| 00:00.32 | *** join/#asterisk xpot (n=jim@c-71-195-241-115.hsd1.ma.comcast.net) |
| 00:01.17 | *** join/#asterisk RabidW (n=RabidW@adsl-75-28-149-104.dsl.wlfrct.sbcglobal.net) |
| 00:07.05 | galeras | Any idea why Disa is having troubles recognizing dtmf from some kind of phones? |
| 00:14.35 | Yourname` | Ok, fine new question. |
| 00:16.07 | Yourname` | One box transfers a call via sip to another box. Except the other box is not working out properly. And the other box is not connecting to my ITSP. Because it doesn't need to, it needs to simply handle inbound sip calls. But it's not taking it. |
| 00:16.15 | Yourname` | If someone is listening, I'd like to go into details. |
| 00:16.31 | Yourname` | I'm pretty sure it's something reaaaal small. |
| 00:16.47 | Yourname` | Spent four hours trying to get it straight too. :( |
| 00:17.33 | *** join/#asterisk NirS_ (n=Nir@87.68.144.61) |
| 00:17.39 | JT | details are pretty important |
| 00:18.05 | *** join/#asterisk freecity (n=will@adsl-75-4-100-156.dsl.emhril.sbcglobal.net) |
| 00:18.12 | Yourname` | Thanks, JT. |
| 00:18.21 | *** join/#asterisk cygar (n=cygar@200.26.191.3) |
| 00:18.31 | fujin | could anyone take a look at http://rafb.net/p/qQVnmm92.html |
| 00:18.36 | fujin | seem to be having some issues |
| 00:18.37 | Yourname` | Basically, this was setup by someone else, and now he's fired.. so I'm trying to do this myself by looking into it further. |
| 00:18.39 | fujin | creating folders/files |
| 00:19.00 | fujin | check your sip.conf on both servers Yourname` |
| 00:19.26 | Yourname` | First detail is, a call goes to extension 1012 on the inbound box, where agents from 1000-to 1010 are logged in. And when the calls are transferred to the inbound box, all these agents are supposed to ring. |
| 00:19.32 | Yourname` | fujin, checked and checked. |
| 00:19.49 | Yourname` | Let me give you the details on the inbound box n pastebin, JT. |
| 00:20.42 | coldsteal | umm i dont have ulaw i dont think |
| 00:20.58 | coldsteal | <PROTECTED> |
| 00:21.16 | coldsteal | how do i get the ulaw codec |
| 00:21.25 | fujin | it should have one |
| 00:21.28 | fujin | it should have ulaw built in |
| 00:21.30 | fujin | rather |
| 00:21.46 | coldsteal | lol well i dont think i do |
| 00:21.48 | fujin | have you got allow=ulaw disallow=all in sip.conf? |
| 00:22.45 | coldsteal | no |
| 00:23.09 | Yourname` | JT: sip.conf http://pastebin.ca/index.php |
| 00:23.13 | JT | coldsteal: you will have ulaw unless something went terribly wrong |
| 00:23.16 | JT | Yourname`: wrong url. |
| 00:23.56 | Yourname` | Damm, lol |
| 00:24.02 | Yourname` | extensions.conf http://pastebin.ca/639982 |
| 00:24.14 | Yourname` | sip.conf http://pastebin.ca/639979 |
| 00:24.20 | coldsteal | well i got reid of allow=ulaw disallow=all and its still doesnt work |
| 00:25.23 | coldsteal | what order should i put allow=ulaw disallow=all |
| 00:25.25 | Yourname` | extensions_astguiclient.conf http://pastebin.ca/639985 |
| 00:26.12 | Yourname` | extensions_broadast.conf http://pastebin.ca/639987 |
| 00:26.23 | Yourname` | That's all the confs on the inbound box. |
| 00:26.38 | Yourname` | 1012@boxip address is what the call is being sent to. |
| 00:27.40 | Yourname` | If you want I can msg all those pastebins to you. JT. |
| 00:27.41 | Hmmhesays | destination field is completely usesless in cdrs if you dial from an s extension |
| 00:27.56 | coldsteal | look |
| 00:27.57 | coldsteal | disallow=all |
| 00:28.00 | coldsteal | [Jul 30 19:27:39] NOTICE[18826]: chan_sip.c:5256 process_sdp: No compatible codecs, not accepting this offer! |
| 00:28.06 | coldsteal | idk what happened |
| 00:28.09 | Yourname` | PS: All this setup on the inbound box is by him. |
| 00:28.17 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 00:28.24 | Hmmhesays | heh, disallow=all doesn't all any codecs |
| 00:29.01 | *** join/#asterisk remmo (n=junk@203.62.147.8) |
| 00:29.19 | JT | sysreq: but it has allow=ulaw |
| 00:29.21 | Hmmhesays | icky thump bitches |
| 00:29.37 | coldsteal | umm okay now im not getting any NOTICEs |
| 00:31.13 | Yourname` | So weird. |
| 00:32.26 | *** join/#asterisk ming_zy1 (i=ming_zym@nat/yahoo/x-340c8f26ece6844c) |
| 00:33.07 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 00:33.18 | *** part/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 00:33.36 | Yourname` | Let me also get you a debug. |
| 00:33.42 | codebuster | how do i in * assign that to a viaiable ie if there is a value there say assign it to $cfwd ? exten => s,2,GotoIf($["${DB(Callforward/master)}" != ""]?7)/ |
| 00:36.34 | Yourname` | Anything funny in there JT? |
| 00:36.37 | Yourname` | Stil collecting the debug. |
| 00:37.18 | coldsteal | this is what im having problems with http://rafb.net/p/1FhXHv70.html |
| 00:38.20 | Yourname` | Here you go. http://pastebin.ca/639997 |
| 00:38.23 | Yourname` | SIP debug. |
| 00:38.51 | coldsteal | what was changed? |
| 00:39.02 | wothinn | Is there a way to zap a global variable out of existence from within the dialplan? I built a big (for me) dialplan application and need to do some garbage collecting. |
| 00:39.55 | Yourname` | Is anyone around at all? |
| 00:40.14 | coldsteal | Yourname`: was that link for me? |
| 00:40.20 | Yourname` | NO dude. |
| 00:41.02 | coldsteal | lol u sead Here you go |
| 00:41.37 | coldsteal | its confused me |
| 00:43.11 | JT | english please |
| 00:43.24 | coldsteal | *it confused me |
| 00:43.42 | JT | lol u sead |
| 00:43.47 | coldsteal | o |
| 00:43.53 | coldsteal | lol im bad at spelling |
| 00:45.02 | Yourname` | lol |
| 00:45.20 | Yourname` | JT? |
| 00:46.33 | JT | Yourname`: what? |
| 00:46.36 | *** join/#asterisk remmo (n=junk@203.62.147.8) |
| 00:46.52 | Yourname` | Just wondering if you're looking at those pastebins, and willing to help me out. |
| 00:48.35 | JT | Yourname`: that system is a mess of AGI and dialplans |
| 00:48.42 | JT | i have no idea what it's trying to do |
| 00:48.48 | coldsteal | okay this has my error and sip.conf http://rafb.net/p/8QUqGS59.html |
| 00:48.50 | JT | hell, i have no idea what your error is |
| 00:49.24 | Yourname` | haha, me neither. I'm just tryng to accept calls on 1012 and make sure that all agents logged in get the ring and anyone of them answer the call. |
| 00:50.11 | wothinn | Apparently I've learned the system enough that I'm asking questions people can't answer... mixed blessing. :-) |
| 00:53.52 | Yourname` | [TK Defender? |
| 00:54.02 | Yourname` | Or Strom/ |
| 00:54.20 | Yourname` | These guys are always here, and today they ain't. :( |
| 00:54.22 | fujin | I think I know the feeling, wothinn |
| 00:54.30 | fujin | or people are ignoring/annoyed :) |
| 00:55.50 | JT | Yourname`: what about them? |
| 00:55.59 | fujin | Anyone here setup priority queues which work based on the callerID of the caller? |
| 00:56.16 | fujin | I can't seem to find any examples of people doing it. The contractor who did the runners on us said it could be done, but the code he left doesn't work |
| 00:56.40 | JT | i'm sure it could be done, if you setup the queues right |
| 00:57.27 | fujin | yeah |
| 00:57.31 | Yourname` | JT: They are helpers, who are usually around. And when they ask a question reciprocating my questions, it usually means they're on my case and are about to offer help which makes it worth waiting. :) |
| 00:57.36 | fujin | I was thinking maybe a macro? and then pass the callerid variable to it |
| 00:57.42 | fujin | but not sure how to match it inside of the macro |
| 00:57.43 | Yourname` | And they usually help me, and other needy people. |
| 00:57.50 | Yourname` | Or people in need, lol |
| 00:58.11 | JT | Yourname`: you won't find much sympathy from people if you keep requesting specific people, you'll also piss those people off |
| 00:58.26 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 00:58.55 | Yourname` | Am I pissing you off? |
| 00:59.01 | JT | i absolutely hate it when someone harrasses me about some problem when i have not been already working on it, because i helped them weeks ago, or they saw me help someone else, it makes me what to not help them at all |
| 00:59.19 | JT | s/what/want/ |
| 01:00.09 | Yourname` | I'm not harassing you. I'm just trying to gather if you're really going to be helping me out and I should be anticipating it or not, that's all. |
| 01:00.24 | Yourname` | I'm not here often to see you helping others. |
| 01:00.34 | JT | i'm just saying don't harrass [tk]-defender or strom either |
| 01:00.42 | JT | if someone can help they can |
| 01:00.46 | Yourname` | Don't get me wrong, defender and strom have their own ways of helping others.. but the least they do is point to the right direction. |
| 01:00.47 | JT | if not they can't |
| 01:00.57 | Yourname` | I'm not harassing them if they ain't here, now am I? :) |
| 01:01.38 | JT | well you were having a whinge at the least then |
| 01:01.41 | Yourname` | You know what, it looks like I'm pissing you off because I'm clearly looking for someone else other than you. So, I'm going to drop this conversation. If you don't wanna help me out, that's cool. |
| 01:01.47 | JT | you haven't even bothered to specify what your problem is yet |
| 01:01.58 | JT | no-one can help you until you say what the problem is |
| 01:02.20 | JT | ~Question |
| 01:02.21 | jbot | [question] If you have a question and want people to give useful answers, make sure you have read this first: http://www.catb.org/~esr/faqs/smart-questions.html |
| 01:02.21 | Yourname` | You really need to pay more attention then, mate. |
| 01:02.29 | JT | nope |
| 01:02.43 | Yourname` | Whenever anyone has questions in this channel, they are requested to provide conf pastes in pastebin. |
| 01:02.51 | JT | you just said you were trying to get something working in sip |
| 01:02.55 | Yourname` | I've done that, and even gave a little brief summary of what is going on. |
| 01:03.00 | JT | yes with a DETAILED description of the setup and the PROBLEM |
| 01:03.20 | Yourname` | Yourname`>One box transfers a call via sip to another box. Except the other box is not working out properly. And the other box is not connecting to my ITSP. Because it doesn't need to, it needs to simply handle inbound sip calls. But it's not taking it. |
| 01:03.39 | Yourname` | Did you need a question mark in the above para to make it look like a question? |
| 01:03.48 | Yourname` | If so, then you're right... I did not ask a question. |
| 01:03.52 | JT | no, i'd like it to be clear |
| 01:04.00 | JT | "not working" doesn't help us |
| 01:04.09 | JT | what behaviour did it exhbit |
| 01:04.12 | JT | what did you try |
| 01:04.21 | Yourname` | Exactly. And when "not working" doesn't help you, my pastebins usually help everyone else out. |
| 01:04.44 | *** join/#asterisk nighty^ (n=nighty@p2136-adsau16honb13-acca.tokyo.ocn.ne.jp) |
| 01:04.44 | JT | Yourname`: what errors did you receive |
| 01:04.47 | JT | what action did you take |
| 01:04.53 | JT | you used what to make a call |
| 01:05.10 | Yourname` | Nevermind dude. I'm going to drop it and wait for some other time to ask this question, because you're clearly having a bad day. :) |
| 01:05.15 | Yourname` | Happens to the best of us. |
| 01:05.23 | JT | no |
| 01:05.42 | JT | don't assume things that are not there |
| 01:05.51 | JT | you must decribe exactly what you did |
| 01:05.54 | JT | and what error you got |
| 01:06.04 | JT | that gave you the "not working" impression |
| 01:06.24 | Yourname` | No, I'm not assuming. I never do. |
| 01:06.32 | JT | because you're clearly having a |
| 01:06.32 | JT | <PROTECTED> |
| 01:06.40 | JT | ^assumption |
| 01:06.42 | *** join/#asterisk techie (n=gus@routedsystems.com) |
| 01:07.04 | Yourname` | This behavior of mine is because of how I always came in, and said what I usually do.. and everyone needed pastebins to back it up. All the errors and stuff, they usually see in the sip debug. |
| 01:07.08 | Yourname` | Which was also provided. |
| 01:07.17 | *** join/#asterisk hohum (n=dcorbe@dhcp64-134-231-170.shs.nyc.wayport.net) |
| 01:07.18 | Yourname` | Ok, sorry wrong assumption. |
| 01:07.46 | JT | the stuff in pastebins only tell us one dimension of the problem |
| 01:08.11 | JT | they don't say "attempted to call using a <blah> sip phone to call <x number> etc" |
| 01:09.41 | Yourname` | How come I never really had to mention things like that before? |
| 01:09.45 | *** join/#asterisk _DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 01:10.05 | JT | i have no idea |
| 01:10.20 | JT | but stop being silly and arguing about what may or may not have happened in the past |
| 01:10.34 | JT | and tell us what steps were taken, and what problem was encountered |
| 01:11.13 | Yourname` | I have no idea. It was setup by someone else, and now he's gone. |
| 01:11.20 | Yourname` | So, I'm trying to pick up from pieces here. |
| 01:11.30 | fujin | <PROTECTED> |
| 01:11.32 | fujin | learn2asterisk |
| 01:11.38 | fujin | probly the best way to fix your problem (honestly) |
| 01:12.06 | JT | no-one can help you if you keep falling back to the line that it was set up by someone else, and move on, no more weasel words |
| 01:12.07 | Yourname` | That's what I'm doing on the side, fujin, lol. Currently working on another box too. |
| 01:12.12 | Yourname` | But I still want to figure this thing out. |
| 01:12.22 | JT | surely you must be using a phone or something? |
| 01:12.25 | Yourname` | haha ok, JT. |
| 01:12.31 | JT | can you at least tell us about that |
| 01:13.31 | Yourname` | I'm using an asterisk dialer. It plays a message to a person, and if the person presses 1.. the call will be accepted there and then transferred to the inbound box on an extension 1012, via sip. |
| 01:14.13 | JT | i can see why you're trying to hide what you're doing so much now :P |
| 01:14.57 | fujin | anyone have any ideas how I'd do the callerId based queue priority joining? |
| 01:15.02 | fujin | having a brainfart |
| 01:15.25 | JT | there's quite a few ways to check callerid |
| 01:15.30 | sevard | pfffffffffffffffffft |
| 01:15.50 | Yourname` | I'm not hiding anything tho, lol |
| 01:16.05 | fujin | JT: but matching it against a list of 10000~ numbers is probably stupid, right? |
| 01:16.10 | JT | Yourname`: oh, i thought you would've known most people hate telemarkeing diallers :P |
| 01:16.13 | fujin | I can see that putting massive load on Asterisk |
| 01:16.34 | JT | fujin: some form of database query would be the sanest for that application |
| 01:16.40 | Yourname` | Oh, haha, no this isn't telemarketing. :) |
| 01:16.42 | Yourname` | Trust me. |
| 01:16.48 | fujin | right, using AGI or similar? |
| 01:17.03 | JT | fujin: you can embed queries in the dialplan too |
| 01:17.30 | fujin | oh, didn't know that |
| 01:17.39 | JT | ODBC |
| 01:19.49 | fujin | is DBget/DBput bad? |
| 01:19.52 | fujin | as opposed to odbc |
| 01:20.07 | JT | they'd work i guess |
| 01:20.12 | JT | haven't had to use them much |
| 01:20.19 | fujin | I see |
| 01:20.21 | fujin | thanks |
| 01:20.23 | fujin | will take a looksy |
| 01:21.29 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.149) |
| 01:23.00 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 01:27.12 | galeras | someone with alcatel and asterisk integrated? |
| 01:36.36 | Yourname` | ~nat |
| 01:36.36 | jbot | somebody said nat was Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 01:36.43 | Yourname` | ~sip-nat |
| 01:36.49 | Yourname` | ~sipnat |
| 01:36.49 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 01:38.00 | *** join/#asterisk Strom_C (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 01:41.29 | *** join/#asterisk ManxPower (n=manxpowe@71-8-56-17.dhcp.leds.al.charter.com) |
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| 01:51.06 | *** part/#asterisk galeras (n=galeras@201.245.103.169) |
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| 02:01.35 | *** join/#asterisk kiwoneka (n=kiwoneka@KTNRON06-1168103823.sdsl.bell.ca) |
| 02:02.52 | kiwoneka | good evening to all |
| 02:03.34 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 02:03.35 | kiwoneka | may someone point me to a good resource and how to of setting up h323 using polycom ip601 |
| 02:04.08 | mcab | kiwoneka: I don't believe the 601 supports H323 |
| 02:04.16 | JT | lol h.323 |
| 02:04.22 | kiwoneka | just the 501 |
| 02:04.24 | JT | you like to joke ;) |
| 02:04.40 | kiwoneka | ? |
| 02:05.08 | mcab | kiwoneka: actually, I'm not even sure about that. you may have to dig up a 500... |
| 02:05.24 | JT | kiwoneka: why not use sip? |
| 02:05.41 | kiwoneka | oh i already am |
| 02:05.55 | *** join/#asterisk ukris (n=ukris@aa20060807547d355914.userreverse.dion.ne.jp) |
| 02:05.58 | kiwoneka | it just i am trying to connect it to an avaya system |
| 02:06.41 | kiwoneka | it is a piece of sh** ip406 |
| 02:06.56 | JT | what has this got to do with asterisk? |
| 02:07.59 | kiwoneka | i am seeking advice on how to setup an asterisk box using h.323, so i know what is involved |
| 02:08.04 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 02:08.07 | kiwoneka | a good learning experience |
| 02:08.23 | JT | good way to drive yourself crazy |
| 02:08.27 | kiwoneka | i am not out to waste anyone's time |
| 02:08.28 | kiwoneka | lol |
| 02:08.36 | JT | easier to setup a yate box as a H.323 to SIP translator |
| 02:08.51 | kiwoneka | yate? |
| 02:09.11 | kiwoneka | tell me more |
| 02:09.18 | JT | off topic for this channel |
| 02:09.21 | kiwoneka | point me to resources |
| 02:09.25 | kiwoneka | ok |
| 02:09.25 | JT | you can easily find it with google.com |
| 02:09.27 | kiwoneka | thanks |
| 02:10.20 | *** join/#asterisk hijacked (i=90ut@cerebus.clandestineresearch.com) |
| 02:12.29 | kiwoneka | JT, I thank you that is exactly what i was/am looking for |
| 02:13.20 | *** join/#asterisk ManxPower (n=manxpowe@71-8-56-17.dhcp.leds.al.charter.com) |
| 02:13.42 | *** join/#asterisk stridernzl (n=neville@222-155-185-55.jetstream.xtra.co.nz) |
| 02:18.51 | *** join/#asterisk [T]ank (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 02:21.27 | [T]ank | i am periodically dropping calls. This is the output on the CLI when it happens. Where can I begin to look for the cause? |
| 02:21.28 | [T]ank | http://pastebin.ca/640097 |
| 02:21.46 | [T]ank | the calls are coming in from another server via IAX2 |
| 02:22.01 | [T]ank | sometimes they work just fine, sometimes they die after about 30 seconds. |
| 02:27.30 | ManxPower | [T]ank: set notransfer=yes in iax.conf. No this does NOT impact the ability of USER generated transfers, only the IAX2 equiv of a SIP REINVITE |
| 02:28.08 | [T]ank | so does that allow it to do reinvites? |
| 02:28.30 | ManxPower | it disables reinvites, which are prolly your problem |
| 02:29.17 | ManxPower | -- Channel 'IAX2/agent1-4' unable to transfer |
| 02:29.17 | [T]ank | ok... cuz i did have canreinvite=no set |
| 02:29.28 | ManxPower | there is no such option for iax.conf |
| 02:29.38 | ManxPower | unless they changed the name in 1.4 |
| 02:30.04 | ManxPower | there is no such thing as a "reinvite" in IAX, since there is no such thing as an "invite" in IAX |
| 02:30.11 | ManxPower | hence the different name of the option |
| 02:30.44 | [T]ank | ok... trying it, thanks for the clarification. |
| 02:31.17 | ManxPower | check /path/to/src/asterisk/configs/iax.conf.sample to make sure the option is still listed and not changed name. |
| 02:31.39 | ManxPower | BTW, Asterisk does NOT complain when you have invalid options in the config files. |
| 02:32.55 | [T]ank | 'notransfer' is deprecated in favor of 'transfer' which has options 'yes', 'no', and 'mediaonly' |
| 02:33.24 | [T]ank | so i want transfer=no, right? |
| 02:36.16 | [T]ank | hmmm... still dumping my calls |
| 02:36.54 | [T]ank | but now i am not even getting errors... it is just hanging up and saying that it did. |
| 02:37.18 | [T]ank | <PROTECTED> |
| 02:37.18 | [T]ank | <PROTECTED> |
| 02:39.54 | ManxPower | [T]ank: I guess it is time to do an iax2 debug and wade thru the massive amounts of information |
| 02:40.04 | ManxPower | but don't ask me to help. *shiver* |
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| 02:44.03 | *** part/#asterisk techie (n=gus@routedsystems.com) |
| 02:49.04 | stridernzl | if i get this: Executing DBget("Zap/3-1", "cfwd=Callforward/master") in new stack |
| 02:49.04 | stridernzl | Jul 31 15:45:00 WARNING[5272]: app_db.c:226 get_exec: This application has been deprecated, please use the ${DB(family/key)} function instead. |
| 02:49.20 | stridernzl | does that mean it worked .. but was not so happy or didn't work ? |
| 02:49.36 | ManxPower | stridernzl: you did not read UPGRADE.txt, that is what it means. |
| 02:50.05 | ManxPower | yes it worked, no it will not work in the next major version of Asterisk |
| 02:50.10 | stridernzl | Manxpower: 500 readme files I was bound to miss one :) |
| 02:51.12 | stridernzl | manxpower: that then turns to this: xecuting Dial("Zap/3-1", "Zap/g1/$cfwd||t") in new stack |
| 02:51.12 | stridernzl | <PROTECTED> |
| 02:52.31 | stridernzl | but then it just hangs up .... :( is it related to the fact i have used the first bit wrong ? |
| 02:52.40 | Juggie | $cfwd isnt a variable |
| 02:52.47 | Juggie | ${cfwd} would be |
| 02:53.25 | stridernzl | juggie: oh so i just need to wrap it in {} ? |
| 02:53.29 | Juggie | yes |
| 02:54.00 | stridernzl | juggie: cheers .. I'll go try that .. thanks ... only taken me 2 days to get that far ... you could be a life saver! |
| 02:56.05 | Juggie | your welcome |
| 02:56.18 | *** part/#asterisk svm_invictvs (n=patrick@unaffiliated/svminvictvs/x-938456) |
| 02:57.14 | [T]ank | ManxPower: lol... i rebooted the server. It may have fixed it. ;-) |
| 03:02.21 | *** join/#asterisk ptiggerdine (n=ptiggerd@203-219-14-182.static.tpgi.com.au) |
| 03:02.27 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:04.06 | *** join/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 03:08.41 | ManxPower | cool |
| 03:10.21 | *** join/#asterisk NoIdea465 (n=NoIdea46@d53-64-116-212.nap.wideopenwest.com) |
| 03:11.15 | *** part/#asterisk workaphobia (n=workapho@ool-44c30ab1.dyn.optonline.net) |
| 03:21.59 | *** join/#asterisk gardo (n=gardo@121.97.195.87) |
| 03:25.46 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 03:28.39 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 03:33.13 | fujin | what formats can Playback() handle? |
| 03:33.15 | *** join/#asterisk Trevor_b (n=tbenson@69.12.220.201) |
| 03:35.16 | ManxPower | fujin: "show formats" |
| 03:35.34 | fujin | no such command |
| 03:36.15 | ManxPower | fujin: go read UPGRADE.txt to see what the 1.4 version of that command is. |
| 03:39.51 | fujin | that'd be useless wouldn't it? |
| 03:39.55 | fujin | being that I'm runnign 1.2 and everything. |
| 03:40.17 | fujin | I couldn't ge ta file to play, but it was because I was specify the extension of the file |
| 03:40.23 | fujin | instead of like /path/to/file i had /path/to/file.wav |
| 03:40.25 | fujin | =fail |
| 03:41.37 | ManxPower | "show application playback" should have mentioned you don't need the file extension |
| 03:41.51 | fujin | wow, does too |
| 03:41.54 | fujin | didn't know about show application |
| 03:42.06 | ManxPower | You have a long path ahead of you, grasshopper. |
| 03:43.07 | ManxPower | NEVER look on the Wiki or anywhere else for information on specific asterisk applications. |
| 03:43.22 | ManxPower | they are always out of date, and sometime not accurate. Show application is the first place you look. |
| 03:43.37 | fujin | sweet |
| 03:43.41 | fujin | thanks dude |
| 03:45.20 | ManxPower | "show applications" for a list of all of them,. |
| 03:45.26 | ManxPower | "help" is a good command too., |
| 03:46.47 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 03:47.57 | tengulre | hi,all |
| 03:48.02 | tengulre | anybody alive? |
| 03:50.31 | tengulre | Oh,,Come on! |
| 03:51.15 | NoIdea465 | hey |
| 03:51.24 | *** join/#asterisk DaveCanoe (n=Dave@belbrrcnas08-1088684068.dial.bell.ca) |
| 03:52.09 | *** join/#asterisk JT (n=jon@unaffiliated/jt) |
| 03:52.30 | NoIdea465 | I had to factory reset my 7970 and its now asking for jar70sip.8-0-3-34.sbn |
| 03:52.44 | *** join/#asterisk bmg505 (n=leon@196.209.177.87) |
| 03:52.49 | NoIdea465 | but jar70sip.8-0-3-33.sbn is what was included in the file |
| 03:54.00 | NoIdea465 | a search for jar70sip.8-0-3-34 gives me nothing... |
| 03:56.52 | snuff-work | it should ask u only what you specify in the cnf.xml |
| 03:57.14 | snuff-work | besides cisco have been known to make errors naming files |
| 03:59.56 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 04:00.40 | NoIdea465 | TFTP logs don't show it touching any of my xml files |
| 04:01.03 | NoIdea465 | it goes straight to the default.loads file |
| 04:04.28 | NoIdea465 | The screen is also compleatly dark with the line buttons cycling |
| 04:05.31 | NoIdea465 | headset light is green, mute turns red after it downloads a file. Then it restarts and the loop continues.. |
| 04:05.59 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 04:06.21 | NoIdea465 | All files listed in the default.loads file do not match the files that ere in the archive |
| 04:06.39 | NoIdea465 | Changing the names does not help |
| 04:10.04 | NoIdea465 | Hmmm.... |
| 04:10.06 | NoIdea465 | I'm about to hang myself with this cat5 |
| 04:11.05 | fujin | k |
| 04:11.11 | fujin | Let me know how that goes. |
| 04:15.22 | *** join/#asterisk gammah (n=gammah@cpe-70-113-89-51.austin.res.rr.com) |
| 04:17.06 | *** join/#asterisk ptiggerdine_ (n=ptiggerd@203-206-171-23.perm.iinet.net.au) |
| 04:17.25 | *** join/#asterisk adeeln (n=adeel@c-67-161-185-121.hsd1.ca.comcast.net) |
| 04:17.42 | adeeln | quick question....does anyone know if a2billing supports asterisk realtime? or is it static? |
| 04:21.40 | Juggie | that would be a question for a2billing support |
| 04:22.00 | *** join/#asterisk Primer (n=vi@sh.nu) |
| 04:22.35 | Primer | Anyone using skinny? All I get is a dial tone now. I was using chan_sccp in 1.2 just fine, but that's unmaintaned now |
| 04:25.15 | *** join/#asterisk ReDNeQ- (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 04:26.02 | *** join/#asterisk ReDNeQ (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 04:27.23 | *** part/#asterisk adeeln (n=adeel@c-67-161-185-121.hsd1.ca.comcast.net) |
| 04:28.23 | *** join/#asterisk iBuMp- (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 04:29.01 | *** part/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 04:32.05 | *** join/#asterisk iBuMp (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 04:36.08 | Primer | and now it just works...wtf |
| 04:36.10 | *** part/#asterisk Primer (n=vi@sh.nu) |
| 04:37.23 | NoIdea465 | *wish my 7970 would just work |
| 04:43.13 | *** join/#asterisk froek (n=digitals@S01060001023c40c6.gv.shawcable.net) |
| 04:43.26 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:43.28 | froek | flenders yo |
| 04:44.04 | froek | flenders I solved my dropped calls problem! |
| 04:45.26 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-82-81-220-132.cablep.bezeqint.net) |
| 04:46.46 | SwK | anyone have altigen's interconnected w/ asterisk over PRI? |
| 04:47.30 | flenders | froek: hey |
| 04:47.47 | flenders | froek: what was the problem? |
| 04:47.51 | flenders | double nat? |
| 04:49.02 | froek | oh hey, yeah.. well double nat had something to do with it, but the main problem was two NIC's in the asterisk box were sharing IRQs |
| 04:49.03 | SplasPood | Anyone happen to know what settings I'd need (compared to your usual US line) to configure a "POTS" line in mainland china? Using a TDM400P |
| 04:49.48 | froek | and as a result, it was dropping sip packets too often |
| 04:50.01 | froek | well, just not getting them properly or something |
| 04:50.55 | froek | i haven't had one dropped call yet! |
| 04:50.59 | froek | <knocks on wood> |
| 04:51.02 | flenders | great news |
| 04:51.24 | froek | yeah! so happy :D |
| 04:51.36 | froek | now i can concentrate on feature development for the asterisk box :D |
| 04:52.41 | *** join/#asterisk mtaht4 (n=m@host99.72.248.107.conversent.net) |
| 04:59.39 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 05:04.18 | leif_madsen | y0 |
| 05:04.36 | Juggie | word |
| 05:04.42 | leif_madsen | what is up my brother |
| 05:04.52 | Juggie | i will only speak to blitzrage |
| 05:05.40 | Juggie | noo |
| 05:05.40 | blitzgare | hi! |
| 05:05.48 | Juggie | close but not close my friend :) |
| 05:06.07 | blitzragie | oh wait... maybe this is what I WANT |
| 05:06.15 | blitzragie | ugh... I had to spend $75 to get home tonight |
| 05:06.22 | Juggie | hahahahahahahahahhaahhhahaahaha |
| 05:06.23 | blitzragie | I totally missed my last train back from volleyball |
| 05:06.48 | leif_madsen | I was hitting on a drunk girl... that was my problem |
| 05:07.03 | leif_madsen | then she got REALLY drunk, and I got bored with her |
| 05:07.21 | Juggie | hahah |
| 05:07.25 | Juggie | was she hot |
| 05:07.39 | leif_madsen | she had a nice rack and a cute face |
| 05:07.45 | Juggie | sounds good :) |
| 05:07.51 | leif_madsen | actually ya, she's pretty cute |
| 05:08.10 | tengulre | SplasPood: Where are u? China? |
| 05:08.25 | Juggie | leif_madsen, then you should have stuck with it :) |
| 05:08.36 | tengulre | a vote video: http://v.youku.com/v_show/id_cj00XMTM5MDU2.html |
| 05:08.40 | leif_madsen | first time I've really talked to her. She was like... "hey! do you know who I am?"... and I'm like... "ummm... no... I think your name is Meghan right?", and we went from there :) |
| 05:08.56 | leif_madsen | Juggie: nah... she was all over the place man... couldn't keep her attention after too much beer |
| 05:08.57 | Juggie | did you get a number |
| 05:09.22 | leif_madsen | nah, I honestly wasn't trying that hard. She got super drunk and I wasn't too interesting in talking to her anymore |
| 05:09.26 | leif_madsen | I didn't drink enough |
| 05:09.29 | Juggie | silly boy |
| 05:09.31 | leif_madsen | meh |
| 05:09.40 | leif_madsen | seriously... she was outta control... it was useless |
| 05:09.50 | Juggie | reminds me of my old roomate vicki |
| 05:09.51 | leif_madsen | I'll try in 2 weeks when I play next |
| 05:10.00 | Juggie | she made a mess of herself @ the wedding friday |
| 05:10.00 | leif_madsen | I played awesome tonight though |
| 05:10.03 | leif_madsen | lol |
| 05:10.46 | leif_madsen | I'm supposed to go to a wedding on the 18th of Aug., but that is the same weekend as my volleyball tournament, and I'm not sure I really even want to go to this wedding |
| 05:10.46 | leif_madsen | I hate weddings |
| 05:10.46 | leif_madsen | as I soon realized when I went to one this summer |
| 05:10.51 | leif_madsen | religious ceremonies make me uncomfortable |
| 05:10.57 | Juggie | i have 3 in 1.5 weeks |
| 05:11.06 | leif_madsen | and I've been to plenty of church :) |
| 05:11.40 | Juggie | i just stand in church and smile |
| 05:11.42 | Juggie | i dont do the prayers etc... |
| 05:11.57 | leif_madsen | same |
| 05:12.16 | Juggie | i'm definitally athiest :) |
| 05:12.36 | leif_madsen | I don't believe in organized religion anymore (actually... I pretty haven't since grade 6 when I started finding contradictions in the bible... but I was lucky... I went to a Catholic school, so I could study it :)) |
| 05:12.41 | leif_madsen | I'm not atheist anymore |
| 05:12.57 | leif_madsen | just for fun, read Conversations with God by Neil Donald Walsh |
| 05:13.11 | Juggie | i am rc as well. |
| 05:13.13 | leif_madsen | even if you don't actually believe how he wrote it, it's still really good advice |
| 05:13.23 | Juggie | (roman catholic) |
| 05:13.26 | leif_madsen | I'm not RC... just went to an RC school |
| 05:13.30 | Juggie | ah |
| 05:13.37 | leif_madsen | technically I'm presbyterian |
| 05:13.49 | Juggie | your not atheist anymore? |
| 05:13.51 | leif_madsen | well, I was when I was a kid |
| 05:14.18 | leif_madsen | nah, not atheist anymore |
| 05:14.33 | Juggie | now? |
| 05:14.33 | leif_madsen | that CWG book is pretty enlightening... and I'm a stubborn bastard when it comes to my convictions |
| 05:14.46 | Juggie | you believe in god again? :P |
| 05:14.49 | leif_madsen | non-religious... I would consider myself "spiritual",.... whatever that means |
| 05:14.54 | leif_madsen | I believe I am god |
| 05:14.58 | Juggie | hahah |
| 05:15.00 | leif_madsen | I believe we are all god |
| 05:15.04 | Juggie | i believe theres something out there |
| 05:15.12 | Juggie | but it isnt someone standing on the clouds |
| 05:15.16 | leif_madsen | nope |
| 05:15.20 | Juggie | none the less this isnt really a good conversation for asterisk |
| 05:15.24 | flenders | hangon, I AM GOD! |
| 05:15.31 | leif_madsen | I read a lot of philosophy and cosmology too |
| 05:15.42 | leif_madsen | flenders: funny thing, is I believe you are :) |
| 05:15.52 | *** join/#asterisk A-jay (n=quirc@62.217.245.97) |
| 05:15.54 | flenders | you are right, son |
| 05:15.59 | leif_madsen | we are all god, simply experiencing ourselves in the physical form |
| 05:16.25 | flenders | oh my god |
| 05:16.27 | leif_madsen | but I'm not trying to push any of my convictions on anyone... everyone is free to believe as they wish |
| 05:16.29 | flenders | I mean, oh your god |
| 05:16.29 | leif_madsen | it makes no difference |
| 05:16.33 | leif_madsen | you're* |
| 05:16.33 | flenders | I mean, oh YOU! |
| 05:16.40 | leif_madsen | OH GOD OH GOD OH GOD |
| 05:16.46 | leif_madsen | that's what I make the girls scream |
| 05:16.47 | leif_madsen | :D |
| 05:17.03 | leif_madsen | see? I have proof even |
| 05:17.09 | *** join/#asterisk MdeP (n=mdep@155-83-22-190.adsl.tie.cl) |
| 05:17.26 | flenders | are they all drunk when they scream that? |
| 05:17.36 | flenders | :D |
| 05:17.37 | leif_madsen | not all the time |
| 05:17.50 | leif_madsen | sometimes sober |
| 05:18.05 | leif_madsen | I'm not an ugly computer nerd thank god ;) |
| 05:18.20 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 05:18.20 | *** mode/#asterisk [+o denon] by ChanServ |
| 05:18.42 | leif_madsen | tooth decay... hehe |
| 05:19.19 | denon | :) |
| 05:20.18 | SplasPood | tengulre: nah, in NYC.. which makes debugging this a bit more... painful |
| 05:23.31 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 05:26.00 | tengulre | :( |
| 05:27.22 | SplasPood | that, combined with the fact that I have NO idea what settings I should be using if anything |
| 05:28.23 | leif_madsen | china? only thing you should really need to change, on analog, would be the defaultzone and loadzone (if I remember the correct parameters) |
| 05:29.01 | tengulre | CN for it |
| 05:29.50 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 05:30.03 | leif_madsen | yep... zonedata.c says 'cn' |
| 05:30.04 | SplasPood | leif_madsen: hrm, got those |
| 05:30.06 | tengulre | loadzone=cn, defaultzone=cn ... |
| 05:30.16 | leif_madsen | that should be all you really need |
| 05:30.27 | leif_madsen | other than fxsks=1 and fxoks=2 |
| 05:30.40 | leif_madsen | (assuming those modules on those ports... obviously) |
| 05:30.52 | tengulre | I m in china, and use asterisk for owner company PBX. very COOL! |
| 05:31.08 | SplasPood | hrm.. |
| 05:31.16 | leif_madsen | is VoIP legal / usable there? |
| 05:31.26 | SplasPood | we have a small development office there |
| 05:31.29 | tengulre | It has one asterisk card (4 FXO+4FXS) in that box. |
| 05:31.30 | leif_madsen | does China have the crazy bandwidth infrastructure Japan has? |
| 05:31.37 | SplasPood | decided to use asterisk as a "pbx" tied to the analog line |
| 05:31.38 | SwK | china has voip |
| 05:31.40 | tengulre | FXS to connect old PBX |
| 05:31.48 | leif_madsen | SwK: a lot... or a little? |
| 05:31.52 | SwK | loads of people use skype in china |
| 05:31.53 | leif_madsen | in relation to N.A. |
| 05:31.54 | tengulre | a lot |
| 05:32.11 | SplasPood | fxsks=3-4 |
| 05:32.11 | SplasPood | defaultzone=cn |
| 05:32.11 | SplasPood | loadzone=cn |
| 05:32.20 | tengulre | very big VoIP company in china. |
| 05:32.24 | tengulre | avaya. |
| 05:32.28 | *** join/#asterisk jarod14 (n=jarod14@mail.viatelecom.com) |
| 05:32.31 | tengulre | cisco |
| 05:32.32 | leif_madsen | they are big in Europe too |
| 05:32.36 | tengulre | huawei. |
| 05:32.38 | tengulre | 3com |
| 05:32.42 | tengulre | <PROTECTED> |
| 05:32.46 | SwK | avaya != voip company... avaya is a pain in the ass PBX vendor |
| 05:32.57 | leif_madsen | right... but I'm curious about Asterisk based solutions |
| 05:33.30 | *** join/#asterisk obnauticus (n=obnautic@c-71-236-219-178.hsd1.wa.comcast.net) |
| 05:33.58 | tengulre | asterisk is simple. only suit small company. like owners. |
| 05:34.16 | SplasPood | i wouldn't say that... |
| 05:34.18 | tengulre | too simple |
| 05:34.23 | bmg505 | ? |
| 05:34.28 | SwK | you would be suprised the places asterisk is deployed then tengulre |
| 05:34.36 | SplasPood | tengulre: what would you say it lacks? |
| 05:34.47 | tengulre | sometimes it will halt. |
| 05:34.50 | tengulre | ;( |
| 05:34.53 | CunningPike | tengulre: It's not too simple for the municipal government I work for |
| 05:35.05 | CunningPike | Oh Jeez |
| 05:35.05 | tengulre | COOL. |
| 05:35.07 | bmg505 | <tengulre> too simple <----------- eeish u obviously never got into custom dialplans |
| 05:35.09 | SplasPood | tengulre: well thats been rare for me |
| 05:35.14 | CunningPike | I crashed it, therefore it's crap |
| 05:35.24 | CunningPike | Good night |
| 05:35.28 | SplasPood | tengulre: but yes, I'd say thats a major drawback of asterisk compared to some of the more "established" PBX vendors |
| 05:35.31 | leif_madsen | tengulre: I never get lockups on my Asterisk boxes... and even if one did, I have a cluster of them ;) |
| 05:35.34 | SwK | SplasPood, I would say asterisk's 2 biggest lacking features are scalability and fault tollerance |
| 05:35.35 | SplasPood | s/vendors/systems |
| 05:35.41 | SplasPood | SwK: agreed |
| 05:35.47 | leif_madsen | SwK: I would agree with that |
| 05:35.58 | leif_madsen | I've made it (mostly) work... but hold crap what a pain in the butt |
| 05:35.59 | SwK | i dont thing anyone can disagree with it ;) |
| 05:36.02 | SplasPood | SwK: stability has gotten much better, but there's still a lot of scary lurking in that code I wager |
| 05:36.11 | leif_madsen | SwK: not anyone who really uses it |
| 05:36.13 | tengulre | sorry, I only point call center function in it. ; |
| 05:36.17 | SwK | hah I wont take that bet SplasPood |
| 05:36.25 | SwK | tengulre, even then it doesnt scale very well |
| 05:36.42 | SwK | tengulre, I work on carrier equipment and broadcasting stuff |
| 05:36.48 | leif_madsen | SplasPood: actually... those who test and know how to test and report bugs often don't have those problems because the developers can be very responsive if you give them the information they need. |
| 05:36.48 | SplasPood | I recently gave freeswitch and callweaver a spin |
| 05:36.56 | SplasPood | freeswitch seemed interesting, but... undocumented |
| 05:37.06 | tengulre | I know you are very like asterisk. but I have to say, it's too professional . |
| 05:37.07 | SplasPood | callweaver was well... .. yea |
| 05:37.07 | leif_madsen | normal in open source telephony... unfortunately |
| 05:37.16 | SwK | tengulre, when i say i work on carrier gear I should clarify i do a ton of work for a carrier equipment manufacturer |
| 05:37.16 | tengulre | can not suit many people. |
| 05:37.19 | leif_madsen | tengulre: what does that really mean? |
| 05:37.26 | leif_madsen | "too professional" ? |
| 05:37.42 | SwK | SplasPood, freeswitch is documented sorta... its got the RTFSC documentation heh |
| 05:37.45 | leif_madsen | just like Remote Access BBS system isn't good because it's too professional? :) |
| 05:38.00 | tengulre | config is hard. funtion is simple. |
| 05:38.04 | SplasPood | SwK: heh |
| 05:38.06 | tengulre | that's my mean. |
| 05:38.13 | leif_madsen | I will agree the configuration has a learning curve |
| 05:38.19 | SplasPood | leif_madsen: Always preferred MBBS myself |
| 05:38.28 | SwK | leif_madsen, you have a copy of Remote Access laying around? |
| 05:38.32 | leif_madsen | but once you *understand* how things work in Asterisk, most of it makes sense |
| 05:38.34 | SplasPood | I likely do |
| 05:38.35 | leif_madsen | SwK: I wish |
| 05:38.44 | leif_madsen | and I'm not that old :) |
| 05:38.48 | SwK | SplasPood, i have a copy og wg2.0 around here somewhere |
| 05:38.57 | SwK | ahve 3.0 too |
| 05:39.01 | leif_madsen | I tried every system... RA made the most sense |
| 05:39.04 | SwK | (that I actually paid for) |
| 05:39.09 | *** part/#asterisk ming_zy1 (i=ming_zym@nat/yahoo/x-340c8f26ece6844c) |
| 05:39.11 | leif_madsen | wg? |
| 05:39.19 | wg_ | what? |
| 05:39.23 | SwK | worldgroup... the renamed upgrade of mbbs |
| 05:39.25 | SplasPood | SwK: I've got em all, and I recently found a copy of the 1.0 WG Unix release! |
| 05:39.26 | leif_madsen | wg_: sorry... not you :) |
| 05:39.28 | wg_ | oh |
| 05:39.31 | wg_ | stop using my initials you bastards ;P |
| 05:39.34 | SwK | SplasPood, kool |
| 05:39.37 | SwK | hah |
| 05:39.38 | leif_madsen | wg_ |
| 05:39.44 | SwK | wg_ :P |
| 05:39.57 | wg_ | nah |
| 05:40.00 | wg_ | just lights up a number |
| 05:40.02 | leif_madsen | that sucks |
| 05:40.07 | wg_ | 6i'll ignore it now ;) |
| 05:40.14 | leif_madsen | oej's makes a crazy submarine ping sound |
| 05:40.23 | jarod14 | hi guys ! |
| 05:40.31 | leif_madsen | hoi |
| 05:41.38 | lmadsen | ugh... |
| 05:41.47 | lmadsen | oops... nevermind, it worked, heh |
| 05:42.42 | tengulre | the manage api is rubbish . |
| 05:42.46 | SwK | lmadsen, arent you supped to be blitzrage? |
| 05:43.17 | lmadsen | SwK: am I? I don't know... I'm trying something new :) |
| 05:43.34 | SwK | oh |
| 05:43.35 | lmadsen | I wish I could be leif.madsen |
| 05:43.41 | SwK | ok |
| 05:43.43 | lmadsen | underscore sucks |
| 05:44.06 | lmadsen | someone too 'leif' |
| 05:44.10 | lmadsen | s/too/took/ |
| 05:44.14 | SplasPood | tengulre: haven't played with it in 1.4 much, but I'd agree there |
| 05:46.31 | tengulre | SplasPood: I developt a manager app with asterisk manage api, look here! http://voip.zrtek.com/products.php?id=19 |
| 05:48.16 | jarod14 | I'm trying to find a way to match an extension of unknown length but ending with a # . there is my current code in extensions.conf : http://pastebin.fr/125 . I would like to replace the several pattern by only one. but I do not know how to wait for the # DTMF character indefinetely. I tried _X.# but it failed. Any idea ? |
| 05:48.36 | flenders | tengulre: why not in english? |
| 05:49.01 | SwK | english is over rated |
| 05:49.08 | tengulre | flenders: it have english version. do you like ,I sent it to u! |
| 05:49.11 | *** join/#asterisk ptiggerdine (n=ptiggerd@203-219-14-182.static.tpgi.com.au) |
| 05:49.15 | tengulre | s/have/has |
| 05:49.21 | flenders | I would like to have a look, yes |
| 05:49.22 | *** join/#asterisk ReDNeQ (n=ibump@cpe-66-68-37-190.austin.res.rr.com) |
| 05:49.27 | SwK | tengulre, does it have an engrish version? |
| 05:49.36 | tengulre | SwK: yes! |
| 05:49.40 | SwK | PERFECT! |
| 05:50.05 | tengulre | I only a little english. I want upload to sf.net. |
| 05:50.28 | flenders | you're a little english man? |
| 05:50.33 | flenders | :D |
| 05:50.35 | tengulre | anybody want it, tell me , I send it to you. |
| 05:50.40 | tengulre | ;( |
| 05:50.52 | tengulre | I known. |
| 05:50.55 | tengulre | only a little. |
| 05:51.21 | SwK | tengulre, C#? |
| 05:51.46 | tengulre | SwK: No, VC6.0++ |
| 05:52.00 | *** join/#asterisk ptiggerdine_ (n=ptiggerd@203-219-14-182.static.tpgi.com.au) |
| 05:52.05 | tengulre | so I think the manage api is too simple. ;( |
| 05:52.19 | *** join/#asterisk Penggu (i=foobar@220-245-200-87.static.tpgi.com.au) |
| 05:52.44 | *** join/#asterisk ming_zy1 (i=ming_zym@nat/yahoo/x-f7c15e34f3ef0a7a) |
| 05:52.54 | Penggu | hi all. ive got a snom320 and a GN-Netcom GN 6210 BT headset... anyone know if calls can be picked up on headset itself? |
| 05:53.37 | flenders | tengulre: do you have screenshots and/or a page in english? |
| 05:53.41 | flenders | or just the app? |
| 05:55.07 | tengulre | flenders: pls wait a monment, I upload screenshots now. |
| 05:55.44 | flenders | thanks |
| 05:57.37 | *** join/#asterisk metfan2007 (n=metfan20@189.180.215.129) |
| 05:58.04 | metfan2007 | hi all!!! Is there any way to know the channel used by a given extension using AMI??? |
| 06:06.05 | tengulre | OK |
| 06:06.29 | tengulre | http://voip.zrtek.com/temp/001.JPG ... 005.JPG |
| 06:06.37 | tengulre | notice: JPG not jpg. |
| 06:07.48 | flenders | interesting |
| 06:07.56 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 06:09.30 | tengulre | application is http://voip.zrtek.com/temp/asteriskmanager.rar |
| 06:10.32 | flenders | is that chinese? |
| 06:10.45 | metfan2007 | any clue getting channel data using only an extension with AMI??? pls! :) |
| 06:11.02 | tengulre | no. |
| 06:11.06 | tengulre | english version. |
| 06:11.19 | flenders | the other one... was it chinese? |
| 06:11.58 | tengulre | metfan2007: yes. |
| 06:12.28 | metfan2007 | tengulre: how? |
| 06:12.32 | tengulre | flenders: yes. it has two versions. |
| 06:13.05 | tengulre | metfan2007: split strings. |
| 06:13.48 | metfan2007 | tengulre: what command do I use in AMI, I only have the extension dialed, and I want to know witch channel is using.. |
| 06:14.30 | tengulre | call 'show ' command. and got the response string then ... |
| 06:15.02 | metfan2007 | tengulre: show channels??? |
| 06:15.13 | tengulre | if it has bug, pls email me . email: tengulre@163.com subject: About AsteriskManager. |
| 06:15.19 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 06:15.43 | *** join/#asterisk ptblank (n=MURDER1@cpe-75-84-221-219.socal.res.rr.com) |
| 06:15.51 | tengulre | metfan2007 : what kind of your channel. |
| 06:16.12 | metfan2007 | tengulre: this is for a high volume traffic callcenter.... I need something more particular, I think using "show channels" for every call and parsing all the text will take more asterisk resources |
| 06:16.14 | tengulre | pls , http://voip-info.org. search it. |
| 06:16.32 | metfan2007 | tengulre: zap channels |
| 06:16.56 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 06:17.19 | tengulre | metfan2007: zap channels not more api , so you can not got more status about zap channels. |
| 06:18.31 | Penggu | speaking about zap channels... anyone know of an implemntation where outgoing callers can be queued to wait for a free zap line? |
| 06:19.39 | metfan2007 | tengulre: this is for an external hangup web application, so I only know the number dialed, but for a HangUp command in AMI I need to know first the channel that is using... |
| 06:20.52 | *** join/#asterisk obnauticus (n=obnautic@c-71-236-219-178.hsd1.wa.comcast.net) |
| 06:21.36 | tengulre | metfan2007: if it have new call, asterisk will give the zap new name like, zap/1-20003, so you must got this name then can hangup it. |
| 06:24.13 | metfan2007 | tengulre: I tried that, but I have a problem, this is a high volume call center, so, if the customer hangsup before agent the zap channel gets free, and other call can take that channel, so If the agent try to hangup the channel after that, he hangups a channel that is used by another call |
| 06:24.17 | *** join/#asterisk saftsack (n=oliver@p54A7C4BD.dip.t-dialin.net) |
| 06:26.54 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 06:28.50 | tengulre | metfan2007: usually agent has not access to hangup a call in call center. your problem I cannot solve it. |
| 06:29.24 | tengulre | Penggu: your problem is my problem too. |
| 06:32.13 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 06:33.26 | metfan2007 | tengulre: Ok, this is another point of view, but need almost the same data. The same agent has in the DB stored an extension number for a client if it has an IVR. The agent doesn't has access to the phone key pad, so he click a button tu send DTMF tones via AGI with SendDTMF() app. The same, he does not knows the channel, only the number dialed. :) |
| 06:34.06 | tengulre | haha... |
| 06:38.45 | metfan2007 | what? |
| 06:41.49 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 06:48.49 | *** join/#asterisk saftsack (n=oliver@p54A7D730.dip.t-dialin.net) |
| 06:54.20 | tengulre | how to known the zap line is null(no line), I have 4 zap channles in a group g1, but when channel 1 has not line. |
| 06:57.52 | *** join/#asterisk menil (n=meni@bzq-179-153-168.static.bezeqint.net) |
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| 07:01.27 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 07:02.05 | badcfe | hello. is there information loss in transcoding between alaw and ulaw? |
| 07:02.32 | JT_ | it would be close to negligible |
| 07:02.46 | badcfe | JT_: and does it consume cpu? |
| 07:03.23 | JT_ | close to negligible |
| 07:03.26 | badcfe | JT_: about a few percent cpu consumation compared to 711/729 no? |
| 07:03.26 | JT_ | show translation |
| 07:06.17 | badcfe | JT_: thanks. |
| 07:08.50 | *** join/#asterisk saftsack (n=oliver@p54A7D60E.dip.t-dialin.net) |
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| 07:18.24 | *** join/#asterisk phsdshft (n=nkoenig@204.56.88.94) |
| 07:19.27 | phsdshft | Hello.. The reason code for a disconnect for a channel (defined in include/channel.h) is an integer... but I do not see where it is specified what integer is for what disconnect reason. |
| 07:19.34 | phsdshft | Where is that specified (if at all) |
| 07:21.01 | phsdshft | i.e. if the reason is '3' or '1' where is it defined (in comments somewhere I would assume) what those mean.. i.e. busy signal, etc. |
| 07:22.22 | *** join/#asterisk JT (n=j@unaffiliated/jt) |
| 07:23.56 | *** join/#asterisk JT (n=j@unaffiliated/jt) |
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| 07:36.09 | *** join/#asterisk waiter (i=waiter@2001:6f8:333:bbbb:bbbb:bbbb:bbbb:bbbb) |
| 07:36.29 | waiter | hi |
| 07:37.03 | codey | is ist possible to do a dial on the cli on a dummy-device via channel Zap/4 or Zap/5? |
| 07:37.26 | *** join/#asterisk matsk (n=mk@194.68.102.171) |
| 07:40.19 | *** join/#asterisk bhrobinson (n=Flagg732@northtx1-static.telwestonline.com) |
| 07:40.23 | waiter | I have a problem with a configuration sip. In the log, one says to me that it is the password, but I am sure of my passwords. |
| 07:40.25 | waiter | WARNING[21043]: chan_sip.c:9813 handle_response_invite: Forbidden - wrong password on authentication for INVITE to '"Dect Central" <sip:065******@voip.belgacom.be>;tag=as024aa321' |
| 07:40.25 | waiter | SIP/Belgacom_065$$$$$$_out-0815e870 is circuit-busy |
| 07:40.25 | waiter | == Everyone is busy/congested at this time (1:0/1/0) |
| 07:42.40 | bhrobinson | is there anyone here that can help with a PRI card? |
| 07:43.58 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
| 07:47.40 | phsdshft | hmmm only people with questions, no one with answers are awake right now :) |
| 07:47.59 | bhrobinson | pretty much |
| 07:48.31 | bhrobinson | you got answers? |
| 07:48.33 | bhrobinson | :) |
| 07:48.39 | waiter | :D |
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| 07:58.09 | *** join/#asterisk geniesis (n=geniesis@c220-239-144-224.eburwd1.vic.optusnet.com.au) |
| 07:58.19 | geniesis | why would i be getting power alarms? |
| 07:58.51 | geniesis | everytime i get a power alarm, the fxs port stops working (no dialtone) and i have to unload and reload the zaptel modules |
| 07:59.13 | *** join/#asterisk af_ (n=getsmart@81-174-47-169.dynamic.ngi.it) |
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| 08:06.03 | tzafrir_laptop | geniesis, what do you mean by "power alarm"? |
| 08:06.16 | tzafrir_laptop | what hardware? |
| 08:07.30 | geniesis | zaptel TDM844B |
| 08:07.39 | geniesis | Kernel message is "Power alarm on module 2, resetting!" |
| 08:07.40 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 08:08.00 | geniesis | module 2 being the FXS port |
| 08:08.06 | tzafrir_laptop | hmmm.... no idea. contact digium support? |
| 08:09.10 | tzafrir_laptop | maybe a hardware/software issue with the power supply |
| 08:09.53 | geniesis | the only problem i have is that i am running asterisk and the digium card on a non-idealized platform |
| 08:11.06 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-76-175-64-30.socal.res.rr.com) |
| 08:11.11 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 08:11.44 | geniesis | i have Trixbox 2.2beta 3 which uses asterisk 1.4 with zaptel 1.2.19; Its physically running on a Cisco Content Engine (CE-507) , which is essentially a P3 800Mhz with 512mb ram with intel nics and north/south bridge and scsi HD on a custom cisco made motherboard |
| 08:12.42 | *** join/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 08:13.12 | creativx | trixxxxxboxxxxxx |
| 08:14.08 | bhrobinson | I would be willing to bet it is a power supply issue... |
| 08:14.23 | bhrobinson | get a p/s tester and check it |
| 08:14.52 | geniesis | its possible |
| 08:15.01 | geniesis | although it is a cisco based psu |
| 08:15.04 | tzafrir | a p/s being? |
| 08:15.11 | bhrobinson | power supply |
| 08:15.14 | geniesis | and they generally don't skimp on things |
| 08:15.31 | geniesis | the psu is not standard issue |
| 08:15.38 | bhrobinson | if it is a p3 based machine... it also is not new :) |
| 08:15.44 | geniesis | and its certainly not atx form |
| 08:16.01 | *** join/#asterisk dharrigan (n=david@host12.williamhill.co.uk) |
| 08:16.25 | tzafrir | did it work well before? |
| 08:16.28 | geniesis | i don't have a digital multimeter so i can't test how accurate it is |
| 08:16.50 | geniesis | well...to be true, i haven't actually had it working stable yet |
| 08:17.00 | geniesis | until now its pretty much been a driver issue |
| 08:17.31 | geniesis | with random noise and stuff |
| 08:17.49 | geniesis | however now that i have sorted that issue, its now the power alarm |
| 08:18.39 | geniesis | funnily enough using the 1.4 branch of the zaptel drivers i get excessive noise but no power alarms.....using the 1.2 branch i get perfect sound but there are power alamrs |
| 08:18.58 | *** join/#asterisk af_ (n=getsmart@81-174-47-190.dynamic.ngi.it) |
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| 08:22.02 | bhrobinson | if you have access to an old AT power supply, hook it to the dsub on the card and power the unit... |
| 08:22.28 | bhrobinson | I would be willing to bet the +5 is floating down to 3.5 or up to 6 |
| 08:22.40 | geniesis | i have an old AT psu but its rails are far from stable |
| 08:30.52 | *** join/#asterisk falco_toadfoot (i=papir@104.80-202-255.nextgentel.com) |
| 08:31.24 | falco_toadfoot | anyone notice that AGI: GET DATA never returns the first digit pressed? only the following ones |
| 08:32.00 | *** join/#asterisk grEvenX (n=even@1ult2p8.ip.hipercom.no) |
| 08:32.24 | falco_toadfoot | that's in 1.2.13 |
| 08:33.49 | falco_toadfoot | I've checked with agi debug, and this is what I get from pressing 123: |
| 08:34.20 | falco_toadfoot | AGI Tx >> 200 result=23 (timeout) |
| 08:36.35 | *** part/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
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| 09:08.09 | *** join/#asterisk MrTelephone (n=test@S010600195b946fa3.ls.shawcable.net) |
| 09:08.32 | MrTelephone | ${CALLERID(ani} will tell you wether a call is collect or not? |
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| 09:22.41 | Modcuts | How do you sort priorty when you transfer a call to a extension that is part of a ring group?, how do you make sure the phone being transfered to doesn't pick up the incoming call? |
| 09:22.46 | *** join/#asterisk Strom_C (n=strom@netblock-66-159-243-60.dslextreme.com) |
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| 09:29.36 | *** join/#asterisk alin` (n=user@193.226.173.50) |
| 09:30.48 | alin` | To me SIP SHOW SUBSCRIPTIONS shows this line: |
| 09:30.48 | alin` | 192.168.144.149 s 3c26700a46c 222@s Idle dialog-info+xml <none> |
| 09:30.48 | alin` | What I must do in order to change «Idle» to something else? |
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| 09:46.22 | *** join/#asterisk masus (n=tet@88.248.14.186) |
| 09:46.27 | masus | hi all, |
| 09:47.36 | yonahw-work | when i try to use make menuselect it asks me to install ncurses although both nucurses and ncurses-devel are installed. anyone have any ideas why and/or how to fix? |
| 09:48.06 | masus | --> Linux pbx 2.6.18-4-686 'Asterisk 1.2.13' Registration from '<sip:100@192.168.1.1>' failed for '192.168.1.149' - Username/auth name mismatch |
| 09:48.22 | masus | can anybody help ? |
| 09:49.19 | creativx | fix the username |
| 09:49.53 | *** join/#asterisk DaveCanoe (n=Dave@belbrrcnas02-3637524643.dial.bell.ca) |
| 09:50.02 | *** join/#asterisk bdheeman (n=bsd@122.162.2.189) |
| 09:50.22 | masus | ? |
| 09:50.35 | masus | how it's the same one in sip.conf |
| 09:51.31 | masus | type=friend username=100 fromuser=100 secret=100 host=dynamic dtmfmode=rfc2833 canreinvite=yes nat=no qualify=yes disallow=all |
| 09:51.45 | masus | and my grandstream has the same user and pass |
| 09:55.17 | *** part/#asterisk bdheeman (n=bsd@122.162.2.189) |
| 09:56.37 | Daviey | I'm planning to use astfax, 1 channel licence is free - which means i can queue outgoing faxes.. What happens if an inbound fax happens if i am sending at the same time? |
| 09:56.44 | *** join/#asterisk z_mitya (n=mitya@dsl5400B0D4.pool.t-online.hu) |
| 09:57.47 | z_mitya | hi everybody |
| 09:58.19 | z_mitya | can i have a little question ? |
| 09:59.55 | z_mitya | is it possible to do an announcement server with * ? |
| 10:00.05 | *** join/#asterisk shtoom (n=shtoom@59.181.111.61) |
| 10:00.54 | z_mitya | when the INVITE arrives, play back the audio with 183 Session in progress, then redirect the client with 302 ? |
| 10:04.49 | z_mitya | is anybody here ? |
| 10:04.52 | *** join/#asterisk |dennis| (n=dennis@200.32.233.84) |
| 10:10.58 | masus | no only jack is here |
| 10:17.02 | z_mitya | do you know a place somewhere, where sy can answer my question ? |
| 10:17.28 | z_mitya | because the asterisk-users mailing list is terribly slow |
| 10:20.20 | masus | u can ask here |
| 10:20.28 | masus | and wait for an answer |
| 10:24.54 | *** part/#asterisk masus (n=tet@88.248.14.186) |
| 10:24.55 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
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| 11:18.40 | Daviey | Bah.. at least £11.70 (UKP) per channel for PRI in the UK... arg |
| 11:18.52 | Daviey | --> per month |
| 11:20.28 | *** join/#asterisk nayfan (n=nate@193.82.139.227) |
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| 11:41.41 | Swat2 | can anyone show me what an SLA Station extension should look like in sip.conf? |
| 11:41.46 | Swat2 | bit confused by the pdf :( |
| 11:45.10 | *** join/#asterisk Aurs (n=Aurs@1ult2p8.ip.hipercom.no) |
| 11:48.29 | Swat2 | do i just change the context? |
| 11:50.40 | E-bola | Hmm i didnt knwo what an SLA was, so i googled it, i dont really see what the use is though |
| 11:50.57 | Rienzilla | it's a service level agreement |
| 11:51.06 | E-bola | not int his context |
| 11:51.10 | E-bola | http://www.asterisk.org/node/48342 |
| 11:51.12 | Rienzilla | oh |
| 11:51.13 | Rienzilla | nm :) |
| 11:51.32 | E-bola | i knew it as service level agreement too, but it didnt make sence in his question :) |
| 11:51.37 | Rienzilla | hehe |
| 11:51.38 | Swat2 | SLA Shared line Appearance |
| 11:51.44 | Swat2 | its like a keysystem for * |
| 11:51.58 | E-bola | whats a keysystem? the webpage talks about that too |
| 11:52.37 | Swat2 | mmm |
| 11:52.52 | Swat2 | really old telephone system |
| 11:53.13 | Swat2 | where you can pick up and put on hold on the line |
| 11:53.18 | Swat2 | and anyone can pick up the lines |
| 11:53.36 | E-bola | hmm ok |
| 11:53.48 | Swat2 | but im struggling to get BLF working as well :( |
| 11:54.00 | E-bola | i fixed blf on snom and grandstream phones last month |
| 11:54.07 | E-bola | was pretty easy |
| 11:54.10 | Swat2 | ooer |
| 11:54.13 | Swat2 | talk to me |
| 11:54.19 | Swat2 | i got snom 320's |
| 11:54.29 | Swat2 | and finding that my hints keep saying the line is on hol |
| 11:54.30 | Swat2 | d |
| 11:54.58 | E-bola | i used snom 320's too |
| 11:55.14 | E-bola | did u restart them after setting up the hints and subscribing to them? |
| 11:55.24 | Swat2 | yeah |
| 11:55.30 | Swat2 | i think the phone is doing the right thing |
| 11:55.33 | Swat2 | i think * isnt |
| 11:55.55 | Swat2 | or, more to the point, i think i've stuffed up somewhere :P |
| 11:56.59 | Swat2 | in the phone did you just set it to "extension" then the extension number? |
| 11:57.36 | E-bola | yes |
| 11:57.44 | *** join/#asterisk luisjose (n=ljd@nelug/coreteam/luisjose) |
| 11:57.45 | E-bola | no not the extension number |
| 11:57.46 | E-bola | the hint number |
| 11:57.47 | JT | Daviey: sounds fairly standard |
| 11:58.01 | Swat2 | mmmm |
| 11:58.14 | Swat2 | so i assume the hint number should be different to the extension number? |
| 11:58.18 | E-bola | swat: do show hints |
| 11:58.21 | E-bola | on the * console |
| 11:58.28 | E-bola | and try and read up on it a bit |
| 11:58.35 | E-bola | it can be a bit tricky ont he hpones |
| 11:58.41 | E-bola | but the asterisk part is straight forward |
| 11:58.54 | Swat2 | 1002@ext-local : SIP/1002 State:Idle Watchers 1 |
| 11:58.54 | Swat2 | <PROTECTED> |
| 11:59.21 | E-bola | is ur snom phone usibng newest firmware? |
| 11:59.24 | SuurMyy | hi ? |
| 11:59.32 | Swat2 | i think so |
| 11:59.47 | SuurMyy | Is it okay to ask q:s abt. config here ? |
| 12:01.15 | Swat2 | E-bola: does that look right? |
| 12:01.21 | E-bola | if you dont call them q:s then sure :) |
| 12:01.31 | SuurMyy | k... |
| 12:01.38 | E-bola | swat2: if u are trying to monitor the state of extension 1002 then yes |
| 12:01.49 | Daviey | JT: Poo.. can't wait for truly reliable & cheap broadband (ie 1Gbps with QoS) |
| 12:02.36 | SuurMyy | I have Asterisk behind firewall, and I try to connect to a phone in lan to another phone thru sipphone.com. I can hear the other party, the other party can't here me, and Asterisk quite fast auto-disconnects. Firewall should be ok. |
| 12:02.55 | Swat2 | E-bola: should it light up if that extension is active? |
| 12:03.27 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 12:03.44 | Swat2 | when i say active, i mean, on & registered |
| 12:03.48 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 12:03.51 | alin` | I have a user called `200`, and a user called `100`; I tried to define the hint |
| 12:03.51 | alin` | exten => 445,hint,SIP/200 |
| 12:03.51 | alin` | to see on the phone `100` when the phone `200` is busy, but I receive the answer: |
| 12:03.51 | alin` | <PROTECTED> |
| 12:03.54 | alin` | <PROTECTED> |
| 12:04.12 | *** join/#asterisk nfi|ermes (n=ErMeS@217.220.121.62) |
| 12:04.12 | alin` | What is wrong to me? |
| 12:04.26 | SuurMyy | sorry, badly formed: Have phone in lan, connected to asterisk locally, Asterisk is natted, but firewall forwards correctly. Can hear other party thru sipphone, but cannot be heard at that end. At that end, no firewall. |
| 12:04.27 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 12:05.33 | *** join/#asterisk lirakis (n=etamme@65.200.191.253) |
| 12:05.36 | alin` | It is ok, I forgot to restart the * :) |
| 12:05.57 | *** join/#asterisk ccesario (n=ccesario@200-158-226-62.dsl.telesp.net.br) |
| 12:06.38 | *** join/#asterisk darviria (n=darviria@194-105-181-29.ifb.co.uk) |
| 12:06.59 | nfi|ermes | hi all |
| 12:07.13 | nfi|ermes | how can i activate cdr_mysql |
| 12:07.15 | nfi|ermes | ? |
| 12:07.36 | JT | Daviey: but broadband is a totally different product to PRI. |
| 12:07.44 | nfi|ermes | i installed asterisk addons, i created the database and the table |
| 12:08.23 | alin` | It still does not work... What I must write after `hint` to monitor the status of a phone registered to * ? |
| 12:10.18 | Daviey | JT: i know that; but when you can reliably route all calls via VOIP provider - costs will plummet |
| 12:10.53 | JT | Daviey: yeah good luck with that |
| 12:10.59 | Daviey | heh |
| 12:11.02 | JT | Daviey: they're not even parallels |
| 12:11.10 | JT | pri is a totally different service |
| 12:11.17 | JT | run on far better equipment |
| 12:11.25 | JT | telco grade |
| 12:11.26 | Swat2 | E-bola: did you have to use subscribecontext? |
| 12:11.36 | Daviey | I know that.. hence "can't wait for reliabilty" |
| 12:12.32 | JT | Daviey: hence "keep waiting" |
| 12:14.54 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 12:16.21 | alin` | So Nobody can explain me why it does not light up on `200` when I call a number using the phone `100`? |
| 12:16.44 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 12:16.49 | Swat2 | alin`: blf question? |
| 12:16.52 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 12:17.36 | E-bola | Swat2: yes |
| 12:17.55 | Swat2 | in the cli type show hints |
| 12:18.05 | alin` | Swat2: exactly |
| 12:18.05 | Swat2 | and see if it has any watchers |
| 12:18.52 | alin` | <PROTECTED> |
| 12:18.53 | alin` | <PROTECTED> |
| 12:18.53 | alin` | <PROTECTED> |
| 12:19.13 | alin` | Meetme wrork very nice when I make a conference |
| 12:19.39 | Swat2 | what phone is ext 100 |
| 12:19.46 | Swat2 | sorry |
| 12:19.50 | Swat2 | what phone is ext 200 |
| 12:19.52 | alin` | However the extension 445 works wring |
| 12:20.03 | alin` | 200 is ARTDIO |
| 12:20.04 | mvanbaak | and how does the config look like ? |
| 12:20.08 | alin` | 100 is SNOM 320 |
| 12:20.34 | alin` | mvanbaak: what config? |
| 12:20.38 | Swat2 | which phone are you trying to get the BLF to work on? |
| 12:21.25 | *** join/#asterisk IPmonger (n=ipmonger@c-71-224-31-246.hsd1.pa.comcast.net) |
| 12:21.50 | alin` | Swat2: fron the SNOM 324 I subscribed to * to see the extension `200` (200 is a ARTDIO phone) |
| 12:22.17 | alin` | From the SNOM 320 I subscribed to see the meetme room, and it works perfectly! |
| 12:22.41 | DrAk0 | Jul 31 14:22:16 WARNING[26082]: chan_iax2.c:1768 attempt_transmit: Max retries exceeded to host 190.38.184.242 on IAX2/190.38.184.242:4569-4 (type = 6, subclass = 1, ts=12, seqno=0) |
| 12:22.51 | DrAk0 | can't find the problem... |
| 12:23.05 | DrAk0 | why i cannot send calls from one asterisk to the other |
| 12:23.08 | alin` | meetme has the extension 444. the extension 445 is dead completely |
| 12:23.17 | DrAk0 | but i can call from the other asterisk to this one. |
| 12:23.31 | alin` | [100] |
| 12:23.31 | alin` | exten => 444,hint,Meetme:1234 |
| 12:23.31 | alin` | exten => 445,hint,SIP/200 |
| 12:23.31 | alin` | That is the config |
| 12:23.43 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 12:28.21 | *** join/#asterisk saftsack (n=saftsack@pD9E06F0E.dip.t-dialin.net) |
| 12:29.24 | *** join/#asterisk jkiff (n=jkiffmey@unaffiliated/vorondil) |
| 12:29.51 | nfi|ermes | i installed asterisk-addons but i don t find in filesystem cdr_addon_mysql.so |
| 12:30.11 | nfi|ermes | perhaps in asterisk-addons 1.4 , it shouldn t be there ? |
| 12:30.54 | Swat2 | alin`: asterisk 1.4? |
| 12:31.21 | alin` | Swat2: 1.4.9 |
| 12:33.00 | Swat2 | alin`: i too am having problems, i just stumbled on this: http://freepbx.org/trac/ticket/2127 |
| 12:33.45 | alin` | [Jul 31 17:22:28] ERROR[15918]: pbx.c:1523 ast_func_read: Function DEVSTATE not registered |
| 12:33.47 | alin` | <PROTECTED> |
| 12:33.47 | alin` | How can I register the devstate() application in * ? |
| 12:34.05 | alin` | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+BristuffDevstate |
| 12:35.20 | *** join/#asterisk duity (n=some1@static-71-98-242-250.tampfl.dsl-w.verizon.net) |
| 12:36.35 | tzafrir | alin`, no, the function devstate is from 1.4 |
| 12:36.42 | tzafrir | maybe you don't have it built? |
| 12:37.27 | alin` | tzafrir: my asterisk does not support devstate |
| 12:37.29 | alin` | Why? |
| 12:39.27 | duity | We have a 300 MHz box setup with linux 2.6 and asterisk. We're using voicepulse to make calls to the outside. We never need more than 2 people to make calls at the same time. Is the box fast enough for that? |
| 12:39.55 | alin` | I do not understand why * has 2 stable versions: 1.4.9 and 1.2.23. |
| 12:39.56 | duity | The CPU usage hardly ever goes up at all... not sure if it's really telling me the right numbers there. |
| 12:40.05 | alin` | 1.2.23 is not too old? |
| 12:40.40 | E-bola | alin: so does the linux kernel? |
| 12:42.27 | *** join/#asterisk joaop (n=taken@201.47.25.109.adsl.gvt.net.br) |
| 12:42.42 | duity | Anybody else use voicepulse? |
| 12:43.19 | alin` | E-bola: ?? |
| 12:43.33 | tzafrir | E-bola, not exactly |
| 12:44.31 | tzafrir | alin`, because the Asterisk developers have not managed to stabalize 1.4 in time and thus people stuck with 1.2 |
| 12:44.52 | *** join/#asterisk GingerDog (n=GingerDo@irc.palepurple.co.uk) |
| 12:45.07 | alin` | tzafrir: a, ok |
| 12:45.09 | E-bola | this is uber normal in big software projects? |
| 12:45.50 | tzafrir | E-bola, supporting for longer time is always more fun. But takes more resources |
| 12:45.53 | GingerDog | hello; I have digit and response timeouts set, but WaitExten seems to not want to pick up numbers entered from the handset, any suggestions? (it seems to have almost broken overnight which is most annoying) |
| 12:46.16 | lmadsen | alin`: it's like the difference between the 2.4.x and 2.6.x kernel |
| 12:46.51 | alin` | lmadsen: ok. |
| 12:47.00 | E-bola | lmadsen: ohh really? :) |
| 12:47.10 | alin` | For shared line I need devstate also? |
| 12:47.11 | *** join/#asterisk HaMYaI (n=LAMER@ppp-58.8.3.180.revip2.asianet.co.th) |
| 12:47.16 | *** part/#asterisk HaMYaI (n=LAMER@ppp-58.8.3.180.revip2.asianet.co.th) |
| 12:47.30 | lmadsen | well.... I'm using it as an example of two things that are developed simultaneously :) |
| 12:47.49 | lmadsen | and I just woke up about 30s ago, so you should probably ignore anything I say for at least 30 mins |
| 12:47.56 | creativx | slacker |
| 12:50.06 | lmadsen | heck ya |
| 12:50.14 | creativx | ssshh! |
| 12:50.21 | creativx | so how come your name is madsen, leif |
| 12:50.32 | creativx | where in the great scandinavia did it spring out of |
| 12:50.39 | lmadsen | eh? |
| 12:50.45 | lmadsen | my Dad named me that :) |
| 12:51.06 | lmadsen | my grandfather on my fathers side was from Denmark |
| 12:51.30 | *** join/#asterisk anonymouz666 (n=anonymou@189.25.49.71) |
| 12:52.04 | anonymouz666 | good morning! |
| 12:52.28 | creativx | ah i see |
| 12:53.26 | Sci_05 | FYI guys http://securitytracker.com/alerts/2007/Jul/1018472.html |
| 12:59.20 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 13:01.29 | EricL | If I have all my hints layed out by extension, but I have the extensions done using a regex, is that going to prevent presence from working? |
| 13:02.18 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:03.05 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 13:03.18 | *** join/#asterisk myiagy (n=myiagy@201.56.113.74) |
| 13:04.39 | *** join/#asterisk hohum (n=dcorbe@gate.globecommsystems.com) |
| 13:06.55 | *** join/#asterisk nighty^ (n=nighty@p2136-adsau16honb13-acca.tokyo.ocn.ne.jp) |
| 13:16.18 | festr__ | hello, anyone have A102D or A101D? |
| 13:16.21 | *** join/#asterisk hohum (n=dcorbe@gate.globecommsystems.com) |
| 13:16.22 | festr__ | with HW echocan |
| 13:17.07 | festr__ | i've bad expirience with hw echocan. when turning off ec faxes are about 95%. hw echo on cause that fax cannot even train |
| 13:22.13 | lmadsen | echo can needs to be suppressed with faxing |
| 13:22.15 | *** join/#asterisk _DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:22.17 | wothinn | Are there any *-compatible softphones that run on Windows with a global hotkey to mute/unmute? |
| 13:22.18 | lmadsen | it'll break faxes |
| 13:23.05 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:23.05 | Maliuta | softphones just blow goats |
| 13:23.14 | anonymouz666 | I wonder how can I make a call to be recorded when I digit *08 to pickup. |
| 13:23.44 | wothinn | Maliuta: Thanks for the insight. :-P |
| 13:24.17 | JT | wothinn: it's true though |
| 13:24.20 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:24.27 | JT | festr__: echo cancellation must be off for faxes |
| 13:24.29 | Maliuta | wothinn: but it's true, even the best IAX/SIP softphone I have used isn't as good as the hardware SIP unit I have |
| 13:24.38 | datachomper | *somebody* has to get around to making a descent softphone :OP |
| 13:24.53 | JT | datachomper: still a horriblee support nightmare |
| 13:24.58 | tzafrir | It is also the fact that hardphones have a dedicated system |
| 13:25.05 | JT | making sure someone's speaker and mic volumes are all set right |
| 13:25.08 | creativx | morning JT :) |
| 13:25.14 | Maliuta | I love my cisco 7940 |
| 13:25.15 | tzafrir | If you build your OS right, you can use a soft phone just fine |
| 13:25.16 | JT | making sure they're using standard headsets |
| 13:25.21 | JT | hi creativx |
| 13:25.24 | wothinn | True though it is, it doesn't help. I'm playing with the idea of using * to replace Ventrillo for game-like chatter and need a push-to-talk in app_conference to pull it off. |
| 13:25.32 | wothinn | I'm not going to be able to do that with a hardphone. |
| 13:25.41 | JT | wothinn: there is no push to talk |
| 13:25.46 | wothinn | No matter how much I like my Polycom IP500, it just can't do that. :) |
| 13:25.49 | creativx | softphone hell |
| 13:25.52 | JT | Maliuta: prefer my polycoms :) |
| 13:25.57 | datachomper | Lol, I've got a polycom500 myself |
| 13:25.57 | creativx | esp with plantronic usb headsets |
| 13:26.03 | creativx | they have crazy microphones |
| 13:26.07 | Maliuta | wothinn: there are heaps of "usb voip" headsets out there .... |
| 13:26.17 | Maliuta | if you don't mind being forced to use skype |
| 13:26.32 | wothinn | Not interested in Skype... I have a nice app_conference setup already. |
| 13:26.41 | JT | wothinn: ptt? |
| 13:26.41 | Maliuta | JT: I got the cisco at a good price |
| 13:26.46 | wothinn | I just want something I can press a global hotkey and have it mute or unmute. |
| 13:26.47 | JT | Maliuta: free? |
| 13:26.52 | Maliuta | and I'm going to get the DECT headset for it |
| 13:27.11 | Maliuta | JT: not quite, but cheaper than I found a polycom unit |
| 13:27.19 | Maliuta | 'specially when it is for home |
| 13:27.25 | JT | Maliuta: the polycoms units ARE worth more ;) |
| 13:27.29 | wothinn | Thought one of you guys might know a softphone that does that rather than my going to install every softphone I can find to find out. |
| 13:28.02 | JT | wothinn: app_rpt does ptt, think it's iax only though |
| 13:28.04 | Maliuta | JT: I don't doubt it, just can't justify that spend for home ATM |
| 13:28.13 | creativx | wothinn: write an app that subclasses x-lite ;) |
| 13:28.18 | JT | Maliuta: USD$86? |
| 13:28.29 | Maliuta | JT: if you hired me for 3days a week in brisbane for what I am worth ... ;) |
| 13:28.29 | wothinn | Hmm... Thanks, JT, I didn't think of doing it in the server. I'll look at that. |
| 13:28.38 | Maliuta | JT: wasn't that cheap |
| 13:28.56 | JT | Maliuta: telephonydepot.com > phones > polycom > IP320 |
| 13:29.10 | *** join/#asterisk Daviey (i=daviey@ubuntu/member/daviey) |
| 13:29.53 | Maliuta | JT: don't tempt me to spend money I don't have (or need to move from hellbourne back to brisbane) |
| 13:30.07 | JT | mm melbourne |
| 13:30.23 | Maliuta | _hell_bourne |
| 13:30.42 | JT | pfft what's so bad about it? |
| 13:31.14 | Maliuta | it's not brisbane |
| 13:31.24 | *** join/#asterisk RSAMan (n=a@dsl-242-44-247.telkomadsl.co.za) |
| 13:31.25 | Maliuta | I have been here for almost 2 years now |
| 13:31.28 | RSAMan | hello guys |
| 13:31.36 | RSAMan | once i have modified sip.conf |
| 13:31.40 | RSAMan | what should i reload |
| 13:31.41 | RSAMan | ? |
| 13:31.50 | RSAMan | and the same for extensions.conf |
| 13:31.59 | Maliuta | probably shouldn't have even gone back to work so soon after transplant, let alone move my life 2000kms south |
| 13:32.04 | JT | the only thing i really liked about brisbane was riprarian tower/plaza (sp?) |
| 13:32.21 | Maliuta | ???? |
| 13:32.28 | Maliuta | sure you got then name right? |
| 13:32.33 | Maliuta | or even close |
| 13:32.35 | JT | a nice piece of architecture in the city |
| 13:32.52 | datachomper | RSAMan, Are you in the CLI? |
| 13:33.04 | JT | Maliuta: Riparian Plaza |
| 13:33.19 | RSAMan | yes |
| 13:33.25 | RSAMan | datachomper : yes |
| 13:33.29 | Maliuta | not ringing bells, and I know brisbane pretty well |
| 13:33.44 | Maliuta | sure locals don't call it something else? |
| 13:33.59 | wothinn | RSAMan: 'sip reload' and 'dialplan reload' on 1.4.x |
| 13:34.17 | RSAMan | 1.2.x? |
| 13:34.43 | JT | Maliuta: maybe it's called "out of the loop" :P |
| 13:34.59 | Maliuta | I'm not that far out of the loop |
| 13:35.06 | JT | Maliuta: it's googleable. |
| 13:35.06 | Maliuta | got a crossroads for it? |
| 13:35.07 | datachomper | RSAMan, you want "dialplan reload" and "sip reload" |
| 13:35.14 | JT | and for brisbane, it's a huge building |
| 13:35.15 | datachomper | oh, pfhh |
| 13:35.16 | RSAMan | kk |
| 13:35.17 | RSAMan | thanks |
| 13:35.18 | JT | on the water |
| 13:35.54 | *** join/#asterisk yannj_fr (n=yannj@APuteaux-152-1-61-113.w82-120.abo.wanadoo.fr) |
| 13:36.09 | yannj_fr | Hi |
| 13:36.09 | Maliuta | JT: oh, that thing ... I didn't know it had a name |
| 13:36.24 | Maliuta | JT: down near waterfront place |
| 13:36.36 | Maliuta | I remember them planting the foundations |
| 13:36.50 | yannj_fr | Is there anybody who knows how to generate the diguest for resync and reboot by SIP notify on Grandstreams |
| 13:36.51 | yannj_fr | ? |
| 13:37.07 | JT | Maliuta: well it's built now, they just never update the web site |
| 13:38.20 | Maliuta | JT: yeah, it went up rapidly compared to some of the other recent additions to the skyline |
| 13:38.47 | JT | it was the only thing interesting in brisbane ;) |
| 13:39.10 | Maliuta | not really, I guess you just haven't looked in the right places :P |
| 13:39.24 | JT | sure |
| 13:39.35 | Maliuta | I know brisbane better than most people who can't be bothered to look |
| 13:39.57 | tzanger | morning |
| 13:40.03 | tzanger | man I gotta get these RTCP events into a db |
| 13:40.17 | *** join/#asterisk viraptor (n=viraptor@87.194.164.154) |
| 13:40.37 | viraptor | hey |
| 13:40.49 | Maliuta | is facebook like a virus or something? |
| 13:40.56 | JT | Maliuta: ok, great, you like brisbane, you've lived there for a long time or whatever. great. |
| 13:41.16 | JT | i haven't been infected by facebook |
| 13:41.17 | Maliuta | you get an account to look at one persons photos and it never ends |
| 13:41.28 | viraptor | is there any newer version of IAX draft than april 2007? |
| 13:42.44 | creativx | Maliuta: it is |
| 13:43.10 | Maliuta | it's just disturbing, where do people find the time for this crap? |
| 13:44.13 | wothinn | It just takes away from their time they could be spending playing with app_conference. :-P |
| 13:45.00 | [TK]D-Fender | Maliuta : Getting in touch with old friends, reunions, etc. Sharing changes in their lives, blogging, general fun.... |
| 13:45.08 | Maliuta | obviously my toy are cooler than theirs :P |
| 13:45.24 | JT | [TK]D-Fender: seems to be mostly a nort american thing |
| 13:45.27 | [TK]D-Fender | wothinn: Most of the * world works just fine with Meetme. Let me guess... you're trying to VM your * server, right? |
| 13:45.28 | JT | north |
| 13:45.31 | Swat2 | 1002@ext-local : SIP/1002 State:Hold <--- Cant seem to get rid of the Hold state, any one know how to fix? |
| 13:45.42 | Maliuta | [TK]D-Fender: if I wanna get intouch I'll call them, from my asterisk box :P |
| 13:45.52 | wothinn | [TK]D-Fender: No, I have no timing source because the ZT modules don't run on OpenBSD. |
| 13:45.57 | wothinn | So app_conference it is. |
| 13:46.11 | [TK]D-Fender | wothinn: OH you're a MASOCHIST! |
| 13:46.19 | wothinn | Possibly. :) |
| 13:46.21 | creativx | JT: it's come as far as norway too, so .. |
| 13:46.40 | Maliuta | ahhh, people who buy digium kit for openbsd boxen without looking first :) |
| 13:46.41 | JT | creativx: it's in australia, it's just not quite as popular |
| 13:47.03 | JT | Maliuta: who bought the digium gear? |
| 13:47.18 | Maliuta | JT: maybe because most bogons don't know how to use the 'net ;) |
| 13:47.43 | Maliuta | a guy I know bought a TMD400 to put into an openbsd box |
| 13:47.48 | JT | nice |
| 13:47.59 | Maliuta | about a month before I bought mine to put into a linux box |
| 13:48.13 | Maliuta | mine works |
| 13:48.31 | Swat2 | whats wrong w/ digium kits? |
| 13:48.42 | Maliuta | Swat2: nothing, I like mine |
| 13:48.49 | *** join/#asterisk rbooz (n=rbooz@66.28.136.226) |
| 13:49.02 | Maliuta | Swat2: there is just no openbsd support (or there wasn't when I cared to look) |
| 13:49.04 | JT | i think he was talking about bsd |
| 13:49.10 | Swat2 | ahh |
| 13:49.15 | JT | but many have other issues like interupts and timing |
| 13:49.35 | Swat2 | mmm |
| 13:49.55 | Swat2 | im just having trouble with "show hints" not updating and always saying state:hold :( |
| 13:49.56 | Maliuta | JT: I don't particularly care what type of *nix it is, aslong as I can get my GNU's *nix is *nix is *nix :) |
| 13:50.41 | rbooz | morning all. I have a very annoying Zaptel issue that I've searched high and low on (even tried to talk briefly with Tzafrir Cohen yesterday) but can't get resolution on - wondering if someone has a few minutes to listen/maybe give a fresh look at |
| 13:51.06 | rbooz | Zaptel 1.2.18, Digium TDM400P (4 FXO cards) |
| 13:52.05 | mocker | ~ask |
| 13:52.06 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 13:52.09 | *** join/#asterisk KDan (n=KDan@87-194-122-30.bethere.co.uk) |
| 13:53.01 | [TK]D-Fender | Maliuta : Some are just "more equal" than others ;) |
| 13:53.05 | rbooz | jbot: totally understood - sorry - I know that's bad form... |
| 13:53.36 | [TK]D-Fender | rbooz: Don't talk back to scripting engines... its only midly funny and we need quality humour here :) |
| 13:53.44 | Maliuta | [TK]D-Fender: are you refering to "Fisher Price: My First *nix" ? ;) |
| 13:53.47 | rbooz | Been using Asterisk and the TDM card for 2 years - very little problems - three different "servers" in that time |
| 13:54.10 | rbooz | [TD]D-Fender: realized that name after I sent it... :-) |
| 13:54.30 | rbooz | we just moved offices and suddenly I get random "on-hook" events int he middle of calls |
| 13:55.23 | Maliuta | rbooz: have you looked into the possiblity that it's solar flare activity? |
| 13:55.41 | *** join/#asterisk bintut (n=bintut@cm16.gamma176.maxonline.com.sg) |
| 13:55.46 | rbooz | I've tried two different servers, one of which we never saw the problem on in the old office - both will randomly drop callers in the middle of calls. Timing isn't consistent, POTS line that drops is not consistent |
| 13:55.46 | Maliuta | tried wrapping the ports in alfoil? ;) |
| 13:55.56 | yonahw-work | when i try to use make menuselect it asks me to install ncurses although both nucurses and ncurses-devel are installed. anyone have any ideas why and/or how to fix? |
| 13:56.32 | rbooz | Maliuta: haven't gone that route yet... although new company dress code with the new office was all aluminum jackets... could that have something to do with it? |
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| 13:57.08 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:57.15 | *** join/#asterisk LuXten (n=noname@81.74.200.194) |
| 13:57.22 | Maliuta | rbooz: let me run your SAN kit past my MRI machine so I can make sure :P |
| 13:57.41 | rbooz | Maliuta: sounds good - let me know what you find... |
| 13:57.54 | datachomper | yonahw-work, try installing libncurses5 |
| 13:58.04 | Maliuta | rbooz: the answer is probably "platters, and shite" |
| 13:58.54 | tzafrir | yonahw-work, re-run ./configure ? |
| 13:59.40 | bintut | tzafrir: need your help. i'm trying to build a zaptel-module but i always get an error. please check http://paste.debian.net/33756 |
| 13:59.42 | yonahw-work | datachomper: thanks I will try |
| 13:59.50 | yonahw-work | tzafrir: tried that to no success |
| 14:00.27 | LuXten | Hi I have asterisk-1.2.23 and asterisk-addons-1.2.7 on a CentOS 4. I use a .mp3 file for MOH. Every 2-3 days asterisk segfaults. It always crashes on line 242 of format_mp3.c which says "s->buf[s->buflen] = s->dbuf[(s->buflen-save)+s->dbufoffset];". Could somebody help me in diagnosing this issue? |
| 14:01.10 | *** join/#asterisk rbooz (n=ryan@66.28.136.226) |
| 14:01.14 | *** join/#asterisk CM3_1_2_632 (n=CM3_1_2_@cm222-166-6-33.hkcable.com.hk) |
| 14:01.20 | *** join/#asterisk nettie (n=nettie@ns.coolgadgets.it) |
| 14:01.55 | tzafrir | bintut, dpkg -l linux-headers-`uname -r` # is it installed ? |
| 14:02.07 | rbooz | Maliuta: sorry about that - must have been a solar flare.... :-) |
| 14:02.10 | nettie | Hi guys, anyone know if I could specify in sip.conf the maximum channels phone could use please? |
| 14:02.28 | creativx | nettie: call-limit |
| 14:02.34 | bintut | tzafrir: afaik, yes. ii linux-headers- 2.6.18.dfsg.1- Header files for Linux 2.6.18 on PPro/Celero |
| 14:02.36 | rbooz | tzafrir: we talked briefly yesterday in #asterisk-il about my zaptel issue... any other thoughts? |
| 14:02.42 | Maliuta | rbooz: if you take the platters out of the SAN kit it _might_ work :) |
| 14:02.46 | tzafrir | I saw that |
| 14:02.51 | nettie | I need it to disable call-waiting from the server |
| 14:02.56 | tzafrir | I don't have any grand ideas |
| 14:02.57 | nettie | creativx uhmm thanx.. trying it.. |
| 14:03.12 | *** join/#asterisk Uatec (n=uatecuk@adsl.ntsols.com) |
| 14:03.16 | Uatec | Hi there |
| 14:03.30 | Uatec | i'm stuck using bri, for reasons best known to somebody who isn't me. |
| 14:03.47 | Uatec | I've already implemented an asterisk system using ABE and a B410p |
| 14:03.48 | rbooz | tzafrir: OK |
| 14:03.50 | rbooz | thanks |
| 14:03.57 | tzafrir | rbooz, I tend to blame hardware :-( |
| 14:04.01 | *** join/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 14:04.18 | Uatec | I think we're going to use ABE again, but what hardware would you guys recommend for BRI? |
| 14:04.19 | creativx | nettie: although it seems (atleast here with 1.2.10) that using call-limit on sip peers kills extensionstatus: events in the AMI |
| 14:04.20 | Maliuta | tzafrir: I tend to blame dell engineers for that |
| 14:04.26 | *** join/#asterisk MdeP (n=mdep@200.124.36.28) |
| 14:04.46 | rbooz | tzafrir: meaning the Digium card or the server? |
| 14:04.51 | *** join/#asterisk MdeP (n=mdep@200.124.36.28) |
| 14:04.53 | yonahw-work | datachomper: libncurses5 was already installed. any other ideas? |
| 14:05.00 | datachomper | libncurses5-dev |
| 14:05.02 | rbooz | (I'm happy to blame the hardware - I just need to fix the problem...) |
| 14:05.08 | bintut | tzafrir: what do you think? |
| 14:05.39 | nettie | creativx didnt work.. |
| 14:05.47 | nettie | creativx call-limit=1 |
| 14:06.07 | nettie | I was in a conversation and I could still receive calls |
| 14:06.12 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 14:06.16 | nettie | got a callwaiting message |
| 14:07.37 | creativx | nettie: sip reload? |
| 14:07.40 | nettie | ehehe |
| 14:07.46 | nettie | I did that of course |
| 14:08.04 | yonahw-work | datachomper: I have the ncurses-devel package on CentOS, is that by chance what you are referring to? |
| 14:08.13 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 14:08.32 | creativx | nettie: exactly where did you put it |
| 14:08.42 | datachomper | yonahw-work, Yep, I'm just covering everythign I usually have to install to get * to compile. |
| 14:08.44 | *** join/#asterisk ashd (n=ashleydr@user-194-248-151-83.e7even.com) |
| 14:09.18 | yonahw-work | datachomper: yeah I got those installed, hence my confusion when I am told to install ncurses |
| 14:09.55 | Uatec | does nobody have any opinion on BRI? |
| 14:10.05 | datachomper | yonahw-work: Trying re-doing a ./configure, and grepping for ncurses to see if it's finding it or not |
| 14:10.24 | *** join/#asterisk ManxPower (n=manxpowe@71-8-56-17.dhcp.leds.al.charter.com) |
| 14:10.29 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 14:10.51 | yonahw-work | datachomper: thanks i'll try that |
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| 14:11.07 | *** join/#asterisk echo--- (n=echo@64.184.118.232) |
| 14:11.12 | nettie | creativx: in sip.conf as an option of the phone I would like to limit |
| 14:11.13 | bintut | tzafrir: i just purged the linux-headers that i currently have and reinstalled a new one but still got the same error.. :( |
| 14:11.24 | brodiem | What are people using as a GUI to view call recordings/CDRs? |
| 14:11.31 | EricL | If I have all my hints layed out by extension, but I have the extensions done using a regex, is that going to prevent presence from working? |
| 14:11.41 | Uatec | brodiem, i'm using a custom written CRM system |
| 14:11.54 | yonahw-work | datachomper: i get "checking for initscr in -lncurses... yes" |
| 14:12.23 | *** join/#asterisk Dan0maN_Work (n=dschuh@64.149.174.136) |
| 14:12.26 | brodiem | Uatec you mean a public mod for a CRM or you personally customized it? |
| 14:14.47 | creativx | nettie: then im unsure. i tried it myself, but it killed all the fun functionality in my service app that chews AMI messages |
| 14:14.54 | datachomper | yonahw-work : Hmm, I'm out of ideas. You could grep that exact error message in the source tree and track it down in the code :O/ |
| 14:16.07 | yonahw-work | datachomper: sounds like fun, appreciate the help |
| 14:16.42 | datachomper | yonahw-work, sure |
| 14:17.16 | Uatec | brodiem, no, i wrote the CRM |
| 14:18.26 | [TK]D-Fender | EricL: Hints have to be hard numbered. Everything else is irrenevent. |
| 14:18.41 | [TK]D-Fender | irrelevant* |
| 14:19.09 | Maliuta | like resistance |
| 14:19.20 | wothinn | No, that's futile. |
| 14:19.37 | Maliuta | depends on how many ohms |
| 14:20.29 | EricL | [TK]D-Fender: That's how I have it and for some reason, the users state isn't updating at all (both on the phone and within *) |
| 14:20.41 | Chris-NB | hi |
| 14:20.54 | [TK]D-Fender | EricL: then I guess you'd better show us the relevant bits..... |
| 14:20.59 | Chris-NB | is it possible to get the version of a running asterisk from outside? |
| 14:21.14 | Chris-NB | tried via manager port and sip. but there is no version included : / |
| 14:21.15 | [TK]D-Fender | Chris-NB: outside of rhat? |
| 14:21.36 | Chris-NB | [TK]D-Fender, not logged in, from remote |
| 14:21.38 | tzafrir | bintut, busy now. I'll re-check that later this evening. I need to rebuild zaptel on my laptop anyway |
| 14:21.40 | [TK]D-Fender | Chris-NB: use AMI Command and use "show version" |
| 14:22.02 | Chris-NB | [TK]D-Fender, and without userlogin? like banner grabbing |
| 14:22.06 | bintut | tzafrir: ok. thanks.. :) |
| 14:22.09 | Chris-NB | is this possible? |
| 14:22.20 | [TK]D-Fender | Chris-NB: ?! |
| 14:22.45 | [TK]D-Fender | Chris-NB: You just asked via AMI and I said yes, you could. Whats the issue? |
| 14:22.47 | Chris-NB | [TK]D-Fender, for example telnet to port 22 shows running ssh version, to poert 25 shows running mailserver version |
| 14:22.58 | creativx | Chris-NB: the banner doesnt show exact version |
| 14:22.59 | Chris-NB | [TK]D-Fender, only interest. |
| 14:23.13 | creativx | AMI + auth + show version |
| 14:23.15 | [TK]D-Fender | Chris-NB: AMI is secured (relatively). You should know the rules for this. |
| 14:23.19 | Chris-NB | creativx, jep, but could. I'm just interested |
| 14:23.39 | [TK]D-Fender | Chris-NB: And no I don't believe there is any way to sniff off the version without authing somehow. |
| 14:23.48 | tzafrir | Maliuta, hardware (n), the part of the computer system you can kick. I can't kick the Dell engineers. |
| 14:23.58 | Chris-NB | [TK]D-Fender, ok, thats what I wanted to know. thanks |
| 14:24.15 | rbooz | tzafrir: so seriously - would you recommend I focus on the TDM card or the server (which does happen to be a new Dell...) |
| 14:24.23 | Maliuta | tzafrir: I just threaten to gouge their eyes out ... if I ever find the dick who came up with OMSA .... |
| 14:24.37 | *** join/#asterisk Cresl1n (i=matt@nat/digium/x-e89ab7fb3886596d) |
| 14:24.37 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 14:24.46 | creativx | open manage hell |
| 14:24.56 | Maliuta | rbooz: so _you_ want to gouge out the eyes of the OMSA dick aswell??? ;) |
| 14:25.07 | EricL | [TK]D-Fender: http://www.pastebin.ca/640568 |
| 14:25.11 | rbooz | Maliuta: I gave up on OSMA a long time ago... |
| 14:25.33 | Maliuta | rbooz: one of my co-workers suggested using a teabag when I was looking for something blunter and softer than a spoon |
| 14:25.49 | Maliuta | creativx: "open" |
| 14:25.59 | Maliuta | and lets not get started on DRAC |
| 14:26.07 | yonahw-work | datachomper: make distclean and then a ./configure solved my problem |
| 14:26.10 | Maliuta | call that "access"? |
| 14:26.11 | creativx | Maliuta: i have dell servers, but im not using any of that :) |
| 14:26.12 | tzafrir | rbooz, what happens if you replace the hardware? use a simple analog phone instead of the card? |
| 14:26.24 | creativx | Maliuta: im glad im closely located to the rack. hehe |
| 14:26.32 | Maliuta | creativx: I have to run all dell kit to, and not by my choice |
| 14:26.40 | *** join/#asterisk coppice (n=chatzill@93.195.17.210.dyn.pacific.net.hk) |
| 14:26.43 | tzafrir | Will you still have problems? |
| 14:26.46 | [TK]D-Fender | EricL: Did you enable presence in your Polycom sip.conf? Also show me your phone entries masking only PW's |
| 14:26.55 | Maliuta | creativx: I would _love_ to be able to drop in on my servers in London |
| 14:27.03 | tzafrir | That's a bit extreme method of testing here, right |
| 14:27.20 | creativx | Maliuta: you poor soul, heh |
| 14:27.31 | ManxPower | EricL: You don't have a <sd> for Eric |
| 14:27.34 | creativx | drac, the dell reboot access card |
| 14:27.41 | rbooz | tzafrir: actually, I was thinking of that - the problem is that it's random - and it's hard to tell the one client that we seem to have consistent problems with, "hey, can we call you and test our phone system?" - but, it's worth a shot |
| 14:27.43 | ManxPower | sd is the speed dial entry, it MUST be uniwue |
| 14:27.45 | ManxPower | unique |
| 14:28.19 | EricL | ManxPower: Because that entry is for my phone. Do I need to have a speed dial on my phone for myself? |
| 14:28.31 | ManxPower | Yes |
| 14:28.44 | ManxPower | Or at least "It should not hurt to have it" |
| 14:28.48 | ManxPower | also listen to [TK]D-Fender |
| 14:28.52 | tzafrir | rbooz, you saw in the logs that you get an on-hook event from the card, right? |
| 14:28.55 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 14:29.16 | rbooz | just looking again - got an email that it happened an hour ago for another user... |
| 14:29.19 | tzafrir | So it's either the card initiating it somehow, or getting a hangup from the line |
| 14:29.21 | ManxPower | [TK]D-Fender: feel free to point people to http://www.fnords.org/~eric/polycom-config-examples/ |
| 14:29.46 | ManxPower | tzafrir: he's not using busydetect or callprogress is he? |
| 14:29.51 | EricL | ManxPower: I looked at that example and I still don't understand why you have the mac-a mac-b etc entries. |
| 14:30.27 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 14:30.28 | tzafrir | <rbooz> chan_zap.c: Got event On hook(1) on channel 1 (index 0) |
| 14:30.39 | ManxPower | EricL: They are just the SIP username. If I was a newbie and wanted to paint myself into a corner I could have used the extension number, as long as each SIP userid is unique. |
| 14:30.49 | [TK]D-Fender | EricL: Please pastbin up the info I requested. |
| 14:31.04 | ManxPower | EricL: listen to [TK]D-Fender |
| 14:31.07 | EricL | [TK]D-Fender: Working on it. |
| 14:31.33 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 14:31.37 | rbooz | tzafrir: yep, here's the logs each time (realizing that the channel can be 1-4 depending on the incoming line) |
| 14:31.40 | rbooz | chan_zap.c: Exception on 13, channel 1 |
| 14:31.54 | rbooz | chan_zap.c: Got event On hook(1) on channel 1 (index 0) |
| 14:32.00 | rbooz | chan_zap.c: disabled echo cancellation on channel 1 |
| 14:32.06 | rbooz | channel.c: Didn't get a frame from channel: Zap/1-1 |
| 14:32.12 | rbooz | channel.c: Bridge stops bridging channels Zap/1-1 and SIP/27-08eb4bc8 |
| 14:32.15 | rbooz | and that's it |
| 14:32.45 | EricL | [TK]D-Fender: http://www.pastebin.ca/640576 |
| 14:33.20 | rbooz | tzafrir: if the card was initiating the on-hook - what would cause that in mid call? |
| 14:33.23 | [TK]D-Fender | EricL: What * ver? |
| 14:33.34 | EricL | [TK]D-Fender: 1.4.8 |
| 14:34.05 | [TK]D-Fender | EricL: You need to change from "type=friend" to "type=peer", and set "call-limit=100" |
| 14:34.52 | EricL | [TK]D-Fender: Does call-limit need to be set on an individual basis or is the global value sufficient? |
| 14:34.54 | rbooz | ManxPower: where is busydetect/callprogress defined? |
| 14:35.04 | rbooz | (I don't think I'm using it...) |
| 14:35.06 | [TK]D-Fender | EricL: I would recommend individual |
| 14:35.10 | *** join/#asterisk ido (n=ido@unaffiliated/ido) |
| 14:35.27 | ManxPower | robin_z: in zapata.conf |
| 14:35.42 | ManxPower | you should NOT have them defined or set them to no. |
| 14:36.18 | rbooz | OK: they are not defined... |
| 14:36.19 | Swat2 | 1002@ext-local : SIP/1002 State:Hold <--- Cant seem to get rid of the Hold state. Phone isnt on hold or on any calls yet it continues to say it's on hold.. any ideas? |
| 14:36.50 | yonahw-work | I have unixODBC installed and configured properly as well as unixODBC-devel yet ./configure can not find SQLConnect. anyone know how I can fix this problem |
| 14:37.17 | *** join/#asterisk afrosheen (n=cj@207.71.49.137) |
| 14:37.39 | nettie | creativx there's something wrong I'm sure.. |
| 14:37.41 | EricL | [TK]D-Fender: Alright, I made the changes for the phones and I am now restarting them... |
| 14:37.42 | [TK]D-Fender | Swat2: what is that line from>? |
| 14:37.44 | afrosheen | hey, does anyone have experience with the latest polycom firmware and how it traps the first 4 digits you dial? |
| 14:37.54 | Swat2 | [TK]D-Fender: show hints |
| 14:38.05 | [TK]D-Fender | Swat2: check your CHANNELS |
| 14:38.18 | [TK]D-Fender | Swat2: And pastebin EVERYTHING |
| 14:38.28 | [TK]D-Fender | afrosheen: Clarify please... |
| 14:38.55 | ManxPower | afrosheen: that is TOTALLY configured in the dialplan of the phone |
| 14:38.57 | afrosheen | [TK]D-Fender, if you pick up the phone and start dialing, after the first 4 digits it gives you a dialtone but lets you keep dialing until you hit send |
| 14:39.21 | afrosheen | this is all on the default config for the latest polycom firmware |
| 14:39.28 | *** join/#asterisk myiagy (n=myiagy@201.56.113.74) |
| 14:39.28 | [TK]D-Fender | afrosheen: depends on the phones dialplan. |
| 14:39.59 | ManxPower | afrosheen: that is totally configured in the dialplan for the phone. |
| 14:40.19 | afrosheen | it's pretty stupid for a default config, guess I need to dig into the files |
| 14:40.22 | ManxPower | [TK]D-Fender: is this going to be "I don't like your answer so I'll ignore it" day today? |
| 14:40.51 | EricL | [TK]D-Fender: Ah, so that was the difference. Presence doesn't work with type=friend? |
| 14:40.57 | bintut | tzafrir: i just able to build the zaptel-module. thanks anyway.. :) |
| 14:41.14 | [TK]D-Fender | EricL: That + the call limit for channel detect... some whacked changes came in with 1.4 |
| 14:41.27 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 14:41.29 | tzafrir | bintut, any idea what the problem was? |
| 14:41.31 | [TK]D-Fender | EricL: "friend" is pretty much dead in 1.4 |
| 14:42.35 | *** join/#asterisk yacc (n=andreas@85.127.77.158) |
| 14:42.37 | EricL | [TK]D-Fender: Alright...excellent. |
| 14:42.38 | *** join/#asterisk axisys (i=iqbala@outbound.silenceisdefeat.org) |
| 14:42.43 | EricL | [TK]D-Fender: You did it once again. |
| 14:42.49 | EricL | [TK]D-Fender: Thanks. |
| 14:43.04 | [TK]D-Fender | EricL: np |
| 14:43.04 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 14:43.18 | ManxPower | [TK]D-Fender: I would called that some whacked changes. |
| 14:43.28 | ManxPower | what exactly does calllimit have to do with device state anyway? |
| 14:43.45 | yannj_fr | Nobody knows how to generate the digest to reboot GrandStream Phones with SIP NOTIFY? |
| 14:43.49 | [TK]D-Fender | ManxPower: All things being equal (which never turns out to be the case), I'd agree.... |
| 14:44.20 | ManxPower | yannj_fr: the Wiki and your extensive search of the mailing list archives was not helpful? |
| 14:45.06 | yannj_fr | ManxPower : I found the patch on Mantis, but there were talking about a toolkit, googling didnt helped me |
| 14:46.04 | ManxPower | yannj_fr: want to know where I found the info to reboot my polycoms using SIP notify? |
| 14:46.12 | ManxPower | or options, I don't recall which SIP message. |
| 14:46.19 | ManxPower | On the Wiki, IIRC |
| 14:46.34 | yannj_fr | Hum, I saw of to reboot Snom, thomson ... |
| 14:46.36 | ManxPower | hell, there is even a sip_notify.conf file in asterisk to configure it |
| 14:46.41 | EricL | Since you all are Polycom users as well, is there a way to remote rebooy a Polycom? |
| 14:46.47 | ManxPower | EricL: Yes. |
| 14:46.52 | afrosheen | EricL, yep |
| 14:46.58 | yacc | Just wondering, is there a simple voice mail app (not a PBX), that listens on a SIP address, and after x rings plays a message and records the call? |
| 14:46.59 | yannj_fr | but Grandstream need to include a digest in the sip notify |
| 14:47.06 | bintut | tzafrir: i just reinstalled all linux-headers** and related packages.. |
| 14:47.21 | afrosheen | there are actually a few ways to reboot polycoms |
| 14:47.22 | ManxPower | ~gs |
| 14:47.22 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 14:47.28 | EricL | I found a Perl script that someone wrote to do it, but it doesn't seem to work. |
| 14:47.32 | ManxPower | jbot knows all |
| 14:47.32 | jbot | and don't you forget it |
| 14:47.43 | afrosheen | EricL, that's the one that depends on ARP right? |
| 14:47.48 | [TK]D-Fender | ~jbot |
| 14:47.49 | jbot | i guess jbot is a hack!, or known to have only said one useful thing. |
| 14:47.51 | yannj_fr | My boss joke : REboot the POE switch |
| 14:48.17 | EricL | afrosheen: Yep. |
| 14:48.22 | Mercestes | ~mercestes |
| 14:48.23 | jbot | mercestes is definitely a total nub |
| 14:48.28 | *** join/#asterisk MrTelephone (n=MrTeleph@h697179-171.picriverisp.net) |
| 14:48.34 | EricL | afrosheen: Do you have another one that works? |
| 14:48.37 | Mercestes | ~[TK]D-Fender |
| 14:48.38 | jbot | [TK]D-Fender is the Zen Master of the blatantly obvious. |
| 14:48.41 | afrosheen | EricL, you can reboot the phones if you know their IP addresses by hitting the web interface on the phone |
| 14:48.42 | Swat2 | [TK]D-Fender: http://www.pastebin.ca/640598 |
| 14:48.52 | MrTelephone | how do you verbose a dial plan var in cli? |
| 14:49.06 | afrosheen | EricL, OR you can touch the config file for that phone..the polycom will think it's updated and reboot to soak up the "new" settings |
| 14:49.16 | Capps- | Does anyone know of some good documentation on how to setup voicemail emails through Asterisk? |
| 14:49.48 | Mercestes | Capps-, It was so simple we chose not to overly document it. |
| 14:49.59 | brodiem | Capps- search voicemail.conf on wiki. |
| 14:50.00 | EricL | afrosheen: I touch'd the config file and it didn't reboot. |
| 14:50.09 | Mercestes | Capps-, You set up sendmail, then add the 3rd field as the email address you want to go to on the Wiki |
| 14:50.11 | afrosheen | EricL, is it using tftp or ftp |
| 14:50.12 | ManxPower | MrTelephone: Noop(EXTEN is ${EXEN}) |
| 14:50.26 | [TK]D-Fender | Swat2: "limit-call=1" = bad , "call-limit=1" good |
| 14:50.35 | EricL | afrosheen: tftp |
| 14:50.43 | ManxPower | The wiki is pretty much totally useless for things like application or option documentation, but it is handy for a "how do I?" |
| 14:50.53 | Swat2 | argh lol |
| 14:51.01 | Capps- | Mercestes: I figured so. I found the voicemail.conf documentation and forwarded it to the tech that is setting up the voicemail email and he said "This doesnt help." |
| 14:51.07 | [TK]D-Fender | EricL: You set a flag in sip.cfg and then dend the phone a chkconfig packet (via * CLI) |
| 14:51.10 | [TK]D-Fender | send* |
| 14:51.12 | afrosheen | EricL, maybe that trick doesn't work for tftp or the phone is on a polling timer...it works for us with FTP but I forgot how long it took |
| 14:51.18 | Mercestes | Capps-, That really is all there is to it. |
| 14:51.19 | MrTelephone | exten => _229XXXX,2,Verbose(0|${CALLERID(ani)} ${CALLERANI2}) |
| 14:51.22 | MrTelephone | hows that look? |
| 14:51.32 | Mercestes | Capps-, Just put in the email address. As long as "sendmail" works....your golden |
| 14:52.08 | EricL | [TK]D-Fender: Any flag? And then how do I send the phone a chkconfig packet (via the * CLI)? |
| 14:52.40 | [TK]D-Fender | EricL: No, there is one that tells the phone to reboot regardless of changes to the file. |
| 14:52.57 | [TK]D-Fender | EricL: That is well documented on the WIKI. Go read. |
| 14:52.59 | [TK]D-Fender | ~wikis |
| 14:52.59 | jbot | it has been said that wikis is http://www.voip-info.org |
| 14:53.12 | EricL | [TK]D-Fender: I have been on the wiki looking for it for the past 10 minutes. |
| 14:53.16 | EricL | I will keep looking. |
| 14:53.40 | Swat2 | [TK]D-Fender: you da man! |
| 14:54.07 | [TK]D-Fender | Swat2: ywc |
| 14:54.15 | afrosheen | Swat2, yes he is |
| 14:54.21 | EricL | Ah, found it in the forums. |
| 14:54.26 | Swat2 | now if i could just figure out how to make my park extensions show up in blf i'll be laughing |
| 14:55.24 | bintut | gtg now.. thanks.. :) |
| 15:00.13 | MrTelephone | how does the phone company know if a call is collect or not? |
| 15:01.04 | *** join/#asterisk saftsack (n=saftsack@pD9E06F0E.dip.t-dialin.net) |
| 15:02.03 | rbooz | why would a TDM400P (fxo) initiate an on-hook in the middle of a call? Could it be DTMF somehow? |
| 15:02.16 | *** join/#asterisk wunderkin (n=wunderki@ip72-223-86-126.ph.ph.cox.net) |
| 15:02.58 | nfi|ermes | i installed asterisk-addons but i don t find cdr_addon_mysql.so in filesystem |
| 15:03.08 | nfi|ermes | so i can t load module |
| 15:03.31 | Swat2 | http://www.pastebin.ca/640613 <-- is that correct syntax to add the park extensions to hints? |
| 15:03.39 | afrosheen | nfi|ermes, so you didn't see it in the asterisk-addons build folder after you ran make on it? |
| 15:04.17 | nfi|ermes | i only find cdr_addon_mysql.c in sources directory |
| 15:04.39 | afrosheen | nfi|ermes, sounds like it didn't get built then, run make again and watch for errors |
| 15:04.55 | nfi|ermes | ok |
| 15:05.30 | afrosheen | and by watch for errors I don't mean turn on Itchy and Scratchy Go To The Moon, be like me and stare at the terminal while it builds :) |
| 15:05.52 | nfi|ermes | it s so fast terminal |
| 15:06.04 | nfi|ermes | i try make >> debug |
| 15:06.31 | afrosheen | anyone know the prerequisites for building that module? |
| 15:07.07 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 15:07.14 | *** join/#asterisk rpm (n=russell@66.183.28.233) |
| 15:07.30 | [TK]D-Fender | Swat2: Looks about right. |
| 15:09.33 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 15:09.37 | Katty | allo. |
| 15:10.34 | afrosheen | zombie cat will do your bidding |
| 15:10.39 | *** join/#asterisk Voicemeup (n=VoiceMeU@modemcable159.131-56-74.mc.videotron.ca) |
| 15:11.32 | Voicemeup | Any reason when asterisk sends a 44 code pri.. all circuits busy SER doesnt do load next_gw ? i get a 403 back from asterisk |
| 15:11.36 | Voicemeup | shdount it be a 5XX ? |
| 15:12.15 | MrTelephone | how come there isn't much documentation on collect call information.. is noone else having issues with customers using collect? |
| 15:12.44 | afrosheen | MrTelephone, I guess nobody is calling from jail |
| 15:12.59 | MrTelephone | i got a 1 minute call from oklahoma 12 dolalrs |
| 15:13.18 | MrTelephone | no wonder why that guy blew up that building |
| 15:13.23 | afrosheen | lolcats |
| 15:13.35 | MrTelephone | voicemeup, hey what happened on sunday.. full pri? |
| 15:13.58 | Rienzilla | pff |
| 15:14.02 | MrTelephone | im wondering if collect calls usually have a special ani2 code.. the documentation isn't very clear |
| 15:14.46 | Strom_C | MrTelephone: read this |
| 15:14.49 | Strom_C | http://nanpa.com/number_resource_info/ani_ii_assignments.html |
| 15:14.58 | Strom_C | also, note it's ANI II, not "ani2" |
| 15:15.08 | ManxPower | MrTelephone: I just tell my telco "no collect calls, no 900 calls, no 976 calls" and then not worry about it |
| 15:15.37 | MrTelephone | 976 is another toll npa? |
| 15:15.47 | MrTelephone | some of the businesses want collect calls :-/ |
| 15:15.55 | ManxPower | MrTelephone: it is like a 900 number, but specific to the area code. |
| 15:16.01 | ManxPower | i.e. 504-976-1234 |
| 15:16.55 | MrTelephone | ani2 I said because thats what the variable is named :-/ |
| 15:17.05 | MrTelephone | so 29 is for prison and collect calls |
| 15:17.21 | Strom_C | well, whoever named the variable wasn't checking their references correctly :) |
| 15:17.52 | MrTelephone | some people lack intelligence |
| 15:18.10 | MrTelephone | one guy was accepting colelct for .20/min when he could have dialed them for .05/min |
| 15:19.30 | EricL | Any idea how come the state of a phone is perpetually Hold even though I have rebooted the phone twice and done a 'sip reload' a few times? |
| 15:19.41 | jcolp | EricL: what version of Asterisk? |
| 15:19.48 | EricL | jcolp: 1.4.8 |
| 15:20.00 | jcolp | that has been fixed |
| 15:20.21 | Swat2 | still doing it to me in 1.4.9 |
| 15:20.32 | EricL | It's not all phones, just a few of them. Most of the phones are behaving properly, its just 2 that aren't playing nice. |
| 15:20.56 | jcolp | Swat2: fixed in subversion |
| 15:21.22 | Swat2 | jcolp: know when it's scheduled for release |
| 15:21.23 | Swat2 | ? |
| 15:21.36 | jcolp | there is no schedule for it |
| 15:21.41 | EricL | But why is it happening to some phone and not others? |
| 15:21.52 | russellb | svn co http://svn.digium.com/svn/asterisk/branches/1.4 asterisk-1.4 |
| 15:21.56 | Uatec | hey, where is the mixmonitor default recording location? |
| 15:21.58 | jcolp | EricL: are they doing transfers? |
| 15:21.58 | russellb | that's how you can get it right now :) |
| 15:22.20 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 15:22.22 | EricL | jcolp: Nope, they aren't even on the phone. |
| 15:23.15 | jcolp | EricL: if you restart Asterisk they will be reset to hold, a sip reload won't do it and a reboot of the phone won't |
| 15:23.39 | *** join/#asterisk ManxPower (n=manxpowe@71-8-56-17.dhcp.leds.al.charter.com) |
| 15:23.43 | EricL | jcolp: The State should be registering sa Idle. |
| 15:24.15 | jcolp | EricL: yes, I realize that... and it's a bug that it isn't, which should be fixed in 1.4 from subversion... but I'm just telling you how to reset it to idle |
| 15:24.21 | *** join/#asterisk tbic (n=tbic@207.148.218.162) |
| 15:24.48 | EricL | jcolp: Ok. |
| 15:25.17 | nfi|ermes | afrosheen, no problem came out from make command |
| 15:25.42 | afrosheen | nfi|ermes, so it probably got built then right |
| 15:27.21 | nfi|ermes | is it possible that i should choose wich mods to enable with something about menuselect ? |
| 15:28.26 | Katty | jcolp?! |
| 15:28.35 | jcolp | Katty: !!! |
| 15:28.35 | Katty | jcolp: why are you going by jcolp? |
| 15:28.39 | afrosheen | well the module has to exist first, if you don't see a .o and a .so after it built...then it didn't build, because even a make install will leave those files there and only copy them to the target dir. |
| 15:28.46 | jcolp | Katty: why not? |
| 15:28.50 | Katty | jcolp: you're file :< |
| 15:28.57 | Katty | jcolp: :<<< |
| 15:28.58 | jcolp | Katty: this is true |
| 15:29.02 | jcolp | I am also jcolp |
| 15:29.07 | Qwell[] | no you aren't |
| 15:29.08 | Katty | clearly. |
| 15:29.18 | Katty | but file is more fun |
| 15:29.28 | Katty | or at least more framilier :P |
| 15:29.41 | Corydon76-work | framilier? |
| 15:29.58 | [TK]D-Fender | Corydon76-work: Exactry! |
| 15:30.11 | Katty | familiar :> |
| 15:30.11 | Strom_C | that sounds like the French word for someone who makes picture frames |
| 15:30.17 | nfi|ermes | here's the output of my make command |
| 15:30.21 | nfi|ermes | http://pastebin.com/m1d478800 |
| 15:30.25 | Katty | Corydon76-work: i do tend to typo a lot :P |
| 15:30.37 | Corydon76-work | orly? |
| 15:30.43 | Katty | yesrly |
| 15:31.10 | afrosheen | nfi|ermes, what...did you do make menuselect? this is new to me |
| 15:31.45 | nfi|ermes | nothing, perhaps for for me too :D |
| 15:31.48 | Corydon76-work | That nick reminds me of Ralph Fiernes |
| 15:32.19 | Katty | afrosheen reminds me of eddie murphy in those cop movies. |
| 15:32.31 | Katty | banana in the tail pipe movie |
| 15:32.34 | afrosheen | I'm so dumb, I was playing Castlevania Symphony of the Night and near the end of the game realized that the main character, Alucard, is Dracula backwards |
| 15:32.36 | Katty | which i can't think of ^_- |
| 15:32.42 | Dan0maN_Work | bev hills cop |
| 15:32.48 | Katty | oh yeah. |
| 15:32.49 | datachomper | Do the laugh Eddie. |
| 15:32.58 | Yourname` | ~sipnat |
| 15:32.58 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 15:33.38 | Dan0maN_Work | more like coming to america (also with eddie murphy), and the "soul glow" product ;) |
| 15:33.50 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 15:33.51 | *** join/#asterisk Trevor_b (n=tbenson@69.12.220.201) |
| 15:33.55 | *** join/#asterisk SwK (n=SwK@63.96.55.2) |
| 15:34.00 | Katty | SwK: ! |
| 15:34.33 | Katty | any ideas for lunch? |
| 15:34.55 | afrosheen | Dan0maN_Work, yeah, you nailed it |
| 15:34.56 | Qwell[] | Katty: steak - and lots of it |
| 15:35.06 | Katty | Qwell[]: that's not very good for keeping my girlish figure. |
| 15:35.08 | afrosheen | the scene with the grease spots on the sofa = classic |
| 15:35.15 | Qwell[] | Katty: steak - and moderate amounts of it |
| 15:35.16 | Katty | Qwell[]: there's also that.. steak part |
| 15:35.29 | Katty | i don't like steak, it's icky. |
| 15:35.36 | Qwell[] | fine, carrots |
| 15:35.37 | [TK]D-Fender | Katty: Moooooooooooooooooooooooooooooo |
| 15:35.37 | Qwell[] | :p |
| 15:35.51 | afrosheen | nfi|ermes, check your build directory again, I see nothing related to mysql in your pastebin stuff |
| 15:36.07 | datachomper | Anybody here work at Digium, know Ryan Brindley? |
| 15:36.13 | Qwell[] | datachomper: yeah |
| 15:36.14 | [TK]D-Fender | Katty: I already take my Filet Mignon blue & seared.... and a girl I'm hooking up with likes hers tartar.... I will be bold :) |
| 15:36.40 | Qwell[] | datachomper: why? |
| 15:37.11 | Katty | [TK]D-Fender: ! |
| 15:37.26 | Katty | fender and a GIRL!? |
| 15:37.42 | afrosheen | Katty, tequila is a wonder drug |
| 15:37.51 | [TK]D-Fender | Katty: You've seen my dashing self..... don't act so surprised ;) |
| 15:37.52 | Katty | mm, tequila. |
| 15:37.58 | Katty | [TK]D-Fender: true, true. |
| 15:38.05 | Katty | [TK]D-Fender: /clap. |
| 15:38.35 | Katty | afrosheen: tequila isn't so 'wonder' drug on me... it just sorta makes me sick after awhile :/ |
| 15:38.41 | Swat2 | jcolp: that SVN version fixed the prob - ta |
| 15:38.49 | Katty | afrosheen: but now, RUM... |
| 15:38.54 | afrosheen | Katty, yeah but it probably puts you in the sweet spot for 30 minutes |
| 15:38.54 | Katty | afrosheen: rum is fun ^_^ |
| 15:39.02 | Katty | afrosheen: not really. |
| 15:39.11 | Katty | afrosheen: i go from 1 margarita to... ugah, i knew i shouldn't have had that. |
| 15:39.19 | afrosheen | Katty, I guess you haven't spent much time in Texas :) |
| 15:39.22 | Katty | afrosheen: and that's what.. a shot? (= |
| 15:39.23 | Dan0maN_Work | heh |
| 15:39.23 | MrTelephone | datachomper, who do you work for? |
| 15:39.44 | Katty | afrosheen: i've never been to texas :P |
| 15:39.46 | MrTelephone | where is the next asterisk convention in toronto |
| 15:39.56 | MrTelephone | in canada i mean |
| 15:39.56 | Katty | afrosheen: tho i hear it's uberpretty :> |
| 15:40.01 | Katty | afrosheen: and good for shopping :> |
| 15:40.01 | MrTelephone | nerd convention :-/ |
| 15:40.03 | Dan0maN_Work | mexican martini's for me ;) |
| 15:40.12 | afrosheen | Katty, it's a great place to get drunk ;) |
| 15:40.14 | Katty | ugah, martinis. |
| 15:40.15 | *** join/#asterisk NoIdea465 (n=NoIdea46@d53-64-116-212.nap.wideopenwest.com) |
| 15:40.21 | Katty | pure vodka and olive juice... *shudder* |
| 15:40.50 | Katty | afrosheen: i can get drunk watching star trek every night :P |
| 15:40.52 | Dan0maN_Work | no. /mexican/ martini's. no vodka. tequila. |
| 15:40.57 | Katty | afrosheen: that's a great way to get drunk :P |
| 15:41.11 | afrosheen | Katty, hahaha classic |
| 15:41.13 | Katty | Dan0maN_Work: oh great. so instead of being able to mostly tolerate it, i get stuck being sick :P |
| 15:41.41 | Katty | my star trek collection is... huge. |
| 15:41.50 | Katty | and by huge i mean taller than me. |
| 15:41.55 | Dan0maN_Work | heh. jeager then? that's my poison really. |
| 15:42.00 | Katty | [TK]D-Fender: no comments out of you about my height. |
| 15:42.09 | *** join/#asterisk sketc1 (i=ryan@nat/digium/x-6e56e83d254977f5) |
| 15:42.24 | Katty | omgjager. |
| 15:42.28 | Katty | we named our puppy after jager! |
| 15:42.37 | MrTelephone | googletalk speaks with asterisk? |
| 15:42.38 | anonymouz666 | anyone in here already use SRTP with *? |
| 15:42.39 | MrTelephone | thats waesome |
| 15:42.41 | afrosheen | Katty, does he taste disgusting also? |
| 15:42.50 | Katty | afrosheen: pfft. no. he's sickeningly sweet ^_^ |
| 15:43.11 | Katty | afrosheen: you don't like licorice, do you |
| 15:43.25 | afrosheen | Katty, no but for some reason everyone's grandma does |
| 15:43.44 | Dan0maN_Work | Katty: hated it growing up. now if it gets me gone, i'm in ;) |
| 15:43.58 | MrTelephone | as if all the cool new stuff is for asterisk 1.4 |
| 15:44.02 | MrTelephone | thats crappy |
| 15:44.06 | Katty | Dan0maN_Work: haha |
| 15:44.34 | Katty | afrosheen: i think licorice has a way of growing on you. |
| 15:44.47 | Katty | afrosheen: kinda like brocolli. |
| 15:44.49 | afrosheen | Katty, so does fungus, but I try to scrub my feet once a day |
| 15:44.52 | afrosheen | ;) |
| 15:45.01 | *** join/#asterisk zuesman (n=pb0036@dsl017-122-055.mci1.dsl.speakeasy.net) |
| 15:45.15 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:45.18 | Katty | i wish this debian cd would download faster. |
| 15:45.26 | Strom_C | yell at the bits |
| 15:45.49 | Katty | now why didn't i think of that?! that will surely speed things up! |
| 15:46.03 | MrTelephone | get a faster connection |
| 15:46.07 | Uatec | put butter on the network cables |
| 15:46.10 | Dan0maN_Work | i hear a lot of debian talk here. is that the norm? i had planned to test out * on centos |
| 15:46.21 | afrosheen | Dan0maN_Work, it's wonderful on centos also |
| 15:46.36 | MrTelephone | its wonderful on windows 95 |
| 15:46.36 | Dan0maN_Work | k. thanks |
| 15:46.38 | Katty | Dan0maN_Work: i just use debian cause i'm most familiar with it. |
| 15:47.02 | afrosheen | Katty, I just gave debian another chance for the 18th time and this time it's working so far :) |
| 15:47.16 | afrosheen | up until this point I'd used Centos and RHEL exclusively |
| 15:47.28 | Katty | afrosheen: yeah, i've had a couple problems with it before. mostly before asterisk ever gets downloaded |
| 15:47.37 | afrosheen | Katty, exactamundo |
| 15:47.46 | MrTelephone | debian has too many updates all the time |
| 15:47.52 | Dan0maN_Work | heh |
| 15:47.52 | MrTelephone | then u accidently do a kernel update |
| 15:48.01 | Katty | hehe |
| 15:48.01 | MrTelephone | and ur fucked when the power goes out |
| 15:48.13 | Katty | MrTelephone sounds like a war vet! |
| 15:48.15 | MrTelephone | because it boots up and the zaptel drivers don't work because the kernel is the wrong version |
| 15:48.15 | afrosheen | yeah you reboot into a new kernel with no sangoma/zaptel drivers |
| 15:48.22 | Dan0maN_Work | i like centos. it's YUMmy |
| 15:48.27 | afrosheen | I've had that happen before...once |
| 15:48.27 | MrTelephone | yeah that gets me all the time |
| 15:48.28 | MrTelephone | hah |
| 15:48.40 | afrosheen | until I got a brain and added the kernel to the exclude list |
| 15:48.43 | Katty | centos! the bash maker! |
| 15:48.48 | afrosheen | haha |
| 15:48.50 | MrTelephone | there is an exclude list? right on |
| 15:49.00 | afrosheen | there's an exclude list for most distros |
| 15:49.02 | MrTelephone | but then you think.. maybe i'll get some more irq room with the new version :-/ |
| 15:49.05 | Katty | i never knew there was an exclude list either |
| 15:49.13 | Katty | i didn't know that updates went on their own, even |
| 15:49.20 | MrTelephone | they don't |
| 15:49.24 | Katty | it tells me all the time in the gui that there's some waiting |
| 15:49.25 | MrTelephone | but I like using dselect |
| 15:49.27 | afrosheen | yeah they don't |
| 15:49.29 | Katty | and i always go through them. |
| 15:50.00 | Dan0maN_Work | i guess before i knock it, i should actually try deb. never touched it |
| 15:50.27 | Dan0maN_Work | been a redhat slut since 6.2 |
| 15:50.34 | datachomper | debian > all, imho |
| 15:50.36 | afrosheen | Dan0maN_Work, it's more difficult to work with than centos/rhel but it has its merits |
| 15:50.54 | *** join/#asterisk ToyMan (n=Stuart@72.168.167.241) |
| 15:51.33 | Katty | the reason i like debian so much is because i was using mandrake. |
| 15:51.37 | afrosheen | brb, boring ole meeting time |
| 15:51.44 | Katty | and despite the fact that mandrake has a billion and one wizards, they never worked... |
| 15:51.52 | Dan0maN_Work | heh |
| 15:51.55 | Katty | so someone suggested debian, then i learned about apt-get, and.. yeah |
| 15:51.56 | datachomper | Lol, true that. |
| 15:52.08 | [TK]D-Fender | CenotOS & Slackware = instant success for me... |
| 15:52.19 | datachomper | I cheated on Debian with Gentoo for a while. I'm not proud of it, but I learned my lesson. |
| 15:52.26 | Katty | [TK]D-Fender: how does that go over with the ladies? |
| 15:52.26 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 15:52.35 | Katty | i've heard good things about gentoo.... |
| 15:52.39 | Swat2 | the new deb version's pretty nice :) |
| 15:52.47 | Katty | but also that it's kinda hard. i'm not ready for too hard yet. |
| 15:52.52 | Katty | Swat2: yeah, it is. |
| 15:52.57 | *** join/#asterisk NamNT_ (n=NamNguye@cm246.delta196.maxonline.com.sg) |
| 15:52.58 | Katty | Swat2: the install is a lot easier too |
| 15:53.04 | Swat2 | heh |
| 15:53.09 | [TK]D-Fender | Katty: How does what go over? |
| 15:53.12 | NamNT_ | hi |
| 15:53.33 | Katty | [TK]D-Fender: that instant slackware/centos success. |
| 15:53.39 | Katty | NamNT_: hi (= |
| 15:53.54 | Yourname` | Hello. Outbound box is not firewalled, neither is inbound box. Just the agents are behind NAT. The call is placed from outbound box, and if the other party has answered the call the call is sent via sip to the inbound box, where there is an agent... now NAT seems to be a problem. The called person can hear the agent, but the agent cannot hear the other person. I've checked the NAT documentation, a |
| 15:53.55 | Yourname` | nd it's exactly the way it should be. Yet, it's giving me problems. The chart is at http://img260.imageshack.us/my.php?image=chartjj0.jpg |
| 15:53.58 | [TK]D-Fender | Katty: The plug-ins are functional, but the system is somewhat unstable ;) |
| 15:54.00 | Swat2 | i've been using Debian since "Rex" (1.2) |
| 15:54.06 | NamNT_ | where can i find cheap digium cards? |
| 15:54.08 | Yourname` | Outbound box sip.conf http://pastebin.ca/640672 |
| 15:54.16 | Yourname` | Inbound box sip.conf http://pastebin.ca/640668 |
| 15:54.20 | Katty | [TK]D-Fender: heh |
| 15:54.23 | [TK]D-Fender | Yourname` : Read the following |
| 15:54.25 | [TK]D-Fender | ~sipnat |
| 15:54.25 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 15:54.28 | *** join/#asterisk zdsl (i=aguilar@nat/google/x-a3c30cc7e1623ef8) |
| 15:54.30 | zdsl | hello |
| 15:54.44 | zdsl | wow this is a big channel |
| 15:54.54 | Yourname` | [TK]D-Fender: I did.. since last night. And everything is according to that, trust me. Please look at the sip.confs if you think I'm lieing. |
| 15:55.01 | Swat2 | very observant, zdsl. |
| 15:55.06 | Yourname` | I just don't know what logic it is.. |
| 15:55.39 | [TK]D-Fender | Yourname`: remove every commented out line PERMANENTLY and repastebin |
| 15:56.04 | Katty | zdsl: a bit :P |
| 15:57.19 | [TK]D-Fender | Yourname`: your outbound box does not have a localnet setting. And reinvites have NOT been disabled. |
| 15:57.32 | [TK]D-Fender | Yourname`: You're ability to follow those directions seems rather poor |
| 15:57.35 | MrTelephone | hahah |
| 15:57.45 | Katty | [TK]D-Fender: simmer down now |
| 15:57.49 | [TK]D-Fender | your* |
| 15:58.03 | Yourname` | [TK]D-Fender: Does it need a localnet? Because both the boxes have public IPs and are not behind any firewall. |
| 15:58.17 | MrTelephone | whats with this IVR database crap im reading on the web |
| 15:58.19 | Katty | fender's turning into a bitter old man. |
| 15:58.20 | lirakis | "Yourname`: I just don't know what logic it is.." .. lol |
| 15:58.22 | [TK]D-Fender | Katty: Simmer is still pretty hot... I'm not boiled over yet ;) |
| 15:58.25 | MrTelephone | an IX does lookup requests on the AIN? |
| 15:58.33 | MrTelephone | how do you do that with asterisk :-/ |
| 15:58.55 | [TK]D-Fender | Yourname`: it should. Please clean them both up and re-pastebin. |
| 15:59.00 | Katty | the things i put up with.. |
| 15:59.04 | Yourname` | One sec. |
| 15:59.08 | Swat2 | Yourname` hasnt learnt that what [TK]D-Fender should be taken as law, and adheared to as that. |
| 15:59.15 | MrTelephone | tkfender, whats the grep command for that |
| 15:59.24 | Yourname` | Swat2: I already do know that, lol ;) |
| 15:59.24 | Swat2 | + says |
| 15:59.26 | MrTelephone | how do you negate /^;.*/ |
| 15:59.28 | [TK]D-Fender | "I *AM* the law!" - Judge Dredd |
| 15:59.32 | Yourname` | [TK]D-Fender: Outbound http://pastebin.ca/640679 |
| 16:00.21 | zdsl | Has anyone used asterisk for telephone conferencing and video? |
| 16:00.40 | [TK]D-Fender | Yourname`: register => 1012@64.34.139.97 is out or order. it has to be AFTER all your other parm declarations. You have NOT DISABLED REINVITES either. And you should have a Localnet |
| 16:00.53 | Katty | zdsl: i'm getting ready to start on it :> |
| 16:01.12 | *** join/#asterisk JSegatto (n=pirc@225.89.203.68.cfl.res.rr.com) |
| 16:01.18 | JSegatto | is there a place to get default sounds for asterisk? I don't see any in the /var/lib/asterisk folder |
| 16:01.26 | zdsl | Katty: can we exchange ideas on this |
| 16:02.01 | Katty | zdsl: umm, sure. |
| 16:02.05 | Katty | zdsl: but i'd like to keep mine. |
| 16:02.08 | Katty | zdsl: :> |
| 16:02.16 | *** part/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 16:02.37 | Yourname` | [TK]D-Fender: Inbound http://pastebin.ca/640684 |
| 16:03.07 | lirakis | Yourname: you aslo mispelled disallow |
| 16:03.37 | lirakis | Yourname: diallow=all -> disallow=all |
| 16:03.40 | MrTelephone | cat sip.conf | grep "!^;*" |
| 16:03.42 | MrTelephone | whats that do |
| 16:04.27 | *** part/#asterisk NamNT_ (n=NamNguye@cm246.delta196.maxonline.com.sg) |
| 16:04.44 | zdsl | Katty: I am researching tarting today what is commercial and opensource solutions |
| 16:04.51 | Yourname` | whoa |
| 16:04.56 | lirakis | MrTelephone: that doesnt do anything ... but it looks like its meant to be a regular expression for grep... |
| 16:05.06 | MrTelephone | how do you grep all lines not beginning with ; |
| 16:05.13 | MrTelephone | its nto working :-/ |
| 16:05.23 | fetcher | MrTelephone: grep -v ^\; |
| 16:05.57 | MrTelephone | good point |
| 16:06.24 | MrTelephone | whats what Yourname needed |
| 16:06.55 | [TK]D-Fender | Yourname`: and you are still mis-ordering [general] Register has to come at the END. |
| 16:07.06 | MrTelephone | whose got a british cell phone exchange? |
| 16:07.11 | MrTelephone | its 40 cents a minute there |
| 16:07.16 | Yourname` | [TK]D-Fender: And about the localnet, I was following #6 from http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions , because both the boxes are on public IPs and not in a LAN. The register was put in right where sip.conf's original example was. |
| 16:07.17 | MrTelephone | isn't that a bit high? |
| 16:07.18 | [TK]D-Fender | Yourname`: And you put nat=yes yet put no externip, etc... |
| 16:07.29 | [TK]D-Fender | Yourname`: read the FIRST guide : |
| 16:07.34 | [TK]D-Fender | ~sipnat |
| 16:07.35 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 16:07.35 | wg_ | US$0.40 or 40p? |
| 16:07.39 | MrTelephone | us .40 |
| 16:07.39 | Yourname` | ok |
| 16:07.46 | wg_ | seems cheap |
| 16:07.47 | MrTelephone | british land line is cheap |
| 16:07.49 | wg_ | shrug |
| 16:07.52 | *** join/#asterisk Keltus (n=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 16:07.53 | MrTelephone | why does that seem cheap? |
| 16:07.57 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 16:07.57 | wg_ | british land line is expensive |
| 16:08.01 | MrTelephone | its 2 cents to call north america |
| 16:08.03 | wg_ | well, its timed |
| 16:08.09 | MrTelephone | british land line is 13 cents or something |
| 16:08.16 | MrTelephone | just take vonage for example |
| 16:08.16 | wg_ | 13 cents a minute? |
| 16:08.29 | *** join/#asterisk ptblank (n=MURDER1@cpe-75-84-221-219.socal.res.rr.com) |
| 16:08.40 | wg_ | i pay AU$0.15 for untimed national calls |
| 16:08.47 | wg_ | to any land line in australia |
| 16:08.58 | wg_ | mobiles are a lot more expensive |
| 16:09.11 | *** part/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 16:09.11 | MrTelephone | if you call mobile in canada its free though |
| 16:09.11 | wg_ | more like AU$0.60/min i think |
| 16:09.13 | coppice | I can call my mum in London for less than it costs her to talk to her friends nearby :-\ |
| 16:09.18 | MrTelephone | if the mobile is in the same area code |
| 16:09.25 | wg_ | oh, weird |
| 16:09.30 | MrTelephone | i know |
| 16:09.31 | wg_ | go canada ;] |
| 16:09.37 | MrTelephone | the cell companies in europe are scamming people bad |
| 16:10.00 | wg_ | different market |
| 16:10.15 | wg_ | just like people think bandwidth in australia is extraordinarily expensive. but its an entirely different market. |
| 16:10.30 | MrTelephone | but australia has an excuse |
| 16:10.40 | MrTelephone | its far from any major continent |
| 16:10.50 | MrTelephone | like north america or europe |
| 16:10.54 | wg_ | yeah and the city i live in is 3000km from the next city :P distance is a bitch |
| 16:10.56 | *** join/#asterisk ashd (n=ashleydr@user-194-248-151-83.e7even.com) |
| 16:11.10 | Yourname` | I feel like crying now. Or I'm just blind, the first guide presents the sip.conf as a sample scenario of an asterisk server behind NAT. And one extension it is showing is behind NAT, and one isn't.. same way in the outbound box sip.conf, I put the extension as not behind NAT. |
| 16:11.15 | MrTelephone | theres a land -> cell premium in britain and I can't figure out why |
| 16:11.36 | Yourname` | I've also set canreinvite=no in both sip.confs |
| 16:11.52 | MrTelephone | they actually market channel banks with sim cards for dialing out on cell because it doesn't cost cell -> cell |
| 16:12.01 | MrTelephone | pretty cool stuff |
| 16:12.09 | wg_ | whack |
| 16:12.28 | [TK]D-Fender | Yourname`: you did it AFTER your register statment. fix the ORDER and the rest of your parms, localnet, etc. |
| 16:12.42 | Yourname` | ok |
| 16:12.44 | rbooz | If I rarely get 100% with ztteat (although pretty consistently 99.987793%) could that in any way be a cause for Zaptel problems giving random hangups (on-hook event) in the middle of a call? |
| 16:12.46 | [TK]D-Fender | Yourname`: And stop thinking you can just shove things anywhere you want |
| 16:12.50 | Yourname` | lol |
| 16:13.00 | Yourname` | I didn't though! I was just following the order of examples in sip.conf |
| 16:13.11 | [TK]D-Fender | Yourname`: * is like a restaurant. Likely to cause indigestion, and its all about LOCATION, LOCATION, LOCATION, <------------- |
| 16:13.20 | Swat2 | lol |
| 16:13.31 | [TK]D-Fender | Yourname`: No-one puts register in the MIDDLE of their [general] section. |
| 16:15.24 | Yourname` | You know.. I'm first going to remove the damn example lines. |
| 16:18.04 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 16:18.18 | Yourname` | Ok, now the register is at the end of the [general] |
| 16:18.27 | Yourname` | On the outbound. Now it's time for the inbound. |
| 16:19.00 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 16:19.03 | nettie | uhmm |
| 16:19.30 | nettie | I'm runnign asterisk 1.2.17 how do I know which addon version I should use? the latest for 1.2.x brnach is supposed to be compatible? |
| 16:20.26 | [TK]D-Fender | nettie: Check the changelogs for the release dates. |
| 16:20.52 | generalhan | hey all ... i dont know if this is an issue at all but something different started happening today when using "safe_asterisk" to start * ... 'ps' shows 2 asterisk proccesses running is this normal? http://generalhan.pastebin.ca/640701 |
| 16:22.28 | generalhan | normally it would only show one process -- the safe_asterisk one |
| 16:23.41 | Yourname` | Done! |
| 16:23.45 | Yourname` | Now it's still not working. |
| 16:23.58 | Yourname` | Register is the last line in the [general] |
| 16:24.06 | Yourname` | canreinvite is disabled. |
| 16:24.19 | *** join/#asterisk marcan (i=1337@10.Red-88-24-250.staticIP.rima-tde.net) |
| 16:24.20 | Yourname` | Let me do another pastebin for both. |
| 16:24.38 | MrTelephone | 16 dollars a minute to phone this place Inmarsat West Atlantic - Mini M |
| 16:25.17 | nettie | [TK]D-Fender thax |
| 16:26.40 | wg_ | inmarsat is about $3.50/min here from memory |
| 16:26.57 | *** join/#asterisk jacky59 (n=marc@jacky62.maxiscreen.com) |
| 16:27.03 | jacky59 | hello there; |
| 16:27.05 | MrTelephone | thats retarded |
| 16:27.14 | mvanbaak | MrTelephone: here in .nl we have free cell -> cell calls as well |
| 16:27.22 | Yourname` | [TK]D-Fender: New sip.confs. Outbound http://pastebin.ca/640714, and inbound http://pastebin.ca/640712 |
| 16:27.25 | mvanbaak | when all cellphones are in the same business account |
| 16:27.28 | wg_ | i get free cell->cell calls to other cells on my phone network |
| 16:27.29 | jacky59 | got a small question regarding digitmap :/ |
| 16:27.40 | wg_ | so all calls between Hutchison-3G are free |
| 16:27.42 | *** join/#asterisk qdk (n=qdk@193.164.155.10) |
| 16:27.44 | mvanbaak | ah |
| 16:27.49 | MrTelephone | mvanbaak, nice, did you figure out that courier auth daemon yet? |
| 16:27.52 | jacky59 | i'm having an mgcp phone somewhere that is sending digit after digit the number dialed |
| 16:28.05 | mvanbaak | we have a voiceblue to call our coworkers for free |
| 16:28.09 | jacky59 | due to high latency, it's missing some digits sometimes if I do not wait |
| 16:28.19 | mvanbaak | MrTelephone: nope, not yet |
| 16:28.37 | MrTelephone | im trying to figure out my collect call situation |
| 16:28.38 | jacky59 | if I want to create a digitmap: should it be done on the asterisk server or on the mgcp client ? |
| 16:28.56 | [TK]D-Fender | Yourname`: You are still failing to list your localnets, also what have you forwarded to *? |
| 16:29.12 | Hmmhesays | anyone futz around with the share line settings on an spa-942? |
| 16:29.32 | Yourname` | Ok, pardon me for being so ignorant. Localnet, I don't have one. Both the boxes are two different public IP boxes. |
| 16:29.42 | Yourname` | Can you please explain to me how I'd need it? |
| 16:29.59 | Yourname` | And still, what would I put there? 192.168.1.1? There isn't anything like that at all. |
| 16:30.24 | Yourname` | Some other developer also did this similar thing, but I don't see localnet being defined in any of his confs, I'm simply trying to copy and understand. |
| 16:30.47 | *** join/#asterisk Strom_M (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 16:31.54 | Yourname` | I've forwarded the outbound call made by the outbound asterisk via sip extension 1012 to another asterisk inbound box. |
| 16:32.06 | jacky59 | nobody knows for my digitmap where I have to set it up ? |
| 16:33.12 | *** join/#asterisk el_critter (n=chatzill@190.74.124.49) |
| 16:34.34 | Yourname` | [TK]D-Fender: Even in the first link of ~sipnat, it shows that localnet is a local area network address. |
| 16:34.35 | *** join/#asterisk levi_home (n=levi@levi.dsl.xmission.com) |
| 16:35.27 | [TK]D-Fender | Yourname`: it is your loca network & mask. |
| 16:35.52 | [TK]D-Fender | Yourname`: I was reffering to PORTS. what PORTS have you forwarded to *? |
| 16:36.04 | Yourname` | They are all open. No firewall at all. |
| 16:37.39 | *** part/#asterisk IPmonger (n=ipmonger@c-71-224-31-246.hsd1.pa.comcast.net) |
| 16:37.45 | jacky59 | where should you define digitmaps for phones: |
| 16:37.54 | jacky59 | is it on the phone itself, or on the asterisk ? |
| 16:38.14 | jacky59 | I thought we should define it on call agent, but can't find any options about that ? |
| 16:39.23 | *** join/#asterisk Rickalicious (n=Rickalic@c-76-29-83-134.hsd1.il.comcast.net) |
| 16:39.35 | Yourname` | [TK]D-Fender: They are all open. No firewall at all. |
| 16:40.19 | Yourname` | Just great. |
| 16:40.23 | Yourname` | I had to reboot the boxen. |
| 16:40.29 | Yourname` | And now it works. |
| 16:40.48 | [TK]D-Fender | Yourname`: [outbound] Should also be set to "nat=no", and your allow/disallow order is bad. |
| 16:41.00 | Yourname` | ah |
| 16:41.01 | [TK]D-Fender | Yourname`: Disallow first, then allow |
| 16:41.06 | Yourname` | Gotcha! |
| 16:42.11 | Yourname` | Hmm, sip.conf looks very nice all cleaned up. |
| 16:42.41 | *** part/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 16:42.56 | generalhan | [TK]D-Fender: what distro do you run your asterisk box on ? |
| 16:43.04 | rbooz | tzafrir: let me try one last approach... :-)... does this URL potentially have any bearing on my Zaptel issue? |
| 16:43.05 | rbooz | http://lists.ckts.info/pipermail/voip/2006-February/002433.html |
| 16:43.33 | [TK]D-Fender | generalhan: Typically CentOS or Slackware. |
| 16:43.46 | generalhan | [TK]D-Fender: ok thanks |
| 16:44.20 | *** join/#asterisk CVirus (n=GoD@212.12.250.74) |
| 16:45.53 | tzafrir | rbooz, others here know digium cards better than me |
| 16:46.07 | rbooz | tzafrir: OK - thanks |
| 16:46.27 | *** join/#asterisk gcbirzan (n=gcbirzan@pida/gcbirzan) |
| 16:47.55 | rbooz | anyone - my PSTN lines randomly hangup in the middle of a call. Log shows that the line got an "on-hook" event (although neither side of the call actually did) |
| 16:48.06 | gcbirzan | Meh. I'm trying to do an AGI script and having problems using GET DATA. Same thing works with STREAM FILE, but with GET DATA I get a "no such file or directory" error. I've tried with and without the .wav, same thing: Jul 31 19:44:29 WARNING[12123]: file.c:512 ast_openstream_full: File /tmp/robot-35922da170b32e3deeef715cd5569f86-text.wav does not exist in any format |
| 16:48.13 | rbooz | a two-year old TDM400P - 4 fxo cards |
| 16:48.41 | rbooz | New server (Dell PE 840, SATA Raid 1 - hardware) and new office |
| 16:49.04 | *** join/#asterisk _ViperNetworks (n=Nitesh@66.184.39.174) |
| 16:49.05 | rbooz | **appears** to be more office then server as the old server started having the problem in this office, but had no problems in the old office |
| 16:49.24 | [TK]D-Fender | gcbirzan: in ther you clearly see it USING the exten, which is BAD |
| 16:49.33 | rbooz | but just so hard to find any consistency - some calls last for over an hour and no problems - others drop in 2-3 minutes |
| 16:49.55 | gcbirzan | Hm. |
| 16:51.08 | *** join/#asterisk _ViperNetworks (n=Nitesh@66.184.39.174) |
| 16:51.25 | jacky59 | gcbirzan, could it be a permission issue ? |
| 16:51.40 | *** join/#asterisk myiagy (n=myiagy@201.56.113.74) |
| 16:51.41 | _ViperNetworks | Hello All... any PHPAGI expert around..? |
| 16:51.58 | *** part/#asterisk zuesman (n=pb0036@dsl017-122-055.mci1.dsl.speakeasy.net) |
| 16:51.59 | gcbirzan | No, it actually works now. |
| 16:52.19 | gcbirzan | I figured, kind of, what was wrong before. I was deleting the file when my object got deleted, which was before asterisk got a chance to play it. |
| 16:52.36 | _ViperNetworks | need small help with PHPAGI... to generate call. |
| 16:52.39 | jacky59 | humm... I really need this digitmap but can't find where it is supposed to be configured on the server (on mgcp ) |
| 16:53.07 | jacky59 | could have been a permission issue gcbirzan ;) |
| 16:53.26 | gcbirzan | Ah, damn. *kicks* |
| 16:53.37 | gcbirzan | I think I need to answer first. :-P |
| 16:54.45 | jacky59 | in mgcp.conf I can't find anything about digit map |
| 16:54.53 | jacky59 | where this could be possibly detailed ? |
| 16:55.22 | _ViperNetworks | Can anyone help on how to join two calls generated by PHPAGI...? |
| 16:56.39 | Rickalicious | I am trying to write a custom context so that DID's are written to the CDr database, and then handed off to the normal ext-did context. I can get it to write the DID to the Userfield in the CDR database, but then I can't figure out how to get it to return to the Ext-did context. |
| 16:57.43 | s0ck | any misdn chaps about |
| 16:57.47 | gcbirzan | Hm, must be something with GET DATA... I changed only the command to STREAM FILE and it works. |
| 16:57.55 | s0ck | moved a working pbx from one office to another... no longer works lol |
| 16:58.21 | s0ck | both isdn |
| 16:58.21 | gcbirzan | Is GET DATA the right way to get multiple keys from the user? (I've no idea, first time playing with AGI :-P) |
| 16:58.21 | s0ck | not sure what could have changed... |
| 16:59.23 | gcbirzan | Ahhh. Apparently, I need to specify how many digits. |
| 16:59.31 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.148.87) |
| 16:59.53 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 17:00.27 | s0ck | MDL_ERROR:REQ (tei_l2) <---- mean anything? |
| 17:02.41 | JSegatto | I accidentally deleted the AMPWEBADDRESS line in amportal.conf, what is supposed to be there besides the IP address of the machine? |
| 17:04.09 | [TK]D-Fender | JSegatto: You're in the wrong channel. |
| 17:04.11 | _ViperNetworks | gcbirzan: Yes |
| 17:04.23 | Qwell[] | gcbirzan: You are teh nub. |
| 17:04.38 | jacky59 | anyone knows how to configure a digitmap on asterisk ? |
| 17:04.50 | Qwell[] | gcbirzan: also, g0ats r fun |
| 17:05.15 | _ViperNetworks | gcbirzan: this is the PHPAGI implementation $ans = $agi->get_data('beep', 6000, 1); |
| 17:05.37 | *** join/#asterisk pepse (n=pepse@71-223-121-15.phnx.qwest.net) |
| 17:05.52 | pepse | Qwell: Around? |
| 17:06.11 | pepse | Qwell[]: ? |
| 17:06.31 | Qwell[] | I'm always here.. it's just a matter of whether I'm paying attention |
| 17:06.55 | pepse | Qwell[]: I think it was you that said to let you know if I got the 7910 working with PoE with that special cable on the voip-info wiki |
| 17:07.01 | Qwell[] | yeah |
| 17:07.05 | pepse | Qwell[]: well, I did :). |
| 17:07.14 | Qwell[] | just swapped some pairs around, right? |
| 17:07.39 | pepse | yeah i think it swaps 3 4 and 7 8? .. something |
| 17:07.44 | Qwell[] | weird |
| 17:07.53 | pepse | i just followed what it said on the voip-info entry |
| 17:08.05 | pepse | regular color order on one side and a funky color order on the other |
| 17:08.06 | Qwell[] | and I assume it didn't catch fire? |
| 17:08.10 | pepse | it did not :) |
| 17:08.15 | Qwell[] | fun |
| 17:08.20 | Qwell[] | I may have to get a POE switch or something |
| 17:08.27 | pepse | normal straight through cable did nothing, so i figured the phone was fried or something |
| 17:08.35 | pepse | but when i tried a regular 48v brick on the phone, it powered on |
| 17:09.42 | gcbirzan | Qwell[]: First of all, the docs say "ah, if you don't specify, then you can press as many keys as you want" |
| 17:09.51 | Qwell[] | gcbirzan: I wasn't commenting on your issue |
| 17:09.55 | gcbirzan | Qwell[]: And, secondly. http://www.goat-trauma.org/stats.shtml |
| 17:10.05 | Qwell[] | just that - in general - you're a nub :p |
| 17:10.08 | gcbirzan | Yes, yes. |
| 17:10.16 | gcbirzan | And I've got to get this to work. :-P |
| 17:10.32 | pepse | anyone using ipkall? |
| 17:10.33 | Qwell[] | I can't believe I just went to a URL with "goat" in it |
| 17:10.41 | pepse | i suddenly get oneway audio on a machine that's set as dmz host |
| 17:12.35 | jacky59 | anyone to help me with dialplan / digitmap ? |
| 17:13.17 | gcbirzan | Qwell[]: fUN, THOUGH |
| 17:13.20 | gcbirzan | Ugh. Silly caps. |
| 17:14.10 | *** join/#asterisk Jingles (n=dfbarth@39.183.dowl.anc.borealisbroadband.net) |
| 17:14.21 | pepse | JackEStorm: i can try |
| 17:14.23 | pepse | oops |
| 17:14.25 | pepse | jacky: |
| 17:15.24 | jacky59 | ok let me describe |
| 17:15.30 | jacky59 | I got a few mgcp phones |
| 17:15.58 | jacky59 | if I do not define a specific digitmap for my dialplan, they will have problems communicating with asterisk because of the delay |
| 17:16.03 | [TK]D-Fender | jacky59: Which? |
| 17:16.08 | *** join/#asterisk ta^3 (n=tacvbo@189.146.191.75) |
| 17:16.15 | JSegatto | [TK]D-Fender: isn't that an asterisk file? |
| 17:16.17 | jacky59 | (hey user pressed 0; 200 ok; heyy user pressed 2; ...) |
| 17:16.26 | [TK]D-Fender | JSegatto: No. |
| 17:16.42 | [TK]D-Fender | JSegatto: those configs are made by FreePBX and it is NOT supported here |
| 17:16.51 | jacky59 | but untill my phone gets the 200 OK, he won't accept new digits from user.. (won't send infos) |
| 17:16.55 | JSegatto | ahh my bad, sorry |
| 17:17.09 | [TK]D-Fender | jacky59: What model of phone are you using? |
| 17:17.19 | jacky59 | various old stuff |
| 17:17.31 | jacky59 | (old prototype my company is producing) |
| 17:17.49 | jacky59 | btw, what I need is to tell asterisk to tell the phones : |
| 17:18.09 | jacky59 | you can dial 1xxx,00T,0[1-9]xxxxxxxx |
| 17:18.28 | jacky59 | but where to tell asterisk that ? that's where i'm stuck :p |
| 17:19.15 | jacky59 | in the MGCP exchange, asterisk is supposed to send the phone this information |
| 17:19.20 | [TK]D-Fender | jacky59: My first guess would be your DIALPLAN. |
| 17:20.02 | jacky59 | ok, let's imagine the one I just gave: |
| 17:20.10 | jacky59 | 4 digits starting with 1 = local calls |
| 17:20.18 | jacky59 | 00 + timeout ( 00T) |
| 17:20.29 | *** part/#asterisk Cresl1n (i=matt@nat/digium/x-e89ab7fb3886596d) |
| 17:20.30 | jacky59 | and 0[1-9]xxxxxxxx |
| 17:21.00 | jacky59 | so somewhere, in one config file, I should add: |
| 17:21.15 | jacky59 | 1xxx|00T|0[1-9]xxxxxxxx |
| 17:21.24 | jacky59 | probably in extensions.conf ? |
| 17:21.49 | *** join/#asterisk krdian_ (i=krdian@killer.radom.net) |
| 17:21.52 | krdian_ | hi |
| 17:22.25 | jacky59 | dialplan is part of extensions.conf? |
| 17:24.02 | [TK]D-Fender | jacky59: dialplan = extensions.conf |
| 17:24.15 | Yourname` | It works now, thanks [TK]D-Fender :) |
| 17:24.21 | [TK]D-Fender | Yourname`: Good to hear |
| 17:24.47 | jacky59 | ok, so what to write in extensions.conf so asterisk understands it and tells my phones what's the plan ? |
| 17:25.39 | [TK]D-Fender | jacky59: If you have to ask that you haven't even started to learn *. Go download THE book and get started |
| 17:25.41 | [TK]D-Fender | ~book |
| 17:25.41 | jbot | book is probably a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 17:26.08 | jacky59 | well, I can assure you i've been loooking for that for past 4 hours ! :( |
| 17:26.16 | *** join/#asterisk harlequin516 (n=sham@styk.net) |
| 17:26.47 | harlequin516 | How can I call a number and then dial an extension from the dialplan? (Considering that Dial, does not return until the call has terminated?) |
| 17:26.49 | jacky59 | but nothing containing the word digitmap is revelant |
| 17:28.11 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-155-117-70.red.bezeqint.net) |
| 17:29.03 | MindTheGap | we're getting lots of : |
| 17:29.04 | MindTheGap | sip_alloc: Allocating new SIP dialog for NWFmMTBlNzUwN2ZjOTI2YjVmOTY3MDViYzdhNmRlMjk. - SUBSCRIBE (No RTP) |
| 17:29.10 | [TK]D-Fender | jacky59: 4 hours? * will take many more to learn...... and NO, that word is NOT relevant |
| 17:29.15 | MindTheGap | what does that mean? |
| 17:29.20 | _ViperNetworks | Please need help with PHPAGI Call generation... |
| 17:29.30 | [TK]D-Fender | jacky59: You need to read up on how the dialplan works. Thats largely Chapeter 5 |
| 17:29.49 | jacky59 | I understood most of it, but i'm confused compare to my knowledge :p |
| 17:30.03 | Hmmhesays | There should be a global extension h |
| 17:30.11 | sweeper | _ViperNetworks: to start a call, you either need AMI or call files |
| 17:30.13 | jacky59 | I know how it works, the protocol and so on, but how to enable the said option... |
| 17:30.15 | Hmmhesays | on that always gets called no matter what context you are in |
| 17:30.35 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 17:31.39 | [TK]D-Fender | jacky59: Stop thinking in MGCP terms. This is * we're talking about here and its time to get your hands dirty. Go download a SIP soft phone to use to LEARN *'s dialplan mechanics before adding MGCP to the mix.. |
| 17:31.45 | _ViperNetworks | sweeper: thats what I do... I start a call with a ".call" file and then send the call to AGI file which will provide a menu select option to the caller... I want when the caller press 1, it should dial another number and both calls should be joined together. |
| 17:32.14 | [TK]D-Fender | jacky59: Your raw knowledge of MGCP will be of little help and the syntax is very differnt. |
| 17:32.31 | _ViperNetworks | sweeper: are you on Asterisk mailing list? |
| 17:32.46 | MindTheGap | we also get : chan_sip.c:15202 handle_request_subscribe: Got SUBSCRIBE for extension 203@ddd from 10.0.0.16, but there is no hint for that extension. And it repeats the ddd context and the ip put theres one line for every number we got listed on sip.conf |
| 17:33.01 | jacky59 | I know the syntax is different, i know what I need in terms of protocol, just "how to write it in asterisk language" |
| 17:33.07 | MindTheGap | s/put/but |
| 17:33.23 | jacky59 | right now, the config works, |
| 17:33.28 | [TK]D-Fender | MindTheGap: fix your hints / phone. |
| 17:33.50 | jacky59 | it's just that I want my phone to be given the dialplan before he sends the message to the call agent |
| 17:34.09 | [TK]D-Fender | jacky59: We cannot help you. Get off your ass and read the book. no-one wants to waste their time writing your configs for you from scratch. |
| 17:35.05 | sweeper | _ViperNetworks: only the biz one |
| 17:35.14 | jacky59 | but my config is done, my server has been running for over a year |
| 17:35.36 | jacky59 | I know the bloody chapter5 you told me already :/ |
| 17:35.44 | jacky59 | I think I miss explained |
| 17:35.51 | jacky59 | or you miss understood |
| 17:35.59 | sweeper | _ViperNetworks: eh, that sounds pretty striaght forward. just use a Read command to get the number from the user, then a gotoif to send them to a dial command |
| 17:36.01 | MindTheGap | <[TK]D-Fender>, thanks, gonna find out what hints are first...:) |
| 17:36.26 | [TK]D-Fender | jacky59: PASTEBIN is your friend. Go show us something useful |
| 17:36.28 | [TK]D-Fender | ~pb |
| 17:36.28 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 17:36.29 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 17:37.44 | ashd | i am having some problems configuring asteriskNOW - the #asterisknow channel is quiet - should i ask here? |
| 17:39.25 | [TK]D-Fender | ashd: No. |
| 17:41.01 | _ViperNetworks | sweeper: No i dont want to do that way... let me paste my scenario here |
| 17:41.23 | s0ck | MDL_ERROR:REQ (tei_l2) <---- mean anything to anyone? (misdn) |
| 17:41.25 | Katty | ugah, i'm full. |
| 17:41.30 | _ViperNetworks | sweeper: 1) At certain time the system generates a ".call" and make a call to User A. |
| 17:41.41 | _ViperNetworks | sweeper: 2) When User A picks up the phone call, system will play a menu select |
| 17:41.41 | _ViperNetworks | > option. |
| 17:41.41 | _ViperNetworks | > a) Press 1 to call your supervisor. |
| 17:41.41 | _ViperNetworks | > b) Press 2 to call your manager. |
| 17:41.41 | _ViperNetworks | > c) Press 3 to leave a voice message. |
| 17:41.59 | _ViperNetworks | sweeper: 3) When the User A press 1 to call his supervisor... The system has to |
| 17:41.59 | _ViperNetworks | > put the User A on hold and place a call to the supervisor. |
| 17:42.01 | [TK]D-Fender | _ViperNetworks: Nowhere does that require AGI. |
| 17:42.23 | _ViperNetworks | sweeper: 4) Once the supervisor picks up the call, User A has to be in session |
| 17:42.23 | _ViperNetworks | > with his supervisor. |
| 17:42.40 | *** join/#asterisk saftsack (n=saftsack@pD9E06F0E.dip.t-dialin.net) |
| 17:43.40 | _ViperNetworks | Using PHPAGI.. cant we join two channels..? |
| 17:45.26 | [TK]D-Fender | _ViperNetworks: You are complicating things... why is the caller placed on hold while the supervisor is called? |
| 17:45.43 | jacky59 | [TK]D-Fender, can you give me an additional minute , |
| 17:45.45 | jacky59 | ? |
| 17:45.51 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 17:46.00 | [TK]D-Fender | jacky59: If you've got something useful to add.... |
| 17:46.11 | _ViperNetworks | [TK]D-Fender: I am writing a IVR based notification system... |
| 17:46.18 | jacky59 | well would love to re-explain in fact because I'm pretty sure we are missunderstanding :/ |
| 17:46.26 | codefreeze | Juggie: Hmmmm. Ok. I think I fixed CDRbatch, and have it updated. |
| 17:46.27 | _ViperNetworks | [TK]D-Fender: almost all modules are done... |
| 17:46.31 | *** join/#asterisk shinao1 (n=shinao1@41.205.184.14) |
| 17:46.37 | _ViperNetworks | [TK]D-Fender: I am just stuck with this... |
| 17:47.14 | jacky59 | the dialplan here is done, works perfectly. just that I do not know how to send in mgcp format the digitmap. |
| 17:47.25 | Katty | does anyone live in vegas here? |
| 17:47.42 | Katty | i have completely non-geeky, and unrelated asterisk questions. |
| 17:47.46 | *** join/#asterisk scooby2 (n=scooby2@unaffiliated/scooby2) |
| 17:47.59 | [TK]D-Fender | jacky59: send what in MGCP format? You aren't making any sense. |
| 17:48.27 | _ViperNetworks | [TK]D-Fender: Its a part of the system... if the User A is late to login at work... IVR system will generate a call to User A saying "you are late to clock in" and then provide some menu select option... |
| 17:48.33 | mishehu | Katty: what happens in vegas, stays in vegas |
| 17:48.33 | jacky59 | *6x|*7x|2xxx|[2-9]xxxxxxxxx for example |
| 17:48.46 | Katty | mishehu: i was thinking about moving there. |
| 17:48.48 | [TK]D-Fender | jacky59: You don't send that to *. |
| 17:48.58 | [TK]D-Fender | jacky59: You are fainling to understand how * works. |
| 17:49.00 | *** part/#asterisk scooby2 (n=scooby2@unaffiliated/scooby2) |
| 17:49.12 | mishehu | Katty: you'll never be able to tell us anything then unless we live there too! |
| 17:49.14 | jacky59 | the star key ? |
| 17:49.15 | [TK]D-Fender | jacky59: * has its own dilaplan and doesn't give a hoot about your phone. |
| 17:49.21 | [TK]D-Fender | ASTERISK <------- |
| 17:49.22 | Katty | mishehu: ^_^ |
| 17:49.46 | *** join/#asterisk ccesario_ (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 17:49.51 | jacky59 | I don't use star key anyway, that's a copy paste from a web page somewhere talking about my problem |
| 17:50.14 | sweeper | _ViperNetworks: fine, so you need a larger set of options. you still don't need to put people on hold and then connect them |
| 17:50.21 | Katty | [TK]D-Fender: you're being nice this afternoon, right? |
| 17:50.21 | sweeper | just freaking dial : |
| 17:50.22 | sweeper | :v |
| 17:50.32 | Katty | [TK]D-Fender: not being snappy and sarcastic. |
| 17:50.46 | jacky59 | but there should be a command in extensions.conf or other file to put |
| 17:50.55 | sweeper | Katty: maybe if you rub his belly... |
| 17:51.00 | jacky59 | dialplan=1xxx|00T for example no ? |
| 17:51.22 | Katty | sweeper: i don't think mister fender would like me doing that. |
| 17:51.30 | _ViperNetworks | sweeper: ok fine forget about putting ppl on hold.. but how to call the supervisor and join the User A call... so they both can talk to each other. |
| 17:51.41 | sweeper | _ViperNetworks: DIAL |
| 17:51.43 | sweeper | DIAL |
| 17:51.46 | s0ck | hmm |
| 17:51.49 | anonymouz666 | hi Katty |
| 17:52.05 | Katty | anonymouz666: :> |
| 17:52.06 | s0ck | just changed te_ptmp to te_ptp and i can make a call in... |
| 17:52.29 | ManxPower | jacky59: The phone dialplan determines what digits are allowed to be sent to asterisk and when the phone should send the block of digits to the server. It does NOT send the dialplan to asterisk, just the digits that matched the phone's dialplan. Then the digits must match ASTERISK'S dialplan,. |
| 17:53.04 | jacky59 | ManxPower, yes |
| 17:53.06 | jacky59 | agreed |
| 17:53.16 | [TK]D-Fender | jacky59: NO. I am not talking about the "star key". * = ASTERISK. and you do NOT send "*6x|*7x|2xxx|[2-9]xxxxxxxxx" to ASTERISK. |
| 17:53.18 | jacky59 | and we have no way from the asterisk to force the phone dialplan ? |
| 17:53.40 | [TK]D-Fender | jacky59: that is your phone's job. It has ITS dialplan, Asterisk has its OWN. |
| 17:54.16 | ManxPower | So if the dialplan on your phone is *6x|*7x|2xxx|[2-9]xxxxxxxxx and you try to dial "15551212" the phone will reject that asterisk won't even know you did anything. If you dial "5045551212" then the phone will allow that and send "5045551212" as one block of digits to asterisk. |
| 17:54.33 | _ViperNetworks | sweeper: Can I use PHPAGI? |
| 17:54.41 | jacky59 | RFC: 3435: chapter 2.1.5 digitmaps |
| 17:54.47 | ManxPower | now, you must still have an exten => line to match 5045551212 in extensions.conf |
| 17:55.16 | jacky59 | "The Call Agent can ask the gateway to collect digits dialed by the |
| 17:55.17 | jacky59 | <PROTECTED> |
| 17:55.24 | [TK]D-Fender | jacky59: http://www.voip-info.org/wiki/view/Asterisk+config+mgcp.conf |
| 17:55.31 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 17:55.44 | ManxPower | jacky59: NOTHING about RFC3435 applies to Asterisk or SIP, EXCEPT for the digit map patterns do apply to Polycom phones. |
| 17:55.48 | [TK]D-Fender | jacky59: Read this. This is what you can configure for MGCP in Asterisk. Your phone has its job and * MGCP support sucks. |
| 17:55.59 | ManxPower | jacky59: SIP is not MGCP. MGCP sends the digits to asterisk as you dial them, SIP does not. |
| 17:56.14 | jacky59 | yes I know ManxPower ... |
| 17:56.21 | ManxPower | Which protocol are you using? |
| 17:56.25 | [TK]D-Fender | ManxPower: And that is merely what the PHONE will do with whats dialed and has nothing to do with what * will accept. Yup. |
| 17:56.33 | jacky59 | mgcp |
| 17:56.41 | ManxPower | ah. You poor sod. |
| 17:56.43 | [TK]D-Fender | ManxPower: He's developing an MGCP phone |
| 17:56.49 | coppice | MGCP was an attempt to be dumber than even SIP |
| 17:56.54 | ManxPower | [TK]D-Fender: Cool! I needed a good laugh today. |
| 17:56.58 | jacky59 | i'm not developping mgcp phone, that's an old stuff |
| 17:56.59 | [TK]D-Fender | coppice: SUCK-cess! |
| 17:57.08 | ManxPower | coppice: Actually I love the IDEA of MGCP. |
| 17:57.16 | *** join/#asterisk sakic (n=sakic@cpe-071-075-118-121.carolina.res.rr.com) |
| 17:57.21 | *** join/#asterisk techie (n=gus@routedsystems.com) |
| 17:57.24 | jacky59 | mgcp idea wasn't bad at all.. |
| 17:57.27 | ManxPower | coppice: It's just that the implementation and vendor support (phones and asterisk) is terrible. |
| 17:57.33 | jacky59 | still widely used in massive deployments |
| 17:57.34 | coppice | the gateways and switches idea is fine. its MGCP that sucks |
| 17:57.39 | *** part/#asterisk rbooz (n=ryan@66.28.136.226) |
| 17:58.35 | jacky59 | so basically if I understood well, I can't tell asterisk to tell the mgcp phones that the dialplan is .....(whatever) |
| 17:58.35 | ManxPower | I tried MGCP when I was first using Asterisk. I realized it was neither in my best interest, nor my customer's best interest to use a technology that was poorly supported across the board. |
| 17:58.51 | ManxPower | jacky59: that is correct. |
| 17:58.58 | jacky59 | oh :( |
| 17:59.28 | jacky59 | I should have used a cirpack ( :p ) |
| 17:59.43 | ManxPower | you should be able to use whatever provisioning features of the phone to do that. |
| 18:00.08 | jacky59 | I can do that too, just that expected we could do it from the protocol directly |
| 18:00.28 | ManxPower | jacky59: because the phone sends every digit to asterisk as it is dialed, you might just be able to do x. in your phone dialplan |
| 18:00.31 | *** join/#asterisk sweeper (i=sweeper@softcheese.net) |
| 18:00.33 | [TK]D-Fender | jacky59: * does not configure your phones. Thats YOUR job. |
| 18:00.40 | ManxPower | then the phone will allow all digits to be dialed and asterisk can do what it does. |
| 18:01.04 | sweeper | _ViperNetworks: sorry about the discon |
| 18:01.06 | sweeper | anyways |
| 18:01.31 | jacky59 | [TK]D-Fender, in fact, you use the T for timeout |
| 18:01.35 | jacky59 | for example: |
| 18:01.47 | jacky59 | 1xxx immediately dial the number (example 1234) |
| 18:02.08 | sweeper | if you use a call file, in the dialplan, it looks as tho User A had dialed the extension and context you speficy in the dialplan |
| 18:02.16 | *** join/#asterisk AdamB0122 (n=Adam@207.200.28.175) |
| 18:02.21 | jacky59 | but if you have 00T, you dial 001234567890 it will wait 3( or whatever 5 ?) seconds before sending it back to CA |
| 18:02.24 | *** join/#asterisk cullenincrease (n=cp@c-75-64-44-200.hsd1.tn.comcast.net) |
| 18:02.25 | cullenincrease | hey |
| 18:02.52 | cullenincrease | how do i set up chanspy where i enter an extension to connect, then be able to enter the extension of a member to listen in on them? |
| 18:03.04 | sweeper | so once you've done the ivr bits that do "press 1 for blah", you somehow obtain the destination number, and just dial the phone |
| 18:03.12 | cullenincrease | i've tried exten => 556,1,ChanSpy(Member/) but that didnt work |
| 18:03.18 | jacky59 | after for the rest not matching a specific line of the dialplan, you will still send digits by digits |
| 18:04.22 | jacky59 | shame this can't be configured in * |
| 18:05.28 | jacky59 | ok, then i'll move to sip this phone |
| 18:05.37 | ManxPower | cullenincrease: Member/ is not a channel |
| 18:06.27 | _ViperNetworks | sweeper: No worries... got it working... using DIAL |
| 18:06.35 | sweeper | ^_^ |
| 18:06.50 | cullenincrease | eh? |
| 18:06.56 | cullenincrease | so would i use SIP/ then? |
| 18:07.19 | _ViperNetworks | sweeper: PHPAGI has a function to Dial and join the calls. |
| 18:07.27 | skymeyer | morning, any1 experience with SNOM phones ? |
| 18:07.33 | _ViperNetworks | sweeper: Thanks once again.. to put some light. |
| 18:07.52 | ManxPower | cullenincrease: no you would use SIP/sipconfuserid |
| 18:08.11 | ManxPower | did you read "show application chanspy" ? |
| 18:08.28 | cullenincrease | yeah but want im trying to do is after i enter the ext to connect to chanspy i would like to then be able to enter a specific extension to spy on |
| 18:08.36 | cullenincrease | yes also read the wiki entries and forum posts |
| 18:08.39 | cullenincrease | just cant quite figure it out :( |
| 18:08.41 | ManxPower | _ViperNetworks: it is usually bad to Dial from an AGI |
| 18:08.59 | ManxPower | cullenincrease: then use Read to get the digits, then pass them to chanspy |
| 18:09.24 | cullenincrease | hm |
| 18:09.29 | cullenincrease | how would i do that? |
| 18:10.19 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 18:10.50 | ManxPower | cullenincrease: you cannot spy on extensions, only channels. |
| 18:11.02 | cullenincrease | i think i got it figured out now |
| 18:11.18 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 18:11.21 | cullenincrease | i used ChanSpy(SIP/) and it automatically connects to a random ext |
| 18:11.25 | ManxPower | You could do chanspy(SIP) then dial the digits associated with the sip.conf userid you want to spy on |
| 18:11.32 | cullenincrease | but then if i enter an ext followed by # it jumps to that one |
| 18:11.56 | sweeper | anybody got a script for mass-pagining via autoanswer and meetme laying around? |
| 18:12.01 | cullenincrease | is there any way to tell via dialing an ext what users are currently on a call? |
| 18:12.01 | ManxPower | Where in "show application chanspy" does it show a / after the chanprefix? |
| 18:12.09 | neverblue | anyone VOIP providers in the channel, I am looking for service.... |
| 18:12.24 | ManxPower | cullenincrease: maybe after a couple of months of using asterisk you'll be able to write something to do that. |
| 18:12.36 | cullenincrease | is that pretty complicated? |
| 18:12.45 | ManxPower | neverblue: All VoIP providers suck. teliax.com seems to suck less than both. |
| 18:12.54 | neverblue | both? |
| 18:13.05 | neverblue | there are alot more providers than 'both' |
| 18:13.10 | ManxPower | cullenincrease: an extension is a series of digits that are in extensions.conf to map to a specific device. |
| 18:13.28 | ManxPower | neverblue: sorry "teliax.com seems to suck less than most" |
| 18:13.34 | neverblue | hehe |
| 18:13.35 | neverblue | :) |
| 18:13.47 | *** join/#asterisk CunningPike (n=arodgers@d207-6-165-254.bchsia.telus.net) |
| 18:13.53 | *** join/#asterisk sketc1 (i=ryan@nat/digium/x-563f3ddf20db0a03) |
| 18:14.02 | jacky59 | ManxPower, basically all development for mgcp on asterisk halted some monthes (if not years) ago ? |
| 18:14.04 | ManxPower | neverblue: just like the phone company maps a series of digits (phone number) to a physical port on their switch |
| 18:14.17 | ManxPower | jacky59: You would have to look at the change log. |
| 18:14.33 | _ViperNetworks | ManxPower: Why whats wrong? |
| 18:14.57 | ManxPower | _ViperNetworks: well your ago won't continue after the dial |
| 18:15.02 | ManxPower | AGI, that is. |
| 18:15.23 | neverblue | that reply wasnt for me, right ManxPower ? |
| 18:15.25 | coppice | i think the MGCP stuff in * was just created in one short burst and never touched thereafter |
| 18:15.25 | Katty | CunningPike: :> |
| 18:15.26 | ManxPower | _ViperNetworks: as has been discussed on the mailing lists over the past few days |
| 18:15.45 | ManxPower | coppice: It really should be removed. |
| 18:15.46 | _ViperNetworks | ManxPower: I didnt get you...? what will not continue..? |
| 18:15.51 | CunningPike | Hey, Katty |
| 18:15.56 | CunningPike | How are you? |
| 18:15.56 | ManxPower | if it's not being activly developed |
| 18:16.01 | CunningPike | Long time no mew |
| 18:16.01 | sweeper | ManxPower: what do people usually use? |
| 18:16.09 | ManxPower | _ViperNetworks: your AGI will never continue to execute after your run a dial |
| 18:16.24 | ManxPower | sweeper: use for what? |
| 18:16.29 | *** join/#asterisk myiagy (n=myiagy@201.56.113.74) |
| 18:16.29 | sweeper | instead of Dial |
| 18:16.37 | sweeper | to connect to another node |
| 18:16.58 | *** join/#asterisk Bensin (n=chatzill@c-1a4de253.615-1-64736c11.cust.bredbandsbolaget.se) |
| 18:17.11 | ManxPower | run the AGI, set dialplan variables, exit the agi, next priority in the dialplan run Dial using the variables set in the AGI, then after the Dial run an AGI to do cleanup. You'll also have to run the AGI in exten h |
| 18:17.17 | jacky59 | do we have ideas on what's the remaining proportion of mgcp/ H323 ? |
| 18:17.41 | jacky59 | (except voip deployed by massive ISPs for their "closed" systems) |
| 18:17.43 | sweeper | ManxPower: oh. that's freaking nasty |
| 18:17.53 | sweeper | but I guess it works~ |
| 18:17.54 | ManxPower | jacky59: almost nobody uses those techs because they are poorly supported in Asterisk |
| 18:18.09 | jacky59 | i'm figuring this out :/ |
| 18:18.10 | ManxPower | sweeper: in MY world, we never have to run an AGI after a Dial |
| 18:18.14 | _ViperNetworks | ManxPower: hmmm funky... let me check... |
| 18:18.26 | sweeper | ManxPower: yea, most times I don't either |
| 18:18.31 | jacky59 | damn, I promised a friend a free phone, sent him an old mgcp device and should have sent him a new sip one :( |
| 18:18.48 | ManxPower | jacky59: MGCP in Asterisk has NO support for NAT. |
| 18:18.58 | jacky59 | ManxPower, humm |
| 18:19.01 | _ViperNetworks | ManxPower: Thanks for the Alert man.. |
| 18:19.06 | jacky59 | my version does :p |
| 18:19.08 | ManxPower | Prolly because MGCP has no support for passwords. |
| 18:19.10 | Katty | CunningPike: yep (= |
| 18:19.13 | Katty | CunningPike: how beith? |
| 18:19.49 | jacky59 | true true... I remember I had to manually modify the code to enable the nat |
| 18:20.28 | CunningPike | Katty: Not bad, thanks - union's on strike, but I'm about to go on vacation, so hopefully it'll all be over by then |
| 18:20.45 | CunningPike | Katty: Got some contract work to keep me going until I go away |
| 18:20.51 | CunningPike | Katty: You? |
| 18:21.03 | jacky59 | wanted to submit modification long time ago, but it was so dirty... i guess that's better this way |
| 18:21.14 | jacky59 | thx for help |
| 18:21.14 | krdian_ | anybody knows any good tool/software for test sip signaling |
| 18:21.25 | jacky59 | krdian_, ethereal very nice |
| 18:21.26 | Katty | CunningPike: i'm doin good. got two ferrets, a dog, and moved to a new house. |
| 18:21.32 | Katty | CunningPike: so, staying busy as well (= |
| 18:21.45 | CunningPike | Katty: ;) |
| 18:21.57 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 18:22.05 | CunningPike | krdian_: It's called wireshark now, but it's still good |
| 18:22.15 | x86 | hey guys, I have a weird issue with my dialplan... exten => 9XXX9NXXNXXXXXX,1,Dial(Zap/g3/${EXTEN:5}|1000|RTt) |
| 18:22.36 | x86 | seems like it matches fine, but gives me a dialtone then i have to dial the number all over again |
| 18:22.36 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 18:22.56 | x86 | ohhhh crap, is it because i'm missing a _ in the beginning? |
| 18:23.14 | x86 | nope, i got that just forgot to type it |
| 18:23.42 | Hmmhesays | god I hate web interfaces |
| 18:23.45 | krdian_ | CunningPike: ok, but i like to have raports etc. |
| 18:24.03 | krdian_ | CunningPike: all-in-one tester |
| 18:24.14 | *** part/#asterisk JSegatto (n=pirc@225.89.203.68.cfl.res.rr.com) |
| 18:24.25 | x86 | http://pastebin.ca/640824 |
| 18:24.30 | Katty | Hmmhesays: i'm sure they hate you as well. |
| 18:24.32 | x86 | anyone see a problem with this? |
| 18:24.44 | *** join/#asterisk galeras (n=galeras@200.31.204.42) |
| 18:24.47 | ManxPower | x86: what else did you forget to type. COPY and PASTE it |
| 18:24.57 | x86 | ManxPower: i did, see pastebin ;) |
| 18:25.09 | x86 | 13:24 < x86> http://pastebin.ca/640824 |
| 18:25.13 | x86 | did you miss that? |
| 18:25.22 | Katty | simmer down now. |
| 18:25.26 | Hmmhesays | yes they do |
| 18:25.32 | Katty | Hmmhesays: :> |
| 18:25.32 | ManxPower | x86: Is that everything in the context and pastebin the CLI output of a call. |
| 18:25.37 | [TK]D-Fender | x86: AAAA-NPA-NXX-XXXX != _9XXX9NXXNXXXXXX |
| 18:25.43 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 18:25.46 | x86 | [TK]D-Fender: nevermind the comments ;) |
| 18:25.59 | x86 | [TK]D-Fender: we changed the format of our account codes |
| 18:26.12 | x86 | (and no, it didn't work before the change eitheR) |
| 18:26.20 | *** join/#asterisk claudiotainen (n=claudiot@ppp-15-180.33-151.iol.it) |
| 18:26.22 | *** join/#asterisk obnauticus (n=obnautic@c-71-236-219-178.hsd1.wa.comcast.net) |
| 18:26.28 | [TK]D-Fender | x86: pastebin some sample OUTPUT please... |
| 18:26.48 | [TK]D-Fender | x86: We can only guess what kind of garbage you're actually DIALING ;) |
| 18:26.57 | claudiotainen | I would like to set french as Asterisk's default voice |
| 18:27.16 | x86 | http://pastebin.ca/640829 |
| 18:27.24 | x86 | that includes the macro to do the actual dialing |
| 18:27.46 | ManxPower | x86: only people that ride the short bus to school use TtR on their Dial line |
| 18:28.02 | [TK]D-Fender | x86: pastebin some sample OUTPUT please... |
| 18:28.04 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 18:28.08 | ManxPower | x86: we are still waiting |
| 18:28.08 | x86 | http://pastebin.ca/640832 |
| 18:28.13 | x86 | [TK]D-Fender: was working on it ;) |
| 18:28.26 | Katty | you people are acting like a couple grumpy old men! |
| 18:28.34 | x86 | [TK]D-Fender: the CLI shows no error, and it actually does the Dial, right before I hear the second dialtone |
| 18:28.51 | [TK]D-Fender | x86: -- Executing Dial("Zap/12-1", "Zap/g3|17084307575|100|tRP") in new stack <- does that "| " after "zap/g3" look good to YOU? |
| 18:28.55 | claudiotainen | I've inserted a exten => s,1,SetLanguage(it) in sip.conf first context (after general) |
| 18:29.10 | x86 | [TK]D-Fender: then when you get to the second dial tone, and you dial the number again (NPANXXXXXX format), then it says something about "strange state" |
| 18:29.13 | [TK]D-Fender | x86: exten => s,n,Dial(${LDTRUNK}|${ARG2}|100|tRP) <- bad |
| 18:29.18 | ManxPower | claudiotainen: good for you, it won't do anything |
| 18:29.25 | [TK]D-Fender | x86: exten => s,n,Dial(${LDTRUNK}/${ARG2}|100|tRP) <- good |
| 18:29.25 | Mercestes | I'm not a grumpy old man. |
| 18:29.32 | x86 | [TK]D-Fender: ah, you rock! |
| 18:29.33 | x86 | thanks :) |
| 18:30.00 | [TK]D-Fender | Katty: I told you not to mention anything about the bondage gear in here! ;) |
| 18:30.05 | claudiotainen | so would you please tell me what I have to do ? |
| 18:30.14 | Katty | hi Mercestes (= |
| 18:30.19 | Mercestes | bondage? |
| 18:30.26 | ManxPower | claudiotainen: I have no idea, but "s" is never the answer to anything |
| 18:30.31 | *** join/#asterisk ashd (n=ashleydr@user-194-248-151-83.e7even.com) |
| 18:30.44 | [TK]D-Fender | claudiotainen>I've inserted a exten => s,1,SetLanguage(it) in sip.conf first context (after general) <---------- this is a line the belongs in extensions.conf, not sip.conf, and WHERE it is exactly matters.. |
| 18:31.08 | ManxPower | exten => s only matches when the extension dialed is "" i.e. EMPTY. |
| 18:31.11 | [TK]D-Fender | ManxPower: ..... whats the next letter in the alphabet following "r"? :) |
| 18:31.23 | ManxPower | [TK]D-Fender: "t" |
| 18:31.33 | [TK]D-Fender | ManxPower: SMRT :) |
| 18:31.40 | claudiotainen | yes ok I meant extensions.conf |
| 18:31.49 | ManxPower | Smarter than I look, as my grandmother always told me. |
| 18:32.01 | [TK]D-Fender | claudiotainen: We'd need to see a LOT more of your configs to tel you where you've gone wrong. PASTEBIN is your friend |
| 18:32.03 | [TK]D-Fender | ~pb |
| 18:32.03 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 18:32.45 | [TK]D-Fender | claudiotainen: And when you asked us about setting french as the default language... why are you showing us a line setting ITALIAN instead? |
| 18:33.07 | [TK]D-Fender | ManxPower: I've grown used to it. |
| 18:33.24 | claudiotainen | http://pastebin.com/d471ca62c |
| 18:33.42 | Katty | you're all crazy. |
| 18:33.55 | *** join/#asterisk guillote_GNU (n=guillote@190.7.27.17) |
| 18:33.58 | Mercestes | and so am I |
| 18:34.10 | claudiotainen | I think it's the same isn't it? |
| 18:34.30 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 18:35.04 | Mercestes | Which CD? |
| 18:35.05 | [TK]D-Fender | claudiotainen: What is using that context you showed us? |
| 18:35.42 | *** join/#asterisk ashd (n=ashleydr@user-194-248-151-83.e7even.com) |
| 18:35.47 | claudiotainen | er sip UA |
| 18:36.21 | Katty | Mercestes: debian one. |
| 18:36.23 | [TK]D-Fender | claudiotainen: Well that line :exten => s,1,SetLanguage(it) means NOTHING to it. |
| 18:36.36 | *** join/#asterisk irule (n=irule@189.164.47.106) |
| 18:36.39 | [TK]D-Fender | claudiotainen: You should set "language=[languagecode]" in their SIP.CONF entry |
| 18:36.39 | Mercestes | ahh. |
| 18:36.45 | Katty | [TK]D-Fender: i'm about to start my very first asterisk install that will use a t1 :> |
| 18:36.50 | Mercestes | get gentoo. It has great taste, and it's less filling. |
| 18:36.53 | Katty | err s/ [TK]D-Fender / Mercestes |
| 18:37.00 | Katty | Mercestes: no thanks (= |
| 18:37.00 | [TK]D-Fender | Katty: that'd BETTER be a PRI this time! |
| 18:37.01 | Mercestes | ... |
| 18:37.10 | Mercestes | omg....you said my name wrong. |
| 18:37.12 | Mercestes | I can't believe it... |
| 18:37.14 | Katty | i know |
| 18:37.19 | Mercestes | In our most intimate moment....you said *his* name. |
| 18:37.20 | [TK]D-Fender | Mercestes: Oh no.... its definately "full of it"... you RICER! |
| 18:37.21 | Katty | i fail at life. |
| 18:37.21 | Mercestes | =/ |
| 18:37.25 | Katty | *hee* |
| 18:37.43 | coppice | PRI is for wimps. real men whistle down their T1s :-) |
| 18:37.45 | Mercestes | who am I kidding? ...i don't care.... |
| 18:37.48 | Mercestes | I'm yours regardless. |
| 18:37.49 | [TK]D-Fender | Katty: You'll go from administering a system, to COMPILING one on a daily basis ;) |
| 18:37.53 | Mercestes | abuse me....I'll take it. |
| 18:38.47 | *** join/#asterisk oej (n=olle@84-20-108-55.no.ventelo.net) |
| 18:40.14 | Mercestes | I can be fender if you want me to be. |
| 18:40.15 | Mercestes | :( |
| 18:40.44 | Mercestes | .... |
| 18:40.51 | Mercestes | wow, I silenced a sea of n00bs trying to make Asterisk work. |
| 18:41.00 | Mercestes | your all staring at me...aren't you? |
| 18:41.09 | Dan0maN_Work | heh |
| 18:41.36 | Katty | me? making a new one a day?! |
| 18:41.40 | Katty | my life is over!! |
| 18:41.52 | Mercestes | nooo... |
| 18:41.56 | Mercestes | I'll be your life.... |
| 18:42.01 | Mercestes | here, have mine |
| 18:42.18 | claudiotainen | tanks fender ;) |
| 18:42.20 | Katty | does it have a warranty? |
| 18:42.22 | Katty | :P |
| 18:42.53 | Mercestes | Umm. |
| 18:42.57 | Hmmhesays | css sure doesn't like table borders on php generated tables |
| 18:42.58 | Hmmhesays | stf |
| 18:42.59 | Hmmhesays | wtf |
| 18:43.15 | Mercestes | no, not really, but you can abuse it, neglect it, mishandle it, misuse it, or otherwise violate it in any way you see fit. |
| 18:43.58 | Katty | oh no. |
| 18:44.06 | Katty | my boss is trying to get me to buy refurb 501s. |
| 18:44.16 | Katty | because they're a whole 20 bucks cheaper than new. |
| 18:44.21 | datachomper | 501s are good phones, oh... |
| 18:44.24 | Katty | ..and have no warranty |
| 18:44.25 | [TK]D-Fender | Katty: how much? |
| 18:44.36 | Katty | [TK]D-Fender: 159 for refurb |
| 18:44.36 | [TK]D-Fender | Katty: And don't you have PoE there? |
| 18:44.36 | Dan0maN_Work | what are the major differences in teh 501s vs the 550s? |
| 18:44.37 | Mercestes | Quick, take a FXO card and use it to generate a 90v ring signal directly to his brain. |
| 18:44.42 | Katty | [TK]D-Fender: ha! |
| 18:44.50 | Katty | Mercestes: hahaha |
| 18:45.28 | [TK]D-Fender | Katty: Oh. and he's only saving **8$** |
| 18:45.41 | Katty | datachomper: the 501s are nice little phones. the 601s and 650s (don't remember the difference) have a side card expansion thing AND support custom logos tho. that's pretty nifty in my opinion (= |
| 18:46.02 | datachomper | Katty, I have all my 501's setup with custom logos |
| 18:46.08 | [TK]D-Fender | Katty: IP 501 supports custom logog & MB as well... |
| 18:46.12 | Katty | oh? |
| 18:46.17 | Katty | where do i go to set this up? |
| 18:46.19 | [TK]D-Fender | Katty: indeed. |
| 18:46.25 | datachomper | Yep, One sec for linkage |
| 18:46.26 | [TK]D-Fender | Katty: same place as for a 601 :) |
| 18:46.27 | Katty | k |
| 18:46.35 | datachomper | http://www.voip-info.org/wiki/view/Polycom+Idle+Images |
| 18:46.36 | Katty | [TK]D-Fender: well aren't you handy. |
| 18:46.39 | Katty | datachomper: thank you dear. |
| 18:46.46 | datachomper | Katty, sure. |
| 18:46.51 | *** join/#asterisk Splat (n=splat@home.heehawhills.com) |
| 18:47.29 | datachomper | Whoever wrote that wiki entry is a genious. I found /no/ other documentation supporting that feature from polycom. |
| 18:47.41 | datachomper | They *barely* mention it in the admin guide. |
| 18:48.01 | [TK]D-Fender | datachomper: Enough :) |
| 18:48.42 | *** join/#asterisk Terminus (n=jjereza@210.213.140.84) |
| 18:48.53 | [TK]D-Fender | brb |
| 18:49.22 | Katty | datachomper: is the /path/to/image/name an http:// or a \\server\ path? |
| 18:49.23 | Terminus | hello. anybody have any suggestion for a good provider in the US? |
| 18:49.26 | Katty | datachomper: it doesn't really say. |
| 18:49.29 | Terminus | for business lines i mean. |
| 18:49.43 | datachomper | Katty: It can be either with a 501, from what I understand |
| 18:49.57 | Mercestes | Katty, Likely the root folder of the polycom config files |
| 18:50.00 | datachomper | Katty: I can only verify that it's a /path/to/address |
| 18:50.07 | el_critter | is it worth the effort to compile 1.4.* rather than use the precompiled 1.2 (in distros such as ubuntu) ? |
| 18:50.13 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 18:50.13 | Katty | Mercestes: oh, yeah, that would make sense. |
| 18:50.22 | wunderkin | compile.. effort? ... |
| 18:50.47 | el_critter | yeah.. pentium 2 |
| 18:51.17 | el_critter | 1.2 precompiled deb vs 1.4 .tar.gz |
| 18:51.46 | Mercestes | Katty, If you did /path/to/file it would be from the server root, but if you just specify a file name then it's the FTP(tftp,sftp,http) folder location. |
| 18:51.47 | wunderkin | that's lame if there isn't a 1.4 package yet.. as of tomorrow, 1.2 is no longer maintained |
| 18:51.57 | x86 | ok |
| 18:52.15 | datachomper | Mercestes is correct. |
| 18:52.20 | x86 | so changing the | to a / seemed to fix the Dial command, but now the Dial command is putting people right into voicemail |
| 18:52.40 | Mercestes | x86, Why? |
| 18:56.20 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 18:57.28 | Mercestes | Ok, quiet, he's back |
| 18:57.41 | wunderkin | stop talking about him |
| 18:57.43 | Voicemeup | <PROTECTED> |
| 18:58.26 | Mercestes | ... |
| 18:59.26 | Mercestes | Houston Tx, on Global Crossing Network |
| 18:59.34 | Mercestes | Global Crossing sucks for faxes btw. |
| 18:59.36 | Voicemeup | yes |
| 18:59.40 | Voicemeup | wanna be sure |
| 18:59.42 | Voicemeup | ;) |
| 18:59.46 | Mercestes | be sure. |
| 18:59.54 | Mercestes | they are a cut rate "carrier." |
| 19:01.03 | Voicemeup | imho they all suck in usa except L3 |
| 19:01.11 | Voicemeup | i mean everyone |
| 19:01.15 | _ViperNetworks | ManxPower: Is it possible to use phpagi-asmanager.php and do the same as DIAL via AGI |
| 19:01.39 | Mercestes | there is only one common element in all that suckiness |
| 19:04.32 | tzafrir_laptop | wunderkin, unstable has had 1.4 for a long time now |
| 19:04.36 | *** join/#asterisk mattchis (n=mattchis@216.54.143.246) |
| 19:08.18 | galeras | "make install" for libpri-1.2.5 get this error: "/sbin/ldconfig: /usr/lib/libid3tag.so.0 is not a symbolic link" |
| 19:08.22 | galeras | Thanks for any idea |
| 19:08.48 | tzafrir_laptop | galeras, this is not a problem of libpri |
| 19:09.21 | tzafrir_laptop | this is generally a problem of your system. you'll get that warning again when running 'lsconfig' as root |
| 19:10.38 | galeras | tzafrir_laptop: you are right, i will check, thanks |
| 19:10.43 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 19:12.25 | *** join/#asterisk y7n (n=na@85-210-9-252.dsl.pipex.com) |
| 19:12.47 | y7n | how do I hangup a sip channel from the CLI? |
| 19:12.51 | y7n | without warning |
| 19:13.42 | wunderkin | soft hangup |
| 19:13.47 | y7n | thanks |
| 19:14.00 | brodiem | I'm trying to have queue recordings set the agent ID within the file name (I need to record from queues.conf due to the requirement of the unique ID on that leg of the call). I set setinterfacevar=yes in queues.conf and included ^{MEMBERINTERFACE} in a MONITOR_EXEC var, but that var never gets set... any ideas? |
| 19:14.30 | tzafrir_laptop | galeras, my mistake: run 'ldconfig' as root |
| 19:17.15 | Katty | 13% downloaded and going. |
| 19:17.43 | Jingles | so, what do you do when you're trying to test a remote location because they've got poor call quality - but Asterisk's console isn't reporting ANY ERRORS. |
| 19:17.49 | *** join/#asterisk centrex (i=centrex@nat/digium/x-52ccf9b402b6b904) |
| 19:18.13 | ManxPower | Jingles: poor call quality is not an error. |
| 19:18.24 | Jingles | right. |
| 19:18.28 | Jingles | so, I didn't know we were having issues. |
| 19:18.43 | Jingles | until someone said 'oh, I never use that shitty phone system - calls are always broken up, or dropping off' |
| 19:18.49 | ManxPower | You check for IRQ conflicts, check zttest when disk access is happening, duplex mismatch on the ethernet port |
| 19:19.13 | Jingles | ManxPower : ok. The rest of the company (specifically everyone *local*) doesn't have this problem. |
| 19:19.21 | Jingles | just one phone, out at a remote office. |
| 19:19.32 | ManxPower | Oh! Just one phone at a remote office. |
| 19:19.35 | Jingles | yessir. |
| 19:19.36 | ManxPower | So it is a network issues. |
| 19:19.41 | Jingles | right. |
| 19:19.48 | ManxPower | I'll bet you are using SIP, right? Are you using 1.2 or 1.4? |
| 19:20.00 | Dan0maN_Work | this is probably a very nooby question, but i'll do it anyway. the books and online articles all talk about 1.4 being a beta. is it still? the site doesn't seem to point that out. |
| 19:20.07 | Jingles | they've got 1 meg up and down, and the ISP is holding tight to 'they're getting what you're paying for'. |
| 19:20.14 | Jingles | SIP, and 1.2. |
| 19:20.20 | *** join/#asterisk Voicemeup (n=VoiceMeU@modemcable159.131-56-74.mc.videotron.ca) |
| 19:20.21 | ManxPower | Dan0maN_Work: no. 1.4.8 or 1.4.9 is current |
| 19:20.22 | brodiem | Jingles use a low b/w codec - i.e. g729 |
| 19:20.27 | Jingles | and right now, upgrading to 1.4 is not an option. |
| 19:20.30 | Yourname` | Hmm, this is weird. I have an itsp.. but when I do sip show registry.. it's not registered with the itsp. |
| 19:20.33 | Dan0maN_Work | k. thank you |
| 19:20.35 | ManxPower | Jingles: Do you know what a "jitterbuffer" is and why it is important? |
| 19:20.36 | Jingles | (it will be soon - but it is not *yet*) |
| 19:20.53 | ManxPower | Yourname`: then the itsp can't send you calls. |
| 19:20.57 | Jingles | ManxPower : if I'm to understand, it's similar to a 'read ahead buffer' to smooth out the call quality. |
| 19:21.42 | ManxPower | Jingles: In 1.2 only IAX2 has a jitterbuffer. No other protocol has jitterbuffer support. |
| 19:21.48 | Jingles | *nods* |
| 19:21.52 | Jingles | I remember reading that. |
| 19:22.04 | Jingles | but 1.4 uses jitterbuffer for SIP? |
| 19:22.09 | ManxPower | Now think about lack of a jitterbuffer, the network connection and SIP. |
| 19:22.11 | Voicemeup | how you debug fax again apart from udtpl |
| 19:22.12 | Voicemeup | lol |
| 19:22.22 | Voicemeup | and man cleanup source lol res_agi.c: Wow, connected! is not |
| 19:22.24 | Voicemeup | a good sign |
| 19:22.29 | Jingles | ManxPower : that makes me think about shots of vodka until I can't think any more. |
| 19:22.29 | ManxPower | Jingles: SIP is a signalling protocol. AUDIO uses RTP and 1.4 has RTP jitterbuffer. |
| 19:22.39 | Voicemeup | bah im dumping asterisk .. cee y |
| 19:22.44 | Jingles | right, right. SIP just makes the phones ring. ;) |
| 19:22.46 | *** part/#asterisk Voicemeup (n=VoiceMeU@modemcable159.131-56-74.mc.videotron.ca) |
| 19:23.03 | centrex | That was interesting. |
| 19:23.14 | Qwell[] | way to get business for yourself |
| 19:23.17 | *** join/#asterisk link55 (n=andy@rrcs-24-136-121-163.nyc.biz.rr.com) |
| 19:23.22 | brodiem | ManxPower doesn't 1.4's JB require a Zap channel in the leg of the call? |
| 19:23.38 | ManxPower | Jingles: We don't use ANY VoIP over non-QoS links because of the lack of an RTP jitterbiffer. |
| 19:23.45 | ManxPower | brodiem: hell if I know. |
| 19:23.52 | link55 | Anyone use IAX realtime in 1.4? |
| 19:24.02 | centrex | link55, I have before but it's been a second. |
| 19:24.05 | Jingles | ok. Which totally stands to reason - now I need to figure out how to set up QoS. |
| 19:24.17 | Jingles | but - all that aside. |
| 19:24.18 | ManxPower | brodiem: I don't use 1.4 because until VERY recently I've not considered it production quality. |
| 19:24.25 | link55 | I'm having problems with all the threads "locking up", so all the IAX traffic just sits and queues up |
| 19:24.29 | Jingles | on a 1 meg up and down I still shouldn't have any issues with bandwidth. |
| 19:24.29 | ManxPower | Jingles: What type of network connection is between the two locations? |
| 19:24.49 | centrex | link55, Hrm never seen that one, sorry :/ |
| 19:25.01 | ManxPower | Frame Relay, *DSL, Point to Point T-1, ATM? |
| 19:25.08 | Jingles | we've got DSL out at the remote office (1MBit up/down) and similar bandwidth (much more) here at the main office. |
| 19:25.08 | link55 | 1.2 is ever worse since it's not threaded, there are problems with the Realtime causing the IAX thread to not respond |
| 19:25.09 | Jingles | DSL. |
| 19:25.32 | ManxPower | Jingles: you will NEVER EVER get real QoS on DSL without the carrier doing it. |
| 19:25.38 | Jingles | crapsteaks. |
| 19:25.44 | brodiem | ManxPower found the answer here if you're interested: http://asterisk.org/node/48317 |
| 19:25.45 | ManxPower | We use T-1s |
| 19:27.16 | ManxPower | brodiem: Every server we have has a Zaptel card and PRI on it. |
| 19:27.36 | Jingles | That begs the question - why would my remote offices in Hawaii (my main office is here in *Alaska*) have better call quality than the one office phone out in a town that's only 50 or so miles away. |
| 19:28.10 | ManxPower | Jingles: I did not say your problems were jitter. That is just the most common issue. |
| 19:28.14 | Jingles | *nods* |
| 19:28.37 | Jingles | So, if you were trying to test to see if jitter were the problem, what exactly might you do? |
| 19:28.43 | ManxPower | Jingles: some DSL modems have large packet buffers in them, some ISP routers are configured with large packet buffers. |
| 19:29.12 | ManxPower | do a 500 packet ping from each location to the main location. Look at the VARIENCE in latency, not the actual latency. |
| 19:29.25 | ManxPower | I would expect the office with poor call quality to have large variences. |
| 19:29.43 | Jingles | spiking up and down. That's a good test. |
| 19:30.08 | ManxPower | Are there VPNs involved? |
| 19:30.08 | Jingles | in fact, I should connect *straight* into the DSL device, and run that test. |
| 19:30.16 | Jingles | so that the rest of the office equipment isn't a factor. |
| 19:30.16 | ManxPower | no you should not. |
| 19:30.19 | Jingles | really? |
| 19:30.38 | Yourname` | ManxPower: When I type sip show registry.. what should show? My VoIP ITSP, right? |
| 19:30.42 | ManxPower | that would eliminate any possibility of some host at the poor quality location being infected by a virus that spikes the bandwidth usage |
| 19:30.55 | ManxPower | Yourname`: do you have a register => whatever in sip.conf? |
| 19:31.13 | Yourname` | Yes! That's the answer, lol.. lemme look |
| 19:31.34 | ManxPower | Jingles: first you don't want change anything, you want to ping in the same enviroment the poor call quality is happening in. |
| 19:31.39 | Jingles | ManxPower : well, I've got some ideas now of what to go and look for. |
| 19:31.50 | Jingles | so, now I'ma go look. Thanks for the pointers. They're very much appreciated. |
| 19:31.51 | ManxPower | if you see large latency variations, THEN try to determine why |
| 19:31.55 | Jingles | *nods* |
| 19:32.12 | Dan0maN_Work | also, you may have more hops to the major backbones from the poor quality office than the one in hawaii. could be a cause |
| 19:32.25 | ManxPower | I find half the time it's some asshole running bitttorrent or something |
| 19:32.33 | Jingles | I don't know that we even *have* any major backbones up here. |
| 19:32.50 | Jingles | so, the main office and the hawaii offices may be enjoying a benefit because of that. |
| 19:33.17 | ManxPower | Jingles: Also remember than many ISPs make ping (ICMP) very low priority, many routers also do that. |
| 19:33.21 | Jingles | anyone. off to the remote office. if I have any other questions, I'll log back in and ask them. ;) |
| 19:33.29 | ManxPower | So ping MAY not give you a good picture, but it is a place to start |
| 19:33.33 | Jingles | *nods* |
| 19:35.02 | lirakis | can i set up a sip extension on an asterisk server... so that another asterisk server can use it as a trunk? |
| 19:35.10 | *** join/#asterisk sferley (n=Testme@64.141.113.130) |
| 19:35.53 | lirakis | .. or rather......... does anyone know a nice howto.. or resource of info on this?? FYI: i have "the book" |
| 19:37.21 | lirakis | quiet.. ?? netsplit?? |
| 19:37.23 | sferley | Hi, i was woundering if anyone knew this problem here. I am running asterisk 1.2-x and i have a remote cisco phone that seems to not be able to dial out, even though it registers just fine. I get on the phone a "Calling out <inv>" 9 times out of 10. One call might go through and then won't dial out again. It can receive calls just fine, as in i can call it. |
| 19:37.59 | lirakis | sferley: sounds like multiple devices trying to register the same extension... |
| 19:38.18 | Dan0maN_Work | prolly just busy atm. they've been really active today |
| 19:38.29 | sferley | hmm.. looking on the console, i only see one registration log. |
| 19:39.01 | ManxPower | sferley: Registration ONLY applies to Server -> Phone calls. It has NOTHING to do with Phone -> Server calls. |
| 19:40.14 | Katty | so, i just updated my sip.cfg and changed the DefaultIdle thingy and made a greyscale image of my company's logo |
| 19:40.15 | sferley | ManxPower: hmm so you are saying there could be more phones then one that are trying to send calls as (extension on question is 101) to the asterisk server. |
| 19:40.23 | Katty | teh phone is rebooting :> |
| 19:40.39 | ManxPower | sferley: no, I am saying something is not configured correctly |
| 19:40.42 | Dan0maN_Work | heh |
| 19:40.58 | Katty | you know what's not configured correctly? ManxPower's brain. |
| 19:41.01 | Katty | it's /all/ screwed up ;) |
| 19:41.10 | Katty | and my logo didn't show up either :< |
| 19:41.18 | ManxPower | Katty: I think the term you are looking for is "sick pervert" |
| 19:41.26 | Katty | well i wasn't going to go that far. |
| 19:41.30 | Katty | but if you want to admit it, that's fine :P |
| 19:41.36 | datachomper | Katty, are you using poly 501's ? |
| 19:41.41 | Katty | datachomper: yes'r. |
| 19:42.09 | sferley | ManxPower: if the phone is on a different network then the asterisk server is, even though there is on nat, should one turn on nat on the extension config ? |
| 19:42.15 | datachomper | Katty, does it still show the date in the middle of the lcd, or did it move down and shows the polycom logo? |
| 19:42.15 | Katty | i have too many pixels! |
| 19:42.28 | Katty | no polycom logo |
| 19:42.34 | Katty | same as before my reboot. |
| 19:43.11 | datachomper | In your global sip.cfg, make sure this is set to 1, not 0 <indicators ind.idleDisplay.enabled="1"> |
| 19:43.11 | Katty | i also have a dash in my filename, i'm gonna take that out too |
| 19:43.21 | Katty | okay. i'll double check that too (= |
| 19:47.16 | *** part/#asterisk lirakis (n=etamme@65.200.191.253) |
| 19:54.29 | Aces1Up | does anyone here know much about |
| 19:54.44 | Aces1Up | Mexican Licensing procedures? for providing telephony services? |
| 19:57.08 | Katty | datachomper: :>>>>>>>> |
| 19:57.13 | sweeper | Aces1Up: no clue. but I lay good odds that a few thousand pesos in the right place should clear things up.... |
| 19:57.27 | datachomper | :OP |
| 19:58.22 | Aces1Up | sweeper, well its funny that all these companies allow you to purchase DID's in any country, i don't understand the regulations if you want to provide a service on these DID's. |
| 19:58.53 | ManxPower | sferley: if there is no nat then using nat=yes is useless |
| 19:59.08 | xpot | can anyone help with a system cmd issue? I am trying to run a script on voice file written after hangup... the script runs but nothing happens to the designated files, I am guessing the file is not completly written when h is executed |
| 19:59.54 | ManxPower | xpot: that should not be the issue, but if you are paranoid add a Wait(1) as the first priority of extension "h" |
| 20:00.07 | ManxPower | Do you SEE your System(.. command being executed on the Asterisk CLI? |
| 20:00.16 | *** join/#asterisk joetester (n=joeteste@216.191.34.13) |
| 20:00.28 | xpot | I do, however, if I add a wait(1) then I do not |
| 20:00.34 | xpot | but I do see the wait |
| 20:00.57 | *** part/#asterisk joetester (n=joeteste@216.191.34.13) |
| 20:02.47 | xpot | the reasoning I have for the file not finished writing, is I have the files being mixed and changed from wav to mp3... this process may not be quick enough when my other script runs |
| 20:03.31 | ManxPower | xpot: then you have a priority problem with the wait and system command |
| 20:03.53 | *** join/#asterisk |dennis| (n=dennis@200.32.233.84) |
| 20:04.03 | ManxPower | pasteb in your "h" extension |
| 20:04.46 | neverblue | anyone VOIP providers in the channel, I am looking for service.... |
| 20:05.50 | ManxPower | neverblue teliax.com |
| 20:05.51 | xpot | http://pastebin.com/d440b062a |
| 20:06.13 | ManxPower | You MUST have one and only one priority "1" |
| 20:06.24 | Katty | i see the sangoma t1 cards are about 500 bucks more than the digium ones. is it really worth it? |
| 20:06.32 | neverblue | ManxPower, repetition is not necessary |
| 20:06.53 | ManxPower | If I needed an ITSP teliax is who I would use. |
| 20:07.00 | xpot | I see, my first h is a write to database |
| 20:07.29 | ManxPower | Katty: Generally EC adds about $500 to the cost of the card. Sangoma and Digium cards are similar in price for similar features. |
| 20:07.50 | ManxPower | xpot: is the first "h" priority 1? |
| 20:07.54 | xpot | yes |
| 20:08.07 | ManxPower | xpot: I don't have any other ideas. |
| 20:08.43 | xpot | ok, I do appreaciate your atempt. -=0) |
| 20:11.00 | Katty | ManxPower: oh.... i see. |
| 20:11.10 | Katty | ManxPower: i must have been comparing digium non EC to sangoma EC |
| 20:11.14 | [TK]D-Fender | ManxPower: mORE LIKE ABOUT 300$ |
| 20:11.25 | Katty | [TK]D-Fender: holy capslock batman. |
| 20:11.37 | Katty | [TK]D-Fender: :> |
| 20:12.16 | *** join/#asterisk sensei (n=erik@northernmost.org) |
| 20:12.20 | [TK]D-Fender | Katty: wHaT ArE YoU TaLkInG AbOuT?!?!?!? |
| 20:12.40 | sferley | on a cisco phone, anyone know what garp_enable is ? |
| 20:12.46 | fetcher | anyone know if a PAP2 can be made not to fall through to re-order after the far end hangs up? |
| 20:13.05 | fetcher | preferably returning to a dialtone instead? |
| 20:13.13 | sensei | Is the owner of the chan-sccp-users list here by any chance? |
| 20:15.44 | *** join/#asterisk madcap (i=madcap@unaffiliated/madcap) |
| 20:17.21 | *** join/#asterisk y7n (n=na@85-210-9-252.dsl.pipex.com) |
| 20:19.48 | Dan0maN_Work | sferley: http://en.wikipedia.org/wiki/GARP |
| 20:20.02 | y7n | i'd like to send SMS messages from asterisk to GSM mobile phones. Can anyone recommend a suitable GSM modem that works well with Linux + Asterisk. |
| 20:20.04 | yakkop | hi -- I've got a strange problem with iptables: a certain period after restarting ip tables I find asterisk in a state where it no longer detects incoming SIP messages. I have iptables sending any rejected messages to the log -- and see nothing there, but I see incoming SIP messages with a portcap. any ideas what could be happening? |
| 20:20.38 | ManxPower | y7n: what country are you in? |
| 20:20.48 | y7n | UK |
| 20:21.21 | ManxPower | y7n: you should not need a "GSM modem". You would want to use app_SMS and have it dial your SMSC to send the SMS message |
| 20:22.03 | ManxPower | And, no there are o really good docs for this. Your best bet is to search the mailing list archives or the Wiki |
| 20:22.08 | ManxPower | ~mailinglist |
| 20:22.08 | jbot | Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.htmm |
| 20:22.10 | ManxPower | ~wiki |
| 20:22.29 | ManxPower | ~voipinfo |
| 20:22.40 | ManxPower | fine. The Wiki is at voip-info.org |
| 20:22.51 | y7n | ok, thanks. :) |
| 20:23.03 | *** join/#asterisk toot (n=toot@84.19.255.123) |
| 20:23.13 | y7n | So in theory i should be able to dial my operators SMSC number |
| 20:23.18 | y7n | im not sure they'd allow that |
| 20:27.26 | centrex | astmasters.net? rofl |
| 20:27.52 | datachomper | Well there are plenty of free online SMS message webpages out there, so /somebody/ is allowing it :OP |
| 20:34.36 | zdsl | Hello |
| 20:34.42 | zdsl | Any consultants out there? |
| 20:34.47 | sweeper | ManxPower: how fast can you send SMS's that way? |
| 20:34.59 | sweeper | like 1 per second, or a lot faster? |
| 20:35.14 | mvanbaak | have a look at bayhamsystems |
| 20:35.25 | ManxPower | sweeper: I'm in the USA, we don't have SMSCs here. |
| 20:35.44 | ManxPower | zdsl: I am a consultant. Are you interested in applying to become a customer? |
| 20:36.52 | zdsl | depends |
| 20:38.09 | ManxPower | I don't do one-off contracts. |
| 20:38.43 | ManxPower | The customer that I have had for the shortest has been a client for 4 years -- the longest has been a customer for 11 years. |
| 20:39.02 | ManxPower | (I do non-asterisk related consulting) |
| 20:39.06 | ManxPower | as well as asterisk |
| 20:39.49 | centrex | I was about to ask how you've been an asterisk consultant for 11 years =) |
| 20:40.10 | Mercestes | I'm a consultant |
| 20:40.23 | *** join/#asterisk galeras (n=galeras@200.31.204.42) |
| 20:40.42 | Mercestes | .... |
| 20:40.46 | Mercestes | What? I am.... |
| 20:40.47 | Mercestes | really..... |
| 20:41.17 | galeras | Sirs: is there some way to read channel variables in other channel? |
| 20:41.31 | galeras | i mean without ami |
| 20:41.52 | Mercestes | galeras, what exactly are you trying to do? |
| 20:42.21 | centrex | galeras: I'm not positive, but I think you would have to save that to a global variable in one channel, and then read the global variable in another. |
| 20:42.54 | *** join/#asterisk beighto (n=chatzill@12.176.156.130) |
| 20:43.03 | ManxPower | centrex: I've been using Asterisk since late 2003. I do networking LAN/WAN consulting as well. |
| 20:43.14 | galeras | mmm, long story, basically to emulate chanspy's ${SPYGROUP} with zapbarge |
| 20:43.23 | centrex | ManxPower, I figured, I was joking for the most part =). |
| 20:44.24 | ManxPower | 8-) |
| 20:45.19 | beighto | Any recommendations on a good HUD for Asterisk besides Fonality? |
| 20:46.07 | ManxPower | I didn't know Asterisk supported Heads Up Displays 8-) |
| 20:46.08 | Corydon76-work | HUD? |
| 20:46.18 | centrex | I think he means a GUI =) |
| 20:46.31 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 20:47.08 | Corydon76-work | galeras: see ImportVar |
| 20:47.09 | beighto | I thought you could set up some sort of httpd without installing something like trixbox |
| 20:47.33 | Corydon76-work | and why would you want to do that? |
| 20:47.52 | wunderkin | heads up, 7 up? |
| 20:47.56 | beighto | For a help desk phone system. The boss wants to know which agents are logged in at one time. |
| 20:48.12 | galeras | Corydon76-work: because unlike zapbarge, chanspy doesn't guaranty exact channel match |
| 20:48.17 | tzafrir_laptop | beighto, FOP? |
| 20:48.34 | Corydon76-work | Yeah, I'd go with FOP as well |
| 20:48.48 | beighto | what is FOP? |
| 20:48.59 | Corydon76-work | Flash Operator Panel. See the book. |
| 20:49.01 | datachomper | A rival product to Dapper Dan |
| 20:49.02 | Corydon76-work | ~thebook |
| 20:49.03 | jbot | i heard thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 20:49.16 | datachomper | "I don't want no FOP, I'm a Dapper Dan man" |
| 20:49.22 | beighto | ahh, flash operator panel |
| 20:49.25 | centrex | Corydon76-work is lying. It's the fraternal order of police, according to google. |
| 20:50.11 | Corydon76-work | centrex: hush, you |
| 20:50.18 | centrex | k |
| 20:50.24 | galeras | Corydon76-work: thanks seems is exactly what i need |
| 20:50.30 | tzafrir_laptop | BTW: is there some SIP client program that will just register and do BLF? |
| 20:50.51 | tzafrir_laptop | not even necessarily handle audio |
| 20:51.07 | *** part/#asterisk sensei (n=erik@northernmost.org) |
| 20:51.08 | beighto | I don't think FOP will work. There are more than 30 people that can be logged in at once, I don't think FOP can display all of them at once. |
| 20:51.27 | tzafrir_laptop | beighto, you can scale the buttons |
| 20:51.46 | tzafrir_laptop | but with more than 30 or so people, what exactly do you want to show? |
| 20:52.46 | beighto | tzafrir_laptop- I'll give it a try. It is supposed to show all the help desk agents that are registered to the server so the boss knows who is working. |
| 20:56.41 | beighto | you can have 100 buttons on a screen at once! woo! |
| 20:57.26 | Corydon76-work | And if you have them log in and log out pretty quickly, you can simulate pong. woo! |
| 20:57.57 | Qwell[] | Corydon76-work: I want a sample of that. |
| 20:58.47 | Corydon76-work | Qwell[]: if we only had 100 people with phones with nothing better to do... |
| 20:59.02 | Qwell[] | pfft, call files or sipp |
| 20:59.04 | beighto | sweet! |
| 20:59.19 | *** part/#asterisk harlequin516 (n=sham@styk.net) |
| 20:59.21 | datachomper | Simulate pong in the FOP? haha |
| 20:59.29 | datachomper | That would make a funny youtube video |
| 20:59.59 | Corydon76-work | Pretty lame, really. That was the original "screen" for pong in the minicomputer days... |
| 21:00.17 | beighto | or do the tetris thing like those MIT guys did |
| 21:00.18 | Corydon76-work | (it was the status lights on the front of a minicomputer) |
| 21:00.33 | *** join/#asterisk BudgetDedicated (n=BudgetDe@s5593c2e9.adsl.wanadoo.nl) |
| 21:01.11 | datachomper | I thought pong was invented on an oscilliscope? |
| 21:02.46 | BudgetDedicated | Hi, is it possible to replace MusicOnHold with something else (like an agi script). I've been experimenting with php-agi and have made some nice things with computervoices and interactive programs. But i was wondering if I could create something (like a game you play against the computervoice) and put that in place of moh |
| 21:03.16 | centrex | speaking of which, apparently debian has a version of pong in opengl.. |
| 21:03.27 | datachomper | Omg, it will be the new MUD :OP |
| 21:04.01 | *** join/#asterisk Ebola (n=Ebola@host86-138-208-67.range86-138.btcentralplus.com) |
| 21:04.20 | BudgetDedicated | pong could btw be a very good example of a MOH app. if a person is looking at our website while calling we could have them play pong with phone contorls ;) |
| 21:04.35 | ManxPower | BudgetDedicated: No, you cannot |
| 21:05.13 | BudgetDedicated | ManxPower: so replacing moh with something else is not possible ? |
| 21:05.28 | ManxPower | BudgetDedicated: you can replace it with any audio files you want. |
| 21:05.29 | anonymouz666 | Does anyone know any webphone (it could be IAX)? |
| 21:05.39 | centrex | anonymouz666, : kiax ? |
| 21:05.55 | centrex | Oh webphone, sorry |
| 21:06.08 | neverblue | anyone VOIP providers in the channel, I am looking for service.... |
| 21:06.37 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 21:06.44 | *** join/#asterisk jarrod (i=anon@theos.org) |
| 21:07.09 | anonymouz666 | centrex: something to put on website and click.. and boom. I can also do it through Manager but that would require a softphone on the visitor. |
| 21:07.12 | *** join/#asterisk snook3r (n=ariel@bzq-219-46-202.isdn.bezeqint.net) |
| 21:07.18 | BudgetDedicated | ManxPower: i would like to replace it with a hook into an agi script. If i put someone on hold i'd like the agi to 'pick up' that line for the time being. I have a lot of possibilities in mind. (like letting people choose their music, reading them the weather forcast or news, playing games, checking the status of their server etc) |
| 21:07.44 | BudgetDedicated | so it were possible somehow it would open up unlimited possibilities |
| 21:08.30 | zdsl | hello |
| 21:08.40 | zdsl | sorry was someone trying to contact me? |
| 21:09.04 | zdsl | I am looking for some asterisk gurus to help on a project of mine |
| 21:09.22 | anonymouz666 | 1 dollar/hour |
| 21:09.26 | anonymouz666 | :) |
| 21:09.33 | *** part/#asterisk jarrod (i=anon@theos.org) |
| 21:11.16 | *** join/#asterisk BrokenArrow (n=Lp@wikipedia/BrokenArrow) |
| 21:12.06 | *** join/#asterisk gammah (n=gammah@70-253-197-131.ded.swbell.net) |
| 21:12.43 | Capps- | zdsl: asterisk's website has a forum where you can place your want ad. |
| 21:12.55 | fujin | BudgetDedicated: just Answer() and then do whatever you like with agi |
| 21:14.32 | BudgetDedicated | fujin: I want it to happen once I put someone on hold, not just when they call in. If I answer and let is go into the agi than how will i take over once i pick up the phone? |
| 21:15.09 | fujin | mod the code |
| 21:15.11 | fujin | heh |
| 21:15.17 | fujin | I don't think it's something you can do |
| 21:15.52 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 21:19.31 | *** join/#asterisk Beirdo (n=gjhurlbu@unaffiliated/beirdo) |
| 21:19.43 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 21:21.54 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 21:24.21 | fujin | Has anyone here done inbound-caller prioritising based on callerID? |
| 21:28.57 | sferley | After running some debugs here on a Cisco 7960 phone, i noticed i am getting "SIP/2.0 401 Unauthorized" authorization errors on the first 2 of the 3 channels i configured for the 7960. All extensions have been configured exactly the same. It seems only the 3rd extension (same 101 extension for all) registers. Anyone have any idea? |
| 21:29.53 | *** join/#asterisk dharrigan (n=dharriga@82-71-62-76.dsl.in-addr.zen.co.uk) |
| 21:31.04 | [TK]D-Fender | sferley, we suggest you show us everything related including SIP debug so we can see where it went wrong. |
| 21:31.11 | [TK]D-Fender | sferley, PASTEBIN is your friend. |
| 21:31.12 | [TK]D-Fender | ~pb |
| 21:31.13 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 21:34.20 | *** join/#asterisk dijungal (n=kdaniel@64.86.52.254) |
| 21:34.46 | sferley | jbot,[TK]D-Fender: http://pastebin.com/d9429450 |
| 21:35.06 | dijungal | hi all.... 1. while spying using Chanspy() is there anyway to use a key to exit chanspy() and move to the next priority line? |
| 21:35.09 | generalhan | haha, LOVE addressing the bots !!! |
| 21:35.20 | *** join/#asterisk sangee (i=ravi@206.191.114.82) |
| 21:36.37 | sangee | i got fake and real ringback tone (both ring) when i use sipura or PAP2 device on asterisk, anyone know how to solve it? |
| 21:37.21 | dijungal | 2. I exempt the ,b option but chanspy() still does not allow me to spy on channels that are not on a call |
| 21:38.52 | sferley | [TK]D-Fender: any ideas? |
| 21:39.23 | [TK]D-Fender | sferley, And the reason I don't see your sip.conf entries for this is....? |
| 21:42.30 | sferley | http://pastebin.com/dc35f1e2 |
| 21:43.24 | galeras | dijungal: what about zapbarge? |
| 21:43.58 | dijungal | i've never used zapbarge, because i want to monitor sip agents taking inbound zap calls |
| 21:44.14 | dijungal | and everyone seems to think chanspy is better thatn zapbarge |
| 21:45.24 | [TK]D-Fender | sferley, well if you put the user as 101 and password as 123456 you should be fine... you'll have to show something more for any fiurther comments. |
| 21:45.51 | *** join/#asterisk |dennis| (n=dennis@200.32.236.20) |
| 21:46.29 | galeras | dijungal: for me zapbarge has a couple nice things: 1. # key lets you end barge and go to the next priority line. 2. channel match is exact. |
| 21:49.03 | Yourname` | The most confusing problem, ever. |
| 21:49.17 | centrex | Yourname`, quite a bold statement! |
| 21:49.34 | Yourname` | You said the same thing to me before, are you a bot? |
| 21:49.58 | centrex | I don't remember ever saying anything to you before, but yes, I am a bot. |
| 21:50.23 | sferley | [TK]D-Fender: http://pastebin.com/d6e1b770a .. included the Cisco SIP<mac>.cnf ... one thing i have just noticed which is strange is when i switch NAT=yes on the sip config, i get 1 channel to register (3rd one), if i change that back to NAT=never, none register.. but the first 2 out of the 3 still don't register no matter what .. |
| 21:51.56 | *** join/#asterisk pierceb (n=pierceb@216.36.132.161) |
| 21:53.29 | Yourname` | Ok, so basically, the inbound box I was mentioning earlier today accepts SIP transfers to extension 1012, and when that happens agents logged in on the inbound box 1000-1010 get rings. But now, once every few calls, instead of going to the agents.. the called person hears dead air. |
| 21:53.39 | Yourname` | What's going wrong? |
| 21:53.48 | Yourname` | And it happens once every few calls. |
| 21:54.30 | Yourname` | SIP debug on inbound box http://pastebin.ca/641053 |
| 21:54.57 | Yourname` | Anything else you need, please ask. |
| 21:55.36 | *** part/#asterisk mtaht4 (n=m@host99.72.248.107.conversent.net) |
| 21:55.53 | *** part/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 21:59.07 | dijungal | galeras: true but i am interested in monitoring Agents not ZAP calls |
| 21:59.10 | karleeto | i'm trying to compile zaptel with hpec, and i'm using a trixbox 2.2 system. at first i couldn't compile zaptel b/c i didnt have the kernel sources installed, (i guess trixbox doesn't leave the sources there). well, i got the kernel source rpm that matched my kernel and installed that, but it gave me linux-2.6.9.tar.gz and a TON of patch files in /usr/src/redhat/SOURCES, so now i've got the source, i extracted it under /usr/src, and made the linux s |
| 21:59.28 | dijungal | galeras: so i want to monitor agents on calls and agents that are not on calls |
| 21:59.38 | karleeto | i was thinking now that the problem was that either the kernel wasn't built, or maybe that its not patched |
| 21:59.56 | karleeto | how can i make it patch with all those millions of patch files? |
| 22:00.18 | *** join/#asterisk cspot (i=cspot@ip68-1-63-100.pn.at.cox.net) |
| 22:00.26 | *** join/#asterisk NoIdea465 (n=NoIdea46@d53-64-116-212.nap.wideopenwest.com) |
| 22:02.41 | dijungal | Is there a way to 'exit' ChanSpy and continue down the dialplan? Or is a caller stuck in ChanSpy until they hangup the phone? |
| 22:08.12 | centrex | karleeto, I think trixbox uses centos, why didn't you just use yum to install the sources? In fact, i think trixbox even has it's own repositories set up by default. yum install kernel-devel (or kernel-devel.i686 possibly?) to install your sources. |
| 22:08.21 | *** join/#asterisk _deg_ (n=deg@200.195.161.164) |
| 22:08.35 | _deg_ | Is there someone that could help me on it? |
| 22:08.37 | _deg_ | st_expr2.fl: ast_yyerror(): syntax error: syntax error, unexpected '=', expecting $end; Input: |
| 22:14.19 | *** join/#asterisk Starvirus2 (n=starviru@infected.with.a.starvir.us) |
| 22:14.36 | dijungal | Is there a way to 'exit' ChanSpy and continue down the dialplan? Or is a caller stuck in ChanSpy until they hangup the phone? |
| 22:15.19 | *** join/#asterisk sysreq (n=sysreq@modemcable064.127-81-70.mc.videotron.ca) |
| 22:18.18 | ManxPower | dijungal: does "show application chanspy" say you can exit it? |
| 22:20.20 | Hmmhesays | I really wish the destination field in the could be something other than the extension called |
| 22:20.21 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 22:21.48 | dijungal | hmm.. |
| 22:22.35 | s0ck | MDL_ERROR:REQ (tei_l2) <---- mean anything to anyone? (misdn) |
| 22:23.15 | s0ck | im beginning to think there's some kind of regional variation in isdn in the uk |
| 22:23.18 | dijungal | ManxPower: no |
| 22:23.39 | ManxPower | then it prolly doesn'. |
| 22:23.56 | dijungal | i see there was a revision that introduced an 'X' option... which version of asterisk was that implemented in? |
| 22:24.00 | ManxPower | Hmmhesays: of CDR? |
| 22:24.17 | Hmmhesays | ManxPower yes |
| 22:24.26 | Hmmhesays | cause I have a lot of s extensions |
| 22:24.28 | Hmmhesays | and that just sucks |
| 22:26.09 | killfill | hey.. is it possible to do video-sip calls with nokia phones?.. i.e. n95 |
| 22:26.15 | killfill | or is just for gsm calls... |
| 22:26.21 | Hmmhesays | I am wondering if there is any way around that |
| 22:27.47 | zdsl | killfill: yes you can |
| 22:27.58 | zdsl | Wifi Sip application software |
| 22:28.45 | karleeto | centrex: yes, i just used yum and installed the sources |
| 22:29.00 | killfill | zdsl: is that a sip client for nokia? |
| 22:29.14 | karleeto | centrex: and i see that when my zaptel build (STILL) fails, its using the sources that yum just installed from the trixbox repository |
| 22:29.40 | karleeto | centrex: so why can i not build zaptel? i really need to be able to rebuild it with HPEC enabled |
| 22:30.31 | killfill | zdsl: you dont happend to have a url hidden somwhere, do you?.. :) |
| 22:31.02 | *** join/#asterisk _DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 22:31.12 | karleeto | http://rafb.net/p/HyzzSJ63.html |
| 22:31.20 | karleeto | there is the log of zaptel build failing |
| 22:31.28 | karleeto | I MUST have this resolved and working before morning |
| 22:33.07 | *** join/#asterisk sweeper (i=sweeper@softcheese.net) |
| 22:33.49 | *** part/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 22:34.33 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 22:38.29 | Corydon76-work | In DUNDi, does a TTL of 1 make sense? i.e. does the 1 count the present machine or only how far along the peer chain it asks? |
| 22:39.00 | fujin | SWEET! I got caller-ID based queue prioritisation working! |
| 22:39.06 | fujin | matching against a list of our customers numbers |
| 22:39.10 | fujin | am I awesome or what? :) |
| 22:39.49 | JT | what did you use to acheive that? |
| 22:40.06 | sweeper | magnets |
| 22:40.12 | fujin | macros :( |
| 22:40.21 | tzafrir_laptop | ~centosbug |
| 22:40.21 | jbot | rumour has it, centosbug is a problem with the 2.6.9-42 kernels prior to 2.6.9-42.0.1. If you can't compile zaptel, do a 'yum update', you're running an old kernel. If you HAVE to run an old kernel, the fix is "sed -i s/rw_lock/rwlock/ /usr/src/kernels/*/include/linux/spinlock.h" |
| 22:40.31 | fujin | with goto(${CALLERID(num)}. |
| 22:40.33 | JT | fujin: macros using what to look up the number? |
| 22:40.34 | fujin | seems to be quick |
| 22:40.44 | fujin | I have a pre-generated list of numbers |
| 22:40.46 | *** join/#asterisk s0ck (n=m@unaffiliated/s0ck) |
| 22:40.57 | fujin | like, exten=>s,NUMBER,Macro(dosomething) |
| 22:40.58 | JT | wait, doesn't that mean you need thousands of extensions if you have thousands of numbers? |
| 22:40.58 | tzafrir_laptop | karleeto, look at what jbot just said |
| 22:41.04 | fujin | yes! |
| 22:41.07 | fujin | kind of. |
| 22:41.08 | JT | :o |
| 22:41.10 | fujin | yes. |
| 22:41.21 | fujin | But it's all in-ram. |
| 22:41.28 | fujin | or I assume, anyway |
| 22:41.41 | fujin | I'll pasty what I have. |
| 22:42.30 | *** join/#asterisk axisys (i=iqbala@outbound.silenceisdefeat.org) |
| 22:42.34 | tzafrir_laptop | jbot, no, centosbug is a problem with the 2.6.9 kernels prior to 2.6.9-42 If you can't compile zaptel, do a 'yum update', you're running an old kernel. If you HAVE to run an old kernel, the fix is "sed -i s/rw_lock/rwlock/ /usr/src/kernels/*/include/linux/spinlock.h |
| 22:42.35 | jbot | okay, tzafrir_laptop |
| 22:43.20 | tzafrir_laptop | but the best thing is to upgrade to a newer kernel |
| 22:43.27 | fujin | http://rafb.net/p/wWZ3CT57.html |
| 22:43.32 | fujin | probably not the smartest way to do it |
| 22:43.34 | fujin | but it appears to work :) |
| 22:44.25 | *** join/#asterisk flynux (n=flynux@2a01:38:0:0:0:0:0:1) |
| 22:45.02 | fujin | see how it works jt? |
| 22:46.19 | JT | yeah, not the neatest, but i guess it might work |
| 22:46.38 | JT | you should probably not use commands that were deprecated centuries ago :P |
| 22:46.40 | JT | exten=>s,n,SetVar(QUEUE_PRIO=0) |
| 22:46.56 | fujin | I'm runnin Asterisk 1.2.12.1 built by buildd @ rothera on a i686 running Linux on 2006-09-29 11:07:22 UTC |
| 22:47.00 | fujin | that's why |
| 22:47.05 | fujin | is there a better way to set queue priority now? |
| 22:47.07 | JT | err |
| 22:47.14 | JT | Set is in 1.2 |
| 22:47.22 | fujin | ah |
| 22:47.23 | JT | i never use SetVar in 1.2 |
| 22:47.27 | fujin | so just Set(QUEUE_PRIO=x)? |
| 22:47.32 | JT | yes |
| 22:47.35 | fujin | sweet :) |
| 22:47.36 | fujin | thanks. |
| 22:48.16 | JT | also you don't actually need parentheses after commands with no arguments, ,1,Answer() |
| 22:48.36 | fujin | oh true? |
| 22:48.39 | fujin | awesome |
| 22:48.41 | fujin | didn't know bout that. |
| 22:48.42 | *** part/#asterisk dijungal (n=kdaniel@64.86.52.254) |
| 22:49.27 | JT | which is strange considering you also have 099139683,n,Hangup |
| 22:49.27 | JT | ;) |
| 22:49.47 | fujin | I thought you had to answer() |
| 22:49.48 | fujin | ;( |
| 22:49.48 | Yourname` | Ok, so basically, the inbound box I was mentioning earlier today accepts SIP transfers to extension 1012, and when that happens agents logged in on the inbound box 1000-1010 get rings. But now, once every few calls, instead of going to the agents.. the called person hears dead air. |
| 22:50.36 | JT | fujin: Answering before playback is good, but usually you won't want to Answer before Dial |
| 22:50.52 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 22:53.21 | fujin | nah, I don't answer before dial |
| 22:53.57 | fujin | JT: I don't suppose you care to suggest a better way for me to achieve my callerid matching |
| 22:53.57 | JT | a database :) |
| 22:54.12 | s0ck | JT |
| 22:54.16 | s0ck | can you humour me for a sec |
| 22:54.44 | s0ck | isdn box works fine at site a, move to site b, refuses to make calls in/out |
| 22:54.46 | s0ck | any ideas? |
| 22:54.54 | JT | isdn box.... |
| 22:55.01 | s0ck | asterisk+b410p etc |
| 22:55.07 | s0ck | misdn |
| 22:55.08 | JT | can you be more specific |
| 22:55.09 | JT | well |
| 22:55.17 | *** join/#asterisk Stridernzl (n=neville@125-239-179-163.jetstream.xtra.co.nz) |
| 22:55.19 | JT | are is the telco setup the same? |
| 22:55.26 | s0ck | uk |
| 22:55.28 | s0ck | everything is bt |
| 22:55.36 | s0ck | to my knowledge, all isdn installs are equal |
| 22:55.38 | JT | the isdn settings |
| 22:55.54 | JT | well, one could be set to PTP mode, the other could be PTMP |
| 22:55.55 | s0ck | i assumed it would just work because it was 'just' isdn |
| 22:56.00 | s0ck | i've tried both |
| 22:56.04 | s0ck | neither work |
| 22:56.15 | JT | also, can you define not working |
| 22:56.16 | s0ck | once |
| 22:56.24 | s0ck | te_ptp allowed a single inbound call |
| 22:56.27 | s0ck | which was 90% static |
| 22:56.31 | s0ck | and then that never worked again |
| 22:56.41 | s0ck | i took the box back to site a |
| 22:56.44 | s0ck | instantly worked |
| 22:56.45 | karleeto | tzafrir_laptop: well, doing a yum update shows a lot of packages to be updated, but the kernel is not one of them |
| 22:56.47 | JT | you're saying the pc hardware is exactly the same unit? |
| 22:56.50 | *** join/#asterisk axisys (n=axisys@ip70-174-179-120.dc.dc.cox.net) |
| 22:56.51 | s0ck | yes |
| 22:57.07 | *** join/#asterisk bcnl (n=mike@S010600131078957c.vc.shawcable.net) |
| 22:57.09 | JT | i see, as i said, define not working, are you getting errors? |
| 22:57.13 | s0ck | i'm hoping i've made some schoolboy error here |
| 22:57.16 | s0ck | the error... |
| 22:57.18 | tzafrir_laptop | karleeto, then skip it and use the original fix, or |
| 22:57.28 | s0ck | MDL_ERROR:REQ (tei_l2) <---- |
| 22:57.30 | bcnl | has anyone used IMAP voicemail storage with MicroSoft Exchange 2003 as the back end imap server? |
| 22:57.39 | tzafrir_laptop | revert the respective #ifdef of RWLOCK |
| 22:57.42 | JT | s0ck: did you use the same cable between the NT1 and the card? |
| 22:57.55 | tzafrir_laptop | in zaptel.c and torisa.c |
| 22:58.02 | s0ck | i tried several straight rj45 cables |
| 22:58.06 | JT | ugh, misdn errors are almost indecipherable, but it's blabbering something about Layer 2 and TEI |
| 22:58.07 | s0ck | all produced the same problems |
| 22:58.29 | s0ck | am i right in thinking it should really have just worked? |
| 22:58.42 | JT | maybe |
| 22:58.52 | JT | is there any other bri devices at this problem site? |
| 22:59.12 | s0ck | fresh isdn2 instal |
| 22:59.20 | *** join/#asterisk SwK (n=SwK@24.248.196.141) |
| 22:59.28 | JT | it could be broken, test with something else |
| 22:59.28 | s0ck | i notce the boxes were labelled NTE8 |
| 22:59.40 | s0ck | which are not on the site a boxes which just have 'isdn2e' |
| 22:59.52 | JT | and ask the telco what mode it's set in, and if it's up and running |
| 22:59.56 | JT | not sure what NTE8 is |
| 22:59.59 | bcnl | I'm thinking it's a problem with NTLM authentication? but I don't know for sure |
| 23:01.00 | s0ck | i wasn't sure if i needed all 4 ports plugged into the wall |
| 23:01.03 | s0ck | tried 2 and 4 |
| 23:01.03 | s0ck | same tho |
| 23:01.20 | *** join/#asterisk SwK (n=SwK@24.248.196.141) |
| 23:01.33 | Yourname` | Hello, JT. Doin' good today? :) |
| 23:01.48 | JT | what do you mean all 4 ports plugged into the wall? |
| 23:01.57 | JT | Yourname`: not bad, got into work not that long ago |
| 23:02.09 | s0ck | 4 ports on the back of the b410p, yeh? |
| 23:02.21 | Yourname` | That's good. Would you be willing to help me out today? :D |
| 23:02.22 | s0ck | there are two ports per isdn unit on the wall |
| 23:02.22 | JT | s0ck: yes, there are |
| 23:02.28 | s0ck | again, four in total |
| 23:02.40 | JT | Yourname`: if you're willing to describe your problem |
| 23:02.43 | s0ck | i noticed i can make/receive two calls with just one rj45 cable in the site a |
| 23:02.44 | JT | s0ck: umm |
| 23:02.47 | s0ck | so er, wasn't sure |
| 23:02.47 | Yourname` | I will. |
| 23:02.55 | JT | s0ck: how many bris do you have installed? |
| 23:03.01 | s0ck | 2 |
| 23:03.18 | JT | how many cables connect from the b410p? |
| 23:03.29 | s0ck | i tried one to each unit |
| 23:03.33 | s0ck | then two to each |
| 23:03.40 | JT | well two won't work |
| 23:03.55 | s0ck | nein? |
| 23:03.57 | JT | you need 1 connection to the S-bus connector |
| 23:04.09 | s0ck | the who-bus? |
| 23:04.14 | JT | it's a digital bidirectional differential serial bus |
| 23:04.29 | JT | the S-bus is the digital circuit coming out of the NT1 |
| 23:04.45 | s0ck | in laymans terms, the rj45 port? |
| 23:05.03 | JT | rj-45 sockets can have anything on them |
| 23:05.07 | JT | it should be labelled. |
| 23:05.12 | Yourname` | JT: Basically, I have two boxes. One is an outbound box, and then when the called person picks up, the outbound call transfers that call to extension 1012 on the inbound box. Now, when the inbound box sees an incoming call, via SIP transfer or inbound DID call, it will ring all agents' phones. Perfect scenario. Everything is working. Now, once every few calls like this, the transfer comes in... |
| 23:05.13 | Yourname` | ...on the inbound box via SIP, BUT, in the CLI it shows "Called 1012@itsp" and little more stuff.. and none of the agents phones rings, and neithr do I see anything in the CLI. That's the problem. |
| 23:05.56 | s0ck | not sure about that.... |
| 23:06.04 | s0ck | there are two rj45 per unit |
| 23:06.11 | s0ck | and one standard bt analogue too |
| 23:06.15 | JT | s0ck: what other connections are there? |
| 23:06.31 | s0ck | are you uk? i forget... |
| 23:06.37 | JT | no |
| 23:06.40 | JT | australia |
| 23:07.01 | JT | there's a few variants of NT1s, but they all do pretty much the same thing |
| 23:07.17 | s0ck | http://www.draytek.co.uk/products/image/isdn2e_box1.jpg |
| 23:07.20 | s0ck | that's what it looks like |
| 23:07.33 | s0ck | two of those on the wall |
| 23:07.37 | s0ck | each has two rj45 |
| 23:07.51 | s0ck | one per 64kb channel i assumed |
| 23:07.55 | JT | very small pic |
| 23:07.58 | JT | bad assumption |
| 23:08.03 | s0ck | admittedly, it's a shit pic |
| 23:08.27 | s0ck | the assumption came from when i had home isdn: |
| 23:08.28 | s0ck | http://www.draytek.co.uk/products/image/highway_box1.jpg |
| 23:08.38 | s0ck | i could plug each pc in the (blue) rj45 |
| 23:08.41 | JT | i thought you said you had an NTE8 this time, not an isdn2e |
| 23:08.49 | s0ck | or just one into one and get 128k |
| 23:09.02 | s0ck | it says something like nte8 on it |
| 23:09.06 | s0ck | but it looks like the first pic |
| 23:09.08 | JT | if you have two cables connected at the other site, it's a miracle it works |
| 23:09.08 | s0ck | standard isdn |
| 23:09.21 | s0ck | i dont, it was only ever one at the other site |
| 23:09.45 | s0ck | so, even for a 4 channel setup |
| 23:09.48 | s0ck | only one cable...? |
| 23:10.00 | JT | you can't do 4 channels on one BRI. |
| 23:10.07 | JT | you'll need one cable per bri |
| 23:10.10 | s0ck | k |
| 23:10.16 | *** part/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 23:10.47 | s0ck | this is what i had originally when i walked in this morning |
| 23:10.51 | s0ck | alas, it would not work |
| 23:11.14 | s0ck | didn't try te_ptp with just one per bri tho |
| 23:11.19 | s0ck | so that's something i can try tomorrow |
| 23:11.48 | JT | here's what to do |
| 23:11.52 | JT | ring BT tech |
| 23:12.06 | JT | and ask "hello, what mode is my bri set to, is it up and running?" |
| 23:12.14 | JT | and specify each circuit |
| 23:12.57 | s0ck | "what do you mean, "mode". i am not very familiar with isdn i think it must be your pbx at fault, what pbx did you say it was? let me get someone from sales to call you." |
| 23:13.03 | s0ck | that's the response i got last time |
| 23:13.13 | s0ck | admittedly, i did have it configured wrong but still |
| 23:13.15 | s0ck | :P |
| 23:13.24 | s0ck | i will give them a ring tomorrow, you've been a big help :) |
| 23:14.34 | JT | speak to the right person |
| 23:14.51 | s0ck | that was a bt engineer, believe it or not |
| 23:14.59 | JT | "is my isdn basic rate circuit in point to point or point to multipoint mode?" |
| 23:15.05 | *** join/#asterisk joaop (n=taken@189.30.80.198) |
| 23:15.11 | JT | probably the wrong "engineer" |
| 23:15.18 | s0ck | tried to tell me there was rj11 on the isdn box at first |
| 23:15.26 | s0ck | i knew it was going to be an entertaining conversation |
| 23:15.51 | JT | also your termination could be set wrong |
| 23:16.06 | s0ck | whassat? |
| 23:16.15 | JT | s-bus termination |
| 23:16.21 | s0ck | would that be the in/out switches on the bottom of the unit |
| 23:16.21 | JT | how long is the cable? |
| 23:16.30 | s0ck | 2-3M |
| 23:16.55 | JT | i'm talking about appropriate levels of termination with resistors to prevent reflections which would cause errors |
| 23:17.02 | s0ck | k |
| 23:17.33 | JT | the shorter the cable, the more likely you will get reflections |
| 23:17.45 | JT | you can set termination in the NT1, and on the card usually |
| 23:17.54 | JT | not familiar with the B410P |
| 23:18.02 | JT | but it might have termination settings |
| 23:18.05 | *** join/#asterisk ming_zy1 (n=ming_zym@124.254.52.110) |
| 23:18.19 | bkruse | can you do inline if's with Set? like Set(CALLERID(num) ? ${CALLERID(num)} : 'idk the cid') |
| 23:18.21 | s0ck | there are jumpers on it, if that's any help |
| 23:18.29 | s0ck | but at the end of the day, it works at site a |
| 23:18.33 | s0ck | so... |
| 23:18.40 | JT | so it's a different site. |
| 23:18.57 | JT | my octobri card has jumpers for te/nt mode and termination |
| 23:19.06 | *** join/#asterisk remmo (n=junk@203.62.147.8) |
| 23:19.14 | s0ck | the card is definitely set to te |
| 23:19.24 | s0ck | dunno about the termination stuff |
| 23:20.35 | Hmmhesays | does anyone have a british telecom rate list? |
| 23:20.36 | JT | in the meantime i suggest you read up on bri |
| 23:22.21 | bkruse | anyone know about a dialplan set statement such as that? |
| 23:22.21 | s0ck | rate of what |
| 23:23.01 | s0ck | the sbus you speak of i think is terminated as rj45 with only the centre pair used |
| 23:23.05 | s0ck | is it not rj45 in oz? |
| 23:23.21 | JT | yes it's an 8P8C connector here |
| 23:23.28 | JT | commonly mistakingly refered to as RJ45 |
| 23:23.29 | JT | :) |
| 23:23.33 | s0ck | hehe |
| 23:23.40 | s0ck | the service light locks on when the cable is attached |
| 23:23.44 | JT | rj45 specifies connector and technology |
| 23:23.59 | Stridernzl | if i have a sip client logged into asterisk .... but have call forwarding enabled can i still pump out which line was ringing to the sip client without interferring with the call forwarding process ... eg all i want the sip client to do is tell me which line was ringing? |
| 23:24.08 | JT | and yeah, only centre pair used? that's wrong. |
| 23:24.32 | s0ck | pairs |
| 23:24.33 | s0ck | my bad |
| 23:24.36 | JT | pins 3,4,5,6 are used |
| 23:25.59 | s0ck | can you recommend any literature |
| 23:26.16 | JT | i had some good ones, but that pc crashed |
| 23:26.29 | JT | for cabling http://www.zytrax.com/tech/layer_1/cables/telephone.html |
| 23:26.34 | Yourname` | Jul 31 15:28:34 NOTICE[3058]: pbx.c:1773 pbx_extension_helper: No such label 'agent_talk' in extension '1012' in context 'default' |
| 23:26.38 | Yourname` | What does this mean? |
| 23:26.47 | Yourname` | And what if I want it to actually execute "agent_talk" |
| 23:27.59 | JT | then you should probably add an agent_talk priority label |
| 23:28.03 | s0ck | cheers dood |
| 23:28.15 | Yourname` | priority label? Context? |
| 23:30.14 | bcnl | has anyone used IMAP voicemail storage with MicroSoft Exchange 2003 as the back end imap server? |
| 23:30.58 | s0ck | no... sounds interesting tho |
| 23:31.00 | s0ck | does it work |
| 23:31.15 | Yourname` | lol dude, it's hard for people to get regular answers RELATED to asterisk.. you're gonna find an answer to an off-topic unrelated question? lol.. |
| 23:31.34 | Yourname` | But wait, it actually does look slightly related. |
| 23:31.39 | *** part/#asterisk keith4_ (n=kbe2@216-164-160-36.c3-0.eas-ubr10.atw-eas.pa.static.cable.rcn.com) |
| 23:31.45 | JT | s0ck: http://www.ralphb.net/ISDN/ |
| 23:31.59 | bcnl | s0ck: nope |
| 23:32.06 | JT | s0ck: the agregate bitrate specified there for E1 is wrong though |
| 23:32.08 | bcnl | Yourname`: yea it's kinda related, it's imap voicemail |
| 23:33.01 | bcnl | since the newer 4.2 blackberry OS supports the wav files that asterisk generates for voicemail, it'd be dreamy to have truely centralized messaging |
| 23:33.05 | *** join/#asterisk BadPacket (n=NoCarrie@unaffiliated/badpacket) |
| 23:34.18 | s0ck | e1 is pri init? im only on the small stuff atm :P |
| 23:34.40 | JT | E1 can be PRI |
| 23:34.51 | *** join/#asterisk Jon335 (n=chatzill@unaffiliated/jon335) |
| 23:34.52 | s0ck | Terminal Endpoint Identifiers |
| 23:34.58 | s0ck | was wondering what that stood for |
| 23:35.08 | bcnl | Yourname`: it workes from the mtest command line option as part of the imap sourcecode |
| 23:35.24 | Yourname` | ah |
| 23:35.55 | ManxPower | TEI is mostly a BRI term, I believe. |
| 23:36.10 | JT | it is |
| 23:36.18 | bcnl | problem is the docs for imap storage are pretty sparse |
| 23:38.27 | s0ck | bcnl |
| 23:38.30 | s0ck | now you mention it |
| 23:38.34 | s0ck | i've done just that |
| 23:38.47 | s0ck | i've had * email me the messages |
| 23:38.52 | s0ck | and i use exchange 2003 |
| 23:38.58 | bcnl | s0ck: yea I have emailing of messages work |
| 23:39.13 | bcnl | but * can actually store the messages in a IMAP folder |
| 23:39.30 | bcnl | so when you click on the message in the folder of your email client, it's marked as read on your MWI |
| 23:39.38 | Stridernzl | can a message just be sent to a sip device without making it ring ? and where to look for such docs/tutorials ? |
| 23:39.39 | s0ck | ah |
| 23:39.43 | s0ck | that would be rather cool |
| 23:40.20 | s0ck | i guess it needs a local imap client for that to work? |
| 23:42.09 | bcnl | I've compiled it with it |
| 23:42.21 | bcnl | I'm even watching it fail with tcpdump |
| 23:42.32 | bcnl | * seems to LOGOUT right after getting a response to CAPABILITIES |
| 23:42.50 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 23:43.12 | ManxPower | Stridernzl: That depends on the SIP device. Some support some don't. All that do support it do it differently. |
| 23:44.50 | s0ck | is CAPA sent with imap too? |
| 23:44.51 | Stridernzl | manxpower: ... I have a remote extension .. but its not so good quality .. so have have anable call forwarding at the * machine .... I have 2 lines so thats all easy .. but i would still like to send a message to the sip client (eyebeam) and say X line is ringing .. that way I would know exactly what line was called and how best to answer the phone |
| 23:44.56 | s0ck | thought it was pop3 |
| 23:45.41 | s0ck | btw, general * question |
| 23:45.51 | s0ck | can someone recommend me a phone that actually supports blf |
| 23:46.03 | bcnl | so besides exchange, is anyone using asterisk voicemail with the IMAP store? |
| 23:46.52 | VxJasonxV | If you have one person with a hard IP phone, and soft phone, both configured to use the same extension, will each device clobber the other (i.e. kick the other one off) when Asterisk recieves a REGISTER for that extension? |
| 23:49.07 | *** join/#asterisk xpot (n=jim@c-71-195-241-115.hsd1.ma.comcast.net) |
| 23:49.47 | fujin | JT: still around? |
| 23:50.06 | ManxPower | Stridernzl: find out what protocol Eyebeam uses for text messages, then come back. |
| 23:50.20 | ManxPower | s0ck: Polycom |
| 23:50.22 | fujin | it uses sip |
| 23:50.25 | *** join/#asterisk sxntixgo (n=santiago@127-11-17-190.fibertel.com.ar) |
| 23:50.46 | ManxPower | fujin: Which RFC? |
| 23:51.06 | ManxPower | VxJasonxV: A device is not an extension. |
| 23:51.09 | fujin | http://www.faqs.org/rfcs/rfc3428.html |
| 23:51.11 | sxntixgo | hi all, i need some help installin asterisk and h323 |
| 23:51.12 | fujin | 3482 |
| 23:51.31 | ManxPower | an extension is a digit or text mapping to/from a device (in your case SIP) |
| 23:51.56 | ManxPower | You can have the same exten => line find multiple devices. You cannot have different devices register as the same SIP userID |
| 23:52.29 | ManxPower | fujin: that is a messaging extension for SIP. |
| 23:52.44 | ManxPower | If nothing else you could use sipsak to do the text message using the System command. |
| 23:52.47 | fujin | yes, and that's what Eyebeam uses for text messages |
| 23:53.08 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 23:53.09 | ManxPower | You can also try the SendText application in Asterisk, I don't which specific implimention it uses for SIP |
| 23:53.22 | JT | fujin: so it's not actually sip. |
| 23:53.26 | JT | fujin: and yes i'm here |
| 23:53.31 | fujin | it's an extension to SIP |
| 23:53.41 | JT | right |
| 23:53.43 | fujin | JT: was just going to ask what you meant by use a database |
| 23:53.46 | JT | it's not sip itself |
| 23:53.50 | ManxPower | Generally it is a VERY VERY VERY bad idea to make your extensions in extensions.conf be the same as your SIP User IDs in sip.conf |
| 23:54.18 | JT | a lot of things won't support said extension |
| 23:54.18 | ManxPower | It limits what you can do, it limits your thinking, and brainwashes you into thinking a device is an extension |
| 23:54.52 | JT | fujin: postgresql or similar |
| 23:55.21 | fujin | with odbc? |
| 23:55.26 | ManxPower | Polycoms, for example support MSN (or is it Microsoft) Messenger text messages |
| 23:55.38 | JT | fujin: yes |
| 23:55.45 | ManxPower | I believe they also support SIMPLE messages. They do not support SendText |
| 23:55.51 | fujin | this might be pushing it, but got an example? |
| 23:56.02 | JT | nope |
| 23:56.11 | fujin | sweet |
| 23:56.28 | fujin | are you saying I can database the exten=>s,THISPART, of my stuff? |
| 23:56.47 | sxntixgo | hi all, i need some help installin asterisk and h323 |
| 23:57.12 | fujin | apt-get install asterisk |
| 23:57.34 | ManxPower | sxntixgo: You are far beyond our help then. |
| 23:57.52 | sxntixgo | oh... but im ussing fedora |
| 23:57.55 | s0ck | ManxPower: you've had it working personally? flashy light/call pickup/etc? |
| 23:58.00 | Stridernzl | Manxpower: .... Umm thats hard to find ? But i also run skype and msoft messenger on that machine so I could send a message to any of them / ie perhaps it easier to message one of the other apps ?? |
| 23:58.18 | fujin | yum install asterisk |
| 23:58.26 | sxntixgo | lol |
| 23:58.26 | ManxPower | Stridernzl: Do you have Polycoms? |
| 23:58.32 | JT | fujin: stop recommend people use packages |
| 23:58.42 | JT | sxntixgo: compile from source |
| 23:58.54 | fujin | lol |
| 23:58.54 | Stridernzl | manxpower: no .. all clients are soft |
| 23:58.57 | fujin | what's wrong with packages |
| 23:58.58 | ManxPower | Stridernzl: most people do not use text messaging with SIP phones. You would should read the mailing list archives. |
| 23:59.01 | ManxPower | ~mailinglist |
| 23:59.01 | jbot | Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.htmm |
| 23:59.04 | sxntixgo | im traing to compile but i get an error |
| 23:59.05 | JT | ... |
| 23:59.10 | JT | especially with h.323 |
| 23:59.12 | ManxPower | fujin: for one thing they always put all the files in weird ass places. |
| 23:59.30 | fujin | they put the files in $PREFIX |
| 23:59.33 | fujin | just like make install would |
| 23:59.46 | fujin | the ones I use do, anyway |
| 23:59.56 | ManxPower | The file you need to read is /path/to/src/asterisk/docs/README.whatever Oh, you are using a package? Then find that file your own damn self. |
