverse on the credit card |
| 17:35.02 | putzz | I did a search on the owner I got alot of his private info so its up to him how he wants to deal with it ;-) |
| 17:35.27 | b11d | yeah I bet |
| 17:35.40 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 17:35.44 | CrashSys | Hmmmm |
| 17:35.45 | CrashSys | sip.cfg |
| 17:35.59 | CrashSys | feature.calllist-missed |
| 17:36.07 | putzz | anyone recommend anything good for me? |
| 17:36.35 | b11d | yeah, food and water.. |
| 17:37.00 | a1fa | putzz : you better get your money back first |
| 17:37.10 | Bobthehunter | ok so the OSP example on digium talks about ospauth in dialplan and its not in the module lol |
| 17:37.54 | putzz | a1fa: im not worried like I said I got alot of info if I have to go get it I will. |
| 17:38.04 | putzz | but my credit card company will refund me anyways |
| 17:39.40 | b11d | I hope your lesson was well learned. |
| 17:40.42 | Bobthehunter | putzz voicemeup.com |
| 17:41.09 | putzz | Im not that worried I can garantee u I will go knock at their house and have some fun with the idiots that want to have fun with my money |
| 17:41.30 | Bobthehunter | how much was it ? |
| 17:41.34 | Bobthehunter | reverse it with visa |
| 17:41.40 | putzz | im reversing it |
| 17:41.45 | putzz | but im still going to find them |
| 17:42.04 | CrashSys | microsoft let hotmail.com expire :) |
| 17:42.04 | Bobthehunter | gott one who owes me 45k |
| 17:42.06 | Bobthehunter | one 10k |
| 17:42.22 | Bobthehunter | and first data stole 450,000$ from my boston bank merchatn # |
| 17:42.46 | Bobthehunter | no explanation nothing.. lawyers letter never answered.. they are the corporate domination.. the banks |
| 17:42.58 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 17:43.05 | mercestes | CrashSys: Sounds promising |
| 17:43.07 | Bobthehunter | if you go in person they call cops and say you are there to istrub the peace.. |
| 17:43.12 | putzz | I got couple boxes at a colo in seattle that I need to get to do some maint. so I will find them |
| 17:43.19 | putzz | hotmail set to expire 2014 wanna wait |
| 17:43.20 | putzz | ? |
| 17:43.20 | putzz | lol |
| 17:43.31 | Bobthehunter | putzz waht you need.. 1 seatle did + term ? |
| 17:43.34 | Bobthehunter | wahts your target price |
| 17:43.37 | Bobthehunter | and volume |
| 17:44.19 | putzz | im making some decisions still |
| 17:45.07 | Lann | just laying out the content in html |
| 17:46.05 | Lann | im just noticing the pictures appear dark everywhere but at my house |
| 17:46.33 | *** join/#asterisk NDT (n=nunya@cpe-74-70-211-81.nycap.res.rr.com) |
| 17:46.43 | in-pt | how do i have to load cdr_addon_mysql.so with asterisk 1.4 ? |
| 17:46.55 | in-pt | anyone any idea plz ? |
| 17:46.57 | Bobthehunter | any reason wy ${SIPCHANINFO(peerip)} gets the INTENRAL ip ? |
| 17:47.58 | NDT | we have someone sending to us sip from a meta switch they say...keep getting WARNING[2762]: chan_sip.c:3546 process_sdp: Insufficient information for SDP (m = '', c = '') Anyone run into that before? |
| 17:48.10 | Bobthehunter | check your nat |
| 17:50.37 | *** join/#asterisk Araluccl0 (n=ciccio@adsl-ull-230-7.46-151.net24.it) |
| 17:51.58 | *** join/#asterisk tim0123 (n=cash247@adsl-75-39-213-70.dsl.rcsntx.sbcglobal.net) |
| 17:52.30 | tim0123 | Anyone know how to record call queues |
| 17:55.23 | NDT | yeah nat is fine...what I don't understand heh |
| 17:57.26 | tim0123 | Nobody know how to record inbound calls |
| 18:01.36 | sevard | you can record with Record() or Monitor() iirc |
| 18:02.02 | wunderkin | check queues.conf |
| 18:02.04 | sevard | monitor |
| 18:03.58 | *** join/#asterisk topping (n=topping@h-67-100-91-18.snfccasy.covad.net) |
| 18:04.10 | *** join/#asterisk Crescendo (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 18:05.47 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 18:06.26 | The_DoC^ | . |
| 18:06.39 | *** join/#asterisk mxtreme311 (n=mxtreme3@208.71.143.66) |
| 18:07.16 | mxtreme311 | can someone help me with a Zaptel cofig? |
| 18:07.16 | *** part/#asterisk tdi (n=tdi@reykin.pozman.pl) |
| 18:07.20 | mafkees | any info on ipv6 in iax ? |
| 18:08.08 | tim0123 | I know that are there any examples tho |
| 18:08.36 | mxtreme311 | I'm working on a setup that I've never done before and would like some help. |
| 18:09.51 | mxtreme311 | is anyone available for a little help? |
| 18:11.31 | shido6 | ? |
| 18:11.36 | shido6 | whats up? |
| 18:11.51 | mafkees | mxtreme311: what's the problem ? |
| 18:12.11 | *** part/#asterisk a1fa (n=a1fa@unaffiliated/a1fa) |
| 18:12.19 | mxtreme311 | I have a PRI and a regular outbound only T-1 I'm trying to configure on my 2 port TE205P, the regular T-1 has E&M Wink signalling, and the PRI has |
| 18:12.19 | mxtreme311 | zaptel.conf: |
| 18:12.19 | mxtreme311 | span=1,1,0,esf,b8zs |
| 18:12.19 | mxtreme311 | span=2,1,0,esf,b8zs |
| 18:12.19 | mxtreme311 | e&m=1-24 |
| 18:12.20 | mxtreme311 | bchan = 25-47 |
| 18:12.22 | mxtreme311 | dchan = 48 |
| 18:12.24 | mxtreme311 | loadzone=us |
| 18:12.26 | mxtreme311 | defaultzone=us |
| 18:12.28 | mxtreme311 | zapata.conf: |
| 18:12.30 | mxtreme311 | [channels] |
| 18:12.32 | mxtreme311 | group=1 |
| 18:12.34 | mxtreme311 | language=en |
| 18:12.36 | mxtreme311 | signalling=em_w |
| 18:12.38 | mxtreme311 | usecallerid=yes |
| 18:12.40 | mxtreme311 | callerid=asreceived |
| 18:12.42 | mxtreme311 | context=default |
| 18:12.44 | mxtreme311 | echocancel=64 |
| 18:12.46 | mxtreme311 | echocancelwhenbridged=yes |
| 18:12.48 | mxtreme311 | rxgain=1.0 |
| 18:12.50 | mxtreme311 | txgain=1.0 |
| 18:12.52 | mxtreme311 | channel => 1-24 |
| 18:12.54 | mxtreme311 | group=3 |
| 18:12.56 | mxtreme311 | language=en |
| 18:12.58 | mxtreme311 | switchtype=dms100 |
| 18:13.00 | mxtreme311 | signalling=pri_cpe |
| 18:13.02 | mxtreme311 | usecallerid=yes |
| 18:13.04 | mxtreme311 | callerid=asreceived |
| 18:13.06 | mxtreme311 | context=default |
| 18:13.08 | mxtreme311 | echocancel=64 |
| 18:13.10 | mxtreme311 | echocancelwhenbridged=yes |
| 18:13.12 | mxtreme311 | rxgain=1.0 |
| 18:13.14 | mxtreme311 | txgain=1.0 |
| 18:13.16 | mxtreme311 | channel => 25-47 |
| 18:13.18 | mxtreme311 | But my CLI is outputting this almost every few seconds: |
| 18:13.18 | sevard | dude |
| 18:13.20 | mxtreme311 | == pri_find_dchan: No D-channels available! Using Primary channel 48 as D-channel anyway! |
| 18:13.21 | sevard | pastebin.ca |
| 18:13.22 | mxtreme311 | == Primary D-Channel on span 2 up |
| 18:13.34 | mxtreme311 | sorry |
| 18:13.36 | sevard | ~pb |
| 18:13.46 | jbot | i heard pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 18:14.48 | mxtreme311 | I was trying to say, I have a E&M Wink T1 and a PRI and I'm trying to configure the two to both work on one TE205P card. the full problem is posted here: http://forums.digium.com/viewtopic.php?t=13373 |
| 18:16.05 | mxtreme311 | does anyone have any wisdom they would like to share on my little problem? |
| 18:16.27 | mafkees | not me, sorry |
| 18:17.39 | *** join/#asterisk UlbabraB (n=salama@host241-43-static.72-81-b.business.telecomitalia.it) |
| 18:20.17 | mxtreme311 | shido6, did you get that? can you please help? |
| 18:22.46 | mxtreme311 | is there someone on here that is good with configuring Zaptel cards? |
| 18:25.06 | CrashSys | dont you include the d-channel in the channel config? |
| 18:25.12 | CrashSys | like channels => 25-48 |
| 18:26.43 | CrashSys | hmmm, nope... |
| 18:26.57 | mxtreme311 | yeah, not that I am aware of |
| 18:27.02 | CrashSys | You sure you have the PRI Line plugged into the right port? |
| 18:27.23 | mxtreme311 | Yes. I've configured many T-1s and PRIs, but never on the same card. |
| 18:27.24 | BrokenNoze_ | anyone know a good place to look for asterisk related jobs/contracts? |
| 18:27.41 | CrashSys | it sounds like (from the error) that it's looking for the d-channel on channel 48, but that' it's not there... |
| 18:27.43 | mxtreme311 | That's my problem here... |
| 18:27.59 | mxtreme311 | so the config looks correct? |
| 18:28.12 | CrashSys | To my eyes, yes... |
| 18:28.27 | CrashSys | Can you call in on the CAS T1? |
| 18:28.43 | mxtreme311 | i don't have the inbound setup quite yet... |
| 18:29.15 | CrashSys | Just for fun, swap the PRI and CAS lines... see what happens... |
| 18:29.28 | *** join/#asterisk chainey (n=chainey@haw-207-182-233-32.vel.net) |
| 18:29.29 | CrashSys | or can you dial out the CAS T1? |
| 18:29.47 | mxtreme311 | yeah, but it goes up and down... |
| 18:30.09 | CrashSys | so you can connect and talk, but it hang-up? |
| 18:30.16 | mxtreme311 | yeah |
| 18:30.24 | CrashSys | hmmm |
| 18:31.10 | mxtreme311 | it's constantly outputing that there are no D-channels available! Using Primary channel 48, then it will say the Primary D-Channel on span 2 is up. |
| 18:31.29 | CrashSys | which one you using for timing source? |
| 18:31.34 | *** join/#asterisk RoyK (n=roy@ti211310a080-5551.bb.online.no) |
| 18:31.50 | mxtreme311 | where can I find that? |
| 18:32.50 | Lann | has anyone here ever set up dialplan based voicemail? |
| 18:33.07 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 18:33.07 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 18:33.10 | Lann | (looking for link to walkthrough) |
| 18:33.10 | *** join/#asterisk RoyK (n=roy@ti211310a080-5551.bb.online.no) |
| 18:33.54 | *** join/#asterisk J4k3 (n=jsuter@mhset-growth7.memorialhealth.org) |
| 18:34.54 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 18:35.47 | CrashSys | mx: try setting the E&M as the secondary timing source... maybe? span=1,2,0,blah blah |
| 18:36.14 | mxtreme311 | I'll try that, thanks. |
| 18:36.50 | mxtreme311 | would it matter that these two circuits are from different providers? |
| 18:39.47 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-221.ph.ph.cox.net) |
| 18:39.59 | mafkees | nope |
| 18:40.19 | mxtreme311 | makfees: is that answer to my question? |
| 18:41.10 | mafkees | yeah |
| 18:41.13 | mafkees | it does not matter |
| 18:41.22 | mxtreme311 | thanks. |
| 18:41.22 | Lann | hmm...what would someone charge for a custom voicemail system? |
| 18:41.29 | mafkees | brb, food |
| 18:44.49 | *** join/#asterisk J4k3 (n=jsuter@mhset-growth7.memorialhealth.org) |
| 18:46.44 | CrashSys | mx: Does ztcfg -vvv list the d-chan on it's output? |
| 18:47.24 | mxtreme311 | yes... |
| 18:47.30 | CrashSys | and channel 1-23 as "Clear Channel"? |
| 18:47.37 | mxtreme311 | yes... |
| 18:47.41 | CrashSys | Hmmm |
| 18:47.48 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 18:47.55 | CrashSys | err 25-47 as clear |
| 18:48.00 | CrashSys | and 48 a D-chan |
| 18:48.10 | *** join/#asterisk J4k3^ (i=jsuter@wls-a032.intrastar.net) |
| 18:48.19 | mxtreme311 | correct. |
| 18:48.25 | mxtreme311 | 1-24 show as E&M Wink |
| 18:48.30 | *** join/#asterisk duel (n=spam@mail.asaiatm.com) |
| 18:49.09 | CrashSys | Is this PRI active? |
| 18:49.19 | mxtreme311 | yes |
| 18:49.22 | CrashSys | Maybe they haven't turned it up on their end? |
| 18:49.53 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 18:51.51 | CrashSys | Does the PRI and E&M have inbound? |
| 18:52.03 | *** join/#asterisk J4k3^ (n=jsuter@mhset-growth7.memorialhealth.org) |
| 18:52.06 | mxtreme311 | it seems that changing the timing on the T-1 to secondary has corrected the problem |
| 18:52.21 | CrashSys | ok... |
| 18:52.29 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-154-54-119.red.bezeqint.net) |
| 18:52.43 | CrashSys | You can possibly run into timing problems between providers if one uses GPS and the other uses atomic timers... |
| 18:52.48 | CrashSys | super-rare... but it happens... |
| 18:53.27 | mxtreme311 | well thank you so much... I'll check with my providers and see what they are using. |
| 18:53.32 | mxtreme311 | again, thank you so much |
| 18:53.40 | CrashSys | yup |
| 18:53.55 | CrashSys | mx: It's only a problem if we have incredibly-bad solar flares :) |
| 18:54.02 | CrashSys | they happen like every 10-12 years or so |
| 18:54.08 | mxtreme311 | lol |
| 18:54.13 | CrashSys | last one was a couple years ago |
| 18:54.27 | mxtreme311 | well then it looks like I'm good to go... |
| 18:54.53 | CrashSys | Yup... i'd still try in/out bound calling, see if you can maintain a 5-minute call without dropping tho :) |
| 18:55.03 | mxtreme311 | will do thanks |
| 18:55.05 | CrashSys | later mang |
| 18:58.08 | *** join/#asterisk KuJaX (n=one@67.164.251.181) |
| 18:58.27 | lenne_dk | Can * play a soundfile to the transferrer at a succesful transfer? |
| 18:58.29 | *** join/#asterisk dasenjo (n=dasenjo@190.5.197.47) |
| 18:59.33 | dasenjo | Hello, I'm getting a warning saying channel.c:2049 ast_indicate: Unable to handle indication 3 for 'SIP/avatar_dasenjo-4a6d' |
| 19:00.03 | dasenjo | when this occurs the caller side does not have ring indication, ¿what can I do? |
| 19:04.10 | riddlebox | wohoo I have a meeting with two of my bosses to show them asterisk and how we can sell it |
| 19:05.32 | *** part/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 19:06.53 | Corydon-w | CrashSys: ping |
| 19:08.49 | CrashSys | cory: Hmm? |
| 19:09.08 | CrashSys | ohh, wait |
| 19:09.10 | CrashSys | corydon-w: pong |
| 19:09.25 | Corydon-w | CrashSys: I'll have another patch for you in a few minutes |
| 19:10.34 | CrashSys | cory: you find something in it? |
| 19:10.39 | CrashSys | or you just making one for chan_alsa? |
| 19:10.52 | Corydon-w | No and yes. |
| 19:10.58 | CrashSys | ahhhh |
| 19:11.03 | Corydon-w | http://asterisk.drunkcoder.com/patches/20070205__alsa_flags.diff.txt |
| 19:11.39 | Corydon-w | This one is untested, because I don't have the facilities here to test it |
| 19:11.45 | CrashSys | I guess for posterity's sake I should try running the stock chan_oss and see if that throws up on me... |
| 19:11.47 | Corydon-w | but it should work the same |
| 19:11.49 | CrashSys | i'll test it for ya... |
| 19:11.56 | CrashSys | later tonight that is... |
| 19:12.08 | *** join/#asterisk Op3r (n=op3r@125.212.124.138) |
| 19:12.13 | CrashSys | By test I mean patch the file and see if it works |
| 19:12.29 | Corydon-w | If it works, tell me, and I'll get it into trunk |
| 19:12.34 | *** join/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 19:12.35 | fall0ut | question, if you see invite through one proxy that has a record-route, and you then receive a REFER that bypasses the last proxy, that is considered out-of-dialog, correct? |
| 19:12.55 | fall0ut | since the record-route is not there through the last proxy you saw the call from |
| 19:13.05 | *** join/#asterisk bkw_ (n=brian@adsl-70-142-59-250.dsl.tul2ok.sbcglobal.net) |
| 19:13.56 | *** join/#asterisk DaveCanoe (n=Dave@dhcp3022.nanog.merit.net) |
| 19:14.22 | CrashSys | Cory: I was also getting like 160% CPU usage with the chan_oss patched... as reported by top... must be something locally to me... |
| 19:14.50 | Corydon-w | Must be |
| 19:17.39 | *** join/#asterisk Pretor1ab (n=root@82-34-254-218.cable.ubr07.enfi.blueyonder.co.uk) |
| 19:17.43 | Pretor1ab | hello all |
| 19:18.47 | Pretor1ab | i am having trouble setting up zaptel |
| 19:19.24 | Pretor1ab | i seem to have compiled it an installed it okay but the service does not startup |
| 19:20.05 | Pretor1ab | --> service zaptel start == line 0: Unable to open master device '/dev/zap/ctl' |
| 19:20.21 | Op3r | ztcfg -vvvv |
| 19:20.45 | Pretor1ab | line 0: Unable to open master device '/dev/zap/ctl' |
| 19:21.36 | mercestes | does dev/zap/ctl exist? |
| 19:22.10 | Pretor1ab | no appears not |
| 19:22.30 | *** mode/#asterisk [+o mitcheloc] by ChanServ |
| 19:22.57 | mercestes | did you modprobe zaptel? |
| 19:23.22 | Pretor1ab | modprobe zaptel --> FATAL: Error inserting zaptel (/lib/modules/2.6.19-1.2895.fc6/misc/zaptel.ko): Invalid module format |
| 19:23.31 | mercestes | nice. |
| 19:23.34 | mercestes | recompile zaptel then. |
| 19:23.44 | Pretor1ab | ok |
| 19:23.58 | Pretor1ab | so... |
| 19:24.04 | Pretor1ab | cd zaptel dir |
| 19:24.08 | Pretor1ab | ./configure |
| 19:24.09 | Pretor1ab | make |
| 19:24.12 | Pretor1ab | make install |
| 19:24.18 | Pretor1ab | is that okay? |
| 19:24.52 | mercestes | *might* want to throw a make clean in there but otherwise, yea |
| 19:25.01 | Pretor1ab | i was thinking that too |
| 19:25.27 | Pretor1ab | brb |
| 19:25.59 | Pretor1ab | ./configure --> configure: *** Zaptel build successfully configured *** |
| 19:26.00 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 19:26.29 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 19:27.05 | Pretor1ab | make clean --> okay |
| 19:27.40 | Pretor1ab | *** WARNING: |
| 19:27.40 | Pretor1ab | *** If you had custom settings in /etc/modprobe.d/zaptel, |
| 19:28.02 | Pretor1ab | modprobe zaptel --> FATAL: Error inserting zaptel (/lib/modules/2.6.19-1.2895.fc6/misc/zaptel.ko): Invalid module format |
| 19:28.15 | mercestes | what distro? |
| 19:28.19 | Pretor1ab | fedora core 6 |
| 19:28.22 | mercestes | ew |
| 19:28.28 | mercestes | fresh install>? |
| 19:28.46 | Pretor1ab | nope long standing |
| 19:28.54 | mercestes | thought so. |
| 19:29.01 | Op3r | update ur kernel |
| 19:29.02 | mercestes | ok, drastic time. make clean && make distclean |
| 19:29.07 | mercestes | (backup /etc first) |
| 19:29.24 | Pretor1ab | ok done that |
| 19:29.33 | mercestes | ./configure |
| 19:29.37 | mercestes | then make && make install |
| 19:29.49 | Pretor1ab | uname -r --> 2.6.19-1.2895.fc6 |
| 19:30.13 | mercestes | Yea, it's fc6, I don't need to know what version of broken yoru running..;) |
| 19:30.23 | CrashSys | LOL |
| 19:30.27 | mafkees | gheh |
| 19:30.41 | Pretor1ab | whos he talking to me ?? |
| 19:30.57 | mercestes | lol. Yea, it's a joke. |
| 19:31.05 | mercestes | I'm a !redhat person. |
| 19:31.13 | Pretor1ab | which broken yoru you running? |
| 19:31.15 | mercestes | mostly because redhat has been mean to me. |
| 19:31.16 | CrashSys | mercestes attempted to make a funny |
| 19:31.23 | mercestes | I'm running a broken gentoo system. |
| 19:31.27 | mafkees | apt-get install debian |
| 19:31.37 | mercestes | hehe...Distrowars! |
| 19:31.38 | Pretor1ab | i have suse debian fedora and centos running here |
| 19:31.40 | Corydon-w | like there's any other kind of Gentoo system? |
| 19:31.45 | CrashSys | I use slack... it doesn't have enough things in it to be broken |
| 19:31.51 | CrashSys | I prefer to think of it as "Limited" |
| 19:31.53 | *** join/#asterisk duel (i=duel@mail.asaiatm.com) |
| 19:31.54 | Qwell[] | Corydon-w: a properly admined one :P |
| 19:31.54 | Pretor1ab | if you cant beat - install them all |
| 19:32.09 | Corydon-w | Qwell[]: see "install debian" |
| 19:32.15 | mercestes | Corydon-w: Um... emerge broken-sources emerge riced-sources emerge fomg-optomized-sources yea, pretty much it. |
| 19:32.32 | Pretor1ab | make clean; make distclean; make; make install; modprobe zaptel --> FATAL: Error inserting zaptel (/lib/modules/2.6.19-1.2895.fc6/misc/zaptel.ko): Invalid module format |
| 19:32.59 | mercestes | supersweet! |
| 19:33.00 | Corydon-w | Pretor1ab: sounds like your kernel source version doesn't match your kernel version |
| 19:33.07 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 19:33.07 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 19:33.14 | Pretor1ab | how do i know that ? |
| 19:33.19 | CrashSys | The principal behind Gentoo seems good... not sure i'd like it for production... i'm a fan of packages that (atleast, in theory should) have patches applied for said distribution to eliminate bugs... |
| 19:33.26 | Corydon-w | Pretor1ab: uname -a |
| 19:33.44 | duel | what is roots passwd for vmware image? i have admin pass, but admin doesnt seem to be the same as root. no info on root password ?? |
| 19:33.50 | mercestes | CrashSys: I use gentoo for production...but mainly because I can install *NOTHING* and get a clean system fairly easily. |
| 19:34.06 | Pretor1ab | uname -a --> Linux home.pc 2.6.19-1.2895.fc6 #1 SMP Wed Jan 10 18:32:37 EST 2007 i686 i686 i386 GNU/Linux |
| 19:34.10 | CrashSys | Hmmm... that intrigues me too :) |
| 19:34.15 | mercestes | it just seemed to be in harmony with my brain. Only reason I like it. |
| 19:34.16 | CrashSys | the spartan approach |
| 19:34.17 | mafkees | OpenBSD griever.vanbaak.info 4.0 GENERIC.RAID#0 i386 Intel Pentium II ("GenuineIntel" 686-class, 512KB L2 cache) |
| 19:34.36 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 19:34.53 | mercestes | See, OpenBSD is an awesome concept....and I would use it...if it's creator and entire fanbase wasn't one...big...pulsating....ass. |
| 19:35.06 | CrashSys | :o |
| 19:35.10 | CrashSys | I like my OBSD |
| 19:35.11 | mercestes | zOMG woe to you if you ask even a slightly uninformed question in #openbsd |
| 19:35.26 | bkw_ | Darwin taz.local 8.8.1 Darwin Kernel Version 8.8.1: Mon Sep 25 19:42:00 PDT 2006; root:xnu-792.13.8.obj~1/RELEASE_I386 i386 i386 |
| 19:35.29 | mercestes | I like my oBSD too, but it's not linux, and your not allowed to ask questions about it. |
| 19:35.48 | mafkees | almost no need to ask questions |
| 19:35.53 | mafkees | the docs are beautiful |
| 19:35.55 | mercestes | the MAN pages and the online manual and the books you can by should (do) suffice for everything you need to know. |
| 19:36.04 | mercestes | heh. |
| 19:36.12 | CrashSys | There's not much in OpenBSD to ask questions about... |
| 19:36.12 | Pretor1ab | well ???? |
| 19:36.13 | mercestes | The docs are beautiful if you speak that language. |
| 19:36.24 | mafkees | I'm always frustrated when going back to linux |
| 19:36.27 | mercestes | Pretor1ab: have you considered/planned on updating yoru kernel? |
| 19:36.27 | CrashSys | It barely comes with enough utils to make it a working OS |
| 19:36.30 | bkw_ | CrashSys, yes their is.. why does it suck at anything but routing and filtering? |
| 19:36.50 | bkw_ | OBSD is only good for a router or firewall.. thats about it. |
| 19:36.52 | Pretor1ab | no i have not considered planning updating my kernel - i considered planning to install asterisk |
| 19:36.54 | mercestes | bkw_: ROFLMAO |
| 19:37.03 | mafkees | bkw_: it works great as desktop too |
| 19:37.06 | bkw_ | bullshit |
| 19:37.09 | mafkees | nope |
| 19:37.12 | bkw_ | its bullshit |
| 19:37.14 | mercestes | bkw_: google openbsd sucks It's not even good for a router/firewall. |
| 19:37.14 | bkw_ | I call bullshit |
| 19:37.18 | CrashSys | OBSD as a desktop? |
| 19:37.23 | mafkees | yeah |
| 19:37.32 | Corydon-w | Okay, we're getting a bit off topic, folks |
| 19:37.35 | mafkees | I'm running it as my main desktop since 3.2 |
| 19:37.44 | Pretor1ab | yeah I have an install problem i need help with please |
| 19:37.48 | mafkees | asterisk runs great on it |
| 19:37.59 | CrashSys | maf: No zap? |
| 19:38.06 | mercestes | ....dude, you don't even RUN OpenBSD..LOL |
| 19:38.07 | mafkees | no zap, but who needs that :) |
| 19:38.12 | mafkees | it's VOIP |
| 19:38.15 | mafkees | ;) |
| 19:38.20 | CrashSys | Umm, I need that... |
| 19:38.29 | *** join/#asterisk hi365_m (n=hi365_m@ti100710a081-1255.bb.online.no) |
| 19:38.32 | Pretor1ab | can anyone help please ??? |
| 19:38.43 | mercestes | Pretor1ab: Ok..so no updates to kernel. Well, yo ucould recompile your old kernel sources..:D or...download new kernel sources and compile those. |
| 19:38.54 | mafkees | Pretor1ab: do you have the kernel sources for your running kernel installed ? |
| 19:39.04 | Pretor1ab | yes they were needed to make zaptel |
| 19:39.10 | mercestes | Pretor1ab: Normally I would just emerge kernel-headers but....meh, that's gentoo |
| 19:39.22 | Pretor1ab | i dont use Gentoo |
| 19:39.37 | Pretor1ab | i am using fedora core 6 |
| 19:39.40 | mercestes | Pretor1ab: Then you need to compile the sources you have in order for zaptel to run. DISCLAIMER to follow. |
| 19:39.56 | mercestes | Pretor1ab: If Zaptel isn't compiling against your sources tho...your sources do *NOT* equal what you are running now |
| 19:39.57 | duel | what is roots passwd for vmware image? i have admin pass, but admin doesnt seem to be the same as root. no info on root password ?? |
| 19:39.58 | Pretor1ab | they are in the repository |
| 19:40.13 | mercestes | duel: try sudo su - |
| 19:40.15 | *** part/#asterisk hi365_m (n=hi365_m@ti100710a081-1255.bb.online.no) |
| 19:40.31 | Pretor1ab | mercestes: me ? |
| 19:40.40 | Pretor1ab | oh sorry |
| 19:40.45 | duel | mercestes: it asks for password though |
| 19:40.46 | mercestes | Pretor1ab: So you may be breaking something by compiling the new kernel sources or going either backwards or forwards in your revision. They dont' appear to be matching |
| 19:40.54 | mercestes | duel: Enter the admin password |
| 19:41.16 | Pretor1ab | right so i need to install a new kernel with sources? |
| 19:41.22 | duel | i did, wrong pass |
| 19:41.43 | mercestes | Pretor1ab: Or compile the ones you have...but...again, they aren't equal to what you are running obviously |
| 19:41.56 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
| 19:41.58 | Pretor1ab | how would they not be - i only used yum |
| 19:42.06 | Op3r | Pretor1ab: uname -r |
| 19:42.07 | Pretor1ab | i did not compile anything myself |
| 19:42.18 | mercestes | yea, I'm not a fan of yum either. |
| 19:42.19 | CrashSys | d00d... it's like yum day on here... |
| 19:42.21 | Op3r | then install kernel sources that correspond to that |
| 19:42.25 | mercestes | yea. |
| 19:42.26 | duel | mercestes: i did, wrong pass |
| 19:42.30 | Pretor1ab | uname -r --> 2.6.19-1.2895.fc6 |
| 19:42.36 | *** join/#asterisk a1fa (n=a1fa@unaffiliated/a1fa) |
| 19:42.47 | mercestes | then download and install 2.6.19-1.2895.fc6 |
| 19:42.54 | CrashSys | Yum must operate like Windows Update... they release updates that break other things... until they release more updates for the updates... which still break other things... |
| 19:42.55 | Pretor1ab | i have the kernel sources for 2.6.19-1.2895.fc6 |
| 19:42.55 | Pretor1ab | <PROTECTED> |
| 19:43.08 | mafkees | Pretor1ab: get kernel-headers-2.6.19-1.2895.fc6 |
| 19:43.15 | mercestes | duel: Dunno then...you miskeyed or somethign has gone terribly wrong. You can try google default root password <whatever it is yoru playign with> |
| 19:43.17 | Pretor1ab | i have kernel headers |
| 19:43.25 | mercestes | duel: sudo su - should work tho with existing or no passwords |
| 19:43.32 | Lann | has anyone here ever set up custom voicemail (custom menus and prompts)? Where can i find documentation or examples? |
| 19:43.47 | mercestes | pretor1ab: And as we stated previously, they are wrong and do not match your running kernel |
| 19:43.52 | mafkees | Lann: all you need is the vm-* soundfiles |
| 19:44.04 | mercestes | Pretor1ab: One or the other must change.... |
| 19:44.08 | Pretor1ab | can you explain how you know this ? |
| 19:44.15 | mafkees | Lann: but the menu options are hardcoded, so if you want to change tthem you have to hack app_voicemail.c |
| 19:44.27 | Pretor1ab | they have the same revisions |
| 19:44.41 | CrashSys | Lann: You could write one in AGI... if you wanted... |
| 19:44.43 | mafkees | Pretor1ab: you have a link /usr/src/linux ? |
| 19:44.48 | Lann | agi would be nice |
| 19:44.50 | *** join/#asterisk gerhard7 (n=gerhard@82-171-117-191.dsl.ip.tiscali.nl) |
| 19:45.00 | Lann | which would be easier, i want certain menu options in voicemail to invoke scripts |
| 19:45.09 | Lann | and for each voicemail to have some associated data |
| 19:45.20 | CrashSys | ohh |
| 19:45.20 | mafkees | sounds like agi indeed |
| 19:45.23 | Pretor1ab | nope |
| 19:45.27 | *** join/#asterisk rrocha (n=ruy@116-144-142-200.mcmtelecom.com.br) |
| 19:45.29 | CrashSys | well you need to write the whole thing then... |
| 19:45.35 | Lann | each message is an appointment that you can accept or decline |
| 19:45.39 | CrashSys | if you want extra data the built-in voicemail doesn't capture |
| 19:45.46 | mercestes | Pretor1ab: .... oh, USE="symlink" emerge gentoo-sources |
| 19:45.49 | mercestes | Oh yea! your not on gentoo |
| 19:45.59 | mafkees | lol |
| 19:46.00 | mercestes | Pretor1ab: do your ln -s then and make a symlink.. :) |
| 19:46.05 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 19:46.08 | Pretor1ab | to what ? |
| 19:46.14 | mercestes | your kernel sources |
| 19:46.17 | mercestes | that should be there. |
| 19:46.20 | Pretor1ab | which are where? |
| 19:46.20 | Lann | where can i find an AGI based voicemail example? |
| 19:46.23 | mercestes | because yum would never fail. |
| 19:46.26 | *** join/#asterisk Zaw (i=zaw@unaffiliated/zaw) |
| 19:46.29 | mercestes | linux -> kernel sources |
| 19:46.30 | mafkees | cd /usr/src && ln -s kernel-headers-blablablabla linux |
| 19:46.36 | CrashSys | Lann: No clue... I tell my clients to deal with comedian |
| 19:47.06 | Lann | custom code people? |
| 19:47.06 | Lann | never heard of comedian |
| 19:47.06 | mercestes | Lann: Ok ok ok, We don't normally tell n00bs this, but there is a secret way to hack apart voicemail withou thaving to do any coding. |
| 19:47.10 | CrashSys | Comedian Voicemail = Asterisk's voicemail |
| 19:47.15 | Lann | heh |
| 19:47.20 | J4k3^ | what odd is |
| 19:47.34 | J4k3^ | depending on what codec I call with |
| 19:47.35 | CrashSys | Like when you dial into voicemail it says "Comedian Mail" |
| 19:47.41 | J4k3^ | I get "asterisk mail" or "comedian mail" |
| 19:47.45 | J4k3^ | I *swear* it sounds different |
| 19:47.46 | Lann | hmm |
| 19:47.47 | *** join/#asterisk ta^3 (n=tacvbo@189.136.33.8) |
| 19:47.48 | Lann | i never knew that |
| 19:47.56 | CrashSys | which for the longest time I had clients thinking it said "Idiot Mail" |
| 19:47.57 | CrashSys | lol |
| 19:47.59 | Lann | cant you implement it with AGI? |
| 19:47.59 | Pretor1ab | might come back later - this is not making sense |
| 19:48.05 | mercestes | it involves you catting /dev/urandom directly onto your harddrive. |
| 19:48.08 | J4k3^ | well, g729 tends to connect about halfway through the first word ;) |
| 19:48.31 | mafkees | can be solved with an answer() and a wait() |
| 19:48.40 | mercestes | J4k3^: Maybe it's cutting off the "co" and your getting "median mail" It does that on some codecs |
| 19:48.47 | Lann | well honestly...if i think about it in AGI terms, i really just need a script to be able to pipe out voicemails generated by asterisks normal voicemail system to the caller |
| 19:48.47 | CrashSys | I think I had everyone in the company pulling me aside and going "I dont wanna tell you how to do your business, but I think the voicemail says Idiot Mail!" |
| 19:49.02 | shido6 | sometimes u get ....... mmmmmmmMeeeedian MAil..... mailbox? |
| 19:49.21 | J4k3^ | mine will just go "beep" eventually... if you don't know to put a password in there, you don't deserve voicemail ;) |
| 19:49.23 | mafkees | lol CrashSys |
| 19:49.31 | Lann | can an AGI script read a voicemail to a user/ |
| 19:49.41 | mafkees | sure |
| 19:49.50 | mercestes | Lann: You could do it with some severely complex records and backgrounds if you *really* want to get into that. |
| 19:49.50 | Lann | that would do what i need |
| 19:49.55 | mafkees | it's stored in the format you specify in voicemail.conf |
| 19:50.03 | CrashSys | Lann: Could even have the AGI script read the TXT file associated with the voicemail... |
| 19:50.12 | Lann | they have txt files? |
| 19:50.15 | mafkees | yup |
| 19:50.18 | CrashSys | Believe so |
| 19:50.22 | mercestes | how do ppl get "idiot" out of "Comedian"? |
| 19:50.24 | Lann | what is that some sort of voice synthesis? |
| 19:50.25 | CrashSys | records the voicemail info, like caller, time, etc... |
| 19:50.29 | CrashSys | mercestes: They're end users |
| 19:50.30 | *** join/#asterisk anthonyl (n=anthonyl@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 19:50.31 | Lann | oh ok |
| 19:50.35 | mafkees | leave a voicemail and look in /var/spool/asterisk/voicemail/context/mailbox/ |
| 19:50.47 | Lann | yeah if i can cat out the audio to the caller via agi, i could easily code my entire voicemail app |
| 19:50.51 | mercestes | Lann: Festival, cepstral. |
| 19:51.00 | Lann | i use Flite |
| 19:51.05 | mercestes | .... |
| 19:51.09 | mercestes | festival, cepstral. |
| 19:51.12 | mafkees | yeah |
| 19:51.13 | mafkees | festival |
| 19:51.17 | mafkees | sounds GREAT ! |
| 19:51.18 | mafkees | ;) |
| 19:51.26 | mercestes | esp on Obsd |
| 19:51.30 | Lann | heh |
| 19:51.35 | Lann | flite sounds pretty bad |
| 19:51.48 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 19:51.50 | mafkees | mercestes >) |
| 19:52.02 | Lann | but my system is just gonna say "You have 3 appointments" ...you can read them and accept or deny, upon accept it sends receipt to the person who made the appointment |
| 19:52.05 | Lann | thats what im trying to do |
| 19:52.12 | syzygyBSD | how can I make a zap card recover from a red alarm, or will it do it automatically when my provider stops sucking? |
| 19:52.32 | J4k3^ | red alarms are usually facilities based |
| 19:52.33 | CrashSys | Lann: just record the words... same way asterisk prompts work |
| 19:52.46 | J4k3^ | yellow alarms sometimes require user invention, but most telco stuff is designed to come back up ASAP |
| 19:52.47 | CrashSys | faster then a speech engine... |
| 19:52.48 | mercestes | you can use my voice. |
| 19:53.00 | syzygyBSD | J4k3^: I know what it is, and I know the cause, I just need to know if it will come back automatically or If I can go to lunch |
| 19:53.06 | mercestes | I sound like Arnold Schwartzeneggar on ether. |
| 19:53.29 | CrashSys | I get drunk when I record prompts... cause then I dont stutter :) |
| 19:53.35 | mercestes | syzygyBSD: If your telco is sucking and they cease to suxxorz and you don't change anything then the redness should subside by itself. |
| 19:53.47 | *** join/#asterisk saftsack (n=oliver@p54A7DA96.dip.t-dialin.net) |
| 19:53.59 | saftsack | hi |
| 19:53.59 | mercestes | I sound like a well articulated minotaur. |
| 19:54.16 | mercestes | ....or James Earl Jones on helium. |
| 19:55.15 | CrashSys | the redness should subside? |
| 19:55.15 | CrashSys | Sounds like you need penicillin |
| 19:55.20 | mercestes | or a new telco. or both. |
| 19:55.50 | mafkees | or a fireaxe |
| 19:55.57 | mafkees | always solves my problems |
| 19:56.21 | CrashSys | I like where Mafkees is coming from... |
| 19:56.34 | mafkees | gheh |
| 19:56.35 | saftsack | about what are you talking atm? |
| 19:57.04 | CrashSys | saft: Tecnical support... (damn I cant spell) |
| 19:57.05 | Lann | crashsys: so you are saying to use asterisk prompts instead of voicemail type stuff? |
| 19:57.07 | CrashSys | technical |
| 19:57.17 | J4k3^ | everything works better after being molested with a fireaxe |
| 19:57.17 | CrashSys | Lann: Instead of a speech engine |
| 19:57.37 | Lann | oh... |
| 19:57.41 | CrashSys | I would rather use recorded prompts then a speech engine |
| 19:57.52 | Lann | well if i feel like recording 1-31 and every month |
| 19:57.55 | Lann | and then times |
| 19:58.02 | CrashSys | Asterisk already has those... |
| 19:58.06 | CrashSys | I think |
| 19:58.09 | mercestes | it does |
| 19:58.14 | mercestes | under ./digits |
| 19:58.15 | CrashSys | dunno about the months |
| 19:58.27 | Corydon-w | It has those, too |
| 19:58.33 | CrashSys | There ya go |
| 19:58.51 | Corydon-w | and formulations for every number up to 999,999,999 |
| 19:58.54 | CrashSys | So all you need to record is "appointments" |
| 19:59.07 | Lann | mmm...sushi |
| 19:59.24 | CrashSys | hand-roll or makimono? |
| 19:59.37 | Corydon-w | Maki or Nigiri? |
| 19:59.42 | Lann | the inside-out one |
| 19:59.43 | CrashSys | that too |
| 19:59.47 | CrashSys | ... |
| 20:00.02 | Lann | wasurimashita |
| 20:00.03 | mercestes | I like the ones with the raw fish in them. |
| 20:00.25 | mercestes | how do I make IRC stop doing the annoying red line? I tried to type LAST and I did somethign terribly wrong. |
| 20:00.26 | mafkees | the ones with nemo |
| 20:00.29 | Lann | hai watashi to sakana ga suki |
| 20:00.35 | mercestes | mIRC I mean. |
| 20:00.47 | mafkees | get xchat ;) |
| 20:00.54 | sevard | Yes, get exploited. |
| 20:01.01 | *** join/#asterisk alrs (n=lars@dsl093-066-021.lax1.dsl.speakeasy.net) |
| 20:01.57 | mercestes | woo! exploitation! |
| 20:01.57 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-258255125a5e7f13) |
| 20:01.58 | Lann | they should make one called vagina |
| 20:01.58 | J4k3^ | irssi 4 life |
| 20:01.58 | sevard | "XChat, the surefire way to get ruined on IRC." |
| 20:01.58 | Lann | s/Xchat/vagina |
| 20:02.02 | mercestes | Would you like to exploit my xchat. |
| 20:02.06 | mafkees | I agree with J4k3^ |
| 20:02.15 | mercestes | I've never used my xchat. |
| 20:02.16 | Lann | im using mirc |
| 20:02.32 | mercestes | Bring it on hackers, my xchat has never been penetrated. |
| 20:02.39 | mercestes | yea, s/xchat/vagina is fun! :D |
| 20:02.46 | Moobius | hahah |
| 20:02.53 | Lann | hahaha |
| 20:02.53 | CrashSys | Yeah, I have mIRC too |
| 20:02.57 | CrashSys | :( |
| 20:03.05 | Lann | set mode=promiscuous |
| 20:03.09 | sevard | I use epic. |
| 20:03.11 | sevard | represent. |
| 20:03.19 | CrashSys | No BX eh? |
| 20:03.21 | mercestes | lol |
| 20:03.43 | sevard | I'd use irssi/BitchX if they didn't make my eyeballs bleed |
| 20:03.44 | mafkees | is BX still developed ? |
| 20:03.51 | CrashSys | I dunno |
| 20:03.52 | Lann | hahhahahaha |
| 20:03.54 | CrashSys | it still exists |
| 20:03.55 | mercestes | yea, linux is so full of dirty little utilities. touch, finger, mount, nmap. |
| 20:03.55 | Lann | google "define promiscuous" |
| 20:04.01 | Lann | thats such an undercover joke |
| 20:04.15 | CrashSys | probe |
| 20:04.17 | Lann | fsk! |
| 20:04.22 | Lann | fsck rather |
| 20:04.28 | mercestes | indeed. |
| 20:04.41 | *** join/#asterisk StyleWarz (i=stylewar@gayanalfisting.biz) |
| 20:04.44 | CrashSys | Who here wore the t-shirt in high-school that says "fsck me" |
| 20:04.47 | StyleWarz | Hey guys |
| 20:05.15 | StyleWarz | Any ideas how i can give priorities (not penalties) for ringing in a queue |
| 20:05.25 | CrashSys | :( |
| 20:05.36 | CrashSys | I guess no one also wore the shirt that said "C: |
| 20:05.37 | mercestes | Stylewarz: Creative penalties? |
| 20:05.54 | CrashSys | "C:\dos;C:\dos\run;run\dos\run" :( |
| 20:06.05 | *** join/#asterisk supjigatr (n=syslod@152.53.16.10) |
| 20:06.08 | supjigatr | Hi. |
| 20:06.12 | sevard | Goodbye. |
| 20:06.22 | StyleWarz | mercestes: What are those? |
| 20:06.23 | mercestes | lol |
| 20:06.36 | mercestes | CrashSys: Omg, that's funny. run\dos\run LOL |
| 20:06.38 | *** join/#asterisk mc74909 (n=mcl@dD57665A0.access.telenet.be) |
| 20:06.51 | supjigatr | Anyone know how to get national on normal calls and have 411 and 911 marked as unknow on PRI calls via asterisk? |
| 20:06.51 | mercestes | CrashSys: I always liked the "C:<enter> ###" |
| 20:06.52 | sevard | mercestes: You haven't seen that t-shirt? It's been around since like.... 1995 |
| 20:06.59 | mercestes | never seen it. |
| 20:07.02 | StyleWarz | So no ideas here? :( |
| 20:07.02 | Lann | that path would break dos |
| 20:07.47 | mercestes | StyleWarz: what you are asking for is a deviance from positive to negative numbers which is largely unnecessary if you simply deviate into greater degrees of positive numbres. |
| 20:07.50 | sevard | Lann: good job. |
| 20:08.00 | Lann | heh |
| 20:08.13 | mercestes | StyleWarz: There is nothing that priorities could facilitate that penalties do not in a call queue except the indulgence of laziness. |
| 20:08.22 | sevard | when I'm in DOS i'm always typing cd path/path/path and swearing when it doesn't work |
| 20:09.15 | Lann | "C:/( *.*)--/" |
| 20:09.29 | StyleWarz | mercestes: e.g. Penalty 2 only rings when for 1 is busy. But in some offices i might want to ring in a special order |
| 20:09.52 | CrashSys | I ran desqview in the days of BBS'... so it would just crash a window... |
| 20:10.27 | mercestes | StyleWarz: first ppl to ring, 0, second ppl to ring 1, third ppl to ring 3. Regardless of how you define it penalties and priorities are equivalent on the same linear scale with varying points of origination and perspective. |
| 20:10.30 | CrashSys | On my trust 386-16 4-meg ram |
| 20:10.49 | mercestes | Stylewarz: even if you could go 3, 2, 1, 0, -1, -2 it wouldn't matter because you coul dget the same thing with 0, 1, 2, 3, 4, 5 |
| 20:10.57 | sevard | mercestes: use smaller words with him. |
| 20:11.20 | StyleWarz | sevard: that would be great ;) |
| 20:11.26 | StyleWarz | sevard: English is not my native tounge :) |
| 20:11.53 | mercestes | oh. |
| 20:11.56 | mercestes | StyleWarz: no. |
| 20:12.04 | sevard | :D |
| 20:12.14 | mercestes | :) |
| 20:12.27 | sevard | I think that deserves a bash.org quote, gfi |
| 20:12.52 | anonymouz666 | mercestes: did you read that document about realtime openser and asterisk integration? |
| 20:13.04 | mercestes | sevard: ROFL |
| 20:13.25 | mercestes | anonymouz666: no, I filter all results with an implicit -openser in google. |
| 20:13.53 | *** part/#asterisk Araluccl0 (n=ciccio@adsl-ull-230-7.46-151.net24.it) |
| 20:14.12 | CrashSys | When I set priorities on a queue with two members, one with no penalty, another with a penalty of 1... it never rings the second guy, even tho the first guy is on the phone... |
| 20:14.20 | CrashSys | Unless i'm understanding something wrong |
| 20:14.49 | anonymouz666 | mercestes: asterisk asking for auth is still killing me :/ |
| 20:14.57 | mercestes | Crashsys: Something is broken then. |
| 20:15.01 | CrashSys | Yeah |
| 20:15.03 | CrashSys | the administrator |
| 20:15.05 | CrashSys | :) |
| 20:15.28 | mercestes | anonymouz666: You turn *off* authentication on Asterisk and accept all calls coming to * and then isolate it implicitly by IP from any host except OPenser and force all authentication through OpenSer. |
| 20:15.51 | mercestes | CrashSys: works for me on 1.2.13. Try upgrading.....or.....downgrading.. =/ |
| 20:16.04 | mercestes | Fire the second guy. He's using DND |
| 20:16.21 | CrashSys | I haven't tried with 1.2.14... last time I tried was 1.2.10 |
| 20:16.30 | anonymouz666 | mercestes: I can't just turn off authentication on asterisk by setting insecure=very, allowguest=yes, etc. It does not work. |
| 20:16.38 | CrashSys | they're end users, they dont know how to user the phones... |
| 20:16.40 | anonymouz666 | always ask for auth with a 407. |
| 20:17.05 | CrashSys | blah |
| 20:17.45 | mafkees | rm -rf endusers |
| 20:17.58 | mafkees | there, problem solved |
| 20:18.06 | *** join/#asterisk perd (i=[U2FsdGV@207.44.158.6) |
| 20:18.13 | perd | anyone have a configuration reference pdf for 7960 sip phones, i had one but i cant find the damn pdf anymore |
| 20:18.25 | CrashSys | maf: I like the fireaxe idea |
| 20:18.34 | mafkees | CrashSys: wanna borrow mine ? |
| 20:18.42 | anonymouz666 | insecure=yes works on [general] section of sip.conf? or only on peers/clients? |
| 20:18.48 | mercestes | CrashSys: You'll have to paste configs then. |
| 20:18.49 | mafkees | CrashSys: it's still a bit sticky with the leftovers of today |
| 20:18.51 | CrashSys | No, I plan to make a career out of this... I need to invest in my own equipment... |
| 20:19.13 | mafkees | perd: the wiki has a lot of info about it |
| 20:19.17 | mercestes | perd: sell the 7960 and get you a polycom. |
| 20:19.25 | perd | i wish i could mercestes |
| 20:19.28 | CrashSys | polycom! |
| 20:19.41 | *** join/#asterisk tim0123 (n=cash247@adsl-75-39-213-70.dsl.rcsntx.sbcglobal.net) |
| 20:19.45 | Lann | "insecure=very" heh |
| 20:19.50 | anonymouz666 | yes insecure=very |
| 20:19.53 | perd | does the 7960 sip firmware transmit silence by default? the 7912 doesnt |
| 20:20.00 | mafkees | or convert the 7960 to sccp |
| 20:20.17 | CrashSys | The worst thing about polycom's is where they say how many missed called you have on the LCD screen... |
| 20:20.25 | tim0123 | Is there anyway to record queues without getting hold time |
| 20:20.45 | J4k3^ | ... boss walks in "you have 17 missed calls" |
| 20:20.48 | J4k3^ | "you're fired!" |
| 20:21.08 | CrashSys | No, boss walks in "you have 17 missed calles" ... "Call the phone guy! This system is broken!" |
| 20:21.26 | J4k3^ | hehe |
| 20:21.53 | CrashSys | and the fun ensues |
| 20:22.23 | tim0123 | Nobody know how to record queues without recording hold time |
| 20:22.37 | *** join/#asterisk syzygyBSD_ (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 20:22.38 | wunderkin | heh i just had a problem report and they were dialing the wrong thing.. ugh |
| 20:23.08 | hal2385612 | hi all. I have been investigating the problem of the following errors when attempting to install the zaptel drivers on centos 4.4 http://pastebin.ca/341315 |
| 20:24.19 | hal2385612 | I am told that the sources for zaptel do not conform properly to normal good RPM packaging practice as per http://www.oldrpm.org/ |
| 20:24.21 | CrashSys | Why would you want to record a person in a queue? |
| 20:24.34 | perd | crashsys: to laugh at them |
| 20:24.48 | CrashSys | perd: d00d, that's almost reason enough for me to be interested... |
| 20:24.52 | hal2385612 | can you suggest anyone I can speak to in order to let them know of this issue? |
| 20:25.01 | *** join/#asterisk flying_Luck (n=melifaro@ppp85-141-152-80.pppoe.mtu-net.ru) |
| 20:25.05 | hal2385612 | has anyone else has similar problems? |
| 20:25.12 | *** join/#asterisk nn (n=joseph@cpe-72-229-104-43.nyc.res.rr.com) |
| 20:25.13 | CrashSys | I dont use RPM |
| 20:25.14 | mafkees | indeed, nice idea |
| 20:25.16 | *** join/#asterisk _Vile (n=vile@bc182112.bendcable.com) |
| 20:25.17 | CrashSys | I use sources... |
| 20:25.32 | nn | Is there a way to config * to look up extensions/etc from an ldap db? |
| 20:25.48 | CrashSys | I haven't used RPM's since I last messed with redhat 7-something |
| 20:25.48 | Corydon-w | Not currently, no |
| 20:25.49 | mafkees | nn: yeah. there's res_ldap |
| 20:26.02 | Corydon-w | mafkees: not in a release, there isn't |
| 20:26.04 | nn | Such that i could put a current contact info (SIP URL, mobile, etc) for users into their LDAP roaming profile, and have it use custom queries to ask for contact info on a user |
| 20:26.08 | mafkees | Corydon-w: ah |
| 20:26.09 | nn | Corydon-w: cvs/svn is fine |
| 20:26.32 | nn | im trying to LDAPify everything that involves user location and authentication |
| 20:26.37 | Corydon-w | It's not in there, either |
| 20:26.49 | mafkees | it's not in the official asterisk tree |
| 20:26.52 | Corydon-w | There's a proposed patch on the bugtracker |
| 20:26.53 | nn | since all my users are mobile, it would be great |
| 20:27.25 | Corydon-w | There is, however, ODBC abstraction |
| 20:27.26 | mercestes | hal2385612: Don't use RPMs |
| 20:27.50 | Corydon-w | and the ODBC abstraction is well-tested |
| 20:28.17 | hal2385612 | merc - hello again - yes, I think you are right. If I hadn't used checkinstall, things would have installed and I would have been none-the-wiser |
| 20:28.21 | nn | hmm |
| 20:28.25 | robl^ | I dislike anything ODBC |
| 20:28.28 | nn | Corydon-w: and the ODBC stuff can talk to LDAP? |
| 20:28.35 | nn | If not, it's of no use for this |
| 20:28.39 | Corydon-w | Don't think so |
| 20:28.43 | nn | then its useless :) |
| 20:28.58 | Corydon-w | The LDAP patch is not well-tested |
| 20:29.03 | nn | its impossible to maintain a seperate DB for this |
| 20:29.15 | *** part/#asterisk mc74909 (n=mcl@dD57665A0.access.telenet.be) |
| 20:29.31 | robl^ | nn you could alwyas create a script that polls LDAP and then gnerates an asterisk extensions file.. or populates a database table |
| 20:29.33 | nn | i'll have to check it out |
| 20:29.37 | mafkees | nn: look here: http://bugs.digium.com/view.php?id=5768 |
| 20:29.42 | CrashSys | it compiled |
| 20:29.50 | CrashSys | have to see if it stays stable later on |
| 20:29.51 | nn | robl^: thatd be terribly ineffective |
| 20:29.56 | Corydon-w | CrashSys: well, of course it compiled |
| 20:30.07 | CrashSys | :) Ya never know... |
| 20:30.08 | nn | robl^: users move rapidly and their location is determined via bluetooth and other means :) |
| 20:30.11 | Corydon-w | Whether it works correctly is another matter |
| 20:30.31 | robl^ | nn: its a possible solution until native LDAP support is production ready |
| 20:30.40 | Corydon-w | nn: but their phones don't change authentication |
| 20:31.01 | robl^ | nn: ahh!! how dare users move! just invest in some rope and tie the users to their desk |
| 20:31.04 | nn | Corydon-w: the bluetooth is just used to determine approximate location |
| 20:31.40 | *** join/#asterisk hohum (n=dcorbe@mercury.sunrocket.com) |
| 20:31.42 | mafkees | nn: did you have a look at that link I gave you ? |
| 20:31.50 | nn | mafkees: yea, looking at it now |
| 20:32.06 | mafkees | I did not test it or anything |
| 20:32.38 | nn | we use a bluetooth presence daemon which basically updates LDAP with the current room it sees us in, forwards calls to the nearest desk (SIP) phone, if we're not present in the building, it routes calls to our mobile |
| 20:33.04 | mafkees | nice |
| 20:33.06 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 20:33.07 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 20:33.14 | mafkees | combine that with the stuff in plutohome.org |
| 20:33.20 | mafkees | makes a nice setup |
| 20:33.42 | nn | each room has a 800mhz via nemiah running off 12V (comes via PoE) |
| 20:33.46 | ucionas | my asterisk does a core-dump when using "i" in Queue() |
| 20:33.56 | ucionas | any idea? |
| 20:34.03 | *** join/#asterisk h0 (i=fakhir@unaffiliated/fakhir) |
| 20:34.18 | nn | mafkees: working on setting up clustering, so no servers will be needed here :) |
| 20:34.35 | CrashSys | Man... I purposely try to avoid getting caught near a ringing phone... that whole set-up just sounds like trouble :( |
| 20:34.49 | nn | CrashSys: this is in my house/office :) |
| 20:34.54 | CrashSys | Ohh |
| 20:34.58 | CrashSys | makes sense then |
| 20:35.14 | nn | just makes life more convenient |
| 20:35.16 | CrashSys | other then you have to carry a bluetooth device with you at all times... |
| 20:35.26 | nn | CrashSys: my phone is usually with me anyways |
| 20:35.38 | robl^ | nn: what happens when the receptionist walks down the all?? al incoming calls ring in each office as she passes on her way to the restroom? |
| 20:35.43 | CrashSys | X10 the power and write a script to auto on/off the lights |
| 20:35.45 | mafkees | get one of the nokia E70 phones |
| 20:35.56 | mafkees | they do sip and gsm |
| 20:35.57 | nn | robl^: you've got to be present for a bit before the BTPD picks you up and updates |
| 20:36.06 | CrashSys | use asterisk and voice recognition and microphone your house, issue commands... |
| 20:36.06 | mafkees | lol robl^ |
| 20:36.08 | CrashSys | USS-NN |
| 20:36.09 | anonymouz666 | e61 does not work with asterisk |
| 20:36.13 | anonymouz666 | only register with ser |
| 20:36.19 | mercestes | ucionas: oh! your on a FreeBSD 2006.1 install using asterisk 1.2.12-r4 with the optional ztdummy interfaces and the RTA options with asterisk-addons. Yea, try updating to r5 and recomplining and it will solve everything. |
| 20:36.20 | nn | also, if two people are in one area, the one their longest keeps the 'seat' |
| 20:36.32 | anonymouz666 | e61 does not know how to answer a 407 for register handshake |
| 20:36.33 | nn | allows 'visiting' other people briefly |
| 20:36.41 | CrashSys | Get the enterprise sound and PA the house, have it play the door swoosh sound when you walk in... |
| 20:36.49 | nn | CrashSys: hah |
| 20:36.59 | mafkees | lol CrashSys |
| 20:37.00 | wunderkin | just dont stay too long :P |
| 20:37.14 | mafkees | now all you need is a 'beam me up' room |
| 20:37.18 | CrashSys | d00d, i'd be allll star-trekked out if I had that... |
| 20:37.28 | CrashSys | maf: it's in the basement... with billy bong thorton... |
| 20:37.42 | nn | goal is to minimize power usage and at the same time, maximize ease-of-use |
| 20:38.03 | robl^ | and their bluetooth device battery dies.. so.. "they are not in the building". Which means ring their cell phone, which coincidentally is the bluetooth device with the dead battery |
| 20:38.05 | nn | another example, if no one's in a room for more than 10 minutes, the lights turn off |
| 20:38.31 | nn | robl^: thats when you manually set your presence |
| 20:39.24 | CrashSys | You call #1 and tell him to turn on your ready-room lights |
| 20:39.29 | CrashSys | Cause the borg are coming |
| 20:39.34 | nn | hah |
| 20:39.34 | mafkees | will it start the coffee machine when I get in the house after a leave of 10 minutes or more ? |
| 20:39.55 | nn | mafkees: If scripting for the BTPD stuff where present and you had an X10 capable coffee maker, sure |
| 20:40.13 | CrashSys | Get an old-school auto-off coffee maker... |
| 20:40.19 | CrashSys | plug it in, it makes coffee... |
| 20:40.36 | nn | that works too |
| 20:40.43 | mafkees | my coffee machine is connected to this velleman relaiscard connected to my printerport |
| 20:40.43 | nn | and use an X10 switchable outlet |
| 20:40.44 | J4k3^ | heh... you mean bluetooth works? |
| 20:40.44 | *** join/#asterisk suma (n=suma@cm207.omega186.maxonline.com.sg) |
| 20:40.51 | CrashSys | Could make an X-10 coffee make... two X10 outlets... |
| 20:40.51 | mafkees | I can dialin to activate it |
| 20:40.54 | suma | i have a problem checking out zaptel |
| 20:41.03 | suma | Fetching external item into 'oct612x' |
| 20:41.03 | suma | svn: Can't read directory 'oct612x': File exists |
| 20:41.17 | robl^ | *contemplates X10 and a vibrating bed* |
| 20:41.19 | suma | and stops there |
| 20:41.19 | *** join/#asterisk RoyK (n=roy@ti211310a080-5551.bb.online.no) |
| 20:41.32 | *** join/#asterisk Lymers (n=Lyme@pool-71-244-47-70.dllstx.fios.verizon.net) |
| 20:41.35 | SplasPood | anonymouz666: E61 works with my asterisk.. |
| 20:41.36 | J4k3^ | kinky. |
| 20:41.43 | suma | svn: Can't read directory 'oct612x': File exists , how can i remove this svn error please |
| 20:41.45 | nn | robl^: the coolest part is, anything you can get X10 or BT hw for, you can do with some creative scripting |
| 20:41.46 | CrashSys | X10 and something that vibrates... |
| 20:42.41 | J4k3^ | does Nokia sell a pure-wifi device yet? I refuse to purchase anything with GSM attached. |
| 20:42.47 | mafkees | J4k3^: yeah I think so |
| 20:42.53 | nn | why would you not want GSM? |
| 20:42.59 | mafkees | they have this pda thingie |
| 20:43.20 | J4k3^ | nn: because its horribly inferior on a whole lot of levels. I'm in the USA where we aren't stuck with that old crap. |
| 20:43.21 | robl^ | oh man! that could be used for some great pranks. pair a secretary's cell phone with BTPD.. then as she walks around, have lightsflash on and off. Make her think she's being followed by a poltergeist |
| 20:43.49 | mafkees | gheh |
| 20:43.56 | *** join/#asterisk remmo (n=chatzill@203.22.186.225) |
| 20:43.59 | mafkees | or play gunsounds on the radio |
| 20:44.09 | mafkees | make her think they are shooting at her |
| 20:44.18 | nn | robl^: hahah |
| 20:44.41 | nn | i need to move the floor plan in the LDAP soon |
| 20:44.59 | nn | so i can start populating rooms with devices and their IPv6/X10 ID |
| 20:45.21 | CrashSys | Why IPv6? |
| 20:45.21 | CrashSys | out of curiousity |
| 20:45.30 | robl^ | IPv6 is the future! |
| 20:45.36 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 20:45.36 | mercestes | CrashSys: because IPv4 is dead and gone. |
| 20:45.36 | CrashSys | ohh |
| 20:45.55 | *** join/#asterisk gerhard7 (n=gerhard@82-171-117-191.dsl.ip.tiscali.nl) |
| 20:45.57 | Lymers | anyone have any advice for troubleshooting crackling noise on the asterisk side of the line, but the caller does not hear anything |
| 20:46.00 | nn | CrashSys: with ipv6, i can give each item in my house an IP |
| 20:46.05 | mercestes | it's like VHS....it's dead...we all know it's dead ,but it still lingers around refusing to be gone. |
| 20:46.10 | nn | and eventually make anything i desire publically accessible |
| 20:46.26 | robl^ | except that its impossible to get a true IPv6 connection from 99.9% of the ISPs. you have to to IPv^ tunneling over IPv4 |
| 20:46.27 | mercestes | Lymers: First, is it voip or not? |
| 20:46.35 | CrashSys | One of these days I should read-up on IPv6 |
| 20:46.49 | mafkees | ipv6 is nice |
| 20:46.58 | mercestes | ipv6 is a pain in the arse. |
| 20:47.23 | mafkees | gheh, because you cannot remember the ip's ? |
| 20:47.30 | nn | <3 dns |
| 20:47.36 | mafkees | indeed |
| 20:47.48 | nn | I've spent the last 2 weeks configuring dns, ipv6, kerberos, ldap, etc |
| 20:48.02 | CrashSys | I still like my 192.168 :( |
| 20:48.04 | Lymers | mercestes: its both |
| 20:48.19 | nn | CrashSys: ah i still have ipv4 for the antiques (windows machines, etc) |
| 20:48.20 | mercestes | trying to memorize that fe80::fi70:fo108:fum69 is like getting brain surgery with no anesthesia. |
| 20:48.28 | nn | mercestes: Use DNS then. |
| 20:48.40 | nn | how hard is it to make a zone file? Not hard at all |
| 20:48.50 | J4k3^ | ipv6 is an extension of a bad idea. |
| 20:48.51 | mercestes | nn: PAIN! |
| 20:49.14 | *** join/#asterisk oys (n=eoyslebo@cm-84.208.78.084.chello.no) |
| 20:49.24 | anonymouz666 | what hardware do you recommend for using 20 analogic phones? |
| 20:49.34 | mercestes | Lymers: ....is the crackle on voip? or not voip? |
| 20:49.43 | mercestes | Lymers: describe your call. |
| 20:49.52 | nn | next i have to set up an IVR :) |
| 20:50.05 | mercestes | Yea, that's easy. |
| 20:50.09 | nn | yea |
| 20:50.19 | nn | I want tts and voice rec tho :) |
| 20:50.24 | nn | got Cepestral just need to install it |
| 20:50.25 | suma | zaptel-1.2 svn checout breaks |
| 20:51.08 | Lymers | Lymers: its on voip connections, and zaptel connections... is doesnt matter... any incomming call has crackling issues |
| 20:51.24 | nn | gotta do something to send an SMS with an OTP in addition to smart cards tho, too for root login |
| 20:52.01 | CrashSys | Lymers: what's your load? |
| 20:52.21 | nn | that way root is verified in addition to having a smart card by having to have the registered (in LDAP) mobile # |
| 20:53.06 | mercestes | Lymers: ....what type of phones? |
| 20:53.13 | mercestes | Lymers: voip doesn't make "cracking noises." |
| 20:53.32 | Lymers | okay |
| 20:53.39 | Lymers | the 2 lines that are lit up are zap lines |
| 20:53.54 | *** join/#asterisk jarg (n=jarg@189.157.102.39) |
| 20:54.00 | Lymers | the phones are sipuras |
| 20:54.19 | Lymers | load is only like 2 calls right now... |
| 20:54.20 | Grnd-Wire | So what is the possibility that I got a bad Digium board? |
| 20:54.48 | Grnd-Wire | Port 1 isn't connecting calls, but the other 3 ports are.. Could this be a misconfiguration issue, or has anyone seen something like this before? |
| 20:55.11 | mercestes | Lymers: Zap lines can make crackly noises. sip cannot. Now if you are going from Sipura to * to Zap lines...that's not a voip call. |
| 20:55.34 | mercestes | Lymers: Now i fyou are going from Sipura to * to SIP or IAX to some IP address or DNS name, then it is a VoIP call. |
| 20:55.40 | Lymers | yes i know.... but im just kinda passing down info that was given to me |
| 20:55.40 | Lann | where could i find an example of php opening one of asterisks' voicemail files and playing it to caller? |
| 20:56.02 | Lymers | but the crackling seems to be comming mainly on non voip lines |
| 20:56.12 | mercestes | Lymers: That mainly makes sense. |
| 20:56.23 | CrashSys | pstn! |
| 20:56.23 | mafkees | Lann: no idea, but it shouldn't be too hard |
| 20:56.25 | mercestes | Lymers: if the crackling noises were entirely coming from non-voip lines, then it would make entire sense. |
| 20:56.32 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 20:56.34 | mafkees | Lann: you can use the Playback app for that |
| 20:56.47 | Lann | yah if i can do that, i can just delete the file with php for delete, and easily associate data with a file using maybe mysql and the filename |
| 20:56.59 | mercestes | Lymers: And you would have problems with your PRI/BRI and would need to contact the telco responsible for that T1 and tell them you have noise on yoru lines. |
| 20:57.26 | Lann | do asterisk voicemail files have some sort of unique naming system? |
| 20:57.38 | Lann | like a timestamp and such |
| 20:57.46 | oys | anyone had any luck getting a pci-modem working as an fxo? looking at my options for buying a cheap fxo for home use |
| 20:58.03 | CrashSys | oys: Yes, they are called X100's |
| 20:58.03 | Lymers | they are straight phone lines, not a entire t1, and also if i plug in a normal phone the crackling stops |
| 20:58.04 | Corydon-w | Lann: no, they're ordered |
| 20:58.21 | Lann | is the ordering unique? |
| 20:58.31 | Corydon-w | Lann: 0000, 0001, 0002, 0003, etc. |
| 20:58.41 | Lann | for anyone's voicemail or just a given extension/ |
| 20:58.51 | Corydon-w | Anyone |
| 20:59.00 | oys | CrashSys: thank you.. will google/research.. CrashSys: do they work well (for a home setup, that is)? |
| 20:59.09 | Lann | so bob's first mail is 0001, and bill's first mail that was created afterwards is 0002? |
| 20:59.16 | mafkees | no |
| 20:59.22 | mafkees | per mailbox |
| 20:59.23 | Corydon-w | Lann: we use directories for that |
| 20:59.28 | mercestes | Lymers: Any reason you withheld thsi information until the end of your query? |
| 20:59.35 | Lann | k well thats enough still for me to associate mysql data |
| 20:59.44 | Lann | woot |
| 20:59.56 | Lymers | ehh, haha.... sorry |
| 21:00.36 | mercestes | Anyone else get this feeling? That #asterisk is kinda like a game of CLUE(tm)? Ppl give you a little information...and you guess, "Sir bob, in the library, with the candle-stick!" and they go "NO!" and then give you a bit more informatino? |
| 21:01.00 | Grnd-Wire | heh.. yeah! |
| 21:01.04 | Lymers | haha i know the feeling, sorry about that... |
| 21:01.17 | mercestes | Lymers: So you have *, and FXO ports, and regular phone lines? |
| 21:01.37 | The_DoC^ | wouldn't it be called spoon feeding if they told you the exact answer and not make you try and learn for yourself |
| 21:01.56 | mercestes | Lymers: And these regular phone lines go to........just telco? Nothing freaky like another PBX or channel bank or something? |
| 21:02.05 | mercestes | s/fxo/fxs/ |
| 21:02.16 | Lymers | yes |
| 21:02.38 | CrashSys | I have a problem with crackle on phone lines from the PSTN... it used to be line #3... so they called verizon, they come out, fix it, and now line #1 is crackling :D |
| 21:02.46 | CrashSys | basically, when it rains, it crackles... |
| 21:02.57 | Lymers | well we are in a wharehouse, there is a possibility these lines go through something else, but i have no clue where or control of that part |
| 21:02.59 | mercestes | Lymers: Ok, then you likely have a wiring problem between your FXS ports and your POTS interface. |
| 21:03.06 | The_DoC^ | thats verizon for you CrashSys |
| 21:03.08 | Grnd-Wire | CrashSys: There is moisture in the splices.. |
| 21:03.23 | saftsack | is it possible to have an global telephone book on the asterisk server which is read out by the telephones? (polycoms) |
| 21:03.31 | mercestes | Yea, it's called "water on your lines" and is very common. |
| 21:03.40 | Lymers | well more info would be, it doesnt happen all the time, only sometimes |
| 21:03.41 | mercestes | It's common because...the telco doesn't want to spend the $$$ to find and fix it. |
| 21:03.52 | mercestes | Lymers: Oh! In that case...... |
| 21:03.53 | CrashSys | Aint no moisture on my side... brand-new month-old Cat3 plenum cabling from verizon's demarc to the sangoma A200 (one-piece) |
| 21:04.01 | mercestes | Lymers: Ok, then you likely have a wiring problem between your FXS ports and your POTS interface. |
| 21:04.34 | Lymers | so in other words its not on my end? |
| 21:04.38 | The_DoC^ | CrashSys : its probably from a RT to your DEMARC |
| 21:04.54 | The_DoC^ | or from the co to the rt |
| 21:04.57 | mercestes | Also if it' sonly on incoming lines and you have some sort of PBX doing trunking for you then testing with a phone isn't going to be very productive. |
| 21:04.58 | *** join/#asterisk UlbabraB (n=salama@host241-43-static.72-81-b.business.telecomitalia.it) |
| 21:05.03 | Grnd-Wire | The_DoC^: From the LMNO to the OPQRST? |
| 21:05.03 | Lymers | its more or less who controls what my phone lines do after i plug em in? |
| 21:05.05 | CrashSys | The_DoC: Prolly... old-school GTE laid a lot of paper-insulated cable in the ground around here... |
| 21:05.08 | mercestes | Lymers: It's not on *. |
| 21:05.13 | CrashSys | and verizon aint replacing it like they should... |
| 21:05.43 | The_DoC^ | I know how that is, same situation here |
| 21:05.52 | Lymers | uhrm.... well i figured it was because when i hook up a normal phone to the same phone jacks... the lines are clear |
| 21:06.00 | mercestes | Lymers: I suppose it is possible that the DC circuitry is some how loose or shorted out in yoru box and instead of frying yoru mobo/powersupply or tripping the power supplies voltage/amp/resistance sensors it's generating crackling instead. |
| 21:06.09 | |Vulture| | Anyone have issues with Asterisk reading voicemail timestamps incorrectly? |
| 21:06.11 | CrashSys | So instead you drive down the street, with nitrogen tanks every 4-blocks, trying to evacuate the cable... |
| 21:06.59 | mercestes | Lymers: Ok, the fxo/fxs ports *could* be bad, but not likely. |
| 21:07.15 | |Vulture| | everything is reading 3hrs into the future |
| 21:07.19 | mercestes | Lymers: is there another PBX in the picture? Or is it just 4 telco lines with 4 DIDs? |
| 21:07.41 | Lymers | 4 telco lines, 4 dids |
| 21:07.42 | CrashSys | I'm all for doing like they do in europe and have the cable/wiring infrastructure be part of the utilities and the telco's just lease copper from the cities... |
| 21:07.59 | The_DoC^ | when they would do a MLT test on my lines here when it rained it would come up 1000 feet longer than what it is |
| 21:08.03 | Lymers | well there is a preinstalled system by the phone company |
| 21:08.04 | mafkees | |Vulture|: did you check the asterisk clock ? |
| 21:08.16 | Lymers | like a rollover for busy lines |
| 21:08.25 | |Vulture| | mafkees: on the local server, as in uptime? yes it reads correct local time |
| 21:08.32 | mercestes | Lymers: An dyou tested all 4 lines? |
| 21:08.37 | Lymers | yes |
| 21:08.42 | mafkees | timezones setup correctly ? |
| 21:08.47 | mercestes | Lymers: ok...so yo uhave a BOX that outputs 4 analog lines. |
| 21:08.59 | anonymouz666 | what about a te1XXP and a TDM with 2 FXO (backup) in the same machine? |
| 21:09.11 | mercestes | Lymers: hook in an analog phone and then call the # and answer the call on the phone. |
| 21:09.21 | Lymers | actually 5, but the 5th line is seperate and does go through another hpone system, and has no probs |
| 21:09.25 | |Vulture| | mafkees: in /etc/localtime and /etc/sysconfig/clock |
| 21:09.26 | CrashSys | I need a butt set... |
| 21:09.34 | CrashSys | someone liberated mine last year :( |
| 21:09.59 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 21:10.05 | The_DoC^ | was it a analog or a analog/digital |
| 21:10.16 | CrashSys | Harris TS22 |
| 21:10.17 | CrashSys | analog |
| 21:10.22 | CrashSys | no ISDN |
| 21:10.23 | Lymers | im actually out of the office, i just keep getting asked about it |
| 21:10.26 | *** join/#asterisk JimmyBerulla (n=jimmy_be@86.107.118.203) |
| 21:10.41 | Lymers | all i have is telnet/remote administration |
| 21:10.54 | *** join/#asterisk fiber0pti (n=John@207.114.199.107) |
| 21:11.21 | *** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com) |
| 21:11.23 | fiber0pti | Does anyone know where I might find out how to configure a Fanstel phone on Asterisk? i.e. Fanstel ST2118 |
| 21:11.36 | CrashSys | Fanstel? |
| 21:11.44 | mafkees | fanstel ? |
| 21:11.44 | *** join/#asterisk crich1999 (n=crich@port-212-202-210-130.dynamic.qsc.de) |
| 21:11.44 | saftsack | is it possible to have an global telephone book on the asterisk server which is used by the telephones? (polycoms) |
| 21:11.47 | mafkees | never heard of that |
| 21:11.54 | fiber0pti | Nod: http://www.fanstel.com/cgi-bin/products.exe?VoipBusiness |
| 21:12.10 | CrashSys | saftsack: Yeah, look at the directory.xml file |
| 21:12.27 | CrashSys | sounds like a grandstream knock-off |
| 21:12.40 | fiber0pti | heh |
| 21:12.41 | fiber0pti | nod |
| 21:12.45 | fiber0pti | packet brands them |
| 21:12.47 | fiber0pti | err packet 8 |
| 21:12.56 | fiber0pti | actually a lot of voip providers are branding them |
| 21:12.56 | The_DoC^ | I have a harris ts42 |
| 21:13.07 | *** join/#asterisk w0ls0n (n=Me@43-141-135-64.dsl.sacoriver.net) |
| 21:13.32 | CrashSys | The_DoC: big spender :) |
| 21:13.39 | CrashSys | All I ever need is polarity and dial-tone |
| 21:13.40 | w0ls0n | hi all. |
| 21:13.44 | CrashSys | I dont need much else |
| 21:13.53 | The_DoC^ | I didn't pay for it |
| 21:13.55 | w0ls0n | I have a phone provider thats a monopoly. |
| 21:13.59 | The_DoC^ | well I kinda did |
| 21:14.07 | CrashSys | The_DoC: In that case, wanna sell it? :) |
| 21:14.10 | w0ls0n | Basically any viop provider doesn't have a number in my area |
| 21:14.16 | The_DoC^ | no, I do use it |
| 21:14.25 | w0ls0n | any way around having to pay long distance calls on port forwarding? |
| 21:14.31 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 21:14.51 | CrashSys | The_DoC: You work for a TelCo? or do wiring? |
| 21:15.22 | The_DoC^ | I do tele-data part time |
| 21:15.34 | *** join/#asterisk karmatronic (n=karmatro@84.77.131.137) |
| 21:15.42 | CrashSys | same here |
| 21:15.49 | CrashSys | Start from layer 1 and go up |
| 21:15.50 | saftsack | CrashSys, cool, thanks :) |
| 21:15.56 | The_DoC^ | just the cable installs |
| 21:16.51 | The_DoC^ | smaller setups I do the full install |
| 21:17.12 | *** part/#asterisk nn (n=joseph@cpe-72-229-104-43.nyc.res.rr.com) |
| 21:17.27 | CrashSys | I will do parts, but typically deliver a full install... |
| 21:17.39 | CrashSys | primarily cause I hate dealing with half the hacks down here |
| 21:17.50 | The_DoC^ | its a spin-off of a wireless co-op I run in my area |
| 21:18.33 | CrashSys | I hate trying to deliver a server and the wiring is crap cause the kid down the street who did it for $50 didn't have a punch-tool for the 110 jacks (not a joke)... |
| 21:18.46 | w0ls0n | theres no local numers |
| 21:18.51 | w0ls0n | well our "local" exchange is 929,727,247 |
| 21:18.57 | CrashSys | or do the wiring and their MCSE comes in and bitches cause I used T568a instead of T568b... |
| 21:18.57 | w0ls0n | and the closest number for voip is a portland number and thats long distance |
| 21:19.03 | w0ls0n | so If I tell my phone provider to port forward our current number to a portland number, we're gonna be paying for long distance |
| 21:19.16 | *** join/#asterisk andyd (n=andy@grumps.nosignal.org) |
| 21:19.28 | andyd | How can you work out which mac address a g.729 license is registered to, please ? |
| 21:20.08 | The_DoC^ | been there, soneone once used a butter knife to do the punchdowns and half of the termitations were shit and I had to go through and replace the jacks cause the contacts were trashed |
| 21:20.26 | saftsack | every person which is on the other side of my grandstream phone says that the sound is more than crap *G* |
| 21:21.28 | CrashSys | The_doc: that beats me... worst I had is where they tried to pull the wire down into the 110's and had like 1/2" of bare copper pulled out the side of the keystone jacks... |
| 21:21.44 | CrashSys | and like 3/4" of insulation piled up in the 110 |
| 21:22.16 | CrashSys | Another install they wired all the Cat5 jacks in series (like you would a TelCo POTS line) |
| 21:22.53 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 21:23.00 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 21:24.15 | CrashSys | The stupidest call I ever got from an IT guy was when he called me telling me my wiring was crap... I asked why, he said cause it wasn't wired to T568b standard... to which I replied "No, it's T568a"... and he says, and I quote, "Well that would be the old version, that's why it's a"... |
| 21:24.17 | *** join/#asterisk twisted[asteria] (n=twisted@pdpc/supporter/active/twisted) |
| 21:24.17 | *** mode/#asterisk [+o twisted[asteria]] by ChanServ |
| 21:24.22 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 21:24.31 | Lymers | whats the command to enable record on demand? |
| 21:24.42 | *** join/#asterisk adop (n=adonis@ppp091-090.dsl.hol.gr) |
| 21:24.47 | aptura | Anyone know if the polycom ip 500/501 uses a generic power adapter or one with the polycom logo on it? |
| 21:24.50 | The_DoC^ | you terminate in T568A , all I do is T568B unless otherwise requested |
| 21:25.17 | CrashSys | I find that T568a does a neater job at the patch panel |
| 21:25.25 | CrashSys | patch cords I do in B |
| 21:25.43 | CrashSys | or whatever is there before me |
| 21:26.13 | *** part/#asterisk w0ls0n (n=Me@43-141-135-64.dsl.sacoriver.net) |
| 21:26.19 | CrashSys | They're electrically equivalent... |
| 21:26.28 | The_DoC^ | I know mostly edu and gov stuff is done in T568A |
| 21:26.59 | CrashSys | Most home stuff is B |
| 21:27.07 | CrashSys | most commercial is A |
| 21:27.27 | CrashSys | but it's all electrically equivalent... as long as you dont mix them on segments your good to go.. |
| 21:27.32 | The_DoC^ | pretty much anymore its what the IT contrator wants |
| 21:27.46 | The_DoC^ | contractor even |
| 21:28.15 | *** part/#asterisk StyleWarz (i=stylewar@gayanalfisting.biz) |
| 21:28.41 | |Vulture| | wow * time is being a whore |
| 21:28.59 | |Vulture| | systemtime reads EST, but voicemail seems to think I am in PST |
| 21:29.06 | |Vulture| | even though tz= is not set |
| 21:29.27 | *** part/#asterisk adop (n=adonis@ppp091-090.dsl.hol.gr) |
| 21:29.38 | aptura | Atacom is still a major reseller of polycom phones? I am having a little difficulty looking up there site. I think thay are located in the upper mid western states? |
| 21:29.52 | CrashSys | atacomm |
| 21:29.55 | Corydon-w | Minnesota |
| 21:29.56 | aptura | ooookay |
| 21:29.57 | CrashSys | www.atacomm.com |
| 21:30.09 | aptura | yea got my phone from them |
| 21:30.16 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 21:30.18 | CrashSys | They treat me good... |
| 21:30.38 | CrashSys | shipping is a lil high on average... |
| 21:30.44 | CrashSys | But you can call and get a real person :D |
| 21:30.54 | CrashSys | no damn e-mail |
| 21:32.43 | Lann | if you delete a voicemail in a user's directory, does the naming scheme start at the highest present number? |
| 21:33.07 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 21:33.07 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 21:33.33 | Corydon-w | Lann: if you delete a voicemail, the remaining messages are renumbered |
| 21:33.58 | twisted[asteria] | yep |
| 21:34.10 | twisted[asteria] | if no msg0000.* exists, then asterisk will report VM for you, but not let you hear it ;) |
| 21:34.38 | Corydon-w | twisted[asteria]: actually, that should be solved now |
| 21:34.48 | twisted[asteria] | it's a nifty feature for driving assistants insane |
| 21:34.56 | twisted[asteria] | Corydon-w, yea well |
| 21:36.17 | aptura | anyone here have a ip500 running next to them? |
| 21:36.40 | twisted[asteria] | i have a 601 |
| 21:36.41 | twisted[asteria] | ;) |
| 21:38.53 | robl^ | I have a 601, but its 12 miles away. close enuff? |
| 21:40.10 | Jason99 | anyone know the Alert-Info: data required for distinctive ring on D-Link ATA? |
| 21:44.05 | *** join/#asterisk sivana (n=sivana@208.68.91.47) |
| 21:46.37 | *** part/#asterisk a1fa (n=a1fa@unaffiliated/a1fa) |
| 21:48.20 | aptura | need to find somone who has a working ip500 |
| 21:49.33 | aptura | did any of the old polycoms use a transformer that said "Potrans" on them? |
| 21:49.35 | *** join/#asterisk brussel_ (n=brussel@cpe-75-80-175-84.san.res.rr.com) |
| 21:49.59 | CrashSys | IP300's |
| 21:50.03 | CrashSys | cause I got 3 |
| 21:50.35 | aptura | CrashSys so thay say Potrans on them? |
| 21:51.11 | aptura | whats the amp and volt output of them? |
| 21:51.11 | CrashSys | 38kv a 5amps |
| 21:51.11 | CrashSys | :) |
| 21:51.16 | aptura | im not joking about this. |
| 21:51.20 | CrashSys | i would half to like, get under the desk tho... |
| 21:51.22 | CrashSys | :( |
| 21:51.28 | CrashSys | hold my beer, i'll BRB |
| 21:52.06 | CrashSys | 12v@400ma center positive |
| 21:52.30 | aptura | wow really. What does it say on the power supply? |
| 21:52.35 | *** join/#asterisk foxxtrot (n=craig@c-67-185-0-172.hsd1.wa.comcast.net) |
| 21:52.41 | aptura | Because this one is 400 ma with 12vdc out |
| 21:53.18 | CrashSys | 120v AC |
| 21:53.36 | CrashSys | output is 12v DC @ 400ma with Center Positive on plug |
| 21:53.48 | CrashSys | I'm going back to my beer now |
| 21:55.01 | CrashSys | Or did you want like model numbers? |
| 21:55.59 | Lymers | Can someone please give me a possible diagnosis? All my incomming and outgoing calls over my zaptel lines have a crackling noise, they are normal telco lines (4) i have used analog phones on these lines and have no noise. The voip line on the same * box has no noise. When i reboot my system the crackling goes away for a few hours and then comes back slowly at first and gets worse. Ideas? |
| 21:56.19 | aptura | what is it? |
| 21:56.34 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 21:56.52 | CrashSys | What kind of Zap card? |
| 21:56.58 | Lymers | tmd400 |
| 21:56.59 | *** join/#asterisk eryc (i=enkrypte@71.4.147.219.ptr.us.xo.net) |
| 21:57.35 | Lymers | i think thats it, is is a fxo (4) digium card |
| 21:57.41 | aptura | CrashSys who are you taking to |
| 21:58.17 | *** join/#asterisk dseeb_ (n=dcb@CPE-58-169-80-108.vic.bigpond.net.au) |
| 21:59.10 | *** mode/#asterisk [+o angler] by ChanServ |
| 22:00.13 | *** join/#asterisk CrashSys (n=kumba@208.177.233.66.ptr.us.xo.net) |
| 22:00.25 | CrashSys | Potrans model # WD411200400 |
| 22:00.35 | CrashSys | Input: 120vac 60hz 9w |
| 22:00.45 | CrashSys | output: 12vdc @ 400ma LPS |
| 22:00.55 | aptura | yea thats the same one. |
| 22:01.00 | aptura | and the only one |
| 22:01.15 | aptura | I just measured its output on my fluke meter. 17 volts |
| 22:01.18 | *** join/#asterisk atca_pres (n=rick@modemcable128.204-81-70.mc.videotron.ca) |
| 22:01.36 | aptura | I hope the power supply did not cook my phone! |
| 22:01.39 | CrashSys | sounds like a bad regulator |
| 22:01.57 | CrashSys | or high input voltage |
| 22:01.57 | aptura | in the portrans? |
| 22:02.01 | atca_pres | hi somebody :) |
| 22:02.07 | aptura | standard 120 volts |
| 22:02.07 | CrashSys | dunno if these wall-trans actually have active regulation |
| 22:02.19 | wunderkin | i heard something about bad power supplies before but i dont know the details |
| 22:02.58 | aptura | 119.5 VAC at the wall |
| 22:03.31 | CrashSys | RadioShack! |
| 22:04.20 | atca_pres | I need help/opinion on a weird rtp path I have with my Asterisk box...... anyone ? |
| 22:05.46 | CrashSys | Anyone ever done a callpickup group on a sip? |
| 22:05.55 | CrashSys | show features is listing the #8 as available |
| 22:06.35 | atca_pres | never did sry ... |
| 22:06.38 | *** join/#asterisk a1fa (n=a1fa@unaffiliated/a1fa) |
| 22:07.49 | aptura | Yea radio shack might be the only option. |
| 22:08.18 | aptura | I just hope this power supply did not cook my phone. |
| 22:08.58 | CrashSys | Being a polycom it's prolly got pretty hefty protection circuitry in it... |
| 22:09.00 | *** join/#asterisk map7 (n=map7@teksup41.lnk.telstra.net) |
| 22:10.45 | atca_pres | I need help/opinion on a weird rtp path I have with my Asterisk box...... anyone ? |
| 22:11.35 | *** join/#asterisk netlouis (n=netlouis@2001:b18:400d:0:20b:cdff:fe86:231c) |
| 22:13.36 | *** join/#asterisk skopii (n=skopii@fuckcingularwireless.com) |
| 22:13.47 | *** join/#asterisk oej (n=olle@204.11.231.198.static.etheric.net) |
| 22:13.52 | atca_pres | I need help/opinion on a weird rtp path I have with my Asterisk box...... anyone ? |
| 22:13.57 | skopii | hello, I am trying to get the IDLE default images working on my polycom 500 |
| 22:14.06 | skopii | does anyone know how to turn up the logging for that? |
| 22:14.34 | atca_pres | never worked with polycom phones yet ... |
| 22:14.42 | *** join/#asterisk booray (n=booray@static-66-15-145-196.bdsl.verizon.net) |
| 22:14.44 | skopii | http://www.voip-info.org/wiki/view/Polycom+Idle+Images |
| 22:14.52 | skopii | I have followed those instructions exactly |
| 22:15.03 | skopii | and the phone displays nothing where it previously displayed the time. |
| 22:15.14 | skopii | I am guessing that I created the image wrong |
| 22:15.37 | skopii | foo=`curl http://pbx/cgi-bin/queuestatus.cgi`;convert -size 112x52 xc:white -gravity center -annotate 0 "`echo $foo`" -colors 16 IdleDefault.bmp |
| 22:15.39 | atca_pres | try to use irfanview for your picture |
| 22:15.41 | *** join/#asterisk k84 (n=pyromani@86.84-48-44.nextgentel.com) |
| 22:15.53 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 22:16.22 | k84 | How can i choose which voip service i want to dial via, if got several voip providers. |
| 22:16.49 | atca_pres | K84 : contexts, prefixes ? |
| 22:16.49 | k84 | e.g i dial the number, then i get a voice prompt asking Press 1 to dial via Provider 1, Press 2 to dial via Provider 2, Press 3 ....' |
| 22:17.04 | atca_pres | you need to create a menu then |
| 22:17.17 | k84 | but how do i transfer the number? |
| 22:17.24 | k84 | Macro? |
| 22:17.26 | atca_pres | with ${EXTEN} |
| 22:17.56 | atca_pres | you can use Dial($EXTEN@serviceprovider,,) |
| 22:18.05 | atca_pres | you can use Dial(${EXTEN}@serviceprovider,,) |
| 22:18.36 | k84 | so something like: |
| 22:18.45 | *** join/#asterisk TekAdmin20XX (n=prodroom@c-68-55-105-37.hsd1.md.comcast.net) |
| 22:19.10 | TekAdmin20XX | Hello everyone |
| 22:19.14 | atca_pres | hi there |
| 22:20.21 | TekAdmin20XX | I am curious if the asterisk system would be capable of the following , we are working with people at wikiversity, but I figured I would post in here so people were aware of what was happening as well |
| 22:22.04 | atca_pres | K84 : but waiting for a menu seems a pain. I suggest you look in adding prefixes to filter the isp |
| 22:22.11 | atca_pres | or providers |
| 22:22.40 | atca_pres | I need help/opinion on a weird rtp path I have with my Asterisk box...... anyone ? |
| 22:22.52 | Lann | so if you delete a message it renumbers the remaining messages...wtf why |
| 22:23.22 | atca_pres | message 4 becomes message 3 if you delete msg 3 ? |
| 22:23.29 | Lann | someone said that |
| 22:23.36 | TekAdmin20XX | http://en.wikiversity.org/wiki/User:Drjayztekradio/latest_news#02.5C03.5C2007 |
| 22:23.38 | k84 | atca_pres, thanks, i'll try to look into it. |
| 22:23.40 | atca_pres | I don't think so |
| 22:23.54 | booray | Does anyone know of a way to make a callback queue have a separate ring than that of a normal call? |
| 22:23.55 | TekAdmin20XX | there's the link everyone |
| 22:26.02 | atca_pres | TekAdmin20XX : radio over VoIP ? ...wow that'll need some work |
| 22:26.05 | *** join/#asterisk flenders (n=fserto@unaffiliated/flenders) |
| 22:26.13 | atca_pres | I mean CD over VoIP |
| 22:26.38 | De_Mon | heh, if I do a system(echo -e "createcallfile" > callfiledir) I get an error as asterisk reads the callfile before its done. if I do the same thing from a bash script and call that. it works like a charm. |
| 22:27.26 | atca_pres | does the script try to see if callfiledir exists before the echo ? |
| 22:28.25 | flenders | De_Mon: what I do is run a shell script that creates the callfile and then move it to the outgoing dir... |
| 22:28.54 | *** join/#asterisk rrocha (n=ruyrocha@200-203-209-114.ctame7004.dsl.brasiltelecom.net.br) |
| 22:29.26 | *** join/#asterisk [shodan] (n=shodan@ip137.99-113-216.pppoe4.joliette.intermonde.net) |
| 22:29.49 | flenders | De_Mon: exten => 85,11,System(/root/callback_fs&) |
| 22:30.16 | atca_pres | not familiar enough with bash to help you :( |
| 22:30.32 | *** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com) |
| 22:30.59 | atca_pres | I need help/opinion on a weird rtp path I have with my Asterisk box...... anyone ? |
| 22:31.12 | De_Mon | flenders I'm creating the file from a bash script that echos it to a file. |
| 22:31.28 | De_Mon | even uner load its not screwing up. shrug |
| 22:31.32 | *** join/#asterisk toot (n=toot@84.19.255.123) |
| 22:32.03 | atca_pres | My rtp path : caller to asterisk, asterisk to called, called to caller |
| 22:32.54 | atca_pres | I found out that it's the Tt dial cmd options, but why would it do THAT ? |
| 22:33.05 | atca_pres | half bridging ? |
| 22:33.06 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 22:33.08 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 22:33.56 | wunderkin | atca_pres, if you supply those options then asterisk has to be listening for the dtmf |
| 22:34.05 | atca_pres | not in SIP info |
| 22:34.24 | *** join/#asterisk Carp1 (n=none@cpe-24-92-37-135.nycap.res.rr.com) |
| 22:34.26 | atca_pres | and if it need to listen, why not bridge them BOTH to himself and not only the caller ? |
| 22:34.33 | Corydon-w | Asterisk core doesn't know anything about the SIP protocol |
| 22:34.38 | *** join/#asterisk lirakis (n=lirakis@ool-45775b9b.dyn.optonline.net) |
| 22:34.46 | toot | anyone on from digium or asterisk just now? |
| 22:34.54 | flenders | De_Mon: why are you echoing? just cp |
| 22:34.56 | Corydon-w | ~ask |
| 22:35.10 | jbot | from memory, ask is Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily. See also http://catb.org/~esr/faqs/smart-questions.html |
| 22:35.10 | Carp1 | Does anyone know the default web password for a SIPURA 2100? |
| 22:35.12 | The_DoC^ | is it possible to hook a modem up to a ata and connect to another ata with a modem? |
| 22:35.51 | atca_pres | yes |
| 22:36.02 | toot | we have developed a sort of asterisk ui and i was hoping to show a few folk to get inital comments - too early for public demos just yet :) |
| 22:36.19 | lirakis | can some one do me a favor and try to register a phone/softphone to my pbx?? i think everything is working but.. i have not been able to test to other sip devices outside my network |
| 22:36.38 | lirakis | I just want to see if it rings when i call it, or if my extension rings if you dial me |
| 22:37.13 | Corydon-w | lirakis: take a phone with you home tonight |
| 22:37.23 | lirakis | Corydon-w: im at home |
| 22:37.27 | lirakis | ha ha |
| 22:37.33 | Corydon-w | Take one to work |
| 22:37.38 | lirakis | i have one there too |
| 22:37.46 | Corydon-w | Then what's the problem? |
| 22:37.56 | lirakis | .. but when i dial it.. it does not say "ringing" it just sits there... |
| 22:38.09 | lirakis | but when I dial any extension in my house.. it says "ringing" ... on the cli |
| 22:38.43 | Corydon-w | lirakis: "sip debug". It's very useful to figure out what's happening |
| 22:38.50 | lirakis | hmm... |
| 22:38.58 | The_DoC^ | corrydon-w " I can test |
| 22:39.36 | atca_pres | Corydon-w : Asterisk core doesn't know anything about the SIP protocol-> I was hoping for a second opinion, you already told me that, doesn't explain the half-bridge tho asterisk should try to be in both rtp path, not only one |
| 22:39.43 | lirakis | Corydon-w: wow.. uhh.. thats a lot of info to process |
| 22:39.55 | Corydon-w | lirakis: yep |
| 22:40.16 | lirakis | is it just packet headers? |
| 22:40.17 | toot | well if anyone wants a wee nosey at our asterisk interface (along with a few other features) give me a shout - i'm just at the feedback stage but i welcome any comments |
| 22:40.19 | atca_pres | lirakis I output my ssh session to a log file -> easier to read |
| 22:40.33 | Corydon-w | atca_pres: you were hoping that someone different might tell you what you want to hear? |
| 22:41.46 | atca_pres | corydon-w no just want to be explain. To me it's not normal to have this kind of rtp path with both Tt options. Doesn't makes sense to me |
| 22:41.46 | lirakis | Corydon-w: one of the reasons i wanted to get some one to try and help out is b/c this phone at work... i think needs to get rebooted.. since i just did a total reinstall of my asterisk server.. i mean.. a new centos, and a new asterisk install... |
| 22:41.46 | Corydon-w | atca_pres: it's just as likely that one of your SIP phones is violating spec and trying to cut the gateway out of the RTP path |
| 22:41.55 | flenders | toot: is it gonna be free? |
| 22:41.57 | lirakis | so.. im not sure if it just needs a "clean start" .. |
| 22:42.22 | atca_pres | actually it's the Asterisk 200 ok that doesn't cut the path |
| 22:42.33 | Corydon-w | lirakis: you could just wait for the registration to timeout |
| 22:42.54 | JT | The_DoC^: hooking up modems to atas may only work sometimes, under ideal conditions |
| 22:43.05 | lirakis | Corydon-w: well .. it shows that it is registered with "sip show peers" ... but .. i just dont know.. it seems "funny" |
| 22:43.07 | Corydon-w | atca_pres: it's not supposed to cut the path |
| 22:43.07 | *** part/#asterisk k84 (n=pyromani@86.84-48-44.nextgentel.com) |
| 22:43.33 | Corydon-w | lirakis: then it is registered |
| 22:43.40 | lirakis | Corydon-w: i know that... |
| 22:43.40 | JT | atca_pres: it's misleading to give a "yes" answer to The_DoC^'s question |
| 22:43.59 | The_DoC^ | JT : all I need to do is download a updated firmware to a Settop box3 |
| 22:44.00 | atca_pres | Corydon-w: Well Asterisk try to bridge them together at Invite and at 200 ok bridge the caller to him, but the 200 ok to t he called doesn't |
| 22:44.11 | JT | The_DoC^: over a lan with g.711 it might work |
| 22:44.40 | Corydon-w | atca_pres: you're going to need a 'sip debug' output to explain that |
| 22:44.50 | toot | large parts of it will certainly be open sourced at the appropriate time - but thats not right this minute, or this month |
| 22:44.53 | The_DoC^ | yea, I was just going from one port on my ata to the * box then back to the second port on the ata |
| 22:44.56 | *** part/#asterisk jpablo (n=jpablo@linuxuanl.org) |
| 22:45.01 | atca_pres | I put it in my bug : 8941 |
| 22:45.17 | JT | The_DoC^: what ata? |
| 22:45.23 | The_DoC^ | pap2 |
| 22:45.59 | JT | hrm ok |
| 22:46.08 | The_DoC^ | I am not fond of setting up 2 modems as a null via at commands |
| 22:46.28 | atca_pres | Corydon-w I'll be back in half-hour at most |
| 22:47.11 | Corydon-w | atca_pres: this log is unreadable |
| 22:49.27 | lirakis | Corydon-w: hrmm i guess i have some network issue.. (nat) issue.. because it rings when some one dials me.. but i get no audio. it is wierd though.. because it works when i use my trunk to go out |
| 22:49.45 | atca_pres | Corydon-w Try the ethereal |
| 22:50.31 | atca_pres | Corydon-w I don't understand how it came out that way tho ...sry |
| 22:53.00 | lirakis | should I always set dtmfmode=rfc2833 ??? |
| 22:53.44 | lirakis | i used to use trixbox .. and i feel kind of brainwashed about it... so .. im just curious about the reasoning of setting a certain mode |
| 22:54.07 | Corydon-w | Depends upon what your phone supports |
| 22:55.41 | lirakis | Corydon-w: all my phones support in audio, rfc2833... |
| 22:55.55 | Corydon-w | lirakis: what about SIP INFO? |
| 22:56.12 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 22:56.21 | lirakis | Corydon-w: ? |
| 22:56.40 | Corydon-w | lirakis: SIP INFO is the other type of DTMF signalling |
| 22:56.46 | lirakis | oh |
| 22:57.07 | Corydon-w | lirakis: you set on the Asterisk server what your phone will support |
| 22:57.24 | Corydon-w | If it supports all of them, pick one |
| 22:57.26 | lirakis | okay.. it doesnt matter what the pstn uses or anything like that |
| 22:57.41 | Corydon-w | PSTN uses inband |
| 22:57.45 | voipman | anyone familiar with flash timers in zaptel? |
| 22:57.45 | lirakis | alright.. well i have been using rfc2833 .. so i will just stick with it.. i was just asking. |
| 22:57.49 | atca_pres | Corydon-w is the Ethereal conslusive ? |
| 22:58.03 | Corydon-w | Is it what? |
| 22:58.21 | atca_pres | Corydon-w Do you see the problem in the capture |
| 22:58.32 | Corydon-w | No, I don't |
| 22:58.43 | Carp1 | When parking calls, I include parkedcalls in the context, but when I press #700, i justhear the tones in the phone and the call doesnt park. |
| 22:58.45 | atca_pres | corydon-w : check the rtp |
| 22:58.49 | lirakis | oh .. i think i need sip.nat.conf setup... because my pbx is natted |
| 22:59.27 | atca_pres | corydon-w : with Tt on, one rtp path does't get back to asterisk |
| 22:59.31 | *** join/#asterisk uwe (n=uwe@a46-75.adsl.paltel.net) |
| 22:59.40 | Corydon-w | atca_pres: what type of phone |
| 22:59.57 | atca_pres | corydon-w : Mediatrix 2102 units |
| 23:00.11 | Corydon-w | atca_pres: try a different SIP phone. |
| 23:00.41 | Corydon-w | Mediatrix is not known for being compliant |
| 23:00.44 | voipman | having problems with getting flash's to regocnize on a zap channel using an analog gateway, should i be looking at rxflash timers and trying to change them? |
| 23:00.51 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 23:01.20 | Corydon-w | In fact, Mediatrix is not known for interop with anything other than itself |
| 23:01.42 | atca_pres | lol I never found out a more interop unit ! but out-of-suject |
| 23:02.04 | Corydon-w | Crap doesn't even come close to what I think of Mediatrix units |
| 23:02.08 | atca_pres | corydon-w : no a bad idea except I don't see why the sdp of Asterisk would be treated diffrently |
| 23:02.34 | *** join/#asterisk pronai (n=fortheta@uu-217-199-146-246.uunet.co.ke) |
| 23:02.48 | atca_pres | Asterisk cleary put in the SDP the address of the caller party , and not it's own |
| 23:03.36 | Corydon-w | atca_pres: where do you see that? |
| 23:04.25 | *** join/#asterisk Juggie (n=Juggie@CPE00c049d9f271-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 23:04.25 | atca_pres | in the invite to the called party |
| 23:04.43 | Corydon-w | atca_pres: which packet? |
| 23:04.43 | atca_pres | packet 13 in Notworking.pcap |
| 23:05.20 | Corydon-w | atca_pres: no, it didn't |
| 23:05.29 | Corydon-w | atca_pres: see the Via header? |
| 23:05.51 | atca_pres | via is ok the the address in the sdp isn't |
| 23:06.41 | pronai | anyone in east africa region? |
| 23:08.21 | lirakis | muwahaha! fixed |
| 23:09.12 | pronai | Anyone developing asterisk solutions in east africa? |
| 23:10.09 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 23:10.11 | lirakis | Corydon-w: i had to set externhost because my pbx is natted ;) |
| 23:10.20 | *** join/#asterisk ChicagoBud (n=Bud@adsl-70-228-35-78.dsl.chcgil.ameritech.net) |
| 23:10.28 | *** join/#asterisk k-man__ (n=jason@ppp244-232.static.internode.on.net) |
| 23:10.31 | Corydon-w | atca_pres: yep, looks wrong |
| 23:10.33 | Carp1 | When parking calls, I include parkedcalls in the context, but when I press #700, i justhear the tones in the phone and the call doesnt park. |
| 23:10.54 | Carp1 | Can someone help me with this issue? |
| 23:11.03 | lirakis | Corydon-w: thanks for your.. help/responces... some times its a little defeating when no one says anything when you have a question.. so even if it isnt directly answered.. responces keep me at it.. so thanks! |
| 23:11.09 | atca_pres | Corydon-w: Well glad you see it, makes my day ! :) |
| 23:11.11 | k-man__ | so.... with asterisk running, should I be able to telnet to port 5060 and see something? |
| 23:11.24 | atca_pres | Corydon-w: btw I appreciate you taking the time for this |
| 23:11.32 | oys | it seems that people have had a lot of trouble with varios x100p cards out there. Anyone here able to recomend a clone that works ok? |
| 23:11.40 | Corydon-w | atca_pres: in the future, please don't upload zip files to bugs. |
| 23:11.51 | atca_pres | Corydon-w: understood |
| 23:12.03 | ChicagoBud | Here is a softball for someone. How do I call into voicemail so it knows what extension I am calling from. Right now I go into voicemail main and have to enter the extension I am calling from |
| 23:12.41 | Carp1 | Voicemaiil(100) |
| 23:12.47 | atca_pres | VoicemailMain(${CALLERIDNUM}@Voicecontext) |
| 23:12.49 | Carp1 | or whatever mailbox number |
| 23:12.58 | *** join/#asterisk karmatronic (n=karmatro@84.77.131.137) |
| 23:13.57 | atca_pres | Corydon-w: I already should be on my way. What's going to happen with this issue ? Should I do something, do want anything more ? |
| 23:14.33 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 23:15.12 | k-man__ | do i have to tell asterisk which ip to bind to when it listens on 5060 |
| 23:15.16 | Corydon-w | atca_pres: I reopened the issue, with a different subject |
| 23:15.28 | ChicagoBud | Never mind. I can answer my own question: exten => NNNN,1,VoicemailMain(${CALLERIDNUM}) |
| 23:15.43 | atca_pres | Corydon-w: Thanx a lot |
| 23:16.11 | JT | ChicagoBud: ${CALLERID(num)} is the way to get callerid now |
| 23:16.16 | Carp1 | Does anyone know the default web password for a SIPURA 2100? |
| 23:16.57 | riddlebox | Carp1, I believe it is admin admin isnt it |
| 23:17.55 | Carp1 | no |
| 23:18.03 | Carp1 | I tried it and it says its wrong. |
| 23:18.25 | riddlebox | Carp1, is this one that you bought? |
| 23:18.34 | riddlebox | or did you get it from like broadvoice? |
| 23:19.08 | Carp1 | I got it already programmed |
| 23:19.15 | Carp1 | but I reset it (I think I did anyways) |
| 23:19.57 | riddlebox | Carp1, if it came from like broadvoice or something like that they set it up to go out and get all settings and stuff, which means you would have to call them for the password |
| 23:20.23 | Carp1 | I did a factory reset |
| 23:21.17 | riddlebox | Carp1, I got my sipura 2100 from broadvoice, I had to call them to get the password, even after I did a factory reset, it is provisioned to go on their network |
| 23:21.46 | flenders | guys, is there a way to reduce the amount of time it takes to learn the echo (cancel)? |
| 23:21.56 | Carp1 | really? |
| 23:21.57 | Carp1 | hmmm |
| 23:22.04 | flenders | now it takes a few seconds, and people are finding it annoying |
| 23:23.26 | karmatronic | has anyonw ever made anything with chan_bluetooth? |
| 23:23.36 | *** join/#asterisk nfi|ermes (n=ermsewrk@88-149-168-240.f5.ngi.it) |
| 23:25.00 | *** join/#asterisk connorp (n=connorp@c-67-190-8-41.hsd1.co.comcast.net) |
| 23:25.56 | connorp | hey all, I'm running 1.2.13 and am having quality issues. Upgrade to 1.4.0 first, or ask questions here first? |
| 23:26.18 | perd | quality issues will probably not be solved by upgrading |
| 23:26.28 | perd | it's likely a codec/network/hardware issue |
| 23:26.35 | anonymouz666 | exten => 206,1,GotoIf($["${CALLERIDNUM}" = "303"]?3) |
| 23:26.46 | anonymouz666 | does > and < works on gotoif ? |
| 23:26.58 | Qwell[] | anonymouz666: no |
| 23:27.02 | Qwell[] | but it DOES work in a $[] block |
| 23:27.13 | perd | rimshot |
| 23:27.59 | *** join/#asterisk ctooley (n=ctooley@jc1-111.moment.net) |
| 23:27.59 | anonymouz666 | GotoIf($["${CALLERIDNUM}" > "303"]?3) |
| 23:28.15 | Qwell[] | Those are strings, don't forget |
| 23:28.34 | Qwell[] | So "3021" is *not* higher than "303" |
| 23:28.35 | *** join/#asterisk yassine (n=yassine@dsl.voicint.com) |
| 23:29.30 | anonymouz666 | what's the best approach to see calleridnum and do something different if it is bigger than 303? |
| 23:29.51 | Corydon-w | $[0${CALLERID(num)} > 303] |
| 23:30.05 | connorp | tell me what kind of problem this is, if someone can: If I call my IVR from a PSTN line (my home phone or cell) I can get through to my SIP extension. However, if I dial out on that same sip exension through the same SIP trunk that I dialed into before, no audio is transmitted on either side. |
| 23:30.30 | Carp1 | When parking calls, I include parkedcalls in the context, but when I press #700, i just hear the tones in the phone and the call doesnt park. |
| 23:30.33 | Carp1 | any ideas? |
| 23:31.17 | *** join/#asterisk alrs (n=lars@dsl093-066-021.lax1.dsl.speakeasy.net) |
| 23:31.57 | Qwell[] | Corydon-w: we need a func_S_OR :p |
| 23:32.04 | Corydon-w | Heh |
| 23:32.30 | Corydon-w | Qwell[]: Well, you can get it with a combination of ISNULL() and IF() |
| 23:32.37 | Qwell[] | bah |
| 23:32.37 | Qwell[] | :p |
| 23:32.41 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 23:32.52 | Qwell[] | $[${S_OR(${CALLERID(num)},0} > 303] |
| 23:33.06 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 23:33.07 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 23:33.11 | Corydon-w | I suppose we could write IFNULL |
| 23:33.15 | *** part/#asterisk karmatronic (n=karmatro@84.77.131.137) |
| 23:33.19 | Qwell[] | nah |
| 23:33.36 | Corydon-w | IFNULL is semantically equivalent to S_OR |
| 23:34.03 | Corydon-w | in SQL, though |
| 23:34.27 | anonymouz666 | Corydon-w: i didn't understand the 0? |
| 23:34.48 | Corydon-w | anonymouz666: if it's blank, the 0 keeps the expression valid |
| 23:35.30 | Corydon-w | anonymouz666: same purpose as your quotes, except that you can do numeric comparison |
| 23:35.41 | anonymouz666 | thanks! |
| 23:36.04 | *** join/#asterisk Dr-Linux|home (n=Dreamer@DSL-202-59-73-131.nexlinx.net.pk) |
| 23:36.17 | Corydon-w | Remember that in the expression parser, all numbers are evaluated as decimal, whether or not they're preceded by a 0. |
| 23:36.30 | Corydon-w | There's no octal in the expression parser |
| 23:36.36 | JT | float? |
| 23:36.44 | Corydon-w | No float, either |
| 23:36.50 | JT | int? |
| 23:36.51 | Corydon-w | If you want float, use MATH() |
| 23:37.12 | JT | right, just wondering what specificly you meant by decimal |
| 23:37.19 | riddlebox | is this correct now? $[${CALLERID(num)} |
| 23:37.23 | Corydon-w | base 10 integers |
| 23:37.29 | JT | ok |
| 23:37.44 | Corydon-w | as opposed to hexadecimal, octal, or binary |
| 23:39.13 | JT | dec int, right |
| 23:39.45 | Corydon-w | riddlebox: to do what? |
| 23:40.46 | hal2385612 | hi all - could someone direct me to a page that clearly shows what the main additional features asterisk 1.4 gives over 1.2? |
| 23:40.55 | riddlebox | Corydon-w, to take a number that comes in and send it to another priority, I used to have it setup in 1.2 but in 1.4 it has changed |
| 23:43.52 | anonymouz666 | Corydon-w: I am in doubt how to put that expression in a condition on dialplan |
| 23:46.25 | Dr-Linux|home | my VLAN2 softphone register, but doesn't send audio packets. What could be the reason? |
| 23:47.30 | *** join/#asterisk ruied (n=ruied@bl7-208-85.dsl.telepac.pt) |
| 23:47.48 | anonymouz666 | if I can't use > or < inside gotoif.. only in $[] how can I jump to priorities? |
| 23:47.52 | *** join/#asterisk h0 (i=fakhir@unaffiliated/fakhir) |
| 23:48.16 | russellb | then you use $[] inside the gotoif ? |
| 23:48.49 | anonymouz666 | heh |
| 23:49.41 | anonymouz666 | GotoIF($[0${CALLERID(num)} > 303]) ? |
| 23:49.51 | Qwell[] | anonymouz666: just like Corydon-w said, yes |
| 23:50.28 | *** join/#asterisk coppice (n=chatzill@55.157.17.210.dyn.pacific.net.hk) |
| 23:50.38 | Carp1 | Does anyone want to share theur working extensions.conf with features like MOH and Parking? :) |
| 23:51.16 | anonymouz666 | parking is easy as hell |
| 23:51.16 | anonymouz666 | look the book |
| 23:51.30 | ruied | Hi there! I have a tdm400 already configured and working, my problem is when I make a call to the fxs channel (from the outside) it rings three times before asterisk handles it, for example if I play a 'hellow-worl' it rings from the client side, but when asteerisk pick it up I just hear "world" not the "hello" like if it's playing while it is ringing at the client side... |
| 23:51.48 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
| 23:52.03 | Carp1 | I did |
| 23:52.06 | Carp1 | When parking calls, I include parkedcalls in the context, but when I press #700, i just hear the tones in the phone and the call doesnt park. |
| 23:52.06 | BSDTech | anyone here got spandsp fax working in 1.4 |
| 23:52.25 | BSDTech | or are they just all using t.38 pass threw |
| 23:55.17 | ruied | If I place a call from the internal phone (fxs channel or sip account) asterisk picks the call imediately, but if it is from the outside operator, asterisk picks after two or three rings... |
| 23:58.57 | *** join/#asterisk RoyK (n=roy@217-175-39.100710.adsl.tele2.no) |
