| 00:00.02 | JT | Hymie: to do PoE on a 501 you must buy an expensive cable |
| 00:00.03 | Hymie | wouldn't that just work with POE when you go to POE? |
| 00:00.08 | JT | Hymie: no. |
| 00:00.09 | Hymie | weird |
| 00:00.28 | karlhaines | any Gentoo users? |
| 00:00.36 | Hymie | they already have power going in via the network jack.. wtf is with needed ing special cable to do POE then, heh |
| 00:00.48 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 00:00.57 | wunderkin | well it would have a speaker, just listen only.. i know 301 is like that but too lazy to check the others since it is not for me and it is readily available on the website |
| 00:00.59 | JT | Hymie: 802.3af uses -48VDC with special power enabling signalling |
| 00:01.19 | Hymie | well, sure.. ok, but why on earth would theyt do that wonky power business.. they're just strange |
| 00:01.27 | JT | the 802.3af compliant switch will not send power until the device has signalled they can take power |
| 00:01.37 | Zipper_32 | Hymie: Unfortunately no. |
| 00:01.37 | Zipper_32 | Right now I'm in a tossup between the 430 and 330. And it appears that the 330 is best for retail and the 430 as an office phone with speakerphone. Does that sound right? |
| 00:01.40 | JT | so they don't fry non PoE devices |
| 00:01.42 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 00:02.09 | Hymie | does the 430 come with a normal power adapter too? |
| 00:02.28 | JT | yes, quite a nice one |
| 00:02.32 | Hymie | hmm, well |
| 00:02.32 | *** join/#asterisk asteriskguy (n=learnast@cpe-75-80-111-113.socal.res.rr.com) |
| 00:02.34 | Hymie | interesting then |
| 00:02.36 | JT | not a stupid wall wart |
| 00:02.54 | JT | a tiny little inline switchmode universal voltage PS |
| 00:03.01 | JT | which doesn't get very warm |
| 00:03.05 | XVampireX | Can anyone please call me? |
| 00:03.12 | XVampireX | sip:17476499050 |
| 00:03.15 | Hymie | oh |
| 00:03.22 | Hymie | it doesn't support a headset out of the box? :( |
| 00:03.31 | XVampireX | just testing ringtone |
| 00:03.35 | JT | XVampireX: err that won't work, people need your hostname too |
| 00:03.44 | JT | Hymie: the 430 has headset capability |
| 00:03.50 | XVampireX | Hmm, moment |
| 00:03.54 | Hymie | doesn't list it on the doc... |
| 00:03.55 | wunderkin | they all do dont they? |
| 00:04.01 | JT | sip is only a protocol |
| 00:04.04 | Hymie | not that you're wrong... |
| 00:04.04 | JT | yes they do |
| 00:04.10 | XVampireX | proxy01.sipphone.com |
| 00:04.11 | JT | i have mine connected to a plantronics |
| 00:04.18 | Hymie | oh, that whole part about the headset |
| 00:04.27 | Hymie | well, how shoudl anyone notice thaT? ;) |
| 00:04.27 | *** join/#asterisk [hC] (n=hardcore@206.108.27.93) |
| 00:04.41 | XVampireX | sip:17476499050@proxy01.sipphone.com |
| 00:04.57 | LeddyHM | Anyone see this before? http://pastebin.ca/487729 |
| 00:05.14 | Hymie | it would be nice if these had gigabit.. |
| 00:05.16 | LeddyHM | /etc/init.d/asterisk installed by "make config" is bunk |
| 00:05.25 | Hymie | yes, I saw polycom's response to that :( |
| 00:06.24 | JT | polycom make a little switch for gigabit |
| 00:06.26 | JT | but meh |
| 00:06.38 | JT | just get 2 ethernet ports, not that hard |
| 00:06.59 | JT | because i don't think gigabit operation and 802.3af PoE are compatible |
| 00:07.02 | JT | it's one or the other |
| 00:07.44 | Zipper_32 | 802.3af is gigabit capable. |
| 00:07.51 | JT | proof? |
| 00:08.07 | Zipper_32 | Argh, gotta dig up my textbook, gimmie a moment. |
| 00:08.17 | JT | if you know how the two protocols worked, you'd know it's impossible |
| 00:08.22 | JT | s/know/knew/ |
| 00:09.12 | Zipper_32 | You mean that gigabit uses all 4 pair, and power needs those pair too? |
| 00:09.31 | JT | of course |
| 00:09.32 | Zipper_32 | And that you can send data over the same pair as the power. |
| 00:09.42 | Zipper_32 | It's very possible. |
| 00:09.52 | JT | the only way you can pump gigabit speeds over crappy twisted copper is by using all 4 pairs |
| 00:10.47 | Hymie | JT: well, add 'currently' to that ;) |
| 00:10.53 | Hymie | JT: it's always 'currently' ;) |
| 00:11.11 | Zipper_32 | It's been possible for at least a year now. |
| 00:11.35 | Zipper_32 | Just search for gigabit POE switches, you'll get results. |
| 00:11.43 | JT | Zipper_32: they usually switch modes |
| 00:11.51 | *** join/#asterisk ltdwk- (n=z@203-173-10-9.perm.iinet.net.au) |
| 00:11.59 | JT | to 100Mbit PoE or Gigabit no PoE |
| 00:12.04 | Daejeo1 | JT: can I use JP |
| 00:12.14 | Daejeo1 | japan |
| 00:12.21 | JT | Daejeo1: use whatever you like, it's just tones |
| 00:13.55 | JT | Zipper_32: understand the underlying layer 1 technologies, then come back to me and tell me how it's possible |
| 00:14.07 | JT | i know in analogue telephony you can overlaw AC over DC |
| 00:14.18 | JT | but it's not so easy in ethernet |
| 00:14.33 | JT | at least in a way that's compatible with existing ethernet signalling standards |
| 00:14.38 | JT | s/overlaw/overlay/ |
| 00:14.49 | Zipper_32 | JT: Please read http://en.wikipedia.org/wiki/Power_over_Ethernet#Currently_recommended_.28IEEE_802.3-2005.29 |
| 00:16.17 | JT | Zipper_32: i'm not sure if anything actually supports that yet |
| 00:16.27 | JT | and that is not a primary source |
| 00:16.30 | Hymie | ah, wikipeida... always so factural and accurate ;) |
| 00:16.34 | JT | that's only wikipedia |
| 00:16.42 | JT | which often has wrong information in it |
| 00:17.03 | JT | Zipper_32: i know it may be possible in theory, but i'm talking about commerically available |
| 00:17.04 | Hymie | JT: no, it's part of the new, open source methodology, so therefore it must be better than any other source ;P |
| 00:17.12 | JT | heh |
| 00:17.35 | JT | most of the stuff to do with voip on wikipedia was absolute rubbish |
| 00:17.41 | JT | i had to rewrite a lot of it |
| 00:17.59 | Hymie | I bet by now it's all been re-rubbishized (is that a word? ;) |
| 00:18.18 | Zipper_32 | JT: Picture this: Gigabit Device Signals out --> POE Injection ontop of 4 pair --> 4 Pair Line --> Power removed from line to power device --> Differential current (resulting signal) is used to signal remaining device. |
| 00:19.06 | Hymie | the only value I can see for PoE anyhow is in an office where no one uses computers, but wants voip telephony |
| 00:19.11 | Hymie | are there that many of such? |
| 00:19.21 | JT | Hymie: there's a lot more value than that |
| 00:19.34 | Hymie | anyhow, most phones in the 80s even required a normal phone jack + additioal power |
| 00:19.50 | JT | most pabx phones don't require additional power |
| 00:19.52 | Hymie | JT: well... I just don't see the necessity for it. |
| 00:19.55 | JT | pabx/pbx |
| 00:20.02 | JT | Hymie: there are a LOT of advantages |
| 00:20.11 | JT | no power point taken at every station |
| 00:20.16 | JT | no power supply clutter |
| 00:20.20 | JT | less cables |
| 00:20.27 | JT | centralised power backup and management |
| 00:20.29 | Hymie | perhaps... there are a lot of advantages to mechano-electrical steering, but I think its disadvantages are far worse, than the advantages that appear |
| 00:20.32 | *** join/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 00:20.50 | Hymie | actually, #4 I agree with if there are not computers in each office (with UPSes) |
| 00:21.01 | JT | ideally phones should only have one cable to the wall anyway |
| 00:21.17 | *** join/#asterisk SECGOD (i=SECGOD@c-71-57-36-106.hsd1.il.comcast.net) |
| 00:21.31 | Hymie | JT: when I see most modern offices, it's more cable than brain per square foot anyhow? ;) what's one less cable? |
| 00:21.42 | JT | Zipper_32: i understand the theory quite well, little point trying to teach me that sort of stuff |
| 00:21.52 | Hymie | JT: what's your email, I'll send you the specs on electro-mechanical steering |
| 00:22.03 | JT | you talking about for cars? |
| 00:22.07 | Hymie | yeah |
| 00:22.35 | Zipper_32 | So how is it impossible? Cisco uses the technology now - http://www.cisco.com/en/US/netsol/ns340/ns394/ns147/ns412/netbr09186a00801f4b9b.html |
| 00:22.41 | Hymie | I just received a nice writeup on it.. unfortunately, it doesn't get as technical as I'd like, but it's a nice mid-level look |
| 00:22.50 | Zipper_32 | <PROTECTED> |
| 00:22.51 | JT | Hymie: planes have been doing fly by wire for years |
| 00:23.00 | JT | Zipper_32: I KNOW THAT |
| 00:23.05 | Hymie | JT: this isn't fly by wire (and, we're talking about cars) |
| 00:23.07 | JT | Zipper_32: stop beating a dead horse |
| 00:23.11 | *** join/#asterisk plasmid (n=noway@c-68-46-97-136.hsd1.pa.comcast.net) |
| 00:23.29 | Hymie | JT: it's quite neat.. it's mechanical steering with an electric steering assist instead of a steering pump/pressure |
| 00:23.37 | JT | Hymie: interesting |
| 00:23.46 | Hymie | JT: so, if the power dies.. you can still steer (and not die ;) |
| 00:24.11 | JT | Hymie: you should hear about the systems used in military aircraft |
| 00:24.13 | Hymie | JT: my only problem is that there _is_ a motor there that provides pressure, and could try to override the driver's wishes when in a fault state... |
| 00:24.26 | JT | like the C-17 Globemaster heavy lift jet transport |
| 00:24.33 | JT | it has quadruple redundant fly by wire |
| 00:24.39 | JT | with dual mechanical backup |
| 00:24.42 | plasmid | when I punch in my DID# on my cell like this 215-xxx-xxxx it tells me it cannot complete the call.. but if I dial it like this: 1-215-xxx-xxxx then it completes the call.. How can I take out the "1" off? |
| 00:24.43 | Hymie | JT: sweet |
| 00:25.06 | plasmid | off asterisk dial plan that is. I am missing some operand. |
| 00:25.32 | Zipper_32 | plasmid: ${EXTEN:1} |
| 00:25.34 | JT | Zipper_32: point it, most hardware can't do it |
| 00:26.01 | Zipper_32 | JT: I just don't see why you're telling me that the function is impossible. Especially if you say that you understand it./ |
| 00:26.26 | JT | Zipper_32: impossible with most current hardware i should've said |
| 00:26.52 | Zipper_32 | I don't mean to upset you in any way, but I fail to understand why I receive flak when I bring something to ones attention that they 'knew' wasn't possible. |
| 00:27.13 | Zipper_32 | Ahh, I see then. |
| 00:27.15 | Zipper_32 | Understood. |
| 00:27.15 | killfill | hey |
| 00:27.17 | plasmid | Zipper_32, not sure I follow u. |
| 00:27.31 | killfill | what do you guys use for graphing stadsistics from a CDR database? |
| 00:27.33 | Zipper_32 | plasmid: Are you trying to remove the 1 from the dialplan? |
| 00:27.50 | killfill | is there a cool util for that?.. |
| 00:27.50 | Zipper_32 | erm, remove the 1 from the outbound call? |
| 00:28.05 | killfill | i.e. hitogram from mon-fri, etc |
| 00:28.39 | JT | Zipper_32: also, if people really need gigabit for their workstations, it shouldn't be that hard to get another ethernet link for their phones i'd think |
| 00:29.16 | plasmid | Zipper_32, err.. some ppl complain that when they call my number they have to punch in 1 b4 the area code... i just want it removed from my dial plan so ppl can just dial straight 215-xxx-xxxx |
| 00:30.41 | JT | as going through a phone would reduce the performance a little bit |
| 00:30.53 | Zipper_32 | JT: I would just keep the ethernet connection dedicated to phone. In my opinion, it just makes it easier in the closet with multiple patch panels dedicated to separate purposes. |
| 00:33.43 | JT | yeah |
| 00:33.43 | JT | if you have limited ports, then 100Mbit for pc and phone should be fine |
| 00:33.43 | Zipper_32 | plasmid: You're talking about people who are using your pbx, right? |
| 00:33.43 | JT | but if you have a serious environment that needs 1000Mbit/s for each PC, you can afford another port |
| 00:33.43 | Zipper_32 | killfill: This might help you, I've never tried it though: http://www.voip-info.org/wiki/view/Asterisk+CDR+Areski+GUI |
| 00:33.44 | killfill | ooh its the one used by freepbx.. |
| 00:33.44 | plasmid | Zipper_32, yes.. ppl that call my pbx DID# |
| 00:33.44 | Zipper_32 | plasmid: from inside your pbx? |
| 00:33.45 | Zipper_32 | Otherwise you're just long-distance from them, and they'll always need 1 before the number. |
| 00:33.53 | plasmid | Zipper_32, what about ppl that live on my state, they always have to dial in a 1 to reach me? |
| 00:34.10 | Zipper_32 | Are they considered long-distance? |
| 00:34.27 | plasmid | no.. they have 215 area code too.. so most of them dont punch in 1 |
| 00:36.12 | JT | plasmid: are they calling in from normal phones via the PSTN? |
| 00:36.23 | Zipper_32 | If you're not considered long-distance, you have to take it up with your phone company as to why somebody has to push 1 first. 1 is required for long-distance calls. If they really ARE long distance, you can possibly get a DID in their local area and route it to your phone using asterisk. |
| 00:37.47 | Zipper_32 | Case point: I have a 604 number. Not ALL 604 numbers are local for me. |
| 00:38.37 | killfill | some sites, has a "click here to live support" thing.. you guys know an app that can make call to support team via web?.. (i.e. java applet or somethng) |
| 00:38.55 | JT | click to call |
| 00:39.01 | JT | they pretty much always cost money |
| 00:39.16 | JT | it's easier to make a web page that calls the person's normal phone |
| 00:39.35 | JT | who wants to support their retarded headset and sound settings too just to talk to them maybe once off? |
| 00:40.29 | killfill | cost money?.. |
| 00:40.59 | killfill | well not all ppl have a configure sip/iaz phone setted up.. |
| 00:41.18 | JT | killfill: yes, these java applets cost money |
| 00:41.28 | JT | you buy a click to call applet from someone |
| 00:41.58 | killfill | ah.. |
| 00:42.03 | JT | killfill: yes, and who want's to talk to a non-tech customer if they don't have their pc all setup for use as a softphone? way too hard |
| 00:42.09 | JT | s/want's/wants/ |
| 00:42.28 | JT | i think it's a stupid idea, personally |
| 00:42.53 | JT | click to call their normal phone back is a much better idea, but you must be wary of the potential for abuse |
| 00:43.56 | killfill | yeah.. well could be a "Cselling idea".. "hey, with us, when you have problems, you can call us for free, with not setup" |
| 00:44.31 | JT | killfill: yes and do what i just said |
| 00:44.39 | JT | click to call THEIR REAL PHONE |
| 00:44.46 | JT | not a softphone applet |
| 00:45.07 | killfill | Ah, didnt got your idea.. |
| 00:45.13 | killfill | like "we will call you back" |
| 00:45.15 | JT | how hard is it really? |
| 00:45.17 | JT | it's simpler |
| 00:45.23 | JT | no, it calls them |
| 00:45.24 | JT | immediately |
| 00:45.38 | killfill | you just open my eyes.. :P |
| 00:46.32 | killfill | i would have no idea how to do this automattically tho..:P (withouth doing a lame email to support, and then they make the call manually of course) |
| 00:47.07 | JT | the web page generates either a .call file or a command to the asterisk manager interface |
| 00:49.52 | killfill | loved the idea |
| 00:57.39 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
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| 01:11.41 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 01:12.22 | tengulre | hi,all |
| 01:12.24 | tengulre | anybody here? |
| 01:14.30 | *** join/#asterisk ltdwk- (n=z@203-173-10-9.perm.iinet.net.au) |
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| 01:16.39 | tengulre | Hello |
| 01:16.40 | tengulre | hi |
| 01:16.44 | tengulre | ... |
| 01:19.09 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 01:19.13 | JunK-Y | tengulre: fxsks |
| 01:23.10 | tengulre | anybody here. |
| 01:23.10 | mutilator | O_O |
| 01:23.10 | JT | it's like some people have never seen a netsplit or something |
| 01:23.10 | mutilator | maybe they havnt |
| 01:23.10 | mutilator | not everyone sits here all day like us |
| 01:23.10 | *** join/#asterisk remmo (n=junk@smack.isp.net.au) [NETSPLIT VICTIM] |
| 01:24.42 | tengulre | I have 4 FXOs card, how to setting it in /etc/zaptel.conf? |
| 01:24.42 | tengulre | fxoks or fxols? |
| 01:24.42 | *** join/#asterisk ctaloi (n=ctaloi@pool-71-176-69-82.syrcny.fios.verizon.net) |
| 01:24.42 | blitzrage | ks |
| 01:24.42 | blitzrage | ~book |
| 01:24.45 | jbot | methinks book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 01:24.45 | tengulre | fxoks? but failed! |
| 01:26.16 | Daejeo1 | [root@asterisk1 ~]# ztcfg -v |
| 01:26.16 | Daejeo1 | Zaptel Configuration |
| 01:26.17 | Daejeo1 | ====================== |
| 01:26.17 | Daejeo1 | 1 channels configured. |
| 01:26.17 | Daejeo1 | what is wrong? |
| 01:26.17 | JT | tengulre: fxsks |
| 01:26.18 | JT | tengulre: you are using totally wrong signalling, of course it does not work. |
| 01:26.18 | Daejeo1 | JT plz have a look |
| 01:26.18 | Daejeo1 | I am unable to see configuration |
| 01:26.59 | JT | Daejeo1: err what on earth, how are we meant to diagnose anything from that? |
| 01:26.59 | Daejeo1 | this is what I am saying |
| 01:26.59 | JT | what's the problem? |
| 01:28.54 | tengulre | ztcfg -v line 0: Unable to open master device '/dev/zap/ctl' |
| 01:28.54 | Daejeo1 | ztcfg -v does not show the configuration |
| 01:28.55 | JunK-Y | -vvvvv |
| 01:28.55 | tengulre | JT: what 's the wrong signalling... |
| 01:28.55 | JunK-Y | when zaptel.conf is okay. |
| 01:28.55 | JT | tengulre: FXSKS |
| 01:28.55 | JT | tengulre: for an FXO port |
| 01:28.55 | JT | tengulre: how many times to people need to tell you? |
| 01:29.03 | Daejeo1 | JT: how can I check the zaptel version? |
| 01:29.11 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 01:29.14 | tengulre | ztcfg -vvvvvvvvvvvvvvvv ZT_CHANCONFIG failed on channel 1: No such device or address(6) |
| 01:29.27 | JT | tengulre: is the device driver even loaded? |
| 01:29.53 | tengulre | JT: yeah! 1st: modprobe zaptel 2nd: modprobe xxxx |
| 01:30.33 | JT | you do not need to modprobe zaptel. |
| 01:30.41 | Daejeo1 | JT: me? |
| 01:30.42 | JT | you load the module for the correct driver |
| 01:30.48 | JT | Daejeo1: i'm not talking to you |
| 01:32.14 | Daejeo1 | JT: how can I check zap ver? |
| 01:32.29 | JT | Daejeo1: no idea okayt |
| 01:32.40 | JunK-Y | u open zaptel.h and u look. |
| 01:32.43 | JT | if you have sources installed, check them |
| 01:32.52 | tengulre | JT: Notice: Configuration file is /etc/zaptel.conf |
| 01:32.53 | tengulre | line 0: Unable to open master device '/dev/zap/ctl' |
| 01:33.03 | Daejeo1 | I have installed the trixbox |
| 01:33.14 | JT | ~trixbox |
| 01:33.34 | jbot | Trixbox is a full linux distro that includes , FreePBX, and other 3rd party add-ons. It is these things on top of which make it seriously painful to support and hence you will find little help here for it. Try asking in #trixbox , or their forums & WIKI at http://www.trixbox.org |
| 01:33.39 | JT | tengulre: do any kernel messages come up when you load the module for your card? |
| 01:34.23 | tengulre | JT: only got the zaptel messages. not my cards. |
| 01:34.41 | JT | tengulre: well check the messages for the card now |
| 01:34.45 | JT | look in dmesg |
| 01:35.48 | tengulre | JT: http://rafb.net/p/FJDAwf66.html |
| 01:36.34 | JT | tengulre: what zap card do you have? |
| 01:37.56 | tengulre | a compatible card! |
| 01:37.56 | JT | ffs |
| 01:37.56 | JT | what is the card |
| 01:37.56 | JT | or stop wasting our time |
| 01:37.58 | JT | A CARD is not useful |
| 01:39.52 | JT | there is no evidence in that dmesg dump of the card driver even being loaded |
| 01:39.52 | JT | stop loading the zaptel module manually |
| 01:39.52 | JT | it's unnecessary |
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| 01:44.42 | killfill | hm.. |
| 01:44.42 | tengulre | JT: Thanks! |
| 01:44.42 | JT | tengulre: so what's the card |
| 01:44.42 | JT | tengulre: ? |
| 01:46.07 | MindTheGap_ | im setting up a replacement * server w a TE110P card. modules load cleanly but i get a "Unable to open master device '/dev/zap/ctl'" when issuing a ztcfg... Is it an expected message whe the card is not connected to the E1? also, theres a "wcte1xxp: Setting yellow alarm" at /var/log/messages. |
| 01:46.08 | killfill | JT, whats the matter with my .call file?.. http://pastebin.ca/488522 does it look too bad? |
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| 01:46.08 | MindTheGap_ | s/whe/when |
| 01:46.08 | killfill | MindTheGap_: i get that yellow alarm too. never understood what it was... :P |
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| 01:48.11 | MindTheGap_ | I think a yellow allarm is ok for as far as i know, when no signal is received a yellow is sent to the other node... |
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| 01:48.55 | MindTheGap_ | i just want to make sure the "Unable to open master device '/dev/zap/ctl'" is ok when the card is not connected... |
| 01:48.55 | MindTheGap_ | is it? |
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| 01:51.39 | tengulre | JT: openvox card |
| 01:51.47 | tengulre | JT: www.openvox.com.cn |
| 01:53.58 | tengulre | JT: the card is not well |
| 01:54.33 | tengulre | compatibly too low. |
| 01:55.01 | JT | i see |
| 01:57.44 | Zipper_{A} | killfill: Did you figure out that callfile of yours? |
| 01:58.07 | killfill | Zipper_32: actually not.. |
| 01:59.19 | Zipper_32 | killfill: What are you trying to do with it? Just send out ext 600 to user 6002, right? |
| 01:59.54 | Zipper_32 | Well, IAX2/6002 |
| 01:59.54 | killfill | yup |
| 02:00.37 | killfill | ooh how dumb..:P |
| 02:00.43 | killfill | i missed the "2" |
| 02:00.54 | Zipper_32 | Speaking of which, shouldn't you be using iax2, instead of iax? |
| 02:01.03 | Zipper_32 | =) |
| 02:02.40 | Zipper_32 | It's okay, I don't think anybody else saw it |
| 02:02.47 | killfill | hehe |
| 02:02.54 | *** join/#asterisk MrTelephone (n=test@bas13-toronto63-1242453250.dsl.bell.ca) |
| 02:03.03 | MrTelephone | does anyone here use an adit 600? |
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| 02:21.12 | SuperID | I'm trying to configure broadvoice. Their config page says to put a line in sip.conf in the [general] section of the form register=>phone@sip.broavoice.com:xxx:yyy every other option I have set is key=value not key=>value, is their page correct? should it be register=>value? |
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| 02:36.19 | anonymouz666 | blitzrage: problem solved using STRPTIME() |
| 02:36.30 | anonymouz666 | STRFTIME works only with current time |
| 02:37.13 | anonymouz666 | and then Sayunixtime() said the right thing |
| 02:38.01 | ohadz | hi y'll. need assistance with my dial plan. anyone? |
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| 02:47.50 | Zipper_32 | Any explanations as to why, in the situation of a: * <-----> IAX2 <------> * Configuration, one side suddenly sees the other as unreachable, but the other can communicate via IAX to the other side? |
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| 02:51.52 | ohadz | need assistance with my dial plan. anyone? |
| 02:52.06 | ohadz | can't make calls out or receive calls in :/. |
| 02:52.07 | JT | how about explaining the problem |
| 02:52.18 | jutex | hi, i'm trying to implement realtime to my asterisk. My SIP phones can register, but I don't know how to insert the rows to extenstion table. |
| 02:52.27 | JT | people usually ignore questions which don't have enough info |
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| 02:53.00 | jutex | what is the "exten" value should i put there ? |
| 02:54.23 | jutex | in order to call the registered phone, is it the "name" from sip table ? |
| 02:54.47 | *** join/#asterisk stridernzl (n=neville@125-237-116-132.jetstream.xtra.co.nz) |
| 02:56.11 | stridernzl | hi all. seems lots of people still are trying / running asterisk :) Can i ask a question? |
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| 02:57.12 | blitzrage | stridernzl: go ahead -- you don't need to ask to ask |
| 02:57.30 | stridernzl | How many ppl here are running remote extensions ..... that is even overseas .... are they good , is anyone trying to run a remote extension and even that extension being overseas .. ie 3,000 kms |
| 02:58.00 | JT | 3000km isn't very far |
| 02:58.06 | JT | by cable anyway |
| 02:58.08 | stridernzl | basically really remote extensions ... that is New Zealand to australia ? |
| 02:58.21 | JT | new zealand and australia are really close |
| 02:58.54 | JT | it's like 60ms latency at worst |
| 02:59.26 | stridernzl | so basically sitting here thinking its hard ... gonna be crappy then its more oh yeah mate .... its really easy and pretty much something i can do ? |
| 02:59.43 | stridernzl | and expect it to work well |
| 02:59.44 | JT | yeah just don't use satellite bandwidth :) |
| 03:00.11 | stridernzl | yeah anything wifi etc same with sat always seem dodgy to me :) |
| 03:00.20 | JT | the southern cross cable network has hops direct from nz (auckland i think) to sydney |
| 03:00.44 | JT | yeah, <3000km, vs. ~80000km, slight difference :P |
| 03:00.45 | stridernzl | personaly i think its only good for backup :), calble so to speak is the way to go |
| 03:01.01 | JT | geostationary orbit is 35000km above the equator |
| 03:01.11 | JT | 3 times the diameter of the earth |
| 03:01.14 | *** part/#asterisk jutex (n=c@bb219-74-243-186.singnet.com.sg) |
| 03:01.25 | stridernzl | :) yeah well :) .. says it all! |
| 03:01.25 | *** join/#asterisk nick125_lappy (n=nick@atarack/staff/nick125) |
| 03:01.30 | nick125_lappy | evening everyone |
| 03:01.46 | nick125_lappy | Does asterisk native moh support ogg/voribs? |
| 03:01.51 | stridernzl | that truely depends where you live but hello |
| 03:02.01 | nick125_lappy | stridernzl: I hate timezones :p |
| 03:02.06 | stridernzl | JT: so where is you / your setup of * |
| 03:02.19 | JT | the universal time of day on irc is always morning :P |
| 03:02.32 | JT | stridernzl: have a few in australia |
| 03:02.33 | nick125_lappy | lol |
| 03:02.43 | stridernzl | They are great .. I was chatting with TK before using asterisk across in canada it its special! |
| 03:03.37 | stridernzl | JT: o.k me live in CHCH new zeland and have office in Sydney ... So thinking i can get them to act as an extesion to N.Z .. |
| 03:03.43 | *** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com) |
| 03:03.54 | JT | stridernzl: sure, i don't see why not |
| 03:04.02 | stridernzl | take the calls .. and i can go play golf .. or Farcry or something :) |
| 03:05.20 | stridernzl | JT: it would be really cool though ... I think half my struggle is comming to the realisation of what i have just put under my hood with the implentation of asterisk ... |
| 03:06.19 | *** join/#asterisk tenzind (n=tenzind@202.144.144.77) |
| 03:06.40 | JT | yeah it gives you a lof of flexibility compared to other solutions :) |
| 03:06.43 | nick125_lappy | Mhrwehgeghbw |
| 03:06.44 | nick125_lappy | May 14 20:07:26 ERROR[16038]: format_ogg_vorbis.c:224 ogg_vorbis_open: Only monophonic OGG/Vorbis files are currently supported! |
| 03:06.47 | nick125_lappy | :( |
| 03:06.55 | stridernzl | so anyone anyway so to speak can plug into our little pabx server as a remote extension be they in germany or half way across the world? |
| 03:07.11 | stridernzl | What sort of clients do they use ? |
| 03:07.16 | JT | stridernzl: sip phones |
| 03:07.24 | JT | or ATAs |
| 03:07.25 | nick125_lappy | Huh...I'm still using 1.2.14, that might be bad. |
| 03:07.45 | *** part/#asterisk nhudson (n=nhudson@68.113.120.148) |
| 03:07.50 | JT | nick125_lappy: you need mono music |
| 03:07.53 | JT | not stereo |
| 03:08.04 | nick125_lappy | JT: I need to convert a few of my oggs to mono |
| 03:08.26 | *** join/#asterisk Cabal (n=Cabal@unaffiliated/cabal) |
| 03:08.27 | nick125_lappy | Is the situation any different with 1.2.18? |
| 03:08.46 | stridernzl | JT: ATA's ... and presume you more mean hardphones but sip based , we using eyebeam, so fully soft phone |
| 03:08.53 | asteriskguy | <PROTECTED> |
| 03:09.30 | JT | stridernzl: yes hardphones or ATAs |
| 03:09.45 | JT | softphones i don't recommend other than for testing |
| 03:09.49 | JT | asteriskguy: not bad |
| 03:10.02 | nick125_lappy | JT: I don't even recommend them for testing! |
| 03:10.10 | JT | hehe |
| 03:10.13 | *** join/#asterisk flashnet (i=flashnet@admin.mojamreza.org) |
| 03:10.21 | asteriskguy | Cool, not to bad here either |
| 03:10.24 | nick125_lappy | Well..it depends on what you are testing |
| 03:10.33 | nick125_lappy | If you are testing your sanity, then, sure, it would work. |
| 03:10.41 | asteriskguy | been playing around with some things lately |
| 03:10.48 | asteriskguy | I got hylafax+ to work |
| 03:10.50 | asteriskguy | pretty good |
| 03:10.57 | JT | hmm |
| 03:11.12 | JT | asteriskguy: what are you using, modems? |
| 03:11.34 | asteriskguy | iaxmodem |
| 03:11.41 | JT | hrm |
| 03:11.45 | asteriskguy | with digium's T1 |
| 03:11.48 | JT | is it any good? |
| 03:11.56 | asteriskguy | yeah so far so good |
| 03:11.58 | JT | any failed faxes? |
| 03:12.14 | asteriskguy | not that I know of |
| 03:12.19 | asteriskguy | no complaints yet at least |
| 03:12.25 | karlhaines | is that stuff still experimental? |
| 03:12.34 | MrTelephone | what does network disconnect mean in terms of loopstart RBS |
| 03:12.40 | asteriskguy | we're in production where I work |
| 03:13.21 | karlhaines | hmm, i just had all my DIDs ported, and had my fax did ported to a company that email's the faxes to you (callwave i think) |
| 03:13.23 | asteriskguy | asterisk pickup faxes pretty fast and forward them to the appropriate iaxmodem |
| 03:13.58 | JT | karlhaines: what country? |
| 03:14.48 | karlhaines | JT, US |
| 03:14.57 | JT | ah ok |
| 03:15.04 | JT | is the fax service any good? |
| 03:15.22 | karlhaines | dunno yet, the number hasn't ported yet, i'm still waiting |
| 03:15.24 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:15.32 | karlhaines | a friend referred me, supposedly its great |
| 03:15.38 | JT | ah ok |
| 03:15.49 | JT | what fax services do they offer? |
| 03:15.55 | karlhaines | stkn_, i've been looking for gentoo users in here for a few days |
| 03:16.18 | karlhaines | JT, callwave.com |
| 03:16.34 | asteriskguy | oh, I found away to get voicemail to show up on email (old news), but if you delete the voicemail on your inbox it also deletes the one resides on asterisk |
| 03:16.38 | asteriskguy | pretty cool |
| 03:16.59 | karlhaines | asteriskguy, yeah, i thought that was pretty sweet also |
| 03:17.33 | Daejeo1 | anyone help to get rid of echo things |
| 03:17.40 | asteriskguy | yeah, it's in the Asterisk TFOT but the newer version |
| 03:17.54 | asteriskguy | it's not out yet. I got a hold of the uncut version |
| 03:17.56 | Daejeo1 | I have installed TDM400P card |
| 03:18.04 | karlhaines | stkn_, could i query you for a moment? i'd like to talk to you about a project |
| 03:19.04 | asteriskguy | anyone here familiar with using DUNDI & clustering? |
| 03:19.21 | Daejeo1 | ????????????????????????????????????????/ |
| 03:19.48 | karlhaines | what is dundi? |
| 03:19.50 | JT | Daejeo1: are you trying to be disruptive? |
| 03:20.08 | Daejeo1 | JT no sir |
| 03:20.20 | JT | karlhaines: do you know if the pdfs they email are OCRed or not? |
| 03:20.38 | karlhaines | jt: no idea at this point |
| 03:20.39 | Daejeo1 | my keyboard has a trouble |
| 03:20.47 | karlhaines | Daejeo1, lol |
| 03:21.00 | JT | karlhaines: fair enough |
| 03:21.19 | [TK]D-Fender | stridernzl, PING |
| 03:21.23 | asteriskguy | karlhaines: http://www.voip-info.org/wiki-DUNDi |
| 03:23.46 | karlhaines | thanks asteriskguy |
| 03:23.46 | asteriskguy | np |
| 03:23.46 | karlhaines | asteriskguy, wow, that looks sweet! |
| 03:23.58 | asteriskguy | yeah |
| 03:24.08 | *** join/#asterisk inv_arp[work] (i=junya@c-67-191-12-203.hsd1.fl.comcast.net) |
| 03:24.12 | asteriskguy | I know it's possible, just not very much documents out there on it |
| 03:24.19 | demlak | little linux queston.. im writing a batch script.. i know how to put output of, for example "cat /testfile", to a new file with ">> /newfile" but.. how to put the output to a string/variable? |
| 03:24.45 | *** join/#asterisk Strom_M (n=strom@12.175.45.206) |
| 03:24.49 | JT | most fax to email services are incredibly basic |
| 03:25.04 | JT | i wonder when someone will start offering a more fully featured one |
| 03:25.30 | [TK]D-Fender | demlak, time to lear AGI |
| 03:25.37 | karlhaines | JT, yeah, me too! all though, i really wish people would just email and not fax anyway |
| 03:25.54 | *** join/#asterisk santiago (i=santiago@debian/developer/santiago) |
| 03:26.02 | JT | heh |
| 03:26.14 | [TK]D-Fender | learn* |
| 03:26.19 | demlak | [TK]D-Fender no.. it´s a bash script for an embedded device.. no chance to install any software.. just using busybox |
| 03:27.07 | [TK]D-Fender | demlak, Myabe you should be more specific in what you want to do. |
| 03:27.08 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 03:27.50 | karlhaines | any gentoo users in here (other than stkn_, since he's idle ;)) |
| 03:27.52 | demlak | i need the output of this in a string cat /tmpfile | sed '/To:/!d; s/ *(.*)//; s/>.*//; s/.*[:<] *//' |
| 03:28.04 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 03:28.09 | nick125_lappy | karlhaines: I'm a gentoo user, what do you need? |
| 03:28.29 | karlhaines | nick125_lappy, well, i'm actually starting a project and looking for volunteers |
| 03:28.40 | nick125_lappy | karlhaines: What kind of project are you talking? |
| 03:28.47 | karlhaines | nick125_lappy, mind if i msg you? |
| 03:28.54 | nick125_lappy | karlhaines: go ahead |
| 03:37.18 | *** part/#asterisk sevard (i=chuck-th@adsl-71-129-115-242.dsl.irvnca.pacbell.net) |
| 03:38.33 | _VoiceMeUp_COM | real funny how people can have 90 line signatures with all the web sites google can trow at you |
| 03:38.51 | _VoiceMeUp_COM | re: Asterisk High-Capacity Stability in as-users |
| 03:39.14 | JT | rofl |
| 03:41.53 | *** join/#asterisk hacim (n=micah@debian/developer/micah) |
| 03:42.11 | *** join/#asterisk rpm (n=rpm@S010600111155e117.vc.shawcable.net) |
| 03:42.31 | hacim | i've got a friend in brasil who connects to my meetme() and his voice quality is bad, would it be better if he used ulaw or alaw? |
| 03:42.42 | *** join/#asterisk tinrsh (n=claudiu@81.181.94.112) |
| 03:42.47 | nick125_lappy | hacim: What is he currently using? |
| 03:42.47 | tinrsh | hi there |
| 03:43.05 | JT | hacim: does he connect using sip? |
| 03:43.33 | hacim | JT: yeah, via sip |
| 03:43.42 | tinrsh | I have a question, how can I override the CALLERID(num) and CALLERID(name) for the calls outgoing over an sip peer ? |
| 03:43.56 | hacim | nick125_lappy: he can use u-law, a-law and gsm, not sure what he is using |
| 03:44.16 | JT | hacim: maybe his Internet connectivity is no good |
| 03:44.20 | nick125_lappy | ulaw is usually better quality then GSM, but it uses more bandwidth |
| 03:44.34 | hacim | i think he has some packetloss on some hops |
| 03:44.48 | hacim | so I think I want the lowest bandwidth requirement codec |
| 03:44.49 | *** join/#asterisk axisys (n=axisys@ip68-100-236-97.dc.dc.cox.net) |
| 03:45.05 | *** join/#asterisk kiscokid (n=Ron@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 03:45.11 | JT | ilbc can handle packet loss a lot better than the others |
| 03:46.23 | hacim | hrm, the twinkle sip client only can do ulaw, alaw and gsm, maybe he can use a different client |
| 03:49.52 | hacim | actually, twinkle supports ilbc, but fo rsome reason my asterisk isnt allowing any connections with ilbc |
| 03:50.07 | JT | you need to setup ilbc |
| 03:50.33 | tinrsh | I have a question, how can I override the CALLERID(num) and CALLERID(name) for the calls outgoing over an sip peer ? except by changing them from the dialplan ? |
| 03:53.05 | hacim | hmm, so I need to figur eout how to enable ilbc |
| 03:53.17 | *** join/#asterisk bmg505 (n=leon@196.209.182.114) |
| 03:53.57 | hacim | if I set allow=ilbc in sip.conf will other codecs work too? |
| 03:55.14 | JT | depends what your other allow and disallow statements are |
| 03:56.43 | hacim | they are all disabled (default config) |
| 03:56.59 | JT | then nothing else is allowed |
| 03:57.19 | hacim | if they are all commented out -- are they all enabled? |
| 04:00.02 | *** join/#asterisk InHisName (n=InHisNam@c-68-80-56-212.hsd1.pa.comcast.net) |
| 04:03.30 | hacim | hmm, I set 'allow = ilbc' in sip.conf in the [user] section, but not seeing it as a possible codec |
| 04:03.42 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 04:03.57 | *** part/#asterisk santiago (i=santiago@debian/developer/santiago) |
| 04:04.15 | JT | hacim: have you setup the ilbc codec on your asterisk system or not/ |
| 04:04.40 | hacim | JT: um, i didn't realize I had to do that |
| 04:04.49 | JT | you do |
| 04:05.28 | hacim | does that mean installing stuff from http://www.ilbcfreeware.org/ ? dont see anything in a debian package |
| 04:05.49 | JT | i have no idea, i've never needed to install it |
| 04:06.03 | stridernzl | JT: sorry about before I had to run away and do some work. But thanks for your vote of support :) |
| 04:06.38 | stridernzl | JT: remote extensions we will try! |
| 04:07.02 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 04:07.46 | hacim | not much online about it |
| 04:08.04 | JT | stridernzl: sip phones would be the best |
| 04:08.21 | *** join/#asterisk johann8384 (n=johann83@206.80.75.240) |
| 04:08.22 | JT | stridernzl: otherwise ATAs, especially if you need to use a cordless phone |
| 04:09.40 | JunK-Y | JT: do ya know any ATA that could act like a registrar too? ive been asked for that specific purpose today. |
| 04:09.55 | JT | nup |
| 04:09.58 | JunK-Y | personnaly, i dont know any ATA that process register request. |
| 04:10.01 | nick125_lappy | Nooooo, I don't want to have to reboot my server |
| 04:10.11 | JT | a wrt-54g could, as you can run asterisk on it |
| 04:10.25 | JT | nick125_lappy: rebooting? how quaint |
| 04:10.29 | JunK-Y | JT: actually thats not really an ata. |
| 04:10.33 | stridernzl | Yeah I leave that bit unitll we get a working flow happening i think ... |
| 04:10.36 | nick125_lappy | I think the zaptel ebuild has lost it's mind.. |
| 04:10.43 | JT | JunK-Y: no but it's a small cheap embedded device |
| 04:11.04 | nick125_lappy | It says I don't have CONFIG_FW_LOADER enabled in my kernel, but, .config and /proc/config.gz disagree |
| 04:11.04 | JT | stridernzl: never use a cisco ip phone behind nat, btw |
| 04:11.13 | JunK-Y | that could be great if we could install squashfs on pap2. |
| 04:11.43 | JT | will a pap2 even run linux/ |
| 04:11.49 | JunK-Y | ive no clue. |
| 04:12.00 | nick125_lappy | I heard the pap2v2 runs some kind of linux ( |
| 04:12.03 | JT | JunK-Y: why does the ata need to act as registrar |
| 04:12.08 | nick125_lappy | (I'm not sure if it was a rumor or what) |
| 04:13.00 | JunK-Y | jt: cause a customer doesnt have the registrar on that office and would like to connect other soft-phones directly to it. |
| 04:13.08 | JunK-Y | sounds ridiculous, i know. |
| 04:13.37 | JT | do you run virtual pbxes? |
| 04:13.59 | JunK-Y | nope |
| 04:14.23 | *** part/#asterisk kiscokid (n=Ron@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 04:14.38 | JT | why do they need registrars? |
| 04:15.17 | *** join/#asterisk thoughtpolice (n=austin@c75-111-139-133.plaicmtc01.tx.dh.suddenlink.net) |
| 04:15.59 | *** join/#asterisk KaiHanari (n=kai@CPE0013a3bd89d2-CM0011e6c7e1cf.cpe.net.cable.rogers.com) |
| 04:16.14 | KaiHanari | whats the default folder asterisk sounds are in ? |
| 04:16.20 | JunK-Y | cause customer wants to send registers from soft-phones to it. |
| 04:16.26 | nick125_lappy | KaiHanari: /var/lib/asterisk/sounds I think |
| 04:16.28 | JunK-Y | KaiHanari: /var/lib/asterisk/sounds |
| 04:16.38 | JT | JunK-Y: and the softphones will make calls through what? |
| 04:16.47 | *** join/#asterisk sonet (n=darrnh@144.133.204.78) |
| 04:17.01 | JunK-Y | JT: thru a proxy |
| 04:17.09 | nick125_lappy | Wouldn't it just be easier to use a small asterisk box than try to find an ATA that will act as a registrar? |
| 04:17.26 | KaiHanari | ah, there they are :) thanks nick125_lappy , JunK-Y |
| 04:17.27 | JunK-Y | this is what i told him. |
| 04:17.40 | JT | if the soft phones are behind nat, wont they need to register directly to the proxy? |
| 04:17.53 | nick125_lappy | Heck, you could probably setup a Mini-ITX box for about $250 |
| 04:17.56 | JT | a gumstix can be a small asterisk box :) |
| 04:18.11 | JT | gumstix wouldn't be more than USD$150 with ethernet |
| 04:18.20 | nick125_lappy | JT: And it would probably be sufficient CPU power |
| 04:18.56 | JT | especially if it doesn't transcode |
| 04:19.08 | nick125_lappy | $206USD for a 400mhz w/ 64Mb of ram |
| 04:19.13 | nick125_lappy | http://gumstix.com/store/catalog/product_info.php?cPath=26&products_id=170 |
| 04:19.36 | nick125_lappy | pop a $10 CF card in there and you are ready to go |
| 04:20.03 | JT | you don't even need 400MHz |
| 04:20.14 | nick125_lappy | its $20 more for 400mhz than 200mhz |
| 04:21.28 | KaiHanari | Bah, whats a good linux softphone? x-lite doesnt like this pc. |
| 04:22.05 | nick125_lappy | KaiHanari: Error in parsing statement: good + softphone != possibility |
| 04:22.13 | KaiHanari | so true. |
| 04:23.10 | KaiHanari | but i dont have a hardphone yet... gonna get a granstream or something w/ my next pay. need something to do me till then. would be fine if my laptop wasnt broken, x-lite likes that. |
| 04:23.11 | *** join/#asterisk bluelinq (n=bluelinq@dsl-7-36.cofs.net) |
| 04:23.22 | JT | ekiga, idefisk |
| 04:24.10 | bluelinq | Hey guys, I have a 7940 working great. The only issue is that if the person has 2 calls going and a third person calls the extension it sounds busy instead of going to vm. What is the trick? |
| 04:24.10 | nick125_lappy | KaiHanari: By the way, if I were you, I'd probably spend the little bit extra for a PAP2 rather than get the grandstream |
| 04:24.23 | KaiHanari | got any links nick125_lappy |
| 04:24.24 | KaiHanari | ? |
| 04:24.37 | nick125_lappy | KaiHanari: Where are you located at? |
| 04:24.55 | KaiHanari | Newfoundland, can |
| 04:25.50 | nick125_lappy | Just looking at voip supply's CA store, it's $66.95CAD for a PAP2 unlocked |
| 04:26.22 | nick125_lappy | For a Grandstream 286, its $44.95CAD |
| 04:26.38 | KaiHanari | link? |
| 04:26.49 | [TK]D-Fender | ~gs |
| 04:27.01 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 04:27.14 | nick125_lappy | http://www.canadianvoipstore.com/product_info.php?products_id=35 < 286; http://www.canadianvoipstore.com/product_info.php?products_id=1630 - pap2t; |
| 04:27.19 | KaiHanari | JT, im unframiliar with ekiga, registrar is the ip of the pbx right? |
| 04:27.32 | JT | KaiHanari: in your case, yes |
| 04:28.35 | *** join/#asterisk IOscanner (n=IOscanne@cpe-76-187-194-128.tx.res.rr.com) |
| 04:28.40 | bluelinq | hello, I have a sip truck that works great with local calls, but some overseas calls I have a horrendous echo. any tips? |
| 04:29.15 | IOscanner | From the CLI how can I found out how long a call has been going on? |
| 04:29.26 | JunK-Y | IOscanner: core show channel foo |
| 04:29.32 | KaiHanari | oh, nick125_lappy lol. i was gonna get one of those later. first thing im getting is a hardphone... cat5e straight to the phone. |
| 04:29.50 | Daejeo1 | I have installed trixbox, where can I find zaptel source. it is not in /usr/src |
| 04:29.51 | JT | KaiHanari: polycom! |
| 04:29.54 | KaiHanari | bookmarked though ;) good for when i decide to connect the house wiring up |
| 04:29.56 | nick125_lappy | KaiHanari: What kind? Cisco? Polycom? Linksys? |
| 04:30.17 | IOscanner | show channel * shows Elapsed Time: N/A |
| 04:30.29 | KaiHanari | JT, yea, ive used polycom vidphones before, those are nice, had my eye on a voice hardphone on ebay but missed it... |
| 04:30.37 | JunK-Y | IOscanner: take the bridged call. |
| 04:30.39 | nick125_lappy | KaiHanari: I still need to hook my house, its just the rest of the house that doesn't want to move to voip (and save a bunch of money for way more features than qwest) |
| 04:30.40 | KaiHanari | nick125_lappy, granstream i think it was |
| 04:30.50 | nick125_lappy | KaiHanari: What's your budget? |
| 04:31.25 | Daejeo1 | I have installed trixbox, where can I find zaptel source. it is not in /usr/src anyone help plz |
| 04:31.29 | JT | KaiHanari: you're buying a grandstream? :o |
| 04:31.37 | bluelinq | Dae is in the asterisk web site |
| 04:31.44 | KaiHanari | nick125_lappy, right now, $0. when i buy my hardphone, $70 tops before shipping, pref as cheap as possible, while maintaining features (in other words, not the bland d-link cheapies) |
| 04:31.47 | bluelinq | download it |
| 04:32.05 | bluelinq | also look in /tmp I think |
| 04:32.33 | *** join/#asterisk [hC] (n=hardcore@70.68.142.245) |
| 04:32.59 | IOscanner | that got it thanks |
| 04:33.05 | JunK-Y | IOscanner: no problem. |
| 04:33.13 | nick125_lappy | KaiHanari: Hrm, I'd recommend a linksys or a polycom, but, that's a tad bit out of budget (around $144CAD) |
| 04:33.24 | KaiHanari | nick125_lappy, as per hooking the house up, all of my roomates use cellphones, no one has the landline. |
| 04:33.27 | JT | KaiHanari: you won't get a decent phone for $70 unless you get a good deal |
| 04:33.30 | JunK-Y | IOscanner: that would be a good idea to directly changed the NA to the chan->_bridged huh? |
| 04:33.45 | KaiHanari | 1s, i'll pull up what i was looking at on ebay |
| 04:34.15 | nick125_lappy | KaiHanari: Everyone in my house has aleast one cell phone, but, my family is worried about in case of an emergency we couldn't call 911 |
| 04:34.25 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 04:34.46 | JT | nick125_lappy: you should have at least 1 landline for 911 |
| 04:35.18 | nick125_lappy | JT: I was thinking of hooking the Qwest line up to my asterisk box for backup/incoming |
| 04:35.43 | KaiHanari | heh. everyone here doesnt worry about that, cause the cell phone can do that just fine, and its e911, well, my cell anyway. it immediately transmitts GPS data on dial of 911. comforting when hiking and crap, dont have to get voice through, if i get 1s of airtime, i know help is on the way |
| 04:36.07 | JT | nick125_lappy: put an analogue phone directly on the line too |
| 04:36.35 | JT | KaiHanari: cellphones will not survive a disaster (even a small one) |
| 04:36.42 | JT | landlines are better |
| 04:36.46 | *** join/#asterisk evilcyrus (n=evilcyru@bas3-hamilton14-1096561890.dsl.bell.ca) |
| 04:36.53 | JT | and relying on cellphones for hiking is EXTREMELY dodgy |
| 04:36.58 | KaiHanari | one i was looking at is Polycom IP300 Soundpoint, $40 + $90 ship. |
| 04:37.10 | JT | $90 shipping?? |
| 04:37.27 | KaiHanari | its how they make the money, the $40 is buy it now |
| 04:37.36 | JT | ip300 is fairly old now, still does the job though |
| 04:37.40 | JT | the 301 is almost the same iirc |
| 04:37.54 | JT | if you need a speakerphone, it's IP430 and up |
| 04:41.46 | [TK]D-Fender | IP 320 = $95USD w/ speakerphone. |
| 04:41.46 | KaiHanari | just looking for a cool phone i can use, and mess with |
| 04:42.34 | [hC] | the ip320 looks nice. |
| 04:43.47 | [TK]D-Fender | KaiHanari, Serious tip though. Don't cheap out on this. Nother worse than "Buyer's Remorse". |
| 04:44.56 | KaiHanari | lol. im going cheap cause in the end im having at least 2 sip phones. one really nice one (buying later) and one that is well... a sip phone. |
| 04:45.20 | JT | you can never have too many phones |
| 04:45.44 | nick125_lappy | KaiHanari: Well, think if it this way: Would you rather buy a crappy SIP phone, then have to get another crappy one, or just buy one good one? |
| 04:46.33 | [TK]D-Fender | KaiHanari, I would suggest that to start you just get an ATA and use that with an analog phone. much more inexpensive and more readily recyclable. |
| 04:47.10 | JT | i think that's the only sort of device i'm missing at home, ATAs :P |
| 04:47.12 | *** join/#asterisk Daejeo1 (n=chatzill@124.62.150.49) |
| 04:47.58 | [TK]D-Fender | JT : If you're not transfering calls all over the place they're still great. |
| 04:48.55 | JT | i might need one if i decide to get a cordless phone |
| 04:48.58 | JT | actually, that's falso, i'd only get one to play with |
| 04:49.08 | JT | cordless phone will connect to my channel bank just fine |
| 04:49.08 | KaiHanari | i want 1 crappy 1 good :-X one soon, just to have, and learn. one later, so i can move the cheap-ish one to a diff room, and use the good one where i want |
| 04:49.13 | JT | s/falso/false/ |
| 04:49.33 | JT | KaiHanari: why not get an ATA instead of a crappy? |
| 04:50.06 | KaiHanari | cause i dont have a phone, and i was thinking about getting a $5 phone, an ata, and a cheapy :X |
| 04:50.07 | KaiHanari | lol |
| 04:50.42 | KaiHanari | i want 2 ext's to mess with, the 2nd probably not for a few weeks to a mth after the first. then later get a nice fancy phone, when i have money |
| 04:50.57 | JT | so get an ATA instead of a shit SIP phone |
| 04:51.00 | KaiHanari | see, i just moved to the city... money is /kind/ of tight till next mth |
| 04:51.10 | JT | you can connect multiple handsets to a port an an ATA |
| 04:51.22 | mosty | KaiHanari, use a free softphone in the meantime |
| 04:51.32 | [TK]D-Fender | 70$ for a decent 2-port ATA |
| 04:51.45 | [TK]D-Fender | there's your 2 extensions... |
| 04:54.21 | [TK]D-Fender | OMG, the IP 320/330 support a 2.5mm headset instead of the RJ one |
| 04:54.52 | Qwell | is that a bad thing? |
| 04:54.54 | [TK]D-Fender | correction : in ADDITION ot. |
| 04:54.57 | Qwell | nice |
| 04:55.05 | KaiHanari | mosty, thats how the convo started :P |
| 04:55.09 | [TK]D-Fender | Qwell : a very GOOD thing. |
| 04:55.16 | Qwell | with both, yeah |
| 04:55.47 | mosty | KaiHanari, all well a softphone could be free if you already have a microphone or headset |
| 04:55.48 | Qwell | bed time |
| 04:57.04 | KaiHanari | mosty, lol, thats how the convo started, i was wondering about softphone recommends for linux. then mentioned im planning on getting a hardphone soon |
| 04:57.40 | JT | mosty: pc headsets use 3.5mm 3 ring jacks |
| 04:57.42 | JT | not 2.5mm |
| 04:58.02 | JT | actually, dual jacks usually |
| 04:58.19 | [TK]D-Fender | JT : and he never said ANYTHING about what kind.. so get off your rant :) |
| 04:58.19 | mosty | jt, i think you mean [TK]D-Fender |
| 04:59.02 | JT | mosty: ah i thought you said the headset capability would be free if you already had one :) |
| 04:59.10 | JT | [TK]D-Fender: well 2.5mm isn't that useful |
| 04:59.14 | JT | what uses 2.5mm? |
| 04:59.19 | mosty | JT, yes- for a softphone |
| 04:59.44 | [TK]D-Fender | JT : REALLY.... how about the tons of inexpensive headsets already out there in that size? |
| 04:59.50 | [TK]D-Fender | JT : CELL PHONES. |
| 05:00.05 | [TK]D-Fender | JT and every residential cordless phone witha jack. |
| 05:00.13 | [TK]D-Fender | JT : its the NORM. |
| 05:00.21 | JT | does the ip320 take the nokia jack? |
| 05:00.30 | JT | isn't the nokia jack 2.5mm 4 prong or so? |
| 05:00.38 | [TK]D-Fender | JT : Only PC's use the stereo mic / stereo speaker dual prong method. |
| 05:00.44 | JT | uhuh |
| 05:00.56 | [TK]D-Fender | JT : 3 prong 2.5mm |
| 05:01.16 | JT | [TK]D-Fender: does the nokia jack work with the polycom? |
| 05:01.49 | [TK]D-Fender | JT : phones usually suppotr a 3/4 conductor connector. 1 to short for "answer", and the other ones normal. they usually accept dumb ones that way |
| 05:01.58 | Daejeo1 | JT: i have installed trixbox 2.2 . where can I find zaptel source. it is not in /usr/src |
| 05:02.04 | [TK]D-Fender | JT : same jack as described above |
| 05:02.14 | JT | Daejeo1: i don't provide trixbox support |
| 05:02.20 | [TK]D-Fender | Daejeo1, the shouldn't BE any source. Its a binary distro |
| 05:02.32 | [TK]D-Fender | Daejeo1, And read the damn topic. You should know better. |
| 05:02.48 | Daejeo1 | root@asterisk1 ~]# cd /usr/src |
| 05:02.50 | Daejeo1 | [root@asterisk1 src]# ls |
| 05:02.51 | Daejeo1 | asterisk-perl-0.08 freepbx redhat sipsak-0.8.1 |
| 05:03.01 | Daejeo1 | this is what I have in src |
| 05:03.06 | [TK]D-Fender | Daejeo1, Listen up. its a BINARY DISTRO. |
| 05:03.17 | [TK]D-Fender | Daejeo1, there IS NO SOURCE. |
| 05:03.35 | [TK]D-Fender | Daejeo1, the compiled everything and slapped it on an ISO. |
| 05:03.44 | [TK]D-Fender | Daejeo1, You want support, this is not the place. |
| 05:04.23 | Daejeo1 | how can i deal with echo then |
| 05:04.34 | JT | ~trixbox |
| 05:04.46 | jbot | Trixbox is a full linux distro that includes , FreePBX, and other 3rd party add-ons. It is these things on top of which make it seriously painful to support and hence you will find little help here for it. Try asking in #trixbox , or their forums & WIKI at http://www.trixbox.org |
| 05:04.46 | KaiHanari | nick125_lappy, http://cgi.ebay.ca/Asterisk-SIP-IP-Phone-Grandstream-BudgeTone-BT-101-New_W0QQitemZ250113892799QQihZ015QQcategoryZ61840QQrdZ1QQcmdZViewItem |
| 05:04.46 | KaiHanari | is what i was looking at |
| 05:04.54 | JT | KaiHanari: that's an absolute piece of rubbish |
| 05:04.58 | JT | KaiHanari: don't buy it |
| 05:05.01 | [TK]D-Fender | KaiHanari, ..... |
| 05:05.03 | [TK]D-Fender | ~ygwypf |
| 05:05.19 | jbot | i heard ygwypf is You Get What You Pay For. If the sole factor in your decision to purchase a product or service is that it's cheaper than everything else out there, don't be surprised if it's also worse in every other respect than everything else out there. |
| 05:05.20 | [TK]D-Fender | ~gs |
| 05:05.28 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 05:05.28 | KaiHanari | whats wrong with it? |
| 05:05.29 | [TK]D-Fender | ^^^^ |
| 05:05.31 | JT | it's made by grandstream |
| 05:05.39 | JT | and it's one of their worse models too |
| 05:05.42 | *** join/#asterisk luckyone (n=jordan@CPE-65-28-7-102.kc.res.rr.com) |
| 05:05.56 | luckyone | what project is festival in? |
| 05:06.06 | *** join/#asterisk fnordus (n=dnall@24.85.128.203) |
| 05:06.17 | KaiHanari | what about the GXP-2000 ? |
| 05:06.28 | JT | KaiHanari: what about AVOIDING GREANDSTREAM? |
| 05:06.38 | KaiHanari | why? :'( |
| 05:06.52 | mosty | kaihanari: they just plain suck |
| 05:06.56 | [TK]D-Fender | KaiHanari, Perhaps you should reread jbots previous recallection |
| 05:06.58 | Zipper_32 | Because it's the recommendation of people who deal with VOIP all day |
| 05:06.59 | JT | KaiHanari: when heaps of people tell you to avoid rubbish multiple times, for your benefit, there's probably a reason :) |
| 05:07.13 | luckyone | any ideas how to compile festival? |
| 05:07.14 | Zipper_32 | What JT and Fender said. |
| 05:07.18 | KaiHanari | yea, i just like to know the reason |
| 05:07.25 | KaiHanari | and what about soyo? |
| 05:07.26 | JT | flakey firmware |
| 05:07.30 | JT | cheap construction |
| 05:07.34 | JT | terrible audio |
| 05:07.47 | KaiHanari | ok then, soyo? |
| 05:07.54 | [TK]D-Fender | KaiHanari, You need to consider that we who've ben here forever have encountered every common bit of equipment out there and that our experience is probably worth of your consideration before buying your wy into a lot of wasted time and regret. |
| 05:07.55 | JT | never heard of it |
| 05:07.58 | JT | ~phones |
| 05:08.00 | jbot | it has been said that phones is http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. |
| 05:08.42 | Zipper_32 | I've bought loads of cheap phones in the past. Some have survived, but others have simply 'died', quality is/becomes crap, parts break easily, etc. |
| 05:08.45 | Zipper_32 | They're headaches. |
| 05:08.54 | KaiHanari | http://cgi.ebay.ca/ws/eBayISAPI.dll?ViewItem&item=150122718659&fromMakeTrack=true&ssPageName=VIP:watchlink:top:ca |
| 05:08.55 | [TK]D-Fender | KaiHanari, Grandstream produces very inexpensive SHIT. Flakey firmware, crappy construction, feel, audio quality, uner interface, etc. Echo issues and worse |
| 05:08.58 | Zipper_32 | As soon as you buy a high quality phone, you'll never look back. |
| 05:09.08 | KaiHanari | OK , i get the granstream point |
| 05:10.07 | *** join/#asterisk crochat (n=crochat@84-74-150-141.dclient.hispeed.ch) |
| 05:10.40 | friedrich| | I tried it too! |
| 05:11.20 | [TK]D-Fender | KaiHanari, You know what. You seem to aiming bottom dollar and starting with every 2-bit cheap Chinese model produced in sequence. I don't think you're going to learn any other way. GO FOR IT. Buy whichever you feel like and realize that karmic road-kill is only an e-bay away.... |
| 05:12.14 | nick125_lappy | Anyone here know an easy way to convert a stereo ogg into a mono ogg though command line? |
| 05:12.24 | mosty | sox |
| 05:12.34 | KaiHanari | meh. the reason im looking for sub-$70 is cause its not going to be the phone im tied to in the long run. right now i dont have the money for a really nice phone yet, and wont for a few months. i just moved to the city. |
| 05:12.50 | JT | KaiHanari: then why not get an ATA? |
| 05:12.58 | JT | or use a softphone for a little longer |
| 05:14.27 | KaiHanari | i guess i will, unless i can get that polycom ip300... |
| 05:14.31 | nick125_lappy | mosty: Would sox -c 1 <ogg in> <ogg out> work? |
| 05:14.45 | mosty | try it and see |
| 05:15.31 | [TK]D-Fender | KaiHanari, to the negative list add : Soyo, ANYTHING pruduced with a PA168(8) series chip, GRANDSTREAM, Soyo (google up for nightmare stories), Cisco phones not supporting SIP images, Atcom, ArtDio |
| 05:15.51 | [TK]D-Fender | KaiHanari, you mean the one listed for $130 total you saw earlier? |
| 05:16.15 | KaiHanari | after shipping, yes. |
| 05:16.35 | *** join/#asterisk bmd (n=bmd@72.54.252.34) |
| 05:16.44 | nick125_lappy | mosty: It seems to have taken a 4.7MB ogg file and turned it into a 8Kb file...hmm |
| 05:16.45 | KaiHanari | alright, i'll reconsider, $140 after shipping? |
| 05:16.50 | KaiHanari | any decent phone? |
| 05:16.53 | KaiHanari | 2 lines? |
| 05:17.09 | [TK]D-Fender | KaiHanari, You can get a NEW IP 301 for $115, and an IP 320 for About the same including the power brick (I'd personally suggest a PoE Injector) |
| 05:17.10 | *** join/#asterisk ExR90 (n=exr9001@cpe-76-166-105-25.socal.res.rr.com) |
| 05:17.14 | KaiHanari | preferred to have 2 lan ports |
| 05:17.46 | [TK]D-Fender | IP 430 @ 150$ is a great phone. Does it all. |
| 05:17.55 | KaiHanari | i'd love PoE but nothing else i have is PoE compliant. tried some PoE stuff before, the network cards in these systems short those pins to ground. |
| 05:18.00 | nick125_lappy | Aah, I think I figured out why. |
| 05:18.13 | [TK]D-Fender | IP 501 = if you're not planning for POE = you won't think of another for a LONG time. |
| 05:18.21 | nick125_lappy | I was trying to do this: sox -c 1 <source file> <source file> (so it would overwrite the old one) |
| 05:18.28 | ExR90 | I have a 1.4.4 release box with grandstream 2000 phones. BLF stays lit even though call limit and notify is tracking call-state correctly. Any ideas? |
| 05:18.38 | [TK]D-Fender | Anyways, enough for tonight..... |
| 05:18.40 | luckyone | I compiled and installed asterisk-1.4.4 from source on Saturday, I am trying to find out where it put festival so I can start up the festival server, can anyone help me? |
| 05:18.57 | luckyone | it tells me to run /usr/local/festival/bin/festival --server > /dev/null 2>&1 &, but that process dies off quickly |
| 05:19.35 | ExR90 | what does it do when you festival --server only? |
| 05:20.13 | ExR90 | I have a 1.4.4 box and my festival always generates gethostbyname errors in the debug logs in * cli. |
| 05:20.37 | KaiHanari | where online would you be able to buy polycom phones? ebay doesnt seem reliable |
| 05:20.39 | luckyone | ExR90: can you start festival from the * CLI? |
| 05:20.45 | ExR90 | voipsupply.com |
| 05:21.18 | Zipper_32 | KaiHanari: You mentioned that you are in Canada, right? |
| 05:21.21 | Zipper_32 | NB? |
| 05:21.24 | ExR90 | luckyone: Not that I know of, I just run it in another ssh window while having CLI in the first ssh window |
| 05:21.25 | KaiHanari | NL |
| 05:21.28 | Zipper_32 | NL, that's right. |
| 05:21.53 | Zipper_32 | I deal with a company in Ontario, they'll match online prices almost every time: http://www.williamsglobal.com/ |
| 05:22.14 | luckyone | ExR90: hah, i took the > /dev/null off and it says it can't find festival |
| 05:22.20 | ExR90 | ;) |
| 05:22.27 | luckyone | ExR90: where oh where would * have put it? |
| 05:22.36 | ExR90 | * doesnt install it, you must do it |
| 05:22.48 | Zipper_32 | They usually have all the polycom equipment. And you can pay in Canadian dollars. |
| 05:23.10 | luckyone | I read the README.festival in /usr/src/asterisk-1.4.4/contrib and I didn't see how you do that... |
| 05:23.30 | ExR90 | You need to download the festival src, and compile it. |
| 05:23.39 | luckyone | ExR90: from where? |
| 05:23.59 | *** join/#asterisk ModocNet (n=d82e036a@208.106.31.82) |
| 05:24.04 | SwK | anyone seen anything on the Apollo/SIP front yet? |
| 05:24.20 | ExR90 | KaiHanari: canadianvoipstore.com |
| 05:24.22 | SwK | (Apollo being the new light weight runtime from Adobe) |
| 05:24.40 | ExR90 | luckyone: have you even tried looking at google? |
| 05:25.21 | luckyone | ExR90: never heard of it... kidding (wgetting now, but "Download Festival" sounds like a good concert too) |
| 05:25.41 | ExR90 | ;) |
| 05:25.52 | luckyone | ExR90: Shins, Incubus, Wolf Parade, Modest Mouse these are good bands! |
| 05:25.57 | ModocNet | getting a zaptel compliation error when trying to install on CentOS 5 - only thing I have been able to find is: http://forums.digium.com/viewtopic.php?p=50595&sid=af6286e4f0f429f8d8d830fef53f751a |
| 05:26.32 | ModocNet | did a updatedba and locate for xbus_core.c but can't find it |
| 05:27.13 | luckyone | ExR90: festival-1.96 latest version? |
| 05:27.25 | ExR90 | sounds like it. |
| 05:27.51 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 05:28.27 | JT | KaiHanari: that PoE you were using musn't have been IEEE 802.3af compliant |
| 05:29.26 | KaiHanari | JT,late response, and it wasnt the PoE that wasnt compliant, it was the nic's the nics shorted the unused pins to ground. they werent PoE compatable |
| 05:29.49 | JT | KaiHanari: yes, your PoE switch musn't have been compliant |
| 05:29.59 | ExR90 | I have a prob with * 1.4.4 and BLF. Hints are setup and notify'ing correctly. Devices are subscribed. BLF's show up as busy always on grandstream 2000 phone. THis bug looks familiar, but tried its listed fixes no luck. Tried also setting phones as peer and friend in sip.conf, no dice. |
| 05:30.01 | JT | they don't send power unless the device signals it can take it |
| 05:30.03 | ExR90 | any ideas? |
| 05:30.19 | ExR90 | http://bugs.digium.com/view.php?id=8800 |
| 05:30.26 | KaiHanari | PoE switch? i didnt have a PoE switch. this is what im saying .. lol |
| 05:30.51 | JT | KaiHanari: yes, which is why i said it was not IEEE 802.3af compliant |
| 05:31.11 | KaiHanari | lol.... |
| 05:31.18 | *** join/#asterisk keulin (n=cray@AMontpellier-152-1-59-248.w83-197.abo.wanadoo.fr) |
| 05:32.16 | mosty | will using * and # in features.conf prevent me from being able to dial numbers beginning with * or #? or does features.conf stuff only take effect once the call is connected? |
| 05:35.23 | nick125_lappy | mosty: features.conf only effects a currently connected call that is correctly dialed |
| 05:35.40 | nick125_lappy | (It has to have a w or t flag, I cna't remember which) |
| 05:35.49 | mosty | w (or W) |
| 05:36.22 | *** join/#asterisk rudholm (i=rudholmm@nat/yahoo/x-b2e606047d755920) |
| 05:36.25 | luckyone | ExR90: dang it - won't compile... |
| 05:36.27 | mosty | some of my customers have issues dialling *1 *2 etc for one touch recording, but when i change it to * they have no problems |
| 05:38.08 | *** part/#asterisk hacim (n=micah@debian/developer/micah) |
| 05:40.27 | nick125_lappy | Whee, I crashed asterisk 1.2.14 |
| 05:40.47 | ExR90 | luckyone why not |
| 05:42.33 | *** join/#asterisk Putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 05:43.32 | luckyone | ExR90: some integer casting, changing config/config to use gcc3.3 or so |
| 05:44.00 | ExR90 | whoa |
| 05:44.15 | ExR90 | old gcc? |
| 05:45.12 | luckyone | in the INSTALL it said that it only compiles on upto 3.3 |
| 05:45.33 | luckyone | Linux (2.0.30) for Intel (RedHat 4.[012]/5.[012]/6.[01],7.[01],8.0) |
| 05:45.33 | luckyone | <PROTECTED> |
| 05:45.33 | luckyone | <PROTECTED> |
| 05:48.08 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 05:49.06 | luckyone | well, this is fun for tomorrow! |
| 05:49.07 | luckyone | adios amigos! |
| 05:52.20 | nick125_lappy | Ugh...this is a pain in the rear |
| 05:52.33 | ExR90 | see ya |
| 05:53.12 | nick125_lappy | asterisk keeps trying to compile against an old zaptel, and keeps failing. But, I can't upgrade zaptel either |
| 05:54.45 | *** join/#asterisk TheDingy (n=lboyd@h-66-167-118-2.hstqtx02.covad.net) |
| 05:59.33 | russellb | nick125_lappy: why can't you upgrade zaptel? |
| 05:59.52 | nick125_lappy | russellb: Because the ebuild is retarded and says I don't have FW_LOADER enabled |
| 06:00.11 | russellb | have you tried installing from source yourself? |
| 06:00.19 | nick125_lappy | not yet |
| 06:00.39 | russellb | that will probably work ... you don't actually need firmware loading support for all drivers |
| 06:00.50 | russellb | depends on what you use |
| 06:01.02 | nick125_lappy | I just need ztdummy for meetme |
| 06:01.09 | russellb | gotcha, then you don't need it |
| 06:07.39 | *** join/#asterisk grEvenX (n=even@ti500720a080-7918.bb.online.no) |
| 06:08.48 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:11.27 | *** join/#asterisk DarKnesS_WolF (n=wolf@unaffiliated/sherif) |
| 06:20.38 | *** join/#asterisk dlynes_laptop (n=dlynes@d207-216-161-56.bchsia.telus.net) |
| 06:31.07 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 06:38.18 | *** join/#asterisk plasmid2 (n=noway@c-68-46-97-136.hsd1.pa.comcast.net) |
| 06:40.53 | *** join/#asterisk DrukenHME (n=jdumais@CPE001346f4961f-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 06:42.24 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 06:44.33 | *** join/#asterisk mantice (n=richard@125-238-3-57.broadband-telecom.global-gateway.net.nz) |
| 06:46.09 | mantice | Can I use a usb phone such as http://www.trademe.co.nz/Computers/Laptops/Accessories/auction-99874635.htm?p=3 with asterisk? |
| 06:46.41 | *** join/#asterisk matsk (n=mk@194.68.102.173) |
| 06:47.25 | JT | mantice: not directly, all those toys connect to softphones on a pc |
| 06:47.34 | JT | they're just a speaker/mic and keypad |
| 06:47.44 | JT | they don't actually do voip stuff internally |
| 06:47.45 | plasmid | when someone calls my pbx DID# say... 215-xxx-xxxx they tell me that the only way they can reach me is by putting in a 1 in front of it. How can I edit my asterisknow/asterisk/trixbox sip.conf settings to so that ANYONE who calls my pbx doesn't ahve to type in a 1 in front of the xxx-xxx-xxxx number... UNLEss it's international or out of state. |
| 06:48.00 | JT | they rely on a softphone to do all the work |
| 06:48.41 | mantice | JT: So they dont work with out a program on the computer besides Asterisk? |
| 06:48.58 | JT | plasmid: we already explained to you earlier on that this has nothing to do with asterisk, but you seem to have ignored all the explanations you were given |
| 06:49.01 | JT | mantice: correct |
| 06:49.04 | JT | they're toys really |
| 06:50.04 | plasmid | JT, ignored? I got disconnected. Let me see if I can scroll up to find the answer. |
| 06:50.17 | mantice | can you use a modem to get analog into asterisk ? |
| 06:50.17 | JT | plasmid: what happens when someone calls without the 1? |
| 06:50.21 | JT | mantice: no |
| 06:50.50 | plasmid | JT, they get a message saying that they can't connect. |
| 06:50.58 | JT | plasmid: from the telco? |
| 06:51.04 | plasmid | or some sort of error.. especially if they are using a cellphone. |
| 06:51.18 | JT | it means the number isn't local to them |
| 06:51.18 | plasmid | JT, yes.. I believe so. |
| 06:51.32 | JT | you need to get a closer DID if this is a problem |
| 06:52.35 | plasmid | well.. when I call someone out of state via my DID# i punch in: areacode then number: 609-222-2222 for example and NOT 1-609-222-2222. Either way works. |
| 06:53.03 | JT | this has got to be a provider issue |
| 06:53.08 | plasmid | so I am wondering why some ppl tell me they cant call me using the standard xxx-xxx-xxxx format. |
| 06:53.32 | Zipper_32 | plasmid: As I said before, it has to do with other people being outside of the local calling area. |
| 06:53.46 | Zipper_32 | You can't change those calling areas. They're controlled by the telco. |
| 06:53.59 | plasmid | must be then. Thanks. I was trying to pinpoin the issue... but I could have sworn it was an edit from my pbx calling plan to drop the "1" |
| 06:54.34 | Zipper_32 | You can modify *your* pbx to do what ever you like, but you can not change what they dial into their PSTN provider. |
| 06:55.02 | JT | plasmid: easy way to nail it... get one of these phones with trouble to call you while you watch the asterisk CLI with verbosity at at least 10, hell, you can switch on sip debug if you want |
| 06:55.16 | plasmid | Zipper_32, i don't think I am explaining myself right. What I am trying to say is this: I cannot call my pbx from my local cellphone using this format xxx-xxx-xxxx.. I can only call my local pbx using this format 1-xxx-xxx-xxxx. That's all local calls using my cellphone TO the pbx. |
| 06:55.58 | mantice | Could you use this SIP adapter with asterisk ? http://www.trademe.co.nz/Computers/Networking-modems/Other/auction-99846686.htm?p=1 |
| 06:56.03 | JT | plasmid: if the error is from the telco, it's likely outside of asterisk's domain, although there's a very slight chance you have something configured wrong |
| 06:56.33 | JT | mantice: yes but it's made by grandstream, you should avoid their products |
| 06:56.55 | mantice | JT: Thanks for the tip :P |
| 06:57.30 | mantice | JT: would this be a good brand Linksys PAP2-NA Dual Port Analog VoIP Gateway ? |
| 06:57.48 | plasmid | mantice, i have one myself. does the job. :-) |
| 06:57.52 | JT | linsys/sipura is fine for ATAs |
| 06:57.57 | JT | linksys |
| 06:58.10 | plasmid | checking inbound routes.. let me see... |
| 06:58.27 | JT | plasmid: just do the test i said to |
| 06:58.33 | JT | plasmid: will save a lot of time wasting |
| 06:59.49 | mantice | ATA's ? |
| 07:00.00 | JT | ~ata |
| 07:00.21 | jbot | somebody said ata was Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA |
| 07:00.34 | mantice | ah thanks |
| 07:00.39 | mantice | I googled it just then to. |
| 07:02.58 | mantice | If you had a business and wanted ur phones to have a line out would you have to buy heaps of accounts with SIP termination? so you could phone out ? |
| 07:03.11 | JT | no |
| 07:03.20 | JT | well |
| 07:03.49 | JT | depends on what sort of plans were on offer, but i'd want multiple lines to be on the same account |
| 07:05.12 | ido | mantice: most SIP providers let you make as many simultaneous outgoing calls as you want since you pay by the minute |
| 07:05.16 | JT | and i'd never get rid of real lines from a business |
| 07:05.29 | JT | ido: really depends on country |
| 07:05.43 | ido | jt: true |
| 07:05.48 | ido | you could just get an PRI isdn |
| 07:06.18 | mantice | I see. Im just wondering how my work do multiple phones |
| 07:06.39 | JT | they probably have real phone lines |
| 07:06.48 | JT | quite possibly PRI for multiple lines |
| 07:06.49 | ido | yeah, most likely a T1/E1/something |
| 07:07.01 | ido | with like 24-32 banks per T1/E1 |
| 07:07.08 | ido | i mean lines not banks |
| 07:07.08 | justdave | we have one sip account, three different phone numbers come in on that account, and we've had up to 40 concurrently inbound on it |
| 07:07.19 | JT | 23 chs for a PRI T1, 30chs for an E1 |
| 07:07.23 | plasmid | JT, found the issue... it appears on the incoming routes I had specified 1xxxxxxxxxx. I deleted the 1 and now it works. :-) |
| 07:07.27 | ido | 24 and 32, i thought, jt |
| 07:07.33 | mantice | I know we have CAT5 cables that go out to switches for the phone. |
| 07:07.47 | justdave | 24 and 32 total, but you can't use them all |
| 07:07.48 | JT | plasmid: do calls beginning with 1 work now? |
| 07:07.53 | Zipper_32 | plasmid: Unbelievable... |
| 07:07.53 | Zipper_32 | =) |
| 07:07.57 | justdave | T1 uses 1 and E1 uses 2 for signalling |
| 07:07.58 | ido | mantice: then you probably just have SIP/voip set up |
| 07:08.10 | JT | ido: in PRI signalling, 23 and 30, definitely, without a shadow of a doubt |
| 07:08.17 | JT | umm |
| 07:08.23 | JT | T1 uses 1 for a D channel |
| 07:08.24 | mantice | Maybe im on drugs :) |
| 07:08.34 | ido | jt: ok :) thanks, i don't know much about t1/e1 |
| 07:08.37 | mantice | im going to check tomorrow |
| 07:08.38 | JT | E1 uses 1 for a D channel, and 1 for multiframe synch and LoS alarms |
| 07:09.05 | plasmid | JT, checking... |
| 07:09.25 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 07:09.31 | plasmid | JT, yes. :-) |
| 07:09.32 | JT | ido: yes it's 24 and 32 * 64kbit/s, but not in PRI mode |
| 07:09.38 | JT | plasmid: cool |
| 07:09.42 | ido | ok |
| 07:09.50 | JT | ido: not usable B channels anyway |
| 07:10.09 | ido | bah, in the US, there's no reason NOT to use VOIP. |
| 07:10.12 | ido | for a business |
| 07:10.20 | JT | when connecting a T1 to a channel bank, there's 24 chans |
| 07:10.28 | JT | as it's not PRI |
| 07:10.38 | JT | but rather channel associated signalling/robbed bit signalling |
| 07:10.53 | JT | ido: err, WHAT? voip over Internet? |
| 07:10.59 | JT | plenty of reason not to use it |
| 07:11.43 | ido | haha why not jt? there are some really great quality providers out there, and chances are if you are investing in more than 8 phone lines you have the money for a fast enough internet connection |
| 07:12.08 | JT | or a PRI connection |
| 07:12.10 | JT | why not |
| 07:12.12 | mantice | do SIP providers allow you to recieve calls for free ? |
| 07:12.13 | JT | let's see |
| 07:12.24 | JT | Internet connectivity issues |
| 07:12.25 | ido | jt: please take cost into account :) |
| 07:12.29 | JT | ITSP issues |
| 07:12.35 | JT | routing issues |
| 07:12.43 | JT | ido: i am |
| 07:13.12 | ido | jt: ok, well, i am not an expert -- i have only begun dabbling in telephony beyond the simple hacking about asterisk. |
| 07:13.15 | ido | :) |
| 07:13.21 | JT | saying VoIPoI is the way to go most of the time for business' outside lines is near insanity |
| 07:13.28 | JT | it's mainly quality and reliability issues |
| 07:13.51 | ido | but for inside lines it is good? |
| 07:13.56 | Zipper_32 | ido: It would be silly for a company to use pure VOIP over an internet line when a dedicated connection to the provider via T1/PRI is available at the same cost, or slightly higher. Especially since the T1's are required to have far greater uptime than any internet service. |
| 07:14.04 | JT | well most business LANs are pretty reliable |
| 07:14.08 | JT | and have ample bandwidth |
| 07:14.37 | ido | ok, my noobickle is showing :) |
| 07:14.45 | Zipper_32 | If a business can't talk to its customers, that business will die. Simple as that. |
| 07:14.57 | Zipper_32 | *Disclaimer: Pending what type of business it is. |
| 07:15.03 | JT | yeah |
| 07:15.09 | JT | say pizza hut |
| 07:15.19 | JT | they like their callcentre lines to work |
| 07:15.28 | justdave | right now we've got 4 analog lines with a failover to a SIP provider if they're all busy |
| 07:15.44 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 07:15.44 | justdave | we're working on getting a PRI someday when the phone company gets off its butt to put the cable in |
| 07:15.45 | JT | argh analogue, kill, destroy :P |
| 07:16.29 | justdave | (been on order for about 5 months now I think) |
| 07:16.36 | JT | wow |
| 07:16.39 | JT | that's terrible |
| 07:16.44 | ido | so i'm looking at getting a Sipura SPA-3102 (or SPA-3000) for my home asterisk setup |
| 07:16.45 | JT | only in america? :P |
| 07:17.11 | ido | anyone used the sipuras or can recommend an external 1 or 2 line FXO? |
| 07:17.21 | justdave | welcome to the world of monopoly carriers. :) |
| 07:17.38 | Zipper_32 | JT: I'm looking to setup a new retail system, with 8 lines, including 1 for fax, and 1 for a POS system. (10 total), would a PRI system be feasable for that purpose? |
| 07:17.38 | justdave | sure, there's plenty of competition, but they all rent lines from the same company so it doesn't really matter if someone needs a line run |
| 07:17.51 | JT | ido: sipura/linksys are the most commonly used |
| 07:18.38 | JT | Zipper_32: yes, but it's probably easiest to get analogue lines for the fax and POS |
| 07:18.47 | mantice | I was watching a video about Asterisk and the guy said some people cut there lines coming into the house? any one know what thats about ? |
| 07:18.55 | JT | Zipper_32: in the US the smallest fractional PRI you can usually get is 8 channels |
| 07:19.10 | JT | mantice: no phone service |
| 07:19.16 | JT | mantice: i think it's a stupid idea |
| 07:19.20 | JT | but some people do it |
| 07:19.25 | justdave | mantice: never had them hooked up at my current house. |
| 07:19.45 | justdave | when I moved (1.5 miles from the old house) I tried to get SBC to move the number to the new house... |
| 07:19.50 | ido | cut their lines? |
| 07:19.53 | justdave | and they claimed my new house didn't exist so they couldn't do it |
| 07:19.53 | ido | what does that mean? |
| 07:20.03 | JT | ido: disconnected their analogue phone line |
| 07:22.27 | ido | oh |
| 07:22.28 | justdave | so I just went and got VoIP service and ported the number |
| 07:22.28 | ido | link to video please :) |
| 07:22.28 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 07:22.29 | JT | justdave: dsl or cable? |
| 07:22.29 | justdave | fortunately I have fairly reliable broadband here, very seldom have outages |
| 07:22.29 | justdave | cable |
| 07:22.29 | justdave | dsl needs a phone line which they couldn't hook up :) |
| 07:22.29 | JT | cable isn't my idea of reliable broadband |
| 07:22.29 | justdave | depends where you live |
| 07:22.29 | justdave | it's pretty reliable here |
| 07:22.30 | ido | jt: optonline is amazing |
| 07:22.30 | ido | and reliable |
| 07:22.30 | JT | justdave: what about during power failures? |
| 07:22.30 | ido | in the northeast |
| 07:22.30 | JT | ido: during power failures? |
| 07:22.31 | justdave | cable modem and the router are on a UPS |
| 07:22.31 | JT | no |
| 07:22.31 | *** join/#asterisk af_ (n=getsmart@81-174-8-57.f5.ngi.it) |
| 07:22.31 | JT | i asked about the cable network |
| 07:22.31 | JT | they have amplifiers in the streets to make them work |
| 07:22.32 | JT | cable networks do not work when these have no power |
| 07:22.32 | justdave | ah, no clue. power doesn't go out very often here either |
| 07:22.32 | JT | i see :P |
| 07:22.32 | justdave | most of the power cables are underground |
| 07:22.32 | mantice | Im going to get a SIP supplyer and my Linksys PAP2-NA and I will have a phoneline :) |
| 07:22.36 | JT | even so |
| 07:22.50 | Zipper_32 | JT: As someone who has never setup a PRI based Asterisk install, is it really any more difficult than an arrangement of FXO/FXS's on an analog card? |
| 07:22.51 | JT | if you're lucky, your cable network's amps will have some batteries |
| 07:22.54 | justdave | they started getting rid of the above ground ones when the power companies started figuring out it was cheaper to bury them than to come fix them every winter during the ice storms :) |
| 07:23.04 | JT | these might last a half hour at most |
| 07:23.21 | JT | Zipper_32: in some respects, easier |
| 07:23.34 | JT | Zipper_32: some people have more trouble, some have less |
| 07:23.37 | Zipper_32 | And echo is non-existent, correct? |
| 07:23.45 | JT | but if you know what you're doing, their easier |
| 07:23.46 | Zipper_32 | But it's not a huge ordeal? |
| 07:23.51 | JT | unfortunately not correct |
| 07:23.57 | JT | your circuit won't make echo |
| 07:24.07 | JT | remote end analogue circuits will |
| 07:24.13 | Zipper_32 | Understood. |
| 07:24.33 | JT | that's why you should always spend the extra on a pri card with hardware echo cancellation |
| 07:24.53 | Zipper_32 | Thanks, I appreciate it. It looks like I'll be doing a PRI if the numbers add up correctly. |
| 07:25.03 | Zipper_32 | any particular model you recommend? |
| 07:25.19 | justdave | anyone know if it's possible to change the soft-function buttons on the bottom of the idle screen on Polycom phones? (IP430/IP501) |
| 07:25.23 | JT | whatever digium or sangoma looks most appropriate to your needs |
| 07:25.36 | Zipper_32 | justdave: Not of my knowledge, |
| 07:26.12 | justdave | 501's fine actually, it has a button for Call Lists. on the IP430 if you want to look at your callerid history it's buried like 5 levels deep in the menu |
| 07:26.14 | ido | has anyone here used an astribank before? |
| 07:26.20 | JT | Zipper_32: if you EVER have an IVR or anything like that, you'll love having digital |
| 07:26.31 | justdave | there's two buttons that don't do anything on the idle screen, would be nice to put CallLists on one of them |
| 07:26.31 | Zipper_32 | Why is this? |
| 07:26.34 | JT | analogue call progress signalling is mediocre |
| 07:26.57 | JT | Zipper_32: it's very hard for a computer to tell what stage of a call analogue is in |
| 07:27.09 | Zipper_32 | *I have an IVR running on analog right now... |
| 07:27.14 | JT | very easy with digital, they're all expressed in Q.931 signalling messages |
| 07:27.22 | Zipper_32 | Rings 1 to 1.5 times, and then the IVR picks up, |
| 07:27.45 | JT | Zipper_32: you can hang up the remote end instantly, and you can answer a call before the calling end even hears any ringing indication in some cases |
| 07:28.26 | Zipper_32 | That's right, I've experienced that before, |
| 07:28.27 | justdave | if you have callerid on an analog time you have to let it get to the second ring before you answer or you'll lose the callerid data |
| 07:28.27 | Zipper_32 | Which now explains why I noticed the difference. |
| 07:28.27 | justdave | s/time/line/ |
| 07:28.39 | justdave | haha, nice bot :) |
| 07:29.18 | JT | Zipper_32: also, you can set outgoing callerid on a per call basis |
| 07:29.39 | JT | and receive multiple calls to multiple inbound numbers |
| 07:29.41 | Zipper_32 | Dynamically though Asterisk? |
| 07:29.55 | Zipper_32 | erm, set the callerID dynamically? |
| 07:29.56 | JT | the closest thing to that with analogue is distinctive ring, which is flakey and limited |
| 07:30.00 | JT | Zipper_32: yes |
| 07:30.13 | Zipper_32 | Very interesting, =) |
| 07:30.20 | JT | and receive multiple calls to multiple different numbers, and take different actions depending on the numbers |
| 07:30.22 | Zipper_32 | Well thanks, I'll keep these points in mind. |
| 07:30.40 | JT | the signalling of analogue is absolute rubbish compared to digital :) |
| 07:31.54 | mantice | If its free to call any one in your area and you get a SIP provider with a phonenumber from your area. If people ring ur SIP number will it be free for them ? |
| 07:31.55 | JT | also you can send and receive many different types of call termination cause codes, which a computer can act on |
| 07:32.14 | ido | mantice: local calls are local calls |
| 07:32.37 | *** join/#asterisk darkskiez (n=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 07:32.59 | Zipper_32 | mantice: What ido said. |
| 07:33.40 | Zipper_32 | JT: Thanks for your help. I'm off to sleep. I may need to come to you again one of these days for another word or two of advice. |
| 07:33.50 | *** join/#asterisk h3x (i=Justino@64.192.116.17) |
| 07:34.18 | JT | Zipper_32: no probs :) |
| 07:34.53 | docelmo | PUKE'N RALLY! |
| 07:35.42 | mantice | ok cool |
| 07:37.40 | JT | mantice: as long as your provider actually gave you a DID on the PSTN, not just a meaningless "SIP number" that works on their network :) |
| 07:38.30 | mantice | Direct Inward Dialing ? |
| 07:38.41 | JT | yes |
| 07:40.33 | h3x | anybody looked at 9694 |
| 07:40.51 | plasmid | Hmm.. now I need a company that gives 1-888#'s with payphone service. vitelity.net (my itsp) gives out 1-888 but they don't work with payphones. Hmm. |
| 07:41.37 | h3x | of course, theres a .50 surcharge |
| 07:41.37 | h3x | heh |
| 07:42.11 | plasmid | i dont mind the surcharge as long as it works with payphones. |
| 07:42.52 | h3x | that is because hardly anybody is bright enough to deal with ANI-II digits on a voip switch |
| 07:43.03 | h3x | :) |
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| 07:44.03 | plasmid | good point. I should carry a cell phone but I don't.. and I loathe 2 yr contracts with any of these cellphoen companies. Pre-paid cell phoens are a ripoff. |
| 07:44.45 | JT | freecall numbers must be really cheap in the US with the rate you guys buy them at :P |
| 07:45.05 | JT | oh yes, cellphones are a dismal situation in the US |
| 07:45.13 | JT | where unlock phone are not very common at all |
| 07:45.21 | JT | and all these horrible contracts :P |
| 07:45.24 | JT | unlocked |
| 07:45.36 | h3x | some judge ruled that a carrier has to help you unlock your phone now |
| 07:45.40 | h3x | if you call them |
| 07:45.48 | plasmid | yup.. i read that right... unlock the damm phones. |
| 07:45.52 | h3x | and you arent under contract |
| 07:46.32 | plasmid | i carry my voip phone aroudn but these metropolitan philadelphia greedy bastards won't give enough hotspots... so it doesn't work half the time. |
| 07:46.39 | JT | h3x: ANI-II isn't exactly rocket science, is it? |
| 07:46.58 | h3x | theres no standard way to translate ANI-II to SIP |
| 07:47.11 | JT | ah yes, SIP |
| 07:47.13 | h3x | it makes a little more sense on SIP-T |
| 07:47.21 | JT | but does it need to get translated to SIP? |
| 07:47.36 | h3x | well the easiest thing to do is just suffix the ANI |
| 07:47.50 | h3x | and i think the MAX TNT lets you do that for instance |
| 07:47.51 | JT | does it need to go as far as the customer site? |
| 07:47.54 | h3x | but many of the voip gateways dont |
| 07:48.04 | h3x | it needs to get far enough to get into billing |
| 07:48.08 | JT | right |
| 07:48.24 | JT | i'm not sure if ANI-II is even used here |
| 07:48.27 | JT | Australia... |
| 07:48.31 | h3x | oh |
| 07:48.36 | JT | my logs log ANI-II and it's never set |
| 07:48.37 | h3x | maybe not |
| 07:48.54 | h3x | its fairly common on MF signalling at least |
| 07:49.13 | JT | hrm |
| 07:49.26 | JT | is it normally unset if the call is not from a payphone? |
| 07:49.49 | h3x | http://www.nanpa.com/number_resource_info/ani_ii_assignments.html |
| 07:51.48 | JT | right, so it's probably not used here |
| 07:53.03 | h3x | its probably in ETSI SS7 somewhere |
| 07:53.17 | h3x | ISUP |
| 07:53.37 | JT | ss7 probably, isup, maybe |
| 07:53.53 | *** join/#asterisk Dibbler_ (n=Dibbler@host217-45-198-229.in-addr.btopenworld.com) |
| 07:53.54 | h3x | isup is used for setting up and tearing down calls |
| 07:55.00 | JT | i know |
| 07:55.15 | JT | i'm not sure if it would have the equivalent of ANI-II in it |
| 07:58.02 | *** join/#asterisk qdk (n=qdk@193.226.189.115) |
| 07:59.57 | JT | mantice: does no-one in nz use ebay? |
| 08:04.00 | Zipper_32 | JT: I just found another *sweet* function of a PRI from my local provider: "Release Line Trunking. Transfer outside calls to another outside number without relying on two B channels to maintain the connection - the PRI enables your equipment to release the B channels and transfer responsibility for maintaining the connection to us, freeing up resources on your PBX for additional calls." |
| 08:05.25 | JT | yeah but that's hard |
| 08:05.30 | Zipper_32 | Is it? |
| 08:05.33 | JT | probably not supported properly yet in asterisk |
| 08:05.37 | JT | yes |
| 08:05.37 | mantice | JT: no they use trade me |
| 08:05.51 | JT | limited usefulness anyway imho |
| 08:05.58 | JT | mantice: wonder why |
| 08:06.55 | mantice | brb |
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| 08:24.16 | JT | hmm |
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| 08:36.03 | mantice | JT: Trademe sucks. |
| 08:36.50 | JT | mantice: why do people use it then? |
| 08:39.13 | mantice | JT: I dislike the selling fees. Yeah its NZ ebay as you could call it |
| 08:39.46 | JT | is there an ebay nz site? |
| 08:40.48 | mantice | JT: Yeah but its not used. |
| 08:42.33 | JT | hrm |
| 08:42.43 | JT | mantice: do you know why trademe is used? |
| 08:42.48 | *** join/#asterisk ltd (n=z@nox.amused.net) |
| 08:44.11 | mantice | I think its because trademe came before ebay nz |
| 08:44.43 | mantice | JT: Also it spread mouth 2 mouth and on new zealand media like wild fire. |
| 08:45.09 | *** join/#asterisk ctaloi (n=ctaloi@pool-71-176-69-82.syrcny.fios.verizon.net) |
| 08:45.16 | JT | right, i really hate the site design, i hope ebay buy it :P |
| 08:46.01 | mantice | :D |
| 08:46.35 | mantice | KT: Look at this www.italk.co.nz/ |
| 08:47.16 | mantice | I think they are a SIP provider. |
| 08:48.09 | JT | lol |
| 08:48.09 | JT | iTALK, the smart way to make phone calls! |
| 08:48.11 | JT | iTALK is an innovative new service that allows any broadband user to make and receive calls over the internet at heavily discounted rates. |
| 08:48.18 | JT | translation: we are an ITSP |
| 08:50.29 | mantice | so thats what a SIP service is called ? |
| 08:50.39 | JT | ~itsp |
| 08:50.54 | jbot | itsp is, like, Internet Telephony Service Provider. An ITSP is a "VoIP Phone Company" |
| 08:50.55 | mantice | well most ITSP offer SIP? |
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| 08:58.10 | mantice | Have you looked at http://www.freecall.com apparently you can call any one for free and you can use ur Asterisk box |
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| 09:15.48 | mantice | if you wanted ur whole house on Asterisk you would have to buy a for every 2 phones you have. |
| 09:17.12 | JT | what? |
| 09:17.15 | JT | missing words? |
| 09:17.33 | mantice | yeah |
| 09:17.40 | *** join/#asterisk olinux (i=olinux@ip68-107-12-15.sd.sd.cox.net) |
| 09:17.56 | mantice | if you wanted ur whole house on Asterisk you would have to buy a Linksys PAP2-NA for every 2 phones you have. |
| 09:19.37 | JT | i guess, if they all had to be different extensions |
| 09:19.43 | JT | or you could buy sip phones |
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| 09:21.15 | mantice | dont sip phones have to be connected via ethenet |
| 09:21.43 | JT | yes |
| 09:23.16 | mantice | So I could have it so when you pick up the phone dial 1 for outside line push 2 for VOIP and 3 to check messages :) |
| 09:24.07 | JT | sure |
| 09:24.28 | mantice | My house could get interesting. |
| 09:27.32 | olinux | my polycom 501 is not connecting from off site, |
| 09:28.06 | olinux | i thought i heard someone say they have a problem with ips, but domain names work ok |
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| 09:32.53 | shrewd1980 | I've installed the lastest asterisk on a mac os x machine and can't find much information on how to configure it, does someone have a good website on mac os x and asterisk? |
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| 09:41.18 | BrokenNoze | Hi. for some reason my network is running really slow. if I change the router from 192.168.0.50 to 192.168.0.1 it runs fast again. I know this isn't the right place ( where is) for the question but its effecting how long it takes for my endpoints to ring. anyone know what I might have done wrong? |
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| 09:44.58 | tengulre | hi,all |
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| 09:56.35 | penguinFunk | BrokenNoze: sounds like traffic shaping issue ? |
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| 10:37.04 | Siya | BrokenNoze: check your connections (full/half duplex Ethernet) |
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| 12:09.35 | mosty | does bri usually receive DNID data for incoming calls (with misdn)? |
| 12:09.40 | DrukenLPY | god damnit.... we had a massive electrical storm come threw last night.... |
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| 12:14.52 | SoftIce | hi please can somebody look at this post and give me their input? |
| 12:14.53 | SoftIce | http://paste.linux-vserver.org/1830 |
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| 12:22.31 | DrukenLPY | SoftIce: what about it? |
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| 12:24.07 | SoftIce | DrukenLPY: well any idea why ? |
| 12:24.28 | SoftIce | I have a dial commond on a box, passing it to another box I have the account details in iax.conf that passes to a context in extensions |
| 12:24.37 | SoftIce | I cant see why it wont just answer the call |
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| 12:25.33 | DrukenLPY | are you passing to s@ ? |
| 12:25.41 | SoftIce | I dont understand what thatt means |
| 12:25.45 | SoftIce | you mean s,1,answer ? |
| 12:25.50 | DrukenLPY | yeah |
| 12:25.58 | DrukenLPY | s=start @context |
| 12:26.16 | SoftIce | so you would like to see a snip of extensions.conf ? |
| 12:27.18 | SoftIce | http://paste.linux-vserver.org/1832 |
| 12:30.01 | DrukenLPY | i had that problem a while ago... it was an auth issue... |
| 12:30.10 | DrukenLPY | i'd verify your auth between the servers |
| 12:30.17 | SoftIce | user/pass is correct |
| 12:30.26 | DrukenLPY | what's the dial line look like? |
| 12:30.35 | SoftIce | i'm using a basic dial command iax2/user:pass@ip |
| 12:31.13 | DrukenLPY | iax2/user:pass@ip/s |
| 12:31.25 | SoftIce | what is the /s for? |
| 12:31.35 | Aurs | extension |
| 12:31.38 | DrukenLPY | for the start exten |
| 12:32.17 | SoftIce | so it has to have a /s |
| 12:32.17 | SoftIce | ? |
| 12:32.20 | SoftIce | for it to work |
| 12:32.46 | DrukenLPY | if you want to reduce all problems, use iax2/user:pass@ip/s@scds |
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| 12:33.42 | Mavvie | I wonder what he is smoking. |
| 12:34.13 | [TK]D-Fender | SoftIce: No, it has to be SOMETHING |
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| 12:34.30 | Strom_M | he's smoking the buzzword bong, Mavvie |
| 12:34.55 | Mavvie | correct me if I'm wrong, but there is no echocancellation done in RTP streams is there? |
| 12:34.57 | SoftIce | and that will force that, exten => s,1,dial(iax2/9043211:9043211@ip) |
| 12:34.59 | SoftIce | so that wont work |
| 12:35.02 | DrukenLPY | hehe why, you want some? |
| 12:36.10 | SoftIce | DrukenLPY: so /s@scds |
| 12:36.29 | SoftIce | will force box a to read the context of iax.conf on box 2 well context @scds |
| 12:36.49 | DrukenLPY | you can pass along a context with iax2... |
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| 12:37.04 | DrukenLPY | you can allow a peer/user to be in more than one context |
| 12:37.43 | SoftIce | DrukenLPY: yes, but what im trying to ask is, dial .... user:pass@ip/s@scds, wil causes this box i'm using the dial string on to read context on the box i'm dialing in iax.conf ? |
| 12:37.52 | SoftIce | it will read scds in iax.conf on the box i'm dialing ? |
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| 12:40.28 | enioreh | hi |
| 12:41.46 | SoftIce | also the box with the pri, should that do the answer |
| 12:41.52 | SoftIce | or should it do a straight dial out ? |
| 12:42.11 | SoftIce | and if not does it make a difference if immediate is set to yes or no? |
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| 12:53.41 | DrukenLPY | SoftIce: no, you don't have the box with the pri do the answer, the box using the call should do the answer, and it'll do just a stright dial out, but WITH context so it'll go into that context on the receiving server |
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| 13:20.07 | kippi617 | hey |
| 13:20.35 | kippi617 | how can I just reload zapata? |
| 13:20.37 | kippi617 | .conf |
| 13:21.48 | kippi617 | just done a stop and start gussing that will do it |
| 13:22.45 | enioreh | asterisk -rx "reload chan_zapata.so" perhaps |
| 13:23.35 | [TK]D-Fender | kippi617: "reload chan_zap.so" |
| 13:23.51 | *** join/#asterisk MrTelephone (n=MrTeleph@h697179-171.picriverisp.net) |
| 13:24.00 | MrTelephone | has anyone tried d4 framing for an adit 600? |
| 13:24.34 | MrTelephone | My card says its sending RBS bits A=0 B=1 but the adit says its receiving A=1 B=1 which means network disconnect |
| 13:24.42 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 13:24.44 | Katty | morning lovables! |
| 13:24.48 | MrTelephone | I did some t1 analysis and everything seems to be ok |
| 13:25.22 | Katty | anyone in here work for polycom? |
| 13:25.31 | MrTelephone | no but i work with polycom 501s |
| 13:25.40 | Katty | well...goshdangitanyhow. |
| 13:25.58 | MrTelephone | i got a vx6000 in my van in a grocery bag |
| 13:25.58 | Katty | that's what we use here, MrTelephone... they're nicey nice (= |
| 13:26.09 | coppice | would they admit that? :-) |
| 13:26.09 | MrTelephone | yeah I like them better than the cisco 7960 |
| 13:26.34 | MrTelephone | the cisco phone I have is staticy.. maybe a bad handset wire or something |
| 13:26.39 | [TK]D-Fender | Katty: Mew. |
| 13:26.39 | MrTelephone | katty do u have an adit 600? |
| 13:26.39 | Katty | but i don't wanna /gasp/ call polycom |
| 13:26.46 | Katty | [TK]D-Fender: mew. |
| 13:26.50 | MrTelephone | polycom support pisses me off |
| 13:26.54 | Katty | polycom peoples aren't very nice to me. |
| 13:27.02 | MrTelephone | what are you trying to do? |
| 13:27.04 | Katty | i have to pretend to be uberdumb before they'll even talk to me like a human. |
| 13:27.08 | MrTelephone | it took me a day to get tftp provisioning working |
| 13:27.15 | Katty | they're like, oh..a girl..psh |
| 13:27.20 | [TK]D-Fender | Katty: w'sup? |
| 13:27.31 | Katty | [TK]D-Fender: just tryin to do a lil homework for a client. |
| 13:27.40 | MrTelephone | yeah adit 600 is all female and I was kind of worried that theyd just pass me on but they actually knew what they were talking about |
| 13:27.44 | Katty | [TK]D-Fender: it's more...video conferencing...related. |
| 13:27.59 | [TK]D-Fender | Katty: Ah... got a reseller/integrator around? |
| 13:28.02 | MrTelephone | adit 600 support I meant to say |
| 13:28.05 | Katty | [TK]D-Fender: yes'r. |
| 13:28.17 | [TK]D-Fender | Katty: Well call them up.. that IS their job actually.. |
| 13:28.25 | Katty | [TK]D-Fender: eh.... |
| 13:28.38 | Katty | [TK]D-Fender: voipsupply only works with mostly voip stuff... |
| 13:28.38 | MrTelephone | if your specialized with a linksys router you can be a specialist with polycom video products |
| 13:28.43 | Katty | [TK]D-Fender: they're not all that up on the video side just yet |
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| 13:31.22 | MrTelephone | unless if you talk to an engineer for support they are going to talk to you like an idiot |
| 13:31.39 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 13:31.46 | Mercestes | Help! My MOH sounds like it was remixed by anthrax |
| 13:31.47 | MrTelephone | because they have a little book in front of them that they use to guide you through your problems :( |
| 13:31.49 | cpm | but what if I am an idiot? |
| 13:32.07 | MrTelephone | and they never worked with the equipment themselves :( |
| 13:32.40 | MrTelephone | Zap/41-1 s@bandoffice:1 Rsrvd (None) |
| 13:32.43 | MrTelephone | what does that mean |
| 13:32.50 | MrTelephone | Rsrvd? |
| 13:33.13 | Katty | Mercestes: oh noes! |
| 13:33.18 | Mercestes | aye |
| 13:33.32 | MrTelephone | I can't get my adit 600 to recognize the right RBS bits from my pri card so I'm going to try d4 framing instead of esf |
| 13:33.35 | Dovid | hungry* |
| 13:33.35 | Mercestes | my phone server died yesterday. controller card |
| 13:34.14 | Dovid | yummy |
| 13:35.12 | Katty | so, today, i'mma play with fstab. |
| 13:35.34 | Katty | and maybe rc.local if i'm feeling brave. |
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| 13:37.32 | Dovid | hehe |
| 13:37.41 | Dovid | what r u using rc.local for ? |
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| 13:37.52 | Katty | hewwo mog. |
| 13:37.57 | Katty | Dovid: not sure yet. |
| 13:38.02 | Katty | Dovid: probably to start some apps. |
| 13:38.11 | Katty | Dovid: or run some processes. |
| 13:38.19 | Dovid | not that hard |
| 13:38.27 | Katty | i'm sure it isn't. |
| 13:38.37 | Dovid | just go to the end of the file and in what u want it to start |
| 13:39.11 | Katty | one thing at a time, fstab is first ;) |
| 13:39.42 | MrTelephone | how do you know if a t1 controller is screwed |
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| 13:41.19 | coolbeans | I'm sure this is something simple but I've never seen it. I installed 1.4.4 and configured extconfig and res_mysql. Everything "appears" to be working fine but I get this when I do a sip reload: sipsock_read: Recv error: Bad file descriptor ... over and over and over. What's weird is it doesn't do it on the initial asterisk start. Any clues? |
| 13:42.59 | [TK]D-Fender | Katty: If you want to get some better info out of them, call up asking for a local reseller (NOT VoipSupply) |
| 13:43.30 | Katty | i suppose i could do that. |
| 13:44.14 | Katty | erk! |
| 13:45.48 | Mercestes | whose idea was it to remove mpg123-59r from portage? dmanit. If I remove mpg123 will it use native without a reboot? |
| 13:46.14 | Katty | file: make my foot stop hurting :< |
| 13:46.18 | Mercestes | I'm trying to get asterisk to mode=files but "reload" isn't helping |
| 13:46.35 | Katty | woohooooooo!!! |
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| 13:48.20 | Mercestes | The answer to my question is "no, if you remove mpg123 it will just go to silence when you put someone on hold." |
| 13:48.54 | Katty | i love how polycom's find a partner page just forwards you to the contact us page. |
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| 13:54.28 | [TK]D-Fender | Mercestes: Compiled Asterisk-addons for MP3 support? |
| 13:54.54 | *** join/#asterisk luckyone_ (n=hidden@CPE-65-28-7-102.kc.res.rr.com) |
| 13:55.18 | luckyone_ | I am having trouble compiling speech_tools for festival, can anyone help me out? |
| 13:55.53 | *** join/#asterisk kova_ (n=kova@tech.quentris.be) |
| 13:56.41 | kova_ | Hi all! Anyone with experience in video conferencing? |
| 13:57.34 | Katty | kova_: gettin there. |
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| 14:01.42 | Hymie | luckyone_: I hope you don't think festival will be an impressive experience :( |
| 14:02.05 | Mercestes | [TK]D-Fender, yea, checked that. |
| 14:03.35 | kova_ | does the 'v' option in MeetMe actually work? |
| 14:04.40 | coolbeans | I'm sure this is something simple but I've never seen it. I installed 1.4.4 and configured extconfig and res_mysql. Everything "appears" to be working fine but I get this when I do a sip reload: sipsock_read: Recv error: Bad file descriptor ... over and over and over. What's weird is it doesn't do it on the initial asterisk start. Any clues? |
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| 14:07.30 | Katty | [TK]D-Fender: would you look at my fstab mount command and double check it for me? |
| 14:07.53 | [TK]D-Fender | Katty: I could, but I'm very inexperienced with it personally. |
| 14:08.03 | Katty | k |
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| 14:17.38 | toot | anyone from digium on - just emailed info: No permission to create tickets in the queue 'Sales' |
| 14:17.39 | toot | :) |
| 14:17.55 | livesN[box] | hey guys is there a way to set certain dynamic agents in a queue so that calls won't come to them unless all the other agents are busy ? |
| 14:20.21 | livesN[box] | looks like maybe penalities. |
| 14:20.42 | Katty | anyone know where the /real/ rc.local file is in debian |
| 14:22.03 | mosty | katty: locate rc.local |
| 14:22.15 | Katty | mosty: there's a bazillion. |
| 14:22.20 | Katty | mosty: hence wnating to know where the /real/ one is. |
| 14:22.24 | Dovid | katty: it may be in /etc |
| 14:22.37 | Katty | file: do you use debian, dear? |
| 14:22.59 | mosty | Katty, what do you mean by real? |
| 14:23.10 | Katty | i presume that most of these are symlinks. |
| 14:23.19 | mosty | ls -l will tell you for sure |
| 14:23.31 | tzanger | file'll tell you too |
| 14:23.48 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 14:23.50 | Katty | i see. |
| 14:24.03 | mosty | locate rc.local | xargs ls -l $1 |
| 14:25.33 | mosty | and you can open a symlink just the same as a "real" file anyway |
| 14:28.05 | coolbeans | Anybody care to take a stab? -> I'm sure this is something simple but I've never seen it. I installed 1.4.4 and configured extconfig and res_mysql. Everything "appears" to be working fine but I get this when I do a sip reload: sipsock_read: Recv error: Bad file descriptor ... over and over and over. What's weird is it doesn't do it on the initial asterisk start. Any clues? |
| 14:28.31 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:28.44 | mosty | coolbeans, look in the full log with debug and verbose |
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| 14:33.00 | BrokenNoze | anyone help, why do my polycom's suddenly start saying bad file descriptor when i try and dial out? |
| 14:33.53 | [TK]D-Fender | coolbeans: unload chan_brokenrecord.so |
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| 14:43.41 | Mad|Cow | I'm having some issues with my Polycom Soundpoint 500; when ever I dial 1 or #, it goes fast bussy. I checked my digitmap in the phone config and there is no reference to either.... anyone have any ideas why it matches those and tries to dial? |
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| 14:44.31 | heanol | what's the easiest way to via asterisk call a number, play an mp3-file and hangup? |
| 14:44.35 | heanol | i want to have this automated |
| 14:47.16 | MrTelephone | I finally got my RBS issue fixed with the adit 600 but asterisk won't recognize any digits I dial on the phone.. Is there something I'm missing? |
| 14:47.16 | tzanger | callfile |
| 14:47.19 | tzanger | that's the easiest way |
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| 14:50.05 | MrTelephone | it recognizes 8 very well |
| 14:50.13 | MrTelephone | relaxdtmf? |
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| 15:02.46 | `Sean | hrmp is there a good soloution to using fax with VoIP, because id really make life easier as i could get international did's and my associates would be able to fax me documents on the fly |
| 15:03.03 | Strom_M | fax over voice over internet protocol |
| 15:03.16 | Strom_M | bad idea :) |
| 15:03.22 | Strom_M | look at T.38 |
| 15:03.37 | cpm | not as bad as tvoip |
| 15:04.29 | `Sean | Strom_M wusup man :)? |
| 15:04.42 | inv_arp[work] | t.38 has matured... |
| 15:05.04 | `Sean | long time no see, anyhow.. ther thing i was reading was that if youre lan isnt busy as most peoples are then it shouldn't cause problems however due to large bits of data coming in rapidly, it can disturb youre lan |
| 15:05.46 | `Sean | <PROTECTED> |
| 15:05.48 | `Sean | <PROTECTED> |
| 15:05.54 | `Sean | sigh, i gotta go back to uncompressed :| |
| 15:06.11 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 15:06.16 | coolbeans | Tk/mosty: Thanks. |
| 15:06.17 | drako | why i keep getting this: |
| 15:06.18 | drako | *CLI> misdn show channels |
| 15:06.18 | drako | Chan List: (nil) |
| 15:06.22 | Strom_M | even with G.711, fax over voip isnt that reliable |
| 15:06.55 | Qwell[] | s/that // |
| 15:07.25 | `Sean | Strom_M i understand its not reliable as a corporate soloution |
| 15:07.37 | `Sean | my lan is almost always dead |
| 15:07.45 | Qwell[] | `Sean: You've obviously never worked in a large corp, heh |
| 15:07.57 | `Sean | Qwell no |
| 15:08.00 | Qwell[] | "corporate solutions" is extremely laughable |
| 15:08.06 | *** join/#asterisk adker (n=chatzill@74-33-221-202.br1.glv.ny.frontiernet.net) |
| 15:08.09 | `Sean | i know :( |
| 15:08.15 | `Sean | Qwell i just need a quick soloution |
| 15:08.15 | Qwell[] | especially referencing the reliability of one... |
| 15:08.26 | `Sean | where i can still use g729 and still be able to use t.38 |
| 15:08.27 | Qwell[] | heh, *that* sounds more like a "corporate solution" |
| 15:08.27 | [TK]D-Fender | Qwell : "reliable" is a matter of perpective. You can "rely" on something all you want, its just depends on your expectations :) |
| 15:08.32 | Qwell[] | s/corporate/quick/ |
| 15:08.37 | Qwell[] | [TK]D-Fender: touche |
| 15:08.48 | `Sean | or at least have asterisk assign a port for t.38 and raw audio stream |
| 15:09.03 | Qwell[] | [TK]D-Fender: in a job training thingie once, they told us something similar about the word "quality" |
| 15:09.07 | Qwell[] | it was quite funny |
| 15:09.26 | Qwell[] | "We sell quality parts" is a very funny statement |
| 15:11.35 | `Sean | Qwell how can i use my fax machine and as well not get rid of my g729 encoding |
| 15:11.57 | `Sean | can i specify a certain port on the TDM400P that uses no encoding |
| 15:11.57 | Qwell[] | get some ATAs that support T.38 |
| 15:12.02 | `Sean | whilst others use the g729 |
| 15:12.05 | Qwell[] | what? |
| 15:12.09 | Qwell[] | analog ports don't use g729 |
| 15:12.21 | `Sean | i know they dont |
| 15:12.54 | `Sean | err wait, with incomign did's you can specify incoming call encoding format |
| 15:13.13 | coppice | run V.34 over the analogue port, and G.729 over that :-) |
| 15:13.14 | `Sean | or well at least you can with the provider so he doesn't encode them to a different format |
| 15:13.34 | Qwell[] | coppice: stop confusing the nubs :p |
| 15:13.40 | drako | Hey, I'm trying to set up a b410p card but i can't get the channels up |
| 15:13.43 | drako | Chan List: (nil) |
| 15:16.04 | *** join/#asterisk karlhaines (n=karl@unaffiliated/karlhaines) |
| 15:16.24 | russellb | drako: support@digium.com would be happy to help you. |
| 15:19.06 | `Sean | damn this sucks im getting no where with my fax |
| 15:24.35 | Strom_M | here's a silly question: in zapata.conf, is it necessary to type channel => or can one get away with just channel = |
| 15:24.51 | `Sean | you can get away with a = |
| 15:24.59 | anonymouz666 | is it possible to add another field (not userfield) to store on CDR? For example I get the category call and I want to save into a field called "CAT" |
| 15:25.19 | Sweeper | anyone use idefisk? |
| 15:25.23 | coppice | `Sean: print it out. pop it in an envelope. mail it |
| 15:25.38 | `Sean | :( |
| 15:25.45 | `Sean | that would be going backwards in time |
| 15:25.50 | Strom_M | email a pdf |
| 15:25.55 | `Sean | in this day and age where speed is of relevance |
| 15:25.59 | Sweeper | faking faxes |
| 15:26.04 | codefreeze | anonymouz666: you can use CDR(CAT)=val in a Set call |
| 15:26.07 | Sweeper | they are the DEVIL |
| 15:26.14 | [TK]D-Fender | anonymouz666: not without coding up app_queue |
| 15:26.32 | [TK]D-Fender | anonymouz666: oops... scratch that. |
| 15:26.39 | [TK]D-Fender | anonymouz666: parse error :) |
| 15:28.18 | *** join/#asterisk hfb (n=hfb@pool-72-67-156-130.lsanca.dsl-w.verizon.net) |
| 15:28.50 | mosty | i'm trying to dial #123 with a snom phone, and the phone says "not found", and it doesn't try to dial. anyone know what setting is causing this? |
| 15:29.12 | anonymouz666 | codefreeze is CDR king. thanks! |
| 15:29.50 | [TK]D-Fender | mosty: Turn up SIP debug. I'm betting it is, and its 404-ing |
| 15:32.26 | mosty | you were correct, found the problem with my dialplan now- thanks |
| 15:36.02 | *** join/#asterisk DarKnesS_WolF (n=wolf@unaffiliated/sherif) |
| 15:36.03 | Strom_M | mosty: don't start extensions with # |
| 15:36.09 | Strom_M | # means "I'm finished dialing" |
| 15:36.21 | Strom_M | so the phone is likely putting the call through even before you get to dial 1 |
| 15:37.24 | irule | may someone please direct me to a propper doc on setting up a couple * servers as iax peers? one is for production and the second one is for testing, so Id like to call the testing server from the sip phones connected to the production server and avoid reconnecting stuff to make tests |
| 15:37.24 | *** join/#asterisk bawb2 (n=bawb2@129.237.2.66) |
| 15:37.27 | *** join/#asterisk truz_`24 (n=truz_`24@74-129-166-232.dhcp.insightbb.com) |
| 15:37.29 | truz_`24 | o/ |
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| 15:37.43 | DarKnesS_WolF | irule: http://www.voip-info.org/wiki/view/Asterisk+Connect+2+servers |
| 15:37.43 | [TK]D-Fender | Strom_M: I use # all the time, and it only means what you tell it that it means or a STUPID (too smart for YOUR good) phone interprets it as |
| 15:37.58 | truz_`24 | What package is the zaptel module in for ubuntu? |
| 15:38.10 | truz_`24 | apt-get install zaptel; all installed but modprobe zaptel returns nothing |
| 15:38.24 | irule | DarKnesS_WolF thanks |
| 15:38.45 | DarKnesS_WolF | truz_`24: it's there |
| 15:38.49 | DarKnesS_WolF | truz_`24: what ubuntu ur using ? |
| 15:39.35 | Sweeper | ooookay |
| 15:39.40 | truz_`24 | edgy |
| 15:39.44 | Sweeper | so apparnetly, idefisk does its own dns lookups |
| 15:39.47 | Sweeper | fucking nooches |
| 15:39.48 | Strom_M | [TK]D-Fender: in traditional telephony, "#" means "put the call through right now plzkthx" |
| 15:40.02 | Strom_M | so therefore, it's a bad idea to go redefining things just because you think it's a stupid idea |
| 15:40.28 | DarKnesS_WolF | truz_`24: hmm i'm using fiesty and it's there.. |
| 15:40.40 | DarKnesS_WolF | truz_`24: u mean that u don't have zaptel or ztdummy in ur modprobe ? |
| 15:40.53 | mosty | Strom_M, a customer wants to do pbx commands like #33 |
| 15:40.55 | truz_`24 | right |
| 15:40.56 | [TK]D-Fender | Strom_M: If I take a boring AT&T line and attempt to terminate a 7 digit number with "#" what'll happen? |
| 15:40.58 | Vec2 | I am having a problem with faxing, if I try send outgoing faxes it works fine, eg, Fax Machine > ATA > Asterisk > ZAP but when its the otherway round the call lasts for a second and then hangs up, I think its a T38 negotiation bug, http://bugs.digium.com/view.php?id=8736. Anyone know have this has been fixed in asterisk 1.4.4 ? |
| 15:41.37 | *** join/#asterisk jmacz (n=jmacz@190.24.100.67) |
| 15:41.37 | DarKnesS_WolF | truz_`24: hmm don't know really :-s i compile from source always :-s |
| 15:41.39 | mosty | Strom_M, my dialplan was using _X. in a bunch of places (because _. produces warning messages), and that was breaking #<foo> calls |
| 15:41.52 | Strom_M | [TK]D-Fender: the call will go through immediately when you press # |
| 15:41.52 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:41.53 | [TK]D-Fender | Vec2: * doesn't DO T.38 except in pass-through. Since you are terminating to Zap, that can't be it... |
| 15:42.12 | truz_`24 | where is the module stored on your box? |
| 15:42.20 | truz_`24 | or what is it called ? |
| 15:42.34 | [TK]D-Fender | Strom_M: I know starting with it fails... never tried up here on a regular line... lemme see how it works up here |
| 15:42.58 | DarKnesS_WolF | truz_`24: i alaways compile from source. |
| 15:43.01 | [TK]D-Fender | mosty: You must really love dangerous pattern matches |
| 15:44.27 | coppice | anyone who has dated loves dangerous pattern matches |
| 15:44.30 | mosty | [TK]D-Fender, i was using _X. - i thought that was safe |
| 15:44.42 | mosty | it just didn't match #33 |
| 15:45.07 | [TK]D-Fender | mosty: I'm not the least bit surprised ;) |
| 15:45.13 | mosty | though, when i dial *33 the asterisk console shows the call entering the dialplan, and when i dial #33 it shows nothing |
| 15:45.34 | [TK]D-Fender | mosty: guess you've got a * match around there somewhere. |
| 15:45.37 | mosty | [TK]D-Fender, care to enlighten me? i was just following the suggestion of asterisk's own warning messages |
| 15:45.55 | [TK]D-Fender | mosty: Enlighten about what? |
| 15:46.17 | mosty | why you're surprised |
| 15:46.30 | Strom_M | mosty: do I need to repeat myself about #? |
| 15:46.34 | Strom_M | # means "I'm finished dialing" |
| 15:46.41 | Strom_M | so the phone is likely putting the call through even before you get to dial 1 |
| 15:47.19 | mosty | Strom_M, a tethereal tracke shows the phone requestion %2333@mypbx, then the pbx sends back a 404 |
| 15:47.20 | Strom_M | so dont start extensions with # unless you want to confuse people and break with standards and all that jazz |
| 15:47.32 | mosty | ok, point taken |
| 15:47.58 | *** join/#asterisk af_ (n=getsmart@81-174-8-57.f5.ngi.it) |
| 15:48.21 | truz_`24 | DarKnesS_WolF, when you compile from source, it creates the zaptel module, where is this module located on yoru machine? |
| 15:48.52 | justdave | truz_`24: somewhere in /lib/modules/`uname -r`/ |
| 15:49.19 | Strom_M | mosty: also see the nanpa list of vertical service codes |
| 15:49.20 | irule | a little off topic, I have a grandstream phone, it is not registered atm and the screen says it is downloading an update, is it no0rmal that it is not working while in the updating process? can I interrupt it or something or set up something so that it may update only at 3 am or something? thanks |
| 15:49.22 | Strom_M | ~vsc |
| 15:49.24 | jbot | vsc is, like, Vertical Service Codes such as *67, *69, *72, and *82. These codes are generally reserved for specific uses, and it's a bad idea to conflict with the official assignments. A list of assigned VSCs for North America is at http://nanpa.com/number_resource_info/vsc_assignments.html and http://www.nanpa.com/number_resource_info/vsc_definitions.html |
| 15:49.52 | truz_`24 | cool, thx justdave |
| 15:50.48 | [TK]D-Fender | mosty: I said I am NOT surprised. |
| 15:51.32 | [TK]D-Fender | irule: interrupting updates tends to brick phones. |
| 15:52.18 | irule | ok thanks heh |
| 15:52.39 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com) |
| 15:52.52 | vader-- | ok i got international calling problem |
| 15:52.58 | vader-- | i guess i never setup my dial plan for this |
| 15:53.00 | *** join/#asterisk hrmphh (i=patrick@notchill.com) |
| 15:53.10 | vader-- | i have a phone number that is ### ### ### ### ### |
| 15:53.14 | hrmphh | where are voice mail greetings stored? |
| 15:53.18 | vader-- | ive never seen anything in that format |
| 15:53.18 | hrmphh | the ones recorded by users? |
| 15:53.57 | *** join/#asterisk cazze (n=pc@unaffiliated/kammicazze) |
| 15:54.34 | [TK]D-Fender | vader--: exten _XXXXXXXXXXXXXXX,1,Dial(SOMETHIN!!!) |
| 15:54.35 | justdave | hrmphh: look in /etc/asterisk/asterisk.conf |
| 15:55.33 | glogic | anyone having trouble with AMI in trunk? it seems broken at the moment |
| 15:55.42 | *** join/#asterisk d00gster (n=doughant@bas1-toronto12-1128667111.dsl.bell.ca) |
| 15:56.18 | hrmphh | erm |
| 15:56.23 | hrmphh | so its /var/lib/asterisk |
| 15:57.34 | mosty | Strom_M, interesting (service code list), thanks |
| 16:00.28 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 16:04.13 | hrmphh | how can you set it up to trasnfer directly to VM? |
| 16:04.20 | hrmphh | without ringing ... |
| 16:04.30 | hrmphh | like i want to be able to trasnfer to person a's vm greeting instantly |
| 16:04.33 | irule | http://pastebin.ca/489516 I made a few readjustments organizing the trunks, making it cleaner and stuff, and fafter this my sip phones cant call each other, this is weird, any thoughts? thanks |
| 16:04.34 | *** join/#asterisk [shodan] (n=shodan@ip020.99-113-216.pppoe4.joliette.intermonde.net) |
| 16:04.39 | *** join/#asterisk galeras (n=root@200.31.204.42) |
| 16:06.50 | *** join/#asterisk ctooley (n=ctooley@rrcs-71-42-115-242.sw.biz.rr.com) |
| 16:07.07 | ctooley | Anyone in the Dallas/Fort Worth area looking for Asterisk work? |
| 16:07.29 | mutilator | i could move |
| 16:07.30 | mutilator | :P |
| 16:07.35 | SwK | i can do it from remote |
| 16:09.24 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 16:09.34 | [TK]D-Fender | hrmphh: exten => 123,1,VoiceMail(456@default,u) |
| 16:09.53 | ctooley | SwK no you couldn't. I need an experienced person, already in Dallas, that wants a full time position. |
| 16:10.10 | SwK | not for me then |
| 16:10.13 | ctooley | Not necessarily in Dalls, the office is in Irving |
| 16:10.27 | *** join/#asterisk wunderkin (i=wunderki@ip68-108-204-139.ph.ph.cox.net) |
| 16:10.35 | karlhaines | ctooley: i used to work in dallas, for annulet inc, i built their domain registry |
| 16:10.45 | karlhaines | php for the website and python for the backend |
| 16:10.56 | karlhaines | ctooley: is the contract or full time? |
| 16:11.01 | *** part/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 16:11.05 | ctooley | karlhaines you moved away from Dallas? It's full time. |
| 16:12.08 | *** join/#asterisk rene- (n=rene@200.34.66.137) |
| 16:13.21 | hrmphh | hrm |
| 16:13.26 | hrmphh | why wont it let me transfer to *number |
| 16:13.53 | [TK]D-Fender | hrmphh: Got it in your dialplan? |
| 16:13.56 | *** join/#asterisk ZaVoid (n=zavoid@160.79.136.150) |
| 16:14.04 | hrmphh | _*1XXX => { VoiceMail(u${EXTEN}); } |
| 16:14.07 | hrmphh | yes |
| 16:14.10 | hrmphh | and i dialplan reloaded |
| 16:14.12 | ZaVoid | morning |
| 16:14.25 | [TK]D-Fender | hrmphh: debug your phone and see what its sending. |
| 16:14.52 | hrmphh | might be the diaplan on the sipura |
| 16:16.29 | *** join/#asterisk jmacz (n=jmacz@190.24.98.253) |
| 16:17.19 | hrmphh | (*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.) |
| 16:17.31 | hrmphh | would that dial plan on a sipura prevent me from transferring to "*1XXX" |
| 16:17.53 | [TK]D-Fender | hrmphh: because it clearly tells you to STFU after the 2nd "x" |
| 16:18.06 | hrmphh | yeah the first part? |
| 16:18.22 | hrmphh | only wants 2 digits after *? |
| 16:18.22 | [TK]D-Fender | hrmphh: yes, it really IS that clear |
| 16:18.34 | hrmphh | must be the vertical service codes |
| 16:18.38 | hrmphh | this is just the default dialplan |
| 16:18.39 | hrmphh | on the phone |
| 16:18.41 | *** part/#asterisk BSD_tech (n=BSDTech@adsl-69-230-174-37.dsl.irvnca.pacbell.net) |
| 16:18.43 | rene- | hello, what could cause audio problems that happen only on the receiving side of a sip call? i am having lots of gaps and skips, in the past when i was linking two sites using IAX i would enable the jitterbuffer and they would go away, with the sip jb things doesnt seem so simple, apparently people are using for zap-sip communications? what else should i look into? i know i cannot or should not traffic shape and incoming connection? |
| 16:18.45 | [TK]D-Fender | hrmphh: no, its jsut a DIALPLAN. |
| 16:19.08 | [TK]D-Fender | hrmphh: If stops listening after the 2nd digit and thats it. go FIX IT. |
| 16:19.13 | rene- | they dont happen in every call but in most of them |
| 16:19.13 | hrmphh | uh what? no shit its a dial plan, im saying the reason they prob have only 2 there is because all the vertical service codes are 2 digits |
| 16:19.35 | hrmphh | and they have no need for more BY DEFAULT |
| 16:19.56 | mosty | rene-, what's on the other end of these sip calls? |
| 16:20.12 | rene- | it is SIP provider-> asterisk - mitel sip phone |
| 16:20.13 | [TK]D-Fender | hrmphh: No, there are clearly VSC's with 3 digits |
| 16:20.24 | rene- | sip all the way |
| 16:20.33 | hrmphh | not defined on this phone |
| 16:20.56 | [TK]D-Fender | hrmphh: A nifty oversight :) Make it do what YOU want it to do... |
| 16:20.59 | mosty | rene-, ok then it sounds like your internet connection needs QoS |
| 16:21.05 | hrmphh | fyi theres 31 vscs defined by default |
| 16:21.07 | hrmphh | all 2 digit |
| 16:21.27 | [TK]D-Fender | http://nanpa.com/number_resource_info/vsc_assignments.html <- read again |
| 16:21.34 | rene- | mosty: i would think that would be the case on outgoing connections? |
| 16:21.44 | hrmphh | is that url specific to my phone? no? kthx |
| 16:21.58 | mosty | rene-, depends what else is happening on that link |
| 16:22.03 | rene- | true |
| 16:22.09 | [TK]D-Fender | hrmphh: Nope, just by those that seem to define the very term you are using. |
| 16:22.25 | hrmphh | do i care what the full standard is? no |
| 16:22.32 | hrmphh | am i talking about a specific phone's default configuration? yes |
| 16:22.50 | rene- | i am asking my ISP to run a traffic report for me so i can see what my usage is and if that is being the problem |
| 16:22.58 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 16:23.03 | [TK]D-Fender | hrmphh: Well it appears your phone's dialplan doesn't make ANYBODY happy... |
| 16:23.20 | galeras | DummQ: can fxotune be used for E1/T1 cards? |
| 16:23.25 | hrmphh | hmm how does that dialplan even allow 5 digit extension dialing |
| 16:23.44 | hrmphh | rene-; what type of csico eq? |
| 16:23.46 | hrmphh | cisco rather |
| 16:24.14 | [TK]D-Fender | hrmphh: Doesn't look like it should |
| 16:24.35 | rene- | i have one box i think is a 2801 that is T1 voip router and that gives me a t1 to sprint that is flawless |
| 16:24.38 | hrmphh | it does, you have to hit "dial" at the end |
| 16:24.52 | hrmphh | rene-; 2801? just break the box on boot and reset term passwd |
| 16:25.12 | hrmphh | they wont even know |
| 16:25.13 | [TK]D-Fender | hrmphh: SPA-841? |
| 16:25.16 | hrmphh | 941 |
| 16:25.31 | hrmphh | may just be able to grab existing startup-config and brute force passwd |
| 16:25.34 | [TK]D-Fender | hrmphh: that'd be Linksys then :) |
| 16:25.37 | hrmphh | unless its well encrypted |
| 16:25.38 | hrmphh | tk; yeah |
| 16:26.05 | [TK]D-Fender | hrmphh: May be that it'll send anything if you hit send, but respect the dialplan if you wait or hit a match on dialtone. |
| 16:26.31 | [TK]D-Fender | hrmphh: will it dial a 5-digit while off-hook and listening to tone if you follow with "dial"? |
| 16:26.49 | hrmphh | dunno havent tried offhook |
| 16:26.50 | hrmphh | playing w/it now |
| 16:27.12 | hrmphh | prob just going to change from (*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.) |
| 16:27.47 | *** join/#asterisk mightnare (n=mike@p1117-ipad07motosinmat.mie.ocn.ne.jp) |
| 16:27.58 | hrmphh | hrm can you use regex ?'s in the dialplan on a linksys? |
| 16:28.09 | hrmphh | cause i want to add 5 digit dialing optionally beginning with a ? |
| 16:28.19 | hrmphh | *?xxxxx |
| 16:28.21 | [TK]D-Fender | hrmphh: it follows the MGCP dialplan RFC. Well documented |
| 16:28.45 | [TK]D-Fender | hrmphh: (*x.|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.) |
| 16:28.57 | hrmphh | one or more x's? |
| 16:29.01 | hrmphh | err digits |
| 16:29.03 | [TK]D-Fender | hrmphh: yup |
| 16:29.10 | hrmphh | what about 5 digit dialing |
| 16:29.23 | rene- | hrmphh: cool |
| 16:29.33 | hrmphh | think itd be better to just add "*?xxxxx" if that worked |
| 16:29.39 | hrmphh | trying to find this rfc now |
| 16:29.44 | Strom_M | hrmphh: ugh, please dont tell me you're doing five-digit vertical service codes |
| 16:29.52 | hrmphh | strom; no not at all |
| 16:29.56 | hrmphh | its for transfer directly to vm |
| 16:30.00 | hrmphh | we have 5 digit extensions internally |
| 16:30.10 | hrmphh | but if you want to transfer to someone's vm unavail without waiting |
| 16:30.18 | Strom_M | yeah, there's a brilliant idea...conflict with the entire vertical service code space all in one go |
| 16:30.23 | hrmphh | :)( |
| 16:30.25 | hrmphh | err |
| 16:30.32 | hrmphh | have a better suggestion? |
| 16:30.33 | hrmphh | for a digit to use? |
| 16:30.39 | *** join/#asterisk keulin (n=cray@AMontpellier-152-1-31-50.w81-251.abo.wanadoo.fr) |
| 16:30.41 | Strom_M | use one of the available vertical service codes followed by the extension |
| 16:30.43 | Strom_M | ~vsc |
| 16:30.45 | jbot | hmm... vsc is Vertical Service Codes such as *67, *69, *72, and *82. These codes are generally reserved for specific uses, and it's a bad idea to conflict with the official assignments. A list of assigned VSCs for North America is at http://nanpa.com/number_resource_info/vsc_assignments.html and http://www.nanpa.com/number_resource_info/vsc_definitions.html |
| 16:30.52 | Strom_M | there's a range that's reserved for local assignment |
| 16:31.00 | hrmphh | k |
| 16:31.42 | hrmphh | ill still have to change my dial plan tho |
| 16:31.45 | hrmphh | even if i use a reserved one |
| 16:31.52 | hrmphh | cause it cuts you off 2 digits after * |
| 16:32.11 | *** part/#asterisk mutilator (i=WebChat@the.drinkproject.com) |
| 16:32.25 | hrmphh | ah *99, iirc thats what most people use on cisco callmanager |
| 16:32.33 | [TK]D-Fender | Strom_M: You know... you've created a whole new class of neurosis... VSC Zealotry ;) |
| 16:32.38 | Strom_M | muahaahaha |
| 16:33.01 | [TK]D-Fender | Strom_M: And a message from the rest of the Universe : "We don't actually CARE" :) |
| 16:33.08 | Strom_M | fuck you too :) |
| 16:33.31 | [TK]D-Fender | Strom_M: We oughtta go for a beer :) |
| 16:33.57 | [TK]D-Fender | hrmphh: Wait... even better... use #xxxx ! ;) |
| 16:34.00 | Strom_M | hah |
| 16:34.04 | tzanger | mmmm beer |
| 16:34.10 | [TK]D-Fender | Strom_M: pwned |
| 16:34.24 | tzanger | my unemployed friend just popped online to tell me he's enjoying beer, wings and wifi at a local pub |
| 16:34.27 | tzanger | bastard |
| 16:34.35 | [TK]D-Fender | FYI, I use #(X's) myself for direct voicemail. |
| 16:34.42 | cpm | [TK]D-Fender is right, the rest of the universe doesn't care, at all. They only want to reserve the right to bitch like crazy when stuff breaks even though they refuse to learn anything |
| 16:35.41 | hrmphh | (*xx|*99x.|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.) |
| 16:35.44 | hrmphh | hrm |
| 16:35.44 | *** join/#asterisk Enron (n=enron@216.70.173.176) |
| 16:35.46 | Enron | Hi |
| 16:35.51 | hrmphh | apparently phone doesnt like that either |
| 16:35.57 | [TK]D-Fender | cpm: They do that, don't they :) matching *xxxx for VM's does not interfere with VSC's, it just means it won't dial INSTANTLY on the 3rd or 4th digit dialed. BIG DEAL. |
| 16:35.59 | hrmphh | will it break on first match? |
| 16:36.00 | Enron | we are experiencing problem with our phone abit urgent |
| 16:36.02 | hrmphh | the *xx? |
| 16:36.05 | Enron | can someone help diagnose |
| 16:36.07 | hrmphh | should i put the *99x. first? |
| 16:36.16 | hrmphh | lol enron |
| 16:36.16 | Enron | I'm getting Primarty D-channel on span 1 down |
| 16:36.19 | syzygyBSD | hah, Enron asking for help... |
| 16:36.39 | Enron | Chan_zap.c:2438 pri_find_dchhan: No d-channels available |
| 16:36.40 | [TK]D-Fender | FRAUDULENT! |
| 16:36.49 | Corydon-w | Enron: what does 'zap show status' show you? |
| 16:36.53 | SwK | Enron: zap show status' |
| 16:36.54 | Enron | using primary channel 24 as d-channel anyway, !! got I-frame while link state 2 |
| 16:36.58 | cpm | first match wins |
| 16:37.04 | hrmphh | cpm; thank you |
| 16:37.07 | [TK]D-Fender | Enron: Thats because you don't actually HAVE any D-channels... tcheck your OTHER books ;) |
| 16:37.48 | Enron | It was working fine then it went down |
| 16:38.13 | hrmphh | hrm from the cli how can i see what phone# a sip phone is sending? |
| 16:39.04 | [TK]D-Fender | hrmphh: "sip debug" |
| 16:39.12 | hrmphh | thanks |
| 16:39.26 | hrmphh | and can you use '*' in an extension in extensions.ael? |
| 16:39.40 | hrmphh | like _*1XXXX is valid? |
| 16:39.49 | [TK]D-Fender | hrmphh: Should be able to just fine |
| 16:40.24 | Enron | What does no d-channels mean? |
| 16:41.01 | Enron | what is the d-channel |
| 16:41.05 | hrmphh | ok weird |
| 16:41.08 | hrmphh | dialplan reload didnt do it |
| 16:41.11 | hrmphh | i had to restart the whole thing |
| 16:41.15 | hrmphh | for that extension to take |
| 16:41.22 | Enron | [TK]D-Fender ? |
| 16:41.31 | Enron | VoxTrada*CLI> zap show status |
| 16:41.31 | Enron | Description Alarms IRQ bpviol CRC4 |
| 16:41.31 | Enron | Digium Wildcard TE110P T1/E1 Card 0 OK 0 0 0 |
| 16:41.56 | [TK]D-Fender | Enron: Guessing your card may jsut have received a reset. Is it WORKING now? |
| 16:42.11 | Enron | nope :( |
| 16:42.33 | Enron | still busy signal when calling and nobody can call in |
| 16:42.51 | Enron | what else can we check? |
| 16:43.30 | Strom_M | Enron: what kind of circuit do you have plugged into the card? |
| 16:44.17 | Enron | pri 24 chan |
| 16:44.26 | Enron | voice and data |
| 16:44.28 | Strom_M | T1? |
| 16:44.28 | *** join/#asterisk Splat (n=splat@home.heehawhills.com) |
| 16:44.35 | Enron | full 1.5 |
| 16:44.36 | Enron | yea |
| 16:44.38 | *** join/#asterisk Zdrulio (n=BLA@84.238.147.195) |
| 16:44.46 | Strom_M | ok, which channels are voice and which channels are data? |
| 16:45.58 | Enron | all are both |
| 16:46.31 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 16:46.37 | Enron | automatic ajustment |
| 16:46.43 | [TK]D-Fender | Left is right! 4 is 5! |
| 16:46.48 | Enron | ha? |
| 16:46.50 | Strom_M | so it's a basic full ISDN PRI with 23 B channels and a single D channel? |
| 16:47.26 | Enron | it's a T1 |
| 16:48.16 | hrmphh | hmm so apparently when you transfer directly to vm it starts the greeting right away |
| 16:48.26 | Strom_M | yes, ISDN PRI is delivered over Tq1 |
| 16:48.27 | Strom_M | er, T1 |
| 16:48.41 | hrmphh | like receptionist picks up, hits transfer, *991XXXX and then they hang up right? |
| 16:48.44 | Strom_M | but I'm asking you if you know exactly how that circuit is configured |
| 16:48.54 | hrmphh | so from the time they dial the extension to the time they hang up only they can hear the greeting |
| 16:49.16 | Strom_M | yes |
| 16:49.16 | MrTelephone | I had to use the sangoma beta drivers for the adit 600 to work |
| 16:49.28 | MrTelephone | and also had to relax dtmf for tones to be picked up |
| 16:49.29 | Strom_M | hrmphh: you may want to put twelve seconds of ringing in there |
| 16:49.40 | hrmphh | strom twelve? heh |
| 16:49.42 | Enron | Strom_M Ok so T1 and all chans are data and voice |
| 16:49.43 | hrmphh | seems arbitrary |
| 16:49.47 | Enron | What else can we diagnose |
| 16:49.52 | hrmphh | was thinking just pausing for a few seconds? |
| 16:49.54 | Strom_M | Enron: uh, no |
| 16:50.00 | Strom_M | chans cannot be "data and voice" |
| 16:50.16 | Strom_M | either it's ISDN, it's data, or it's voice |
| 16:50.21 | [TK]D-Fender | hrmphh: She should be doing a BLIND transfer |
| 16:50.29 | Enron | can a central office/switch be dropping our Dchan before it reaches telepacific |
| 16:50.30 | hrmphh | how do you do a blind transfer? |
| 16:50.31 | *** join/#asterisk agile (n=mike@63.98.55.146) |
| 16:50.59 | [TK]D-Fender | hrmphh: Look at your soft keys closer |
| 16:51.06 | hrmphh | k |
| 16:51.14 | Strom_M | Enron: i doubt it |
| 16:51.24 | hrmphh | in any case prob want to play some ring |
| 16:51.25 | Enron | why? |
| 16:51.27 | Strom_M | Enron: now please answer my question |
| 16:51.28 | hrmphh | whats the command to do that? |
| 16:51.38 | hrmphh | Ring(seconds?) |
| 16:51.39 | Enron | sorry, what is your question again. |
| 16:51.39 | Strom_M | is the circuit a basic full ISDN PRI with 23 B channels and a single D channel? |
| 16:51.41 | Enron | *? |
| 16:51.53 | hrmphh | agile; depends mostly on cpu |
| 16:51.57 | hrmphh | and of course outside lines :) |
| 16:52.09 | [TK]D-Fender | hrmphh: Why would you want to do that... to give the caller the illusion that they MIGHT get answered? |
| 16:52.26 | Enron | It's a T1 shared pri circuit that has voice and data riding the same pair, automatic |
| 16:52.27 | hrmphh | tk; no to buy some time if its not a blind transfer |
| 16:52.32 | hrmphh | because it could be coming from an analog phone |
| 16:52.37 | [TK]D-Fender | hrmphh: And why wouldn't it be? |
| 16:52.45 | hrmphh | we have an analog cordless phone on an fxs port |
| 16:52.45 | Strom_M | Enron: that's not lilely |
| 16:52.49 | hrmphh | theres no "blind transfer" softkey |
| 16:52.50 | Strom_M | er, likely |
| 16:52.50 | [TK]D-Fender | hrmphh: What phone can't do a blind transfer? |
| 16:53.00 | hrmphh | its hitting a flash button |
| 16:53.01 | Strom_M | Enron: it sounds to me like you don't really know what you're talking about |
| 16:53.02 | hrmphh | and dialing the extension |
| 16:53.05 | hrmphh | then hitting flash again |
| 16:53.10 | Enron | assuming it's just voice |
| 16:53.13 | [TK]D-Fender | hrmphh: I recall there being one when I had my 941... look AGAIN, and read the manual |
| 16:53.17 | Enron | what can we do to troubleshoot further. |
| 16:53.18 | hrmphh | tk |
| 16:53.22 | hrmphh | did you read what i said |
| 16:53.26 | hrmphh | 09:47 < hrmphh> we have an analog cordless phone on an fxs port |
| 16:53.27 | [TK]D-Fender | hrmphh: hitting flash != softkeys |
| 16:53.32 | hrmphh | 09:47 < hrmphh> because it could be coming from an analog phone |
| 16:53.33 | Strom_M | Enron: i need to know what the circuit actually is configured for before I can troubleshoot it |
| 16:53.36 | [TK]D-Fender | hrmphh: What kind of fxs? |
| 16:53.37 | hrmphh | 09:48 < hrmphh> theres no "blind transfer" softkey |
| 16:54.05 | hrmphh | what kind? |
| 16:54.05 | hrmphh | its a regular budget cordless phone |
| 16:54.05 | hrmphh | connected to fxs port on a tdm card |
| 16:54.05 | [TK]D-Fender | hrmphh: I believe its after you press "more" |
| 16:54.09 | Enron | on the phone with provider right now |
| 16:54.13 | Enron | will let you know in a sec |
| 16:54.25 | Enron | btw thanks for helping I really appreciate it. |
| 16:54.27 | [TK]D-Fender | hrmphh: zaptel has a blind transfer feature in features.conf |
| 16:54.33 | hrmphh | o rly |
| 16:54.48 | hrmphh | http://www.voip-info.org/wiki/view/Asterisk+vertical+service+activation+codes |
| 16:54.49 | hrmphh | reading that now |
| 16:54.50 | [TK]D-Fender | hrmphh: rly |
| 16:55.17 | [TK]D-Fender | Strom_M: I bet features.conf has you BOILING, doesn't it? ;) |
| 16:56.17 | hrmphh | http://www.voip-info.org/wiki/view/Asterisk+tips+zap+transfer |
| 16:56.19 | hrmphh | this is how we're doing now |
| 16:56.24 | hrmphh | cant seem to find blind transfer yet |
| 16:56.42 | Strom_M | [TK]D-Fender: cocks |
| 16:56.48 | Enron | Storm it's a Voice over ATM delivered to the building, and from there 24 chan pri out of the adtran. |
| 16:56.49 | hrmphh | altho it seems to be listed here: http://www.voip-info.org/wiki/view/Asterisk+PBX+functions |
| 16:56.57 | Enron | Using G726 Encoding |
| 16:57.11 | Strom_M | Enron: ......ok? |
| 16:57.23 | Enron | this is the info I am provided with |
| 16:57.35 | Enron | The encoding allows us to use half of the chan for data |
| 16:57.43 | Enron | and half for voice, so it's a single circuit |
| 16:57.55 | Enron | This is what i'm being told. so it's basically voice |
| 16:57.58 | [TK]D-Fender | Strom_M: c'mon.... don't hold back on us.. tell us how you REALLY feel ;) |
| 16:57.58 | Enron | ISDN PRI |
| 16:58.17 | Strom_M | is the g726 encoding used on the PRI coming out of the adtran, or on the ATM portion? |
| 16:58.21 | [TK]D-Fender | hrmphh: jsut look directly in features.conf |
| 16:58.33 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 16:58.45 | Enron | it's on the ATM |
| 16:58.57 | hrmphh | hmm ok so # |
| 16:59.07 | hrmphh | so hit flash then #extension and hangup? |
| 16:59.08 | hrmphh | sound right? |
| 16:59.11 | Enron | ISDN PRi btw. |
| 16:59.25 | [TK]D-Fender | hrmphh: not for atxfer |
| 16:59.27 | hrmphh | or are you not supposed to hit flash first |
| 16:59.34 | hrmphh | uatxfer |
| 16:59.38 | hrmphh | err |
| 16:59.40 | hrmphh | uaxfer |
| 16:59.47 | Strom_M | Enron: ok |
| 16:59.52 | Enron | Would it be a component failure? |
| 16:59.56 | Strom_M | so it's a basic ISDN PRI |
| 16:59.57 | hrmphh | blindxfer allows unattended or blind transfers. It works like this: |
| 16:59.58 | hrmphh | While on a conversation with another party, you dial the blindxfer sequence. Asterisk says "Transfer" then gives you a dial tone, while putting the other party on hold. You dial the transferee number and the caller is put through to that number immediately. Your line drops. The caller ID displayed to the person receiving the transferred call is exactly the same as the caller ID presented to you. |
| 17:00.03 | Enron | we have restarted the phone system |
| 17:00.04 | hrmphh | yay |
| 17:00.07 | Enron | Yes |
| 17:00.10 | Strom_M | Enron: pastebin your zaptel.conf and zapata.conf |
| 17:00.11 | Enron | it's a basic ISDN PRI |
| 17:01.16 | Enron | ok one moment getting it. |
| 17:01.20 | hrmphh | Note: You MUST use the T and/or t options in the command Dial() in order to allow the caller and/or callee to use any transfer feature |
| 17:01.21 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 17:01.23 | hrmphh | hmm |
| 17:01.25 | hrmphh | how do i use those options? |
| 17:01.31 | [TK]D-Fender | blind transfer on features.con appears to be #1 |
| 17:01.43 | [TK]D-Fender | hrmphh: "show application dial" |
| 17:02.03 | [TK]D-Fender | hrmphh: And seriously..... go look at your 941 again.. it HAS a blind transfer w/o using features.conf |
| 17:02.29 | hrmphh | dude |
| 17:02.31 | hrmphh | im talking about the analog |
| 17:02.33 | hrmphh | cordless |
| 17:02.33 | *** join/#asterisk bawb2 (n=bawb2@ip50210.estcmp.ku.edu) |
| 17:02.52 | [TK]D-Fender | hrmphh: And I'm back on a rant about the other non-existant features I've been pointing out to you ;) |
| 17:03.39 | Enron | Strom_M http://phpfi.com/234318 <-- zapata.conf |
| 17:03.41 | hrmphh | ouch so you have to put the t option anytime you dial? |
| 17:03.49 | Enron | Where would zaptel.conf be located |
| 17:03.50 | hrmphh | anytime you Dial() rather? |
| 17:04.04 | Strom_M | Enron: /etc/zaptel.conf |
| 17:04.09 | Enron | got it |
| 17:04.28 | Strom_M | Enron: uh, you need to specify an ISDN switchtype in zapata.conf... |
| 17:04.31 | Enron | Zaptel.conf http://phpfi.com/234319 |
| 17:04.52 | Strom_M | also, take the spaces out of your span= line |
| 17:05.02 | Strom_M | do you know what switchtype your circuit uses? |
| 17:05.10 | *** join/#asterisk cayorde (n=flexable@host184-111-dynamic.17-87-r.retail.telecomitalia.it) |
| 17:05.25 | Enron | DMS 100 |
| 17:05.40 | Strom_M | DMS100 signalling, or National ISDN 2 signaling? |
| 17:06.40 | Strom_M | because the DMS-100 can speak both ;) |
| 17:06.41 | Enron | checking one moment |
| 17:07.43 | *** join/#asterisk tr2x (n=alvar@80-218-185-55.dclient.hispeed.ch) |
| 17:08.05 | Enron | Strom_M can you tell me the line I need to edit in the zapata.conf for that |
| 17:08.08 | Enron | we will try both |
| 17:08.12 | [TK]D-Fender | hrmphh: where applicable, yes. BTW... there is a REASON I suggest Linksys SPA FXS over zaptel... |
| 17:08.40 | Strom_M | Enron: I'd rather know what it's supposed to be than go guessing around at it |
| 17:08.46 | Enron | ok |
| 17:10.55 | *** join/#asterisk Avero (n=Avero@216.186.253.120) |
| 17:11.41 | *** join/#asterisk jeremy_g (n=jerms@static-213-115-44-90.sme.bredbandsbolaget.se) |
| 17:11.43 | jeremy_g | hi |
| 17:11.53 | cpm | lo |
| 17:12.23 | jeremy_g | what does this symbol |OO| stands for? its printed to describe a female port on a telco hardware |
| 17:13.20 | Enron | Strom_M what's the value to be put in for NI2 |
| 17:13.35 | Strom_M | Enron: what did the telco say? |
| 17:13.37 | Strom_M | NI2? |
| 17:13.40 | Enron | still waiting |
| 17:13.42 | Strom_M | ok |
| 17:13.45 | Enron | so we are trying 2 of it |
| 17:13.46 | Strom_M | PATIENCE |
| 17:13.49 | Enron | ok |
| 17:14.36 | Strom_M | anyway, i have to break for lunch, so if the telco says "DMS-100", you put switchtype=dms100 in zapata.conf before signalling=; if they say NI2, you put "switchtype=national" |
| 17:14.50 | Strom_M | but DO NOT DO THIS BEFORE THE TELCO TELLS YOU |
| 17:15.01 | [TK]D-Fender | jeremy_g: Looks like a 2-prong power connector to me. |
| 17:15.17 | Strom_M | klunchbye |
| 17:15.53 | Enron | This is Richard. I have 20 years i telecom. I woirked for Nortel. THe Calss 5 switch is a DMS100. I need to know what the correct value for singnally is to set it for NI II. It was set to pri_cpe. Is that even a valid setting? I think we just need to klnow the valid ode for the signally field in the zapata.conf |
| 17:16.30 | Qwell[] | <Strom_M> anyway, i have to break for lunch, so if the telco says "DMS-100", you put switchtype=dms100 in zapata.conf before signalling=; if they say NI2, you put "switchtype=national" |
| 17:16.49 | jeremy_g | Enron: what does |OO| sybol printed on a telco port mean |
| 17:16.52 | jeremy_g | symbol |
| 17:17.02 | jeremy_g | [TK]D-Fender:nopes it aint that |
| 17:17.23 | jeremy_g | [TK]D-Fender:it apparently looks like a female serial port but it should be the monitor port |
| 17:17.29 | jeremy_g | but vga wont go in |
| 17:17.30 | jeremy_g | :( |
| 17:18.03 | [TK]D-Fender | jeremy_g: Well your ASCII art is the WORST sample of "supporting docs" I've ever seen :) |
| 17:18.12 | *** join/#asterisk c4t3l (n=c4t3l@cpe-72-181-205-77.houston.res.rr.com) |
| 17:18.12 | [TK]D-Fender | jeremy_g: And your description lacking |
| 17:18.15 | Qwell[] | jeremy_g: what color is the port? |
| 17:18.29 | Enron | national 2 |
| 17:18.32 | cpm | what type of connector, how many pins, if any |
| 17:19.09 | jeremy_g | Qwell:black |
| 17:19.28 | jeremy_g | its a 9-pin |
| 17:19.39 | jeremy_g | its my media gateway |
| 17:20.19 | jeremy_g | with two ethernet interfaces coming out of it, out of which one is shuved into the signallin gw (linux box) and other to the network |
| 17:20.56 | jeremy_g | Enron:i was expecting an answer out of your 20 years. |
| 17:20.57 | [TK]D-Fender | jeremy_g: could be serial for SMDR |
| 17:21.07 | jeremy_g | female serial |
| 17:21.27 | jeremy_g | but i would like to know the IP of the box whose serial it is |
| 17:21.43 | jeremy_g | supposed to be the monitor but it surely doesnt contain any os |
| 17:21.44 | *** join/#asterisk Braxus (n=braxus@66.147.214.164) |
| 17:22.01 | jeremy_g | ignore the last line plz |
| 17:22.22 | Qwell[] | ^^ that one? |
| 17:22.27 | Enron | jeremy that was richard, he was assisting me, this is atif now. sorry |
| 17:22.49 | aptura | my moh volume is cranked to the point of clipping distortion. Horrid. IE the volume is way to high anyone seen this before? |
| 17:23.29 | c4t3l | aptura: once |
| 17:23.49 | aptura | yea its doing it on my system. blow your ear drum out :) |
| 17:23.59 | c4t3l | have you tried messing with /etc/asterisk/musicongold.conf? |
| 17:24.18 | aptura | I just was in there will see if its a setting that needs to be turned down :) |
| 17:24.23 | c4t3l | hold! hold! musiconhold :) |
| 17:24.24 | Qwell[] | c4t3l: patent pending! |
| 17:24.31 | c4t3l | hehe |
| 17:25.18 | c4t3l | look into quietmp3 perhaps that may fix... |
| 17:26.10 | c4t3l | sorry quietmp3nb |
| 17:26.17 | Katty | herro |
| 17:26.23 | c4t3l | yo |
| 17:26.26 | Katty | anyone framilier with debian. |
| 17:26.30 | Katty | super duper framilier. |
| 17:26.42 | Qwell[] | Katty: I know of Debian |
| 17:26.51 | Katty | really now. |
| 17:27.19 | Katty | can you tell me, if i want to run a mount commmand, wether i should put it in the /etc/rc.local file, or the /etc/init.d/rc.local file? |
| 17:27.57 | aptura | c4t3l yes changed it to that setting same effect. |
| 17:28.04 | Qwell[] | Katty: /etc/init.d/rc.locall is what calls /etc/rc.local |
| 17:28.28 | Katty | so, that's just a symlink |
| 17:28.32 | Katty | to the /etc/rc.local file |
| 17:28.32 | Qwell[] | no |
| 17:28.43 | Katty | then i don't understand. |
| 17:28.46 | Katty | pretend i'm 5 :P |
| 17:28.49 | Qwell[] | it executes /etc/rc.local from /etc/init.d/rc.local |
| 17:28.53 | Katty | oh! |
| 17:28.57 | Qwell[] | GO TO YOUR ROOM |
| 17:29.00 | Qwell[] | NO ICECREAM |
| 17:29.01 | Katty | butbut :< |
| 17:29.05 | c4t3l | aptura: are you running mpg123? |
| 17:29.13 | mosty | Katty, neither. put it in /etc/fstab |
| 17:29.15 | Katty | you're not supposed to talk that way to a 5 year old :P |
| 17:29.17 | aptura | yup |
| 17:29.18 | Katty | mosty: uh, no |
| 17:29.19 | Qwell[] | pfft |
| 17:29.36 | aptura | I am looking into it online. |
| 17:29.39 | mosty | Katty, you're trying to mount a fs at boot? that's what fstab is for |
| 17:29.40 | c4t3l | aptura: have you tried to kill the mpg123 proc and restart? |
| 17:29.41 | Katty | Qwell[]: then talk to me like i'm 22 :P |
| 17:29.43 | Katty | mosty: no |
| 17:29.47 | aptura | c4 i could |
| 17:29.56 | luckyone_ | Hymie: sorry, I was away. So festival isn't that good? |
| 17:30.00 | Qwell[] | Katty: I don't think you want that either |
| 17:30.02 | Qwell[] | ... |
| 17:30.04 | mosty | Katty, what are you trying to do then? |
| 17:30.09 | Katty | mosty: nothing. |
| 17:31.00 | aptura | c4 killed restarted same thing looking online now for a fix. |
| 17:31.22 | c4t3l | hmm... is it a custom mp3 file?? |
| 17:31.24 | Katty | mosty: so nice of you to help me tho. |
| 17:31.28 | Katty | mosty: i do appreciate that (= |
| 17:31.45 | Qwell[] | oh, sure, thank him :P |
| 17:32.03 | file | Qwell[] doesn't do hugs |
| 17:32.06 | c4t3l | Katty: /etc/fstab |
| 17:32.08 | Katty | Qwell[]: you know i appreciate every little thing you do for me (= |
| 17:32.24 | aptura | hmm all default |
| 17:32.59 | c4t3l | aptura: whens the last time the system was rebooted? |
| 17:33.18 | Katty | file: did i tell you a hurt my foot yesterday? |
| 17:33.28 | file | Katty: yes :( I gave you morphine earlier! |
| 17:33.34 | Katty | file: oh yes. hrmm. |
| 17:33.41 | Katty | file: sorry, my memory is failing too. |
| 17:33.50 | Katty | file: i have my mother to thank for that. |
| 17:33.56 | DarKnesS_WolF | may be it's the morphine :P |
| 17:33.57 | file | Katty: did she hurt you?!? |
| 17:34.05 | Katty | file: her genetics, you goofball. |
| 17:34.13 | mosty | Katty, if you're tring to do nothing, put /bin/true in your rc.local file :P |
| 17:35.57 | Qwell[] | mosty: /bin/false |
| 17:35.57 | file | pfft |
| 17:35.57 | Katty | oh for goodness sake. |
| 17:35.57 | Qwell[] | do nothing - unsuccessfully |
| 17:35.57 | *** join/#asterisk KeNroM (n=hg@69.73.197.249) |
| 17:35.57 | Katty | i don't want to put it in /etc/fstab! accept that and move on! |
| 17:35.58 | Katty | kthx. |
| 17:35.58 | Katty | mishehu: ! |
| 17:35.58 | aptura | c4 have not but it sounds like the setting is stuck in memory. |
| 17:35.58 | c4t3l | ooh ooh for fun set your default runlevel in inittab to 6 |
| 17:35.58 | KeNroM | HI guys... i have a slight problem... |
| 17:35.59 | Katty | KeNroM: slight? i have a /real/ problem. |
| 17:35.59 | DarKnesS_WolF | Katty: then use /etc/rc.local |
| 17:35.59 | DarKnesS_WolF | Katty: rc.local is the last thing running before X as i recall |
| 17:35.59 | Qwell[] | c4t3l: real men (and/or women) set it to 0 |
| 17:35.59 | Katty | DarKnesS_WolF: yes, indeed. that's what Qwell[] said about 5 minutes ago. |
| 17:36.00 | mishehu | Katty: howdy, been a while. how's things down in the state of Misery? |
| 17:36.00 | c4t3l | haha |
| 17:36.00 | Katty | mishehu: miserable. hot. humid. |
| 17:36.01 | DarKnesS_WolF | Katty: oh i was writing an email... :-) sorry didn't notice .. |
| 17:36.01 | DarKnesS_WolF | Katty: but why u don't want to use fstab !? |
| 17:36.01 | Katty | mishehu: and the river it wrecking havoc on my allergies :P |
| 17:36.02 | Katty | DarKnesS_WolF: because it is a network share. |
| 17:36.02 | Katty | DarKnesS_WolF: off of a windows machine. |
| 17:36.02 | DarKnesS_WolF | Katty: so ? |
| 17:36.02 | Qwell[] | samba? |
| 17:36.03 | Qwell[] | You can fstab that :D |
| 17:36.09 | Katty | cifs |
| 17:36.09 | DarKnesS_WolF | i add Samba to my fstab |
| 17:36.09 | mishehu | Katty: damn teh river. time to declare war on the terrorist allergens |
| 17:36.09 | KeNroM | i have a closer campaign setup..and i also have a custom php webform..... when the user clicks on webform button it bring out window with this url |
| 17:36.09 | aptura | katty sounds like texas. |
| 17:36.10 | *** join/#asterisk `pariah (n=josh@unaffiliated/pariah) |
| 17:36.10 | Katty | mishehu: indeedy. |
| 17:36.17 | KeNroM | http://astguiclient.sourceforge.net/test_VICIDIAL_output.php?lead_id=112227 |
| 17:36.17 | Katty | it's not samba. it's cifs. |
| 17:36.23 | KeNroM | why does that happens |
| 17:36.25 | Qwell[] | whichever |
| 17:36.26 | mvanbaak | you can fstab cifs too |
| 17:36.26 | Katty | and i have domain and username and passwords and lots of other things |
| 17:36.28 | KeNroM | and how do i fix it.... |
| 17:36.32 | Katty | <PROTECTED> |
| 17:36.33 | *** join/#asterisk Ebola (n=Ebola@host86-137-4-175.range86-137.btcentralplus.com) |
| 17:36.34 | Katty | end of story! |
| 17:36.42 | Katty | next! |
| 17:36.51 | c4t3l | Katty create a pass file |
| 17:37.02 | mosty | Katty, fstab is basically just a series of mount commands broken up into their components |
| 17:37.04 | c4t3l | hehe |
| 17:37.06 | Katty | go fix someone else's problem! |
| 17:37.14 | Katty | help KeNroM :P |
| 17:37.19 | DarKnesS_WolF | Katty: http://forums.fedoraforum.org/archive/index.php/t-96149.html check last comment |
| 17:37.43 | mishehu | Katty: hope that's not an IBM type-M keyboard |
| 17:38.13 | Katty | mishehu: i think it's cute how there's a billion different ways to do something, and if you're not doing it the way someone else would, you're clearly in the wrong :P |
| 17:38.28 | DarKnesS_WolF | Katty: http://docs.hp.com/en/B8724-90067/ch02s07.html here too |
| 17:38.39 | Katty | mishehu: see? :P |
| 17:38.41 | mishehu | Katty: that, adn don't forget that being female automatically makes you wrong when it comes to driving or computing... |
| 17:38.50 | Katty | but thanks so much everyone for your help (= |
| 17:39.00 | Katty | mishehu: oh yeah. i forgot about that part. |
| 17:39.14 | c4t3l | Katty: why not write a shell script and sav it in .. say /usr/sbin, make it executable and call it form /etc/rc.local |
| 17:39.22 | Katty | c4t3l: i think i'd rather have an icecream cone. |
| 17:39.28 | c4t3l | ok |
| 17:39.33 | Katty | now if you guys could offer KeNroM some help... |
| 17:39.34 | c4t3l | that will also work |
| 17:39.37 | Katty | he really needs. |
| 17:39.38 | aptura | tc same. I need to get something to eat. |
| 17:39.47 | c4t3l | agreed! |
| 17:39.49 | mishehu | Katty: next time you're in chicago I'll get you one. my parents own an ice cream store here now. |
| 17:39.58 | Katty | mishehu: oh. is it kosher? |
| 17:40.04 | Qwell[] | yes |
| 17:40.11 | mishehu | Katty: I believe so. |
| 17:40.12 | Qwell[] | no meat at all |
| 17:40.52 | Katty | mishehu: hot dog (= |
| 17:40.53 | Qwell[] | especially not hooved meat |
| 17:40.53 | mvanbaak | lol |
| 17:40.53 | Qwell[] | hot dog? |
| 17:40.53 | Qwell[] | isn't that...very non-kosher? |
| 17:40.53 | KeNroM | can anyone help me |
| 17:40.53 | mvanbaak | shoarma icecream ! |
| 17:41.02 | Katty | KeNroM: i think you better post your problem again. |
| 17:41.18 | Katty | mishehu: i wouldn't mind a lil trip to chicago |
| 17:41.27 | Katty | mishehu: 8 hours on amtrak isn't bad, afterall. |
| 17:41.43 | mishehu | Katty: I know that not all the products are, but many of them are certified by OU. |
| 17:41.43 | mishehu | think the ice cream is one |
| 17:41.43 | mishehu | Katty: but I thought you don't eat ice cream |
| 17:41.43 | mishehu | at least not real ice cream |
| 17:41.53 | KeNroM | i have a closer campaign setup..and i also have a custom php webform..... when the user clicks on webform button it bring out window with this url |
| 17:41.57 | KeNroM | http://astguiclient.sourceforge.net/test_VICIDIAL_output.php?lead_id=112227 |
| 17:42.01 | Katty | mishehu: i'm actually omni again. |
| 17:42.04 | KeNroM | instead of my custom script |
| 17:42.05 | Katty | mishehu: sort of. |
| 17:42.25 | mishehu | Qwell[]: hot dogs can be kosher. Bests, Hebrew NAtional, etc. |
| 17:42.48 | Qwell[] | Bests sound like an uber-generic brand |
| 17:42.52 | mishehu | Katty: good to hear, now you can enjoy ice cream |
| 17:42.59 | Katty | ^_^ |
| 17:43.07 | Qwell[] | and many other wonderful foods ;) |
| 17:43.12 | Qwell[] | like...steak |
| 17:43.15 | Katty | ugah |
| 17:43.15 | Katty | no |
| 17:43.19 | Enron | Just confirmed it was a bad card |
| 17:43.20 | Qwell[] | and hamburgers |
| 17:43.22 | mishehu | ugah bugah |
| 17:43.23 | Enron | Thanks for all the help |
| 17:43.26 | KeNroM | anyonw? |
| 17:43.30 | Qwell[] | well, minus the ham, I suppose? |
| 17:43.37 | Katty | i only eat chicken and turkey. |
| 17:43.43 | Qwell[] | that's not omni :p |
| 17:43.45 | mishehu | Katty: boc boc? |
| 17:43.48 | mishehu | err |
| 17:43.52 | Qwell[] | that's like...vegetarian |
| 17:43.52 | mishehu | Katty: not parrot I hope |
| 17:43.53 | Katty | mishehu: berkok! |
| 17:44.09 | Katty | mishehu: no, i'm not going to eat your parrot. |
| 17:44.20 | Katty | too many feathers. |
| 17:44.25 | mishehu | Katty: she thanks you for not being interested in eating her. |
| 17:44.35 | Katty | hehe |
| 17:44.57 | Katty | mom's bird was kinda fiesty last time i went to visit. |
| 17:45.03 | Katty | don't think he's been handled enough lately. |
| 17:47.06 | KeNroM | can anyone help me plzzzz |
| 17:47.23 | DarKnesS_WolF | KeNroM: what is ur problem ? |
| 17:47.31 | KeNroM | i have a closer campaign setup..and i also have a custom php webform..... when the user clicks on webform button it bring out window with this url |
| 17:47.36 | KeNroM | http://astguiclient.sourceforge.net/test_VICIDIAL_output.php?lead_id=112227 |
| 17:47.59 | DarKnesS_WolF | KeNroM: where is the problem ? |
| 17:48.10 | KeNroM | it is to bring up a script... |
| 17:48.27 | KeNroM | http://www.alpha-barbados.com/scripts/dc/index.php |
| 17:49.05 | KeNroM | but when i click on the webform button it bring up...this |
| 17:49.12 | KeNroM | http://astguiclient.sourceforge.net/test_VICIDIAL_output.php?lead_id=112227 |
| 17:49.59 | DarKnesS_WolF | KeNroM: don't know :-s |
| 17:50.55 | Katty | file: pst. |
| 17:51.06 | file | no :( |
| 17:51.07 | KeNroM | :S..:( |
| 17:51.12 | Katty | file: :< |
| 17:51.16 | Katty | file: pst. not .pst |
| 17:51.34 | mishehu | .pst files suck |
| 17:51.40 | Katty | yes, yes they do. |
| 17:51.47 | mishehu | it's microsoft's way of saying "we're retards" |
| 17:51.52 | Katty | hehe yes |
| 17:51.57 | Katty | let's dump everything into a single file!! |
| 17:52.02 | Katty | that'll be great! |
| 17:52.11 | mishehu | Katty: actually, they had to do that because of how NTFS is... |
| 17:52.39 | aptura | katty so when that file corrupts you loose everything unless of course it was backed up |
| 17:52.47 | Katty | aptura: yep. |
| 17:52.55 | mishehu | NTFS is another alf bassed hackwards idea of theirs |
| 17:53.19 | mishehu | and it cannot handle lots of small files in any moderately efficient manner. |
| 17:53.30 | Katty | alf bassed hackwards? :P |
| 17:53.43 | *** join/#asterisk `pariah (n=josh@unaffiliated/pariah) |
| 17:53.45 | Katty | haha, that's great. |
| 17:53.53 | mishehu | Katty: yes! |
| 17:53.55 | Katty | i'm gonna remember that one ;) |
| 17:54.06 | *** join/#asterisk naitram (n=chatzill@216.77.58.40) |
| 17:55.43 | karlhaines | thats why ext3 rocks ;) |
| 17:55.50 | karlhaines | journaling fs |
| 17:56.11 | karlhaines | i guess reiser is supposed to be really good with small files but i never use it |
| 17:56.22 | Qwell[] | reiser is good at lots of things |
| 17:56.29 | Qwell[] | covering up murder is not one of them |
| 17:56.31 | *** join/#asterisk d00gster (n=doughant@bas1-toronto12-1128667111.dsl.bell.ca) |
| 17:56.40 | *** join/#asterisk flashnet (i=flashnet@admin.mojamreza.org) |
| 17:56.41 | *** join/#asterisk jkiff (n=jkiffmey@unaffiliated/vorondil) |
| 17:57.41 | Juggie | haha |
| 17:58.57 | [TK]D-Fender | Qwell[]: The jury's still out on that one IIRC;) |
| 17:59.25 | Qwell[] | well, I already passed judgment, so it's okay :P |
| 17:59.33 | Qwell[] | "judgment"? why does that look wrong? |
| 18:00.58 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 18:01.46 | [TK]D-Fender | Qwell[]: Because its unnatural to drop the "e" off "judge" to congugate it that way. |
| 18:01.58 | Qwell[] | quite |
| 18:02.03 | [TK]D-Fender | Qwell[]: Its American :) |
| 18:02.12 | Qwell[] | stupid en_US |
| 18:02.15 | [TK]D-Fender | :D |
| 18:02.42 | Katty | it's cold. |
| 18:02.46 | [TK]D-Fender | the problem with being the lowest common denominator is just how far you have to go ;) |
| 18:02.48 | wunderkin | judgemente? judgoument? |
| 18:02.52 | [TK]D-Fender | Katty: Look into the light! |
| 18:02.54 | coppice | why do americans have a fear of the letter U? :-) |
| 18:02.55 | Qwell[] | wunderkin: spanish? |
| 18:03.06 | wunderkin | lol no stupid joke |
| 18:03.08 | Katty | [TK]D-Fender: someone left the server room door open again. :< |
| 18:03.13 | Katty | stupid co-worker. |
| 18:04.19 | *** join/#asterisk crochat (n=crochat@84-74-150-141.dclient.hispeed.ch) |
| 18:09.27 | aptura | katty, make it a rfid accessable. |
| 18:09.52 | aptura | spring loaded doors with a 8 second buzzer. |
| 18:10.27 | *** join/#asterisk hmm-home (n=hmm-home@24-119-176-74.cpe.cableone.net) |
| 18:10.32 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 18:10.32 | *** mode/#asterisk [+o mog] by ChanServ |
| 18:12.09 | mishehu | bah. |
| 18:12.22 | [TK]D-Fender | Katty: Nah... let them in... then throw the switch to open the drop-floor doors and dunk them into the pool of sharks with friggen laser beams on their heads ! |
| 18:13.12 | aptura | or have a mechanic leg kick them as thay exit the door :) |
| 18:13.30 | aptura | mechanical :) |
| 18:15.08 | Katty | mechanical.. |
| 18:15.11 | Katty | mechanical squirrel. |
| 18:15.41 | Katty | [TK]D-Fender: yay, my rc.local worked!! |
| 18:16.05 | DarKnesS_WolF | :D |
| 18:16.29 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 18:16.39 | Katty | [TK]D-Fender: now does the modprobe and ztcfg and asterisk -vblahblah stuff go in rc.local too? |
| 18:17.06 | DarKnesS_WolF | Katty: u can use /etc/modules to run modules in boot time. |
| 18:17.08 | [TK]D-Fender | Katty: Depends. I'm not sure how Debian handles services. |
| 18:17.19 | Katty | [TK]D-Fender: i guess i could try and find out (= |
| 18:17.20 | DarKnesS_WolF | Katty: why u don't want to use init.d scrips for running asterisk in boot time :-s? |
| 18:17.25 | [TK]D-Fender | Katty: in RH you'd install the SysV stuff and set the init levels and that's it |
| 18:17.32 | Katty | sysv? |
| 18:17.53 | Katty | DarKnesS_WolF: i think /you/ think i know more than i do (= |
| 18:18.01 | Katty | DarKnesS_WolF: i'm learning linux right now. |
| 18:18.03 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 18:18.03 | *** mode/#asterisk [+o mog] by ChanServ |
| 18:18.16 | DarKnesS_WolF | Katty: ok great :-) then u can use a tool in debian called rcconf |
| 18:18.22 | DarKnesS_WolF | apt-get install rcconf |
| 18:18.23 | Katty | tool? debian? |
| 18:18.27 | Katty | meh |
| 18:18.28 | Katty | i don't want a tool |
| 18:18.31 | Katty | i want to know how to do it |
| 18:18.38 | DarKnesS_WolF | this tool will help u to put services in boot time. |
| 18:18.49 | cpm | without actually doing my homework, is there such a thing as a sip video phone that actually works? |
| 18:18.50 | DarKnesS_WolF | ok u can put the init script of asterisk into /etc/init.d |
| 18:19.00 | DarKnesS_WolF | and then link it to /etc/rc.runlever ur using. |
| 18:19.05 | Katty | uhh |
| 18:19.08 | DarKnesS_WolF | u can get the runleve from /etc/inittab |
| 18:19.17 | Katty | mew? |
| 18:19.22 | DarKnesS_WolF | :-s |
| 18:19.25 | Katty | can't ijust put everything into rc.local |
| 18:19.27 | Katty | it'd be so much easier |
| 18:19.32 | Katty | one spot |
| 18:19.35 | DarKnesS_WolF | it's not the right way. |
| 18:19.37 | Katty | everything there? |
| 18:19.40 | Katty | does it get the job done? |
| 18:19.49 | Katty | if it gets the job done, why is it not the Right Way(tm) |
| 18:19.54 | DarKnesS_WolF | i think so |
| 18:20.03 | Vec2 | Does anyone know if the T38 pass through problems have been resolved in asterisk 1.4.4 ? |
| 18:20.04 | Katty | then if it gets the job done...who cares? :P |
| 18:20.34 | DarKnesS_WolF | Katty: hmmm do it the way u like :-) but it's always better to do it the right way if u really want to learn. |
| 18:20.41 | [TK]D-Fender | cpm: The Grandstream one works, but being GS I'd follow JBOT's advise. eyeBeam works great. Heard the Tornado works as well (but quality sucks) |
| 18:20.50 | *** join/#asterisk crochat (n=crochat@84-74-150-141.dclient.hispeed.ch) |
| 18:20.51 | Katty | DarKnesS_WolF: right, so, regardless...no tool |
| 18:20.57 | cpm | eyeBeam? |
| 18:20.58 | mishehu | DarKnesS_WolF: the "right way" is very weighted |
| 18:21.07 | cpm | thanks [TK]D-Fender |
| 18:21.08 | mishehu | and also depends on the distro |
| 18:21.14 | DarKnesS_WolF | cpm: yes it's the payed version of xlite it did work here too |
| 18:21.29 | cpm | oh, softphone? |
| 18:21.31 | cpm | no joy |
| 18:21.32 | DarKnesS_WolF | same as ekiga if i recall correctly i did play with video like few months ago. |
| 18:21.38 | mishehu | cpm: take off your beer googles |
| 18:22.08 | DarKnesS_WolF | Katty: rcconf it just a tool to create the link for the services from /etc/init.d/ to /etc/rc.runleve /etc/rc5.d or whatever runlever ur using. |
| 18:22.15 | DarKnesS_WolF | hmm i think we are way offtopic :-D |
| 18:23.05 | Katty | DarKnesS_WolF: and /why/ is it better to do it that way |
| 18:23.25 | [TK]D-Fender | s/we are/I am/ |
| 18:25.24 | mishehu | I say we blame [TK]D-Fender-Bender |
| 18:26.04 | LeddyHM | me too |
| 18:26.06 | LeddyHM | all his fault |
| 18:26.08 | [TK]D-Fender | And people keep saying I don't DO anything... we here you have it! I DO! |
| 18:26.17 | [TK]D-Fender | :D |
| 18:26.49 | Katty | you do everthing. |
| 18:27.40 | Katty | [TK]D-Fender: that's not an excuse for you to go on vacation. |
| 18:28.06 | *** join/#asterisk mikebwilliams (n=mikebwil@12-218-71-62.client.mchsi.com) |
| 18:28.09 | karlhaines | someone in here yesterday said that they worked at a VOIP provider |
| 18:28.19 | [TK]D-Fender | Katty: But I do... everthing! ;) |
| 18:28.20 | coppice | If life gives you lemons, make lemon tea... if life give you tea leaves |
| 18:28.20 | karlhaines | whoever you were, are you here? |
| 18:28.23 | mikebwilliams | hey all, i've got a problem where i get an error when trying to dial out over a pri: chan_zap.c: Unable to determine channel for data PRI/98125843779 |
| 18:28.40 | Katty | [TK]D-Fender: /sob |
| 18:28.51 | mikebwilliams | anybody had that one before? |
| 18:29.32 | Katty | we don't have a pri. |
| 18:29.35 | Katty | so...count me out (= |
| 18:29.57 | Katty | [TK]D-Fender: that new card came in today ^_^ |
| 18:30.08 | mikebwilliams | hmm |
| 18:30.39 | [TK]D-Fender | Katty: neat-o |
| 18:31.00 | [TK]D-Fender | mikeperhaps you should pastebin the entire CLI output of your failed call |
| 18:31.02 | karlhaines | anyone know a place that will port my DIDs in a timely manner?? I've paid les.net to do this, but its been 30 days already, i need to get this done, NOW |
| 18:31.35 | *** join/#asterisk ESCulapio__ (n=The@66.44.88.200.l.sta.codetel.net.do) |
| 18:32.31 | mikebwilliams | [TK]D-Fender: |
| 18:34.28 | mikebwilliams | what's the url to paste things online? |
| 18:35.10 | [TK]D-Fender | ~pb |
| 18:35.21 | jbot | somebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 18:35.33 | *** join/#asterisk asdx (n=diego@200.26.178.39) |
| 18:35.57 | mishehu | and I thought pb was for peanut butter |
| 18:36.11 | mikebwilliams | http://hashbin.com/46b.html |
| 18:36.19 | ESCulapio__ | quien habla espanol |
| 18:36.28 | coppice | peanut butter is a paste, isn't it? |
| 18:36.28 | mikebwilliams | that's the debug output |
| 18:36.40 | mikebwilliams | coppice: i guess |
| 18:38.22 | mishehu | coppice: indeed, and probably before it goes into a jar, it's in a bin |
| 18:38.51 | mishehu | ESCulapio__: no recuerdo mucho espanol. hablo ingles y hebreo |
| 18:39.02 | mikebwilliams | i'll be on and off... vpn' |
| 18:39.42 | ESCulapio__ | mishehu, tengo una pregunta y/o una confusión me puedes ayudar |
| 18:40.50 | [TK]D-Fender | mikebwilliams: May 15 12:12:14 VERBOSE[9715] logger.c: -- Executing Dial("IAX2/4444-1", "ZAP/PRI/98125843779|300|tr") in new stack |
| 18:41.06 | [TK]D-Fender | mikebwilliams: PRI is not part of a valid channel formatting |
| 18:41.21 | [TK]D-Fender | mikebwilliams: You need to pick a channel, or channel group |
| 18:41.22 | mikebwilliams | hmm |
| 18:41.32 | ESCulapio__ | un amigo contrato un usuario en un proveedor, si lo configuro en un softphone puedo recibir llamadas pero si lo registro en asterisk no me entran las llamadas "no puedo recibir las llamadas" |
| 18:41.36 | mikebwilliams | could I name that group "PRI" |
| 18:41.42 | ESCulapio__ | mishehu, un amigo contrato un usuario en un proveedor, si lo configuro en un softphone puedo recibir llamadas pero si lo registro en asterisk no me entran las llamadas "no puedo recibir las llamadas" |
| 18:41.44 | mishehu | ESCulapio__: hay poco que comprender (?) espanol aqui |
| 18:42.00 | ESCulapio__ | mishehu, tratare en ingles |
| 18:42.26 | [TK]D-Fender | mikebwilliams: No. |
| 18:42.39 | luckyone_ | ESCulapio__: un pocito |
| 18:42.53 | mishehu | ESCulapio__: I don't remember enough to even fully understand your problem... or to tell you there are probably very few people who speak spanish |
| 18:42.56 | [TK]D-Fender | mikebwilliams: Go read up on zapata.conf on the WIKI and fix your dialplan |
| 18:43.00 | karlhaines | anyone know a place that will port my DIDs in a timely manner?? I've paid les.net to do this, but its been 30 days already, i need to get this done, NOW |
| 18:43.09 | mikebwilliams | ok, i've been looking through it, but i think i'm confused |
| 18:43.14 | [TK]D-Fender | karlhaines: what area codes? |
| 18:43.19 | mishehu | but anyway, I must throw a pizza into the oven, I'm starving |
| 18:43.23 | mikebwilliams | i'll be back in a few when i'm better informed |
| 18:43.25 | Katty | [TK]D-Fender: before, i used to type modprobe zaptel, then insmod wctdm...and i know i added wctdm as a module option... |
| 18:43.42 | [TK]D-Fender | mikebwilliams: "Dial(Zap/1/123456)" , "Dial(Zap/g1/1234567890), etc |
| 18:43.46 | Katty | [TK]D-Fender: i did a locate, and found it in /lib/modules/mykernel/misc/wctdm.ko |
| 18:43.47 | karlhaines | [TK]D-Fender: 615 |
| 18:44.04 | Katty | [TK]D-Fender: i'm not sure what i'm doing wrong.. |
| 18:44.16 | mikebwilliams | i'm thinking what happened is that i renamed my zap channels in freepbx, and then didn't bother with my channels, in zapata.conf, and didn't even but a "g" before the name in freepbx |
| 18:44.34 | [TK]D-Fender | karlhaines: Tried unlimitel? |
| 18:44.35 | ESCulapio__ | mishehu, a friend contract the service of a supplier "an account sip" |
| 18:44.57 | [TK]D-Fender | ~freepbx |
| 18:45.17 | jbot | i guess freepbx is unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 18:45.30 | [TK]D-Fender | mikebwilliams: And it was a mistake to enter "PRI" wherever you did. |
| 18:45.32 | ESCulapio__ | mishehu, if I form it in softphone I can receive the calls, the calls enter |
| 18:45.44 | karlhaines | [TK]D-Fender: do you have service with them? |
| 18:45.54 | mikebwilliams | [TK]D-Fender: got it, back to using numbers, and more things are working, thanks |
| 18:46.17 | ESCulapio__ | mishehu, but if it registry in asterisk the calls do not enter |
| 18:46.30 | [TK]D-Fender | karlhaines: Not personally, but several of my clients, yes. Good support & quality |
| 18:49.35 | mishehu | ESCulapio__: are you actually seeing it register to asterisk? |
| 18:51.55 | ESCulapio__ | mishehu, yes, the flames leave to me but they do not enter |
| 18:52.45 | ESCulapio__ | and have type=friend |
| 18:58.59 | *** join/#asterisk tschafer (n=tschafer@207.241.143.246) |
| 19:01.25 | naitram | how do you get asterisk to startup on bootup on linux (debian). I now it should be a debian question but haven't figured it out |
| 19:01.26 | Qwell[] | ctooley: still around? |
| 19:02.07 | DarKnesS_WolF | naitram: in /asterisk/source/contrib/init/rc.asterisk.debian |
| 19:02.11 | DarKnesS_WolF | if i recall correctly the path |
| 19:02.22 | DarKnesS_WolF | take this file copy to ur /etc/init.d/asterisk |
| 19:02.42 | DarKnesS_WolF | and then link it with to ur /etc/rcrunlevel.d |
| 19:03.04 | DarKnesS_WolF | or use rcconf to add the services to the boot time after u do copy the init script into ur /etc/init.d |
| 19:03.42 | naitram | DarKnesS_WolF: Ok, got ya. Thanks will try. |
| 19:04.23 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 19:05.03 | karlhaines | [TK]D-Fender: ever heard from any of those customers how long their ports took? (i've sent them a contact req). les.net said it would take 10 days, and its been over a month |
| 19:05.19 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 19:05.32 | [TK]D-Fender | karlhaines: Sorry no..... |
| 19:07.51 | *** join/#asterisk rdb_ (n=rdb@gw.avila.edu) |
| 19:10.32 | *** join/#asterisk bbryant (i=brett@nat/digium/x-17b45750599878fc) |
| 19:11.23 | Enron | anyone know where I can purchase a digium nic in LA |
| 19:11.32 | Enron | don't want to wait overnight |
| 19:11.40 | *** join/#asterisk Chris-NB (n=chris@home.fuerstaller.com) |
| 19:11.46 | Qwell[] | Digium doesn't sell NICs |
| 19:12.32 | Corydon-w | Qwell[]: T1 cards are technically NICs... just not Ethernet NICs |
| 19:13.04 | *** join/#asterisk boch (n=fran@190.48.206.133) |
| 19:13.18 | Qwell[] | Enron: You may be able to get in touch with a distributor, and find a local reseller |
| 19:13.22 | boch | hi |
| 19:13.25 | Katty | so. in 1.4... in the zaptel.conf, you just need loadzone, defaultzone, and your whatcardisit=channels |
| 19:13.28 | Qwell[] | there is a list on digium.com, I believe |
| 19:13.30 | Enron | none seem to have it in stock |
| 19:13.53 | Katty | right? |
| 19:13.53 | Corydon-w | Katty: and a span definition, if applicable |
| 19:13.54 | boch | mates, do you know how to playback all files in a dir? want to listem and delete if apropiated |
| 19:14.01 | Katty | Corydon-w: what's a "span definition" |
| 19:14.13 | Corydon-w | Katty: T1s have span= definitions |
| 19:14.18 | Katty | k |
| 19:14.24 | Katty | none of that just yet (= |
| 19:14.34 | Katty | Corydon-w: thanks (= |
| 19:14.36 | Corydon-w | E1s, too |
| 19:15.18 | Corydon-w | and a trunkgroup mapping, for NFAS or D-channels not on the last channel of a T1 span |
| 19:15.19 | [TK]D-Fender | boch: Go write an AGI |
| 19:15.50 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 19:16.19 | boch | [TK]D-Fender, heh i had foget them thanks |
| 19:16.28 | boch | too much AEL |
| 19:24.04 | errr | When in the voicemail is it possible to get the Caller ID info to display on the phone from the voicemail you are listening to? |
| 19:25.20 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 19:25.33 | *** part/#asterisk ctooley (n=ctooley@rrcs-71-42-115-242.sw.biz.rr.com) |
| 19:27.22 | irule | is there such a thing on line as a regular/traditional PBX flow chart that may help me design my dialplan? |
| 19:27.50 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 19:29.27 | *** join/#asterisk xpot (n=jim@c-71-195-241-115.hsd1.ut.comcast.net) |
| 19:29.50 | Strom_M | irule: not that I'm aware of; is it dialplan planning you're trying to work out, or is it your numbering plan that you're trying to engineer? |
| 19:33.10 | Katty | i'm having brain failure. i'm looking at zapata.conf, and i have a 'from-zap' context with all the settings and such under it, then i have two more contexts with channels under it. |
| 19:33.15 | irule | Strom_M I dont know how you may call it, I an starting to draw a graphical diagram on visio defining all steps that may be taken when a call is handled by [default], I just think it will be easyer down the line to prepare this before actually writing the code to the dialplan since I will know exactly what I am going to do |
| 19:35.45 | Katty | nevermind. |
| 19:37.14 | *** join/#asterisk Zipper_32 (n=None@142.232.142.80) |
| 19:37.24 | *** join/#asterisk ta^3 (n=tacvbo@189.146.195.139) |
| 19:37.37 | Zipper_32 | What parameter can I use in Musiconhold.conf to allow music playback from where it left off? |
| 19:38.14 | Strom_M | irule: well first off, [default] shouldnt be your primary context |
| 19:38.15 | *** join/#asterisk sharp (n=sharp@pool-71-242-178-140.phlapa.east.verizon.net) |
| 19:38.33 | Strom_M | there should be nothing in [default]; everything should be appropriately named |
| 19:39.16 | irule | as far as Im concerned an incomming call goes straight to default, right? |
| 19:39.22 | irule | from zap |
| 19:39.30 | Strom_M | no, that's not a good idea |
| 19:39.37 | Strom_M | incoming calls should land in their own context |
| 19:40.10 | irule | oh, ok, I see; thanks! |
| 19:40.53 | irule | how may I configure some other context to be executed instead of [default]? |
| 19:41.11 | Strom_M | set it up in your channel driver |
| 19:45.27 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 19:45.58 | Katty | rxgain is outgoing? |
| 19:46.02 | Katty | and txgain is incoming? |
| 19:46.07 | Katty | or do i have those backwards? |
| 19:46.17 | *** join/#asterisk afrosheen (n=cj@207.71.49.164) |
| 19:47.12 | *** join/#asterisk angom (n=angom@red-corp-201.143.54.251.telnor.net) |
| 19:47.22 | afrosheen | hey guys, if asterisk won't start at boot because zaptel takes awhile to create the devfs devices, what should I try? |
| 19:47.31 | Qwell[] | wait longer? |
| 19:47.49 | *** part/#asterisk angom (n=angom@red-corp-201.143.54.251.telnor.net) |
| 19:48.06 | afrosheen | like by changing the S number? |
| 19:48.42 | afrosheen | it was working fine with the regular asterisk, but after the business edition install, it's jacked up |
| 19:49.38 | Qwell[] | ...why don't you just...call support? |
| 19:50.33 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 19:55.27 | *** join/#asterisk nick125_lappy (n=nick@atarack/staff/nick125) |
| 19:56.13 | *** join/#asterisk techie (n=gus@voip.routedsystems.com) |
| 19:58.24 | afrosheen | Qwell[], ugh..that's all everyone says when I mention ABE |
| 19:58.36 | Qwell[] | well, that's kinda the point of buying it |
| 19:58.58 | Strom_M | but I don't WANNA use the support I've already paid for |
| 19:59.08 | anonymouz666 | unknow RTP codec 100 received |
| 19:59.14 | afrosheen | man I'm gonna leave before I start flaming..good day |
| 19:59.15 | anonymouz666 | which is codec 100? |
| 19:59.22 | *** part/#asterisk afrosheen (n=cj@207.71.49.164) |
| 19:59.30 | hansin321 | Does anyone know if it makes sense that if I want to create a conference bridge (MeetMe) in a work environment that is SIP, that I could set up an * server and have it act as a User Agent, assign an extension, and have it only to conference bridge stuff? Of does does it need to be something than just another UA in the SIP system? |
| 19:59.44 | *** join/#asterisk Cresl1n (i=matt@nat/digium/x-754dcbeca01b3ef7) |
| 19:59.44 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 20:00.00 | hansin321 | Basically make it another end-point like any other phone. |
| 20:00.45 | hansin321 | So it would have to be an end point that could answer many calls. Does this work? Thanks in advance... |
| 20:02.05 | [TK]D-Fender | hansin321: ...HUH?!!? |
| 20:02.06 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 20:02.20 | [hC] | anyone have any idea why when i execute Busy() it hangs up my call instead of playing busy tones |
| 20:02.39 | [TK]D-Fender | hansin321: MeetMe is an App in *. * is already B2BUA. You should know what all this implies to your needs |
| 20:03.03 | [TK]D-Fender | [hC]: Because the channel has been answered and is capable of support OOB signaling for that state. |
| 20:04.05 | [hC] | [TK]D-Fender: The call is coming in over IAX |
| 20:04.13 | [hC] | what would i look at doing to resolve that? |
| 20:04.26 | [hC] | I do infact do an Answer(), but i didnt think that was bad practice. |
| 20:04.33 | [TK]D-Fender | [hC]: Answer the call, THEN busy it out. |
| 20:04.53 | [TK]D-Fender | [hC]: You may want to use "playtones" instead of Busy if that doesn't solve it |
| 20:05.54 | hansin321 | [TK]D-Fender: Thanks. I need to do more homework I am sure, I just wanted to make sure I can set * up as a conference bridge, and then just assign it an extension like any other phone (we use an external SIP provider). I wasn't sure if a confernce bridge needed any special setup separate from being another UA/end point in the SIP cloud. |
| 20:06.19 | [TK]D-Fender | hansin321: What (if anything) are you using * for right now? |
| 20:06.39 | [hC] | [TK]D-Fender: Huh... I have one box that takes the call in via zap, passes it to me via IAX, my box taking the call over IAX does an Answer() - Plays an IVR. Select an option on the ivr, place an outbound IAX call back to the box w/ the pri in it. The call tries to go out PRI, and returns busy... So, it returns "BUSY" over IAX to the box with the ivr on it, and it plays Busy() |
| 20:06.48 | [TK]D-Fender | hansin321: And yes, * IS just like "any other phone". Thats what being a B2BUA means |
| 20:07.04 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
| 20:07.20 | [TK]D-Fender | [hC]: So it kicks you back to the other box? |
| 20:08.12 | hansin321 | I have, at home set it up on a server to answer SIP calls from the internet and then I can dial out through an FXO card to my regular POTS line. For example, I can call from a laptop from a WiFi hotspot to my server, and then make a call on my home POTS line. But I am pretty new to it. |
| 20:08.23 | [hC] | [TK]D-Fender: Its getting kind of confusing the way im talking about it, not sure what box you mean by the other one... :P |
| 20:08.54 | hansin321 | I know, cell phones are cheaper, but thought this would be cool for international use or something. More fun that anything. |
| 20:08.58 | *** join/#asterisk shinao1 (n=shinao1@196.3.63.252) |
| 20:09.05 | nick125_lappy | GCCTV, whee!! |
| 20:09.07 | [hC] | [TK]D-Fender: lets say, pribox, ivrbox. Call comes in to pribox, goes via IAX to ivrbox. ivrbox takes ivr option, and plces call TO pribox. pribox returns busy over IAX to ivrbox. ivrbox executes Busy() |
| 20:09.14 | [TK]D-Fender | hansin321: I mean in the environment where you are considering using MeetMe. Where is * in that picture NOW? |
| 20:09.45 | [TK]D-Fender | [hC]: So you are using PRIBOX as a mere terminator? |
| 20:09.48 | [hC] | [TK]D-Fender: then, instead of busy tones playing, it just drops the call. |
| 20:09.54 | [hC] | [TK]D-Fender: yes. |
| 20:10.07 | [hC] | [TK]D-Fender: and ivrbox is what does the Answer() before playing the ivr. |
| 20:10.16 | [TK]D-Fender | [hC]: have you confirmed if the end caller gets TELCO busy? |
| 20:10.34 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 20:10.39 | [TK]D-Fender | [hC]: I might try "Playtones" here in this case. |
| 20:10.41 | [hC] | [TK]D-Fender: i am calling and testing it from my cell, i get dropped instead of busy. if i call the nbr direct, its busy. |
| 20:10.58 | [hC] | It seems like instead of it sitting and waiting at Busy() its not figuring its done and finishes its macro |
| 20:11.28 | [hC] | ok |
| 20:11.31 | [hC] | it makes sense now |
| 20:11.34 | [hC] | Busy() does not play tones |
| 20:11.41 | [hC] | it returns the busy oob signal |
| 20:11.42 | [hC] | period |
| 20:11.52 | [hC] | Note that this command does not actually play a busy tone to the user. If you wish to do that, call Playtones(busy) before calling this command. |
| 20:11.54 | [hC] | from wiki |
| 20:12.03 | hansin321 | [TK]D-Fender: It isn't. We recently installed Polycom VoIP phones, and utilize a Qwest MGCP service. We are still paying for a T1 so we can utilize the bridge on our old PBX. I figure I can't do this as easy as long as we are using MGCP, but I think the service may go SIP at some point. I thought, hey, lets just build a bridge with * (or similar) once we fo to SIP, and just assign it an extension like any other phone. You call |
| 20:12.05 | [hC] | in most cases a sip phone will play busy |
| 20:12.07 | [hC] | but im calling from zap. |
| 20:12.14 | [hC] | there's no oob signalling to tell it to play busy |
| 20:13.46 | Katty | what does the command "insmod wctdm" actually...do. |
| 20:13.53 | Katty | i get wctdm is for the tdm card |
| 20:14.03 | Katty | what's insmod? |
| 20:14.46 | Qwell[] | Katty: use modprobe |
| 20:14.46 | LeddyHM | www.google.com |
| 20:14.55 | Qwell[] | insmod is...meh |
| 20:14.55 | LeddyHM | sounds like you need a linux administration guide |
| 20:15.07 | Katty | Qwell[]: so we just don't do it anymore eh? |
| 20:15.09 | Qwell[] | requires full path, plus extension, plus it doesn't automatically load dependencies |
| 20:15.17 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 20:15.18 | Katty | Qwell[]: the modprobe zaptel thing takes care of wctdm driver? |
| 20:15.22 | Qwell[] | Katty: modprobe is far better than insmod, in most cases |
| 20:15.27 | Qwell[] | no, you need to modprobe wctdm |
| 20:15.30 | LeddyHM | make config |
| 20:15.33 | Qwell[] | zaptel will automatically come with it |
| 20:15.41 | [TK]D-Fender | hansin321: You should flash your phones to SIP, and put * in the middle. |
| 20:15.57 | Katty | ok. |
| 20:16.06 | Katty | so modprobe zaptel loads zaptel config files. |
| 20:16.15 | [TK]D-Fender | [hC]: Your ZAP is PRI, and therefor IS OOF |
| 20:16.17 | Katty | and i need to modprobe wctdm /before/ that so it..sees what to load config files for |
| 20:16.17 | [TK]D-Fender | OOB* |
| 20:16.20 | Katty | right? |
| 20:18.22 | [hC] | [TK]D-Fender: I think i know whats up. when the original call comes into pribox, it dials ivrbox with a Macro() |
| 20:18.32 | [hC] | it gets busy back, but then doesnt play the tones, it exits the macro |
| 20:19.01 | *** join/#asterisk cnile (n=canabis@ip70-172-239-41.br.br.cox.net) |
| 20:19.10 | [TK]D-Fender | [hC]: Yeah.. the call IS still OOB being IAX between the two as well |
| 20:19.14 | *** join/#asterisk clive- (n=pirch@dsl-242-139-161.telkomadsl.co.za) |
| 20:19.49 | hansin321 | [TK]D-Fender: I understand, but that wouldn't fly with who makes the decisions. I'm low on the totem pole, and we are a medium sized company. I was encouraged though by a superior to see if I could come up with some sort of solution. By setting up a server that is nothing more than an end-point/UA, if this is possible, it is much more likely that I could get someone to listen and maybe give it a shot. I don't think *'s MGCP supp |
| 20:20.40 | [TK]D-Fender | hansin321: No, at the point where your termination can be converted to SIP, then I would schedule the full re-org |
| 20:20.53 | [TK]D-Fender | hansin321: And indeed.. I wouldn't touch MGCP in *. |
| 20:21.09 | clive- | does anyone here deal with a comnpany called "telcan" ? |
| 20:21.14 | [TK]D-Fender | hansin321: But this company has you by the balls right now. NOT GOOD. |
| 20:21.28 | *** join/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 20:21.35 | *** join/#asterisk mocker (n=user@198.247.173.227) |
| 20:21.41 | hansin321 | [TK]D-Fender: Thanks. So you don't think you could set * as a UA and have it do bridge functionality? |
| 20:21.51 | *** join/#asterisk bkw_ (n=brian@adsl-70-143-39-207.dsl.tul2ok.sbcglobal.net) |
| 20:22.02 | *** join/#asterisk BSD_tech (n=BSDTech@adsl-69-230-174-37.dsl.irvnca.pacbell.net) |
| 20:22.14 | mocker | Am I crazy for thinking there was a Dial variable that purposely kept the call going through asterisk instead of allowing it to be a peer-peer call? |
| 20:22.29 | clive- | brian hi...ever head of a company called telcan/ or callture? |
| 20:23.12 | clive- | ever heard.... |
| 20:23.12 | wunderkin | mocker, well if there are any options specified that require to listen for dtmf.. then it wont reinvite.. |
| 20:23.12 | *** join/#asterisk coolbeans (n=null@adsl-074-247-038-249.sip.bhm.bellsouth.net) |
| 20:23.12 | [TK]D-Fender | hansin321: Well your phones being MGCP would need to send the call to * via that, no? or through your provider more like.... |
| 20:23.12 | clive- | mocker I think its notransfer or something |
| 20:23.13 | coolbeans | Hi all. Anyone have a good aastra.cfg file for asterisk that has working forward and dnd softkeys? |
| 20:23.28 | clive- | canreinvite=no |
| 20:23.52 | clive- | mocker in your sip.conf |
| 20:24.30 | mocker | Ahh, for some reason I thought there was a dial option.. |
| 20:25.07 | mocker | The issue is that when calls are blind transferred, the person being transferred doesn't hear anything until someone (voicemail or person) picks up. |
| 20:25.12 | mocker | Just dead air, no ring or moh. |
| 20:25.32 | mocker | My first thought was that asterisk may be dumping it out of the media path. |
| 20:26.13 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 20:26.44 | hansin321 | [TK]D-Fender: Well, if Qwest switches their service to SIP (it sounds like it might happen), we then basically upgrade the firmware on our Polycom phones (and what ever HW/Router changes we need for what interfaces with Qwest's lines). At that point it might work. As long as we are on MGCP I figure it is still in the idea stage. Thanks for the help. |
| 20:27.27 | [TK]D-Fender | hansin321: NP, and everything is the way it should be for NOW, but "now's" circumstances should be changed ASAP :) |
| 20:27.39 | *** join/#asterisk kink0 (n=k@161.pool62-37-205.static.orange.es) |
| 20:27.40 | kink0 | hi |
| 20:28.00 | mocker | Well, that doesn't seem to be it. |
| 20:29.00 | BSD_tech | ok this is not funny not getting audio |
| 20:29.35 | hansin321 | Well, maybe we'll change it one step at a time ;) |
| 20:31.50 | BSD_tech | when is the codefest ? |
| 20:31.53 | BSD_tech | and where |
| 20:32.04 | kink0 | any idea about how to limit CPS and balance load to N Asterisk boxes ? |
| 20:32.26 | kink0 | I mean, balance based on CPS instead on used channels |
| 20:32.44 | aptura | whats the deal with asterisk not relasing vars out of memory? I had a different path name for asterisk music on hold and it insist on reading from another directory after reloading asterisk |
| 20:33.19 | aptura | pointer issues? |
| 20:33.39 | [TK]D-Fender | ok, back in a few hours |
| 20:34.00 | *** join/#asterisk funkmaster (n=funky1@vhe-364501.sshn.net) |
| 20:34.25 | coolbeans | Hey guys, how would I setup an auto call forward extension? |
| 20:34.41 | funkmaster | hi there :) is there a command i can use to see the last #number of callers or callers within last hour/day etc? |
| 20:35.41 | *** part/#asterisk clive- (n=pirch@dsl-242-139-161.telkomadsl.co.za) |
| 20:36.37 | *** join/#asterisk Dandan (n=dandan@ip68-9-233-149.ri.ri.cox.net) |
| 20:36.39 | Dandan | hey all :) |
| 20:36.50 | Dandan | so it is time to try some web interface for * |
| 20:36.53 | Dandan | any recommendations? |
| 20:38.31 | Dandan | all dead? :) deaf? :) |
| 20:38.52 | rudholm | funkmaster: egrep `date +%F` Master.csv | awk -F , '{ print $2 }' | uniq | wc -l |
| 20:39.10 | rudholm | funkmaster: that should tell you how many unique callers called since the start of today. |
| 20:39.49 | *** join/#asterisk nomadsoul (n=nomadsou@unaffiliated/nomadsoul) |
| 20:39.57 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 20:39.57 | funkmaster | rudholm: thx a bunch :) |
| 20:40.24 | Qwell[] | that's it |
| 20:40.26 | Qwell[] | it's settled |
| 20:40.31 | Qwell[] | I'm changing my name to have a comma in it |
| 20:40.52 | Qwell[] | I'm going to make every company I do business with, special case me |
| 20:40.54 | coolbeans | Windows Update: The VoIP killer. |
| 20:41.45 | Nivex | s/VoIP/\*/ |
| 20:41.55 | mog | Qwell[], you could be jason O'Qwell and do about the same thing |
| 20:42.09 | *** join/#asterisk NoVaZuR (n=novazur@LLamentin-151-13-252.w81-248.abo.wanadoo.fr) |
| 20:42.12 | Qwell[] | mog: most companies already special case '...but none special case a comma :P |
| 20:42.19 | coolbeans | lol |
| 20:42.22 | Qwell[] | I want free hotel rooms, damnit |
| 20:42.25 | Qwell[] | ;) |
| 20:42.53 | mog | i dont think you can legally jam a comma into your name |
| 20:43.01 | mog | but if you try let me know |
| 20:43.02 | Qwell[] | I can if I go through the legal process |
| 20:43.22 | Strom_M | i dont think there's a law that says "names must consist only of letters" |
| 20:43.23 | Qwell[] | OR, I could be like, something utf-8 |
| 20:43.36 | Qwell[] | or, a jpeg |
| 20:43.37 | Strom_M | hell, you could name yourself SanDeE* |
| 20:43.41 | NoVaZuR | hi ! I need some help (with my very bad english). If I call an FXS extension with Zap/1, in the manager, I get an ExtensionStatus Event, if I call it with Zap/1r1 (to have CID on my french DECT phone), I don't get any ExtensionStatus Event. |
| 20:43.49 | Qwell[] | make people convert their firstname column be an image field |
| 20:44.28 | mog | qwell i dont know about federally but in past judges have thrown out peoples attempts to have silly names |
| 20:44.36 | Qwell[] | lame |
| 20:44.37 | NoVaZuR | Someone have an idea to help me ? |
| 20:44.49 | Qwell[] | I'll have to do it in a state that's laid back |
| 20:44.51 | mog | like ahjbfkjnfjwnf , pronounced john |
| 20:45.04 | coolbeans | For you Aastra 480x users, how do you setup a forward softkey on the phone? I'm using aastra.cfg files successfully, but it escapes me how to setup Call Forwarding or just simple extension forwarding. Our Polycom's just sort of work. |
| 20:45.19 | coolbeans | And the book is vague. |
| 20:45.27 | coolbeans | And the page on voip-info.com is blank. |
| 20:45.44 | coolbeans | And any help would be greatly apprecaited. |
| 20:45.48 | coolbeans | And O'Qwell rocks. |
| 20:46.07 | *** part/#asterisk naitram (n=chatzill@216.77.58.40) |
| 20:46.33 | NoVaZuR | nobody ? |
| 20:48.15 | *** join/#asterisk cspot (i=cspot@ip68-1-63-100.pn.at.cox.net) |
| 20:49.13 | *** join/#asterisk nick125_lappy (n=nick@atarack/staff/nick125) |
| 20:51.02 | NoVaZuR | I'm surely not on the right channel... |
| 20:51.04 | *** join/#asterisk Lithium_Ion (n=lithium@d57-71-44.home.cgocable.net) |
| 20:52.05 | Lithium_Ion | Hey all. I have a queue in asterisk that instead of ringing the 3 extentions that are members like its supposed to, it rings every extention. Any ideas? |
| 20:53.48 | coolbeans | Did you reload the config? |
| 20:53.59 | Dandan | oops |
| 20:54.17 | Lithium_Ion | Yes |
| 20:55.00 | coolbeans | Hrm... It will only ring members... |
| 20:55.17 | Lithium_Ion | It's really weird |
| 20:55.19 | coolbeans | Paste your queues.conf and extensions.conf to the pastbin of your choice and paste the URLs in channel. |
| 20:55.58 | Lithium_Ion | pastebin? |
| 20:56.30 | coolbeans | http://www.google.com/search?q=pastebin |
| 20:57.49 | *** join/#asterisk d00gster (n=doughant@bas1-toronto12-1128667111.dsl.bell.ca) |
| 20:59.29 | funkmaster | rudholm: didn't work actually |
| 21:00.26 | funkmaster | was lookgin for a command to see the calls received during the day through the cli console |
| 21:00.30 | *** join/#asterisk nick125_ (n=upirc@atarack/staff/nick125) |
| 21:01.20 | nick125_ | afternoon everyone |
| 21:01.22 | Qwell[] | prepend it with a ! |
| 21:01.30 | nick125_ | yeah! |
| 21:01.35 | Qwell[] | PREpend |
| 21:01.37 | Qwell[] | :p |
| 21:01.40 | Qwell[] | !yeah |
| 21:01.52 | nick125_ | !yeah |
| 21:02.20 | Lithium_Ion | http://pastebin.ca/490097 |
| 21:04.10 | nick125_ | I wish I had tab complete on my treo :( |
| 21:04.46 | aptura | to bad there wasnt a channel related to telcom |
| 21:05.13 | nick125_ | aptura: you could start one :-) |
| 21:05.26 | aptura | Dont think there would be enough interest |
| 21:06.03 | nick125_ | you never know until you try |
| 21:09.00 | Katty | so...who's someone i know that wouldn't mind being a professional consultant for me? |
| 21:10.04 | Lithium_Ion | coolbeans: http://pastebin.ca/490097 |
| 21:10.37 | Qwell[] | Katty: a "professional consultant"? |
| 21:10.54 | aptura | Katty, got your hands full and not enough time for * |
| 21:11.06 | *** join/#asterisk nick125_ (n=upirc@atarack/staff/nick125) |
| 21:11.22 | *** join/#asterisk kiscokid (n=ron@208.106.33.66) |
| 21:11.30 | nick125_ | that button didn't work.... |
| 21:13.29 | nick125_ | so, what's everyone up to? |
| 21:13.31 | Katty | more like i need someone to teach me. |
| 21:13.39 | Katty | too much to learn, not enough time. |
| 21:13.53 | Katty | and then general help when i get in over my head, etc. |
| 21:14.01 | *** join/#asterisk _VoiceMeUp_COM (n=Miranda@145-27.mc.cite.net) |
| 21:14.20 | kink0 | when you need a hand to help you, seek at the end of your arm :) |
| 21:14.23 | kiscokid | Katty: you could go to an Asterisk boot camp |
| 21:14.32 | Katty | they can't be without me here |
| 21:14.37 | Katty | i'm the only tech |
| 21:14.43 | Katty | well, the only one that knows anything heh |
| 21:14.49 | Katty | we've got a cable runner everyone calls a tech ;) |
| 21:15.02 | Iamnacho | what do they do when you take vacation? |
| 21:15.08 | Katty | i uhh |
| 21:15.19 | Katty | haven't had one in awhile |
| 21:15.19 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 21:15.31 | Katty | i think our company might want someone to fly down here and hold my hand :P |
| 21:16.15 | Mercestes | http://pastebin.ca/490117 |
| 21:16.16 | Mercestes | please hlp |
| 21:16.44 | Mercestes | really? I'll come hold your hand, Katty |
| 21:17.06 | Mercestes | I'm not too professional tho, but I can consult. :) |
| 21:17.42 | kink0 | Mercestes, her company wants 10,000 $ for his hand, but if you get both, probably you would get a good discount |
| 21:17.59 | *** part/#asterisk nick125_ (n=upirc@atarack/staff/nick125) |
| 21:18.15 | *** join/#asterisk nick125_ (n=upirc@atarack/staff/nick125) |
| 21:18.21 | Mercestes | kink0, A full set always costs more than the individual pieces. |
| 21:19.15 | kink0 | yes, I guess, girls are like cars :) |
| 21:19.58 | kink0 | sorry for the jokes... I am really desesperate getting CAUSE codes 47 on my span's :( |
| 21:20.24 | Mercestes | they both crash after you drive them hard? |
| 21:20.28 | Strom_M | kink0: you're getting cause 47 when the other party hangs up? |
| 21:20.48 | Mercestes | I'm getting cause 69 as soon as I dial |
| 21:20.50 | kink0 | Strom_M, no..no just when the call is passed to the Zap channel |
| 21:21.00 | Strom_M | ok? |
| 21:21.14 | syzygyBSD | if I have a fax extension, will it catch faxes if I am in a Dial() command? |
| 21:21.34 | Mercestes | syzygyBSD, yes and no |
| 21:21.37 | Katty | Mercestes: can you be flown? |
| 21:21.40 | kink0 | syzygyBSD, no, but you can probably use Dial(.... G ) |
| 21:21.43 | Mercestes | syzygyBSD, actually. entirely no. |
| 21:21.54 | syzygyBSD | ya, that I was thinking |
| 21:22.10 | Mercestes | Katty, ground shipping is cheaper but if you want overnight air, then sure. |
| 21:22.14 | kink0 | if you use G, then the call gone to 2 priority once answered |
| 21:22.22 | kink0 | priority N and N+1 |
| 21:22.27 | Mercestes | Katty, I'm about $2000 for overnight air tho and only $450 shipped ground. |
| 21:22.38 | Mercestes | plus the money for the crate |
| 21:22.39 | Katty | k |
| 21:22.46 | kink0 | all you need later, is to detect fax, and if fax then go to fax , and if voice bind both legs |
| 21:22.59 | Katty | i think the last person they flew out here, to teach us some stuff (not asterisk related) they paid 20k |
| 21:23.00 | *** join/#asterisk tslunj (n=turbo@griffin.linux.hr) |
| 21:23.06 | Mercestes | for how long? |
| 21:23.09 | Qwell[] | Katty: brt |
| 21:23.09 | Katty | 2 days |
| 21:23.17 | Mercestes | lol |
| 21:23.20 | Qwell[] | Mercestes: You still have a trial to sign-up for |
| 21:23.21 | syzygyBSD | ya, problem is I am connecting to another asterisk server with fax detection... |
| 21:23.25 | Qwell[] | you can't go until that's done :P |
| 21:23.26 | Strom_M | kink0: 47 is "Resource unavailable, unspecified" |
| 21:23.31 | Strom_M | kink0: show me your dial line |
| 21:23.36 | Mercestes | Qwell[] My * server hosed a RAID controller. |
| 21:23.43 | Qwell[] | excuses, excuses |
| 21:23.47 | Mercestes | no really |
| 21:23.48 | Mercestes | :( |
| 21:23.49 | Qwell[] | Didn't you have an excuse last week too? :P |
| 21:23.55 | kink0 | Strom_M, yes, that is, I suspect due to memory leack in the telco side switch |
| 21:24.00 | Qwell[] | heh |
| 21:24.02 | *** part/#asterisk tslunj (n=turbo@griffin.linux.hr) |
| 21:24.03 | Strom_M | bullshit |
| 21:24.06 | Mercestes | I'm a bad person, I'm sorry |
| 21:24.12 | Strom_M | telco switches don't have that sort of problem |
| 21:24.15 | Strom_M | show me your dial line |
| 21:24.16 | Mercestes | as soon as I save my job....or...find a new one. |
| 21:24.22 | Mercestes | Katty, what do you need help with?? |
| 21:24.34 | Mercestes | seriously... |
| 21:24.37 | kink0 | Strom_M, let me know your originating IP |
| 21:24.55 | kink0 | ( due are behind fw ) |
| 21:24.56 | Strom_M | ....why? |
| 21:25.09 | Strom_M | i'm asking you to paste text at me |
| 21:25.28 | kink0 | ahhh sorry !! I was thinking you wanna to place a call :) |
| 21:25.29 | syzygyBSD | hmm, how does everyone else do fax detection when the call might be connected to an end device? |
| 21:25.52 | Mercestes | syzygyBSD, It's flakey at best. I suggest avoidance of exten => fax |
| 21:26.12 | kink0 | has no any extra, just Dial(Zap/r1/${EXTEN:2}) |
| 21:26.24 | syzygyBSD | Sadly I have no control over that business decision |
| 21:26.28 | Strom_M | why are you using r on your PRI? |
| 21:26.28 | kink0 | but I have tryed with ,,d ,,hHd and ,,tTd |
| 21:26.39 | Strom_M | try using G1 instead of r1 |
| 21:26.45 | kink0 | to give some time to the same channel recover time |
| 21:26.52 | kink0 | I had used also g and G, but the same |
| 21:26.52 | Strom_M | oh nonsense |
| 21:26.59 | Strom_M | you don't need to give the channel time to recover |
| 21:27.20 | kink0 | is, but I tried with all combinations |
| 21:27.31 | kink0 | the problem starts when over 2 CPS |
| 21:27.41 | kink0 | when one or less CPS, then gone fine. |
| 21:28.00 | Strom_M | CPS? |
| 21:28.06 | kink0 | calls per second |
| 21:28.07 | Qwell[] | call per second |
| 21:28.13 | Strom_M | ok |
| 21:28.35 | Strom_M | 2 cps on the circuit, or on the channel? |
| 21:28.53 | kink0 | 2 cps to the box, where there 4 PRI |
| 21:29.25 | Strom_M | ok, you're skipping steps |
| 21:29.26 | kink0 | and with r/R I see differents channels are choose, from different PRI |
| 21:29.36 | Strom_M | two calls per second out the PRI? |
| 21:29.55 | kink0 | yes, but 2 CPS to the 4 PRI |
| 21:30.24 | Strom_M | ok, maybe i'm not being specific |
| 21:30.24 | kink0 | because sometimes one channel is on the 1th PRI, and the next call is goin to the 2th PRI |
| 21:30.48 | kink0 | lets say, one call gone to Zap/23-1 , the next call to Zap/24-1 .. and so |
| 21:31.05 | kink0 | I have only one group in zapata.conf for the 4 PRI |
| 21:31.14 | Strom_M | does the problem occur when you're placing more than one call per second out the same PRI, or does the problem occur when you're placing more than one call per second in general? |
| 21:31.18 | *** join/#asterisk qdk (n=qdk@193.226.189.115) |
| 21:31.19 | kink0 | but I had tried also to separate in four groups, and the same |
| 21:31.51 | Mercestes | can someone help me with http://pastebin.ca/490117 please? |
| 21:31.59 | kink0 | Strom_M, is the same, X CPS to the Asterisk box is X CPS to the PRI |
| 21:32.13 | Mercestes | all my calls ring to congestion/all circuits busy. ....it's my own internal PRI. It can't be busy. :( |
| 21:32.26 | kink0 | Mercestes, codec problem ? |
| 21:32.46 | Strom_M | kink0: ok, but you just said you have four PRIs |
| 21:32.59 | kink0 | Strom_M, well, there 4 PRI per box |
| 21:33.13 | Strom_M | so I'm asking you if you've narrowed it down to problems on a single PRI, or whether it's a problem that seems to be unrelated to the number of PRIs in use |
| 21:33.28 | kink0 | is unrelated |
| 21:33.42 | kink0 | I tryed setting only one , two or the 3 PRI on the group |
| 21:34.13 | Strom_M | so if you place a call out PRI number one followed immediately by a call out PRI number two, the call out PRI number two gets rejected? |
| 21:34.29 | kink0 | both rejected with 47 |
| 21:34.41 | Strom_M | ok... |
| 21:34.47 | Strom_M | and the PRIs are idle at that point? |
| 21:34.57 | kink0 | when there low CPS, calls passed.. then goes about 50 or 70 channels in use in the group |
| 21:35.09 | kink0 | but here, when more calls arrives, then 47 |
| 21:35.26 | kink0 | yes !! there much PRI channels idle |
| 21:35.30 | Strom_M | call the telco and ask if they're rate-limiting your calls |
| 21:35.31 | Strom_M | which telco is it? |
| 21:35.34 | Mercestes | kink0, it's a pri to pri call. Direct pri. how can it be a codec problem? :( |
| 21:35.40 | kink0 | France Telecom |
| 21:35.46 | *** join/#asterisk thoughtpolice (n=austin@c75-111-139-133.plaicmtc01.tx.dh.suddenlink.net) |
| 21:36.21 | kink0 | Mercestes, then would be signalling or timing in your PRI, or may be you forgot enable ulaw/alaw |
| 21:36.41 | Mercestes | ... |
| 21:37.08 | Mercestes | .....does pri use ulaw/alaw? |
| 21:37.16 | kink0 | yes |
| 21:37.46 | Mercestes | well, these configs worked previously. I'm on my emergency server. I don't think it's a codec issue. |
| 21:37.50 | Mercestes | I can make sip calls |
| 21:37.54 | Mercestes | and pri calls on span 1 |
| 21:38.21 | Mercestes | could you maybe give me a definitive answer. |
| 21:38.23 | Mercestes | ? |
| 21:38.46 | kink0 | do you pass calls SIP->Zap in only one span ? |
| 21:39.21 | Mercestes | I go from Sip Phone to PRI 1, and I accept faxes from PRI 1 directly to PRI 2 which runs to a brooktrout faxboard in a fax server. |
| 21:39.36 | Mercestes | The transition from PRI 1 to PRI 2 is giving me congested. |
| 21:40.10 | kink0 | did you try pri intense debug span x ? |
| 21:42.04 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 21:42.30 | *** join/#asterisk tschafer (n=tschafer@207.241.143.246) |
| 21:44.11 | *** join/#asterisk kombi (n=kombi@213.160.14.18) |
| 21:44.57 | kombi | got asterisk installed and an AVM fritz card stuck in, what next? |
| 21:45.04 | Qwell[] | start it |
| 21:45.15 | kombi | how..? |
| 21:45.19 | Qwell[] | type 'asterisk' |
| 21:45.22 | Qwell[] | then, read... |
| 21:45.23 | Qwell[] | ~book |
| 21:45.28 | jbot | i heard book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 21:45.28 | Qwell[] | ~wikis |
| 21:45.36 | jbot | [wikis] http://www.voip-info.org |
| 21:45.36 | kombi | asterisk you mean, started it is.. |
| 21:46.58 | kombi | Qwell[]: actually, I spend all day reading and havn't really gotten anywhere, I understand I need to configure a channel, being the isdn card and tweaked modem conf, but so far to no avail. show channels still returns a sad empty list.. |
| 21:49.04 | kombi | how do I make sure the isdn card is really there? |
| 21:49.29 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 21:50.10 | *** join/#asterisk Braxus (n=braxus@66.147.214.164) |
| 21:50.23 | kombi | sigh.. |
| 21:50.51 | Mercestes | gah, how do I log output of the CLI to a file without the stupid color codes? |
| 21:52.45 | _VoiceMeUp_COM | hmm |
| 21:52.48 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 21:53.12 | _VoiceMeUp_COM | if i screen -x the asterisk process i get color.. else if i just start without i get none.. also .. note that color codes is mainly the EMULATION in the ssh client |
| 21:53.26 | _VoiceMeUp_COM | EX: LINUX full color VS ANSI or VT100 |
| 21:53.52 | _VoiceMeUp_COM | again maybe im wrong.. but its a pointer |
| 21:54.08 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 21:54.08 | *** mode/#asterisk [+o mog] by ChanServ |
| 21:54.13 | kombi | does "show channels" give information about channels that are configured or channels that are currently in use? |
| 21:54.37 | shido6 | in use |
| 21:54.46 | kombi | thanks shido6! |
| 21:55.08 | shido6 | whatya workin on? |
| 21:55.30 | kombi | shido6: trying to get the isdn card to work.. |
| 21:55.37 | shido6 | isdn not bri |
| 21:55.40 | shido6 | right? |
| 21:56.01 | shido6 | euro or northamerican ? :) |
| 21:56.13 | kombi | I was about to ask.. euro! |
| 21:56.31 | kombi | isdn then, right? |
| 21:56.44 | shido6 | what card are you using? |
| 21:56.53 | kombi | avm fritz 2.0 |
| 21:56.59 | Mercestes | _VoiceMeUp_COM, I get the color codes instead of the color when I > file and vim it. I want th ecolor codes themselves gone |
| 21:57.28 | *** join/#asterisk Downchuck (n=downchuc@c-24-22-20-80.hsd1.mn.comcast.net) |
| 21:57.39 | Downchuck | hallo |
| 21:57.46 | _VoiceMeUp_COM | yeah..but your ssh client has color emulation etc ? maybe theres a negociation between the shell and the terminal compat's |
| 21:58.00 | Downchuck | My asterisk/sip server is behind a firewall. I've opened up 5060 incoming; do I need to open others, for RTP? |
| 21:58.04 | _VoiceMeUp_COM | can you see if you can turn off color from the client then reconnect to ssh and then try ? |
| 21:58.22 | Downchuck | I'm also behind a NAT, which is making it difficult sometimes to test |
| 21:58.24 | shido6 | 10k-20k unless you set it in your phones to use something different UDP |
| 21:58.35 | Downchuck | the sip/asterisk is not, just a plain firewall, static external ip |
| 21:58.40 | shido6 | 10k-20k UDP is great start for RTP |
| 21:59.24 | kombi | shido6: card is good in dmesg, but I havn't found a way to see whether asterisk has found it yet.. |
| 21:59.40 | shido6 | *which* card? |
| 21:59.51 | kombi | shido6: avm fritz pci 2.0 |
| 22:00.19 | Mercestes | _VoiceMeUp_COM, then it just gives me color codes |
| 22:00.49 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 22:00.57 | gerphimum | if i want the ability to add a sip user who is on the internet (not local) do i just have to forward port 5060 or is there more to it than that |
| 22:01.11 | Downchuck | shido: 10k-20k open on the server firewall? |
| 22:01.15 | Downchuck | thx. |
| 22:01.26 | Downchuck | i saw 10 - 20k, thought you were talking kilobits :) |
| 22:01.38 | _VoiceMeUp_COM | weird |
| 22:01.56 | kombi | shido6: show channeltypes should mention something isdn, no? |
| 22:02.12 | shido6 | :) |
| 22:02.18 | shido6 | gimme a sec |
| 22:02.22 | kombi | sure.. |
| 22:02.28 | shido6 | trying to crack into my own box |
| 22:02.42 | kombi | shido6: forgot the word? |
| 22:03.27 | Downchuck | i've only got $10 of termination credits. i feel hackerproof. |
| 22:04.42 | shido6 | currently on a mac trying to use crossover to run the Dell IP KVM s/w |
| 22:06.51 | kombi | shido6: sounds complicated.. no ssh there? |
| 22:07.01 | shido6 | mmm... |
| 22:07.08 | _VoiceMeUp_COM | you could hack the source |
| 22:07.10 | shido6 | ssh is accessible AFTER the bios |
| 22:07.17 | kombi | true.. |
| 22:07.21 | shido6 | IP KVM allows us to see everything, even the reboot process |
| 22:07.30 | kombi | I see.. |
| 22:07.35 | shido6 | so you dont have to drive to the colo |
| 22:07.41 | shido6 | or call and pay the colo engineer guy |
| 22:07.49 | _VoiceMeUp_COM | Mercestes : i think its in term.c in asterisk source |
| 22:07.57 | _VoiceMeUp_COM | <PROTECTED> |
| 22:08.05 | _VoiceMeUp_COM | its looking for htis in your sh environement |
| 22:08.14 | _VoiceMeUp_COM | wahts your set env resutls |
| 22:08.24 | kombi | point taken, that's why I got all the boxes stashed in the basement.. |
| 22:08.35 | Mercestes | _VoiceMeUp_COM, thanks, I fixed it |
| 22:08.38 | _VoiceMeUp_COM | if (!option_console || option_nocolor || !option_nofork) |
| 22:08.39 | _VoiceMeUp_COM | btw |
| 22:08.47 | _VoiceMeUp_COM | so there an option no color.. i think it was -n |
| 22:08.57 | _VoiceMeUp_COM | glad it works |
| 22:09.09 | Corydon-w | export TERM=foo |
| 22:09.20 | kombi | ..with a fixed line |
| 22:09.37 | Mercestes | _VoiceMeUp_COM, %s/(!option_console || option_nocolor/1/g and then a recompile fixed it |
| 22:09.51 | Corydon-w | _VoiceMeUp_COM: those only have an effect when you START asterisk |
| 22:09.55 | _VoiceMeUp_COM | k |
| 22:10.02 | _VoiceMeUp_COM | true.. since -r is a new sessions right ? |
| 22:10.18 | Corydon-w | It's a remote connection |
| 22:10.39 | Corydon-w | to an already started asterisk |
| 22:11.47 | *** join/#asterisk Greek-Boy (n=Greek-Bo@196.45.144.42) |
| 22:12.53 | Mercestes | Corydon-w: you rock |
| 22:13.05 | Mercestes | I don't care what I said about you before. |
| 22:13.27 | Corydon-w | Uh huh |
| 22:13.47 | Mercestes | no, I mean it |
| 22:14.45 | *** join/#asterisk karlhaines_ (n=karl@unaffiliated/karlhaines) |
| 22:18.30 | *** join/#asterisk karlhaines (n=karl@unaffiliated/karlhaines) |
| 22:18.34 | aptura | voicemeup interesting site |
| 22:18.42 | Mercestes | Can someone please look at http://pastebin.ca/490237 ?? |
| 22:20.51 | *** join/#asterisk bill4242 (n=bill@66.60.191.200) |
| 22:22.14 | Mercestes | network congestion but I know there are no calls on this PRI because it's my pRI |
| 22:23.26 | bill4242 | ALL: I've just installed AsteriskNOW with a Digium TDM800P analog card.. It's not listing any of the analog FXS extension in the GUI, only the FXO ports. When I run "# dmesg | grep FXS" I show ports 1-4 :FAILED FXS (FCC) and Wildcard USB FXS Interface driver registered"... |
| 22:23.29 | Greek-Boy | what is a good voip carrier grade router? |
| 22:23.37 | Mercestes | cisco |
| 22:23.42 | Mercestes | or SER |
| 22:26.17 | Mercestes | no help for me? :( |
| 22:26.27 | johann8384 | Greek-Boy: I've heard Network Foundry makes some real nice stuff but I don't know much about it |
| 22:26.29 | bmd | bill4242: you forgot to plug in your card |
| 22:26.56 | bill4242 | bmd: .... |
| 22:27.00 | bmd | which begs the question, why not change the text 'FAILED' to 'UNPLUGGED' in zaptel... t'would save a lot of headaches |
| 22:27.15 | bill4242 | no its plugged in |
| 22:28.03 | *** join/#asterisk coolbeans (n=null@adsl-074-247-038-249.sip.bhm.bellsouth.net) |
| 22:28.04 | bill4242 | are you talking about the MOLEX connector? |
| 22:28.36 | coolbeans | Hey guys, what's the best PRI card for Asterisk 1.2? |
| 22:28.48 | coolbeans | No channelbank, just a PRI right into a server via X card. |
| 22:28.50 | Mercestes | the digium one |
| 22:28.55 | bill4242 | It detects the FXO ports, just not the FXS 4FXS 4FXO |
| 22:29.03 | coolbeans | Is there a specific model you recommend? |
| 22:29.12 | russellb | how many pri's do you need? |
| 22:29.27 | coolbeans | 1 for now, potentially up to 8 at somepoint. |
| 22:29.44 | russellb | well, then i'd get a quad-span |
| 22:29.55 | russellb | and then you can get a 2nd one when you come to it |
| 22:30.06 | russellb | then, you just have to decide if you want the hardware echo can or not |
| 22:30.09 | russellb | (i would seriously recommend it) |
| 22:30.20 | coolbeans | Cool. Which one, the 410? |
| 22:30.30 | high-rez | god i hate 'em |
| 22:30.39 | high-rez | erps |
| 22:30.42 | coolbeans | high-rez: What? |
| 22:30.47 | kombi | tell the fool, which of the many conf-files in /etc/asterisk are actually read? |
| 22:30.50 | high-rez | Wrong channels. Blackberries though. ;) |
| 22:30.52 | russellb | 405/410 or 407/412, depending on your PCI voltage. |
| 22:31.10 | russellb | kombi: all of them? |
| 22:31.41 | coolbeans | Ahh. Got it. |
| 22:31.52 | kombi | russellb: oh no, really? this confuses the shit out of me.. |
| 22:31.56 | coolbeans | Call quality up there with the Digium cards? |
| 22:33.40 | coolbeans | Here's the scenerio: We provided hosted PBX services to about 220 or so customers. We've been using termination via IP providers (les.net, vitelity, voicepulse, etc) and we were considering replacing them with a bank of PRI's but wanted to step into it. |
| 22:33.47 | *** part/#asterisk bill4242 (n=bill@66.60.191.200) |
| 22:34.17 | kombi | which command shows me the card is there and working? |
| 22:36.20 | Hymie | russellb: hardware echo cancellation? |
| 22:36.32 | Hymie | russellb: I have a 4 port card from digium, is that an extra module or what? |
| 22:37.01 | russellb | yes |
| 22:37.27 | Hymie | russellb: can you point me in the direction that I require pointing in? ;) |
| 22:37.40 | Hymie | russellb: I'd never heard of this before |
| 22:38.35 | *** join/#asterisk karlhaines (n=karl@unaffiliated/karlhaines) |
| 22:39.46 | *** join/#asterisk scurb (n=scurb@c-25aae355.14-16-64736c13.cust.bredbandsbolaget.se) |
| 22:40.01 | hansin321 | kombi: I think you can actually omit some of the config files and it will still work, but I don't know which ones. I am reather new at this, but once I did a 'make samples' and then just moved all them into a subdirectory. I copied over only one at a time what I thought I needed. You just are not going to need to config files for features you are not using, or so that is what I should think would be the case... |
| 22:41.04 | JT | Hymie: what are you scratching your head about? |
| 22:41.19 | kombi | thanks hansin321! |
| 22:41.36 | Hymie | JT: russel said there was now a hardware cancellation device from digium, then vanished when I became curious |
| 22:41.43 | hansin321 | but I am not skilled enough to tell you what is what. but extensions.conf is the big one, the 'master' config file. |
| 22:42.02 | JT | Hymie: there's been cards with hardware echo cancellation for ages, they cost more |
| 22:42.08 | JT | but are generally worth it |
| 22:42.23 | Hymie | JT: yes, but not on the digium site ... |
| 22:42.53 | kombi | hansin321: would you know of any way to see that asterisk actually saw the isdn card? |
| 22:43.06 | JT | Hymie: there's everywhere on the digium site |
| 22:43.12 | JT | they're |
| 22:43.31 | Hymie | JT: I didn't say they weren't |
| 22:43.37 | *** join/#asterisk bkw_ (i=brian@adsl-70-143-39-207.dsl.tul2ok.sbcglobal.net) |
| 22:44.08 | JT | Hymie: you just said not on the digium site |
| 22:44.30 | Hymie | no, you said " there's been cards with hardware echo cancellation for ages, they cost more" and I said "not on the digium site" |
| 22:44.51 | Hymie | there were not such beasts on the digium site years ago |
| 22:45.02 | JT | what's not on the digium site then? ;) |
| 22:45.07 | JT | maybe years ago :P |
| 22:45.08 | hansin321 | kombi: I always have trouble with that stuff. I have an el cheapo single port FXO knock-off card. And I get it to work, but I never remember what the heck I did to get it to work, so I don't think I would be much help. |
| 22:45.28 | Strom_M | really? there've been T1 cards with hardware echo cancellation on the digium site for years :) |
| 22:45.41 | Hymie | JT: not maybe, there simply weren't. I haven't looked at the purchasing a card for several years |
| 22:46.11 | JT | several years = a decade in IT |
| 22:46.24 | Hymie | Strom_M: well, when I bought the cards I have, they only had the demo card and hadn't even released the four port modular card they have now |
| 22:46.38 | kombi | hansin321: lol.., sounds very much like me.. thanks anyway! How did you see it was working though? |
| 22:46.52 | Hymie | Strom_M: and when I bought my most recent card, they had just released the 4 port modular card |
| 22:47.15 | Hymie | anyhow, we'd have to debate what "ages" means in IT, and really.. ages is rather arbitrary |
| 22:47.35 | *** join/#asterisk boch (n=fran@190.48.206.133) |
| 22:48.45 | JT | kombi: you need bristuff or misdn to use isdn |
| 22:48.50 | JT | isdn BRI that is |
| 22:49.06 | JT | Greek-Boy: what do you mean by carrier grade router? |
| 22:49.11 | kombi | JT: think I got that.. |
| 22:49.15 | aptura | cisco |
| 22:49.18 | JT | pfft |
| 22:49.22 | aptura | :) |
| 22:49.26 | JT | cisco isn't close to carrier grade |
| 22:49.30 | aptura | heheh |
| 22:49.40 | aptura | Then what is ? nortel and |
| 22:50.06 | JT | ericsson and nokia and alcatel |
| 22:50.12 | JT | and all the other telco manufacturers |
| 22:50.24 | aptura | we have a big nokia r&D facility here |
| 22:50.44 | hansin321 | kombi: well, for my card I can do a 'dmesg | grep -i wildcard' and see if the kernel recognizes it. That is the first step for the card I am using, then after that I can't remember. I hope to take better notes this time around (I have just compiled a fresh install, but haven't done anything yet). |
| 22:50.54 | Hymie | aptura: take a tour.. and .. ."borrow" something on a test bench ;Þ |
| 22:51.11 | hansin321 | kombi: maybe something like 'show channels' or ??? |
| 22:51.15 | JT | kombi: what have you got? |
| 22:52.18 | kombi | passive bri card, looks good in dmesg, but no idea whether asteris sees it |
| 22:52.21 | aptura | Hymie, hears the security there is unreal. |
| 22:52.33 | JT | kombi: what driver is loaded? |
| 22:52.39 | Downchuck | any idea what setting is causing " modprobe: Can't locate module sound-slot-0 " to pop up in my /var/log/messages |
| 22:52.41 | kombi | hiSAX |
| 22:52.46 | JT | eww |
| 22:52.55 | JT | that is old isdn4linux shit |
| 22:53.01 | JT | you must prevent that crap from loading |
| 22:53.02 | kombi | ture |
| 22:53.05 | kombi | true.. |
| 22:53.31 | kombi | JT: couldn't get the proprietary driver to compile.. |
| 22:53.40 | JT | what card is it? |
| 22:53.46 | kombi | avm fritz pci |
| 22:53.55 | JT | i see |
| 22:54.10 | JT | you should see if it works with bristuff or misdn |
| 22:54.22 | kombi | how do I tell? |
| 22:54.29 | hansin321 | Is there a way to call 'help' from the CLI that will only display one page at a time (like 'shoe | less')? I am connected via SSH & screen and I can't scroll up. I assume the Asterisk console is ncurses based; are there specific keys that allow me to scroll up and down pages? Thanks. |
| 22:54.49 | kombi | (because bristuff is there) |
| 22:54.51 | killfill | hi |
| 22:55.05 | kombi | hansin321: have you tried shift + pageUp? |
| 22:55.23 | killfill | when i do ztcfg, i see in var-log-messages this: copyin failed Registered tone zone 0 () |
| 22:55.34 | killfill | what could that mean?.. i see red alter. |
| 22:55.46 | killfill | the line is ok, becouse when i plug in a phone, it sounds ok.. |
| 22:55.49 | *** join/#asterisk bhrobinson (n=brobinso@northtx1-static.telwestonline.com) |
| 22:56.08 | bhrobinson | question on Cisco 7940. How can I get into the phone to configure it? |
| 22:56.46 | Strom_M | you....don't. |
| 22:56.49 | Strom_M | you edit config files |
| 22:57.09 | karlhaines | bhrobinson: RTFM |
| 22:57.16 | bhrobinson | Strom_M: that is what I thought... but I do not have a TFTP server here |
| 22:57.34 | aptura | install one |
| 22:58.20 | bhrobinson | what if the phones will be at an offsite. I cannot seem to unlock the phone to set the tftp server |
| 22:58.29 | Strom_M | apt-get install atftpd |
| 22:58.40 | Strom_M | bhrobinson: look online and find instructions |
| 22:58.43 | Nugget | you can configure a lot on a 7940 from the phone panel. |
| 22:58.54 | Nugget | assuming you have set (and know) the config password |
| 22:58.58 | hansin321 | kombi: Yeah, like I would at a normal linux command prompt. But it doesn't work. It just shows me some stuff that was "behind" the ncurses * stuff. There has to be a way. Most ncurses program AFAIK implement some way of doing this. |
| 22:59.07 | bhrobinson | Nugget: how do I unlock it? I tried to do the **# to no avail |
| 22:59.18 | Nugget | the only way I know is to set it using the tftp configuration first. |
| 22:59.47 | Nugget | if you plan to use cisco phones you should just consider tftp to be unavoidable. the phones will be an unmanageable hemorrhoid without one. |
| 23:00.14 | bhrobinson | lol... sounds good there :) I need to set up one, but have tried to avoid it so far. |
| 23:00.31 | Nugget | it's the path of least resistance by quite a comfortable margin. |
| 23:00.59 | bhrobinson | and set the option 66 in DHCP to assign TFTP? |
| 23:01.34 | Nugget | yeah |
| 23:02.13 | hansin321 | kombi: It appears to be due to me running screen and SSH. Probably nothing to duw with *. |
| 23:02.21 | hansin321 | due that is ;) |
| 23:03.06 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 23:03.06 | *** join/#asterisk marcan (i=1337@198.Red-83-54-248.dynamicIP.rima-tde.net) |
| 23:07.53 | *** join/#asterisk dotSlashW (n=HTP@200.80.197.5) |
| 23:08.36 | dotSlashW | hello, I need some help configuring a SIP trunk from * to a ShoreTel system |
| 23:11.29 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-72-65-27-9.bflony.east.verizon.net) |
| 23:13.17 | SuPrSluG | need help w/ polcom phone. when I nmap it all I get is |
| 23:13.25 | SuPrSluG | All 1663 scanned ports on 192.168.0.25 are: filtered |
| 23:13.27 | SuPrSluG | MAC Address: 00:17:3F:1F:70:6E (Unknown) |
| 23:14.22 | SuPrSluG | nameserver and http aren't showing. can I reboot and flash thru tftp server? |
| 23:15.13 | *** join/#asterisk SwK (n=SwK@user-69-73-37-99.knology.net) |
| 23:16.49 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 23:20.50 | Qwell[] | hansin321: do* |
| 23:21.48 | irule | what does this mean? "ZT_CHANCONFIG failed on channel 1: Inappropriate ioctl for device (25)" |
| 23:27.13 | *** join/#asterisk marcan (i=1337@198.Red-83-54-248.dynamicIP.rima-tde.net) |
| 23:28.37 | *** join/#asterisk crochat (n=crochat@84-74-150-141.dclient.hispeed.ch) |
| 23:32.56 | Hymie | http://www.metro.co.uk/media/viral.html?in_page_id=55&in_mediaext_item_id=4759 |
| 23:35.55 | *** join/#asterisk snuffy22 (n=na@61.29.30.137) |
| 23:53.31 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 23:58.38 | Downchuck | I just can't get sound going :-( |
| 23:59.02 | Downchuck | i can call people up pretending to be the boss.. but i can't say anything |
| 23:59.11 | Downchuck | maybe after dinner. |
| 23:59.20 | JT | a little more info maybe? |
| 23:59.48 | Downchuck | no sound card, so i may have a misconfiguration there.. I've astrisk and SIP working, registred and dialing to phone lines |