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| 00:16.20 | *** join/#asterisk sx-wks (n=sxpert@navsys.org) |
| 00:21.40 | Chris_H_ | I have a wcfxo card, the modules are loaded, but I cant tell if the option internal_timing=yes is actually working -- apart from the fact that when I put a call on hold, asterisk is not generating outgoing frames |
| 00:21.49 | Chris_H_ | how do I find out if this is working correctly or not? |
| 00:22.06 | *** join/#asterisk AndyCap (n=aoy@pdpc/supporter/sustaining/AndyCap) |
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| 00:37.09 | re-pete | does anyone have anything to say about sellvoip.net ? |
| 00:37.26 | re-pete | I have had a ticket open with them for 2 weeks now |
| 00:37.56 | re-pete | does Jed hang out on this channel? |
| 00:38.57 | re-pete | ~seen Qwell |
| 00:39.04 | jbot | qwell is currently on #asterisk (1d 15h 43m 10s). Has said a total of 36 messages. Is idling for 22h 39m 5s, last said: 'Inverted: wtf?'. |
| 00:40.17 | *** join/#asterisk linagee (n=linagee@unaffiliated/linagee) |
| 00:40.35 | re-pete | aren't we a chatty group tonight... |
| 00:41.50 | Chris_H_ | yes they are :( |
| 00:42.12 | Chris_H_ | althought its 13:42 here, so not quite tonight |
| 00:42.35 | linagee | what if two hosts do an IAX register to the same server and account? is that possible? (i want one asterisk host to pick up some DIDs, and another the rest) |
| 00:42.53 | re-pete | I'm at GMT-5 so it's 20:43 here |
| 00:43.23 | Chris_H_ | linagee I am unsure, I would think it would depend on how the remote / far server is set up would it not |
| 00:43.30 | linagee | Chris_H_: hrm.. |
| 00:44.02 | re-pete | interesting... test it out and let us know :) |
| 00:44.52 | Chris_H_ | linagee I have an internal timing source, when I put a caller on hold, my far end has slience supression, I tried the internal_timing option ,but it does not seem to work, |
| 00:45.07 | Chris_H_ | I am on asterisk-1.2.12.1 |
| 00:45.11 | linagee | Chris_H_: ? |
| 00:45.14 | Chris_H_ | do you know if it was included by then? |
| 00:45.31 | Chris_H_ | so the far end hears bursts of audio |
| 00:45.32 | Chris_H_ | ok |
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| 00:53.57 | jo3sm1th | www.sfgate.com/cgi-bin/article.cgi?file=/c/a/2006/08/20/BUG11KJVGJ1.DTL |
| 00:54.56 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
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| 01:21.33 | L|NUX | linagee : forward them :) |
| 01:22.09 | linagee | L|NUX: that would suck down extra bandwidth, right? |
| 01:22.21 | L|NUX | linagee : not really |
| 01:22.35 | L|NUX | linagee : its totally depends upon the codecs you are using |
| 01:22.49 | linagee | L|NUX: PCM/ULAW |
| 01:23.01 | L|NUX | if you think it wikk suck down extra bandwidth then i believe conneting IAX2 will too ;) |
| 01:23.11 | linagee | (i've tested everything else. no need to further degrade voip quality) |
| 01:23.27 | L|NUX | forward them like this |
| 01:23.49 | L|NUX | exten => did1,1,Dial(IAX2/guest@...../1000/tr) |
| 01:24.15 | L|NUX | exten => did2,1,Dial(IAX2/guest@serverB|1000|tr) |
| 01:24.16 | L|NUX | :) |
| 01:24.41 | linagee | that's very strange.... i am noticing that my 866 number is clearer than my local area code number in 858.... |
| 01:24.57 | linagee | L|NUX: hrm... nice. :-D |
| 01:25.17 | L|NUX | linagee : that might be provider problem not yours :) |
| 01:25.19 | linagee | L|NUX: of course i become a point of failure. heh |
| 01:25.39 | linagee | L|NUX: i will give latency an ear test... |
| 01:25.46 | L|NUX | ok |
| 01:26.20 | linagee | alright. i would guess that it's 100ms over landline... |
| 01:26.59 | linagee | no, latency seems about the same. |
| 01:27.23 | linagee | the only time i've been able to noticebly hear latency being different is with landline versus cellphone. heh |
| 01:28.14 | linagee | yeah. cell phone sounds like 300ms to 500ms if i had to take a guess. |
| 01:28.17 | Chris_H_ | L|NUX do you know anything about internal_timing options |
| 01:29.00 | linagee | interesting.... my voip provider has a ping time of about 75ms. |
| 01:29.21 | L|NUX | Chris_H_ : i am n00b still learning :) |
| 01:29.27 | L|NUX | not a voip engineer |
| 01:29.28 | Chris_H_ | ohh ok :) |
| 01:29.35 | jo3sm1th | did anyone else read that article-> www.sfgate.com/cgi-bin/article.cgi?file=/c/a/2006/08/20/BUG11KJVGJ1.DTL |
| 01:29.44 | L|NUX | technically i am linux tech |
| 01:29.45 | L|NUX | :) |
| 01:29.46 | L|NUX | hehe |
| 01:29.50 | Chris_H_ | well,I am now building from source 1.4.0 at the moment |
| 01:29.53 | L|NUX | learning voip just as hobby |
| 01:29.59 | Chris_H_ | right |
| 01:30.00 | L|NUX | i did it ;) |
| 01:30.10 | Chris_H_ | is it good? |
| 01:30.19 | L|NUX | and running my server with chan_jingle but having issues with gtalk |
| 01:30.19 | L|NUX | :) |
| 01:30.30 | Chris_H_ | yeah I read about that, it looks cool |
| 01:30.49 | linagee | i would say AT&T should be dispatching technicians or sending out confirmation letters, not redirecting lines on the fly. heh |
| 01:30.50 | L|NUX | i did this test on 1.2.x |
| 01:30.55 | L|NUX | and it was working at that time |
| 01:31.05 | linagee | jo3sm1th: er, that was for you |
| 01:31.57 | L|NUX | linagee : US government have also an agreement with AT&T to monitor all Calls :) |
| 01:32.22 | linagee | L|NUX: of course they do. if you think public phones are safe from prying ears, you have to be insane. |
| 01:32.47 | Chris_H_ | any idea when Encrypted SIP will become the norm in Asterisk? |
| 01:32.50 | Chris_H_ | Encrypted RTP I should say |
| 01:32.54 | linagee | L|NUX: i would much rather send my CC numbers down an SSL pipe than over the phones |
| 01:33.24 | L|NUX | SSL is also not safe :) |
| 01:33.24 | L|NUX | every heared about monkey attacks |
| 01:33.24 | L|NUX | ;) |
| 01:33.24 | L|NUX | hehe |
| 01:33.24 | linagee | L|NUX: nothing is safe. SSL is safer. |
| 01:33.34 | Chris_H_ | My Brain is safe :) |
| 01:33.38 | linagee | L|NUX: at least you don't have the govt in the middle. |
| 01:34.24 | *** join/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue) |
| 01:34.31 | EyeCue | new iax client comin out soon ladies :D~ |
| 01:35.23 | L|NUX | EyeCue : might be there are ladies but few |
| 01:35.25 | linagee | L|NUX: possible people stealing your number over the phone: any children in the house with the line picked up, anyone tapping into your line outside your house, any technicians working on the line from an intermediate junction box, any operators working for the phone company, the government, anyone outsourced by those last two, the pizza place itself, anyone hacking into the pizza place's DB, the credit card company, anyone hacking into the CC c |
| 01:35.25 | linagee | ompany, did i miss any? :-D |
| 01:35.32 | EyeCue | s'ok :) |
| 01:36.48 | linagee | L|NUX: with SSL what do you have? Any viruses on your computer, your OS itself, anyone spoofing to be the pizza place's server, the security of SSL itself by being bruteforced, any man in the middle attacks, the pizza place itself, the pizza place's DB being hacked into |
| 01:36.50 | L|NUX | hehe |
| 01:36.58 | linagee | seems like a much shorter list. ;) |
| 01:37.19 | EyeCue | Anyone know if theres a registration timeout bug in the iax client lib ? |
| 01:37.21 | L|NUX | :) |
| 01:37.23 | linagee | oh, i forgot the CC company, anyone hacking the CC company |
| 01:37.32 | EyeCue | IE, it continues to return timeouts even when not registering? |
| 01:37.33 | linagee | you'll always have those. heh |
| 01:38.32 | linagee | L|NUX: also, it's not just the length of the list, but how many of those things are fixable or being improved? |
| 01:40.08 | L|NUX | indeed |
| 01:40.51 | linagee | you can kill all the viruses you know about on your computer. you could use an open source software and examine every line of code if you so desired. you could authenticate the pizza place's server (not 100%), you could contribute to newer SSL replacing technologies/protocols, there are ways to avoid man in the middle, etc etc |
| 01:40.59 | razu | anyone familiar with linux hang problem when unloading qozap module ? |
| 01:41.40 | linagee | L|NUX: further down the line, it's not really things you can do, but the company on the other side has to do |
| 01:42.18 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 01:42.18 | *** mode/#asterisk [+o mog] by ChanServ |
| 01:42.36 | linagee | with a phone system, the list of things you can do is very short. use a cell phone instead of a shared land line. oh wait, what can you do beyond that? not much. |
| 01:43.02 | EyeCue | Are deregister events show in logs with lev 3/4 verbosity ? |
| 01:43.02 | linagee | </fruitstand box> |
| 01:44.12 | linagee | alternatively, drive your ass down there and order it yourself. |
| 01:44.25 | razu | linagee : cell phone isn't secure also ... cause it supports unencrypted connection :) |
| 01:45.18 | linagee | razu: that's true too. cell phones open a whole other can of worms. you have people that might listen in, cell phone technicians that might listen in, cell phone operators that might listen in, and then pass it on to the list above. :-D |
| 01:46.06 | razu | linagee : yes ... thats right :) |
| 01:46.23 | linagee | razu: i probably missed a few, but those would be the common ones. ;) |
| 01:47.19 | linagee | i could even make a security list like that for, "once the CC company gets it's numbers". heh |
| 01:47.26 | linagee | or not |
| 01:47.32 | razu | :) |
| 01:47.35 | *** join/#asterisk ShipHead (i=ShipHead@gateway/tor/x-1340085e421f79ba) |
| 01:48.15 | linagee | razu: you have storage products, storage product technicians, server room technicians, application developers, customer service reps, and the list goes on. :-D |
| 01:48.57 | linagee | razu: it's amazing you can go in and order pizza and not be scammed by hundreds of people in the loop each time! heh |
| 01:50.06 | linagee | does anyone know if it's possible for asterisk to time the latency of the line? |
| 01:50.14 | linagee | play a loud tone, listen for it, and time it? |
| 01:52.53 | benjk | linagee, qualify=yes |
| 01:52.55 | justinu|laptop | it's supposed to be using RTCP to track those stats |
| 01:53.03 | razu | linagee : I agree ... there are no totally secure system anywhere, cause even if we have absolutely no access to whatever system we want to hack ... we always have at least 1 person who gives all the access we need :) |
| 01:53.36 | EyeCue | Is there any difference to a registration that logs "Registered IAX2 '<username' (AUTHENTICATED) at <ip>:<port>" and one that doesnt? |
| 01:53.37 | linagee | razu: exactly. there is not one machine in existance that doesn't need a service person to tinker with it every few years or whatever |
| 01:54.09 | justinu|laptop | i disagree |
| 01:54.16 | linagee | justinu|laptop: name one |
| 01:54.17 | justinu|laptop | yoyager's 1 and 2 are still working |
| 01:54.21 | justinu|laptop | as are plenty of comm satellites |
| 01:54.27 | justinu|laptop | mars rovers |
| 01:54.33 | linagee | justinu|laptop: there are technicians tinkering with it |
| 01:54.44 | linagee | justinu|laptop: sending special service codes that may cause communication to stop |
| 01:54.57 | linagee | justinu|laptop: maybe they want to switch over to a backup battery |
| 01:54.57 | benjk | DS9 |
| 01:54.59 | justinu|laptop | voyagers 1 and 2 are pretty much autonomous now... round trip times are so long they can't work on it in real time |
| 01:55.14 | benjk | DS9 is truly autonomous |
| 01:55.22 | justinu|laptop | ds9 is also not real :P |
| 01:55.28 | benjk | it is |
| 01:55.36 | benjk | NASA won an award for the software |
| 01:55.51 | linagee | any system that is "totally anonymous" cannot be proved that until tens or hundreds of years later |
| 01:56.00 | benjk | the software is called remote agent or something like that |
| 01:57.13 | *** join/#asterisk phileeep (n=philip@203.63.126.9) |
| 01:57.48 | EyeCue | Does anyone know if asterisk caches registrations? |
| 01:57.51 | benjk | there are plenty of examples of goal based lisp systems which are truly autonomous |
| 01:58.06 | justinu|laptop | benjk: that's just for academics |
| 01:58.09 | justinu|laptop | get real |
| 01:58.53 | linagee | wow. budgetone 100 for $30 on ebay.... |
| 01:58.55 | benjk | of course you wouldn't find any FOSS script kiddie that would be able to appreciate Lisp |
| 01:58.55 | linagee | tempting. |
| 01:59.03 | phileeep | using the manager interface, is there a way to remove an agent from the Queue without killing the call they are currently on? |
| 01:59.44 | linagee | is a budgetone really that crappy if you just want basic voip? |
| 01:59.49 | benjk | it is |
| 01:59.53 | linagee | :-/ |
| 01:59.59 | linagee | benjk: i have been using one |
| 02:00.03 | benjk | it is ok for very very occasional calls |
| 02:00.10 | linagee | benjk: hasn't been THAT bad. i only to pcm/ulaw |
| 02:00.15 | benjk | if you use it on a daily basis it will wear out |
| 02:00.19 | benjk | the buttons wear out |
| 02:00.30 | benjk | the cable joints / connectors wear out |
| 02:00.35 | linagee | benjk: what if it's not for a business, but a school |
| 02:00.46 | benjk | the display wears out too |
| 02:00.49 | linagee | benjk: how often do classrooms make phone calls |
| 02:01.06 | benjk | class rooms are the most demanding environments I can think of |
| 02:01.14 | benjk | it will take a bashing every day |
| 02:01.18 | linagee | benjk: there is no poe, there is no passthrough ethernet jack... these things are nice, but bt-100 does work.... hrm... |
| 02:01.36 | benjk | the barbietones are ok as lobby phones (very seldomly used) |
| 02:01.43 | *** join/#asterisk crimson__ (n=ryan@cpe-70-125-148-42.satx.res.rr.com) |
| 02:01.50 | linagee | benjk: they have absolutely NO phones in there right now. heh. i want to put one in my mom's class as a testbed. |
| 02:01.51 | benjk | also for your grandparents (call them once every sunday over the internet) |
| 02:02.07 | benjk | that sort of usage pattern they can survive for quite a while |
| 02:02.17 | tuck3r | what is the proper syntax for rages in permit= |
| 02:02.20 | linagee | benjk: to be honest, i want to put one in there because nobody can ever reach her by cell. heh |
| 02:02.26 | benjk | any more frequent use will wear them out very quickly |
| 02:02.39 | tuck3r | permit=0.0.0.0-0.0.0.1? |
| 02:02.51 | benjk | linagee, you could use the ACT P160 |
| 02:02.57 | linagee | ACT? |
| 02:03.05 | benjk | that's a very simple phone, affordable, and it can take a beating |
| 02:03.29 | linagee | google search doesn't turn up much... |
| 02:03.29 | benjk | buttons are good quality and the cabling joints and connectors won't wear off either |
| 02:03.30 | linagee | ? |
| 02:03.39 | benjk | Taiwanese OEM manufacturer |
| 02:03.56 | benjk | Advanced Century Telecom tralalala or something like that |
| 02:04.31 | linagee | not finding anything... |
| 02:04.41 | linagee | hrm.. i think i will get another grandstream... bah |
| 02:04.49 | linagee | i know it's almost sinful, but geez... $30.... |
| 02:05.38 | benjk | http://www.act-tel.com.tw/_pg/products/productItem.asp |
| 02:05.47 | linagee | hrm... BT-101 seems to have better buttons. and it's only $40 |
| 02:06.03 | linagee | (has rubberized buttons) |
| 02:06.13 | linagee | or is that worse? heh |
| 02:06.17 | benjk | go with the P160 |
| 02:06.19 | EyeCue | http://www.voip-info.org/wiki/view/Asterisk+Wishlist <-- anyone know how upto date that is ? |
| 02:06.47 | linagee | hrm... P160 has a strange look. lol |
| 02:06.52 | benjk | those ACT phones won't win a beauty contest, but they are rock solid |
| 02:07.04 | benjk | workhorses |
| 02:07.12 | benjk | and they can take a beating |
| 02:07.27 | justinu|laptop | doesn't look any worse than the latest avaya crap |
| 02:07.32 | mog | <PROTECTED> |
| 02:08.04 | benjk | those $30 barbietones are probably DOA |
| 02:08.05 | EyeCue | ctrl-f unregister, just wondering if that is still a current wishlist item |
| 02:08.19 | linagee | hrm... it does have room for an extension list card |
| 02:08.23 | mog | ctrl f unregister? |
| 02:08.29 | EyeCue | on the page |
| 02:08.32 | benjk | what would you do if you have just spent a few thousand $ on dead phones? |
| 02:08.38 | EyeCue | to see which entry im talkin about |
| 02:08.48 | EyeCue | i just dont seem to see any activity when tcpdumping udp port 4569 when i dereg |
| 02:08.58 | EyeCue | wonder if its a client lib issue |
| 02:09.00 | benjk | but them up for your discounted reseller price on eBay or somewhere, just to get rid of em |
| 02:09.05 | benjk | sort out the RMA's later |
| 02:09.16 | benjk | that's the modern spirit everywhere now |
| 02:09.39 | benjk | chance is that a large percentage of the customers will not even bother to ask for a return or refund |
| 02:09.49 | linagee | benjk: i can't seem to find any place that sells the P160 |
| 02:09.55 | linagee | ebay, froogle... |
| 02:09.56 | *** join/#asterisk asteriskmonkey (n=phil@bas4-toronto12-1168021952.dsl.bell.ca) |
| 02:09.57 | EyeCue | mog, ? |
| 02:10.07 | benjk | email the guys at ACT and ask them for a reseller near where you live |
| 02:10.19 | benjk | many companies resell these phones under their own brands |
| 02:10.29 | mog | i was wondering what you wanted done |
| 02:10.42 | EyeCue | I didnt want anything done, i wanted to know how current that wishlist was |
| 02:10.48 | benjk | for example, here in Japan those phones go under the name Fujitsu |
| 02:10.51 | asteriskmonkey | hey im having some issues with a fxo based installation.. when i dial out i get a long tone then the message saying all my digits have not been entered, anyone come across this before? |
| 02:10.52 | EyeCue | :) |
| 02:10.53 | mog | to know what though EyeCue |
| 02:11.08 | EyeCue | Section: IAX > # provide an Unregister() command or method |
| 02:11.14 | benjk | and they are placed against Cisco 79xx phones |
| 02:11.19 | benjk | which go under the NEC brand |
| 02:11.24 | benjk | NEC 79xx |
| 02:11.38 | benjk | they sell the P104 for about 400 USD |
| 02:11.46 | justinu|laptop | heh |
| 02:11.59 | benjk | whereas if you order it in TW with the original ACT label, you only pay about 100 USD |
| 02:12.00 | EyeCue | mog, Does asterisk not log unregister, and further, does the iax client lib actually dereg, or is it an empty function call atm? |
| 02:12.18 | mog | i believe asterisk handls an unreg call |
| 02:12.23 | kronic | I'm getting an "unauthenticated" error when attempting a manager connection, I haven't changed anything, username/password or ips |
| 02:12.24 | benjk | but it does compete successfully in Japan against the Cisco 79xx |
| 02:12.34 | EyeCue | mog, so it should log it with -vvvvd ? |
| 02:12.57 | mog | iax2 debug should do it, you should see asterisk respond to said unreg |
| 02:13.21 | benjk | and BTW, the ACT phones have IAX2 firmware |
| 02:13.32 | benjk | if you ask them nicely, they might let you have it |
| 02:13.54 | benjk | I don't know why they don't openly release it, but they do have it |
| 02:14.03 | EyeCue | ah console |
| 02:14.05 | EyeCue | *dribbles* |
| 02:14.55 | EyeCue | hmm, no dereg logged |
| 02:15.00 | asteriskmonkey | mog you know much about tdm/asterisk stuff |
| 02:15.00 | DrkShdw | benjk: which models have iax2 firmware? |
| 02:15.17 | EyeCue | mog, accodingly, do you know if the iax client libs actually execute a dereg when called? |
| 02:16.25 | *** join/#asterisk blebleble (n=ble@d149-67-99-160.col.wideopenwest.com) |
| 02:18.14 | blebleble | ok i think i have the wierdest problem ever, if i dial 1-800-495-5283 from my asterisk 1.2.4 box it works fine, if i dial it from my 1.2.12.1 asterisk machine it just rings doesnt pickup, i've tested this with 4-5 pbx's and they all do the same thing, only the older version of asterisk works. The 800 number is a huge medical insurance company (guessing there not using asterisk) anyone have any ideas? feel free to try it |
| 02:18.38 | *** join/#asterisk Winkie (n=urmom@cpc2-stre3-0-0-cust344.bagu.cable.ntl.com) |
| 02:22.09 | benjk | DrkShdw, I think the firmware is the same for all the phones |
| 02:22.19 | benjk | but I was testing the IAX2 firmware on the P104 |
| 02:22.37 | SplasPood | blebleble: you supplying local ring (r option to Dial() ) |
| 02:22.41 | benjk | they had a few things missing in their implementation |
| 02:22.56 | benjk | basically I coerced them to try to implement IAX2 |
| 02:23.14 | benjk | they thought it would be another big effort like SIP or H323, so they said "No way" |
| 02:23.29 | SplasPood | blebleble: there needed to be a ? at the end... |
| 02:23.41 | benjk | but I said "just let your engineer take a look and I think you'll find IAX is tiny compared to those" |
| 02:23.51 | DrkShdw | benjk: I've been looking for a hardphone with iax2 firmware from the manufacturer (rather than hacked up firmware) so.. this interests me. the phones any good? |
| 02:24.01 | benjk | a week later the called me up and said their engineer had implemented it "over the weekend" |
| 02:24.33 | benjk | the funny thing was that all those parts in the IAX2 spec document which didn't have diagrams were not implemented :) |
| 02:24.41 | blebleble | <SplasPood> not supplying the r option i dont believe, other 800 numbers work fine on my newer versions it just this one number, yet if i dial it from my older version works fine |
| 02:24.52 | benjk | so I made all the diagrams for those sections which didn't have any |
| 02:25.39 | benjk | not sure how much more work they put into this, but it was working nicely |
| 02:26.00 | benjk | the phones are rock solid |
| 02:26.06 | benjk | not beautiful, but rock solid |
| 02:26.06 | blebleble | SplasPood: try to call it let me know if it picks up for you, i've tried tons of different pbx's (only have and tried two carriers and they both do the same thing) its really odd |
| 02:27.00 | benjk | keep in mind that they do not officially distribute the phone with IAX2 firmware |
| 02:27.11 | benjk | you will have to ask them nicely to get it |
| 02:27.16 | DrkShdw | I was just looking at the p104, it's kinda.. ugly :P |
| 02:27.19 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 02:27.33 | benjk | yeah, its no beauty |
| 02:27.41 | SplasPood | works fine on a 1.2.somethin |
| 02:27.48 | benjk | however, in real it looks much better than on the photo |
| 02:27.54 | DrkShdw | however, the page on voip-info.org says: "Protocol: SIP (RFC 3261), IAX. " |
| 02:28.11 | blebleble | SplasPood: whos the carrier |
| 02:28.17 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 02:28.18 | SplasPood | blebleble: however, does not if I supply local ring |
| 02:28.34 | SplasPood | blebleble: a lot of 800 numbers don't actually "answer" the call until you pick a menu option, etc |
| 02:28.36 | blebleble | SplasPood: hmm, where is that setting? |
| 02:28.43 | SplasPood | Dial() its the option 'r' |
| 02:28.48 | *** part/#asterisk Wi_Fi (n=OUT@c-24-127-12-85.hsd1.ca.comcast.net) |
| 02:29.00 | *** join/#asterisk r0d3nt|m (n=RatMan@foster.stonedcoder.org) |
| 02:29.02 | blebleble | is it defined in a certain file to use? |
| 02:29.40 | SplasPood | extensions.conf ... |
| 02:29.43 | SplasPood | in your dialplan.. |
| 02:29.52 | blebleble | k wasnt sure if it was a global thing looking |
| 02:31.28 | blebleble | yah no 'r' |
| 02:34.28 | phileeep | using the manager interface, is there a way to remove an agent from the Queue without killing the call they are currently on? |
| 02:35.03 | *** join/#asterisk kimo_sabe (i=nick@zappa.azrackspace.net) |
| 02:45.42 | EyeCue | whoa, core dump on server stop :D |
| 02:46.02 | razu | anyone have experience with juhnghanns quad port isdn card ? |
| 02:46.54 | razu | junghanns* |
| 02:52.44 | *** part/#asterisk blebleble (n=ble@d149-67-99-160.col.wideopenwest.com) |
| 02:53.17 | asteriskmonkey | what the heck is a scn 3 error mean? |
| 02:53.18 | *** join/#asterisk fafnir (n=notfaf@unaffiliated/fafnir) |
| 02:56.49 | *** join/#asterisk lowlevel (n=Stuart@CPE0050ba71c82f-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 02:57.03 | *** join/#asterisk ThreeAll (n=ThreeAll@206-248-172-251.dsl.teksavvy.com) |
| 02:57.23 | ThreeAll | anybody there |
| 02:57.37 | *** join/#asterisk alexmontoanelli (n=jircii@alexmm.unetvale.com.br) |
| 02:57.50 | ThreeAll | sound is working only one way |
| 02:58.16 | ThreeAll | clients can call each other but voice is going one way |
| 02:58.24 | ThreeAll | any tips |
| 03:00.37 | benjk | razu, what's the trouble? |
| 03:02.09 | razu | benjk : I have somekind of compilation issue ... and now my pbx hangs if I unload qozap module :( |
| 03:02.40 | benjk | what's the error? |
| 03:03.00 | *** join/#asterisk teknoprep (n=tekon@unaffiliated/teknoprep) |
| 03:03.27 | razu | benjk : the warning ... check here -> http://razu.pri.ee/qozap.txt |
| 03:04.03 | razu | benjk : as I understand the warning is related to my computer hanging ... |
| 03:05.47 | benjk | do you have qozap.ko |
| 03:09.13 | razu | yes |
| 03:13.31 | razu | benjk : but if the pbx hangs the screen shows "qozap: no version for "zt_receive" Found: kernel tainted." + some few rows more |
| 03:13.41 | benjk | did you try "make clean; make" |
| 03:13.44 | razu | yes |
| 03:14.04 | benjk | is this your first build of BRIstuff? |
| 03:14.08 | razu | no |
| 03:14.24 | benjk | and this same setup worked before? |
| 03:14.46 | *** join/#asterisk SECGOD (i=SECGOD@c-71-57-36-106.hsd1.il.comcast.net) |
| 03:15.38 | benjk | razu, what does your lsmod say? |
| 03:16.14 | razu | actually I havent got qozap module working for good, but I have bristuffed asterisks working fine |
| 03:16.29 | razu | the problem comes only when I stop zaptel |
| 03:16.32 | benjk | type lsmod |
| 03:16.43 | razu | if I don't it works fine ... |
| 03:16.54 | razu | just a sec ... my case is booting |
| 03:17.18 | benjk | what do you mean "stop zaptel" ? |
| 03:17.59 | razu | when I type /etc/init.d/zaptel stop ... or rmmod qozap ... anything that unloads qozap module |
| 03:19.25 | razu | benjk : lsmod -> http://razu.pri.ee/zap.txt |
| 03:20.28 | benjk | I wouldn't expect zaptel to work after an unload |
| 03:20.42 | benjk | I would regard that as normal behaviour |
| 03:20.48 | *** part/#asterisk hyphen (n=hyphen@71.224.213.97) |
| 03:20.49 | razu | emm |
| 03:21.23 | razu | I don't think that machine complete hang is normal ? or is it ? |
| 03:21.43 | benjk | if you have userland software that expects functionality from the kernel |
| 03:22.03 | benjk | and you remove that functionality from the kernel from right under its a$$ |
| 03:22.17 | benjk | I would be very surprised if it didn't hang |
| 03:22.59 | razu | hmm |
| 03:23.17 | benjk | you should only remove those modules for a shutdown or reinstall |
| 03:23.43 | benjk | first you shut down asterisk, then you can remove qozap, then you can remove zaptel |
| 03:24.01 | benjk | then you can do stuff (like rebuilding etc) |
| 03:24.18 | benjk | then you can load zaptel, then load qozap, then start asterisk again |
| 03:24.25 | razu | so but if I remove qozap like "rmmod qozap" ... then I need to plug the computers power ... couse it's not responding :( |
| 03:24.43 | benjk | did you shut down asterisk before doing that? |
| 03:24.48 | razu | yes |
| 03:25.03 | razu | asterisk isn't working at all right now |
| 03:25.42 | benjk | if asterisk is no longer using qozap (because its not running) then you should be able to rmmod qozap |
| 03:25.50 | benjk | unless there is |
| 03:26.00 | benjk | a) something else trying to use qozap |
| 03:26.02 | benjk | or |
| 03:26.09 | benjk | b) qozap has a bug |
| 03:26.50 | benjk | reboot the machine, load zaptel, then load qozap, DO NOT START ASTERISK, then rmmod qozap |
| 03:26.57 | benjk | see if that hangs too |
| 03:27.01 | razu | the computer hangs ... |
| 03:27.13 | benjk | without ever having started Asterisk? |
| 03:27.18 | razu | yes |
| 03:27.22 | *** join/#asterisk tuck3r (n=tuck3r@unaffiliated/tuck3r) |
| 03:27.33 | benjk | then that's a bug in qozap |
| 03:27.40 | benjk | report it to Junghanns |
| 03:27.51 | razu | ok |
| 03:28.16 | tuck3r | for permit and deny do you have to specify a subnet mask? |
| 03:28.33 | benjk | you can always try with another version of qozap and see if that hangs too on removal |
| 03:28.52 | razu | yes ... I'm working on it right now |
| 03:29.25 | benjk | the key is to try the rmmod WITHOUT STARTING ASTERISK before |
| 03:29.56 | razu | benjk : ok ... thx a lot :) |
| 03:43.48 | DrAk0 | is by any chance a motorola surfr modem compatible with asterisk? |
| 03:43.56 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 03:44.17 | blitzrage | OK... so the syntax of Exec can get REALLY sick if you're doing some weird parsing.... |
| 03:44.27 | blitzrage | and don't want to use ExecIf() |
| 03:46.35 | Juggie | blitzrage, are you watching the colbert report? |
| 03:47.31 | *** join/#asterisk jamesdobrien (n=jamesdob@203-213-5-232.static.tpgi.com.au) |
| 03:53.56 | tuck3r | DrAk0: why would it not be? |
| 03:54.31 | *** join/#asterisk jroysdon (n=jroysdon@c-67-181-65-139.hsd1.ca.comcast.net) |
| 04:02.51 | *** join/#asterisk drcode (n=chatzill@87.69.59.186.cable.012.net.il) |
| 04:02.55 | drcode | hi all |
| 04:03.15 | drcode | when I try to use playback or voicemail , I get warning and no sound |
| 04:03.17 | drcode | any idea? |
| 04:03.49 | tuck3r | tell us the warning...? |
| 04:04.05 | drcode | k |
| 04:04.08 | drcode | let me check again |
| 04:04.41 | drcode | Oct 12 06:11:16 WARNING[26616]: app_playback.c:83 playback_exec: ast_streamfile failed on SIP/test1-1687 for /var/lib/asterisk/moh-native/8kout.mp3 |
| 04:04.56 | drcode | also .gsm file |
| 04:05.14 | drcode | can I get more info? |
| 04:06.06 | DrAk0 | tuck3r, well i read there is not good support for external modem to make it work with asterisk |
| 04:06.16 | tuck3r | uh huh |
| 04:06.26 | *** join/#asterisk ApEtc (i=apetc@ip70-162-197-214.ph.ph.cox.net) |
| 04:06.28 | tuck3r | wtf does the modem have to do with asterisk again? |
| 04:08.34 | razu | drcode : can you show us the config ? |
| 04:08.50 | drcode | sip.conf? |
| 04:09.04 | tuck3r | not sure how he could fuckup a playback command |
| 04:09.05 | drcode | or ete.conf? |
| 04:09.19 | *** join/#asterisk blebleble (n=ble@d149-67-99-160.col.wideopenwest.com) |
| 04:09.24 | razu | extensions.conf ... the Playback line |
| 04:09.54 | drcode | exten => 6000,1,Answer( ) |
| 04:09.57 | drcode | exten => 6000,2,Playback(/var/lib/asterisk/moh-native/8kout.mp3) |
| 04:09.58 | drcode | exten => 6000,3,Hangup( ) |
| 04:09.59 | blebleble | why is it if i have the 'r' option on Dial I can't dial a specific 800 number, 99% of 800 numbers work fine, however this one won't work when the 'r' option is on the dial plan |
| 04:10.35 | tuck3r | drcode: are you trying to play MOH? |
| 04:10.52 | drcode | well , I change thie dir |
| 04:10.58 | tuck3r | yes or no |
| 04:10.59 | drcode | in musiconhold to other dir |
| 04:11.14 | drcode | I just want to use the file |
| 04:11.17 | razu | shouldn't it be ? : exten => 6000,2,Playback(/var/lib/asterisk/moh-native/8kout) |
| 04:11.23 | tuck3r | no |
| 04:11.32 | tuck3r | Playback(8kout) |
| 04:11.33 | blitzrage | drcode: don't use an extension on the filename |
| 04:11.40 | drcode | also dosnt work |
| 04:11.45 | tuck3r | just put the file in /var/lib/asterisk/sounds |
| 04:11.55 | drcode | k |
| 04:11.58 | drcode | let me check |
| 04:12.01 | blitzrage | drcode: you can use the full path -- you can't use an mp3 I don't think |
| 04:12.19 | blitzrage | drcode: mp3 is not a voice codec that asterisk is going to be able to transcode |
| 04:12.29 | tuck3r | if he has format_mp3 it should work |
| 04:12.34 | DrAk0 | tuck3r, well i wanted to use it as a FXO |
| 04:12.55 | blitzrage | if |
| 04:12.58 | drcode | k |
| 04:13.02 | drcode | and gsm file? |
| 04:13.08 | blitzrage | just use a wav file |
| 04:13.09 | blitzrage | its easier |
| 04:13.11 | tuck3r | DrAk0: a modem as a fxo...? |
| 04:13.21 | kronic | I'm having some issues with ast realtime (using odbc to specifiy the location of the queue_members in a mysql db) |
| 04:13.26 | blitzrage | DrAk0: depends on the modem |
| 04:13.42 | tuck3r | almost no modem can be used as a fxo |
| 04:13.43 | kronic | if I had a agent to the db, and show queue, it returns no member |
| 04:13.45 | drcode | k |
| 04:13.54 | drcode | I can convert mp3 to wav? |
| 04:13.59 | kronic | yes |
| 04:14.12 | file | blitzrage: what's a Canadian place where I can get some t-shirts made? eh? EH? |
| 04:14.31 | blitzrage | file: no idea... I looked and couldn't find one |
| 04:14.35 | blitzrage | file: if you find a place, please let me know |
| 04:14.41 | file | I can't find one either. |
| 04:15.06 | blitzrage | Juggie: no, I'm working |
| 04:15.10 | jamesdobrien | I'm wondering if anyone has tried the new Intel 5000 series chipsets and if they have had any issues with them. Client is looking at purchasing a HP PowerEdge with one of these chipsets. |
| 04:15.18 | drcode | with lame? |
| 04:15.25 | DrAk0 | blitzrage, a external modem has no chance right? |
| 04:15.29 | kronic | perhaps with mpg321? |
| 04:15.39 | blitzrage | DrAk0: depends on the chipset -- and probably not -- you can't just use "anything" |
| 04:16.02 | DrAk0 | blitzrage, is a motorola surfr 56k v90 external |
| 04:16.07 | drcode | k |
| 04:16.11 | blitzrage | DrAk0: it sure is |
| 04:16.17 | file | modem does not an FXO device make |
| 04:16.31 | DrAk0 | blitzrage, it sure is what? |
| 04:16.41 | tuck3r | file: yoda? |
| 04:16.48 | file | no, I'm file |
| 04:16.49 | blitzrage | DrAk0: it sure is a motorola surft 56k v90 external |
| 04:16.49 | file | nice to meet you |
| 04:16.56 | tuck3r | hah |
| 04:17.04 | DrAk0 | blitzrage, it is |
| 04:17.08 | file | people also call me by my real name, but not many people |
| 04:17.09 | blitzrage | DrAk0: yep, it sure is |
| 04:17.13 | Juggie | JOSH! |
| 04:17.17 | DrAk0 | blitzrage, wtf? |
| 04:17.20 | blitzrage | Mr. Joshua |
| 04:17.24 | file | :D |
| 04:17.33 | blitzrage | DrAk0: you're tell me what kind of modem you have, and I'm telling you it sure is that type of modem |
| 04:17.53 | blitzrage | DrAk0: if it isn't obvious, I'm trying to tell you I have no idea what kind of chipset your modem has |
| 04:17.58 | DrAk0 | blitzrage, im not asking what modem i have :P im asking if it may be used as fxo |
| 04:18.05 | blitzrage | see above |
| 04:18.06 | DrAk0 | blitzrage, riiiight |
| 04:18.09 | DrAk0 | gotcha now |
| 04:18.21 | file | magic eight ball says... "doubt it" |
| 04:18.27 | DrAk0 | i think ill purchase a X100P |
| 04:18.28 | blitzrage | agreed |
| 04:18.44 | blitzrage | just get an analog to SIP adapter |
| 04:19.01 | DrAk0 | is cheaper? |
| 04:19.40 | tuck3r | no but it works, X100P (more like the fake ones) are notorious for sucking |
| 04:19.41 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 04:20.22 | DrAk0 | you mean somthing like a SPA-3000 ? |
| 04:20.28 | tuck3r | i think he did |
| 04:20.39 | blitzrage | yes |
| 04:20.56 | blitzrage | but I don't use analog at all, so I have no recommendations |
| 04:21.14 | blitzrage | hrmmm... to start on this new feature or not... |
| 04:21.27 | tuck3r | http://pastebin.ca/198736 <-- any idea why 65.39.204.72 is getting let through |
| 04:21.48 | drcode | Oct 12 06:28:17 WARNING[26616]: app_playback.c:83 playback_exec: ast_streamfile failed on SIP/test1-475e for /var/lib/asterisk/sounds/test.wav |
| 04:21.56 | drcode | same warnning |
| 04:22.00 | drcode | also in wav file |
| 04:22.50 | DrAk0 | well i was thinking on this one |
| 04:22.55 | DrAk0 | http://www.digitnetworks.com/X100P_FXO_PCI_Card_p/x100p.htm |
| 04:23.22 | tuck3r | ooh, fake card are fun |
| 04:23.43 | benjk | there are no fake cards |
| 04:23.46 | benjk | they are all the same |
| 04:24.07 | benjk | when Digium still sold them, they got them in bulk from the same manufacturers |
| 04:24.13 | rob0 | Hit or miss, and deal with the problems if it does happen to work. Mine crashes regularly. I never know when my FXO is working or not. |
| 04:24.23 | benjk | however, the chips are not longer manufactured these days |
| 04:24.40 | tuck3r | benjk: with a cap pulled off... |
| 04:24.41 | benjk | so the "new" ones you get now, are made with refurbed or even reject chips |
| 04:24.48 | blitzrage | fuck it, I'm goin' to bed... will work on queues() tomorrow |
| 04:25.20 | benjk | those cards are stricly only useful as zaptel timing devices |
| 04:25.29 | benjk | they are not useful for doing any voice stuff |
| 04:25.54 | tuck3r | i think ztdummy works fine with 2.6 now... |
| 04:26.13 | DrAk0 | is for a home pbx |
| 04:26.15 | DrAk0 | home use |
| 04:26.16 | DrAk0 | * |
| 04:26.25 | benjk | even for home use those cards are no good |
| 04:26.33 | rob0 | Even so, it might turn out to be unreliable. |
| 04:26.33 | benjk | get a Sipura 3000 |
| 04:27.09 | rob0 | I'm going to buy a TDM FXO module. (Already have a TDM card with one FXS.) |
| 04:27.15 | tuck3r | i have to agree with everybody else, get a ATA |
| 04:28.25 | *** join/#asterisk blue`sky (n=crazy_gu@ppp-70-225-176-106.dsl.chmpil.ameritech.net) |
| 04:28.28 | blue`sky | hi |
| 04:28.34 | benjk | not an ATA, an FXO gateway |
| 04:28.40 | blue`sky | anyone know about jingle ? |
| 04:28.44 | benjk | and ATA is an FXS gateway |
| 04:29.07 | benjk | Sipura 3000 is both an FXO and FXS gateway, has 1xFXO and 1xFXS port |
| 04:29.29 | tuck3r | so ATA covers both IMO |
| 04:29.31 | *** join/#asterisk Defraz (n=t0tal@fw.centrisys.com) |
| 04:29.36 | benjk | no they don't |
| 04:30.07 | benjk | ATA is an acronym for Analog Telephone Adapter |
| 04:30.07 | tuck3r | ALA then? |
| 04:30.07 | benjk | as in adapter to connect an analog telephone |
| 04:30.16 | tuck3r | or "analog telephone" line |
| 04:30.24 | benjk | you might want to call it ALA, but I don't think anybody will recognise it |
| 04:30.39 | tuck3r | thats why I called it a ATA |
| 04:30.39 | benjk | ATA is universally considered to be FXS only |
| 04:30.57 | benjk | you may want to call your PBX a shoebox if you like |
| 04:31.07 | benjk | but you cannot claim that this is the proper term |
| 04:31.38 | DrAk0 | sipura 3000 is x3 times expensive , well ill think about it |
| 04:31.47 | tuck3r | save up |
| 04:31.50 | drcode | k |
| 04:31.52 | drcode | now it works |
| 04:31.54 | drcode | great |
| 04:32.02 | drcode | I can use same file in voicemail? |
| 04:32.16 | benjk | that's like saying a bycicle is 100 times more expensive than scrap metal |
| 04:32.33 | benjk | you still can't use the scrap metal to drive to work |
| 04:35.37 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 04:35.44 | DrAk0 | us$243 |
| 04:35.47 | DrAk0 | my solution |
| 04:35.48 | DrAk0 | hhmm |
| 04:36.16 | tuck3r | wtf? thats overpriced |
| 04:36.57 | DrAk0 | tuck3r, thats one SPA-3000 and 2 SPA-2001 |
| 04:37.05 | tuck3r | oh... |
| 04:37.24 | DrAk0 | i need at least 4 POTS |
| 04:37.46 | tuck3r | and how were you going to do that with X100Ps? |
| 04:38.09 | DrAk0 | i was saving us$70 |
| 04:38.30 | tuck3r | no, it would never worked anyways, one per box |
| 04:38.39 | DrAk0 | the X100P cost us$19 the SPA-3000 cost us$99 |
| 04:38.44 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 04:39.01 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:39.08 | rob0 | $19 is high for those, but you're right, stay away. |
| 04:39.14 | DrAk0 | anyway i think im going for that setup SPA-3000 + x2 SPA-2002 |
| 04:48.30 | jamesdobrien | Anyone installed asterisk on a new server using Intel 5000V/P/Z chipset with digium digital cards? If not, is there a channel where I might ask this question? |
| 04:49.21 | tuck3r | why wold it not work? |
| 04:51.10 | jamesdobrien | The digium hardware compatibility guide says to avoid Intel 915, E7221, and E7525 chipsets as they are known to be partly incompatible. http://www.digium.com/en/docs/misc/compatibility_notes.php |
| 04:52.44 | jamesdobrien | We had a client purchase a server and we got intermittent call drops with a TE411P. The only thing we could put it down to after much investigation is the E7525 chipset that server used. They replaced the server with a chipset that isn't listed on the guide and there have been no more dropped calls. |
| 04:58.23 | *** join/#asterisk linlin (i=linlin@c-67-173-49-55.hsd1.il.comcast.net) |
| 05:03.11 | jamesdobrien | From my understanding the E7525 is quite a recent Intel chipset. The 5000, however, is newer still and that "guide" may not have been updated since then. |
| 05:05.08 | *** join/#asterisk angom_h (n=angomg@red-corp-200.76.229.73.telnor.net) |
| 05:17.13 | stephane_ | jour |
| 05:32.05 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 05:32.40 | *** join/#asterisk kgpsathish (n=sathish@59.92.180.118) |
| 05:34.33 | kgpsathish | I installed UnixODBC and freetds with the asterisk,after that when I reload asterisk Iam getting 'Loading res_dodbc.so failed ' message.any ideas? |
| 05:35.15 | *** join/#asterisk `Tingles` (n=tingles@S01060011d8ecb1d0.cg.shawcable.net) |
| 05:47.10 | *** join/#asterisk topping (n=topping@207.47.6.182.static.nextweb.net) |
| 05:50.01 | *** join/#asterisk juice (n=juice@mo-76-0-45-81.dhcp.embarqhsd.net) |
| 05:56.43 | *** join/#asterisk lorinc (n=ang@caracas-4756.adsl.interware.hu) |
| 05:57.23 | topping | is 'modprobe wctdm24xxp BOOST_RINGER=1' the right way to get higher ring voltage without recompiling? |
| 05:57.37 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
| 06:02.04 | *** join/#asterisk VibroMax (n=phisto@83.229.70.154) |
| 06:02.26 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 06:02.26 | *** mode/#asterisk [+o mog] by ChanServ |
| 06:06.03 | *** join/#asterisk Aurs (n=Aurs@host-81-191-123-189.bluecom.no) |
| 06:06.21 | topping | ohhh that's lovely |
| 06:06.37 | topping | i have a real bell working off of asterisk now hehe |
| 06:07.23 | mog | woohoo |
| 06:07.58 | topping | found an old western electric set at a flea market a couple of months ago |
| 06:12.17 | *** join/#asterisk tengulre (n=tengulre@61.185.224.66) |
| 06:12.25 | FuriousGeorge | hey all |
| 06:13.53 | tengulre | hey all !! |
| 06:14.49 | EyeCue | iaxtel down?! |
| 06:14.50 | EyeCue | :~( |
| 06:14.54 | mog | eep |
| 06:23.56 | [hC] | topping: howd you connect it? you mean like, a rotary dial? |
| 06:24.34 | topping | [hC]: yes |
| 06:24.46 | topping | just plugged it in! |
| 06:24.49 | [hC] | topping: how do you have asterisk interpret pulse dialing? pulse to dtmf converter in the middle? |
| 06:24.54 | [hC] | or.. you cant dial out :) |
| 06:24.58 | topping | hmm, lemme check if it can dial out |
| 06:25.11 | [hC] | heh. you cant :) |
| 06:25.33 | topping | neh, it works! |
| 06:25.34 | topping | hehe |
| 06:25.43 | [hC] | theres no way!!! |
| 06:25.49 | topping | but there is! |
| 06:25.52 | topping | :) |
| 06:25.53 | [hC] | whats it plugged in to? |
| 06:26.03 | topping | tdm2400 |
| 06:26.10 | [hC] | ohh |
| 06:26.12 | [hC] | haha |
| 06:26.15 | [hC] | that would do it. |
| 06:26.17 | topping | is that smarter? |
| 06:26.36 | [hC] | I thought you were using a standard ATA |
| 06:26.40 | topping | ah |
| 06:26.51 | [hC] | i guess tdm2400's fxs ports understand pulse dialing |
| 06:26.53 | blue`sky | :D |
| 06:28.49 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 06:28.58 | EyeCue | In order to get iaxtel connected, but with an IAX2 client that doesnt support providing a context, how would i modify my extensions.conf to be able to dialout to iaxtel users? |
| 06:29.50 | EyeCue | I've got my register => line in iax.conf > [general] |
| 06:36.31 | *** join/#asterisk af_ (n=af@ip-171-49.sn1.eutelia.it) |
| 06:37.19 | *** join/#asterisk jmls (n=asterisk@host81-159-167-80.range81-159.btcentralplus.com) |
| 06:37.20 | *** join/#asterisk oej_ (n=oej@apollo.webway.se) |
| 06:37.25 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 06:53.21 | topping | if I have ARA set up, is there an app that I can use to statically dial one of them in a dialplan? |
| 06:54.00 | topping | do I just use Dial? |
| 06:55.02 | topping | like if the operator is at extension 110 would I use 'exten => 0,1,Dial(110)'> |
| 06:55.03 | topping | ? |
| 06:57.10 | topping | is that what the Local channel is for? |
| 06:57.14 | *** part/#asterisk kgpsathish (n=sathish@59.92.180.118) |
| 06:57.31 | topping | ah yes ok |
| 06:58.59 | kronic | is dynamic realtime for queue members supported in 1.2.12? |
| 06:59.25 | kronic | I've been told, it apparently doesn't |
| 07:00.02 | *** join/#asterisk ltd (n=z@202-161-26-249.dyn.iinet.net.au) |
| 07:04.36 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 07:06.23 | *** join/#asterisk dir (n=dir@124.106.223.190) |
| 07:09.04 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 07:09.43 | grEvenX | does anyone know if asterisk has impemented the feature to answer a SIP ping? |
| 07:22.44 | kaldemar | a SIP ping? isn't that usually done with the OPTIONS message? if you meant that, i'd say that Asterisk will answer. |
| 07:23.30 | oej_ | grEvenX: We do not support the PING method, as kaldemar said, we do support OPTIONS |
| 07:28.05 | *** join/#asterisk Mugatu_ (n=mugatu@unaffiliated/Mugatu) |
| 07:30.04 | *** join/#asterisk alexdepa (n=alexdepa@host165-124-static.28-87-b.business.telecomitalia.it) |
| 07:30.09 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 07:30.18 | alexdepa | hi all!! |
| 07:31.22 | alexdepa | someone else can help me to enable pickup function on my asterisk box? |
| 07:33.44 | alexdepa | I have add "exten => _7.,1,Pickup(${EXTEN:1})" in from-internal context |
| 07:33.55 | *** join/#asterisk CrazyTux (n=CrazyTux@adsl-75-42-28-214.dsl.hstntx.sbcglobal.net) |
| 07:35.01 | alexdepa | but when i try the feature asterisk tell me "no application pickup for extension..." |
| 07:35.07 | alexdepa | why? |
| 07:36.08 | alexdepa | someone can help me? |
| 07:50.11 | *** part/#asterisk jamesdobrien (n=jamesdob@203-213-5-232.static.tpgi.com.au) |
| 07:53.27 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 07:53.54 | *** join/#asterisk lvp (n=lpressl@interner.SerNet.DE) |
| 07:55.47 | *** join/#asterisk dwery (n=dwery@nslu2-linux/dwery) |
| 07:58.03 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 07:58.05 | dwery | hello! I would like to setup a dual PRI card to do simple forwarding between ports (NT-TE). Which card/driver would work better for that on a 2.6.17 kernel? |
| 07:58.52 | *** join/#asterisk DrAk0 (n=ljd@unaffiliated/luisjose) |
| 08:02.18 | lvp | the question is: which ISDN implementation (BRIstuff, visdn, mISDN) is working with 1.4 anyway - or going to be usefull in a reasonable time frame? |
| 08:02.41 | lvp | All projects seem to be stalled at the moment :( |
| 08:02.58 | dwery | mmmm so basically we have no working isdn stack? :) |
| 08:04.26 | benjk | Unicall is due for an ISDN plugin release some time soon |
| 08:05.03 | lvp | My hope was on visdn some month ago - but now nothing (visible) is happening and it is said that the main author said he would be restructuring major parts of the implementation |
| 08:05.08 | lvp | Unicall? |
| 08:05.11 | benjk | though I have no idea if it will work with 1.4 |
| 08:05.19 | benjk | Steve Underwood's Unicall, yes |
| 08:05.32 | dwery | lvp: I tried vISDN days ago because I needed ppp and found the ppp plugin was unusable. |
| 08:05.33 | benjk | same guy who does SpanDSP |
| 08:06.17 | benjk | www.soft-switch.org |
| 08:07.06 | dwery | the strange thing about isdn and linux is that the time goes on, the stack changes, but there's nothing stable available :( |
| 08:07.50 | dwery | ..but a lot of companies are selling asterisk based isdn soft pbxs... |
| 08:08.05 | dwery | si I might be missing something ;) |
| 08:11.54 | *** join/#asterisk LoneShadow (n=duh@c-67-188-235-220.hsd1.ca.comcast.net) |
| 08:12.16 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:14.57 | alexdepa | I would like to setup pickup features, i tried follow voip-info.org instuctions but doesn't work for me |
| 08:16.23 | alexdepa | someone has setup this feature succesfully? |
| 08:16.54 | *** join/#asterisk dir (n=dir@124.106.223.190) |
| 08:20.44 | *** join/#asterisk kiddy (n=kiddy@59.93.12.236) |
| 08:20.49 | kiddy | Hi |
| 08:21.22 | kiddy | how a user can record both incoming and outgoing calls when necessary ? |
| 08:38.37 | LoneShadow | how is 1.4 beta so far ? |
| 08:39.01 | jmls | LoneShadow: cool. working. try it :) |
| 08:39.18 | LoneShadow | will do it over the weekend |
| 08:39.39 | LoneShadow | need to figure out what all files to backup before installing this |
| 08:39.47 | LoneShadow | probably /etc/asterisk should be sufficient |
| 08:39.53 | jmls | LoneShadow: /etc/asterisk for your configs |
| 08:40.56 | LoneShadow | 1.4 is supposed to support google talk right ? |
| 08:41.33 | *** join/#asterisk Mw3_ (i=mw3@national.t-error.hu) |
| 08:41.49 | *** join/#asterisk enqubis (n=jbrewer@vpn-132.fresno-dc1.brandxnet.com) |
| 08:42.21 | jmls | LoneShadow: um, sort of, yeah. Mog is the person to ask. I beleive there are still some issues which are being worked on |
| 08:43.16 | LoneShadow | according to the forums, looks like someone hit some problems with 1.4 and jabber |
| 08:43.25 | enqubis | ive got an odd iax problem can somebody help? |
| 08:44.36 | LoneShadow | just ask the question, if its non related to freepbx/trixbox, someone will help you out here |
| 08:44.53 | enqubis | k |
| 08:45.10 | enqubis | ive got two asterisk servers both running 1.2.4 |
| 08:45.29 | enqubis | i can register one way, but not the other |
| 08:45.39 | LoneShadow | are you creating IAX trunks ? |
| 08:45.51 | EyeCue | uh, question, the source port for the client connecting to the asterisk server doesnt have to be 4569 as well does it ? |
| 08:45.57 | enqubis | trunk=yes? |
| 08:46.38 | enqubis | eyecue, no it doesnt |
| 08:46.39 | LoneShadow | both asterisk boxes are running zaptel/ztdummy(2.6 kernel) |
| 08:46.44 | EyeCue | hmm |
| 08:47.00 | enqubis | one has zaptel hardware, one is running ztdummy |
| 08:47.33 | LoneShadow | you need ztdummy atleast on each side for the trunking to work |
| 08:47.50 | enqubis | question, the ztdummy doesnt need configuration, does it? |
| 08:47.53 | LoneShadow | one of my box is a linux machine with ztdummy, and the other is a router running openwrt |
| 08:47.53 | enqubis | just the module loaded? |
| 08:47.53 | jmls | LoneShadow: that would be me :) I did have some problems with jabber, but they are all sorted now. We send in excess of 100000 messages per day from * to our IM |
| 08:47.59 | LoneShadow | so trunking works only half way |
| 08:48.56 | LoneShadow | enqubis: no need of config |
| 08:49.21 | LoneShadow | enqubis: when you do iax2 show peers or registry |
| 08:49.28 | LoneShadow | do you see (t) ? |
| 08:49.53 | LoneShadow | jmls: can we have a voice/sip conversation from * to gtalk ? |
| 08:50.19 | enqubis | under port yes |
| 08:50.21 | enqubis | (T) |
| 08:50.42 | jmls | LoneShadow: I have not tried, but I believe that this is the goal for 1.4 |
| 08:51.09 | LoneShadow | jmls: so how do you send messages ? |
| 08:51.28 | jmls | LoneShadow: using the JabberSend dialplan application |
| 08:51.29 | enqubis | when i iax2 debug i keep seeing subclass: INVAL |
| 08:51.37 | LoneShadow | enqubis: were you able to make things work without trunking ? |
| 08:51.54 | enqubis | ill try setting trunk=no |
| 08:52.24 | LoneShadow | hmm, I was trying to see if it was a nat issue :D |
| 08:52.31 | enqubis | well |
| 08:52.32 | enqubis | it could be |
| 08:52.54 | enqubis | because the machine without zaptel hardware is behind a nat |
| 08:53.23 | enqubis | but i can register to the machine with zt hardware and make calls just fine |
| 08:53.32 | LoneShadow | nat=yes |
| 08:53.36 | LoneShadow | canreinvite=no |
| 08:53.44 | LoneShadow | I have these 2 on my trunk settings |
| 08:53.51 | LoneShadow | mine is a half trunk |
| 08:53.54 | LoneShadow | so config is wierd |
| 08:55.37 | enqubis | still inval's |
| 08:56.00 | enqubis | Tx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL |
| 08:56.06 | enqubis | sorry, i know, pastebin |
| 08:56.34 | LoneShadow | well, I have to sleep soon, its 2am here |
| 08:56.57 | LoneShadow | wont be able to help much anyway, dont have much experience with those error messages |
| 08:57.14 | enqubis | thanks anyway |
| 08:58.12 | LoneShadow | jmls: how do you type the messages ? |
| 08:58.20 | LoneShadow | from your phone ? |
| 08:59.13 | *** join/#asterisk tparcina (n=tparcina@lns02-0286.dsl.iskon.hr) |
| 09:00.05 | tparcina | hi channel! |
| 09:00.15 | tparcina | does anybody use beronet ISDN card? |
| 09:00.16 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 09:00.17 | LoneShadow | oh predefined messages |
| 09:08.09 | kiddy | <PROTECTED> |
| 09:08.34 | kiddy | any additional packages needed for the configuration ? |
| 09:11.21 | *** join/#asterisk jgoo (n=e4b80e21@87.202.222.15) |
| 09:12.05 | jmls | LoneShadow: we do it two ways: * telling a IM monitor of various actions in the dialplan |
| 09:12.11 | *** join/#asterisk HarryR (n=Harry@host-83-146-53-46.bulldogdsl.com) |
| 09:12.17 | *** join/#asterisk razu (n=rasmus@tln-kontor.norby.ee) |
| 09:12.39 | jmls | LoneShadow: second, we use our application to originate a call with the receipient and message as variables for the call |
| 09:12.47 | jgoo | so, I found my charger for my ipaq and axim 30h... so I am thinking of wiring them to docks next to my couch, and using them as universal remotes, and also SIP clients from my * box via wifi... that should pwn - best windows mobile SIP client? :s |
| 09:15.42 | tparcina | beronet ISDN, does anybody use them? |
| 09:15.57 | *** join/#asterisk Greek-Boy (n=grb@193.220.93.162) |
| 09:18.17 | hank | an nt1 usually has two rj45 plugs right? |
| 09:19.35 | hank | one for a card in nt mode and one that provides the s0 bus correct? |
| 09:20.20 | tparcina | hank: i realy don't know |
| 09:20.30 | tparcina | hank: whick card do you use? |
| 09:21.02 | hank | tparcina: then dont answer ;) its a longshine card with a hfc chipset |
| 09:21.13 | kiddy | LoneShadow : Do you know how to configure on demand call recording ? |
| 09:21.42 | kiddy | tparcina : Do you know how to configure on demand call recording ? |
| 09:26.31 | *** join/#asterisk olivier__ (n=olivier@obs92-4-82-239-116-113.fbx.proxad.net) |
| 09:37.01 | *** join/#asterisk ghenry (n=ghenry@82-69-192-46.dsl.in-addr.zen.co.uk) |
| 09:53.48 | *** join/#asterisk RoyK (n=roy@213.160.242.90) |
| 09:57.34 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 10:02.54 | *** join/#asterisk jaike (n=jaike@125.5.144.90) |
| 10:04.06 | jaike | anyone experienced loads of 400+ simultaneous calls in a server going to a queue? |
| 10:04.29 | jaike | wondering if asterisk is ready for this kind of load |
| 10:05.06 | qdk | jaike: * has a hard time reaching 400+ simul. calls, so i guess not. |
| 10:05.56 | benjk | can you spell ... d e a d l o c k ... ? |
| 10:05.59 | jaike | hmmm, am wondering how to loadbalance something like that, a client forwarding calls to a DID |
| 10:06.17 | jaike | that DID receiving the calls going to a queue |
| 10:06.42 | benjk | you could try Bayonne |
| 10:06.56 | benjk | that generally scales much better than Asterisk |
| 10:07.37 | benjk | also FreeSwitch has just recently reached 3000+ concurrent calls, although they may not have implemented any queues yet, need to ask the Freeswitch guys (#freeswitch) |
| 10:07.52 | jaike | thank benjk |
| 10:07.58 | jaike | will look those up |
| 10:08.15 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 10:08.26 | jaike | its a tv marketing client, where they get a rush of calls all at one time |
| 10:08.27 | benjk | just ask the freeswitch guys in the #freeswitch channel |
| 10:08.47 | benjk | Bayonne is part of the GNU Telephony Project www.gnutelephony.org |
| 10:08.56 | benjk | talk to David Sugar |
| 10:10.29 | jaike | benjk: even for pure VOIP setup? we wont be using any cards |
| 10:10.48 | benjk | its not a problem of the interface cards or drivers |
| 10:10.57 | benjk | its a threading and locking problem |
| 10:11.14 | benjk | asterisk only knows one single data structure: single linked list |
| 10:11.33 | benjk | consequently all lookups are linear searches |
| 10:11.51 | benjk | and often an entire list is locked by a thread |
| 10:12.11 | benjk | this causes all other threads to wait |
| 10:12.23 | benjk | there are also deadlock issues |
| 10:12.42 | benjk | its in the core, not the drivers/interfaces |
| 10:19.01 | *** join/#asterisk oQPa (i=Ftv@237.Red-83-44-33.dynamicIP.rima-tde.net) |
| 10:23.02 | *** join/#asterisk soylentgreen (n=fgast@wlan-00.mcbone.net) |
| 10:28.25 | dalbaech | so what's the maximum number of calls that have been made on a single asterisk server? (Has anyone made a record? |
| 10:28.56 | dalbaech | (It would be an interesting thing to know, that's why I ask) |
| 10:32.45 | jaike | same here |
| 10:35.25 | benjk | at 200+ it usually starts getting bad |
| 10:35.40 | kaldemar | http://www.voip-info.org//tiki-pagehistory.php?page=Asterisk+dimensioning |
| 10:36.02 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 10:36.13 | benjk | the best scalability right now is probably with Freeswitch (they recently benchmarked 3000+ calls) |
| 10:36.59 | MrChimpy | yep, i've managed about 180 |
| 10:37.11 | MrChimpy | you need to optimise your AGI seriously |
| 10:37.25 | MrChimpy | and make sure there's no transcoding of prompts etc |
| 10:37.27 | benjk | its the locks that will ultimately kill you |
| 10:37.34 | benjk | not CPU |
| 10:37.56 | MrChimpy | which locks? |
| 10:38.00 | benjk | too much Rocky Mountain locking going on in Asterisk |
| 10:38.36 | benjk | when multiple threads attempt to access the same data structures while being interrupted by each other, bad things will happen |
| 10:38.37 | MrChimpy | if I'd written my FastAGI app in C i could have easily managed a full 240 concurrent |
| 10:38.48 | benjk | to avoid that, you place locks on the data structure |
| 10:39.03 | MrChimpy | but I can't use C, cos if I do the monkeys here can't maintain it. |
| 10:39.34 | benjk | but if your lock is too general and in place most of the time, then most of the concurrent processing will stop as everybody is waiting for the lock to go away |
| 10:39.39 | MrChimpy | so asterisk is overusing mutexes>? |
| 10:40.01 | benjk | overusing and also its using locks that are too broad |
| 10:40.05 | E-bola | Hey any europeans admins here? |
| 10:40.12 | E-bola | how do u solved the am/pm problem with voicemails? |
| 10:40.25 | benjk | more finegrained locking would bring some improvements |
| 10:40.43 | benjk | but the devil is in the detail, the core really has to be rewritten to do stuff differently |
| 10:40.48 | HarryR | benjk, YATE is also pretty scalable (compared to asterisk anyway), but there's no 'official' benchmark |
| 10:40.59 | benjk | I can believe that |
| 10:41.13 | benjk | but the Yate project seems to have an attitude issue |
| 10:41.19 | benjk | not unlike Asterisk ;) |
| 10:41.26 | ShipHead | benjk: How so? |
| 10:41.32 | benjk | I can only handle one such project |
| 10:41.36 | HarryR | i've not had many attitude issues with YATE :) |
| 10:41.53 | benjk | if you ever talk the Yate girl, you'll know what I mean |
| 10:42.02 | HarryR | yeah Diana |
| 10:42.11 | ShipHead | Who's the Yate girl? |
| 10:42.18 | benjk | l-fy |
| 10:42.29 | benjk | that's the IRC nick, anyway |
| 10:42.32 | ShipHead | I like l-fy |
| 10:42.51 | benjk | HarryR, I am putting my bet on OpenPBX |
| 10:43.08 | HarryR | OpenPBX? |
| 10:43.22 | HarryR | oh, yeah but that's still asterisk based |
| 10:43.24 | benjk | well if you like here, fine, I am happy for you, many other people have some difficulties |
| 10:43.44 | benjk | well, we're throwing out more and more Asterisms |
| 10:43.47 | benjk | and replace them |
| 10:44.07 | HarryR | eheh, YATE & Freeswitch just try and Do It Right(tm) from the start ;) |
| 10:44.52 | benjk | for example, we no longer do that wasteful character-by-charcter app name var name context name comparison in the dialplan engine |
| 10:45.08 | benjk | sped up my super low spec test box to gain 2.5 times as many concurrent calls |
| 10:45.22 | benjk | and I am going to throw out all the linked lists, too |
| 10:45.35 | HarryR | that's pretty cool, is openbpx fairly stable at the moment? |
| 10:45.56 | benjk | it is nice and stable and has a number of tricks that Asterisk doesn't |
| 10:46.09 | benjk | full T38 support for example |
| 10:46.23 | benjk | a side effect of using SpanDSP library for the codecs |
| 10:46.45 | benjk | also Zaptel timing dependencies have been removed, all POSIX timer based now |
| 10:47.15 | HarryR | personally I'd like to see a rewrite of the voicemail app and having queues un-borked |
| 10:47.35 | benjk | right now we are mostly working on replacing stuff in the core |
| 10:47.37 | HarryR | can do so much of it with FastAGI, but people don't want that |
| 10:47.49 | benjk | we don't have AGI |
| 10:47.55 | benjk | we have OGI :) |
| 10:47.58 | HarryR | ah |
| 10:48.17 | benjk | to stay clear of any potential trademark hassle |
| 10:48.22 | tparcina | beronet ISDN, does anybody use them? |
| 10:48.37 | HarryR | I don't get people's issue with FastAGI, 'Noo... we want to keep all our complex dialplan logic in the telephony server instead of in a separate iscolated application' idiocy |
| 10:49.26 | benjk | well, if you have a proper dialplan engine, there is nothing wrong with doing it in the core |
| 10:49.40 | benjk | but Asterisk's dialplan engine is really anything else but proper |
| 10:49.55 | benjk | there is of course nothing wrong with offering choice |
| 10:50.15 | benjk | such as pluggable dialplan engines |
| 10:51.19 | HarryR | ah, I suppose all I see it as is just more opportunity for threading problems, and more code to maintain which is considered 'core' |
| 10:51.55 | benjk | well, we've thrown out stuff |
| 10:53.16 | benjk | tparcina, I remember to have helped some guy who was using beronet here this week, but now I can't remember what his nickname was |
| 10:54.16 | benjk | as I understand it, the beronet cards are mostly compatible with Junghanns cards |
| 10:54.34 | benjk | and many people in Europe use those |
| 10:55.08 | benjk | HarryR, rewriting from scratch is clearly nice if you can afford it |
| 10:55.29 | benjk | but I have to do business in the meantime |
| 10:55.50 | benjk | so the gradual replacement approach of OpenPBX suits me better |
| 10:57.09 | *** join/#asterisk folsson (n=folsson@h100n2fls33o985.telia.com) |
| 10:59.11 | HarryR | yah I see :) there's no such thing as an ideal world |
| 10:59.18 | HarryR | I'll have a tinker with it after-hours |
| 11:00.07 | tparcina | benjk: hopefully, someone will help me :)) |
| 11:01.43 | tparcina | benjk: i have found installation guide on Beronet web page. Then I have downloaded script that I was sopouse to, but that script doesn't work as it should... |
| 11:04.32 | benjk | HarryR, we're preparing a release candidate now |
| 11:04.59 | benjk | tparcina, what kind of card is it and what drivers are you using> |
| 11:05.00 | benjk | ? |
| 11:05.56 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:06.03 | jaike | benjk: ever heard of libproxy? to do load balancing of calls between asterisk servers |
| 11:06.04 | *** join/#asterisk topping (n=topping@207.47.6.182.static.nextweb.net) |
| 11:06.28 | benjk | no, I haven't |
| 11:07.09 | benjk | I know, most folks use SER for that sort of thing (if it is SIP) |
| 11:07.09 | HarryR | jaike, how's that differ from something like vserver & ldirectord/piranha? |
| 11:07.57 | jaike | HarryR: dunno, havent done load balancing before. just started doing some research |
| 11:08.25 | HarryR | ah, you should look into them and UltraMonkey, you can get some very neat stuff setup |
| 11:08.33 | benjk | jaike, like I told you before, you're too early ;) |
| 11:09.12 | jaike | benjk: just looking at all the options before telling my boss i cant do it |
| 11:09.19 | jaike | heh |
| 11:12.07 | linagee | does an 800 (or 866) number accept calls from anywhere, any country? do you have to limit your dialplan so you don't get charged up the ying-yang? |
| 11:12.19 | linagee | (or does it work like that) |
| 11:12.54 | h3x | toll free only takes calls from us and optionally canada |
| 11:13.03 | h3x | there are ITFS numbers for international toll free service |
| 11:13.08 | linagee | h3x: how optionally? |
| 11:13.12 | linagee | h3x: ITFS? |
| 11:13.26 | h3x | ITFS arent 800xxxxxxx |
| 11:13.31 | linagee | ah |
| 11:13.37 | linagee | h3x: mine is 866xxxxxxx |
| 11:13.47 | h3x | 8NNNXXXXXX |
| 11:13.47 | h3x | :P |
| 11:13.50 | h3x | err actually |
| 11:13.55 | linagee | h3x: nope. ;) |
| 11:13.56 | h3x | 8YYNXXXXXX to be technical |
| 11:14.01 | linagee | h3x: nope |
| 11:14.15 | linagee | h3x: i live in the 858 area code. that's not toll free. :p |
| 11:14.22 | h3x | you tell your carrier if you want canada origination turned on |
| 11:14.33 | linagee | h3x: i see |
| 11:14.48 | linagee | (i'm sure there are other 8xx area codes that aren't toll free) |
| 11:14.59 | h3x | there are some countries such as the bahamas where you have to pay the carrier to call us toll frees |
| 11:15.13 | benjk | linagee, the 800 numbers have shadow numbers that can be called from any country |
| 11:15.23 | linagee | benjk: shadow numbers? |
| 11:15.25 | h3x | 800, 888, 877, 866, 855, 844, 833, 822 have been reserved |
| 11:15.28 | benjk | the 888, 877 series too |
| 11:15.33 | benjk | but the 866 dont |
| 11:15.42 | linagee | benjk: i see... confusing. heh |
| 11:15.47 | benjk | so if you have an 866 number that is US only, then you have no way to call that from Canada |
| 11:15.58 | linagee | benjk: all i know is, i was able to get an 866 number from voicepulse today |
| 11:15.59 | benjk | and elsewhere also often not |
| 11:16.01 | h3x | you dont really have to worry about somebody calling your toll free and running up a huge bill |
| 11:16.10 | linagee | benjk: aha |
| 11:16.18 | linagee | h3x: no? |
| 11:16.28 | benjk | 866 will only be working in the US, you cant call it from other countries that share the 1 country code (NANPA) |
| 11:16.28 | h3x | canada origination is super expensive on many carriers though |
| 11:16.37 | benjk | get a 877 or 888 or 800 instead |
| 11:16.50 | linagee | benjk: i only want it to work from the US. heh |
| 11:16.56 | benjk | for those numbers there are shadowed area codes |
| 11:17.17 | linagee | benjk: or even just california. i guess i could do caller ID detection on that? hrm.... |
| 11:17.18 | benjk | I think it is 881 for 800, 882 for 888, 883 for 877 |
| 11:17.20 | h3x | with shadowed numbers, the calling party has to pay for part of the call |
| 11:17.30 | benjk | yes |
| 11:17.54 | h3x | i think its really dumb how many companies have a seperate toll free for canada and us |
| 11:17.55 | benjk | its no longer entirely toll free for the overseas caller but at least you can get called |
| 11:18.04 | h3x | if they knew they could just have the resporg set up to LCR the call to two different carreirs |
| 11:18.33 | linagee | benjk: some people have caller ID from a different area. maybe a better idea would be a message saying, "we have detected you live in <insert detected state here, spoken using festival or do all the states recordings>. if you believe this is incorrect, please press zero now." |
| 11:18.57 | linagee | that's one long <>. :) |
| 11:19.29 | h3x | krikey, paypal is paying 5.02% now |
| 11:19.39 | benjk | but for that you have to pick up the call, so they need to be able to call you in the first place and if they call you from Canada on a US toll free number its going to cost you extra |
| 11:19.40 | h3x | what kind of mobster investments are they making |
| 11:20.02 | linagee | benjk: what if i call into my box using a pay phone? are there extra charges? |
| 11:20.16 | h3x | oh yes |
| 11:20.20 | linagee | !!! :( |
| 11:20.21 | h3x | payphone surcharge is $0.50 |
| 11:20.24 | benjk | not if its a toll free number and the pay phone is within the same country |
| 11:20.25 | linagee | bah. |
| 11:20.28 | h3x | you can block that |
| 11:20.31 | benjk | oh |
| 11:20.40 | linagee | $0.50? |
| 11:20.42 | benjk | really? is that provider dependent? |
| 11:20.49 | h3x | that is the current tariffed rate in the US |
| 11:20.53 | benjk | shit |
| 11:20.55 | h3x | some carriers mark it up |
| 11:20.56 | linagee | that sucks |
| 11:21.00 | qdk | benjk: are you working on the openpbx project? |
| 11:21.03 | benjk | that sucks balls |
| 11:21.11 | h3x | that is why we have ANI-II delivery |
| 11:21.11 | benjk | I am a contributor yes |
| 11:21.12 | linagee | h3x: what if someone brought an autodialer to a pay phone? |
| 11:21.20 | benjk | heh |
| 11:21.21 | h3x | ling: its been done before shhh ;) |
| 11:21.39 | linagee | h3x: or if you were a phone company and hired a group of homeless people to sit there and dial all day? :( |
| 11:21.50 | h3x | ANI-II on PRI or SS7 or whatever can tell you in real time if the call was originated from a payphone, hotel, prison, etc. |
| 11:21.52 | qdk | benjk: so you would know if it was completely written in perl or not? |
| 11:21.59 | h3x | linagee: Already seen it done :) |
| 11:22.06 | linagee | h3x: you said there is a way to block payphones? |
| 11:22.20 | h3x | yes, that is another option you can ask from the underlying carrier to put in SMS/800 |
| 11:22.27 | h3x | or they can do it on their switch |
| 11:22.34 | linagee | i wonder if i will get random dialed more often having an 866 number than a local area code number.... |
| 11:22.45 | h3x | payphone compensation is one of those things that is a pain in the ass to deal with |
| 11:22.46 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 11:22.50 | linagee | h3x: SMS/800? |
| 11:22.59 | h3x | the default is to charge the carrier for all the calls, weather or not they got completed with CABS billing |
| 11:23.15 | h3x | then your carrier has to reject records that didn't get completed and try to get their money back |
| 11:23.20 | linagee | what is the caller id on a pay phone like? hehehe |
| 11:23.26 | linagee | time to find out! :) |
| 11:23.34 | linagee | i think i have fifty cents around here somewhere. ;) |
| 11:23.38 | h3x | well, the payphone flag isnt something you usually see |
| 11:23.45 | h3x | oh you dont have to put money in payphones to call toll frees |
| 11:23.55 | h3x | thats why the customer at the other side has to pay for it |
| 11:23.58 | linagee | h3x: i know, i know. :) (that's the point, isn't it?) |
| 11:24.21 | h3x | theres all kinds of CABS billing fraud all the time... |
| 11:24.26 | linagee | ? |
| 11:24.37 | linagee | you mean people don't pay the $0.50? |
| 11:24.38 | h3x | CABS is the format carriers use to bill each other |
| 11:25.13 | h3x | no i mean theres payphone carriers that sometimes fraudulently bill the 50 cents when there was no call ;) |
| 11:25.24 | linagee | wtf? |
| 11:25.25 | h3x | its settled later |
| 11:25.42 | linagee | h3x: you're saying getting a toll free number is a very bad idea?!?!? :-/ |
| 11:25.49 | h3x | but your toll free carrier usually bases the charges to you from the CDRs by storing the ANI-II digits |
| 11:26.04 | h3x | well, like i said they can block payphone origination |
| 11:26.16 | h3x | so they will get a disconnect message "this number dosen't accept calls from payphones" etc |
| 11:26.24 | h3x | by the way |
| 11:26.31 | h3x | many HOTELS consider their room phones to be "payphones" |
| 11:26.39 | linagee | that is weird |
| 11:26.42 | h3x | some asshole got that legally established |
| 11:26.47 | h3x | well, because local calls are billed to the room |
| 11:26.58 | linagee | right, and 800 numbers are usually free |
| 11:27.09 | h3x | so they can get away with charging the toll free customer and their guest if they want to |
| 11:27.25 | h3x | or that |
| 11:28.11 | linagee | h3x: so when you call from a pay phone, what does the caller ID look like? is there a DID set up on pay phones? heh |
| 11:28.26 | h3x | sometimes there is, sometimes it just says out of area |
| 11:28.28 | h3x | theres no standard on that |
| 11:28.34 | linagee | hrm. that sucks |
| 11:28.41 | h3x | a lot of times it says the same # for one company in a whole rate center |
| 11:28.58 | linagee | h3x: i wonder if it costs more than a local call to call a payphone? hehehe. (toll reversal) |
| 11:29.03 | h3x | they just do that becuase theres so many people with telemarketing blockers |
| 11:29.20 | h3x | no but most payphones wont take incoming calls. almost never |
| 11:29.27 | linagee | awww. :( |
| 11:29.34 | h3x | by the way theres a payphone show here in vegas if you wanna come see it sometime |
| 11:29.35 | h3x | hehe |
| 11:29.38 | h3x | thats where i found out a lot of this stuff |
| 11:29.47 | linagee | payphone show? lol |
| 11:29.50 | h3x | convention |
| 11:30.02 | h3x | i think its called APCC |
| 11:30.03 | linagee | that is very strange |
| 11:30.32 | linagee | apcc.net |
| 11:30.35 | *** join/#asterisk hank (n=hank@l4m3.de) |
| 11:30.48 | linagee | American Public Communications Council |
| 11:31.00 | h3x | oh yeah |
| 11:31.15 | linagee | oh cool!! |
| 11:31.19 | linagee | i just realized something |
| 11:31.25 | linagee | my apartment has a fax machine |
| 11:31.33 | linagee | you have to use a calling card |
| 11:31.41 | linagee | hrm... :-D |
| 11:31.46 | linagee | not anymore! :-D |
| 11:31.57 | h3x | haha |
| 11:32.08 | h3x | i so pimped my phones when i went to vancouver for a week vacation |
| 11:32.13 | linagee | yeah, too bad i'm moving out of here. (way too expensive) |
| 11:32.23 | h3x | i bought a prepaid $50 fido sim card off ebay for my moto razr |
| 11:32.24 | linagee | h3x: pimped? |
| 11:32.28 | h3x | for $20 |
| 11:32.34 | linagee | fido sim card? |
| 11:32.37 | h3x | (they unbundled the phone from the sim card) |
| 11:32.41 | h3x | fido is one of the 4 wireless carriers up there |
| 11:32.51 | linagee | i see |
| 11:32.53 | h3x | then, i set up a vancouver DID to point at my asterisk box |
| 11:33.02 | h3x | to run DISA which gives me a dialtone in the default context |
| 11:33.02 | linagee | h3x: do prepaid sim cards offer incoming calls? |
| 11:33.15 | h3x | so i could check my voicemail and call the US without having to pay $0.20 a minute for long distance on the cell |
| 11:33.19 | h3x | yep |
| 11:33.39 | linagee | that is weird. heh |
| 11:33.40 | h3x | and then i forwarded my US numbers to the canadian cell coz its cheap as hell wholesale |
| 11:33.47 | h3x | like .009/min or something |
| 11:33.51 | linagee | nice |
| 11:34.06 | linagee | h3x: what is "wholesale" :) |
| 11:34.12 | h3x | buying too many minutes |
| 11:34.12 | h3x | haha |
| 11:34.19 | linagee | <insert unknown provider here>? :P |
| 11:34.34 | h3x | well i own a istp/carrier |
| 11:34.35 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 11:34.53 | linagee | voicepulse is $0.02/min for canada |
| 11:34.59 | h3x | the big ones, qwest, mci, global crossing, etc. |
| 11:35.28 | h3x | that isnt a bad retail price :) |
| 11:35.56 | h3x | but fuk it i wanna move to canada. people are paying telus and bell .35/min for long distance |
| 11:35.59 | h3x | STILL |
| 11:36.06 | linagee | h3x: oh yeah! |
| 11:36.15 | h3x | calling cards, $0.38/min |
| 11:36.27 | h3x | canadian dollars, which are .90 to USD $1 |
| 11:36.38 | linagee | h3x: it's $0.02/min, but "you're not supposed to use it for a permanent number because they don't have a way to forward the tax to canada" or something. :-D |
| 11:36.40 | linagee | silly taxes |
| 11:36.43 | h3x | the average competitor charges 9c /min |
| 11:37.00 | h3x | now you sound like voipjet |
| 11:37.05 | linagee | h3x: huh? |
| 11:37.11 | h3x | thats the disclaimer on their page too |
| 11:37.28 | h3x | canada's GST is awful |
| 11:37.52 | h3x | BC provincal tax +federal was 13% together |
| 11:38.00 | h3x | that is applied to products AND services |
| 11:38.06 | h3x | they also have parking tax |
| 11:38.12 | h3x | hahaha |
| 11:38.19 | h3x | a pack of cigs is $13 with the taxes up there |
| 11:38.20 | linagee | h3x: so yeah. most anywhere in USA is $0.01/min or less. :-D |
| 11:38.20 | ghenry | HOw do you set what codec of sounds get used in /var/lib/asterisk? |
| 11:38.32 | linagee | h3x: i can call my sister in nevada. hehehe |
| 11:38.35 | h3x | Not really... |
| 11:38.51 | h3x | little ILECs and paging providers are expensive |
| 11:38.52 | linagee | h3x: ($0.01/min or less using voicepulse) |
| 11:38.53 | ghenry | I want to use gsm, but ulaw is getting used. The english UK ones are in gsm |
| 11:38.59 | linagee | h3x: huh? |
| 11:39.06 | h3x | voicepulse just dosent know how to balance their books |
| 11:39.13 | linagee | hehhe. cool. :-D |
| 11:39.26 | h3x | there are some rural areas getting $0.03/min or higher |
| 11:39.37 | linagee | h3x: sometimes you get a bit of static even using pcm with them. (not sure if it's my side) |
| 11:39.45 | h3x | the local carrier gets paid for the long distance calls terminating on them |
| 11:39.53 | h3x | reciprocal compensation |
| 11:40.00 | linagee | it very well could be my side as i'm running asterisk inside vmware. heh |
| 11:40.18 | h3x | That is why its so expensive to call long distance on a ordinary landline (switched long distance) |
| 11:40.40 | linagee | it's not enough noise to cause it to be annoying or anything. |
| 11:40.40 | h3x | the long distance carrier has to share revenue with both the originating and terminating side |
| 11:40.40 | h3x | local providers |
| 11:40.44 | h3x | when you use voip or dedicated T1s, it bypasses one side of it |
| 11:41.00 | ghenry | any ideas on the ulaw/gsm loading for all asterisk sounds? |
| 11:41.03 | linagee | h3x: do you know what this means? "4.9¢/min incoming to toll-free numbers with free CNAM" |
| 11:41.18 | h3x | ghenry: you can use sox to transcode |
| 11:41.19 | linagee | h3x: does that mean i get charged 5 cents a minute for people calling into my toll free? |
| 11:41.32 | h3x | yes |
| 11:41.37 | h3x | CNAM is caller id name service |
| 11:41.39 | linagee | h3x: but CNAM? any ideas? |
| 11:42.14 | h3x | voip toll free is all weird |
| 11:42.16 | ghenry | h3x: yeah. but I just want to switch to gsm for now to test. ANy ideas. I have allow=gsm in sip.conf, but can't see where else to allow the default sounds codec version |
| 11:42.22 | linagee | hrm |
| 11:42.22 | h3x | theres so few carriers that do it |
| 11:42.37 | linagee | h3x: i think voicepulse just started |
| 11:42.38 | h3x | because of some anal retentive people in their legal departments |
| 11:42.56 | h3x | the law supposedly reads "PC to Phone communications is determined by the originating party" |
| 11:43.03 | h3x | the originating party in all toll free calls is a phone |
| 11:43.12 | h3x | they dont seem to think that a Phone to PC is considered VoIP |
| 11:43.17 | *** join/#asterisk sarum4n (n=some@dsl-083-247-031-058.solcon.nl) |
| 11:43.28 | h3x | this is what Qwest's legal schmuck told me |
| 11:43.31 | ghenry | I'll try deleting them all then rm -rf *.ulaw |
| 11:43.35 | linagee | that's... confusing. |
| 11:43.36 | h3x | their head legal schmuck |
| 11:43.43 | h3x | they do sell 8YY wholesale VoIP now |
| 11:44.05 | h3x | another more likely reason most people dont do it is lack of support for ANI-II on their VoIP gateways so they cant bill payphone origination properly |
| 11:44.34 | h3x | or furthermore, maybe their gateway supports it but billing software or SIP proxy dosen't |
| 11:44.42 | ghenry | that the best way |
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| 11:47.55 | tparcina | benjk: its Beronet BN4S0, i'm using their script to install drivers, but that script fails... |
| 11:49.23 | benjk | qdk, see my pm |
| 11:49.49 | tparcina | e164.org, how to check one number without using asterisk? |
| 11:50.09 | benjk | there is Voicetronix (and Aussie company making PCI telephony cards) who have a Perl PBX (I think its not an IP PBX but analog only) |
| 11:50.18 | benjk | this thing is also called OpenPBX |
| 11:50.28 | benjk | the OpenPBX I was talking about is OpenPBX.org |
| 11:50.42 | benjk | that's a fork of Asterisk |
| 11:50.53 | backblue | what do you need about beronet bn4s0? |
| 11:51.02 | backblue | (i was not following) |
| 11:51.58 | qdk | benjk: yesyes.. busy at work here. :-) |
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| 12:07.56 | hank | i do need an ntba(nt1) when i want to connect isdn telephones to asterisk dont i? |
| 12:08.41 | hank | so its hfc in nt mode -> nt1 -S0-> phones |
| 12:08.43 | hank | right? |
| 12:14.06 | backblue | hank: no, if you have power on the bri card |
| 12:14.29 | backblue | hank: if you have a single bri, yes, you will need one NT to give power to the phone |
| 12:15.43 | hank | hmm power on the bri card... the power the card gets from the computer itself would be enough?? |
| 12:15.58 | hank | or would it need an external power supply? |
| 12:16.59 | *** join/#asterisk skywriter (n=test@mail.splendor.net) |
| 12:17.36 | backblue | with single bri cards |
| 12:17.43 | backblue | you will need allways NT's |
| 12:17.57 | *** part/#asterisk jaike (n=jaike@125.5.144.90) |
| 12:18.02 | backblue | but if you use BN2S0 BN4S0 BN8S0 |
| 12:18.05 | hank | so an nt is needed to get power on the s0 line. is that its sole purpose? |
| 12:18.14 | backblue | it has external power supply, and you can connect the phone directly |
| 12:18.34 | hank | but only with a crossed isdn cable, right? |
| 12:18.40 | backblue | hank: no, it's not, it should terminate the circuit with 100ohm resistence. |
| 12:19.11 | backblue | (i wonder why was no one here, in the last year to help me out, when i was with this questions...) |
| 12:19.31 | backblue | hank: no, you need to see which pin-in has you NT |
| 12:19.40 | backblue | each NT have it's own pin-in |
| 12:19.57 | backblue | you have to put the cables in the correct order |
| 12:20.08 | backblue | and terminate the circuit in the dip-switches |
| 12:20.35 | hank | backblue: 'g' sympathetic. i thought i was really dumb for not understanding such things. im trying to get asterisk ready since one month i think... but i dont think i have come any nearer to what i want to reach. |
| 12:20.56 | hank | ok i c |
| 12:21.39 | hank | is there any official documentation about that? i have a bet running with a colleague. i say "one needs an ntba" he says "no, one can plug the phones directly into the isdn card" |
| 12:23.03 | backblue | well, you can, if your phone has his own power supply |
| 12:23.18 | backblue | i dont know your system |
| 12:23.30 | backblue | tell me your ntba references |
| 12:23.40 | backblue | so i can try to search here, if i have the pinout |
| 12:23.41 | hank | i have two longshine card with hfcs chipsets. i have no ntba yet. |
| 12:23.43 | backblue | to give you |
| 12:23.54 | hank | thats what the whole thing is about. |
| 12:24.01 | hank | he says i dont need an ntba, i say i do |
| 12:24.09 | backblue | ok, you need powered phones, or ntba's to connect the other ones |
| 12:24.29 | backblue | it's quit simple, do the phones have power supply? |
| 12:25.40 | hank | we have no phones yet 'g' |
| 12:26.52 | hank | lets assume we had phones with power supply and twisted isdn cables. we could connect them directly to hfc cards in nt nt mode then? |
| 12:27.45 | hank | without the need for resistors? |
| 12:28.45 | benjk | hank, do those cards glow in the dark? |
| 12:29.27 | hank | hmm i think i am missing a joke here... :( |
| 12:29.32 | hank | benjk: what do you mean? |
| 12:30.02 | benjk | Longshine and all that |
| 12:30.20 | benjk | ;) |
| 12:30.21 | hank | 'g' ok |
| 12:32.29 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:33.20 | hank | harr no answer to that last question? |
| 12:34.15 | benjk | maybe he had to go to the bathroom |
| 12:34.34 | hank | benjk: i didnt ask him but the channel ;) |
| 12:34.53 | benjk | not many people use BRI phones, I think |
| 12:35.03 | benjk | I use BRI, but only for the connection with the telco |
| 12:35.06 | hank | really? |
| 12:35.19 | hank | a pity |
| 12:35.25 | benjk | well, as far as the population in this channel is concerned |
| 12:36.02 | benjk | some do, but I wouldn't exactly say they are in a majority |
| 12:36.03 | hank | so thats why getting support for that sh^Wstuff is so difficult |
| 12:37.03 | benjk | there are for sure many more people who use BRI for the telco connection only than there are those who hook up ISDN phones |
| 12:37.27 | hank | mhm probably |
| 12:37.34 | hank | seems like the bet will fall on ice |
| 12:39.36 | benjk | sometimes its also a matter of time of day |
| 12:41.18 | *** join/#asterisk teknoprep (n=tekon@unaffiliated/teknoprep) |
| 12:41.51 | hank | true, true |
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| 12:57.48 | fourcheeze | hey |
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| 13:01.29 | fourcheeze | is there a kind of Polycom IP 501 that can take a PSU into the case rather than POE? |
| 13:02.18 | [TK]D-Fender | fourcheeze: Yes |
| 13:02.26 | *** join/#asterisk cian (n=cian@cian.ws) |
| 13:04.11 | [TK]D-Fender | fourcheeze: IP 501 can be powered 1 of 2 ways : Using a special cable with a PoE circuit inline (bluky dongle), or another special cable that has a small power jack on it for which you plug in a traditional wall-wart. |
| 13:07.18 | *** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net) |
| 13:08.24 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 13:08.25 | fourcheeze | [TK]D-Fender: but each way you need to be near a hub? |
| 13:08.56 | fourcheeze | I've got someone with a long run of utp patch around a room |
| 13:09.13 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:09.31 | [TK]D-Fender | fourcheeze: no, you need to plug the "local" powered cable direct to the phone (its keyed so it HAS to be there unless you mod it). From there you can add a coupler and the run more cable if you want |
| 13:10.19 | *** join/#asterisk pph (n=pph@81.255.164.157) |
| 13:14.49 | fourcheeze | [TK]D-Fender: ok but there's no option to plug something into the phone itself, so at the very least I'll need a coupler |
| 13:15.52 | [TK]D-Fender | fourcheeze: Yeah, thats sums it up. |
| 13:16.23 | [TK]D-Fender | fourcheeze: : OR you can maybe mod the cable and remove that little plastic nodule that makes the jack keyed and then you can move it further back. |
| 13:19.42 | *** join/#asterisk drcode (n=chatzill@87.69.59.186.cable.012.net.il) |
| 13:19.47 | drcode | hia l |
| 13:19.48 | drcode | hi all |
| 13:20.05 | drcode | any recommanded sip client ? |
| 13:20.30 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 13:20.32 | [TK]D-Fender | drcode: Ekiga for Linux, eyeBeam for Windows |
| 13:20.40 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 13:20.51 | drcode | I mean open source x-lite? |
| 13:21.26 | [TK]D-Fender | drcode: Ekiga is GPL, eyebeam is a paid client. |
| 13:21.36 | drcode | I read that ast. support AOPEN standard |
| 13:21.38 | Dr-Linux | [TK]D-Fender: i'm getting again and again these stuff on consol, what should i do to stopd this >> see here >> http://pastebin.ca/199326 |
| 13:21.55 | drcode | how can I know if the modem is aopen voice modem standrd |
| 13:22.16 | [TK]D-Fender | Dr-Linux: Reduce your debug level |
| 13:22.47 | [TK]D-Fender | drcode: Not a question you should ask here, and it would not be compatible for use with *. |
| 13:23.01 | drcode | k |
| 13:23.08 | drcode | I just read it in the forum |
| 13:23.12 | *** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk) |
| 13:24.01 | Dr-Linux | [TK]D-Fender: currenlty below lines are enabled in logger.conf: |
| 13:24.01 | Dr-Linux | console => notice,warning,error,debug |
| 13:24.01 | Dr-Linux | messages => notice,warning,error |
| 13:24.04 | [TK]D-Fender | Dr-Linux: "set debug 0" |
| 13:24.18 | Dr-Linux | [TK]D-Fender: should i remove debug |
| 13:24.22 | Dr-Linux | hhm.. ok |
| 13:24.33 | ghenry | Hi all. What's the best point to start/debug when trying to fix echo between a TDM400P card and Sip phones (Aastra 480i phones) |
| 13:24.37 | [TK]D-Fender | Dr-Linux: DUH! |
| 13:24.41 | ghenry | zapata.conf? |
| 13:24.53 | ghenry | for incoming calls |
| 13:25.02 | ghenry | from pstn |
| 13:25.04 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 13:25.04 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:25.05 | [TK]D-Fender | ghenry: Thats where all of the changes will need to take place. |
| 13:25.10 | sevard | [TK]: you tell dat foo |
| 13:25.41 | Dr-Linux | [TK]D-Fender: i did: |
| 13:25.41 | Dr-Linux | LHR-PBX*CLI> set debug 0 |
| 13:25.41 | Dr-Linux | LHR-PBX*CLI> |
| 13:25.50 | Dr-Linux | but sill i'm getting those stuff |
| 13:26.18 | [TK]D-Fender | ghenry: Give this a good read and then focus on "fxotune" in the middle... http://www.voip-info.org/wiki/view/Asterisk+echo+cancellation |
| 13:26.29 | ghenry | thanks [TK]D-Fender: So tweak |
| 13:26.29 | jmls | Dr-Linux: remove debug from the console line |
| 13:26.33 | ghenry | thanks again |
| 13:26.34 | ghenry | reading |
| 13:26.40 | jmls | console => notice,warning,error |
| 13:26.45 | jmls | from the cli |
| 13:26.45 | [TK]D-Fender | ghenry: Good luck you may need it |
| 13:26.50 | jmls | logger rotate |
| 13:26.54 | ghenry | [TK]D-Fender: Stock Asterisk, latest version was fine |
| 13:27.05 | ghenry | [TK]D-Fender: Trixbox 1.2.1 == echo. hmmmm |
| 13:27.08 | ghenry | chat later |
| 13:27.16 | Dr-Linux | jmls: that's what i was doing, but maybe that will stop debug forever |
| 13:27.36 | [TK]D-Fender | ghenry: Basically says that the zaptel build it includes isn't as good as your "stock" one |
| 13:27.37 | jmls | it will stop debug on the console. you still have debug going to the messages file |
| 13:27.46 | sevard | Might that be a clue that you would want to cencel your trixbox shitinstall and rebuild *? |
| 13:27.49 | ghenry | [TK]D-Fender: Yup. |
| 13:27.58 | [TK]D-Fender | ghenry: Great reason why I highly recommend HWEC cards and not the low end dependent ones |
| 13:28.09 | ghenry | yup, and yup sevard |
| 13:28.19 | [TK]D-Fender | sevard: unload chan_bile.so |
| 13:28.21 | ghenry | maybe just use stock * and freepbx now |
| 13:28.47 | ghenry | don't really have a clue what they've done to the zaptel kernel RPMs! :-( |
| 13:29.30 | Dr-Linux | how can we deny/block an ip address in sip.conf? |
| 13:30.39 | [TK]D-Fender | Dr-Linux: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf |
| 13:33.26 | *** join/#asterisk technwork (n=technoid@host-69-95-124-10.cwon.choiceone.net) |
| 13:35.23 | *** join/#asterisk BrokenNoze (n=SimonK@host86-144-75-221.range86-144.btcentralplus.com) |
| 13:36.20 | BrokenNoze | anyone have time to help a newbie compile zaptel driver? |
| 13:36.31 | drcode | I have qustion |
| 13:36.51 | drcode | I dont know if its ast. or not |
| 13:36.52 | [TK]D-Fender | BrokenNoze: Ok, where are you stuck? |
| 13:37.05 | BrokenNoze | the kernel source install |
| 13:37.11 | BrokenNoze | using FC4 |
| 13:37.41 | BrokenNoze | when I try and make - I get you don't appear to have the kernal-sources installed |
| 13:37.42 | drcode | If I want to call from europe or usa and the local call will translate into sip , so I can get it into my ast box , is it possible? |
| 13:37.43 | [TK]D-Fender | BrokenNoze: You need to install bother the source for your kernel, and the headers. |
| 13:38.06 | BrokenNoze | OK, I've downloaded the src.rpm file |
| 13:38.20 | BrokenNoze | and i've tried to rpm -Uhv it |
| 13:38.42 | [TK]D-Fender | BrokenNoze: try "yum install kernel-source" or something like that. Not sure the eaxt line. You can ask in #fedora like "whats the YUM line to install the source for my kernel" and someone should be able to hand you the 100% answer in seconds |
| 13:39.38 | BrokenNoze | Awesome cheers. i'll try someone there |
| 13:40.13 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 13:41.33 | tparcina | e164.org, how to check one number without using asterisk? |
| 13:43.15 | drcode | tparcina: nice |
| 13:44.03 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 13:44.03 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 13:45.57 | ghenry | [TK]D-Fender: If you recompile zaptel, i.e. for a newer version. Asterisk needs to be recompiled against the version of zaptel, yeah? |
| 13:46.17 | ghenry | silly question, I know the answer is yes |
| 13:48.05 | ghenry | I'm am scared of fxotune! |
| 13:48.31 | [TK]D-Fender | ghenry: Yes, and You may need to match versons for things not to freak out. |
| 13:48.43 | skywriter | how can i dialfrom the command line? |
| 13:48.52 | ghenry | I know. bummer. |
| 13:48.56 | [TK]D-Fender | ghenry: As well you should be. Time to buy a real card with HWEC :) |
| 13:49.15 | ghenry | skywriter: dial appliacation if compiled with it |
| 13:49.37 | skywriter | how do i know |
| 13:49.52 | ghenry | [TK]D-Fender: Yeah, but I just binned a working * install, and wiped it with Trixbox. I should of just installed Freepbx or something on top of it instead |
| 13:50.00 | *** join/#asterisk xnon (n=xnon@200.82.222.85) |
| 13:50.02 | ghenry | [TK]D-Fender: Time to fess up to the client |
| 13:50.20 | ghenry | SkymeyeR: show applications should do it, or typing help |
| 13:50.28 | [TK]D-Fender | ghenry: If you backed them up you should jsut restore things. |
| 13:51.03 | ghenry | [TK]D-Fender: yup. I got /etc/asterisk, just forgot /etc/zaptel.conf, but that's nothing |
| 13:51.12 | skywriter | anyone have solve the following problem |
| 13:51.18 | [TK]D-Fender | ghenry: exactly. 10 second fix |
| 13:51.18 | skywriter | WARNING[3999]: chan_sip.c:9720 handle_response_invite: Forbidden - wrong password on authentication for INVITE t |
| 13:51.36 | [TK]D-Fender | skywriter: Your user or pass is wrong. DEAL WITH IT |
| 13:51.44 | ghenry | [TK]D-Fender: need to wipe trixbox though, no more machine |
| 13:52.01 | [TK]D-Fender | ghenry: Therefor....? |
| 13:52.20 | skywriter | i don think so |
| 13:52.28 | ghenry | [TK]D-Fender: I am an idiot ;-) |
| 13:52.35 | skywriter | i m putting the right password |
| 13:53.01 | *** join/#asterisk hegemoOn (i=dom@80.82.16.205) |
| 13:53.05 | [TK]D-Fender | skywriter: Either the user is wrong or the password is wrong. Doesn't matter what you think is right. You are clearly WRONG. |
| 13:53.17 | skywriter | it s somethiing that have to do with NAT,proxy firewall, cause my atserisk register correctly with the password |
| 13:53.45 | skywriter | but when i make outside calls i get the message |
| 13:53.49 | [TK]D-Fender | skywriter: That message means what it says. |
| 13:54.15 | [TK]D-Fender | skywriter: PM me a PRIVATE pastebin of your sip.conf including those passwords. |
| 13:54.30 | [TK]D-Fender | skywriter: And the CLI output of the attempted call. |
| 13:54.50 | skywriter | no it doesn t , my asterisk register with password,but outside calls say that password is wrong howcome |
| 13:55.13 | skywriter | or the password is wrong for both of them , or it should work for both of them |
| 13:57.02 | *** join/#asterisk L-info (n=linfo@g0962184.demon.co.uk) |
| 13:57.59 | *** join/#asterisk ellisdee (n=ellisdee@69.15.174.113) |
| 14:03.52 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 14:04.46 | *** join/#asterisk M_at (n=matt@lrfadsl01.demon.co.uk) |
| 14:05.49 | M_at | Can anyone help a novice with a dialplan query? Need to strip the leading digit from a passed extension number. |
| 14:06.12 | [TK]D-Fender | M_at: ${EXTEN:1} |
| 14:06.28 | M_at | In context? |
| 14:06.44 | [TK]D-Fender | ${whatevermyvarhappenstobecalled:numberofdigitstostrip} |
| 14:06.57 | *** join/#asterisk X-Gen (n=X-Gen@dsl-145-212-230.telkomadsl.co.za) |
| 14:07.09 | [TK]D-Fender | M_at: Show us what you're working with. use www.pastebin.ca please. |
| 14:07.12 | M_at | So I can place that at the start of the context and it will work from then on? |
| 14:07.55 | [TK]D-Fender | M_at: Depends. Please show us where you are loking to do this. |
| 14:07.59 | *** join/#asterisk Dovid (n=dovi5988@barak.cellcom.co.il) |
| 14:09.32 | M_at | In that case I will probably be better off blatting this machine back to nothing and installing from scratch - it's a bit of a bastard machine right now |
| 14:09.35 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 14:11.46 | BrokenNoze | Hi. still building my zaptels - on a make clean I'm getting an error No sure file or direcory |
| 14:12.08 | [TK]D-Fender | M_at: what do you mean "in that case"? You must already have been very depressed about its state to jump somewhere so negative so fast when I jsut asked to see where you would like to do this in your dialplan.... |
| 14:12.20 | Dovid | BrokeNoze: what version of linux ? and do u have the kernel sources ? |
| 14:12.23 | BrokenNoze | make: *** /lib/modules/2.6.11-1.1369_FC4/build: No such file or directory. Stop. |
| 14:12.33 | Assid | you need the kernel source |
| 14:12.35 | Assid | s |
| 14:12.47 | [TK]D-Fender | BrokenNoze: You need the kernel source AND the headers for your current kernel |
| 14:12.54 | BrokenNoze | I thought I had it! I got the devel |
| 14:13.03 | BrokenNoze | kernel-devel |
| 14:13.22 | BrokenNoze | just the headers in devel? FC4 |
| 14:13.49 | Dovid | BrokenNoze: Yum -y install kernel-sources |
| 14:14.38 | BrokenNoze | nothing to do apparently? |
| 14:15.19 | *** join/#asterisk blebleble (i=godie@caesar.godie.net) |
| 14:15.24 | Dovid | do a yum list kern* |
| 14:16.20 | BrokenNoze | smp packages? |
| 14:16.50 | M_at | [TK]D-Fender: The dialplan was originally a TrixBox one but it's been altered by hand extensively since as part of the learning process. The config is split over too many file for me to easily use pastebin right now - I was hoping that I'd have this one last thing sorted out before stripping the machine back to start from scratch but it may be easier to do it tomorrow |
| 14:17.29 | Dovid | yes |
| 14:18.00 | [TK]D-Fender | M_at: Ok, that explains it. Ditch Trixbox. |
| 14:18.28 | Dovid | M_at: first time I built my own box I copied a friends configs, printed them up and used it as a refrence. and as TK Said "drop the trixbox" |
| 14:18.41 | Dovid | lots of "asterisk admins" - all they know is the trixbox gui |
| 14:18.46 | M_at | I generally have done |
| 14:19.20 | Dovid | M_at: Time to get in to the belly of the beast |
| 14:19.20 | M_at | Using * built from scratch but with FreePBX ontop for some small installs but this is going to be a central box with some funkier routing in place |
| 14:20.50 | M_at | My problem seems to be too much conflicting information out on the web |
| 14:21.15 | BrokenNoze | ok. smp and the smp-devel - same problem. do I need the kernel-xenU stuff? |
| 14:21.24 | Dovid | M_at: between where and where ? |
| 14:21.41 | Dovid | BrokenNoze: no u shouldnt need that |
| 14:21.50 | Dr-Linux | what's this: Oct 12 19:41:07 NOTICE[16989]: chan_sip.c:6275 check_auth: stale nonce received from ... peer ... peer@host.host.host.host ? |
| 14:21.58 | Dovid | what other options do they give u to install if u do: Yum List Kern* |
| 14:22.09 | BrokenNoze | xenu |
| 14:22.17 | Dovid | hmm |
| 14:22.25 | Dovid | this is a question for TK |
| 14:22.28 | BrokenNoze | doc.noarch |
| 14:22.38 | Dovid | i dont think it should matter but maybe a reboot |
| 14:22.47 | BrokenNoze | thats about it. |
| 14:22.51 | BrokenNoze | OK. I'll try that. |
| 14:22.54 | M_at | Dovid: things like StripMSD and the ${EXTEN:1} |
| 14:23.47 | Dovid | what conflicting about ${EXTEN:} ? that is a simple one (granted I had issues with it when is started - was doing ${exten} and NOT {EXTEN} ) |
| 14:24.47 | M_at | Dovid: I mean there's osme people sayinguse stripMSD and others showing ${EXTEN:1} - but nothing that states definitively that one is right and the other old or wrong |
| 14:25.07 | M_at | As both achieve the same result |
| 14:25.11 | pif | has chan_misdn been removed by debian ? |
| 14:25.21 | Dovid | M_at: people give examples and info based on thier own results |
| 14:25.22 | r0d3nt|m | Oct 12 07:25:32 NOTICE[8684]: channel.c:1956 ast_read: Dropping incompatible voice frame on Local/28@PHONE4-9fdd,2 of format ulaw since our native format has changed to slin |
| 14:25.24 | pif | it should be in * 1.2.x |
| 14:25.24 | [TK]D-Fender | M_at: minor stuff. and stripMSD is outdaed. you should always use the standard string notation method. |
| 14:25.26 | r0d3nt|m | any ideas ??? |
| 14:25.43 | Dovid | for instance to get the exten value u do ${EXTEN} and NOT ${EXTEN:1} |
| 14:25.55 | M_at | [TK]D-Fender: Yup, found the patch notes where that was noted from Digium |
| 14:26.06 | Dovid | ${EXTEN:1} will strip the first number off the CID |
| 14:26.20 | Dovid | oops |
| 14:26.23 | Dovid | off the EXTEN |
| 14:26.28 | M_at | ${EXTEN:2:2} will get the 3rd and 4th digits etc |
| 14:26.35 | Dovid | yup |
| 14:26.45 | Dovid | well no |
| 14:26.48 | M_at | That in itself makes sense |
| 14:26.49 | Dovid | first 2 and last 2 |
| 14:27.17 | Dovid | i use voip-info.org for all my gen. questions |
| 14:27.19 | *** join/#asterisk Ebola (n=Ebola@host86-138-123-50.range86-138.btcentralplus.com) |
| 14:27.34 | M_at | Integrating it into the dialplan is where I get a little muddled |
| 14:28.11 | M_at | Of course if my legacy PBX didn't add digits to the beginning of the extension number I wouldn't have this problem ;) |
| 14:28.18 | [TK]D-Fender | M_at: You integrate it anywhere you want o have prefixes to help in routing etc, an then strip off to get the "relevent" parts. |
| 14:28.43 | M_at | Is there a good dialplan tutorial on voip-info.org? |
| 14:28.46 | [TK]D-Fender | M_at: You can use whatever strategy you want with *. I don't have prefixes for anything in my setups |
| 14:28.58 | [TK]D-Fender | M_at: The best out there is the BOOK |
| 14:28.59 | r0d3nt|m | this error popped up after updating to the latest asterisk... it appears to be working normally though.. :-/ " Oct 12 07:25:32 NOTICE[8684]: channel.c:1956 ast_read: Dropping incompatible voice frame on Local/28@PHONE4-9fdd,2 of format ulaw since our native format has changed to slin " Anyone have any ideas or seen this before ???? |
| 14:29.00 | [TK]D-Fender | ~book |
| 14:29.02 | jbot | book is probably a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 14:29.10 | M_at | It's the legacy PBX that adds prefixes |
| 14:29.47 | [TK]D-Fender | M_at: using * inline with it? |
| 14:30.24 | *** part/#asterisk quellhorst (n=quellhor@unaffiliated/rend) |
| 14:31.10 | M_at | [TK]D-Fender: Not yet - using it on a QSIG PRI with the INDeX's networking features. |
| 14:32.24 | M_at | It's going to be the central hub for 3 remote * boxes which need to be able to dial into the legacy PBX |
| 14:32.31 | *** join/#asterisk eonblu[ez (n=eonblu_e@63.247.107.130) |
| 14:32.54 | M_at | But I have a 2 port PRI card in there for the day it becomes the master and needs to run in-line |
| 14:32.56 | eonblu[ez | When i do a blind transfer to another extension, the other party cannot hear the person i transfer to |
| 14:33.00 | eonblu[ez | but we can hear them |
| 14:33.03 | eonblu[ez | what could be up? |
| 14:34.23 | *** join/#asterisk murf (n=steve_mu@216.166.159.235) |
| 14:40.54 | *** join/#asterisk mrc527 (n=marco@81-208-60-205.ip.fastwebnet.it) |
| 14:41.27 | mrc527 | hi all, can i ask you for help? |
| 14:41.58 | Dovid | go ahead. If some one knows a solution they will try ;) |
| 14:42.24 | mrc527 | ok, thx. |
| 14:43.56 | Dovid | mrc527: the quesion is ? |
| 14:44.14 | mrc527 | i have an existent H323 gatekeeper, to make regular calls to regular phones. Now...i have to put in this company a system to translate incoming SIP calls (from other ip systems, like 3G mobiles phones) to that H323 gateway |
| 14:44.33 | mrc527 | how can i make that? It's possible with asterisk? |
| 14:44.51 | *** join/#asterisk wmandra (n=wmandra@c-68-37-251-85.hsd1.nj.comcast.net) |
| 14:44.55 | Dovid | mrc527: i dont have much expirience with h323 but I beleive that asterisk will do it for you |
| 14:45.11 | Dovid | asterisk supports h323 |
| 14:45.25 | r0d3nt|m | Ok, well i found the problem and fix it.. apparently some of the settings changed in the conf files after the upgrade.. |
| 14:45.25 | mrc527 | ah, good, but...did you know how i can configure it? |
| 14:46.11 | Dovid | for h323 specificly or asterisk in general ? |
| 14:46.41 | mrc527 | i'm compiling it on a linux box, but after i have no idea how to make it work |
| 14:46.59 | [TK]D-Fender | M_at: How big is your legacy system? |
| 14:48.26 | mrc527 | did anyone have expirience with H323 and Asterisk? |
| 14:49.09 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 14:51.40 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 14:52.45 | Dovid | mrc527: read rhe book: Asteris: The future of telephony. u can get it here |
| 14:52.46 | Dovid | ~book |
| 14:52.47 | jbot | i guess book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 14:52.50 | Dovid | the* |
| 14:53.05 | mrc527 | mmm, ok, thx |
| 14:53.33 | mrc527 | i'm trying with the support page, but...i dont find anything about my problem:D |
| 14:54.41 | Dovid | mrc527: there is a book there in pdf formt that u should download and read |
| 14:54.47 | *** join/#asterisk xnon (n=xnon@200.82.222.85) |
| 14:54.58 | Dovid | it will teach you everything you need to know to get an asterisk box up and running |
| 14:55.44 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-42.ph.ph.cox.net) |
| 14:57.50 | *** join/#asterisk _deg_ (n=deg@201.15.217.96) |
| 14:58.12 | mrc527 | ok, thx dovid |
| 14:58.38 | Dovid | mrc527: here is another good resource |
| 14:58.40 | Dovid | ~voip-info |
| 14:58.42 | jbot | voip-info is, like, the Voice Over IP wiki. It is a community resource which will answer all of your questions, from Asterisk to ZTDummy. You can find it over at http://www.voip-info.org - well worth bookmarking |
| 15:00.27 | mrc527 | ok thanks! |
| 15:00.39 | mrc527 | i'm searching in that books now..^_^ |
| 15:00.41 | L-info | for a high-load, all-ip conference server running asterisk, am i okay to use ztdummy as a timing source (using a 2.6 kernel), or am i still better of with a digium/sangoma card as a timing source? |
| 15:00.48 | *** join/#asterisk teknoprep (n=tekon@unaffiliated/teknoprep) |
| 15:01.16 | Dovid | mrc527: better to read it thru then just look at h323. u need to understand the fundimentals of asterisk |
| 15:01.19 | [TK]D-Fender | L-info: You intend to do any IAX2 trunked connections (not just user) or MeetMe? |
| 15:01.31 | L-info | MeetMe.. no trunking |
| 15:02.20 | [TK]D-Fender | L-info: Well for a conference server I'd suggest you get a minimal card for timing. just a hardwre guarantee.... safety first, right? |
| 15:02.38 | *** join/#asterisk jake1932 (n=Administ@pool-68-236-19-153.phil.east.verizon.net) |
| 15:03.33 | jake1932 | with this new firmware update my 7960 is dialing alpha instead of numeric. anyone know how to get this back to numeric? |
| 15:03.41 | L-info | thats what i was thinking.. would be interested to hear of any results using the software-based timing source though. thanks TK |
| 15:05.25 | *** join/#asterisk X-Rob_ (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 15:10.12 | aydiosmio | ztdummy? |
| 15:24.11 | *** join/#asterisk CyberMad (n=cybermad@202.73.117.106) |
| 15:25.18 | *** join/#asterisk RoyK (n=roy@213.160.242.90) |
| 15:31.11 | *** join/#asterisk _alex_mx_ (n=alex@200.78.229.18) |
| 15:31.58 | *** join/#asterisk ms345 (n=mike_sim@64.74.198.10) |
| 15:32.17 | _alex_mx_ | hello trying to compile trunk and i'm getting the following error... |
| 15:32.22 | _alex_mx_ | chan_zap.c: In function `zap_send_keypad_facility_exec': |
| 15:32.23 | _alex_mx_ | chan_zap.c:2496: warning: implicit declaration of function `pri_keypad_facility' |
| 15:32.23 | _alex_mx_ | chan_zap.c: In function `pri_dchannel': |
| 15:32.23 | _alex_mx_ | chan_zap.c:10070: error: structure has no member named `call' |
| 15:32.23 | _alex_mx_ | make[1]: *** [chan_zap.o] Error 1 |
| 15:32.23 | _alex_mx_ | make: *** [channels] Error 2 |
| 15:32.30 | *** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk) |
| 15:32.33 | _alex_mx_ | any clues? |
| 15:33.30 | wunderkin | compile zaptel first |
| 15:33.37 | _alex_mx_ | i have |
| 15:33.51 | *** join/#asterisk Cresl1n (i=matt@nat/digium/x-40ef00f1bd965560) |
| 15:33.51 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 15:34.32 | *** part/#asterisk Rez (i=lorez@freenode/staff/lorez) |
| 15:36.04 | *** join/#asterisk xnon (n=xnon@200.82.222.85) |
| 15:36.32 | wunderkin | do a make install in zaptel and make clean and install in asterisk |
| 15:37.15 | *** join/#asterisk freebsd_fan (n=unsure@sw69x3.foto.gu.se) |
| 15:37.46 | _alex_mx_ | make clean, ./configure, make menuselect, make linux26, make install, and make config in zaptel then make clean and install in asterisk |
| 15:37.51 | mrc527 | is someone expert in h323 and asterisk? |
| 15:42.13 | jake1932 | forget it - i found it |
| 15:43.05 | mrc527 | ? |
| 15:47.48 | Juggie | _alex_mx_, 8 ball says update yuor libpri |
| 15:47.48 | olivier__ | _alex_mx_ and libpri ? |
| 15:48.30 | *** join/#asterisk saftsack (n=oliver@IP-213188106101.dialin.heagmedianet.de) |
| 15:48.32 | saftsack | hi |
| 15:48.50 | *** join/#asterisk BrokenNoze_ (n=SimonK@host86-144-75-221.range86-144.btcentralplus.com) |
| 15:49.19 | BrokenNoze_ | looks like the source is install! Wahoo! 2 days to do that before I found you guys! |
| 15:49.42 | saftsack | if i call extension 9991 with my sipphone (9991,1,Playback(on)) i can hear nothing :( |
| 15:49.53 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 15:50.25 | BrokenNoze_ | now getting a cc:Command not found. apparently missing a compiler? |
| 15:51.49 | Juggie | you sure are |
| 15:51.52 | Juggie | what distro |
| 15:52.05 | BrokenNoze_ | fc4 |
| 15:52.16 | Juggie | eugh ;) |
| 15:52.41 | BrokenNoze_ | Oh |
| 15:52.49 | Juggie | and 'whereis gcc' turns up nothing? |
| 15:52.49 | Mercestes | What 8 port fxs card solutions are there out there?? |
| 15:52.55 | Mercestes | ore more than 4 ports in general? |
| 15:52.56 | BrokenNoze_ | well I went for it coz thats what Asterisk was apparently written on |
| 15:53.04 | iq | Good Morning |
| 15:53.25 | BrokenNoze_ | No i get /usr/libexec/gcc |
| 15:53.25 | Juggie | BrokenNoze_, 'whereis gcc' turns up nothing? |
| 15:53.51 | Juggie | '/usr/libexec/gcc -v' |
| 15:54.13 | Juggie | oh |
| 15:54.18 | Juggie | thats just a golder |
| 15:54.21 | Juggie | *folder |
| 15:54.38 | BrokenNoze_ | <PROTECTED> |
| 15:54.42 | Juggie | try, 'yum install gcc' |
| 15:55.01 | saftsack | found the error. not doing answer was the fault |
| 15:55.21 | BrokenNoze_ | Doh! assumed the command was cc - tried yum install cc |
| 15:55.25 | BrokenNoze_ | didn't work. |
| 15:55.35 | BrokenNoze_ | doing now cheers |
| 15:55.53 | *** join/#asterisk ManxPower (n=manxpowe@69-2-85-41.wan.networktel.net) |
| 15:56.02 | mrc527 | hi all |
| 15:56.02 | Juggie | np |
| 15:56.12 | mrc527 | can someone help me? |
| 15:56.26 | *** join/#asterisk SplasPood (n=jwb@nat-loopback.lga4.us.corp.voxel.net) |
| 15:56.32 | mrc527 | i'm trying to make a SIP to H323 gateway |
| 15:56.33 | Juggie | your install may be short on libs too so you might run into a few more missing devlopment libs |
| 15:57.09 | mrc527 | but i dunno how to route incoming calls from SIP to the H323 gateway....someone knows how to do it? |
| 15:57.46 | *** join/#asterisk fiber0pti (n=John@207.114.199.107) |
| 15:58.14 | BrokenNoze_ | octasic-helper? |
| 15:58.36 | mrc527 | ...? |
| 15:58.43 | Juggie | ?RTFM |
| 15:58.56 | Juggie | ~RTFM |
| 15:59.03 | jbot | rtfm is probably Read The F*cking Manual (TM). It is a suggestion to do your homework before posting a question. Sometimes used as RTFM $SPECIFIC_MANUAL to refer to a specific source of information. See also http://en.wikipedia.org/wiki/RTFM |
| 15:59.03 | Juggie | jbot! |
| 16:01.20 | *** join/#asterisk Givur (n=mail@p54BCD8CA.dip.t-dialin.net) |
| 16:01.25 | Givur | Hi Everyone. |
| 16:01.51 | _alex_mx_ | Juggie, svn updated all the packages 30 min ago, something new since then? |
| 16:02.32 | Juggie | did you make; make install in the libpri dir? |
| 16:02.33 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 16:02.46 | _alex_mx_ | yes |
| 16:03.04 | ghenry | [TK]D-Fender: Agreed with the client to roll/wipe trixbox, and just go with stock * and freepbx to get back to how we were. Phew! ;-) |
| 16:03.11 | Givur | I have a little problem with Asterisk 1.2.12. We do some callings to Ireland, some regions have change the regional codes from 5 to 953, this message we get when we call with a normal phone. But when I try to call the number via Asterisk we just get a 'Congestion' back without a message. Is this something what I need to configure in Asterisk to get this message, or is this a problem of my VoIP Provider? |
| 16:04.08 | Juggie | _alex_mx_, do cd zaptel; make clean; make install; cd ../libpri; make clean; make install; cd ../asterisk; make clean; make install |
| 16:04.15 | Juggie | if that doesnt work, report back. |
| 16:07.29 | _alex_mx_ | Juggie, same error |
| 16:08.32 | _alex_mx_ | Juggie, not that it should matter but let me wipe sources and do a fresh checkout and i'll report back |
| 16:09.06 | Juggie | ok |
| 16:09.34 | Juggie | btw is this 1.2 or trunk your checking out |
| 16:10.00 | *** join/#asterisk PhinnFort (n=josteins@unaffiliated/phinnfort) |
| 16:12.37 | PhinnFort | what are the minimum requirements for running asterisk? |
| 16:13.29 | jake1932 | PhinnFort: there are several systems scaled to meet different needs |
| 16:13.51 | PhinnFort | can i run it and make it handle my calls on a pentium 1 system? |
| 16:13.56 | jake1932 | you can run it on a WRT (Linksys router) all the way up to huge powerful boxes |
| 16:14.16 | PhinnFort | i think the box has about 8 megs of ram, will it swap a lot? |
| 16:14.29 | _alex_mx_ | Juggie, 1.4 |
| 16:14.58 | PhinnFort | aster |
| 16:15.04 | jake1932 | PhinnFort: hmm, sounds pretty low powered - but I would think you could do it |
| 16:15.06 | PhinnFort | isk@home wanted 500mhz |
| 16:15.14 | PhinnFort | ill try:D |
| 16:15.24 | jake1932 | yeah - you would just do a minimal install |
| 16:15.42 | PhinnFort | ill just throw a debian unstable on it, i think |
| 16:15.44 | jake1932 | compile a slim distro and just plain asterisk |
| 16:16.00 | PhinnFort | does it need much space? |
| 16:16.05 | jake1932 | nope |
| 16:16.13 | Juggie | PhinnFort, make sure you edit modules.conf |
| 16:16.18 | Juggie | and set autoload=no |
| 16:16.23 | Juggie | and then load only the modules you need |
| 16:16.39 | PhinnFort | Juggie: thank you, ive setup some slim systems before;) |
| 16:19.18 | *** join/#asterisk Dovid (n=dovi5988@barak.cellcom.co.il) |
| 16:19.30 | Juggie | np. |
| 16:24.01 | *** join/#asterisk qdk (n=qdk@0x535eae17.boanxx9.adsl-dhcp.tele.dk) |
| 16:24.56 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 16:24.56 | *** mode/#asterisk [+o mog] by ChanServ |
| 16:30.30 | *** join/#asterisk SwedChef (n=craig@216.215.26.60) |
| 16:30.53 | SwedChef | good morning all |
| 16:31.04 | jake1932 | bork |
| 16:31.19 | eonblu[ez | bork indeed |
| 16:31.21 | SwedChef | i have customers on CenturyTel DSL in central Washington State who no longer can connect from one office to another |
| 16:31.28 | eonblu[ez | my asterisk is borked |
| 16:31.30 | SwedChef | is anyone else noticing a problem like this with CT? |
| 16:31.32 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 16:34.07 | *** join/#asterisk skirmisha (n=viki@87.126.55.7) |
| 16:34.13 | skirmisha | hello guys |
| 16:34.20 | *** join/#asterisk jarg (n=jarg@200.56.225.61) |
| 16:34.27 | *** join/#asterisk Inverted (n=Inverted@66-90-148-38.dyn.grandenetworks.net) |
| 16:34.31 | *** join/#asterisk mavior (n=Miranda@88-149-160-25.f5.ngi.it) |
| 16:34.42 | Inverted | if you have an audio file is there a good day to determine how long it is in seconds? |
| 16:34.44 | skirmisha | have little issue with asterisk send/recive msg command |
| 16:35.07 | skirmisha | how can i configure asterisk to accept text msgs and send them to the proper phone |
| 16:36.21 | Juggie | asterisk doesnt support sip instant messaging |
| 16:36.56 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 16:36.58 | skirmisha | it does |
| 16:37.38 | skirmisha | at least i can send text msg , but the problem is with reciveing |
| 16:37.53 | Juggie | i asssure you, it dos not |
| 16:37.54 | Juggie | *does |
| 16:38.36 | skirmisha | u mean it can not recive, it just send right? |
| 16:38.49 | Juggie | your sent message doesnt go anywhere |
| 16:38.54 | Juggie | look @ your asterisk console |
| 16:39.10 | skirmisha | well i have made couple of tests and sending is working |
| 16:39.16 | skirmisha | i use SendText |
| 16:39.25 | skirmisha | SendText(Message) |
| 16:39.27 | *** join/#asterisk MattB2 (n=MattB2@mail.tricycleinc.com) |
| 16:39.45 | skirmisha | that's send text msg in the open sip channel |
| 16:40.05 | Juggie | thats different then sip messaging between phones/sip devices |
| 16:40.07 | skirmisha | but the problem is when phone send that mesg how should i route it |
| 16:40.16 | MattB2 | hi all... quick question. do you know of any providers that can give me a DID to SIP/IAX that I can get paid for? ie the provider pays me for the number of seconds used on that call |
| 16:40.31 | Juggie | what dont you understand |
| 16:40.36 | Juggie | asterisk does not support sip messaging |
| 16:40.37 | MattB2 | in UK there are premium rate numbers, is there a similar thing in the US? |
| 16:41.03 | Juggie | yes, 1-900 numbers. |
| 16:41.13 | skirmisha | Juggie just to confirm |
| 16:41.30 | skirmisha | i can not send sip msg between sip phones |
| 16:41.32 | MattB2 | anyone know a 1-900 provider that terminates in sip/iax? |
| 16:41.36 | Juggie | skirmisha, no. |
| 16:41.52 | Juggie | not without a patch. its not supported natively within asterisk. |
| 16:41.57 | Juggie | http://www.voip-info.org/wiki/view/Asterisk+SIP+Messaging |
| 16:42.01 | Juggie | i'm going to lunch, bbl. |
| 16:42.25 | skirmisha | thanks |
| 16:43.13 | *** join/#asterisk DarKnesS_WolF (n=wolf@81.10.111.94) |
| 16:44.12 | *** join/#asterisk Dovid (n=Dovid@barak.cellcom.co.il) |
| 16:44.28 | *** join/#asterisk ToTo (n=ToTo@host203-49-dynamic.0-87-r.retail.telecomitalia.it) |
| 16:44.47 | Dovid | hello all |
| 16:50.45 | mavior | oh..forgot before : hi everybody |
| 16:51.02 | mavior | 'cause it's my first attemp here ;) |
| 16:51.07 | *** join/#asterisk MstlyHrmls (n=mh@66.195.193.151) |
| 16:51.48 | Dovid | lol |
| 16:56.30 | *** join/#asterisk audial (n=root@87.240.28.34) |
| 16:59.21 | eonblu[ez | when i transfer a call to another extension, sometimes outgoing audio does not work (i can hear the other side, but they cannot hear me) |
| 16:59.26 | eonblu[ez | we are using cisco phones w/ sip |
| 16:59.28 | eonblu[ez | any ideas? |
| 16:59.58 | audial | hello ! i have a trouble with setting * on my gentoo box. i get the required h323 libz installed. than i got the * from svn [1.4 ] and configure with option --with-h323. After that I 'make opt' at channels/h323 directory - the error occured.... In file included from <command line>:11:../../include/asterisk/autoconfig.h:7:32: asterisk/buildopts.h: No such file or directory |
| 17:00.26 | Dovid | eonblu[ez: are u using NAT at all ? |
| 17:06.41 | eonblu[ez | Dovid -- the phones are behind nat |
| 17:06.42 | eonblu[ez | 10.x.x.x |
| 17:06.51 | eonblu[ez | the phone system is exposed to the world |
| 17:07.12 | eonblu[ez | has ext ip |
| 17:07.13 | eonblu[ez | for now |
| 17:10.04 | *** join/#asterisk noky (n=noky@200.69.211.18) |
| 17:10.05 | noky | hi |
| 17:11.00 | noky | i have a sip channel,.. with a 'default' context... in my extensions.conf i have the default context with only a extension 's' |
| 17:11.07 | noky | must be match here ? |
| 17:11.11 | noky | in 's' extension? |
| 17:11.20 | RoyK | try _X.,1, |
| 17:11.23 | noky | because asterisk answer with a 404 Not Found (SipMessage) |
| 17:11.25 | RoyK | s doesn't match much |
| 17:11.41 | [TK]D-Fender | RoyK: Matches "s" perfectly! |
| 17:11.44 | noky | yes, it's match, but.. why doesn't match with 's' extension? |
| 17:12.02 | RoyK | noky: do you dial 's'? |
| 17:12.06 | Qwell[] | because s isn't a catchall |
| 17:12.08 | [TK]D-Fender | noky: You clearly haven't read what "s" is for in the first place. |
| 17:12.12 | noky | [TK]D-Fender: hi |
| 17:12.19 | [TK]D-Fender | RoyK: Care to do the honours? |
| 17:12.48 | noky | yes.. but it doesn't match with my sip channel, and if i try with "_X." extension .. it matchs ok.. |
| 17:12.52 | RoyK | [TK]D-Fender: what? |
| 17:13.03 | noky | and for example with my oh323 channel matchs ok with the 's' extension |
| 17:13.11 | [TK]D-Fender | RoyK: This is a typical time to see you...... |
| 17:13.11 | noky | and i'm confused |
| 17:13.13 | [TK]D-Fender | ~rtfm |
| 17:13.18 | jbot | i guess rtfm is Read The F*cking Manual (TM). It is a suggestion to do your homework before posting a question. Sometimes used as RTFM $SPECIFIC_MANUAL to refer to a specific source of information. See also http://en.wikipedia.org/wiki/RTFM |
| 17:13.21 | RoyK | :) |
| 17:13.22 | RoyK | and |
| 17:13.23 | RoyK | ~docs |
| 17:13.25 | jbot | somebody said docs was Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 17:13.25 | RoyK | ~book |
| 17:13.27 | jbot | it has been said that book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 17:13.30 | [TK]D-Fender | RoyK: EXACTLY! |
| 17:13.56 | noky | well |
| 17:14.11 | [TK]D-Fender | noky: Because either H.323 doesn't dial direct extens like SIP does, or that * told the other side to dial "s" |
| 17:21.37 | *** join/#asterisk Lucky-- (i=Lucky@adsl-71-138-74-51.dsl.irvnca.pacbell.net) |
| 17:22.03 | Lucky-- | any asterisk users that can recommend a good, cost-effective ATA adapter that will allow me to connect a dial-up timeclock to it, as well as provide for redundancy in case the inet goes down as a trunk |
| 17:25.38 | [TK]D-Fender | Lucky--: ATA + any modem-like device = trouble |
| 17:25.51 | *** join/#asterisk orangey (n=orangey@bas5-london14-1177976513.dsl.bell.ca) |
| 17:25.57 | orangey | hey all! |
| 17:25.59 | [TK]D-Fender | Lucky--: And thinking of running it over the internet = near suicide |
| 17:26.27 | orangey | I have a friend who recently became handicapped, and I'm trying to get him using a telephone via a laptop computer. |
| 17:26.32 | jake1932 | indeed - I'm a lucky survivor |
| 17:26.50 | audial | when i configure * i set --with-h323 flag, but get this lines in the output at the end of configuring: 1) checking h323.h usability... no 2) checking h323.h presence... no 3) checking for h323.h... no What does it mean? |
| 17:27.03 | orangey | I want to make it so that he can talk on the phone via the modem jack of the laptop. |
| 17:27.05 | orangey | any chance of that? |
| 17:27.21 | Qwell[] | orangey: no |
| 17:27.29 | jake1932 | the modem jack is an FXO |
| 17:27.41 | jake1932 | is expecting to talk to a CO |
| 17:28.10 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
| 17:28.10 | orangey | Qwell[]: any idea where I should be looking? |
| 17:28.17 | Lucky-- | [TK]D-Fender: what would you suggest? the timeclock deal is for maybe 1-2 uses until we finish implementing a completely different solution that gets rid of the damn timeclock, but for redundancy what would you suggest, also any good links to getting asterisk/amp to send + receive faxes properly? |
| 17:28.46 | jake1932 | orangey: an ip phone - possibly |
| 17:28.51 | [TK]D-Fender | ~amp |
| 17:28.52 | jbot | amp is, like, NOT supported here! People using it should join #freepbx (FreePBX is the new name of AMP) |
| 17:29.05 | orangey | jake1932: the issue is that at his current center, he has no internet access at all. |
| 17:29.11 | [TK]D-Fender | Lucky--: And as for faxes. Use an analog line temporarily or just crossyour fingers and pray |
| 17:29.24 | jake1932 | <PROTECTED> |
| 17:29.26 | jake1932 | how |
| 17:29.35 | jake1932 | PSTN? |
| 17:29.47 | orangey | jake1932: Does that mean 'a plug in a wall'? |
| 17:29.52 | Lucky-- | So how should i provide for 1 line redundancy to the PSTN in case the internet goes out? what would you recommend? |
| 17:29.53 | orangey | or 'the modem jack in the laptop'? |
| 17:30.20 | *** join/#asterisk aptura (n=s_a_l_e_@S010600a0c93f6f7e.vs.shawcable.net) |
| 17:31.00 | jake1932 | in other words, let's just say you could plug a phone into the laptop, how would the calls from said phone get anywhere? |
| 17:31.12 | jake1932 | (with no internet access) |
| 17:31.29 | orangey | oh. there is a telephone line going to his room. |
| 17:31.40 | orangey | What I was hoping was not to give him a telephone headset. |
| 17:31.51 | orangey | but instead to use his computer as the telephone headset (softphone) |
| 17:31.58 | Qwell[] | ..why? |
| 17:32.00 | jake1932 | oh! |
| 17:32.05 | jake1932 | that's crazy man |
| 17:32.09 | *** join/#asterisk devhen (n=devin@66.119.143.160) |
| 17:32.26 | orangey | Qwell[]: he's paralyzed. he has a phone with ridiculously small buttons in his room, and no use of his hands. |
| 17:32.34 | [TK]D-Fender | Lucky--: SPA-3102 to provide redundancy to *. Plug a big splitter at your demarc and use it for the SPA, your time-clock, and any other sensitive gear. Faxes are best left off your system on dedicated lines. |
| 17:32.47 | jake1932 | hmm |
| 17:32.49 | orangey | but he can move his arm around, so we stuck a tablet pen in there, and he can now 'mouse' with the tablet. |
| 17:33.07 | Qwell[] | Why not just get him a new phone? |
| 17:33.08 | orangey | so he can 'pick up' a telephone if it rings so long as all it takes is some clicks on his screen. |
| 17:33.20 | orangey | Qwell[]: yep. we're definitely looking at that. |
| 17:33.29 | jake1932 | that's not a soft phone per say |
| 17:33.30 | orangey | Qwell[]: but by far the simplest solution would be if he could talk over his modem. |
| 17:33.33 | Qwell[] | just get a good accessable speakerphone |
| 17:33.40 | Qwell[] | erm, I misspelled the hell out of that |
| 17:33.41 | *** join/#asterisk postel_ (n=jp@wikimedia/Postel) |
| 17:33.46 | jake1932 | get him a headest and have the computer p/u and dial |
| 17:33.58 | orangey | p/u? |
| 17:34.03 | jake1932 | (a regular analog headset) |
| 17:34.07 | jake1932 | pick up |
| 17:34.16 | orangey | oh.. yeah, we want to do that. |
| 17:34.19 | orangey | that would be more than sufficient. |
| 17:34.23 | aptura | can he put on the headset? |
| 17:34.44 | jake1932 | like what MS Outlook does - or some sort of handicap accessible app |
| 17:34.58 | orangey | aptura: no. But he doesn't mind wearing headsets for a long period - he already does to manage his music listening and movie watching. |
| 17:35.19 | jake1932 | in other words, only use the computer for dialing, and have the headset on him already |
| 17:35.55 | aptura | another option is a blue tooth ear set. I dont know the batter length of those but that is another option. |
| 17:36.29 | orangey | jake1932: right. |
| 17:36.31 | jake1932 | only the battery would die and he'd be SOL |
| 17:36.40 | jake1932 | (regarding the BT headset) |
| 17:36.42 | aptura | yea that is a issue then. |
| 17:36.55 | orangey | in windows, it's reasonably easy to find 'telephone dialers' that also double as speakerphones and the like. |
| 17:37.13 | jake1932 | orangey: yeah - just don't say softphone :o) |
| 17:37.29 | jake1932 | it's a voicemodem essentally with a good app |
| 17:37.31 | findlay | if I connect to a running server with `asterisk -r` how do I detatch from it? |
| 17:37.38 | jake1932 | exit |
| 17:37.38 | orangey | Linuxfono1 - Linuxfono is a small program (with a XForms interface) that allows you to use your modem as a speakerphone, provided that your modem has voice support. You can plug a microphone and a speaker to the modem and talk. |
| 17:37.55 | orangey | hmm. sounds ideal. |
| 17:38.07 | jake1932 | yeah - that sounds good |
| 17:38.43 | audial | make: *** [ast_h323.o] ïÛÉÂËÁ 1 |
| 17:38.52 | audial | ;\ |
| 17:39.12 | audial | what is the right way to install the * with h323 support? |
| 17:40.24 | *** join/#asterisk Deeewayne (n=dwayne@ool-44c0d56e.dyn.optonline.net) |
| 17:40.38 | audial | i can't get the 1.4 installed. i get the error without any information during compilation. =( |
| 17:40.46 | *** join/#asterisk ellisdee (n=ellisdee@69.15.174.113) |
| 17:41.03 | orangey | lord! all these phone softwares for linux are from the mid 90s. |
| 17:41.18 | *** part/#asterisk MattB2 (n=MattB2@mail.tricycleinc.com) |
| 17:41.23 | *** join/#asterisk ToTo (n=ToTo@host203-49-dynamic.0-87-r.retail.telecomitalia.it) |
| 17:41.39 | sb_mx | orangey, you could try x-lite. if im not mistaken wine supports the windows version |
| 17:42.22 | jake1932 | x-lite is a softphone |
| 17:42.44 | jake1932 | AFAIK |
| 17:43.11 | DrAk0 | where i can find the best documentation for start with asterisk, from scratch |
| 17:43.14 | sb_mx | yeah, my bad. didnt scroll up to read everything :S imma shut up now |
| 17:43.33 | jake1932 | ~docs |
| 17:43.34 | jbot | hmm... docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 17:43.48 | jake1932 | there's so much out there that works |
| 17:43.57 | jake1932 | (or that's close enough) |
| 17:44.16 | *** join/#asterisk audial (n=root@87.240.28.34) |
| 17:44.42 | DrAk0 | is there too much limitation running Asterisk on FreeBSD vs running it on Linux? |
| 17:46.11 | mog | asterisk is primarily developed on linux it should however work fine on freebsd |
| 17:47.13 | eonblu[ez | linux in a stripped down nature is probably just as stable as freebsd, unless you need to use that box for multipurpose bsd-specific things i would go w/ linux |
| 17:47.14 | *** join/#asterisk s0lid (n=jlq@210.213.198.20) |
| 17:49.33 | *** join/#asterisk teknoprep (n=tekon@unaffiliated/teknoprep) |
| 17:49.37 | teknoprep | hi all |
| 17:49.51 | teknoprep | anyone here succesfully setup gxp-2000's with intercom before ? |
| 17:50.16 | [TK]D-Fender | DrAk0: ... |
| 17:50.18 | [TK]D-Fender | ~book |
| 17:50.20 | jbot | somebody said book was a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 17:55.58 | DrAk0 | thanks |
| 17:56.02 | *** join/#asterisk oliviert (n=oliviert@194.2.122.6) |
| 17:56.39 | oliviert | hi guys |
| 17:57.06 | eonblu[ez | hola ponchito |
| 17:57.35 | eonblu[ez | mi ponchito pequeno |
| 17:57.38 | oliviert | let's say i do have 2 incoming line, i wanna know that there's a second incoming call while i'm online on the first one, what module or which config should i need/set ? |
| 17:58.06 | *** join/#asterisk techie (n=techie@c-67-182-22-174.hsd1.ca.comcast.net) |
| 17:58.13 | oliviert | and then be able on the phone to put the first one on hold, and take the 2nd line and vice versa |
| 17:58.45 | oliviert | i did try the queue module, but it does not give me any alert when there's a second line ringing |
| 17:59.10 | oliviert | it place the call on queue, and when i finish the 1st call, the second then ring me |
| 17:59.34 | oliviert | but i even don't know that they were a 2nd call in the queue |
| 18:00.06 | oliviert | any idea about the config to set up or the module to use, perharps a kind of dial plan config. i don't know |
| 18:00.33 | *** join/#asterisk justinu|laptop (n=Justin@12.44.122.130) |
| 18:03.51 | [TK]D-Fender | oliviert: Ok, lets start over a little bit. Describe these 2 lines. Then tell us what kind of phone you are using. |
| 18:06.21 | oliviert | [TK]D-Fender: the 2 lines are zaptel (analog lines) |
| 18:06.30 | oliviert | the phones are thomson ST 2030 |
| 18:06.36 | oliviert | 2 phones |
| 18:06.53 | oliviert | i did test queue and ring group module |
| 18:07.02 | oliviert | they are working fine |
| 18:07.17 | oliviert | but i missed the feature that i explain above |
| 18:07.51 | oliviert | in reality there's 4 zaptel lines |
| 18:07.53 | [TK]D-Fender | oliviert: Well I guess you should be able to simply send it another call it it should either give you a call-waiting style beep, flash another line kety or something like that. |
| 18:08.23 | oliviert | yes, in fact is what i need |
| 18:08.23 | *** join/#asterisk toxap (i=toxap@194.187.128.88) |
| 18:08.27 | [TK]D-Fender | oliviert: Analog call -waiting and * do not mix. So forget the idea that you can count those as 4 lines instead of 2. |
| 18:08.49 | [TK]D-Fender | oliviert: The rest is up to your phone config and there is nothing to be done in *. |
| 18:09.11 | oliviert | well is one phone number and 4 lines grouped under it, and all are analog |
| 18:09.11 | [TK]D-Fender | oliviert: If your phone is set up to accept multiple simultaneous calls then you can just ring it whenever you want and all is good. |
| 18:09.29 | *** join/#asterisk Ebola (n=Ebola@host86-138-123-50.range86-138.btcentralplus.com) |
| 18:09.35 | oliviert | ok, then you think that i missed something in the phone config |
| 18:09.39 | [TK]D-Fender | oliviert: AH, so 4 independant copper pairs with a hunt group associated with it? |
| 18:09.48 | oliviert | yes |
| 18:09.54 | [TK]D-Fender | oliviert: Yes, if its to be fixed, thats where |
| 18:10.11 | [TK]D-Fender | oliviert: Then youi lines are fine. I take it you have a TDM04B or A200? |
| 18:10.26 | oliviert | ok, and with X-lite, i do have the same problem, no simultaneous call |
| 18:10.35 | oliviert | is there also something to config on it ? |
| 18:10.44 | oliviert | TDM04B |
| 18:10.50 | [TK]D-Fender | oliviert: show me your dialplan. |
| 18:11.00 | [TK]D-Fender | www.pastebin.ca |
| 18:11.06 | DrAk0 | ive been having a lot of problem with akiga... is there other nice linux client? |
| 18:11.38 | oliviert | well, the dialplan is a maze, due to the fact that i did use trixbox and freepbx at the beginning to set up the server faster, but while now i'm hacking in the conf file... |
| 18:12.12 | oliviert | but i could paste it anyway, but it's not so trivial, there's a bunch of default routine |
| 18:12.26 | [TK]D-Fender | oliviert: Tell you what. paste it up and we'll see from there |
| 18:12.45 | [TK]D-Fender | DrAk0: linphone, kphone are supposedly ok. |
| 18:12.56 | teknoprep | hmmm |
| 18:13.07 | teknoprep | any how-to on setup of gxp-2000's and Intercom ? |
| 18:13.33 | teknoprep | i have read http://www.grandstream.com.cn/download/other/FAQ_and_Example_for_Asterisk_Configuration_for_GXP-2000.pdf#search=%22Asterisk%20BLF%20key%22 ... not so well written |
| 18:14.24 | *** join/#asterisk pifiu (n=someone@216.5.79.1) |
| 18:14.27 | oliviert | here is the extensions.conf, i'll paste the additional http://pastebin.ca/199701 |
| 18:15.17 | *** join/#asterisk tdonahue (i=tdonahue@207.138.151.58) |
| 18:15.30 | tdonahue | hi all |
| 18:16.22 | *** part/#asterisk ms345 (n=mike_sim@64.74.198.10) |
| 18:16.26 | oliviert | http://pastebin.ca/199707 extensions-additional |
| 18:16.50 | oliviert | the extenstions number that are active are 101 to 104 |
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| 18:17.22 | *** part/#asterisk icepack (n=icepack@207.190.248.178) |
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| 18:18.40 | *** mode/#asterisk [+o Corydon-w] by ChanServ |
| 18:19.58 | [TK]D-Fender | oliviert: Ok, now place a call to that phone, then attempt a 2nd while its on the first call. Pastebin the CLI output of the ENTIRE call. |
| 18:20.32 | oliviert | ok |
| 18:21.52 | *** join/#asterisk oej_ (n=oej@apollo.webway.se) |
| 18:22.13 | aptura | [TK]D-Fender a day ago could not get incomming or outgoing calls on my second line but managed to fix the incomming call issue. Just not sure how to remedy the outgoing issue with the sip provider and line2 |
| 18:23.08 | [TK]D-Fender | aptura: I'd have to see your configs. When I get hom in 2.5 hours I can look for you. |
| 18:23.33 | *** join/#asterisk Tili (n=tili@202.133.67.33) |
| 18:23.50 | aptura | Good could you email me the result then? |
| 18:24.00 | aptura | I probebly wont be around. |
| 18:24.05 | aptura | At least here. |
| 18:25.09 | audial | what is the right way to go through h323 to sip? |
| 18:27.03 | audial | i mean converting |
| 18:27.32 | *** join/#asterisk yogurt2ungue (n=yogurt2u@24-48-231-201.fibertel.com.ar) |
| 18:28.11 | oliviert | Extension 101 has call waiting disabled |
| 18:28.14 | oliviert | [TK]D-Fender: ok,i did it, with debug and verbose=0, and the pb is : |
| 18:28.16 | oliviert | Extension 101 has call waiting disabled |
| 18:28.34 | [TK]D-Fender | I need verbose 10 |
| 18:28.40 | oliviert | then you were right, call waiting is disbled on the extension |
| 18:28.41 | [TK]D-Fender | oliviert: www.pastebin.ca |
| 18:28.57 | oliviert | i did it again with 10 verb and debuf |
| 18:29.01 | [TK]D-Fender | oliviert: I'm not sure you can trust that statment. I want to see it. |
| 18:29.14 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:29.35 | blitzrage | who in here was having problems with Exec() no parsing the commas correctly the other day? |
| 18:30.00 | blitzrage | What I'm doing is putting them into a variable escaped, then just passing the variable to Exec() |
| 18:31.21 | oliviert | ouf the output is too big, and scrolling too fast, i lost the beginning of the log output; is there a way under CLI to send it to a file ? |
| 18:31.45 | aptura | [TK]D-Fender msg you the config info? |
| 18:31.48 | blitzrage | oliviert: asterisk -rvvv | tee /tmp/mylogfile.txt |
| 18:32.14 | *** join/#asterisk psfax (n=GusTavo2@200.49.156.83) |
| 18:32.20 | psfax | hola... alguien que me pueda ayudar ? |
| 18:33.35 | [TK]D-Fender | aptura: You can e-mail them or send me root |
| 18:35.52 | aptura | psfax, hola. no ask..ask justos la pregunta. No hablo español y la mayoría de la otra gente aquí no habla español en lugar de otro que estoy utilizando un Web site que pueda traducir su español al inglés. Http://babelfish.altavista.com goto y entonces hacen su pregunta en inglés. |
| 18:36.20 | aptura | [TK]D-Fender email them? |
| 18:36.29 | aptura | Who is them |
| 18:37.03 | psfax | i've a problem with tdm2400p analog... when I make one llamda telefonica through the PSTN and a person takes care of this call... this does not listen to but I to me if and after seconds the call is cut single |
| 18:37.39 | *** join/#asterisk justinu|laptop (n=Justin@12.44.122.130) |
| 18:38.42 | psfax | when i make a call phone to PSTN and an other person takes this call ... this person can't listen to me... but i yes listen to hem |
| 18:39.06 | psfax | and... later of a time... the call is hangup with de asterisk |
| 18:39.15 | psfax | k |
| 18:39.16 | psfax | <psfax> - |
| 18:39.28 | psfax | <PROTECTED> |
| 18:39.32 | psfax | -Hangup |
| 18:39.36 | aptura | no |
| 18:39.41 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:40.05 | oliviert | [TK]D-Fender: http://pastebin.ca/199743 |
| 18:40.07 | aptura | No exhiba su cli hecho salir aquí. Póngalo en pastebin.ca |
| 18:40.10 | oliviert | the log |
| 18:40.44 | psfax | aptura you understend me ? i'm sorry but my english is very poor :( |
| 18:40.51 | aptura | yes |
| 18:41.16 | aptura | it makes enough sence. Unfortunaly I need to go but it sounds as if your having NAT issues. |
| 18:41.35 | *** join/#asterisk anthonyl (i=anthony@nat/digium/x-6cc6c94e85001666) |
| 18:42.57 | psfax | and... when a person call me to a sip internal from pstn i can listen and hem can listen to me perfect |
| 18:43.09 | psfax | the problem is when i open the channel |
| 18:44.13 | psfax | the asterisk or the hardware not detect when the call is takes care |
| 18:49.07 | *** join/#asterisk Jason99 (n=jason@jason.unitz.ca) |
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| 19:15.02 | sp0n9e- | is there any way to easily remove all the sangoma driver modules and asterisk modules so i can just have a "do-over"? i upgraded my kernel and wanrouter broke |
| 19:15.27 | *** join/#asterisk vodka_liscia (n=data@213-140-17-110.ip.fastwebnet.it) |
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| 19:18.18 | *** join/#asterisk eliXier (i=GTI16V@gti.twice-irc.de) |
| 19:18.40 | gambolputty | Hi. When I make an outgoing SIP URL call with *, I can transfer and hold with no problems, but when I get an incoming SIP URL call, any transfer or hold attempts drop the call. Any ideas on how to fix this? |
| 19:21.22 | *** join/#asterisk Dr-Linux|work (n=Linux@202.59.73.131) |
| 19:21.50 | Dr-Linux|work | hi all |
| 19:22.17 | mog | hi |
| 19:22.20 | gambolputty | hi |
| 19:22.54 | Dr-Linux|work | can we send/recieve SMS's on PRI T1 ? |
| 19:31.46 | jo3sm1th | Hi!! I have a project: I dont have my own PBX and all I need is one simple Direct Inward Dial telephone# that will go to a menu (with recordings I already have saying Press 1 for Sales, Press 2 for Customer Service, etc... and all those extensions will ring to 2 or more different cellular phone#'s can anyone do this or recommend a service |
| 19:32.34 | *** join/#asterisk h3x0r (n=hex@ip68-224-236-92.lv.lv.cox.net) |
| 19:34.54 | Secesh | gambolputty: For your inbound call options, do you have "t" (lowercase) |
| 19:35.23 | gambolputty | don't think so |
| 19:35.39 | gambolputty | I only want my phones to be able to transfer calls, not the inbound caller |
| 19:36.35 | gambolputty | in turn, that t option looks like it is for analog phones only |
| 19:36.47 | Secesh | Right, well, usage of T vs t refers to Caller vs callee (respectivly) -- so when you receive a call, you need t, while when you place a call, you need T |
| 19:37.08 | gambolputty | my SIP phones already have transfer and hold buttons and functions |
| 19:37.37 | gambolputty | the use of the t feature would put transfer functions on one of the 12 standard keypad keys |
| 19:37.39 | Secesh | right, well, you need to include these options in your "extensions.conf" file |
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| 19:40.28 | gambolputty | It does no good at all, an incoming SIP URL still gets hung up with I put the call on hold with that "t" option in the dial command. |
| 19:40.36 | gambolputty | when |
| 19:41.03 | *** join/#asterisk Blackthorn (i=blacktho@72.236.88.10) |
| 19:41.11 | psfax | some body can helpme... i have a problem with tdm2400p analog... when i make a phone call to pstn phone a person take care this call and not listen to me... and later of a time the asterisk hangup de call and send to mi cli this message |
| 19:41.26 | *** join/#asterisk freebsd_fan (n=unsure@200.32.71.142) |
| 19:41.28 | psfax | <PROTECTED> |
| 19:41.39 | psfax | -Hangup |
| 19:43.04 | Blackthorn | Hello, just decided to update asterisk from 1.2.7.1 to 1.2.12.1 and trying to comple I get an eror :chan_zap.c:9026: error: structure has no member named `call' make[1]: *** [chan_zap.o] Error 1 Do you have any suggestions on what I have done wrong? |
| 19:43.41 | Qwell[] | Blackthorn: upgrade zaptel? |
| 19:43.56 | Qwell[] | it was either zaptel or libpri...I'm thinking libpri |
| 19:44.17 | *** join/#asterisk freebsd_fan (n=unsure@sw69x3.foto.gu.se) |
| 19:45.14 | Blackthorn | just did that as well before the asterisk compile. thats what i had been thinking as well. how do i verify the version of libpri and zaptel btw? just to make sure they did upgrade. |
| 19:45.30 | *** part/#asterisk Secesh (n=matt@adsl-070-155-122-032.sip.asm.bellsouth.net) |
| 19:47.55 | *** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
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| 20:04.22 | *** part/#asterisk sp0n9e- (n=sp0n9e@phpurge.com) |
| 20:04.37 | Blackthorn | how do you verify the version of zaptel and libpri that is running on the system? |
| 20:05.58 | *** join/#asterisk thestor (n=thestor@CPE-24-94-241-227.wi.res.rr.com) |
| 20:06.46 | teknoprep | is it possible for Parking Calls BLF ? |
| 20:06.52 | *** join/#asterisk thestor (n=thestor@CPE-24-94-241-227.wi.res.rr.com) |
| 20:06.54 | teknoprep | is there a patch for 1.2 ? |
| 20:06.55 | thestor | i'm trying out trixbox in vmware, i ran netconfig, but it won't save any of the information, if i run netconfig again all the fields are empty, what's going on? |
| 20:07.12 | teknoprep | thestor please join #freepbx for trixbox related questions |
| 20:07.21 | thestor | kk, my bad |
| 20:14.41 | dlynes_laptop | teknoprep: yeah, do a search on bugs.digium.com...there was a patch called metermaid that was committed back in May or something that handled that, and then a new and improved patch called metermaid2 in July or something |
| 20:15.44 | dlynes_laptop | Blackthorn: use the strings command...you should be able to extract some strings from both files that'll give you some idea as to what version it is |
| 20:17.22 | *** join/#asterisk robin_sz (n=you@adsl.redpoint.org.uk) |
| 20:18.00 | teknoprep | dlynes_laptop, i was looking at that.. how would i install it... there are no install instructions |
| 20:18.36 | dlynes_laptop | teknoprep: it's a patch...you patch it against the version of the source code it's written for, or you can look at the diff file, and apply the diff, manually |
| 20:18.48 | teknoprep | hmmm |
| 20:18.50 | dlynes_laptop | teknoprep: type 'man patch' for more information on patching |
| 20:19.03 | teknoprep | would the diff patch require a remake of asterisk ? |
| 20:19.09 | dlynes_laptop | teknoprep: of course |
| 20:19.15 | teknoprep | damn |
| 20:19.40 | dlynes_laptop | It's not a new module |
| 20:19.40 | dlynes_laptop | it's just patching the existing modules |
| 20:21.33 | teknoprep | dlynes_laptop, i found instructions.. only thing i can't find is the metermaid-1.2.7.1.txt |
| 20:21.54 | dlynes_laptop | teknoprep: look through the bug report |
| 20:22.02 | dlynes_laptop | teknoprep: it'll be a downloadable link in one of the messages |
| 20:22.15 | teknoprep | is there a search on that site ? |
| 20:22.23 | teknoprep | i only see a jump and it gives me an error if i search it |
| 20:22.28 | dlynes_laptop | teknoprep: well, you've already got the bug report up, right? |
| 20:22.37 | teknoprep | yup |
| 20:22.44 | dlynes_laptop | i.e. the one that mentions metermaid-1.2.7.1.txt? |
| 20:22.50 | teknoprep | oh no |
| 20:22.52 | teknoprep | can't find it |
| 20:23.07 | dlynes_laptop | then how'd you know what the filename was? |
| 20:23.29 | teknoprep | i found an install how-to |
| 20:23.33 | dlynes_laptop | ah |
| 20:23.40 | teknoprep | but the only thing it didn't give me was a link |
| 20:25.45 | dlynes_laptop | http://svn.digium.com/view/asterisk/team/oej/multiparking/ |
| 20:25.51 | teknoprep | bugs.digium.com/view.php?id=5779 |
| 20:25.57 | teknoprep | hmmm |
| 20:26.00 | dlynes_laptop | erm wait |
| 20:26.01 | dlynes_laptop | nvm |
| 20:26.02 | dlynes_laptop | that's not it |
| 20:26.05 | teknoprep | i found it |
| 20:26.11 | Blackthorn | dlynes.. thankk you for your reply.. but i just don't know what you mean by using the string command |
| 20:26.21 | dlynes_laptop | Blackthorn: 'strings' command |
| 20:26.29 | dlynes_laptop | Blackthorn: it's a linux/unix command line program |
| 20:26.40 | dlynes_laptop | Blackthorn: it dumps all ascii strings from a binary file |
| 20:28.06 | Blackthorn | umm. i understnd what strings is now. i'm assuming i should strings filename. and that file name would be the location of libpri and zaptel files |
| 20:28.26 | *** join/#asterisk [Airwolf] (n=airwolf@53536CE1.cable.casema.nl) |
| 20:28.34 | dlynes_laptop | Blackthorn: like strings /lib/modules/`uname -r`/misc/zaptel.ko |
| 20:29.18 | *** join/#asterisk mitcheloc (n=mitchelo@titaniumsoft.net) |
| 20:29.22 | dlynes_laptop | Blackthorn: or strings /usr/lib/libpri.a |
| 20:29.33 | dlynes_laptop | heya mitch |
| 20:29.43 | Blackthorn | gotach |
| 20:29.45 | dlynes_laptop | long time, no see :) |
| 20:30.22 | intralanman | anyone ever had any problems with LIMIT_PLAYAUDIO_CALLER and LIMIT_PLAYAUDIO_CALLEE not controlling the limit sound on a dial? |
| 20:32.03 | dlynes_laptop | teknoprep: grab oej's test-this-branch |
| 20:32.12 | dlynes_laptop | teknoprep: the metermaid2 patch has been integrated into that branch |
| 20:32.36 | teknoprep | ? |
| 20:32.44 | teknoprep | i run trixbox |
| 20:32.59 | teknoprep | i was just going to build asterisk for the version i am running |
| 20:33.08 | teknoprep | seems quite easy |
| 20:33.24 | dlynes_laptop | teknoprep: ah....metermaid's been merged into multiparking, so there's no separate branch for it now |
| 20:34.03 | dlynes_laptop | teknoprep: yeah...there used to be a patch for it, but it seems to have disappeared now...i'm going to check one other thing...it could be that i've got closed reports filtered out |
| 20:35.18 | teknoprep | dlynes_laptop, http://bugs.digium.com/view.php?id=5779 |
| 20:35.29 | dlynes_laptop | ah...here it is |
| 20:35.37 | dlynes_laptop | yeah...that's the one i just found :0 |
| 20:35.42 | teknoprep | lol |
| 20:35.46 | teknoprep | i pasted that back a bit further |
| 20:35.51 | teknoprep | i had already found it |
| 20:35.56 | dlynes_laptop | You see that file that you can download there, then? |
| 20:35.59 | teknoprep | and told you i did... sorry i didn't indicate your name |
| 20:36.11 | teknoprep | scroll up a bit |
| 20:36.16 | teknoprep | you must have missed what i said |
| 20:36.43 | dlynes_laptop | yeah...must've |
| 20:36.50 | teknoprep | its right underneath your first link you posted |
| 20:37.00 | dlynes_laptop | yeah...i see tha tnow |
| 20:41.04 | robin_sz | tzafrir: so where exactly is the home of you rapid distro stuff? is the xorcom 1.1 stuff the latest release? |
| 20:41.09 | dlynes_laptop | btw, has anyone encountered really crappy voice quality using sipura 2000's hooked up to analog phones, going over cable internet to a pri? |
| 20:41.52 | robin_sz | dlynes_laptop: my sipura 2102 has tx audio that sounds like it has 50Hz hum all over it. |
| 20:42.04 | dlynes_laptop | cool |
| 20:42.14 | dlynes_laptop | no idea what 50Hz hum sounds like, but... |
| 20:42.15 | robin_sz | and a configuration system more complicated than the space shuttle |
| 20:42.31 | dlynes_laptop | i'm guessing it sounds similar to 60Hz hum, though |
| 20:42.45 | robin_sz | like that, but slower |
| 20:43.07 | robin_sz | I think the SIP packet rate is 20ms, so its probably an encoding thing |
| 20:43.29 | dlynes_laptop | yeah, for me it sounds like shizzit, no matter what codec i'm using |
| 20:43.34 | robin_sz | overall, the Sipura ATA has been disappointing |
| 20:43.43 | dlynes_laptop | I thought it was just me |
| 20:43.50 | robin_sz | nearly as bad as my grandstream GXP2000 |
| 20:43.50 | dlynes_laptop | Everyone else seems to rave about them |
| 20:43.56 | robin_sz | really? |
| 20:44.03 | dlynes_laptop | the grandstream bt-102 is a huge pos, too |
| 20:44.15 | dlynes_laptop | well, for what you pay for it, i guess it's pretty good |
| 20:44.19 | dlynes_laptop | but the volume is quite low on it |
| 20:44.34 | robin_sz | the Sip ATA is way WAY too overcomplicated for someting ttrying to connect a simple phone to a sip connection |
| 20:44.41 | *** part/#asterisk hrmmm (i=blake@tata.cluebie.net) |
| 20:45.23 | dlynes_laptop | robin_sz: well, besides...we've had countless hardware problems with the sipuras |
| 20:45.36 | robin_sz | and it it fails to register on the first call, and you get a 'not authorised' error, but second call is fine |
| 20:45.48 | robin_sz | sad isnt it |
| 20:45.53 | aydiosmio | robin_sz: your speech is way too overredundant |
| 20:46.16 | dlynes_laptop | robin_sz: everything from a loud screech on it that never goes away to it not being able to grab dynamic ip addresses to it not registering on the 2nd channel to ... |
| 20:46.32 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 20:46.44 | robin_sz | dlynes_laptop: sounds similar. I think I am am magnet for shitty VOIP hardware :( |
| 20:46.53 | dlynes_laptop | same here :p |
| 20:47.07 | dlynes_laptop | my boss bought 100 pcs of this one crappy voip phone |
| 20:47.15 | dlynes_laptop | I've still got 60 pcs i'm trying to unload |
| 20:47.22 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
| 20:47.33 | robin_sz | dlynes_laptop: I have a few GXP2000s, some Zyxel WiFi phones .. and now this Sipura ATA |
| 20:47.48 | Blackthorn | i've deployed spa-2000 and they have good sounds and no 60Hz noise. This batch of spa-2102 has good sound if you ignor the 60hz humm. Friend said it was because they were using cheap powersupplies. |
| 20:48.14 | robin_sz | Blackthorn: nah, its not from the PSU |
| 20:48.57 | dlynes_laptop | I frankly don't know how yate, freeswitch, bayonne, et al depend on sipura units |
| 20:49.10 | robin_sz | weird |
| 20:49.30 | robin_sz | Blackthorn: got a working config file for asterisk and a 2102? |
| 20:49.45 | *** join/#asterisk diclophis-work (n=jbardin@65.203.37.58) |
| 20:50.11 | Blackthorn | i didn't have to do anyting special to get the 2102 to work. just normal sip.conf files no change from spa-2000 |
| 20:50.30 | robin_sz | Blackthorn: min fails to notice asterisk has closed the call, fails to notify asterisk the phone has gone on hook, sounds carp, fails to register correctly first time out |
| 20:50.58 | Blackthorn | that is if your using ulaw. havn't goten it to work with anything but that yet. which is why I just got the g729 codec's and going to try to install those tommorw. |
| 20:51.21 | robin_sz | Blackthorn: I think mine is still confused about working just on an internal network ... it seems to think it wants to be a router too |
| 20:51.35 | Blackthorn | oh yEA! |
| 20:51.41 | robin_sz | obviously, we have nothing connected to the PC port on it |
| 20:51.57 | robin_sz | just the "internet" port |
| 20:52.16 | Blackthorn | i never could get it to work from the ethernet port. I got it to work by connection the spa to a computer, setting the wan to dhcp, turning on the web interface on port 8080. then pluging wan into the hub. |
| 20:52.30 | robin_sz | yeah, |
| 20:52.36 | robin_sz | thats what we did |
| 20:52.36 | Blackthorn | and removing the ethernet side. |
| 20:52.42 | robin_sz | yeah |
| 20:52.49 | robin_sz | poxy thing |
| 20:53.13 | Blackthorn | the instructions say "never connect the wan port to a switch/hub" but I never could get the ethernet side to connect to anything but a PC (even with cross-over cables) |
| 20:53.54 | robin_sz | ours seems to occasionally respond as a DHCP server on the WAN side, completely buggering up the real DHCP server |
| 20:54.06 | Blackthorn | yes you have to turn that off |
| 20:54.24 | robin_sz | we've ip boxed it for now |
| 20:55.02 | robin_sz | ie, put it in a cardboard box until it learns to play nicely on the network |
| 20:55.06 | Blackthorn | hook your pc to ethernet, set your wan to dhcp (should be by default) turn off the dhcp server, turn on the web monitoring port. set your * ip, username, password. save. unplug unit. plug ethercable via wan to switch and it will come up. |
| 20:55.50 | dlynes_laptop | anyways...gotta run |
| 20:56.05 | robin_sz | i tink I didnt tunr off the DHCP server ... but even so, it should only respond the on the PC side |
| 20:56.13 | Blackthorn | me too, gota meetting across down in 30 minutes. |
| 20:56.23 | robin_sz | it shouldnt leak out onto the wan side ... |
| 20:56.26 | robin_sz | ok, cya |
| 20:57.05 | robin_sz | foollishly, we bought another ATA186 today as we had one of those already that works perfectly ... |
| 20:57.14 | robin_sz | another mistake. |
| 20:57.47 | robin_sz | the first one was configured for SIP. this one seems to be configured for SCCP |
| 20:58.25 | benjk | they do SS7 now on those ATAs? |
| 20:58.28 | benjk | impressive |
| 20:58.30 | benjk | :) |
| 20:58.44 | robin_sz | SS7? |
| 20:59.02 | benjk | SCCP |
| 20:59.09 | robin_sz | apparently |
| 20:59.17 | benjk | Signaling Connection Control Part |
| 20:59.38 | robin_sz | it seems it has to correctly register over SCCp before you can switch it over to SIP |
| 20:59.51 | benjk | different from Cisco's Skinny Client Control Protocol, though |
| 20:59.57 | benjk | I was just pulling your leg |
| 21:00.10 | file | <PROTECTED> |
| 21:00.12 | file | gah |
| 21:00.35 | robin_sz | so, overall today was another day of voip hell :) |
| 21:01.15 | schirpich | as my russian asterisk friend says, "asterisk,,, good! dialplan... not so good" |
| 21:01.35 | robin_sz | weird ... |
| 21:01.45 | robin_sz | after all, you write the dialplan yourself |
| 21:02.11 | schirpich | pebcak :) |
| 21:02.27 | robin_sz | oh and .. thats another weird ting .. this Sipura SPA2102 has a in built dialplan .. like whats that all about, ... |
| 21:02.48 | benjk | its a very primitive dialplan |
| 21:02.58 | robin_sz | undocumented really too. it has "heres the dialplan string useful in the USA" .. and you are supposed to work the syntax out all by youself after that |
| 21:03.04 | *** part/#asterisk jmls (n=asterisk@host81-159-167-80.range81-159.btcentralplus.com) |
| 21:03.08 | benjk | heh |
| 21:03.15 | robin_sz | what I want is this: |
| 21:03.17 | benjk | you need to download the user manuad PDF |
| 21:03.23 | robin_sz | oh I did ... |
| 21:03.30 | robin_sz | what I want is this: |
| 21:03.42 | robin_sz | I dial digits, you pass them to * |
| 21:03.47 | robin_sz | seems reasonable to me. |
| 21:04.56 | robin_sz | anyway, while it still has an audio buzz louder than the voice .. its not a lot of use really |
| 21:05.14 | robin_sz | sorry, whining over. |
| 21:08.53 | *** join/#asterisk ToTo (n=ToTo@host203-49-dynamic.0-87-r.retail.telecomitalia.it) |
| 21:11.50 | *** join/#asterisk osiris (n=osiris@c-71-205-27-131.hsd1.mi.comcast.net) |
| 21:12.22 | [TK]D-Fender | robin_sz: X.T|*.T|#.T |
| 21:12.39 | *** join/#asterisk Jamez^7 (n=martini@modemcable131.214-131-66.mc.videotron.ca) |
| 21:12.50 | [TK]D-Fender | robin_sz: Thats the "STFU and take whatever I feel like giving you" dialplan |
| 21:18.06 | *** join/#asterisk Givemelove (n=foo@208.57.229.162) |
| 21:23.28 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 21:24.59 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 21:30.36 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 21:32.54 | *** join/#asterisk CrazyTux (n=CrazyTux@adsl-75-42-28-214.dsl.hstntx.sbcglobal.net) |
| 21:33.03 | Givemelove | Hey there |
| 21:33.06 | Givemelove | quick question |
| 21:33.22 | Givemelove | is there a way to override the callerID at the zaptel/zapata level? |
| 21:33.32 | mavior | People,can anyone report successful installation of astlinux on ,say pentium 1 minor to 166Mhz , and can report the quality of the service, or it's better trying to do a fresh and minimal installation of debian with only the things needed like reported here http://www.voip-info.org/wiki/view/Asterisk+setup+minimum ? |
| 21:33.37 | Givemelove | and not use the Set(CallerID) |
| 21:34.39 | *** join/#asterisk blebleble (i=godie@caesar.godie.net) |
| 21:36.04 | *** join/#asterisk Muggl (n=oh@p54B106AC.dip0.t-ipconnect.de) |
| 21:37.08 | Corydon-w | Yikes, 166MHz? |
| 21:37.39 | Corydon-w | You might be able to do 1 or 2 on that, with minor transcoding |
| 21:37.51 | Muggl | When I pick up an ISDN phone and dial out, is Asterisk able to convert it internally to IP and dial out as a VOIP call or does ISDN calls stay ISDN calls ? |
| 21:37.58 | mavior | i have one old pentium 1 150 Mhz , 16mb ram and 1.2 gb hdd |
| 21:38.26 | mavior | in my garage and an expensive phone bill :) |
| 21:38.39 | Corydon-w | Sorry, that's not even enough RAM |
| 21:38.55 | mavior | how much at least? |
| 21:39.03 | Corydon-w | You might manage with 96, but I wouldn't generally try with less than 128 |
| 21:40.00 | Corydon-w | None of the modern installers will even load without at least 64MB RAM |
| 21:40.22 | mavior | i only need to manage out 2/3 phone lines and i guess it will never or likely never need a concurrent lines situation |
| 21:40.34 | Givemelove | still mavior |
| 21:40.50 | Givemelove | this is not enough to handle the linux system and the asterisk platform |
| 21:41.14 | Corydon-w | I've run a 200 MMX for 24 channels with no transcoding, but I'm sure you'll want some transcoding |
| 21:41.19 | Pj_ | gentoo ! |
| 21:41.23 | mavior | mmhhh...but seems that for example a text debian installation it's possible on oldest machines |
| 21:41.23 | Pj_ | :D |
| 21:41.35 | Pj_ | mavior: just try it |
| 21:41.35 | Corydon-w | and that 200 MMX was doing hardly any call processing either |
| 21:41.37 | Pj_ | it's free |
| 21:42.24 | mavior | Corydon: i'm pretty new to asterisk so i don't know the different w or w/o transcoding |
| 21:42.47 | mavior | pj , do you think debian it's not enough ? |
| 21:42.47 | mavior | :P |
| 21:43.13 | Pj_ | I think that it won't work |
| 21:43.18 | Pj_ | but you won't know till you try |
| 21:43.32 | Pj_ | and I think if it doesn't work you might want to try a super optimized gentoo system |
| 21:43.33 | mavior | ahermm.. http://www.clevelandlug.net/modules.php?op=modload&name=News&file=article&sid=56&mode=thread&order=0&thold=0 |
| 21:43.43 | Pj_ | which will probably takes you ages to set up if you never did it |
| 21:43.57 | Pj_ | and will be only slightly faster anyway |
| 21:44.01 | mavior | seems someone here in the chan was yet successful |
| 21:45.24 | Pj_ | why don't you just try it ? |
| 21:45.32 | Pj_ | it will take you a couple hour |
| 21:46.10 | mavior | cause at the moment i can't....i'm not at home , so just for give myself an idea and hear other experiences.... |
| 21:47.02 | mavior | ...and to know someone's experience on the reliability and quality of the service with this minimal hardware |
| 21:47.28 | mavior | just to know if i will have to buy some ram or proc... |
| 21:47.29 | Corydon-w | I'm running a single channel system on a 233 with 128MB RAM |
| 21:47.53 | Corydon-w | and that's fine |
| 21:48.04 | mavior | only one line? |
| 21:48.16 | Corydon-w | It's my home answering machine |
| 21:49.13 | mavior | mmhh..what's the software on the machine? |
| 21:49.23 | mavior | i mean the OS |
| 21:49.41 | Corydon-w | Slackware |
| 21:49.58 | mavior | anyone tried astlinux ? |
| 21:51.05 | mavior | or can report some infos about the quality of it? |
| 21:51.43 | L|NUX | when i try to load rxfax module |
| 21:51.47 | L|NUX | i got this error |
| 21:51.47 | L|NUX | Oct 12 16:31:43 WARNING[27903]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/app_txfax.so: undefined symbol: t30_get_far_ident |
| 21:51.53 | L|NUX | its for txfax |
| 21:51.56 | L|NUX | but same for rxfax |
| 21:51.57 | L|NUX | what to do |
| 21:52.24 | Corydon-w | L|NUX: did you install the version of spandsp that comes with it? |
| 21:52.27 | Muggl | Sorry guys, I've got a question regarding ISDN phones.. Can I use an ISDN phone with Asterisk to place a VOIP call? Or do I have to get myself an IP phone? |
| 21:53.00 | Corydon-w | Muggl: you'll need something to interface that phone into the Asterisk system |
| 21:53.48 | Muggl | Corydon-w: ok, will an ISDN card with Netcologne chipset do it? |
| 21:53.49 | *** join/#asterisk ManxPower (n=manxpowe@69-2-85-41.wan.networktel.net) |
| 21:53.50 | *** join/#asterisk Tall-guy (n=noway@207-195-103-110.regn.static.sasknet.sk.ca) |
| 21:54.03 | L|NUX | Corydon-w : yup |
| 21:54.28 | Corydon-w | L|NUX: best bet is to ask the maintainer |
| 21:54.34 | L|NUX | ok |
| 21:54.37 | L|NUX | thanks |
| 21:55.01 | Corydon-w | Muggl: it's really a question of whether it will work with misdn in the kernel |
| 21:57.17 | Muggl | Corydon-w: ok, my problem is that I have dozens of ISDN phones and I want to setup VOIP without having to buy dozens of new VOIP capable IP phones but just keep the ISDN phones. And I don't know if this can be realized as I'm totally new to Asterisk and a bit confused... |
| 21:59.37 | Muggl | Corydon-w: so at the end it is no problem getting an expensive Digium PRI card if Asterisk will save me bucks on IP phones and let me keep my ISDN phones.. |
| 22:00.45 | Corydon-w | Muggl: unfortunately, I know very little about euroisdn cards, so I can't help you with that. |
| 22:00.51 | teknoprep | Oct 12 18:00:02 VERBOSE[8305] logger.c: [chan_sip.so]Oct 12 18:00:02 WARNING[8305] loader.c: /usr/lib/asterisk/modules/chan_sip.so: undefined symbol: ast_jb_configure |
| 22:00.51 | teknoprep | <teknoprep> Oct 12 18:00:02 WARNING[8305] loader.c: Loading module chan_sip.so failed! |
| 22:00.56 | teknoprep | i don't need zap... what should i do ? |
| 22:00.59 | Corydon-w | I know enough to guide you towards misdn, and that's it |
| 22:01.04 | teknoprep | i tryed removing chan_zap.so that didn't work |
| 22:01.09 | teknoprep | oh its sip now ? |
| 22:01.10 | teknoprep | wtf |
| 22:01.52 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 22:02.18 | teknoprep | anyone ? |
| 22:02.31 | robin_sz | Muggl: technically, yes its possible ... ther are a few articles out there on connecting ISDN cards to phones ... |
| 22:02.46 | Corydon-w | teknoprep: what version? |
| 22:02.49 | Muggl | ok, no problem.. May be any other of the 274 folks knows if Asterisk can convert a call that is placed by an ISDN phone to a VOIP call in realtime? |
| 22:03.00 | teknoprep | 1.2.9.1 |
| 22:03.07 | robin_sz | of course it can |
| 22:03.26 | aydiosmio | this isn't My First PBX(TM) |
| 22:03.36 | robin_sz | your problem will be connectiong the phones to the boxen |
| 22:03.42 | teknoprep | Corydon-w, i just rebuilt asterisk doing a make ; make bininstall |
| 22:03.43 | Muggl | robin-sz: ok, great.. thats what I need to know.. |
| 22:03.44 | Corydon-w | teknoprep: sounds like you're running modules from 1.4 with a 1.2 asterisk binary |
| 22:03.54 | robin_sz | Muggl: how many phones do you have? |
| 22:04.14 | robin_sz | roughly ... |
| 22:04.15 | Muggl | robin-sz: about 35, currently connected to a Siemens pbx |
| 22:04.19 | teknoprep | corydon-w that doesn't make any sense |
| 22:04.20 | robin_sz | right .. |
| 22:04.37 | Corydon-w | teknoprep: rm -rf /usr/lib/asterisk/modules, then redo your 'make install' |
| 22:04.51 | robin_sz | Muggl: and looking on your server, would you say it has more or less than 35 available PCI slots? |
| 22:05.08 | teknoprep | Corydon-w, will that overwrite my /etc/asterisk config files ? |
| 22:05.13 | Corydon-w | teknoprep: no |
| 22:05.42 | *** join/#asterisk linlin (i=linlin@67.173.49.55) |
| 22:05.47 | Muggl | robin-sz: I thought I could get 2 Digium PRI cards which can handle up to 32 isdn lines |
| 22:06.32 | robin_sz | Muggl: and split it out to 35 BRI ports how exactly? |
| 22:07.19 | robin_sz | or does it already appear as PRI from some egneric Siemens gear? |
| 22:08.40 | robin_sz | whatever, I've seen articles where people have succesfully connected a Siemens ISDN phone to a asterisk box using a HCF based ISDN card ... |
| 22:09.24 | Muggl | robin-sz: yes, I know about HCF cards.. and the siemens pbx should remain between the telco pri asterisk |
| 22:09.49 | robin_sz | eugh. nasty |
| 22:10.00 | robin_sz | why would you want to do that? |
| 22:10.20 | *** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.mn.comcast.net) |
| 22:10.25 | robin_sz | is this for business use? |
| 22:10.33 | teknoprep | all i would like to say |
| 22:10.36 | Muggl | robin-sz: because I'd like to keep all isdn phones and the possibility of dialing out with isdn |
| 22:10.36 | teknoprep | thank god for VMware |
| 22:10.40 | teknoprep | revert to snapshot |
| 22:10.41 | teknoprep | lol |
| 22:11.06 | robin_sz | you are not thinking of routing your outbound business traffic over The Internet as VOIP to save money are you? |
| 22:11.30 | Muggl | robin-sz: yep |
| 22:11.57 | robin_sz | Muggl: Good Luck. |
| 22:12.31 | robin_sz | bet you are back on ISDN for outgoing within the month |
| 22:13.22 | *** join/#asterisk dasenjo (n=dasenjo@63.245.86.151) |
| 22:13.23 | Muggl | because of? poor quality, incompatibilty or else ? |
| 22:13.56 | robin_sz | poor quality, unreliabel service, poor quality, lack of reliability and poor quality. in that order. |
| 22:14.20 | robin_sz | try a test first with you existing hardware. |
| 22:14.47 | Muggl | hm, sounds that you've already tried exactly what I'm going for now - without luck ;) |
| 22:15.41 | robin_sz | well, we put 30 or so nice Snom phones in, to replace a ISDN Siemens system, with * ... we tried using VOIP outbound, but its crap |
| 22:15.47 | robin_sz | we use ISDN outbound |
| 22:16.05 | robin_sz | we love the * features and the Snom phones and features |
| 22:16.14 | gambolputty | snom is good :) |
| 22:16.23 | *** join/#asterisk jbsolutios (n=jbenson@193.93.153.1) |
| 22:16.25 | robin_sz | but voip over public internet does not work in any meaningful way |
| 22:16.41 | Adam12 | Hmm. For a hardphone, I'm having a hard time deciding between an Aastra (either 480i or 9133i) or a Polycom (IP430). Anybody have a firsthand experience with either and can help me out? |
| 22:16.52 | gambolputty | get a QOS guaranteed internet connection then? |
| 22:17.07 | Muggl | I think I'll stay with ISDN then.. and re-try in 2-3 years.. sounds like voip doesnt make really sense without a leased line to the other branch offices.. |
| 22:17.20 | jbsolutios | Hi Everyone. I have an incoming call which ring with a UK pattern for the caller, but when they enter a queue it has a US ring pattern. Does anyone have any suggestions please? |
| 22:17.32 | jbsolutios | I am having a very dense evening :( |
| 22:20.20 | teknoprep | robin_sz ? |
| 22:20.26 | teknoprep | robin_sz, VoIP for business is very nice |
| 22:21.04 | teknoprep | robin_sz, i have had 0 problems with outbound over inet for Voice so far |
| 22:22.23 | *** join/#asterisk dasenjo (n=dasenjo@63.245.86.151) |
| 22:23.02 | robin_sz | teknoprep: I think there are places int he US whereit can work, but in .eu, its a disaster |
| 22:23.21 | diclophis-work | in asterisk-addons-1.4 is export CFLAGS=-DMYSQL_LOGUNIQUEID automagically set? |
| 22:24.03 | robin_sz | teknoprep: not helped by US voip providers launching regionally branded sub domains that look like they may actually be based in .eu, but are not. |
| 22:25.26 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 22:25.26 | *** mode/#asterisk [+o mog] by ChanServ |
| 22:25.55 | Muggl | Is anyone using VOIP services over COLT ? I think they're too into it for a couple of month... |
| 22:25.59 | *** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.mn.comcast.net) |
| 22:26.21 | robin_sz | not knowing |
| 22:26.53 | robin_sz | we tried it in .ch with about 4 different providers .. none worhtwhile |
| 22:27.08 | *** join/#asterisk TrixVox (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net) |
| 22:28.15 | robin_sz | when you try to call frnace from .ch and you hear a taiwanese error message, you know its baaaaaaad |
| 22:29.14 | Muggl | robin_sz: hm.. u eventually speka german? |
| 22:30.33 | TrixVox | anyone know which option sets the per-phone admin password in a polycom xml file? |
| 22:31.31 | carrar | did you extract sip.conf from the polycome sip source code sip file? |
| 22:31.42 | carrar | and the other conf files? |
| 22:31.59 | carrar | I haven't looked, but I assume it is in there |
| 22:32.18 | carrar | (not in the sip file) |
| 22:32.18 | carrar | but the others |
| 22:32.50 | TrixVox | i looked on the wiki, but i don't have those files... do you know if they're freely available? |
| 22:33.39 | *** join/#asterisk andymul (n=iCallAnd@cpe-74-72-215-143.nyc.res.rr.com) |
| 22:44.54 | diclophis-work | does the chan_jingle driver support video? |
| 22:44.57 | diclophis-work | say from ichat? |
| 22:45.26 | mog | no |
| 22:45.30 | diclophis-work | doh |
| 22:45.32 | mog | not currently |
| 22:45.36 | diclophis-work | what channels do support video? |
| 22:45.38 | diclophis-work | is it planned? |
| 22:45.39 | mog | sip |
| 22:45.42 | mog | and h323 |
| 22:45.43 | mog | iax2 |
| 22:45.53 | mog | chan_jingle will support jingle spec |
| 22:46.01 | mog | i know that ichat does not |
| 22:46.12 | mog | at least currently |
| 22:50.39 | jbsolutios | sorry to bother everyone, but does anyone if you can set the ring tone callers hear when they are in a queue please? |
| 22:53.32 | *** join/#asterisk DrAk0- (n=ljd@unaffiliated/luisjose) |
| 22:53.50 | DrAk0- | what is better ? a spa-3000 or Sangoma FXO-2 Module ? for a home use |
| 23:00.32 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 23:03.07 | *** part/#asterisk jarg (n=jarg@200.56.225.61) |
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| 23:07.33 | DasTech | I need a full dialplan for asterisk is there one out there with all the features ? |
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| 23:12.52 | qdk | DasTech: are you high? |
| 23:13.16 | quid246 | Hmm... what's the purpose of the |
| 23:13.22 | quid246 | "agi_callingani2" variable in AGI? |
| 23:13.25 | EyeCue | laf |
| 23:16.20 | DasTech | no |
| 23:17.16 | kronic | anyone managed to get odbc -> mysql realtime queue_members working? |
| 23:17.21 | kronic | apparently its not possible in 1.2.12 |
| 23:19.52 | *** join/#asterisk fnordus (n=dnall@24.85.128.203) |
| 23:21.05 | DasTech | anyone doing a voice controled ivr with asterisk |
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| 23:23.17 | Inverted | if you wanted to play a file back to a caller, but speed it up by default, how could you do that? i know there is 'control stream file' which uses * and # to slow down and speed up, but I want it to just play it fast by default |
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| 23:27.18 | Coriantum | Is it possible to add more than 24 channels in a zap group? |
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| 23:33.58 | Coriantum | Has anyone been able to add 4 T1s to a single zap group? |
| 23:35.24 | DasTech | anyone here using sphinx with asterisk for voice control |
| 23:42.44 | *** join/#asterisk docelm0 (n=vircuser@24-51-128-113.pittpa.adelphia.net) |
| 23:42.49 | beehive | sphinx has good sound quality? |
| 23:44.52 | *** part/#asterisk robin_sz (n=you@adsl.redpoint.org.uk) |
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| 23:45.01 | *** mode/#asterisk [+o mog] by ChanServ |
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| 23:50.10 | DasTech | sphinx is voice control |
| 23:50.18 | DasTech | flite is the tts engine |
| 23:51.21 | findlay | how do I adjust the gain on an iax channel? |
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