| 00:02.35 | [TK]D-Fender | hmmhesays, IIRC its only the 1st |
| 00:07.48 | ThatKidKel | JunK-Y.. I've defined canreinvite=no in the peer entry in sip, and have it dialing as Dial(SIP/1234567890@PEERNAME) but it is still trying the reinvite |
| 00:11.37 | hmmhesays | [TK]D-Fender: yeah I can't find anything that says otherwise |
| 00:12.01 | hmmhesays | I can't seem to find the microbrowser settings via the phones lcd either, is that config file only? |
| 00:12.22 | De_Mon | putnopvut you around? |
| 00:12.29 | Qwell | De_Mon: he left for the day |
| 00:12.35 | [TK]D-Fender | hmmhesays, web/provisioning |
| 00:12.56 | [TK]D-Fender | hmmhesays, Provisioning is the ONLY way you should be configuring those |
| 00:13.12 | De_Mon | ahh well. I just tested his patch, I'll post bug notes instead |
| 00:13.18 | Qwell | [TK]D-Fender: Do you know where to get some good polycom configs that are commented? |
| 00:13.50 | De_Mon | is it safe to run bootp over the internet? |
| 00:13.51 | [TK]D-Fender | Qwell, admin guide says it all. XML is not made to be commented. |
| 00:13.58 | De_Mon | for voip phones I'm speaking of |
| 00:14.04 | [TK]D-Fender | Qwell, and the WIKI has a good breakdown |
| 00:14.05 | Qwell | [TK]D-Fender: okay, but it says which options can be used, and what they do? |
| 00:14.14 | [TK]D-Fender | Qwell, explains them all |
| 00:15.48 | hmmhesays | [TK]D-Fender: yeah poking around I don't even see where you access the microbrowser |
| 00:17.22 | [TK]D-Fender | hmmhesays, on a 501 its that nice button labelled "Services" :) |
| 00:17.38 | *** join/#asterisk ExplodingLemur (n=change@72.29.166.25) |
| 00:17.46 | hmmhesays | there we go |
| 00:17.50 | hmmhesays | i see it on the 601 |
| 00:18.27 | [TK]D-Fender | hmmhesays, non 6XX series got the MB as of SIP 2.1.0+ |
| 00:18.31 | *** join/#asterisk the007killer (n=the007ki@59.154.97.58) |
| 00:18.35 | the007killer | hi everyone |
| 00:19.41 | the007killer | i have a little asterisk issue i cant seem to work out |
| 00:20.33 | ExplodingLemur | Question with the AsteriskGUI...I've done a fresh install of Asterisk 1.4.15 and the latest SVN checkout of the GUI. What level of configuration do I need to do before the GUI will work properly? Right now I've got only the http.conf and manager.conf files and the GUI complains about a missing users.conf. If I touch users.conf the GUI then doesn't complain about the file being missing, but won't progress past the initial config page where I set t |
| 00:20.38 | hmmhesays | now I wonder if it is possible to make a different home page for each phone |
| 00:21.01 | hmmhesays | without having to create a new sip.cfg for every one |
| 00:21.12 | Qwell | I assume they send mac along with the request |
| 00:21.15 | Qwell | Cisco's do |
| 00:21.20 | [TK]D-Fender | ExplodingLemur, please ask in their channel as listed in the channel topic |
| 00:21.47 | [TK]D-Fender | hmmhesays : yup, just copy the sip.cfg tag section to each phone's config |
| 00:22.04 | ExplodingLemur | Huh....that scrolled all the way past the end of my channel window. Quite the topic line! I'll head over there, sorry. |
| 00:22.12 | *** part/#asterisk ExplodingLemur (n=change@72.29.166.25) |
| 00:22.35 | hmmhesays | [TK]D-Fender: the config file where I specify sip.cfg as the master sip config file? |
| 00:22.42 | *** join/#asterisk fujin_ (n=aj@unaffiliated/fujin) |
| 00:23.10 | [TK]D-Fender | hmmhesays, no. in the base provisioning samples they use sip.cfg & phone1.cfg in your <mac>.cfg |
| 00:23.53 | [TK]D-Fender | hmmhesays, what I do and advise is to make phoneXXX based on your individual phone and line each <mac>.cfg to sip.cfg and its resptective phoneXXX.cfg |
| 00:24.41 | [TK]D-Fender | hmmhesays, so template up the phone1.cfg to 99% complete and then copy it around changing only what makes a phone unique. I usually keep 1 master template per model for linekeys, etc. |
| 00:25.40 | hmmhesays | so does the order you list in your <mac>.cfg matter? |
| 00:26.08 | hmmhesays | sip.cfg, phoneXX.cfg if you have a duplicate setting phoneXX.cfg will overwrite whatever it read out of sip.cfg? |
| 00:26.46 | [TK]D-Fender | hmmhesays, generally yup |
| 00:27.01 | [TK]D-Fender | hmmhesays, kep the same order and you need only rename as I suggested. |
| 00:29.24 | *** join/#asterisk mog (n=mog@c-68-62-172-83.hsd1.al.comcast.net) |
| 00:29.24 | *** mode/#asterisk [+o mog] by ChanServ |
| 00:36.12 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 00:37.17 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 00:39.59 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 00:40.21 | *** part/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 00:59.24 | *** part/#asterisk Navion (n=billp@75-105-41-123.cust.wildblue.net) |
| 00:59.26 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 01:00.17 | the007killer | i have an issue with the config file res_mysql |
| 01:00.39 | the007killer | i seem to have entered the informtion correctly but itdoesn't want to seem to work |
| 01:00.43 | the007killer | cdr_addon_mysql.c:323 my_load_module: MySQL server hostname not specified. Assuming localhost |
| 01:00.55 | the007killer | this is the error i keep getting |
| 01:02.18 | [TK]D-Fender | the007killer, that is a warning, not an "error" |
| 01:02.36 | *** join/#asterisk ming_zym (n=ming_zym@124.14.234.68) |
| 01:03.06 | the007killer | but i get this later |
| 01:03.11 | the007killer | vERROR[23953]: cdr_addon_mysql.c:446 my_load_module: Failed to connect to mysql database ftelaccounts on localhost. |
| 01:03.50 | the007killer | localhost doesn't exist on this computer as it is a vserver hence i type in the complete ip address in res_mysql but it doesn't seem to pick it up |
| 01:04.27 | hmmhesays | getting the microbrowser to display a picture sucks |
| 01:07.59 | [TK]D-Fender | hmmhesays, in what way? |
| 01:15.12 | *** join/#asterisk ak_hepcat (n=ak_hepca@patroler.gci.net) |
| 01:20.46 | *** join/#asterisk the007killer (n=the007ki@59.154.97.58) |
| 01:27.00 | ak_hepcat | anybody able to answer a question on SIP trunk groups, specifically multi-line trunks (>15) |
| 01:31.32 | *** join/#asterisk mog (n=mog@c-68-62-172-83.hsd1.al.comcast.net) |
| 01:31.32 | *** mode/#asterisk [+o mog] by ChanServ |
| 01:35.44 | blitzrage | ak_hepcat: just go ahead and ask |
| 01:37.47 | ak_hepcat | well, i've got simple SIP endpoints defined, and can make calls through them from my IP phone to my providor. In talking with them, they want to offer me a "real" trunk group of numbers (24) with a block-range - which after discussing with them, means i register to the trunk-group and not individual 'lines'. I've been looking for info on how to set that up, and have been frustrated, so i'm basically asking if this is even possible. |
| 01:39.28 | hmmhesays | [TK]D-Fender: i can't get the idle screen to display a pic |
| 01:39.41 | [TK]D-Fender | hmmhesays, PB your MB file |
| 01:40.27 | hmmhesays | <img src="pic1.bmp" width="20" height="40" /> |
| 01:40.34 | hmmhesays | is the page i'm trying to load |
| 01:41.17 | *** join/#asterisk darkskiez (n=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) |
| 01:41.34 | [TK]D-Fender | hmmhesays, avoide the resize and <img src="pic1.bmp"></img> |
| 01:43.50 | [TK]D-Fender | ak_hepcat, I would bet that this would mean that you'd set up 1 account with them and you can set yoru callerid to any DID you have with them and use a single peer entry to dialout. |
| 01:44.28 | [TK]D-Fender | ak_hepcat, and that inbound would all come in from the same auth and target the appropriate DID. Channel limiting would be done by your provider to keep you under your limit. |
| 01:45.23 | ak_hepcat | hmm.. okay. i think i see how, then. I'll give it a try next week when i can get ahold of 'em. Thanks. |
| 01:47.13 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 01:53.15 | *** join/#asterisk optize (i=tyler@ip70-176-254-41.ph.ph.cox.net) |
| 01:53.28 | optize | is there a way to ignore/not answer registration packets? |
| 01:53.35 | optize | i dont want people to be able to register, or get anything bad when they try to register |
| 01:53.59 | [TK]D-Fender | optize, run SER in front of * |
| 01:54.26 | optize | ? |
| 01:54.26 | [TK]D-Fender | ~ser |
| 01:54.26 | jbot | it has been said that ser is Sip Express Router - see http://www.iptel.org/ser/, or an old secret method of obtaining a havoc of NAT problems, or at #ser |
| 01:55.06 | optize | hmm |
| 01:56.18 | optize | thanks? |
| 01:56.19 | optize | heh |
| 01:56.36 | [TK]D-Fender | optize, * cannot do this. So if thats what you wat, you're going to have to have a proxy filtering all SIP communications |
| 01:56.52 | optize | i'll try it out |
| 01:57.00 | optize | thanks :> |
| 01:57.06 | [TK]D-Fender | optize, SER has a big learning curve |
| 01:57.13 | optize | oh, wonderful |
| 01:57.13 | [TK]D-Fender | optize, Whats the issue with registering? |
| 01:57.30 | optize | its a backup asterisk server, i dont want people to be able to register ot it, just invite to it.... so people wont stay connected to it |
| 01:57.37 | optize | and will continue to try to register to the primary |
| 01:57.48 | optize | i use dns srv to control it as a backup |
| 01:58.12 | *** join/#asterisk rcahilig (n=user@203.115.187.98) |
| 02:02.12 | *** join/#asterisk angom (n=Angel@201.170.35.218) |
| 02:02.34 | rcahilig | Hi, Anyone please help me about Asterisk recording using mixmonitor or monitor. We are using Asterisk 1.2 in a 10 seater call center, I want to record each every agents calls and their recordings will be saved in their respective folders. Every agents should have its own folder... |
| 02:04.01 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 02:08.56 | [TK]D-Fender | rcahilig, What kind of Members does your queue use? |
| 02:13.31 | rcahilig | [TK]D-Fender:sorry I'm no expert about queue, I don't think I will need queue, were outbound call center |
| 02:14.10 | [TK]D-Fender | rcahilig, even easier then. youcan base it on their caller ID or similar then |
| 02:15.10 | rcahilig | [TK]D-Fender:WOW, can you assist me? sorry I'm too lame on this... |
| 02:15.45 | [TK]D-Fender | rcahilig, What are your agents using for phones? |
| 02:17.13 | rcahilig | [TK]D-Fender:Were only using soft phones, our asterisk server don't have any hardware, I'm using ztdummy |
| 02:17.50 | rcahilig | the asterisk pbx is connected to a DSL |
| 02:18.11 | [TK]D-Fender | rcahilig, Then do something like "MixMonitor(outbound/${CALLERID(num)}/${TIMESTAMP}.wav" |
| 02:18.41 | [TK]D-Fender | rcahilig, Before they dial out and it'll record their calls into folders based on their softphone's CID |
| 02:19.18 | rcahilig | [TK]D-Fender: do I need to configure queue? |
| 02:19.58 | [TK]D-Fender | rcahilig, I kinda assumed at first you were talking about an INBOUND call center. |
| 02:20.38 | [TK]D-Fender | rcahilig, I have no idea how you'd want to run an outbound one |
| 02:22.11 | rcahilig | <PROTECTED> |
| 02:22.52 | rcahilig | [TK]D-Fender: I will configure this in every user in the dial plan |
| 02:23.11 | [TK]D-Fender | rcahilig, you'd put that before any exten they would dial out using. |
| 02:23.34 | [TK]D-Fender | rcahilig, that is just a sample of course. |
| 02:24.27 | *** join/#asterisk coppice (n=chatzill@235.202.17.210.dyn.pacific.net.hk) |
| 02:25.04 | rcahilig | [TK]D-Fender: example exten => _NXXXXXX,1,MixMonitor(outbound/${CALLERID(num)}/${TIMESTAMP}.wav for the user_1 |
| 02:25.13 | rcahilig | [TK]D-Fender: example exten => _NXXXXXX,1,MixMonitor(outbound/${CALLERID(num)}/${TIMESTAMP}.wav for the user_2 |
| 02:25.39 | [TK]D-Fender | rcahilig, Assuming those extens are in spereate contexts and each user is pointed to their own, sure |
| 02:27.37 | darkskiez | rcahilig: its good having them not mixed, so u can tell who said something =] otherwise the person could say 'you ***t' and claim they'd been called it =] |
| 02:27.54 | rcahilig | [TK]D-Fender: I suppose I have 10 agents, I will create a very long dial plan |
| 02:28.18 | darkskiez | rcahilig: u not have an outbound macro u could put it in? |
| 02:28.19 | *** join/#asterisk znoG (n=gs@130-215-114-200.fibertel.com.ar) |
| 02:28.39 | [TK]D-Fender | rcahilig, why don't you just use 1 context with that in it instead of multiple? |
| 02:31.01 | rcahilig | [TK]D-Fender: 1 context like this and multiple extensions. exten => _NXXXXXX,1,MixMonitor(outbound/${User1(num)}/${TIMESTAMP}.wav |
| 02:31.01 | rcahilig | exten => _NXXXXXX,1,MixMonitor(outbound/${User2(num)}/${TIMESTAMP}.wav |
| 02:31.02 | rcahilig | exten => _NXXXXXX,1,MixMonitor(outbound/${User3(num)}/${TIMESTAMP}.wav |
| 02:31.20 | [TK]D-Fender | rcahilig, CALLERID is afixed FUNCTION. |
| 02:31.42 | blitzrage | anytime you run into that type is situation... you should be using either a macro or some sort of loop (depending on the situation) |
| 02:31.44 | [TK]D-Fender | rcahilig, ${User3(num)} <- this does not exist |
| 02:31.53 | [TK]D-Fender | blitzrage, he needs neither. |
| 02:32.03 | [TK]D-Fender | blitzrage, and simply does not understanda the dialplan at all |
| 02:32.14 | rcahilig | [TK]D-Fender: ahhh, how do I put multiple users in this extension? |
| 02:32.27 | blitzrage | I don't understand why he needs neither -- you should never need to put a specific line in for every unique identifier |
| 02:32.36 | [TK]D-Fender | rcahilig, you don't put users in an EXTENSION. Users DIAL and extension. |
| 02:32.45 | rcahilig | [TK]D-Fender: I understand the dialplan a little bit |
| 02:33.00 | [TK]D-Fender | rcahilig, Looking less and less like you do... |
| 02:33.05 | blitzrage | [TK]D-Fender: chill |
| 02:33.17 | blitzrage | rcahilig: what is it you're trying to do again? |
| 02:33.43 | [TK]D-Fender | blitzrage, just call a simple monitor for his outgoing agents calls. |
| 02:34.04 | rcahilig | I want to record each every agents calls and their recordings will be saved in their respective folders. Every agents should have its own folder to store their recordings |
| 02:34.06 | [TK]D-Fender | blitzrage, and not understanding they can all use the same extens in the same context |
| 02:34.31 | blitzrage | rcahilig: how are you calling the agents? |
| 02:34.36 | [TK]D-Fender | blitzrage, example exten => _NXXXXXX,1,MixMonitor(outbound/${CALLERID(num)}/${TIMESTAMP}.wav) <- recording in folder per agent based on CID |
| 02:34.41 | [TK]D-Fender | blitzrage, OUTBOUND |
| 02:34.47 | blitzrage | that'd work |
| 02:34.47 | [TK]D-Fender | blitzrage, I got caught on that too :) |
| 02:35.00 | rcahilig | [TK]D-Fender: outbound |
| 02:35.07 | blitzrage | outbound from the viewpoint of Asterisk? |
| 02:35.08 | *** join/#asterisk saftsack (n=saftsack@p4FC7764E.dip.t-dialin.net) |
| 02:35.19 | blitzrage | or agents placing outbound calls |
| 02:35.25 | [TK]D-Fender | blitzrage, agents call out and telemarket people :) |
| 02:35.33 | blitzrage | gotcha |
| 02:35.38 | [TK]D-Fender | blitzrage, Shoot him now before its too late! :p |
| 02:36.59 | rcahilig | [TK]D-Fender: Hi here is my dial plan: exten => _NXXNXXXXXX,1,NoOp(${CALLERID(num)}) |
| 02:37.00 | rcahilig | exten => _NXXNXXXXXX,n,Set(CALLERID(num)=8647100) |
| 02:37.00 | rcahilig | exten => _NXXNXXXXXX,n,NoOp(${CALLERID(num)}) |
| 02:37.00 | rcahilig | exten => _NXXNXXXXXX,n,Dial(${B2GO}/${B2GOPRE}${EXTEN}) |
| 02:37.00 | rcahilig | exten => _NXXNXXXXXX,n,Hangup() |
| 02:37.59 | blitzrage | if you need to change the CID before dialing out, you need to save the CID of the unique agent first, then use that variable as the filename, then you can change the CID for the outbound call |
| 02:38.39 | [TK]D-Fender | rcahilig, Here's your NEW step #1 : exten => _NXXNXXXXXX,1,MixMonitor(outbound/${CALLERID(num)}/${TIMESTAMP}.wav) |
| 02:40.18 | [TK]D-Fender | rcahilig, And there is no point in NoOp-ing your CID right after you set it. You KNOW what it's holding |
| 02:42.36 | scooby2 | http://pastebin.com/d594f84cf |
| 02:42.52 | scooby2 | for some reason after playing the welcome message it hangs up immediately |
| 02:43.01 | scooby2 | my IVR |
| 02:45.22 | blitzrage | scooby2: WaitExten() |
| 02:45.38 | jer | hrmm. i must be retarded; is this not correct to strip of 9 for a 10 digit number: exten => _9NXXNXXXXXX,1,Dial(Zap/1/${EXTEN:1}) ? |
| 02:45.41 | blitzrage | core show application WaitExten |
| 02:46.30 | scooby2 | thank you |
| 02:46.35 | scooby2 | is that new in 1.4? |
| 02:47.09 | blitzrage | yes... it's because there is autofallthrough turned on |
| 02:47.28 | blitzrage | we document it (somewhere) in TFoT |
| 02:47.44 | blitzrage | probably in the opening chapters |
| 02:47.44 | Qwell | and in UPGRADE.txt |
| 02:47.50 | blitzrage | oh ya ... that makes sense :) |
| 02:48.25 | scooby2 | whats this TFoT? |
| 02:48.33 | Qwell | ~book |
| 02:48.34 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 02:49.02 | scooby2 | thank you |
| 02:49.50 | scooby2 | i think that confirms this consultant was pretty bad |
| 02:50.14 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 02:51.12 | scooby2 | i really hate consultants |
| 02:51.27 | coppice | no consultant who gets paid is bad. that is their only reason to exist |
| 02:51.44 | Qwell | no consultant who gets paid is bad? |
| 02:51.52 | Qwell | why do I think there should be a second negative there? |
| 02:51.57 | scooby2 | oxymoron |
| 02:52.13 | *** join/#asterisk JayTee52 (n=jforde05@c-69-137-243-25.hsd1.in.comcast.net) |
| 02:52.36 | [TK]D-Fender | scooby2, just set "autofallthrough=no" under [general] |
| 02:52.45 | coppice | on time. on spec. on budget - they 3 key no nos of consulting |
| 02:53.04 | Qwell | coppice: a (good) consultant should do none of those? |
| 02:53.06 | *** join/#asterisk nirz (n=nir@bzq-79-183-111-228.red.bezeqint.net) |
| 02:53.17 | [TK]D-Fender | coppice, Good. Cheap. Fast. Pick 2. |
| 02:53.30 | Qwell | I'll take fast and cheap |
| 02:53.30 | [TK]D-Fender | coppice, on a board in the bar I frequent :) |
| 02:53.37 | Qwell | oh...we're still talking about consultants |
| 02:53.38 | Qwell | NEVERMIND |
| 02:54.13 | [TK]D-Fender | Qwell, "I've never fucked a '10', but in one night I fucked 5 '2s'" - George Carlin |
| 02:54.29 | Qwell | :p |
| 02:54.41 | scooby2 | lol |
| 02:54.50 | coppice | on time and on spec remove the opportunity for follow on work. on budgets means you didn't squeeze the suckers in the first phase. this is totally against the consultants ethical code |
| 02:57.33 | [TK]D-Fender | coppice, Yes, that would be a sample definition for oxymoron :) |
| 02:59.43 | [TK]D-Fender | scooby2, So the IVR was fine on the assumption you had that variable set under [general]. Now what I WOULD discredit him on is letting callers sit in your ivr doing nothing and locking up your lines. If someone wanted to hurt your business and had more lines than you they could call and jsut leave themselves off-hook in your IVR |
| 03:00.44 | scooby2 | adding that to general and restarting makes it sit there but if I hit 5 or 8 or dial an extension it hangs up |
| 03:01.02 | scooby2 | WaitExten does the same |
| 03:01.19 | [TK]D-Fender | scooby2, Oh... and adding to that he's trying to run an IVR in a macro. And it shouldn't hang up. |
| 03:01.29 | coppice | the interesting thing about consultants is the more the adhere to the consultants creed, and suck the blood of the living, the more people pay them. high complaint levels go hand in hand with high payments |
| 03:01.31 | [TK]D-Fender | scooby2, Yeah.. that IS sloppy work... |
| 03:02.15 | [TK]D-Fender | coppice, I must be a bad consultant then :) |
| 03:02.51 | coppice | you're here answering questions for free. how mush worse could you get? |
| 03:03.15 | [TK]D-Fender | coppice, I'm a great drug dealer though... first advise is free :p |
| 03:03.55 | coppice | that trick doesn't work with advice, unless its really bad advice requiring endless remedial work |
| 03:04.10 | [TK]D-Fender | coppice, or I find customers who always want more :) |
| 03:04.30 | [TK]D-Fender | coppice, thing is I don't live off of my consulting, its just helps pay for some of the amenities. |
| 03:05.02 | [TK]D-Fender | coppice, Take a contract, do a good job, pick up a few bucks, get referrals, and when the timings right, maybe callbacks. |
| 03:07.48 | *** join/#asterisk UserReg_CL (n=COB@pc-90-122-214-201.cm.vtr.net) |
| 03:07.54 | UserReg_CL | Mmmm... |
| 03:08.18 | blitzrage | hey, what's wrong with consultants!?! :) |
| 03:08.25 | Qwell | blitzrage: most suck |
| 03:08.34 | Qwell | not the canadian ones though |
| 03:08.36 | blitzrage | that's probably true |
| 03:10.52 | JayTee52 | occasionally our company contracts for consultants. We love getting $150hr idiots who come in and aren't able to fix a difficult problem. Then we hunker down, brainstorm and end up finding a solution and we get to pat ourselves on the back and think, "I'm smarter than a $150hr idiot" although the pay differential takes some of the joy out of that. |
| 03:12.14 | blitzrage | hrmmm... $150/hr? I need to double my pay! :) |
| 03:12.50 | coppice | $150 won't get you a really useless consultant. those are more like $250. |
| 03:12.57 | [TK]D-Fender | blitzrage, the days of your being a cheap whore are over! High-class floozey here you come! |
| 03:12.59 | [TK]D-Fender | :p |
| 03:13.03 | Qwell | coppice: SO true |
| 03:13.11 | blitzrage | [TK]D-Fender: hells ya! |
| 03:13.17 | Qwell | we had these consultants at my last job... it was ridiculous |
| 03:13.20 | blitzrage | get me on the cheap until I can change my website! |
| 03:15.00 | [TK]D-Fender | 8/60 |
| 03:15.28 | blitzrage | Qwell: you owe me four cents |
| 03:15.40 | Qwell | blitzrage: k, I'll even convert to CDN |
| 03:15.41 | [TK]D-Fender | 0.33$ |
| 03:15.48 | blitzrage | wrong |
| 03:16.03 | blitzrage | oops |
| 03:16.10 | blitzrage | (sorry... I thought you were exchanging) |
| 03:16.11 | [TK]D-Fender | (8/60/60)*150 |
| 03:16.17 | blitzrage | ya... I forgot to *8 :) |
| 03:16.21 | blitzrage | lol |
| 03:16.23 | [TK]D-Fender | blitzrage, And us/cad = par now |
| 03:16.36 | blitzrage | [TK]D-Fender: well... close -- bank still takes 2 cents :) |
| 03:17.31 | scooby2 | dollar came back a bit |
| 03:17.42 | scooby2 | 1 CAD = 0.999950 USD |
| 03:18.00 | [TK]D-Fender | scooby2, wrong way! |
| 03:18.31 | scooby2 | 99.995 cents buys you 1 canadian dollar |
| 03:19.17 | [TK]D-Fender | scooby2, I'm *Canadian* :) |
| 03:19.20 | blitzrage | 1.00005 |
| 03:19.20 | blitzrage | <PROTECTED> |
| 03:19.20 | blitzrage | <PROTECTED> |
| 03:19.24 | blitzrage | gah! |
| 03:19.35 | blitzrage | my laptop has gone insane |
| 03:19.38 | scooby2 | [TK]D-Fender: lucky bastard |
| 03:20.08 | [TK]D-Fender | scooby2, I have my prayers out for the US'08 Elections..... |
| 03:20.49 | scooby2 | not I |
| 03:20.57 | [TK]D-Fender | scooby2, Oh? |
| 03:20.57 | scooby2 | every politician is corrupt |
| 03:21.17 | [TK]D-Fender | scooby2, perhaps, but some seems substantially less so than others. |
| 03:21.19 | coppice | no. they are just members of the consultants union |
| 03:22.30 | coppice | corruption in politics is a given. what you need to look for are the ones likely to provide some collateral benefit instead of damage |
| 03:23.06 | scooby2 | I've been behind RP but thats really just a dream |
| 03:25.06 | scooby2 | this country needs a third party badly |
| 03:25.23 | [TK]D-Fender | scooby2, I wouldn't be so sure. the number are mounting staggeringly fast and Romney & Giuliani are having dirt pour out their orifices. |
| 03:25.47 | [TK]D-Fender | scooby2, RP *was* the 3rd party, but is more like the only true part left of the original GOP |
| 03:25.58 | scooby2 | true |
| 03:26.13 | [TK]D-Fender | scooby2, RP, DK, or MG <- Any of these guys get in and I'm a happy camper |
| 03:26.39 | [TK]D-Fender | scooby2, Barack = newb. Hi(tler)llary = *shudder* |
| 03:26.48 | scooby2 | bingo |
| 03:27.02 | coppice | "Oh, please let me be your assistant, Mr Senator. I want to see how government works so badly." |
| 03:27.03 | [TK]D-Fender | scooby2, Any other republican = Bush 2.0 |
| 03:27.04 | coppice | "Well, you've come to the right place. You'll never see it more worse than here." |
| 03:28.02 | scooby2 | i dont blame the whole economy on bush. Americans have always lived way past their means. Now its coming back to get them. |
| 03:28.10 | coppice | [TK]D-Fender: why do you say that? sure many republicans are outraged by the current extremism |
| 03:28.40 | [TK]D-Fender | Dennis Kucinich & Ron Paul Do have a chance in my eyes. Mike Gravel IS a pipe-dream, but a very nice one. |
| 03:29.14 | [TK]D-Fender | coppice, Heard the rest of them in any of the debates? So far all pro-war big-government neocons. CONSERVATIVE my ass... |
| 03:29.31 | [TK]D-Fender | coppice, Dems are largely sellouts as well. |
| 03:29.49 | SwK | repubs and dems are both ass hats |
| 03:29.53 | JayTee52 | Mike Gravel is a bit of a nut case. One video he did he just stared into the camera without saying anything for 2 minutes, then he turned and walked to a pond in the background and threw a rock in, stared at it for a little bit and walked away. |
| 03:30.01 | [TK]D-Fender | much like the rise of OSS, my favour goes to the mavericks appealing to the greater sensibility |
| 03:30.47 | [TK]D-Fender | JayTee52, I agree, but not one that will screw the country. A lot of his policies are quite sound. VERY difficult to implement however. |
| 03:31.00 | coppice | well, this is one of the great historic tricks about starting wars. its very hard to loose an election after starting a war. even though in incumbant can't stand again this time, it colour what every cabdidate says |
| 03:31.33 | [TK]D-Fender | of my favourites, Dennis Kucinich might have the better chance once the other 2 Dems are out of the way |
| 03:32.03 | JayTee52 | I think if a candidate ran on a platform of "Jesus has told me to run for this office. He's told me to raise taxes on the poor and cut taxes to the rich, bomb other countries at random and stamp out freedom wherever it raises it's ugly head." |
| 03:32.18 | JayTee52 | That candidate would win by a landslide |
| 03:32.27 | [TK]D-Fender | coppice, GWB calls martial law following the next fake terrorist attack people predict he'll try to pull to become a full fledged dictator. |
| 03:32.35 | [TK]D-Fender | Oh.. I'm sorry... 'Decider Guy" <---- |
| 03:33.22 | JayTee52 | well, Halliburton has finished most of the internment camps so they're ready for their fake terrorist attack anytime now. |
| 03:34.28 | [TK]D-Fender | JayTee52, And recent legislation, DHS playing around with FIREFIGHTERS (like.. WTF?!), and so on... its all coming into place. |
| 03:34.37 | scooby2 | then you have the people that say bush and his cronies were behind 9/11 |
| 03:34.49 | JayTee52 | I don't believe that |
| 03:35.01 | JayTee52 | I think they just dropped the ball and ignored the warning signs |
| 03:35.18 | JayTee52 | but then they took advantage of the situation |
| 03:35.35 | coppice | governments do that a lot when it benefits their cause |
| 03:35.40 | JayTee52 | yeah |
| 03:36.07 | JayTee52 | and sadly the blind semi-literate couch potato sheep gladly trade their freedom for a false sense of security. |
| 03:36.20 | [TK]D-Fender | ^^^ |
| 03:36.46 | [TK]D-Fender | yup. welcome to the majority of America. |
| 03:36.55 | JayTee52 | as the kid in Live Free or Die Hard said, "During Katrina, it took the government 5 days to get water to the Superdome." |
| 03:38.21 | [TK]D-Fender | JayTee52, Oh and Katrina would qualify as the "other disaster" that could let Bush take control in to ensure "fair government" continues in a "crisis" |
| 03:39.42 | JayTee52 | all I know is that Brownie is "doin a heckuva job!" |
| 03:41.16 | [TK]D-Fender | JayTee52, "Brownie"? |
| 03:41.44 | JayTee52 | Brown was in charge of FEMA |
| 03:42.34 | [TK]D-Fender | JayTee52, Ah, not so familiar with them : http://www.globalresearch.ca/index.php?context=viewArticle&code=ROL20060820&articleId=3010 |
| 03:42.40 | [TK]D-Fender | JayTee52, but that is quite telling :) |
| 03:43.16 | [TK]D-Fender | "This is why FEMA could not respond immediately to the Hurricane Katrina disaster ~ humanitarian efforts were no longer part of its job description under the Department of Homeland Security." |
| 03:44.18 | [TK]D-Fender | I somehow suspect that DHS is up really hing on RP's chopping block |
| 03:44.51 | JayTee52 | RP's? |
| 03:45.03 | [TK]D-Fender | Ron Paul's |
| 03:45.11 | [TK]D-Fender | high* |
| 03:45.34 | JayTee52 | well, Ron Paul has some very good ideas IMHO |
| 03:45.39 | [TK]D-Fender | DHS = super neocon sink-hole. |
| 03:46.18 | JayTee52 | the DHS has been as effective at securing our nation as the DEA/ATF has in The War On Drugs |
| 03:46.24 | [TK]D-Fender | JayTee52, He does indeed. but I am sortof a "soft liberal" as well.... he is a LITTLE hard-core but entirely respectable. |
| 03:46.45 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 03:48.51 | [TK]D-Fender | JayTee52, As long as any of my top 3 get in I don't care. Barack is probably the least dangerous viable candidate of the rest. |
| 03:49.21 | [TK]D-Fender | JayTee52, Can't think of anyone else personally. |
| 03:49.43 | JayTee52 | Barack would never spend an hour in the Oval. He'd be assassinated before he's sworn in. |
| 03:51.01 | *** join/#asterisk hodolo (n=user@85.108.205.137) |
| 03:51.19 | hodolo | hi |
| 03:51.48 | [TK]D-Fender | JayTee52, thats a very sad thought. After all this time Bush/Cheney have EARNED the thought of that you figure Barack wouldn't make it 10 steps |
| 03:52.36 | JayTee52 | I was disgusted with the Republican smear tactics like the attacks on Max Cleland. The idea of attacking someone's patriotism who had lost 3 limbs in Vietnam serving his country goes beyond the pale but the Democrats have really backpedalled and started shovelling the same crap since they've gained the majority in the Senate and House. |
| 03:53.53 | [TK]D-Fender | JayTee52, Neo-con <- the 3rd bi-partisan party ;) |
| 03:53.59 | JayTee52 | I'm just saying I think there is a very large yet very discreet racist underground in this country and that they'd very quickly put Barack out of the picture. |
| 03:54.31 | JayTee52 | and that the neocons are very closely tied with these underground groups |
| 03:54.55 | [TK]D-Fender | JayTee52, Racisms cowering in fear of Islamo-fascists! </hilarity> |
| 03:55.08 | [TK]D-Fender | racists* |
| 03:55.19 | coppice | if he got even close to being elected, he would go the way of bobby kennedy |
| 03:55.39 | JayTee52 | yeah, I work with a couple of "Born again" baptists who preach their holier than thou crap of "I'm saved!" while bashing jews, blacks and gays in the same breath. |
| 03:55.41 | *** join/#asterisk Schumie (i=SteveWri@cpc1-rdng2-0-0-cust441.winn.cable.ntl.com) |
| 03:56.04 | JayTee52 | oh, yeah, and Catholics too |
| 03:56.08 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 03:56.08 | *** mode/#asterisk [+o russellb] by ChanServ |
| 03:56.35 | [TK]D-Fender | JayTee52, its not the sect, its just their visible members :) |
| 03:56.38 | *** join/#asterisk scottpayne (n=sp@n128-227-34-154.xlate.ufl.edu) |
| 03:56.48 | JayTee52 | I don't think any of these people have ever had a rational thought in their entire lives. |
| 03:56.52 | coppice | There's safety in numbers, when you learn to divide, |
| 03:56.54 | coppice | How can we be in, if there is no outside? |
| 03:57.27 | [TK]D-Fender | JayTee52, http://www.youtube.com/watch?v=4i6CSoct4lE&feature=user |
| 03:59.49 | JayTee52 | [TK]D-Fender, hahahahaaa. that's awesome! |
| 03:59.58 | [TK]D-Fender | JayTee52, its a fave :) |
| 03:59.59 | coppice | the biggest problem for the rest of the world, is that as the US becomes more extremist, everyone else gets dragged along |
| 04:00.24 | JayTee52 | coppice, so true |
| 04:00.47 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 04:01.01 | [TK]D-Fender | coppice, up until the poitn where every other major economy dumps the $USD and tells them to unceremoniously "fuck off". |
| 04:01.09 | JayTee52 | what's sad is that right after 9/11 we had most of the world with us and yet it only took the neocons less than a year to destroy all that. |
| 04:01.17 | [TK]D-Fender | China = next world super-power. |
| 04:01.32 | JayTee52 | better get used to liking Ramen |
| 04:01.45 | [TK]D-Fender | ew |
| 04:01.59 | JayTee52 | we'll all be pulling rickshaws for our Chinese Overlords |
| 04:02.42 | coppice | its more than that. The last Chinese president was pretty liberal. The current one, elected after bush took power, is much harder line, and things are heading backwards in china |
| 04:02.51 | [TK]D-Fender | JayTee52, I wouldnt worry about the military, but it'd put "Team America" in their place and maybe we'd see some diplomacy. |
| 04:03.22 | JayTee52 | hahaha, everytime I think of that movie I get nauseous |
| 04:03.47 | JayTee52 | still, it's not nearly as bad as "Two girls, one cup" |
| 04:04.46 | coppice | that's the second time today somebody mentioned that |
| 04:05.28 | outtolunc | weird huh |
| 04:07.45 | coppice | JayTee52: ramen is Japanese. its true that its a Japanese term for a Chinese style of noodles, but its still Japanese. |
| 04:08.05 | JayTee52 | ok, I'll keep that in mind :-) |
| 04:08.15 | UserReg_CL | mmm |
| 04:08.28 | UserReg_CL | need set ping for call ... please helpme |
| 04:08.40 | UserReg_CL | sorry ping = pin :) |
| 04:09.08 | [TK]D-Fender | JayTee52, Another sign of the impending apocalypse. |
| 04:09.38 | JayTee52 | [TK]D-Fender, hahaha, yeah, I was thinking that exact same thing when I saw the video |
| 04:09.58 | [TK]D-Fender | JayTee52, I've seen 2 reaction vids but never the origin |
| 04:14.05 | coppice | Lunchtime. Time for some ęéŗµ (ramen), as our new Chinese overlords would say. |
| 04:16.51 | *** join/#asterisk |omni| (n=rob@c-67-185-144-206.hsd1.wa.comcast.net) |
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| 04:21.27 | Titanous | I just set up asterisk with a TDM400P and keep gettin dialtones left on my VM. What is happening? |
| 04:24.16 | [TK]D-Fender | Titanous, you need to tell your telco to enable "Call Disconnect Supervision" on your line. |
| 04:25.29 | Titanous | [TK]D-Fender, this is a standard POTS line, no T1, will it still work? Also, what exactly dsoes it do? |
| 04:26.19 | [TK]D-Fender | Titanous, CDS is where the telco either sends a momentary cut to the circuit or reverses polarity to indicate that th remote end has hung up. |
| 04:26.25 | [TK]D-Fender | Titanous, and this IS for POTS |
| 04:26.34 | Titanous | OK, thanks alot |
| 04:26.43 | Titanous | I will contact 'Ma Bell' |
| 04:26.49 | UserReg_CL | Mmmm... |
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| 05:30.15 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
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| 05:34.32 | UserReg_CL | mmmm |
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| 05:48.17 | mosty | is the a dialplan command that i can use to return from a macro? |
| 05:48.21 | mosty | is there, rather |
| 05:48.29 | joelsolanki | hi all |
| 05:49.07 | joelsolanki | i m using asterisk. unfortunately last night my serve rebooted. cmos battery was weak so the time changed. |
| 05:49.13 | joelsolanki | and calls were going thru. |
| 05:49.32 | joelsolanki | now all master.csv has cdrs dated 2004 |
| 05:49.50 | joelsolanki | is there any way to set those cdrs to 2007 |
| 05:50.17 | [TK]D-Fender | mosty, doubt it. "s" has to run out, so make a NoOp the last priority and Goto it |
| 05:50.39 | [TK]D-Fender | joelsolanki, "man sed" <- |
| 05:51.42 | mosty | [TK]D-Fender, hmm ok. i was hoping for something more robust |
| 05:52.20 | joelsolanki | i have not used sed |
| 05:52.21 | mosty | joelsolanki, i highly recommend that you use a database for CDR's, then it would have been very easy for you |
| 05:52.36 | joelsolanki | i use mysql database |
| 05:52.46 | [TK]D-Fender | mosty, AEL probably can do it in something that looks "cleaner" if you care enough. |
| 05:52.55 | Corydon76-dig | mosty: MacroExit |
| 05:53.01 | [TK]D-Fender | joelsolanki, then time to learn basic SQL. |
| 05:53.05 | joelsolanki | i have a script which takes data from raw csv cdr and puts in mysql database |
| 05:53.44 | mosty | Corydon76-dig, nice- thankyou |
| 05:55.18 | joelsolanki | but that php script only puts the cdrs which are new by comparing the date |
| 05:55.41 | joelsolanki | as date has gone to 2004 php script is not putting the data in mysql database |
| 05:55.56 | mosty | [TK]D-Fender, i'm a bit wary of AEL since i've not seen many people using it. is it considered to be supported to the same level as a regular dialplan? |
| 05:56.11 | joelsolanki | and somehow if i let him keep then also it is a problem. coz all old cdrs will also copy to mysql database |
| 05:56.17 | mosty | joelsolanki, you should configure asterisk to put the CDR's in the database directly |
| 05:56.38 | [TK]D-Fender | mosty, well it all gets parsed back and I guess it'd depend on your idea of "support" So few of us use it. |
| 05:57.04 | [TK]D-Fender | mosty, and I wasn't advocating it so much as offering it up as a though for your desire for "elegance" |
| 05:57.33 | mosty | [TK]D-Fender, ok then that's enough for me to not consider using AEL on anything important |
| 05:58.22 | joelsolanki | <[TK]D-Fender>: can i change the date in master.csv cdr with sed ? |
| 05:58.48 | Corydon76-dig | mosty: the issue isn't the stability of the code; it's merely that its usage simply hasn't hit critical mass yet |
| 05:58.48 | mosty | it's not elegence so much as robustness that i want here. i just like to avoid the pitfalls that i found in BASIC when i was a kid |
| 05:59.33 | [TK]D-Fender | joelsolanki, master.csv is a text file. what do you think SED is for? *sigh* |
| 05:59.39 | Corydon76-dig | Nobody will start using it until many people are already using it. ;-) |
| 05:59.51 | joelsolanki | :) |
| 06:00.03 | [TK]D-Fender | mosty, and what you have to ask yourself is.. do I want to be a Ground Zero when it does? ;) |
| 06:00.19 | mosty | [TK]D-Fender, the answer is "no, probably not" heh |
| 06:00.42 | mosty | joelsolanki, maybe you can open the csv file in a spreadsheet and edit it that way |
| 06:00.56 | joelsolanki | hmm |
| 06:00.58 | joelsolanki | got it. |
| 06:01.09 | joelsolanki | good idea mosty |
| 06:01.11 | [TK]D-Fender | SEARCH & REPLACE. |
| 06:01.14 | Corydon76-dig | [TK]D-Fender: do you also squeeze the trigger with your eyes closed? |
| 06:01.30 | mosty | joelsolanki, but you really should setup asterisk to insert CDR's directly into your database, and not use csv files |
| 06:01.40 | [TK]D-Fender | Corydon76-dig, Don't do guns anymore, only swords :) |
| 06:01.41 | joelsolanki | yes i understand |
| 06:11.22 | coppice | I like to see a Ken do attitude |
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| 06:30.17 | *** part/#asterisk techie (n=techie@adsl-76-214-21-156.dsl.lsan03.sbcglobal.net) |
| 06:42.52 | mosty | does the dialplan have array variables? |
| 06:44.04 | mosty | or do i need to emulate them with something like ${<basename>_<index>} ? |
| 06:53.43 | *** join/#asterisk nhuisman_work (n=nhuisman@cpe-76-88-144-85.hawaii.res.rr.com) |
| 06:53.47 | nhuisman_work | hey anyone awake? |
| 06:54.06 | nhuisman_work | I was wondering how asterisk handles redundancy and failover |
| 06:54.14 | Corydon76-dig | mosty: no arrays, but you can deal with fields inside a single variable by using CUT |
| 06:54.33 | nhuisman_work | I have an aging cisco system and i'm looking to dump it |
| 06:54.42 | nhuisman_work | but I need to be sure asterisk is going to be robust |
| 06:55.34 | Corydon76-dig | mosty: FIELDQTY is the equivalent of knowing how many items are in the "array" |
| 06:55.52 | mosty | Corydon76-dig, i was interested in creating a hash table, but i can get the job done another way |
| 06:56.25 | Corydon76-dig | mosty: in 1.6, there's a HASH function |
| 06:57.03 | Corydon76-dig | It's meant to be used with func_odbc |
| 06:57.05 | mosty | i'm only just moving to 1.4 now, 1.6 is not an option for me |
| 06:58.00 | Corydon76-dig | You can also use the func_odbc backport and get HASH |
| 06:58.59 | mosty | i think my alternate solution is simpler, i'll stick with that |
| 06:59.28 | russellb | nhuisman_work: there is no failover support in the sense that calls automatically move to another server if a problem occurs, but you can easily design your setup such that if a server fails, all calls are automatically routed to another box |
| 06:59.49 | russellb | and by the first part, i meant _existing_ calls |
| 07:03.02 | *** join/#asterisk apardo (i=apardo@172.64.220.87.dynamic.jazztel.es) |
| 07:08.23 | *** join/#asterisk kn0x (n=pinochle@75.127.83.151) |
| 07:15.34 | *** join/#asterisk bmg505 (n=leon@196.209.176.47) |
| 07:17.05 | nhuisman_work | russellb: yeah I don't mind if calls get dropped |
| 07:17.13 | nhuisman_work | but I want it to fail over to the second server automatically |
| 07:17.16 | nhuisman_work | for new calls |
| 07:17.23 | russellb | yep, that's a common setup ... |
| 07:17.50 | nhuisman_work | I have a T1 PRI so i guess i'd have to physically move the cable |
| 07:17.56 | russellb | that's correct |
| 07:18.16 | nhuisman_work | It seems like asterisk business edition is about $1k a lic from digium |
| 07:18.27 | nhuisman_work | then maybe another 3k a year for 5x9 support |
| 07:19.11 | nhuisman_work | I already have two hp dl 320 g5 servers that I was going to use for call manager so I could put asterisk on them |
| 07:19.24 | russellb | yes, you should :) |
| 07:19.35 | russellb | asterisk is awesome ;) |
| 07:19.42 | nhuisman_work | the cisco solution is the suck expensive |
| 07:20.03 | nhuisman_work | can't believe how much they charge for phone device licenses |
| 07:20.33 | nhuisman_work | something like 7 k for my 60 phones |
| 07:20.47 | russellb | per year |
| 07:20.51 | nhuisman_work | no |
| 07:20.53 | nhuisman_work | just one time |
| 07:21.11 | russellb | ah, i'm thinking of some $10/year/phone fee ... |
| 07:21.26 | russellb | and how people haven't paid it, and then they come in here trying to find firmware :) |
| 07:21.28 | nhuisman_work | i have to ask cisco if they can fix our upgrade problem first |
| 07:21.42 | nhuisman_work | we have call manager 3.1 |
| 07:21.42 | russellb | nooooo, just switch to asterisk. |
| 07:21.51 | RypPn | ccm 4 blows as well |
| 07:21.51 | nhuisman_work | well if I can get the upgrade for free |
| 07:22.00 | nhuisman_work | yeah i'd goto 6 |
| 07:22.02 | nhuisman_work | on linux |
| 07:22.19 | russellb | asterisk is still better :-p |
| 07:22.22 | RypPn | fed up re-running the bat tool when the uploads fail |
| 07:22.49 | nhuisman_work | I think I might have a hard time selling my boss on asterisk |
| 07:23.14 | nhuisman_work | she is a big cisco person and wants something very robust. Money is kind of not an issue up to perhaps 25k. |
| 07:23.25 | nhuisman_work | but I'd love to only spend 5k on asterisk |
| 07:23.32 | nhuisman_work | for the software + t1 cards |
| 07:23.55 | russellb | nhuisman_work: it's not like the required level of support isn't there for asterisk |
| 07:24.12 | russellb | and the amount of freedom and room for customization just blows ccm away |
| 07:24.21 | nhuisman_work | yeah I saw several companies who provided support |
| 07:24.38 | nhuisman_work | for me that isn't that much of an issue, our install is pretty standard |
| 07:25.01 | russellb | yep, lots of support options out there |
| 07:25.09 | russellb | i recommend digium, of course :-p (i work there) |
| 07:25.11 | nhuisman_work | i'd probably just go with digium |
| 07:25.20 | nhuisman_work | i think 5x9 would be enough |
| 07:25.28 | russellb | yeah |
| 07:25.31 | nhuisman_work | since 7x24 is quite a bit more expensive |
| 07:26.19 | nhuisman_work | is it possible to autofailover the voicemail too? |
| 07:26.26 | nhuisman_work | ie, keep them both in sync |
| 07:26.42 | russellb | yeah, you can have multiple servers share the same voicemail storage system |
| 07:26.50 | nhuisman_work | ah |
| 07:26.55 | russellb | using something like NFS, or by having voicemails stored in a database |
| 07:27.03 | russellb | or even having voicemail stored on an IMAP server |
| 07:27.13 | nhuisman_work | stored on imap eh |
| 07:27.14 | russellb | so people can access their voicemail using an email client |
| 07:27.18 | nhuisman_work | interesting |
| 07:27.26 | nhuisman_work | as what, mp3s? |
| 07:27.47 | russellb | wav files generally, but mp3 would work |
| 07:27.48 | nhuisman_work | we have an imap server so that would be interesting, i'd be a little afraid of what happens if it's for some reason not available when a call comes in |
| 07:28.04 | russellb | voicemail gets lost then, i guess |
| 07:28.13 | russellb | there is another voicemail + email option |
| 07:28.14 | nhuisman_work | doesn't queue up eh |
| 07:28.24 | russellb | having it stored on disk or database, and just having it emailed as an attachment |
| 07:28.32 | nhuisman_work | ah |
| 07:28.59 | russellb | the difference is, the mailbox isn't the storage medium, so deleting it from your email doesn't make the blinky light on your phone go off |
| 07:29.14 | nhuisman_work | that might be better |
| 07:29.22 | nhuisman_work | although we have some stupid low attachment size limit |
| 07:29.31 | russellb | either you have the message automatically deleted, and the blinky lights never work, or people have to go back to voicemail later to delete it so their light goes off |
| 07:29.44 | russellb | silly email. |
| 07:29.52 | nhuisman_work | silly my boss limiting it to 2mb |
| 07:29.55 | nhuisman_work | pretty useless |
| 07:30.06 | nhuisman_work | 10mb seems more like it |
| 07:30.20 | russellb | 2mb should be plenty for voicemail i think |
| 07:30.28 | nhuisman_work | yeah if it's an mp3 |
| 07:30.38 | nhuisman_work | or other compressed audio |
| 07:30.59 | nhuisman_work | so how do the servers fail over? |
| 07:31.05 | nhuisman_work | heartbeat type setup? |
| 07:31.30 | *** join/#asterisk jmls (n=jmls@62.49.235.130) |
| 07:31.57 | jmls | could someone look at this debug output and tell me what I'm doing wrong .. |
| 07:31.59 | jmls | http://pastebin.com/m2763a071 |
| 07:32.07 | russellb | nhuisman_work: yeah, usually |
| 07:32.38 | nhuisman_work | i'm a little scared of heartbeat, seen it fail more then a few times. Could have been just the setup though. |
| 07:33.02 | jmls | all I'm doing is a Dial((Zap/g3/444444,120,T) |
| 07:33.55 | nhuisman_work | i wonder if there are any asterisk support people in hawaii |
| 07:34.33 | nhuisman_work | snickr |
| 07:34.42 | jql | I'll put up my shingle... sometime after 8:22pm Sunday |
| 07:35.32 | *** join/#asterisk h3x (i=Justino@64.192.116.17) |
| 07:35.43 | nhuisman_work | russellb: you don't happen to know if digium has any partners in hawaii do you? |
| 07:36.25 | russellb | nhuisman_work: not sure ... i think there are some distributor/reseller type listings on digium.com somewhere |
| 07:36.48 | nhuisman_work | well there was a form I just filled out requesting a partner |
| 07:36.52 | nhuisman_work | i'll look for a list |
| 07:37.27 | russellb | https://www.digium.com/en/ecosystem/resellers/locate.php |
| 07:37.39 | russellb | yeah, that's a form requesting info on a reseller near you .. |
| 07:37.54 | *** join/#asterisk techie (n=techie@adsl-76-214-21-156.dsl.lsan03.sbcglobal.net) |
| 07:38.01 | russellb | guess there isn't a list |
| 07:40.25 | nhuisman_work | there is a list but none in hawaii |
| 07:40.31 | nhuisman_work | only like 5 in the us |
| 07:41.30 | russellb | hm, hawaii? |
| 07:41.55 | russellb | you know ... you could convince me to come help you if you pay for travel and lodging for myself and fiance :) |
| 07:42.45 | nhuisman_work | laugh |
| 07:42.53 | russellb | always wanted to visit there .. |
| 07:42.59 | nhuisman_work | i've lived here my whole life |
| 07:43.02 | nhuisman_work | i want to leave |
| 07:43.14 | nhuisman_work | where do you live now? |
| 07:43.31 | russellb | Huntsville, Alabama ... where Digium HQ is located |
| 07:43.41 | nhuisman_work | hmm yeah i'd rather goto seattle |
| 07:44.18 | russellb | so ... cold ... |
| 07:44.28 | nhuisman_work | good tech market |
| 07:44.36 | nhuisman_work | a lot warmer then some other states in the winter |
| 07:44.44 | nhuisman_work | but yeah it does it get cold |
| 07:44.46 | nhuisman_work | and wet |
| 07:44.51 | nhuisman_work | but I'm used to the wet from living in hilo |
| 07:45.24 | russellb | i'm telling you ... tell the boss that one of the main developers will come set it up for a free trip to Hawaii :) |
| 07:46.04 | nhuisman_work | yeah somehow I doubt there is a way to do that from a university |
| 07:46.10 | russellb | darn! |
| 07:46.14 | nhuisman_work | just no mechanism for such a money expenditure |
| 07:46.19 | russellb | which university? |
| 07:46.33 | nhuisman_work | university of hawaii is who we are connected to |
| 07:46.40 | nhuisman_work | we are actually the institute for astronomy |
| 07:46.40 | russellb | gotcha |
| 07:46.46 | russellb | ooooh :) |
| 07:46.54 | russellb | what size of a system are we talking about? |
| 07:46.57 | nhuisman_work | pretty small |
| 07:47.05 | nhuisman_work | it's a satellite office |
| 07:47.11 | nhuisman_work | bout 60 phones |
| 07:48.19 | russellb | something else you might be interested in is http://www.switchvox.com |
| 07:48.32 | nhuisman_work | that an applianced? |
| 07:48.35 | nhuisman_work | appliance |
| 07:48.36 | russellb | it's a pbx based on asterisk, with an awesome web interface |
| 07:48.38 | russellb | yeah, it is |
| 07:48.52 | nhuisman_work | the digium appliance is just under our size |
| 07:49.00 | russellb | ah, this is different than that |
| 07:49.57 | russellb | http://www.switchvox.com/sv?cmd=selection_wizard |
| 07:50.54 | nhuisman_work | hmm |
| 07:50.58 | nhuisman_work | bout 6k per server |
| 07:51.14 | nhuisman_work | with redundant psu, server class mobo, rackmount and enough cpu for our needs |
| 07:51.34 | nhuisman_work | i do already have the hardware though |
| 07:51.42 | nhuisman_work | i have two dual core xeon boxes |
| 07:51.51 | russellb | gotcha |
| 07:52.11 | nhuisman_work | i could reuse them |
| 07:52.12 | nhuisman_work | but shrug |
| 07:52.12 | russellb | just throwing out options :) |
| 07:52.17 | nhuisman_work | thanks :) |
| 07:53.05 | *** part/#asterisk techie (n=techie@adsl-76-214-21-156.dsl.lsan03.sbcglobal.net) |
| 07:53.16 | blitzrage | russellb: go to bed! :) |
| 07:53.26 | russellb | blitzrage: ha, yes, i should consider that. |
| 07:53.39 | russellb | blitzrage: i've been in the zone |
| 07:53.46 | blitzrage | good to hear! |
| 07:53.51 | russellb | learning about something i want to integrate with asterisk ;) |
| 07:54.17 | blitzrage | I've been working on building TFoT into HTML from DocBook and some other network infrastructure stuff like my website... |
| 07:54.26 | russellb | neato |
| 07:54.31 | blitzrage | (i.e. I've been puttering... but all stuff that needed to be done) |
| 07:55.14 | blitzrage | but I think I'm gonna go read some more Guns, Germs, and Steel, then call it a night and actually try to sleep in tomorrow |
| 07:55.21 | russellb | heh, cool |
| 07:55.25 | blitzrage | night mang |
| 07:55.26 | russellb | i think i'm headed out, too |
| 07:55.31 | russellb | my brain is done |
| 07:55.37 | russellb | g'night |
| 07:55.39 | russellb | nhuisman_work: good luck to you ... |
| 07:55.45 | nhuisman_work | nite |
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| 08:00.02 | jmls | guys, I've really got a problem here, and need some help. |
| 08:00.25 | jmls | I'm not a * noob, but maybe I'm doing something really wrong and just can't see it |
| 08:00.46 | jmls | I can receive inbound calls on my zap isdn, but not make them |
| 08:00.58 | jmls | # |
| 08:01.06 | jmls | <PROTECTED> |
| 08:01.06 | jmls | <PROTECTED> |
| 08:01.06 | jmls | <PROTECTED> |
| 08:01.45 | jmls | It starts ok, but then hangs up with a CHANUNAVAIL |
| 08:13.16 | jmls | ffsl. you struggle for ages trying to find the problem. You post a help message to IRC. you then spot your error |
| 08:13.42 | jmls | had a problem with the zap config, and had configured the wrong group. |
| 08:13.52 | jmls | sorry for the waste of bandwidth |
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| 10:20.38 | duki | hello everyone, |
| 10:21.49 | duki | I would like to be sure if a real time kernel (patch) bring some performance on asterisk server (and or) agents/clients ? |
| 10:21.53 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.144.121) |
| 10:24.56 | duki | Or it is only advantageous for music real time? |
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| 11:01.10 | *** join/#asterisk ntazi (n=user@41.251.73.4) |
| 11:01.16 | ntazi | hi all |
| 11:01.38 | ntazi | can someone help me please i have i issue |
| 11:02.00 | BBHoss | ! ask |
| 11:02.03 | BBHoss | !ask |
| 11:02.10 | BBHoss | sup? |
| 11:02.32 | ntazi | i have a long time wait before dial to a e1 trunk |
| 11:02.52 | ntazi | i am using a digium 2 e1 card |
| 11:02.58 | *** join/#asterisk davevg-btwtech (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net) |
| 11:03.03 | ntazi | !ask |
| 11:03.15 | BBHoss | !ask |
| 11:03.18 | BBHoss | ~ask |
| 11:03.19 | 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. |
| 11:03.49 | BBHoss | ntazi, i would suggest calling support for issues, you've paid for it |
| 11:04.46 | ntazi | not very usefull i am in morroco |
| 11:05.15 | ntazi | i cannot paid the support from my country |
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| 11:07.47 | BBHoss | i would try but i have no experience with t1/e1/j1 cards |
| 11:07.57 | obnauticus | BBHoss, hey |
| 11:08.03 | BBHoss | sup |
| 11:08.06 | obnauticus | I just came up with some interesting idea while... trying to get to sleeps |
| 11:08.07 | obnauticus | :| |
| 11:08.20 | BBHoss | heh |
| 11:08.21 | obnauticus | you know uhh bluetooth evesdropping eh? |
| 11:08.31 | BBHoss | yeah |
| 11:08.39 | obnauticus | SAy you have a laptop with it |
| 11:08.40 | obnauticus | mobilly |
| 11:08.41 | obnauticus | and EVDO |
| 11:08.45 | ntazi | i have no probleme with iax trunk or sip trunk |
| 11:08.57 | obnauticus | you have a listening socket on your asterisk server... |
| 11:08.58 | ntazi | only with the e1 trunk ! |
| 11:09.03 | obnauticus | and use chan_cell with it. |
| 11:09.14 | obnauticus | and get a softphone on your laptop. |
| 11:09.20 | obnauticus | use their cell phone to make calls :| |
| 11:09.28 | BBHoss | yeah |
| 11:09.31 | BBHoss | its not fun |
| 11:09.32 | obnauticus | I dunno how that would work. |
| 11:09.39 | obnauticus | WOULD that work though? |
| 11:09.44 | obnauticus | I was just sitting in bed and was like hey! |
| 11:10.09 | BBHoss | if you're talking about running IP over EVDO then yes, it will WORK, but cell networks are hell for latency-sensitive data |
| 11:10.23 | obnauticus | well |
| 11:10.24 | BBHoss | go to a verizon store and ping something from there laptop :) |
| 11:10.35 | obnauticus | ya good point |
| 11:10.40 | obnauticus | but i mean you could use any net connection |
| 11:10.47 | obnauticus | EVDO is just for extreme-mobility and anywhere/ |
| 11:10.47 | BBHoss | yeah |
| 11:10.54 | obnauticus | but isn't that an interesting idea? |
| 11:10.55 | BBHoss | i had a phone that had wifi/evdo |
| 11:11.05 | BBHoss | and there are sip clients for them |
| 11:11.25 | BBHoss | the quality was horrible, but thats probably because it was running g.711u |
| 11:11.31 | obnauticus | :| |
| 11:11.32 | obnauticus | ew. |
| 11:11.41 | obnauticus | God damn |
| 11:11.46 | BBHoss | the xv6700 i think |
| 11:11.47 | obnauticus | fucking voipjet still hasn't given me my MD5 hash |
| 11:12.00 | obnauticus | it's been 2 weeks since i've sent int a query to their `fast support' which is `always avaliable' |
| 11:12.05 | BBHoss | heh |
| 11:12.17 | BBHoss | thats too bad |
| 11:12.45 | obnauticus | im thinking of something to code for what i said before :| |
| 11:13.12 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 11:13.35 | obnauticus | You don't even need it to interface with a remote * server, you can just get a bluetooth dongole, attach an antenna to it run * locally and chan_cell. |
| 11:13.43 | obnauticus | And use other people's cells to issue calls |
| 11:13.44 | obnauticus | yes |
| 11:13.45 | obnauticus | lol |
| 11:14.27 | BBHoss | yeah, but to place calls requires more than the stuff to eavesdrop i think |
| 11:14.35 | obnauticus | well chan_cell just uses uhh |
| 11:14.37 | obnauticus | bluetooth |
| 11:14.50 | obnauticus | you could use common exploits to gain access and spoof stuff |
| 11:14.58 | obnauticus | I still don't do much with bluetooth |
| 11:14.58 | BBHoss | never used it |
| 11:15.32 | BBHoss | yeah verizon locks down pretty much every useful bluetooth profile, so its basically useless on most phones |
| 11:15.46 | obnauticus | Even dialing via bluetooth? |
| 11:16.02 | BBHoss | i hate verizon, i think im gonna start my own cell service company :) |
| 11:16.13 | obnauticus | How do you plan on doing that? |
| 11:16.29 | BBHoss | heh, just a thought, you know... |
| 11:16.56 | BBHoss | focus on having GOOD customer service, over 24/7/365.25 etc |
| 11:17.11 | obnauticus | It's hard to do that |
| 11:17.24 | obnauticus | you need a F*ck load of money to get decent coverage |
| 11:17.36 | obnauticus | and then you need plans with public roaming providers |
| 11:17.44 | BBHoss | no i mean not have 24/7 |
| 11:18.08 | BBHoss | basically be the geeks phone service |
| 11:18.42 | BBHoss | not well thought out, i was just pissed off the other day and started thinking :) |
| 11:18.51 | obnauticus | Well if google gets the C block I imagene there will be a whole bunch of people trying to use google's servioce to cell their VoIP services. |
| 11:19.06 | BBHoss | yeah |
| 11:19.32 | BBHoss | or if UWB got approved, i could launch a few blimps and service all of the US |
| 11:20.25 | ntazi | exit |
| 11:20.27 | ntazi | quit |
| 11:20.34 | obnauticus | Fail? |
| 11:20.44 | obnauticus | UWB? |
| 11:20.45 | obnauticus | lol |
| 11:20.51 | BBHoss | http://www.processor.com/editorial/article.asp?article=articles%2Fp2809%2F09p09%2F09p09.asp |
| 11:20.57 | BBHoss | ultra-wideband |
| 11:21.09 | obnauticus | Why wasn't it approved/ |
| 11:21.16 | BBHoss | another product developed locally :) |
| 11:21.19 | BBHoss | well |
| 11:21.19 | obnauticus | and I hope comcast's DOCSIS 3 comes sooner here. |
| 11:21.24 | obnauticus | 160/120mbps |
| 11:21.27 | obnauticus | better F*Cking hurry. |
| 11:21.28 | BBHoss | much speculation insues |
| 11:21.43 | BBHoss | the FCC is a red-tape nightmare of course |
| 11:21.50 | obnauticus | ya |
| 11:22.03 | BBHoss | and then it is MUCH harder to eavesdrop on than standard radio emissions |
| 11:22.13 | BBHoss | so the CIA/NSA isnt happy about that |
| 11:22.36 | BBHoss | 35 milliwatts has an 18 MILE range |
| 11:22.50 | obnauticus | Ew, |
| 11:22.58 | BBHoss | you can also see through walls and shit with it |
| 11:23.02 | BBHoss | its very nice |
| 11:23.06 | obnauticus | ya well |
| 11:23.09 | BBHoss | timedomain.com |
| 11:23.10 | obnauticus | if you think of err |
| 11:23.17 | obnauticus | You said blimps? |
| 11:23.21 | BBHoss | yeah |
| 11:23.28 | obnauticus | That's like 80mi coverage |
| 11:23.29 | obnauticus | lol |
| 11:23.35 | obnauticus | That chick on that page |
| 11:23.36 | obnauticus | i'd do her. |
| 11:23.55 | BBHoss | heh |
| 11:24.00 | obnauticus | on timedomain |
| 11:24.05 | BBHoss | yeah |
| 11:24.43 | BBHoss | http://en.wikipedia.org/wiki/Ultra-wideband |
| 11:24.55 | BBHoss | read up on it, its a VERY interesting technology |
| 11:25.43 | BBHoss | the wikipedia page doesnt really explain though |
| 11:25.56 | BBHoss | well, graphically |
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| 11:31.15 | BBHoss | damn openoffice.org is a beast, been compiling for 7 hours now! |
| 11:31.26 | obnauticus | LOL |
| 11:31.44 | obnauticus | ya BBHoss |
| 11:31.49 | BBHoss | hell, all of gnome AND the deps didnt take that long |
| 11:31.59 | obnauticus | i think it wouldn't be too hard to code a module for Asterisk to use bluejacking as a channel. |
| 11:32.34 | BBHoss | im not really familiar with bluejacking, so i dunno |
| 11:32.49 | obnauticus | Neither me but I am reading up |
| 11:33.00 | obnauticus | I'm gonna make some docs and modules and etc, i need a project tomorrow lol |
| 11:35.06 | BBHoss | you outa school yet? |
| 11:35.10 | obnauticus | No. |
| 11:35.18 | obnauticus | Well tomorrow is the weekend :| |
| 11:35.23 | BBHoss | yeah |
| 11:35.35 | BBHoss | i have good ol exams coming up this week |
| 11:38.07 | obnauticus | BBHoss think about this lol |
| 11:38.15 | obnauticus | find a mobile phone store |
| 11:38.21 | obnauticus | make an embedded box with EVDO and solar power |
| 11:38.36 | obnauticus | BT jack all the phones inside have it connect to your asterisk server lol. |
| 11:38.41 | obnauticus | use any phone in the store as a channel. |
| 11:38.41 | BBHoss | yeah if you could get in range :) |
| 11:38.45 | obnauticus | it's possible |
| 11:38.49 | BBHoss | but of course its illegal... |
| 11:38.50 | obnauticus | you can extend it apperentally :| |
| 11:39.01 | obnauticus | ya |
| 11:39.11 | obnauticus | just some ideas i have |
| 11:39.12 | obnauticus | for some reason |
| 11:40.49 | obnauticus | http://arstechnica.com/news.ars/post/20071130-really-truly-official-google-to-bid-on-700mhz-spectrum.html |
| 11:40.52 | obnauticus | k im going to bed nao |
| 11:40.54 | obnauticus | cya :{ |
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| 12:25.27 | DarKnesS_WolF | anyone using a2billing ? i'm having aproblem i'm billing with every sec....i want the 1 sec become a charge of min. |
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| 13:24.24 | *** join/#asterisk FCOJ (n=mordur@157-157-239-243.dsl.dynamic.simnet.is) |
| 13:25.41 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 13:26.15 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 13:33.35 | *** join/#asterisk syneus (n=syneus@host5-35-dynamic.4-87-r.retail.telecomitalia.it) |
| 13:42.34 | *** join/#asterisk Buhntz (i=Boones@port-212-202-41-252.dynamic.qsc.de) |
| 13:46.35 | adamowitz | how do i implement sms receive capability on my * box? |
| 13:50.10 | *** join/#asterisk coldsteal (n=coldstea@unaffiliated/coldsteal) |
| 13:50.12 | coldsteal | hello |
| 13:50.19 | DarKnesS_WolF | hello coldsteal |
| 13:50.27 | DarKnesS_WolF | adamowitz: there is no chan_sms or so ? |
| 13:51.25 | coldsteal | im trying to call an international # and every time i do i get a busy signal |
| 13:51.28 | adamowitz | well, i've read alot already, and encountered something like mobile_chan, but I get the impression that it can be done without that, using the PSTN. But I think it requires a gateway of some sort. |
| 13:51.56 | coldsteal | but its not even trying to call it as far as i can tell from the asterisk cli |
| 13:54.02 | coldsteal | im not sure my extenstion correct |
| 13:55.33 | coldsteal | http://rafb.net/p/LGYgBP82.html |
| 13:57.19 | DarKnesS_WolF | coldsteal: try it like this |
| 13:57.48 | DarKnesS_WolF | exten => _011,1,Dial(SIP/axvoice/${EXTEN}) |
| 13:57.51 | DarKnesS_WolF | only |
| 13:57.52 | adamowitz | anyone here use wengophone or openwengo? |
| 13:57.57 | DarKnesS_WolF | no need for the rest for now |
| 13:58.09 | coldsteal | okay |
| 13:59.29 | *** join/#asterisk leikh (n=chatzill@port162.ds1-amb.adsl.cybercity.dk) |
| 14:00.03 | *** join/#asterisk perf3kt (n=perf3kt@adsl-68-73-146-81.dsl.ipltin.ameritech.net) |
| 14:00.10 | leikh | "exten => 1234,6,Voicemail(1234)" plays the default greeting. How can I play a custom greeting? |
| 14:01.15 | coldsteal | DarKnesS_WolF: i get a busy |
| 14:01.22 | DarKnesS_WolF | coldsteal: do that |
| 14:01.34 | DarKnesS_WolF | sip show registry |
| 14:01.39 | coldsteal | and i get http://rafb.net/p/8jwWW097.html |
| 14:02.15 | coldsteal | http://rafb.net/p/PF6P3C78.html |
| 14:03.10 | DarKnesS_WolF | coldsteal: ok do that |
| 14:03.45 | DarKnesS_WolF | exten => _011,1,Dial(SIP/username:password@sip.axvoice.com/${EXTEN}) |
| 14:07.31 | *** join/#asterisk Cybertoy (n=Cybertoy@swillux.swill.org) |
| 14:07.49 | Cybertoy | hi |
| 14:08.05 | Cybertoy | I upgraded to asterisk 1.4.15 last night... and this morning it crashed twice on me... |
| 14:08.30 | coldsteal | DarKnesS_WolF: same thing |
| 14:09.14 | leikh | am I on to something with this? "exten => 1234,6,Voicemail(1234),Playback(custom-greeting)" |
| 14:09.25 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584058.dsl.bell.ca) |
| 14:11.46 | coldsteal | it seams like it doesnt even got to the * box |
| 14:12.17 | leikh | or perhaps: "exten => 1234,6,Voicemail(Playback(custom-greeting),1234)"? |
| 14:16.38 | DarKnesS_WolF | coldsteal: ahh i got it :-D |
| 14:16.48 | DarKnesS_WolF | exten => _011XX.,1,Dial(SIP/axvoice/${EXTEN}) |
| 14:18.09 | coldsteal | omg it works |
| 14:18.13 | coldsteal | thanks so much |
| 14:18.20 | coldsteal | lol but i cant even call it |
| 14:18.41 | coldsteal | i get a message saying the # im trying to call cant be called from my calling level |
| 14:18.44 | coldsteal | lol |
| 14:18.51 | DarKnesS_WolF | coldsteal: |
| 14:18.52 | DarKnesS_WolF | :P |
| 14:19.06 | coldsteal | okay i have to go now thanks |
| 14:19.53 | DarKnesS_WolF | coldsteal: send me 1000000 USD on my account :P |
| 14:20.30 | Cybertoy | third crash |
| 14:27.01 | *** join/#asterisk sysreq` (n=sysreq@unaffiliated/sysreq) |
| 14:28.59 | *** join/#asterisk RoyK (n=roy@ip-231-27-149-91.dialup.ice.no) |
| 14:29.21 | *** join/#asterisk Greek-Boy (n=Greek-Bo@41.221.58.2) |
| 14:32.01 | leikh | what is the prefix for playing custom greeting with voicemail? |
| 14:34.24 | *** join/#asterisk d-k-t (n=dt@125.120.137.80) |
| 14:37.55 | *** join/#asterisk syneus (n=syneus@host5-35-dynamic.4-87-r.retail.telecomitalia.it) |
| 14:38.56 | DarKnesS_WolF | leikh: u have to save it 1st |
| 14:40.51 | leikh | DarKnesS_WolF: It was a serious question... |
| 14:44.27 | *** join/#asterisk [Mr_X] (n=mrx@78-62-74-62.ip.zebra.lt) |
| 14:50.47 | *** join/#asterisk ming_zym (n=ming_zym@124.14.236.40) |
| 14:52.24 | jer | i'm getting a "handle_request_subscribe: Received SIP subscribe for peer without mailbox: (null)" whenever one of my phones renews its subscription. that phone does have a mailbox=.. that is valid, calls go to voicemail, it can check its voicemail; i'm a little unsure as to why i'm getting that, any ideas? |
| 14:53.42 | DarKnesS_WolF | leikh: and i did say a serious answer ... i do have my own u / b message |
| 14:54.07 | DarKnesS_WolF | and teh system automatic will take the recorded one and use it instade of the orginal * one .. check out the options in the * menu in voicemail |
| 14:54.32 | DarKnesS_WolF | jer: what kind of phones ? |
| 14:54.33 | DarKnesS_WolF | snome ? |
| 14:54.37 | DarKnesS_WolF | sone * |
| 14:54.42 | DarKnesS_WolF | snom* |
| 14:54.50 | jer | DarKnesS_WolF, snom 370 |
| 14:55.14 | DarKnesS_WolF | i think i had this before it is configurations issue for the vmail indication |
| 14:55.37 | leikh | DarKnesS_WolF: I don't have clients. The asterisk is solitary sip-client, accepting incoming calls. They are all directed to voicemail and then emailed. |
| 14:55.40 | jer | ok, any ideas on what fixed it? |
| 14:55.59 | jer | or could you point me at some documentation? googling is being frustraiting =/ |
| 14:56.51 | adamowitz | in general, is sms to landlines possible in the usa? |
| 14:57.52 | jer | adamowitz, i would suspect it is, it's possible in Canada; and well, our two systems are basically identical |
| 14:58.18 | adamowitz | thanks jer: do you do it yourself in Canada? |
| 14:58.47 | jer | adamowitz, yes, and it works |
| 14:59.29 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 14:59.34 | jer | when i send a sms message from my cell phone to my landline which connects into the asterisk server and forwards all calls to a hunt group, it comes in i pick it up, and it speaks to me |
| 15:00.02 | Zeeek | jer is that a problem? |
| 15:00.14 | jer | Zeeek, no it's not, i was responding to adamowitz's query |
| 15:00.23 | Zeeek | just got here, wasn't sure |
| 15:01.18 | jer | no worries =] |
| 15:03.00 | Zeeek | because in this country sms tries to read the msg to you if you send it to box 0 |
| 15:03.45 | Zeeek | which is irritating if that isn't what you wanted ;) |
| 15:04.24 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 15:04.24 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:04.28 | jer | Zeeek, yeah, same here |
| 15:04.35 | Zeeek | uk? |
| 15:04.38 | jer | Canada |
| 15:04.42 | Zeeek | ah. |
| 15:04.53 | jer | i mean, for 99% of uses, it's perfectly fine |
| 15:04.54 | Zeeek | in what labnguage does ti read the msg? |
| 15:04.58 | jer | english |
| 15:05.29 | jer | but i've not tried to send an sms message into one of the trunks into my asterisk server from a non-english carrier |
| 15:08.57 | *** mode/#asterisk [+o bmd] by ChanServ |
| 15:09.49 | Zeeek | it's funny once |
| 15:10.42 | *** join/#asterisk rmayorga_ (n=rmayorga@unaffiliated/rmayorga) |
| 15:12.08 | *** mode/#asterisk [+o bkruse] by ChanServ |
| 15:14.43 | leikh | problem with language/sounds; exten => 666,1,Wait,10,Set(LANGUAGE()=no) - and still asterisk plays the English greeting :-/ |
| 15:15.02 | *** join/#asterisk saftsack (n=saftsack@pD9E04A1D.dip.t-dialin.net) |
| 15:18.06 | *** join/#asterisk ZakMcRofl (n=unknown@91.65.148.228) |
| 15:18.13 | ZakMcRofl | hi all |
| 15:18.50 | ZakMcRofl | i have a huge problem: when I try dialing out from the internal ISDN via SIP a truncated number gets dialed |
| 15:19.10 | ZakMcRofl | i.e. i dial "123456789" and asterisk tries to dial "1234" |
| 15:19.34 | ZakMcRofl | the extension is called _XXXX. => { so i guess it just matches too early |
| 15:20.04 | ZakMcRofl | if i dial with a different phone (that probably submits the number as a whole) it works as intended |
| 15:20.13 | russellb | correct |
| 15:20.48 | ZakMcRofl | i think its called DID, isn't it? if it dials digit by digit |
| 15:20.57 | russellb | not really, no ... |
| 15:21.07 | russellb | it's just how certain channel technologies work |
| 15:21.20 | russellb | someones asterisk gets digit by digit, with others, it gets the whole number at once |
| 15:21.48 | RoyK | ZakMcRofl: using misdn? |
| 15:21.51 | ZakMcRofl | but if i pick up the phone, dial digit by digit, how can i tell asterisk to wait X seconds until no further numbers are dialed |
| 15:21.53 | ZakMcRofl | i use CAPI |
| 15:21.56 | ZakMcRofl | its a fritzbox |
| 15:21.59 | RoyK | k |
| 15:22.05 | ZakMcRofl | shall i pastebin the configS? |
| 15:22.45 | ZakMcRofl | is it something i need to configure in the ISDN/CAPI channel settings or in * |
| 15:23.14 | ZakMcRofl | i recon that feature isn't widely used |
| 15:23.46 | ZakMcRofl | i'm using AEL, could that be a part of the problem? |
| 15:23.58 | russellb | nope |
| 15:24.04 | russellb | you just need to adjust your patterns |
| 15:24.10 | russellb | to be more exact |
| 15:24.30 | *** part/#asterisk leikh (n=chatzill@port162.ds1-amb.adsl.cybercity.dk) |
| 15:24.32 | ZakMcRofl | but the pattern says 4 numbers or more shall be dialed via sipgate |
| 15:24.36 | ZakMcRofl | whats wrong with that |
| 15:24.45 | ZakMcRofl | i mean how can the pattern know nothing more is coming |
| 15:25.02 | russellb | well you're experiencing what is wrong with that |
| 15:25.09 | russellb | :) |
| 15:25.28 | ZakMcRofl | well outbound numbers aren't a fixed length |
| 15:25.41 | ZakMcRofl | so how can a patter ever tell the number is complete |
| 15:26.07 | ZakMcRofl | could i use autofallthrough to make it work? |
| 15:26.11 | russellb | well, usually you have a much more defined set of patterns that are valid ... |
| 15:26.21 | russellb | anyway, try changing the pattern to ... _XXXX! |
| 15:26.37 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:26.37 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 15:27.48 | ZakMcRofl | by the way, is there a way to reload the config without dropping current calls (total noob here) |
| 15:28.20 | blitzrage | 'reload' |
| 15:28.28 | blitzrage | just don't "restart now" |
| 15:28.37 | ZakMcRofl | cheers |
| 15:29.00 | DarKnesS_WolF | blitzrage: i recall something like restart grace ? |
| 15:29.00 | DarKnesS_WolF | or so ? |
| 15:29.12 | DarKnesS_WolF | so it do the restart but after the calls are done ? |
| 15:29.33 | DarKnesS_WolF | restart gracefully <---- should do restart after the calls are done right ? |
| 15:29.37 | blitzrage | DarKnesS_WolF: ya... but that is bad, because it could take a long time for asterisk to be able to restart, and you might not be watching it when it does -- and it could fail |
| 15:30.02 | blitzrage | DarKnesS_WolF: that'll stop taking calls in the meantime, but let the current ones finish their call |
| 15:30.07 | DarKnesS_WolF | blitzrage: mmmm not me :-) i always keep an eye on asterisk CLI |
| 15:30.12 | blitzrage | DarKnesS_WolF: restart when convenient will just wait for 0 calls |
| 15:30.22 | blitzrage | DarKnesS_WolF: even if it took 2 days? |
| 15:30.25 | DarKnesS_WolF | blitzrage: mmmm right |
| 15:30.34 | DarKnesS_WolF | blitzrage: nah in this case i had hangup the lines :P |
| 15:30.54 | blitzrage | whatever works -- I don't like using the 'restart ...' commands |
| 15:31.06 | DarKnesS_WolF | :-) |
| 15:31.30 | ZakMcRofl | trying the _XXXX! now |
| 15:32.01 | *** join/#asterisk tobias (n=tobias@user-0ce2hr3.cable.mindspring.com) |
| 15:32.35 | ZakMcRofl | doesnt help |
| 15:32.43 | ZakMcRofl | i feel something is really fucked up with my config |
| 15:33.03 | blitzrage | what are you trying to do, and pastebin the relevant parts |
| 15:33.10 | russellb | you need better patterns is all |
| 15:33.14 | russellb | it's not "really fucked up" |
| 15:33.19 | blitzrage | lol |
| 15:33.19 | russellb | just "not correct" :) |
| 15:33.31 | ZakMcRofl | again i dont see how a pattern can fix it |
| 15:33.56 | ZakMcRofl | it matches the pattern after 4-5 digits and starts sipping out |
| 15:33.58 | blitzrage | I joined the channel to late to know what he was trying to do :) |
| 15:34.17 | russellb | blitzrage: he has _XXXX. because he wants any number, 4 or more, to go out somewhere |
| 15:34.18 | blitzrage | _XXXX! will match 4 or more |
| 15:34.21 | ZakMcRofl | i have internal ISDN bus and when i call from there, the outbound number gets truncated |
| 15:34.28 | blitzrage | _XXXX. matches 5 or more |
| 15:34.29 | russellb | blitzrage: but he's using ISDN where he's getting digit by digit |
| 15:34.49 | russellb | blitzrage: so it gets matched as soon as it hits 4, or 5, even if there are more digits. |
| 15:35.00 | ZakMcRofl | i want it to wait until the users finished dialing (x seconds no digit dialed) |
| 15:35.12 | blitzrage | Set(TIMEOUT(digit)=2) ? |
| 15:35.20 | blitzrage | or is that only in Background()... |
| 15:35.32 | ZakMcRofl | i think i tried that once |
| 15:35.41 | ZakMcRofl | let me try it again |
| 15:35.42 | blitzrage | ya. I don't think it works in pattern matching |
| 15:35.42 | ZakMcRofl | where to put that? start of extension? |
| 15:35.43 | blitzrage | (I just woke up) |
| 15:36.07 | ZakMcRofl | i saw there's a command called WaitExten() maybe that could help |
| 15:36.13 | russellb | i bet if you list multiple patterns ... _XXXX., _XXXXX., _XXXXXX., ... it will wait |
| 15:36.29 | russellb | or even _XXXX., and _XXXXXXXXXXXXXX |
| 15:36.36 | ZakMcRofl | hehe |
| 15:36.40 | ZakMcRofl | lets try that |
| 15:36.44 | russellb | as it will wait to see if it matches the long one, and eventually time out |
| 15:36.52 | russellb | based on whatever the digit timeout is, i think |
| 15:37.05 | russellb | but i don't know what i'm talking about. |
| 15:37.10 | blitzrage | russellb: that makes sense |
| 15:37.10 | russellb | blitzrage is the expert here. |
| 15:37.23 | blitzrage | russellb: and was what I was about to type up, but you beat me to it :) |
| 15:37.35 | blitzrage | (before I got distracted by a firefox update) |
| 15:37.44 | russellb | heh |
| 15:39.09 | ZakMcRofl | hmm it doesn't work |
| 15:39.17 | ZakMcRofl | still dials the first 4 numbers only |
| 15:39.23 | ZakMcRofl | maybe if i switch their order |
| 15:39.28 | blitzrage | order doesn't matter |
| 15:39.36 | blitzrage | asterisk orders them when it loads the dialplan |
| 15:39.56 | russellb | ZakMcRofl: what asterisk version? |
| 15:40.00 | ZakMcRofl | 1.4 |
| 15:40.02 | blitzrage | 'dialplan show <context>' will show you the order asterisk will match them |
| 15:40.06 | ZakMcRofl | using AEL, |
| 15:40.24 | russellb | i swear it should wait ... |
| 15:40.33 | russellb | eh .. |
| 15:40.40 | davevg-btwtech | i think Zak's problem is more of a phone config issue rather than an * config issue. my .02 |
| 15:40.48 | blitzrage | davevg-btwtech: agreed |
| 15:40.51 | russellb | on BRI ? |
| 15:41.03 | blitzrage | the phone itself is an ISDN phone? |
| 15:41.08 | russellb | this is incoming calls on a BRI |
| 15:41.15 | blitzrage | oh... |
| 15:41.21 | blitzrage | now I'm all caught up.... heh |
| 15:41.29 | ZakMcRofl | http://pastebin.com/m953608e |
| 15:41.34 | davevg-btwtech | hah me too, i thought he said earlier it worked with a different phone |
| 15:41.35 | DarKnesS_WolF | mm mark is not here anymore ? |
| 15:41.51 | ZakMcRofl | yeah on a different phone it works |
| 15:42.02 | ZakMcRofl | but only if i dial the whole number and THEN pickup |
| 15:42.18 | ZakMcRofl | if i pickup first and dial digit by digit it fails on the other phone too |
| 15:42.19 | blitzrage | still seems like something is forcing the phone to dial 4 digits... |
| 15:42.21 | russellb | DarKnesS_WolF: not much, no |
| 15:42.27 | davevg-btwtech | what brand phones? |
| 15:42.34 | ZakMcRofl | hagenuk |
| 15:42.34 | DarKnesS_WolF | russellb: cuz of time diffrent i dont' see him much :-s |
| 15:42.44 | ZakMcRofl | old wireless ISDN phones |
| 15:42.56 | DarKnesS_WolF | anyone using a2billing ?? i want to start billing from 1st sec as it is a minut. |
| 15:43.58 | ZakMcRofl | i think the issue is only phone releated because the one phone dials the complete number at once (which works) and the other dials it digit by digit |
| 15:44.05 | ZakMcRofl | even if you entered it all before picking up |
| 15:44.26 | ZakMcRofl | too bad i didn't try it before switching to AEL |
| 15:44.34 | ZakMcRofl | and changing my entire config |
| 15:44.53 | blitzrage | it boggles me the number of people who don't have a test system for their phone systems |
| 15:44.58 | russellb | it has nothing to do witih AEL |
| 15:45.12 | blitzrage | AEL just goes back into dialplan logic from the viewpoint of Asterisk |
| 15:45.32 | blitzrage | AEL is really just an abstraction layer from the dialplan |
| 15:45.37 | ZakMcRofl | ok but maybe the compiled result affects looks different to what i had earlier |
| 15:46.04 | blitzrage | you don't have the old config to compare against? (again... 'dialplan show <context>') |
| 15:46.22 | ZakMcRofl | i still have it but it lacks a few of the features i build |
| 15:46.40 | ZakMcRofl | ok hold on |
| 15:46.54 | ZakMcRofl | isn't there a way to tell asterisk to wait for a few seconds |
| 15:47.20 | DarKnesS_WolF | ZakMcRofl: wait(sec) |
| 15:47.43 | ZakMcRofl | but i would have to do that before it enter _XXXX! |
| 15:47.45 | ZakMcRofl | right? |
| 15:47.57 | blitzrage | you can't.... Wait() will happen after the pattern match |
| 15:47.59 | ZakMcRofl | so i need a second context it can wait in |
| 15:48.01 | blitzrage | during one of the priorities |
| 15:48.12 | blitzrage | [start] |
| 15:48.27 | blitzrage | nevermind |
| 15:48.30 | ZakMcRofl | i go context outgoing_isdn {... dial }; |
| 15:48.41 | ZakMcRofl | *got |
| 15:48.50 | davevg-btwtech | i guess as a hack, you can set up that phone to go to its own context, with a _.,1,WaitExten(), then do a dial |
| 15:48.51 | ZakMcRofl | where it goes directly from capi |
| 15:49.13 | davevg-btwtech | err or a read, so you can concat it |
| 15:49.20 | davevg-btwtech | with what it matched already |
| 15:49.37 | ZakMcRofl | WaitExten() will wait for more digits and then relaunch the context? |
| 15:50.03 | davevg-btwtech | do a read instead, so you have the ${EXTEN} variable, and the variable that was read, concat them together and dial |
| 15:50.23 | davevg-btwtech | ugly hack, but may work |
| 15:50.40 | ZakMcRofl | i think i saw that in one of the WaitExten descirptions. |
| 15:50.53 | ZakMcRofl | http://www.voip-info.org/wiki-Asterisk+cmd+WaitExten |
| 15:51.20 | davevg-btwtech | but, waitexten, you will lose the initial digits on the extension that was matched which called the waitexten |
| 15:51.27 | *** join/#asterisk docelmo (n=docelmo@c-76-99-153-242.hsd1.de.comcast.net) |
| 15:52.23 | ZakMcRofl | In order to obtain the behavior described above, make sure that autofallthrough is not set. |
| 15:52.47 | ZakMcRofl | i read autofallthrough a couple of times too |
| 15:53.05 | ZakMcRofl | its when * keeps trying to dial until the number exists, isn't it? |
| 15:53.14 | docelmo | say anyone in here know of an easy way to pull out the sip response code from a channel when its disposition is congestion or chanunavail? |
| 15:53.40 | *** join/#asterisk tzafrir_laptop (n=tzafrir@89.130.6.136) |
| 15:57.35 | *** join/#asterisk BBishop (n=bbishop@unaffiliated/blackbishop) |
| 16:01.33 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 16:04.19 | Cybertoy | hmm.. I keep having crashes with asterisk-1.4.15 ... anyone else? |
| 16:04.33 | Cybertoy | and worse I can't see anything in the log that would show why... |
| 16:05.47 | *** join/#asterisk Odie_flocon (n=odie@S0106000f3d5eae81.cg.shawcable.net) |
| 16:06.24 | docelmo | Cybertoy if you are crashing look at your core dumps in /tmp |
| 16:07.13 | Cybertoy | nothing there... |
| 16:07.21 | Cybertoy | but perhaps I have to enable core dumps somewhere? |
| 16:09.14 | ZakMcRofl | i don't get AEL. whats wrong with this line: goto(rauswaehlen|${Predigits}${EXTEN}|start); |
| 16:10.42 | ZakMcRofl | ahh... no brackets |
| 16:10.59 | Cybertoy | yeah ... was just about to type that... :) |
| 16:11.13 | *** part/#asterisk thugwh0re (i=plutten@pimping.at.manwh0re.se) |
| 16:11.57 | *** join/#asterisk harryv (n=harry@0x55508034.adsl.cybercity.dk) |
| 16:12.49 | harryv | hello. in my incoming context i only have one extension: exten => _X.,1,SIP/myphone |
| 16:14.32 | harryv | how do i tell the register-statement in sip.conf to use that one? it delts to 's' and therefore rejects incoming calls. |
| 16:15.08 | russellb | harryv: add /something to the end of the register statement |
| 16:15.10 | Cybertoy | harryv, you should either create a s ext... or register to something that would match that regexp |
| 16:15.28 | Cybertoy | harryv, such as register ........ /100 |
| 16:17.15 | [TK]D-Fender | harryv, and this : exten => _X.,1,SIP/myphone <--- you MIGHT want to have an actual application in there too..... |
| 16:17.21 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 16:17.51 | [TK]D-Fender | harryv, and - register => user:pass@host/extentodialotherwises |
| 16:17.51 | harryv | [TK]D-Fender: i do - agi(http://localhost) |
| 16:18.13 | [TK]D-Fender | harryv, well the exten you showed was not valid. |
| 16:18.23 | russellb | [TK]D-Fender: lol at the exten line ... didn't even notice :) |
| 16:18.37 | harryv | exten => _X.,1,AGI(agi://localhost) |
| 16:18.49 | harryv | that's the actual one. works fine. |
| 16:18.58 | ZakMcRofl | ok waitexten isn't helping either |
| 16:19.03 | [TK]D-Fender | harryv, 51st times the charm! |
| 16:19.44 | *** join/#asterisk Schumie (i=SteveWri@cpc1-rdng2-0-0-cust441.winn.cable.ntl.com) |
| 16:25.13 | ZakMcRofl | any idea what this means: |
| 16:25.13 | ZakMcRofl | [Dec 1 16:24:37] WARNING[1419]: chan_capi.c:4151 show_capi_conf_error: ISDN3#02 |
| 16:25.13 | ZakMcRofl | : conf_error 0x2001 PLCI=0x1103 Command=SELECT_B_PROTOCOL_CONF,0x8495 |
| 16:26.48 | jer | i've got a snom370 phone, and my asterisk reports every time it renews its subscription that it's a sip subscribe for peer without a mailbox: (null). in the sip.conf, i've got the proper mailbox listed, call it phone rings, don't answer it goes to voicemail. i can tap the retrieve button to go to voicemail; not sure where the error is, be it on asterisk or the phone -- google isn't being forthcoming in other people who've had this issue |
| 16:27.12 | BBishop | so .. I guess it isn't possible to use ubr900 with asterisk .. :| |
| 16:27.58 | *** part/#asterisk harryv (n=harry@0x55508034.adsl.cybercity.dk) |
| 16:28.11 | [TK]D-Fender | jer, pastebin your sip.conf entry for it |
| 16:28.20 | jer | sure, one sec |
| 16:29.09 | jer | [TK]D-Fender, http://pastebin.com/m74c71dc1 |
| 16:29.26 | [TK]D-Fender | BBishop, well... what protocol does it use? Do you have access to its configuration? |
| 16:29.43 | [TK]D-Fender | jer, yup, its just your phone then. |
| 16:29.49 | jer | ok |
| 16:30.01 | [TK]D-Fender | jer, can't tell you how it is you have to configure it though... |
| 16:30.25 | jer | [TK]D-Fender, that's fine, at least i have an idea now |
| 16:30.54 | BBishop | [TK]D-Fender, sure .. I'm on serial console .. it uses h323 ? |
| 16:33.34 | [TK]D-Fender | jer, IINM * "pushes" VM notifications to the phone instead of a "subscription" by the phone to a given box. |
| 16:36.10 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 16:38.58 | russellb | well, you can set a mailbox as well as an option that says don't sent MWI unless the phone subscribes to it |
| 16:39.03 | russellb | subscribemwi is the option, i think |
| 16:40.47 | adamowitz | is it possible to buy/lease a US-based mobile phone number from a VoIP service provider to connect to my asterisk box as another line, along with PSTN lines? |
| 16:41.39 | Cybertoy | adamowitz, not sure what you mean but ipkall.com not good enough for that? |
| 16:42.23 | BBishop | :| |
| 16:42.53 | adamowitz | i'm reading there now... what i mean is, can i get a mobile phone number rather than a land-line phone number from them. so far, from such services, i've seen only land-line numbers available. |
| 16:43.08 | Cybertoy | adamowitz, define mobile phone number... |
| 16:43.10 | blitzrage | adamowitz: there isn't really a difference in the US.... |
| 16:43.15 | Cybertoy | in the us they're the same |
| 16:43.43 | adamowitz | if you do a reverse number lookup in a database like 411.com, it tells you whether or not the number is a mobile phone number or a land-line number and who the carrier is. |
| 16:43.47 | adamowitz | i want the former |
| 16:44.05 | blitzrage | adamowitz: get a number from a US mobile carrier and port it to an ITSP? |
| 16:44.09 | Qwell | it's only a best guess by NPA/NXX |
| 16:44.34 | blitzrage | from the viewpoint of callers in the US, there is no difference between mobile and landline |
| 16:44.42 | Qwell | WELL |
| 16:44.43 | blitzrage | (i.e. there isn't a separate area code or anything) |
| 16:44.53 | Cybertoy | I ported my mobile number from one carrier to another and 411.com is showing it wrong... |
| 16:44.53 | Qwell | unless you get free mobile-mobile minutes |
| 16:44.54 | blitzrage | in most areas... unless they are big enough :) |
| 16:45.02 | Qwell | but that isn't what he wants :D |
| 16:45.10 | blitzrage | right :) |
| 16:45.20 | adamowitz | Cybertoy: 411 shows what wrong: landline or carrier? |
| 16:45.28 | blitzrage | I gotta setup a phone game on my old cell number.... |
| 16:45.34 | Cybertoy | carrier... as it's still a mobile number... |
| 16:45.39 | blitzrage | (since that number was so cool...) |
| 16:45.49 | Qwell | was it 800-lmadsen? |
| 16:45.52 | blitzrage | no :) |
| 16:45.56 | blitzrage | 519 591 5119 |
| 16:46.01 | Qwell | neat |
| 16:46.13 | Qwell | I've called 519 before |
| 16:46.21 | Cybertoy | I never ported a mobile to landline or vice versa |
| 16:46.22 | blitzrage | that's southwestern Ontario! |
| 16:46.26 | Qwell | I know! |
| 16:46.31 | blitzrage | you do?! |
| 16:46.33 | blitzrage | niiiiice |
| 16:46.39 | Qwell | well, I knew it was Ontario, anyways :D |
| 16:46.47 | blitzrage | heh |
| 16:46.54 | Qwell | or maybe it was 509 O.o |
| 16:46.59 | Qwell | is that also Ontario? |
| 16:47.07 | blitzrage | don't think so |
| 16:47.12 | [TK]D-Fender | 905 <- |
| 16:47.13 | blitzrage | unless maybe it was the norht... ? |
| 16:47.20 | blitzrage | 905, 416, 647, 519 |
| 16:47.29 | [TK]D-Fender | 867-5309! |
| 16:47.29 | Qwell | mmm, wasn't 905 |
| 16:47.57 | BBishop | damn ubr .. grr |
| 16:48.12 | adamowitz | there are web services that require me entering a mobile number to use them, but i don't have a mobile number, so i'm just wondering if i can use asterisk running on my laptop with a number from a VoIP service provider as a substitute. I don't know whether or not those web services have a way to verify that the number provided is indeed a mobile number or not. |
| 16:48.57 | BBishop | they have :) .. the mobile companies in any country can be checked .. |
| 16:49.06 | Cybertoy | adamowitz: I'm thinking they might send you a SMS |
| 16:49.06 | BBishop | don't ask how .. but they have to have a GSM license ... |
| 16:49.11 | BBishop | and those are public somewhere.. |
| 16:49.20 | blitzrage | adamowitz: in the US... no |
| 16:49.41 | blitzrage | adamowitz: you could just use the same number for both fields? |
| 16:50.43 | Cybertoy | ha... |
| 16:50.52 | adamowitz | blitzrage: are you saying that there is no way for them to verify that a number is indeed associated with a mobile phone rather than a landline? |
| 16:50.59 | Cybertoy | i found a friend's mobile phone number that used to be a landline ... and 411.com has it listed as landline... |
| 16:51.04 | Cybertoy | I know it's a mobile though. |
| 16:51.17 | blitzrage | adamowitz: I have no idea what kind of web form actually verifies numbers.... |
| 16:51.18 | BBishop | oh dang .. :| |
| 16:51.37 | adamowitz | ok. |
| 16:51.48 | BBishop | still .. the company that has that number ( even if it shared it to another ) has to know what kind of number is it .. |
| 16:52.42 | [TK]D-Fender | BBishop, just forward your cell to * and set your outbound CID to your cell # |
| 16:53.20 | BBishop | I wish I knew what you were talking about :) |
| 16:53.23 | adamowitz | here's another data point: my friends sometimes send me SMS messages to my landline, and they tell me that they get a message back from their SMSC telling them that that is a landline. And the SMSC must have a way of verifying landline versus mobile (or maybe it's just SMS-capable or not) so that it can sense when to voice-synthesize the SMS. |
| 16:54.03 | [TK]D-Fender | BBishop, ... CALLER-ID |
| 16:54.07 | Cybertoy | yeah .. I'm guessing it's when SMS can't be delivered... that's probably the only way to find out if it's a mobile or landline. |
| 16:54.19 | Cybertoy | there's no SMS to landlines in the US ... |
| 16:54.23 | Cybertoy | at least not that I know of. |
| 16:54.31 | BBishop | well .. I don't have a problem with it yet .. I'm just tryin' to figure out how to make the ubr900 work with a simple asterisk conf :) |
| 16:54.59 | adamowitz | Cybertoy: there is SMS to landlines in the US, but they get voice-synthesized. Or is that what you meant? |
| 16:55.33 | Cybertoy | no ... in Europe you can send SMS to landline phones and if you have the right equipment you can see them on the handset just like on a mobile. |
| 16:55.39 | Cybertoy | i don't think that's possible in the US . |
| 16:55.40 | RoyK | hi. for what i've heard, asterisk recodes dtmf even if inband is specified. is it possible to disable this feature? I just want the data carried untouched in the rtp stream |
| 16:56.04 | blitzrage | RoyK: not in IAX2 afaik |
| 16:56.15 | RoyK | blitzrage: RTP - this is SIP |
| 16:56.21 | blitzrage | RoyK: then you are mistaken |
| 16:56.27 | blitzrage | afaik... :) |
| 16:56.47 | blitzrage | someone with better knowledge can correct me if I'm wrong |
| 16:57.50 | adamowitz | Cybertoy: if I understand you correctly, that's a condition with the US PSTN that prevents true SMS (meaning, not voice-synthesized) capability that the European PSTN does not share. So even if I had the right equipment in the US, I couldn't get true text SMS messages over the US PSTN. Is that what you're saying? |
| 16:58.28 | adamowitz | And if so, can anyone verify that one way or another with certainty? |
| 16:59.09 | Cybertoy | right |
| 16:59.18 | jer | Qwell, 509 isn't in ontario. ontario is: 519, 416, 647, 905, 705 and 613 only |
| 16:59.23 | Cybertoy | I just sent a sms to my landline ... and my carrier just sends me sms back saying that the number is not valid. |
| 16:59.47 | adamowitz | Cybertoy: where are you? |
| 17:00.39 | adamowitz | jer says it works in Canada (SMS to landline) and I confirm it works in the US. Voice-synthesized, but the SMS does not get rejected. |
| 17:00.43 | Cybertoy | adamowitz: New York |
| 17:01.01 | Cybertoy | probably depends on the carrier you're with on the mobile. |
| 17:01.19 | adamowitz | which carrier are you using with your mobile? |
| 17:01.25 | Cybertoy | AT&T |
| 17:01.29 | adamowitz | hm. |
| 17:02.26 | jer | yeah it only works in Canada as voice synthesized (on all the carriers i've ever used, and that's all of them in centeral->eastern Canada) |
| 17:02.44 | Cybertoy | adamowitz: what are you trying to achieve? |
| 17:03.13 | adamowitz | So here's another question: if I configure my landline correctly with asterisk for SMS, and I then have a friend send an SMS to that same number (the landline), does the system now see that as a mobile number (since * handles the incoming SMS)? |
| 17:03.27 | adamowitz | Cybertoy: I mentioned it above. web services that require a mobile number. |
| 17:03.45 | jer | adamowitz, when i send a sms to my landline (which is connected into *) from my mobile phone, the callerid shows up as the mobile number |
| 17:04.14 | *** join/#asterisk BlackSlik (n=James@77.246.64.134) |
| 17:04.17 | Cybertoy | I don't believe landlines are capable of handling sms's in the US ... |
| 17:04.50 | adamowitz | jer: that's what I would expect... do you have the SMS apps in * configured to handle SMS (as per http://www.voip-info.org/wiki-Asterisk+cmd+Sms ) |
| 17:05.15 | BlackSlik | Hello everybody, i have my asterisk set up and all in good shape , how do i start 2 make a call remotely because i am actually doing everything Virtual |
| 17:05.40 | Cybertoy | adamowitz: your linke even speaks about SMS in the USA ... |
| 17:05.50 | adamowitz | right. |
| 17:06.07 | Cybertoy | adamowitz: when I was in Europe I was able to receive and send SMS through my ISDN line ... here in the US having ISDN is non-existant for consumers. |
| 17:06.38 | jer | adamowitz, no i don't |
| 17:06.42 | adamowitz | i haven't looked recently, but i coulda swore that isdn was an option for residential. |
| 17:06.48 | adamowitz | no? |
| 17:06.51 | Cybertoy | adamowitz: I think the easiest for you is to get a US mobile SIM card that allows roaming and use that... :) |
| 17:07.09 | Cybertoy | never saw isdn for residential areas here... |
| 17:07.10 | adamowitz | that's an idea. thanks Cybertoy. |
| 17:07.27 | *** join/#asterisk IgorG (n=FeedomPa@host-90-188-188-119.pppoe.omsknet.ru) |
| 17:08.16 | Cybertoy | adamowitz: AT&T have attractive pre-pay plans that my European friends use when they're here... I'm not sure though if they allow for roaming... look at http://www.gophone.com for example though. |
| 17:08.37 | adamowitz | jer: any interest in fiddling with that? i could work on my * box at the same time and we could compare notes. it's not really clear to me at this point exactly how to do that, in spite of the examples. there is mention of a gateway, but not much detail. |
| 17:08.41 | Cybertoy | then again I've never seen web-sites that require mobile phones... :) |
| 17:08.57 | Qwell | google calendar reminders |
| 17:09.15 | adamowitz | exactly |
| 17:09.33 | Cybertoy | a-ha... |
| 17:09.46 | Cybertoy | so yeah .. anything that tries to send you an sms. |
| 17:10.02 | Qwell | before you get a cell for that - check the list, last I checked, there were a few providers that they couldn't/wouldn't support |
| 17:10.04 | jer | adamowitz, i haven't been following what you want to accomplish? you want to intercept the sms message and display it elsewhere w/o voice synthesis? |
| 17:10.18 | jer | or sending? |
| 17:10.37 | adamowitz | right, i'd like to get a landline to truly handle an sms message; without voice-synthesis. |
| 17:10.45 | adamowitz | receiving is what i mostly care about. |
| 17:11.12 | jer | ah. i have no way to test it though, all the mobiles i have, the providers detect a landline, and just call with the voice synthesized version of the sms |
| 17:11.22 | jer | that happens before it ever hits my * |
| 17:11.30 | adamowitz | oh. |
| 17:12.14 | jer | if there was a known callerid number for sms->landline here, i could detect on that and intercept it. potentially doing speech->text on it and sending that elsewhere; but it doesn't here unfortunately |
| 17:12.26 | adamowitz | see, that's telling me that the networks have a database of numbers with accurate detail as to whether line x is a landline or mobile. other comments in this room indicated that such capabilities didn't exist, unless I misunderstood. |
| 17:13.02 | jer | i don't know if it does in the US or not; but what i've seen, would indicate that such mechanisms exist north of the 49th parallel |
| 17:13.21 | adamowitz | ok, well that's probably indicative of below also. |
| 17:13.35 | *** join/#asterisk rmayorga_ (n=rmayorga@unaffiliated/rmayorga) |
| 17:15.40 | adamowitz | and if so, then i come back to my original question of whether or not it is possible to lease a mobile phone number (not a landline which is what I see offered at every VoIP gateway to the PSTN that I've checked) from some VoIP service provider. If I could do that, then I should be able to get my * box to truly handle sms functionality, right? |
| 17:16.17 | adamowitz | Is anyone doing what's described here: http://www.voip-info.org/wiki-Asterisk+cmd+Sms ? |
| 17:16.23 | adamowitz | or some equivalent? |
| 17:17.10 | adamowitz | BRB |
| 17:17.35 | BlackSlik | Hello everybody, i have my asterisk set up and all in good shape , how do i start 2 make a call remotely because i am actually doing everything Virtual |
| 17:20.03 | *** join/#asterisk Grnd-Wire (n=grundofw@65.101.128.1) |
| 17:20.10 | Grnd-Wire | good morning everyone! |
| 17:21.38 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) |
| 17:25.59 | BlackSlik | Hello everybody, i have my asterisk set up and all in good shape , how do i start 2 make a call remotely because i am actually doing everything Virtual |
| 17:29.28 | [TK]D-Fender | BlackSlik, what do mean "remotely" vs "virtual"? |
| 17:30.03 | BlackSlik | remotely Sir |
| 17:30.26 | [TK]D-Fender | BlackSlik, that is NOT any clearer. |
| 17:31.10 | Grnd-Wire | heh |
| 17:32.14 | BlackSlik | ok what i mean is i have my asterisk installed and runing remotely on a sever, all other stuff on it runing good, Now i want to make a call test from the remote sys |
| 17:32.21 | BlackSlik | how do i do that |
| 17:33.10 | [TK]D-Fender | BlackSlik, server is "remote" from what you just said (whatever that means), and now you want to do what with it? |
| 17:33.28 | Grnd-Wire | [TK]D-Fender: I think he's asking about how to use "console dial" and stuff.. |
| 17:34.02 | [TK]D-Fender | Grnd-Wire, I wouldn't start assuming anything with a person talking like that. |
| 17:34.21 | Grnd-Wire | [TK]D-Fender: I am assuming nothing.. |
| 17:34.25 | [TK]D-Fender | Grnd-Wire, Apparently EVERYTHING is "remote" in his "setup". |
| 17:34.55 | Grnd-Wire | "I have my asterisk installed and runing remotely on a sever.. Now i want to make a call test from the remote sys" |
| 17:35.20 | Grnd-Wire | BlackSlik: Do you have a phone configured on your asterisk server? |
| 17:36.08 | BlackSlik | nope no phone connected but i configured to use a VOIP |
| 17:36.33 | Grnd-Wire | BlackSlik: So what do you expect to accomplish with a test call, since you're not even COMPLETELY setup yet? |
| 17:36.57 | BlackSlik | because i want to basically use it for outbound calls, made a script on perl 2 make an auto call |
| 17:37.26 | Grnd-Wire | BlackSlik: hmm.. So does your Perl script work? |
| 17:38.29 | BlackSlik | yea but i wanna make a test call with series of number , starting asterisk 2 make the call is what i am really confuse :| |
| 17:38.54 | Grnd-Wire | hmm.. I am not even seeing any good resources for how to do that.. |
| 17:39.01 | Grnd-Wire | http://www.asteriskguru.com/tutorials/cli_cmd_14_page6.html |
| 17:39.03 | BlackSlik | umh......... |
| 17:39.20 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 17:39.37 | Grnd-Wire | You ARE looking for "console dial" and "console hangup".. but you'll need some dialplan logic probably to make it useful. |
| 17:39.48 | Grnd-Wire | [TK]D-Fender: Can you add anything at this point? |
| 17:39.58 | BlackSlik | okay |
| 17:40.53 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:43.31 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 17:43.31 | *** mode/#asterisk [+o russellb] by ChanServ |
| 17:44.08 | *** join/#asterisk adamowitz (n=adamowit@ip68-14-27-175.ri.ri.cox.net) |
| 17:45.02 | Grnd-Wire | [TK]D-Fender: I had an idea yesterday, and I am wondering why noone has ever implemented it on any system I have ever heard of.. |
| 17:45.12 | Grnd-Wire | [TK]D-Fender: You wanna know what it is? |
| 17:46.19 | Cybertoy | what is it? |
| 17:47.18 | Grnd-Wire | Why do voicemail systems always ask for extension, wait for a pound, then prompt for password? Since the extension number is usually fixed - why couldn't you just accept 1001234# and parse it out.. |
| 17:47.32 | Grnd-Wire | It would be much much quicker, especially when you're dialing in from a cell phone.. |
| 17:48.04 | [TK]D-Fender | Grnd-Wire, most of the time VM systems know who's box to access and only ask for the password |
| 17:48.07 | Grnd-Wire | As long as your extension length is fairly predictable, the password length can be arbitrary.. |
| 17:48.14 | Cybertoy | you can do that now ... |
| 17:48.16 | Grnd-Wire | [TK]D-Fender: Well, I'm referring to when you call in from the outside. |
| 17:48.38 | [TK]D-Fender | Grnd-Wire, only reason it asks for a box is because you didn't TELL VoicemailMain which box to use |
| 17:48.45 | Grnd-Wire | Cybertoy: I know I can write the code to do it, but it the core functionality already there? |
| 17:49.00 | Grnd-Wire | [TK]D-Fender: uh huh - Because when you call in from the outside, it doesn't know.. |
| 17:49.18 | [TK]D-Fender | Grnd-Wire, which box should it access and how chould it "know"? |
| 17:49.34 | *** join/#asterisk ManxPower (n=manxpowe@31.sub-75-200-19.myvzw.com) |
| 17:49.58 | Grnd-Wire | [TK]D-Fender: The only way would be if you were wasting a ton of DIDs for voicemail access.. or you prompt for "passcode" wich includes your box number.. |
| 17:49.59 | Cybertoy | Grnd-Wire: you'll just have to write the dialplan code around it to call the right box number based on your caller-id. |
| 17:50.15 | [TK]D-Fender | Grnd-Wire, or you check the callerid of the person calling |
| 17:50.33 | Grnd-Wire | [TK]D-Fender: ok, I absolutely hate that behaviour.. |
| 17:51.08 | Grnd-Wire | I can't call my cell from from my asterisk system, because Verizon is too smart for me.. when they see my cell phone's caller ID coming into the system, it assumes that I am checking voicemail - instead of ringing my cell phone. |
| 17:51.09 | [TK]D-Fender | Grnd-Wire, then feel free "Read" a composite box & pass string and jumpt right into the mailbox then |
| 17:52.04 | Grnd-Wire | [TK]D-Fender: Right, my question wasn't could it be done, because I don't need you to tell me that.. My question was.. Is there a reason it hasn't already been done? Are there unforseen consequences, security risks, useability issues, etc? |
| 17:53.47 | *** join/#asterisk Tili (n=tili@203.170.74.29) |
| 17:54.07 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 17:55.25 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 17:58.40 | *** part/#asterisk mog (n=mog@c-68-62-172-83.hsd1.al.comcast.net) |
| 17:59.16 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 18:02.07 | Tili | i sometimes get issues on PRI channels in getting DTMF? what can it be and how can i resolve |
| 18:04.32 | ManxPower | Grnd-Wire: Asterisk is not a PBX. Asterisk is a toolkit that allows you to build a PBX. |
| 18:04.51 | ManxPower | Tili: increase or decrease the rxgain for the PRI channels. |
| 18:05.16 | ManxPower | Grnd-Wire: So basically anyone can get into your voicemail if they spoof their callerid? |
| 18:05.35 | [TK]D-Fender | Grnd-Wire, And frankly you're talking about saving 1 little entry which apparently every system out there makes youdo as well. So since this is wholly unnecessary you can chalk it up to "who cares?" |
| 18:05.39 | Tili | ManxPower: Don't PRI Channels use D-Channel for DTMF which is signalling |
| 18:05.50 | Tili | and digital |
| 18:06.43 | ManxPower | Tili: no, only for call setup. |
| 18:06.52 | Grnd-Wire | [TK]D-Fender: Well, it's the little things that make systems unique.. and I'm sure you're aware companies patent the [usually ccompletely] useless things.. |
| 18:06.59 | ManxPower | and that's not DTMF, that is just a call setup message |
| 18:07.33 | Grnd-Wire | So thanks - You have given me the feedback I required.. |
| 18:07.46 | Tili | ManxPower: thanks. i was thinking all wrong way. I will increase rxgain. |
| 18:08.18 | [TK]D-Fender | Grnd-Wire, Sorry, people hav hacked this stuff before, just that nobody considers it worth making a primary part of *. Feel free to write a patch for it. |
| 18:09.50 | ManxPower | The Asterisk Way is not "build it for you", but is more of "build the tools to let you build what you want". That is the reason so many things must be coded in the dialplan. |
| 18:09.52 | Grnd-Wire | [TK]D-Fender: It doesn't need to be part of asterisk - It's clearly something you can do in the dialplan.. I was asking about the useability side of things. I know that useability engineers who are paid alot more than me at Siemens, Cisco, Avaya, Fujitsu, Nortel would be doing this if it had value. |
| 18:10.09 | Grnd-Wire | <sigh> |
| 18:10.41 | ManxPower | Grnd-Wire: it sounds like you are looking for a PBX, not a PBX toolkit. |
| 18:10.48 | [TK]D-Fender | Grnd-Wire, "This fruitless psychotic rant was brought to you by our sponsors. Ask your doctor is Paxil is right for you!" |
| 18:11.06 | Grnd-Wire | ManxPower: no - I am asking for YOUR personal advice on a specific feature that I am more than capable on putting into my dialplan. |
| 18:11.21 | Grnd-Wire | [TK]D-Fender: no kidding.. ugh - ok, well thanks for your time |
| 18:12.06 | Cybertoy | Grnd-Wire: you might be unclear... I have a phone number that if I dial from my mobile phone asterisk will pick up the callerid and call VoimailMain with the correct mailbox number so that I only have to punch in my password. |
| 18:12.22 | Cybertoy | so I'm doing what you describe within the dialplan. |
| 18:12.45 | [TK]D-Fender | Cybertoy, No, he asked about a single composite entry of box+pass together |
| 18:13.10 | [TK]D-Fender | Cybertoy, not assuming a box and not needing 2 prompts |
| 18:13.32 | Cybertoy | well.. in theory you can do that... but why would you? it's not secure. |
| 18:13.47 | [TK]D-Fender | Cybertoy, Sure it is... you still have to enter a code |
| 18:13.56 | Grnd-Wire | Cybertoy: yeah, explain why it's not? |
| 18:14.09 | Grnd-Wire | AHA! But this is one part of the discussion I wanted to get into.. PERCEPTION of your users.. |
| 18:14.17 | Cybertoy | ah.. wait.. I get it |
| 18:14.26 | Grnd-Wire | hallelujah.. Finally.. |
| 18:14.27 | Cybertoy | you enter the username and password in one entry... that's it? |
| 18:14.35 | Grnd-Wire | ya |
| 18:15.06 | bcnl | yea you could just exten => 1239999,1,VoicemailMain(s123) |
| 18:15.10 | [TK]D-Fender | ok, heading out, back in a while. |
| 18:15.15 | bcnl | no? |
| 18:15.16 | Grnd-Wire | [TK]D-Fender: see ya |
| 18:15.19 | [TK]D-Fender | bcnl, .... no |
| 18:15.40 | bcnl | if your "password" for box 123 was 9999 |
| 18:15.50 | bcnl | but you'd have to manually update the dialplan as users change passwords |
| 18:16.29 | Grnd-Wire | bcnl: In that instance you would just use the normal behaviour of the Voicemail app.. |
| 18:17.09 | bcnl | I only skimmed the backlog, the coffee and bailys has not kicked in and I was awoken by bad ram in a server |
| 18:17.30 | Grnd-Wire | Well one thing is clear to me, no one in here has really given this any thought of there own.. It would seem to be an original idea, as unimportant as it is in the scheme of things. |
| 18:18.13 | bcnl | so you want to run mailbox&pass together in one string |
| 18:18.15 | Tili | why i get that always even though file is owned and mode is all perfect |
| 18:19.09 | Grnd-Wire | bcnl: yeah.. So you just do a Read, then call voicemailMain will the ${VARIABLE:3}, and the password would be ${VARIABLE::4} or whatever.. |
| 18:19.31 | Tili | . /var/spool/asterisk/outgoing/confcall.call: Permission denied, deleting . if it can delete file, why can't it read file? |
| 18:19.33 | Grnd-Wire | If they don't key the password in correctly, Comedian mail will start prompting for it normally at that point.. |
| 18:20.36 | bcnl | hrmm, well best of luck |
| 18:20.42 | Grnd-Wire | heh thanks |
| 18:20.47 | bcnl | adding a hash key doesn't seem like something my users mind doing |
| 18:21.08 | Grnd-Wire | bcnl: It's adding a hash, and having to WAIT .. dude - When I am driving, finding more keys on my PDA phone is a bitch.. |
| 18:21.14 | bcnl | although I have DID's for everyones mailbox, so if they call from a cell they get dumped right into their box |
| 18:21.29 | Grnd-Wire | bcnl: And certainly for bigger installations that is justified.. |
| 18:21.40 | bcnl | and if they call the main number and I have their CID, I just send them to the right box and have them enter their pass |
| 18:21.56 | bcnl | make sure you patent it |
| 18:22.00 | bcnl | ask amazon how |
| 18:22.06 | Grnd-Wire | bcnl: Apparently I should.. :/ |
| 18:22.39 | bcnl | I dunno, almost all of my users are heavy pot smokers (and not all the sharpest knives in the drawer) and I hear no complaints though |
| 18:24.18 | Grnd-Wire | hehehe |
| 18:28.47 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 18:32.58 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 18:42.23 | *** join/#asterisk kickbackit (n=Assi@216.83.78.108) |
| 18:42.52 | ManxPower | Generally, you should not be checking voicemail while driving |
| 18:43.17 | ManxPower | You should also not put on makeup, read the news paper, check e-mail, or have sex while driving. |
| 18:45.05 | kickbackit | I'm having some trouble upgrading to asterisk 1.4 from 1.2.... My dialplan now gives me an error about the call being rejected cause the extention is not found. What is new in 1.4 that is messing my old dial plan up? |
| 18:46.52 | ManxPower | kickbackit: did you read UPGRADE.txt for 1.2 and 1.4? |
| 18:47.06 | ManxPower | Perhaps you should, then come back. |
| 18:47.33 | kickbackit | Thanks, I should... |
| 18:52.38 | bcnl | ManxPower: hey, don't hate on the driving sex |
| 18:52.46 | bcnl | everything else I agree with |
| 18:54.04 | ManxPower | It is never a good thing when the mailserver admin says "opps" |
| 18:55.51 | bcnl | but it's always funny when the mailserver admin goes 'grep -i -l /var/spool/mail/*' |
| 18:55.57 | bcnl | but it's always funny when the mailserver admin goes 'grep -i -l gay /var/spool/mail/*' |
| 18:56.00 | bcnl | damn |
| 18:56.05 | bcnl | way to kill a joke |
| 18:58.54 | *** join/#asterisk bmcghee (n=brentmcg@d66-183-250-149.bchsia.telus.net) |
| 18:59.40 | bmcghee | anyone around that i may pick there brain on a problem with my asterisk install |
| 19:01.35 | bmcghee | Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) |
| 19:01.54 | kickbackit | asterisk -r |
| 19:01.59 | bmcghee | thats what i did |
| 19:02.07 | bmcghee | asterisk isnt launching |
| 19:04.30 | bcnl | is asterisk running? |
| 19:04.42 | bcnl | killall asterisk && asterisk -cvvvv |
| 19:05.30 | bmcghee | asterisk isnt running |
| 19:05.33 | kickbackit | hrm my sip peers don't see to be following the context I send them to... |
| 19:05.57 | bmcghee | i did asterisk -cvvvv and it has no errors |
| 19:06.26 | kickbackit | you sitting at a cli> then? |
| 19:06.46 | bmcghee | nope |
| 19:06.49 | bmcghee | it kicks me out |
| 19:06.52 | bmcghee | when i asterisk -r |
| 19:07.05 | bmcghee | and i do asterisk -cvvvv and it lists then just stops |
| 19:07.08 | ManxPower | If you get that message, it means either Asterisk is not running, or you do not have permission to access that file. |
| 19:07.14 | bmcghee | im root |
| 19:07.30 | ManxPower | bmcghee: asterisk -cvvv should give you an error message where it bombs out. |
| 19:07.35 | bmcghee | i know asterisk isnt loading. i cant figure out why. |
| 19:07.37 | kickbackit | asterisk -c puts you in the cli... You'd have to type quit to get back to the shell at that point asterisk would of stoped. |
| 19:07.56 | jer | my sip trunk is acting funny. it links up, i get "-- Got SIP response 489 "Bad event" back from 66.38.165.231" from the other asterisk server (the one mine connects to) twice, then "sip show channels" on my * doesn't show anything anymore. i can still make calls out, that works fine. it's not a config problem on the other end, because i can add that profile to an ip phone and it works fine. anything obvious i'm missing? (relevant section of sip.conf is h |
| 19:08.05 | bcnl | and if asterisk doesn't start, the last few lines dumped on the screen should tell you what's wrong |
| 19:08.17 | bmcghee | [ Booting... |
| 19:08.18 | bmcghee | [ Reading Master Configuration ] |
| 19:08.18 | bmcghee | [ Initializing Custom Configuration Options ] |
| 19:08.18 | bmcghee | .......... Loading [Sub]Agent Module |
| 19:08.18 | bmcghee | ................................................................................alpha# |
| 19:08.18 | bmcghee | [ Initializing Custom Configuration Options ] |
| 19:08.20 | bmcghee | .......... Loading [Sub]Agent Module |
| 19:08.23 | ManxPower | NO, -c runs asterisk in the foregorund, it does NOT connect you to the existing CLI |
| 19:08.35 | bmcghee | it stops at that part. |
| 19:08.40 | bmcghee | loading sub agent |
| 19:08.42 | ManxPower | bmg505: I suggest you be more careful with your pastes. DO NOT PASTE HERE> |
| 19:08.45 | bmcghee | when just -c |
| 19:08.50 | ManxPower | ~pb |
| 19:08.51 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines 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 |
| 19:08.58 | ManxPower | bmcghee: what version of Asterisk are you using? |
| 19:09.02 | bmcghee | latest |
| 19:09.04 | bmcghee | just dl oit |
| 19:09.05 | bmcghee | it* |
| 19:09.07 | *** join/#asterisk adamowitz (n=adamowit@ip68-14-27-175.ri.ri.cox.net) |
| 19:09.23 | bcnl | ~aastra |
| 19:09.28 | bcnl | ~linksys |
| 19:09.28 | jbot | methinks linksys is a tool of satan |
| 19:09.42 | ManxPower | bmcghee: latest 1.2, latest 1.4, latest AsteriskNOW, latest Asterisk-GUI? |
| 19:09.43 | bcnl | jbot: why do you know not about aastra phones!? |
| 19:09.43 | jbot | why not? |
| 19:09.49 | bcnl | :P |
| 19:09.53 | bmcghee | 1.4 |
| 19:09.54 | bcnl | who's on first? |
| 19:10.56 | ManxPower | bmcghee: now, do asterisk -cvvv again, only this time put the entire output on pastebin.ca |
| 19:11.45 | bmcghee | .ca dont work for me even tho im canadian. it times out. ill post on .com |
| 19:12.05 | kickbackit | Cannot find your TTY (9) |
| 19:12.07 | ManxPower | put it somewhere other than here. |
| 19:12.21 | kickbackit | would that stop it from executing all the way? |
| 19:12.41 | ManxPower | kickbackit: we are still waiting for some useful information |
| 19:12.45 | bmcghee | http://pastebin.com/d443fbe6a |
| 19:14.42 | ManxPower | bmcghee: I suspect the problem is that you are using non-standard paths. |
| 19:15.08 | bmcghee | its bsd |
| 19:15.19 | ManxPower | How did you specify /usr/local using the build process? as a setting in menuconfig or via configure? |
| 19:15.43 | bmcghee | i just ran ./configure |
| 19:15.50 | ManxPower | bmcghee: I don't care if it's ET's OS, installing asterisk in paths it does not expect is troublesome. |
| 19:16.03 | ManxPower | bmcghee: try make menuconfig |
| 19:16.11 | jer | bmcghee, or just use the asterisk port |
| 19:16.20 | bmcghee | i cant get the asterisk port to work |
| 19:16.23 | ManxPower | jer: is the asterisk port 1.4.15? |
| 19:16.29 | jer | ManxPower, oh, not of 1.4.15 |
| 19:16.39 | jer | 1.4.11 currently, until the ports freeze is over |
| 19:16.49 | ManxPower | so it has the massive Postgress security issue |
| 19:17.05 | jer | sure, so work around it, don't use postgres =D |
| 19:17.12 | ManxPower | so Asterisk in ports is 4 versions behind. |
| 19:17.31 | ManxPower | bmcghee: I would suggest you ask on the asterisk-bsd mailing list. |
| 19:17.47 | jer | ManxPower, it's because there's a ports freeze in effect right now (no updates are made unless there's a problem building a port) .. once the freeze is over (when 7.0/6.3 are released) then there will be updates again |
| 19:18.21 | kickbackit | http://pastebin.com/d8d4da69 When someone gets a second |
| 19:19.11 | jer | my sip trunk is acting funny. it links up, i get "-- Got SIP response 489 "Bad event" back from 66.38.165.231" from the other asterisk server (the one mine connects to) twice, then "sip show channels" on my * doesn't show anything anymore. i can still make calls out, that works fine. it's not a config problem on the other end, because i can add that profile to an ip phone and it works fine. anything obvious i'm missing? (relevant section of sip.conf is h |
| 19:19.50 | ManxPower | jer: And that is one of the reasons I recommend building from source. |
| 19:20.02 | ManxPower | ~siptrunk |
| 19:20.03 | jbot | i heard siptrunk is Asterisk does not support SIP Trunks. Set trunk=no in sip.conf and then set up the device normally in sip.conf. |
| 19:20.20 | jer | ManxPower, ah |
| 19:20.46 | file | jer: do you have mailbox= set? if so MWI status is being sent to the other Asterisk, and it doesn't support receiving it... so it sends back 489 Bad Event, nothing really wrong... |
| 19:21.23 | jer | file, ah i don't have mailbox= set, and i know there to be messages there |
| 19:21.38 | ManxPower | *scream* I really wish I could figure out this "Incorrect MAC received on packet" message/prob/em |
| 19:22.56 | *** part/#asterisk coldsteal (n=coldstea@unaffiliated/coldsteal) |
| 19:23.56 | ManxPower | jer: setting mailbox= would create that problem. Since you do not have mailbox= set to something, then it must be some other problem |
| 19:24.11 | jer | ManxPower, right, but it's nice to know |
| 19:24.53 | file | a sip debug would tell you exactly what sort of packet is being sent out to trigger the Bad event response |
| 19:26.58 | *** join/#asterisk spatulamaan (n=ggilmore@67-40-197-239.tukw.qwest.net) |
| 19:27.47 | kickbackit | So anyone see a reason my my context is hosed? http://pastebin.com/d8d4da69 |
| 19:30.36 | jer | urgh; you know what, i'm just going to ask for authorization to set my link up as an IAX2 trunk instead of SIP |
| 19:30.47 | jer | file, thanks, sip debug showed some interesting data |
| 19:32.14 | ManxPower | kickbackit: for one thing you are using a _ prefix on extensions that are not pattern matches. |
| 19:32.21 | ManxPower | Also you sure do have a lot of crap in your sip.conf |
| 19:32.23 | *** join/#asterisk techie (n=techie@adsl-76-214-21-156.dsl.lsan03.sbcglobal.net) |
| 19:33.12 | RoyK | hi. for what i've heard, asterisk recodes dtmf even if inband is specified. is it possible to disable this feature? I just want the data carried untouched in the rtp stream |
| 19:33.31 | ManxPower | also Asterisk can be picky about spaces and you don't have spaces around some of your => |
| 19:33.53 | kickbackit | Yeah I Added the _ at a suggestion from someone. removed them again. Ok I will try the spaces too. |
| 19:33.56 | ManxPower | RoyK: then you would set your device to use inband, and tell asterisk to use INFO or something like htat. |
| 19:34.11 | ManxPower | I am assuming, of course you are talking about SIP./ |
| 19:34.38 | ManxPower | kickbackit: _ is only useful for pattern matches. |
| 19:36.01 | ManxPower | kickbackit: I figured you must have been talking to an idiot since you have dtmf=inband in your sip.conf. That options does not exist. |
| 19:36.05 | RoyK | ManxPower: I mean if asterisk is doing sip/sip bridging - I don't want asterisk to touch the stuff in RTP |
| 19:36.29 | ManxPower | RoyK: Yes, I understand. Do you understand what I said? |
| 19:36.42 | kickbackit | well it was a 1.2 config copied over to 1.4.... was that option there in 1.2? |
| 19:36.54 | ManxPower | RoyK: obviously you have to be using ulaw or alaw codecs for this to work |
| 19:37.00 | ManxPower | kickbackit: no, it was not. |
| 19:37.11 | kickbackit | lol, oh well... |
| 19:37.13 | RoyK | ManxPower: I understood, yes, but if asterisk is using SIP INFO, it might come out in the double at the far end. I'm using ALAW |
| 19:37.18 | ManxPower | kickbackit: as you would know if you looked at sip.conf.sample |
| 19:37.31 | RoyK | afaik g.726 can handle dtmf as well |
| 19:37.34 | ManxPower | RoyK: if the phone is NOT sending info, how would htat happen. |
| 19:37.57 | kickbackit | its actually a config file from my voip provider... so thats interesting. |
| 19:38.00 | ManxPower | RoyK: It would be the first compressed codec in the history of the universe to handle DTMF. |
| 19:38.09 | RoyK | ManxPower: it's in the spec |
| 19:38.12 | ManxPower | kickbackit: like I said, an idiot |
| 19:38.15 | blitzrage | kickbackit: well, the fact that you're call 3006 and only 3005 exists |
| 19:38.26 | RoyK | ManxPower: if asterisk is reading dtmf and resending SIPĀ INFO, it may be bad |
| 19:38.44 | ManxPower | blitzrage: read it again, he's calling 3005 from 3006 |
| 19:38.55 | kickbackit | I did change them |
| 19:38.55 | kickbackit | exten=>3232093006,1,answer |
| 19:38.58 | RoyK | ManxPower: anyway - imho there should be a way to make inband wor just like inband with bridging, so that asterisk doesn't touch the media stream |
| 19:39.03 | bmcghee | w00t asterisk started!!!! |
| 19:39.07 | blitzrage | fuck I'm having a bad day... I can't read anything right :) |
| 19:39.15 | ManxPower | RoyK: if you set the dtmfmode=INFO, asterisk should not be looking for inband dtmf, and in fact many people do this. |
| 19:39.15 | bmcghee | alpha*CLI> |
| 19:39.31 | ManxPower | RoyK: then use SER |
| 19:39.32 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 19:39.46 | *** join/#asterisk mordaunt (n=mordaunt@unaffiliated/mordaunt) |
| 19:39.49 | blitzrage | B2BUA != Proxy :) |
| 19:40.00 | RoyK | ManxPower: but isn't it a little backwards to tell asterisk to use info when what you want is inband and just tell asterisk to not touch it? |
| 19:40.29 | RoyK | ManxPower: we're moving to SER, indeed, but that doesn't make it a bad idea to make asterisk better, does it? |
| 19:40.42 | ManxPower | RoyK: since Asterisk does not have a "don't touch my inband dtmf" option (and probably never will, since Asterisk is not a SIP proxy).... |
| 19:41.19 | RoyK | ManxPower: I'm not saying it's a proxy, but it should be possible to tell it not to meddle with the media |
| 19:41.20 | ManxPower | of course the real issue is why you need inband in the first place. |
| 19:41.36 | ManxPower | RoyK: B2BUAs mess with the media. |
| 19:41.52 | ManxPower | but if you want to submit a patch I'm sure the 4 other people that need this feature will be happy. |
| 19:41.54 | RoyK | ManxPower: bridging as a term is retransmitting what you get. what you're saying is gatewaying |
| 19:42.20 | RoyK | b2buas doesn't necessarily mess with media. using SER as a b2bua doesn't |
| 19:42.23 | ManxPower | RoyK: asterisk uses the term "bridge" for many totally diffent things. |
| 19:42.43 | RoyK | asterisk uses quite a few terms where they shouldn't - I'm aware of that |
| 19:43.06 | ManxPower | I gave you the solution to work around the fact Asterisk does not have the feature you want. |
| 19:43.35 | RoyK | thanks |
| 19:43.59 | RoyK | but still I wonder why you refuse to have it implemented - it's quite elementary, and it shouldn't be too hard |
| 19:44.42 | ManxPower | 1) because I am not a developer and 2) I am not a developer and 3) I have no interest in being a developer. |
| 19:44.52 | kickbackit | argh still rejecting... |
| 19:45.00 | ManxPower | In fact, I hate writing code. |
| 19:45.20 | ManxPower | kickbackit: maybe you should try to simplify it. |
| 19:46.06 | kickbackit | Well I deleted a lot of extentions.conf... It just goes to the context now with no includes... |
| 19:46.59 | ManxPower | kickbackit: http://pastebin.com/m2bb85cce |
| 19:47.39 | kickbackit | without the patern matching I assume? |
| 19:47.47 | ManxPower | without _ |
| 19:51.34 | kickbackit | I added a exten=>s,1,noop("Test") to the top of [local] and it does not display.... |
| 19:51.43 | kickbackit | its not sending the call to the context then? |
| 19:52.22 | ManxPower | It would not, because "s" only matches when there is NO destination extension. |
| 19:52.46 | ManxPower | pretty much only calls from IAXtel, and FXO signaled ports. |
| 19:55.22 | ManxPower | It would have been more correct to use exten => "",1,Noop but that would confuse people, so Asterisk calls it s instead if "" |
| 20:00.32 | ManxPower | I hate having a cold |
| 20:03.35 | bmcghee | max |
| 20:03.36 | bmcghee | manx |
| 20:03.39 | bmcghee | ugh! |
| 20:04.00 | bmcghee | i rm -rf wrong folder |
| 20:04.04 | bmcghee | deleted all asterisk files |
| 20:04.06 | bmcghee | how i reinstall them |
| 20:05.09 | ManxPower | make install |
| 20:05.18 | bmcghee | gmake (bsd) |
| 20:05.21 | bmcghee | and it didnt reinstall them |
| 20:05.27 | bmcghee | how i set a file location |
| 20:05.31 | bmcghee | it didnt set it to default |
| 20:05.35 | ManxPower | of course, you didn't say WHICH directory you removed. |
| 20:05.54 | bmcghee | asterisk |
| 20:05.54 | bmcghee | lol |
| 20:06.02 | ManxPower | it could be /etc/asterisk /var/lib/asterisk /var/spool/asterisk |
| 20:06.07 | bmcghee | /usr/local/etc/asterisk |
| 20:06.17 | bmcghee | /etc/asterisk doesnt exist |
| 20:06.18 | ManxPower | There are no default files in /etc/asterisk |
| 20:06.50 | ManxPower | I will only talk about the default locations, you can translate that to whatever directory structure you used. |
| 20:07.00 | bmcghee | i need all the files that are installed ie (asterisk.conf, sip.conf etc) |
| 20:07.02 | ManxPower | "make samples" will put the non-working sample config files in /etc/asterisk |
| 20:07.29 | ManxPower | but those are NOT default configs, those are sample config files that tries to show every option available (and it doesn't even do that) |
| 20:07.58 | ManxPower | Expect asterisk to not work is you use the sample configs |
| 20:08.38 | bmcghee | the files that are installed by default. those are the samples? |
| 20:09.11 | bmcghee | im missing all the modules also i made a mess |
| 20:10.08 | ManxPower | no, files in /etc/asterisk are not created by "make install" They are created by "make samples" |
| 20:10.10 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.139.102) |
| 20:10.27 | ManxPower | bmcghee: I already told you to look at asterisk -bsd |
| 20:11.12 | ManxPower | And as I said, we can't help you with installing in any directories other than the default ones. |
| 20:12.16 | bmcghee | how i set it to defaults /etc |
| 20:12.22 | bmcghee | it defaults to /usr/local/etc |
| 20:12.51 | *** join/#asterisk RoyKa (n=roy@ip-59-29-149-91.dialup.ice.no) |
| 20:16.35 | ManxPower | ManxPower: bmcghee: I already told you to look at asterisk -bsd |
| 20:16.40 | ManxPower | ~mailinglist |
| 20:16.41 | 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 |
| 20:17.31 | *** join/#asterisk marcan (i=1337@host214-144.cvd.fit.edu) |
| 20:20.41 | MrTelephone | hey a lot of people have problems with static on voicemail prompts and other prompts by asterisk |
| 20:20.46 | MrTelephone | i just started having that problem today |
| 20:21.01 | MrTelephone | is there a possible reason.. ? |
| 20:21.02 | *** join/#asterisk saftsack (n=saftsack@pD9E04A1D.dip.t-dialin.net) |
| 20:23.42 | *** join/#asterisk tobias (n=tobias@user-0ce2hr3.cable.mindspring.com) |
| 20:30.32 | *** join/#asterisk BlackSlik (n=James@77.246.64.134) |
| 20:30.43 | *** join/#asterisk andreiu_x (n=andreiu_@84.126.96.217.dyn.user.ono.com) |
| 20:30.49 | *** part/#asterisk andreiu_x (n=andreiu_@84.126.96.217.dyn.user.ono.com) |
| 20:30.56 | ManxPower | Cite your source for "a lot of people" |
| 20:30.57 | *** join/#asterisk andreiu_x (n=andreiu_@84.126.96.217.dyn.user.ono.com) |
| 20:31.04 | andreiu_x | james ? |
| 20:31.19 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 20:31.37 | BlackSlik | Yea |
| 20:31.39 | BlackSlik | here |
| 20:33.39 | *** join/#asterisk GoldFingaZ (n=wt5@bas13-ottawa23-1177809381.dsl.bell.ca) |
| 20:35.52 | bmcghee | ManxPower: http://pastebin.com/d4952b0a9 |
| 20:36.19 | kickbackit | hrm I reloaded all my old config files from the 1.2 machine and its all working... |
| 20:36.39 | kickbackit | minus some issues with upgraded and depricated functions. |
| 20:36.44 | kickbackit | working on those now. |
| 20:36.51 | kickbackit | thanks ManxPower |
| 20:38.53 | BlackSlik | Good evenin every one, still stuck with what i am tring to do, i am trying to add my asterisk auto dial script to my asterisk Box which also have trixbox , how do i go about it |
| 20:39.10 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 20:39.31 | ManxPower | BlackSlik: you ask on the trixbox channel. |
| 20:40.01 | ManxPower | This is not the trixbox channel. We hate trixbox, we feed it's users to the aligators. |
| 20:40.31 | *** join/#asterisk alephcom (n=chatzill@d142-59-43-74.abhsia.telus.net) |
| 20:41.04 | ManxPower | ~trixbox |
| 20:41.05 | jbot | [~trixbox] Trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
| 20:41.06 | ManxPower | ~gui |
| 20:41.07 | jbot | somebody said gui was (Graphical User Interface) Any system that uses graphics to represent the functions of a program. All Windows operating systems are GUI. or point-and-drool, double-click, drag-and-drop, ... or at http://pla-netx.com/linebackn/guis/index.html. Of course Real Programmers use the command line interface. See cli |
| 20:41.09 | ManxPower | ~zeeek |
| 20:41.09 | jbot | i guess zeeek is someone who once said "learning asterisk using a GUI is like learning sex through masturbation. You'll never get to the good stuff." |
| 20:41.26 | BlackSlik | ook am sorry |
| 20:41.28 | ManxPower | ~amp |
| 20:41.29 | jbot | it has been said that amp is NOT supported here! People using it should join #freepbx (FreePBX is the new name of AMP) |
| 20:41.34 | ManxPower | ~freepbx |
| 20:41.35 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 20:43.29 | bmcghee | ManxPower: http://pastebin.com/d4952b0a9 ?? any ideas |
| 20:44.25 | MrTelephone | ManxPower I've seen other people complain about static on the prompt wavs |
| 20:44.38 | MrTelephone | usually it was an irq issue |
| 20:46.58 | MrTelephone | I just had a chance to stop now and restart and the static is still there |
| 20:47.04 | MrTelephone | shit- |
| 20:47.14 | MrTelephone | probably needs a reboot i guess |
| 20:47.20 | alephcom | Anybody up for some audiocodes issues? |
| 20:47.36 | alephcom | I'm using it, pstn->ip calls far fine. |
| 20:47.42 | MrTelephone | 15:47:31 up 128 days, 5:28, 3 users, load average: 0.06, 0.15, 0.17 |
| 20:47.47 | alephcom | ip -> FXS calls are fine. |
| 20:47.52 | MrTelephone | my pstn calls sound fine |
| 20:48.00 | MrTelephone | just the voicemail chick sounds fuzzy |
| 20:48.02 | alephcom | the dtmf tones aren't working properl on ip -> pstn calls. |
| 20:48.08 | alephcom | http://pastebin.com/m23b2dfcb |
| 20:48.25 | alephcom | $50USD if somebody can help me fix it. |
| 20:50.10 | MrTelephone | shit i'll help |
| 20:50.11 | MrTelephone | lol |
| 20:50.30 | MrTelephone | whats your setup like alephcom |
| 20:51.02 | MrTelephone | i offered 50 bucks CDN to somoene who could help me and i ended up having to do it myself |
| 20:51.14 | alephcom | we have a central asterisk pbx and then we have the audiocodes in a couple of remote shops. |
| 20:51.29 | alephcom | Incoming calls from the pstn route just fine. |
| 20:51.40 | MrTelephone | ip phones -> asterisk? |
| 20:51.48 | MrTelephone | inband dtmfmod sucks |
| 20:51.52 | alephcom | Actually the audiocodes are the mp-118 4fxo/4fxs. |
| 20:51.55 | MrTelephone | so i hope your phones support rfc28533 |
| 20:52.26 | MrTelephone | and the phones connected to the audiocodes are not working? |
| 20:52.44 | alephcom | I'm using asterisk. |
| 20:53.03 | Qwell | asterisk to? |
| 20:53.23 | MrTelephone | the phones that don't pass dtmf are connected as clients to asterisk? |
| 20:53.29 | alephcom | Phones connected to the audiocodes work. The problem is that when I dial, say 17802900167 and route the call out through the audiocodes gateway |
| 20:53.44 | Qwell | so where does "ip" fit into this? |
| 20:53.54 | alephcom | I get the "You have dialed a long distance number please enter 1 or 0" |
| 20:54.14 | alephcom | Everything as far as "IP" goes seems to work. |
| 20:54.20 | MrTelephone | check if asterisk is stripping off the first digit |
| 20:54.31 | Qwell | so what IS the problem? |
| 20:54.32 | MrTelephone | is asterisk placing the call? |
| 20:54.36 | alephcom | the audiocodes is apparently attempting to pass the dtmf tones to the pstn but it's not functioning. |
| 20:54.48 | MrTelephone | qwell what are you doing being a loser on the weekend? |
| 20:54.55 | MrTelephone | burnin out on #asterisk.. geez |
| 20:55.21 | MrTelephone | :P |
| 20:55.45 | MrTelephone | alephcom dial with two 1's instead of 1 |
| 20:55.50 | MrTelephone | see if it works |
| 20:55.55 | alephcom | I'll try it. |
| 20:56.22 | MrTelephone | my cable modems digitmap by default wouldn't except 1 in front of a long distance number |
| 20:56.24 | Qwell | ~msg |
| 20:56.24 | jbot | (1) Use private messages to the bots to reduce channel spam, but don't message people on #debian without asking permission first. Most questions should be asked on channel, so that others can benefit from the question and the answers received. (2) Always feel free to message freenode network staff. They're the people with hostnames ending in 'staff.freenode'. ... |
| 20:56.30 | Qwell | bmcghee: ^ |
| 20:57.03 | alephcom | MrTelephone: what's your paypal address? |
| 20:57.23 | alephcom | :-) |
| 20:57.24 | MrTelephone | why |
| 20:57.34 | alephcom | it's working now |
| 20:57.36 | bmcghee | lol |
| 20:57.38 | MrTelephone | lol |
| 20:57.40 | alephcom | tks |
| 20:57.47 | MrTelephone | send me 10 bucks that advice wasn't worth 50 |
| 20:57.59 | MrTelephone | or send me 0 because its christmas |
| 20:58.01 | alephcom | whatever, I offered $50 and you got it working for me. :-) |
| 20:58.10 | MrTelephone | alright then |
| 20:59.00 | ManxPower | bmcghee: Get your sorry excuse for an ass over to #asterisk-bsd |
| 21:00.28 | ManxPower | he just doesn't get it, does he. |
| 21:02.12 | MrTelephone | he didn't do a make install |
| 21:02.22 | MrTelephone | or he installed the modules in two spots and its defaulting to the wrong spot |
| 21:02.27 | MrTelephone | mcGEE |
| 21:02.50 | ManxPower | MrTelephone: he's using BSD and I suspect he has BSD issues, hence me telling him to try the asterisk-bsd mailing list and the #asterisk-bsd channel. |
| 21:03.12 | bmcghee | i did a make install for your information |
| 21:03.33 | ManxPower | but as far I can tell he's still here asking linux people for help on BSD. |
| 21:05.34 | MrTelephone | oh ok |
| 21:05.38 | MrTelephone | i didn't see what you posted |
| 21:05.55 | MrTelephone | bmcghee, fyi :P |
| 21:06.20 | MrTelephone | bmcghee i don't know jack about bsd :( but you should check out the channel manx told you about |
| 21:06.57 | Qwell | you removed modules from menuselect that you already had installed. Clean out your modules dir |
| 21:07.58 | ManxPower | Qwell: I don't think he even ran menuselect, at least he didn't mention it. |
| 21:07.59 | MrTelephone | what is considered a high value for packets/sec on a standard xeon machine |
| 21:08.06 | bmcghee | i fixed it |
| 21:08.08 | bmcghee | its running |
| 21:08.12 | bmcghee | i downgraded to .11 |
| 21:08.15 | Qwell | ManxPower: of course he didn't mention it |
| 21:08.26 | Qwell | but, regardless, I'm right :) |
| 21:08.27 | MrTelephone | bmcghee, remove all your modules and try make install the newer version |
| 21:08.42 | MrTelephone | bmcghee, send qwell 50 bones |
| 21:08.57 | Qwell | my going rate is a bit higher than that |
| 21:09.11 | CBU[^_^]M`` | hello.. anyone here uses SPA3102 as FXO port? |
| 21:10.45 | *** join/#asterisk xeet (i=xeet@37.sub-75-193-125.myvzw.com) |
| 21:12.32 | MrTelephone | qwell what is it |
| 21:12.39 | MrTelephone | maybe I should change occupations |
| 21:13.00 | MrTelephone | do you work for digium? |
| 21:13.03 | ManxPower | MrTelephone: You should not, trust me. |
| 21:13.11 | CBU[^_^]M`` | how i forward calls from Pstn to a voip extension using the SPA 3102? |
| 21:13.38 | ManxPower | CBU[^_^]M``: there are many FAQs on the internet about that. That's how I found out how to do it like 4 years ago |
| 21:13.58 | MrTelephone | why not manxpower |
| 21:14.03 | *** join/#asterisk lotho (n=lotho@217-20-118-126.internetserviceteam.com) |
| 21:14.06 | lotho | hi |
| 21:14.48 | CBU[^_^]M`` | ive tried everything in the forums and none of the seemed to work :( |
| 21:14.57 | Qwell | MrTelephone: If you have to ask - you can't afford it |
| 21:15.20 | bmcghee | ok here a question you can answer |
| 21:15.31 | ManxPower | Qwell: do you also have different rates for "cool" projects .vs. "what moron decided to do it THAT way" |
| 21:15.31 | bmcghee | how i enable it to use mysql database |
| 21:15.48 | xeet | sorry this is unrelated to asterisk -- anyone know where i can find someone with experience configuring an nec pbx? emergency, phone-only support, paid |
| 21:15.54 | Qwell | ManxPower: when I was doing consulting, I did |
| 21:16.06 | ManxPower | bmcghee: Asterisk does not come with support for MySQL because of licensing issues, but you can download support from asterisk-addons |
| 21:16.19 | bmcghee | k thanks manx |
| 21:16.32 | MrTelephone | i make 20 an hour |
| 21:16.50 | MrTelephone | but thats like 10 an hour in the states |
| 21:16.53 | MrTelephone | j/k |
| 21:17.01 | MrTelephone | oops |
| 21:17.07 | MrTelephone | i mean 30 |
| 21:17.09 | MrTelephone | haha |
| 21:17.13 | MrTelephone | just made fun of myself |
| 21:17.15 | RoyKa | does anyone know where I can find the code inhibiting inband dtmf while bridging to function properly? |
| 21:17.44 | MrTelephone | why would inband be inhibited on a bridge? |
| 21:17.49 | MrTelephone | inband is in the rtp flow |
| 21:18.08 | ManxPower | MrTelephone: he wants to make asterisk not process inband dtmf. |
| 21:18.34 | MrTelephone | rtp.c? |
| 21:19.01 | MrTelephone | does asterisk rape rtp traffic on a bridge? |
| 21:19.58 | bmcghee | ManxPower: i get errors when i try to gmake and gmake install. asterisk-addons |
| 21:20.36 | bmcghee | nvm max |
| 21:20.38 | bmcghee | manx |
| 21:20.43 | bmcghee | i will use ports version |
| 21:21.53 | bmcghee | Autoconf version 2.57 or higher is required for this script o.O |
| 21:23.15 | bmcghee | i have 2.61 >.> |
| 21:23.22 | bmcghee | this box needs to be rm -rf * |
| 21:32.56 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.139.134) |
| 21:36.31 | MrTelephone | thats a sweet command |
| 21:37.06 | bmcghee | lol i know |
| 21:37.14 | bmcghee | it gives you a 10 grand into your bank account |
| 21:37.30 | MrTelephone | one time i typed rm -rf * instead of rm -rf *~ |
| 21:37.35 | MrTelephone | man i was swearing that day |
| 21:37.44 | MrTelephone | hhahaa |
| 21:38.11 | bmcghee | haha shiz that sucks |
| 21:38.23 | MrTelephone | i always use the joe editor and it leaves backups with ~ at the end of the filename |
| 21:38.27 | MrTelephone | which comes in handy sometimes |
| 21:38.37 | bmcghee | lol |
| 21:38.40 | bmcghee | except when u gotta delete them |
| 21:38.53 | MrTelephone | yeah |
| 21:38.54 | bmcghee | i officially give up ! |
| 21:39.00 | bmcghee | i cant get asterisk addons to work |
| 21:39.03 | MrTelephone | use linux isntead of bsd |
| 21:39.06 | bmcghee | lol |
| 21:39.16 | bmcghee | if only it was so simple |
| 21:39.45 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 21:40.10 | ManxPower | bmcghee: Unfortunatly BSD is not well supported by Asterisk. Some people try to make it work well, and there is an attempt to support BSD |
| 21:40.44 | MrTelephone | there is a bsd machine next to the pentagon phone access |
| 21:40.46 | MrTelephone | i see now |
| 21:42.57 | bmcghee | how good is asterisk now? |
| 21:43.57 | *** join/#asterisk Jared555 (n=jared555@dialup-4.159.98.35.Dial1.Chicago1.Level3.net) |
| 21:45.37 | ManxPower | bmcghee: there are companies that send millions of mins thru asterisk servers |
| 21:46.39 | bmcghee | i mean the config installer asteriskNOW |
| 21:47.28 | ManxPower | bmcghee: that's not supported here, so I doubt many people have used it here. |
| 21:47.55 | ManxPower | I thought you meant "how good is asterisk currently" |
| 21:48.28 | ManxPower | http://www.digium.com/en/ecosystem/partners/partners.php |
| 21:50.10 | xeet | sorry this is unrelated to asterisk -- anyone know where i can find someone with experience configuring an nec pbx? emergency, phone-only support, paid |
| 21:51.03 | [TK]D-Fender | xeet, call up NEC and ask for a reseller |
| 21:52.31 | xeet | tried...they want a full support contract, and they can't do that in less than 48 hours |
| 21:52.44 | xeet | thats last resort =) |
| 21:53.24 | xeet | immediately following replacing it with * |
| 21:53.31 | [TK]D-Fender | xeet, best of luck to you... |
| 21:53.52 | xeet | thanks =) |
| 21:56.33 | *** join/#asterisk rmayorga_ (n=rmayorga@unaffiliated/rmayorga) |
| 21:57.44 | Jared555 | is it possible to set up asterisk where you can call in with a cell phone or landline, enter your phone number, and then have asterisk forward all calls to that number until you tell it otherwise? |
| 21:57.56 | Jared555 | (without having to keep the call open continuously) |
| 22:06.23 | *** join/#asterisk inv_arp[work] (n=junya@c-76-26-24-197.hsd1.fl.comcast.net) |
| 22:10.09 | bmcghee | i know this you may not be able to answer but anyone got ideas? http://pastebin.com/m32f7da0 |
| 22:10.19 | bmcghee | i can use any ideas to help me get it to install |
| 22:16.14 | Qwell | bmcghee: it helps if you install asterisk first |
| 22:18.16 | bmcghee | asterisk is INSTALLED and RUNNING! |
| 22:19.01 | CBU[^_^]M`` | anyone here knows how to forward PSTN calls from SPA3102 to a local extension? |
| 22:19.58 | docelmo | Say when a call is hung up the h extension is called correct? |
| 22:20.29 | *** join/#asterisk bmg505 (n=leon@196.209.178.145) |
| 22:24.40 | *** join/#asterisk tiav (n=tiav@ram94-3-82-225-11-215.fbx.proxad.net) |
| 22:28.06 | *** part/#asterisk billybongo (n=rich@82.153.23.79) |
| 22:52.40 | RypPn | bmcghee: tried running it from /usr/src ? |
| 22:52.52 | *** join/#asterisk hesco (n=hesco@c-76-17-99-50.hsd1.ga.comcast.net) |
| 22:53.30 | hesco | . |
| 22:55.25 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-99-132.lns10.syd6.internode.on.net) |
| 23:01.38 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 23:04.07 | *** join/#asterisk Cyon (n=cyon@216.179.31.170) |
| 23:05.40 | [TK]D-Fender | Jared555, You can have * do whatever you want to whatever call come into it. |
| 23:07.06 | Jared555 | [TK]D-Fender, there a guide on how to set up an extension that only forwards to a remote number when the person at that number tells asterisk to start sending calls to them? |
| 23:12.43 | [TK]D-Fender | Jared555, No, there isn't |
| 23:13.43 | [TK]D-Fender | Jared555, its YOUR dialplan. Look how your call comes in. At that point do a check to see if it should send this call out somewhere else rather than continue doing what you'd normally do with it. |
| 23:17.10 | [TK]D-Fender | Jared555, in your your normalhandling of incoming calls you'd have to add the option to flag that calls should be forwarded and to whre (hopefully with some sort of password protection). |
| 23:17.54 | Jared555 | k. thanks |
| 23:17.54 | [TK]D-Fender | Jared555, Then you have to have a way to DEACTIVATE this later. I guess that means you shouldn't just place an out-call while "forwarded" |
| 23:22.10 | *** part/#asterisk jmls (n=jmls@62.49.235.130) |
| 23:25.58 | *** join/#asterisk bkw_ (n=brian@adsl-70-142-57-3.dsl.tul2ok.sbcglobal.net) |
| 23:27.58 | jer | on an incoming call, how can i detect someone has hung up? i.e., i just got an 80 minute long voicemail because they hung up and * didn't detect that |
| 23:28.13 | jer | the voicemail, right from the first second was all dialtone |
| 23:28.24 | *** part/#asterisk ZakMcRofl (n=unknown@91.65.148.228) |
| 23:30.26 | Cybertoy | that never happened to me on sip or iax2 channels... |
| 23:30.47 | Cybertoy | * always detected hangups |
| 23:31.19 | [TK]D-Fender | jer, in voicemail.conf you can set a max limit. And your telco needs to be told to enable "call disconnet supervision" on your line. |
| 23:31.38 | [TK]D-Fender | Cybertoy, tahts because your carrier signals this. |
| 23:32.01 | [TK]D-Fender | jer, CDS is aeither a cut or reversal of polairty to indicate hangup. |
| 23:32.07 | jer | [TK]D-Fender, yeah, my max limit is 80 minutes (call recording and all) =] |
| 23:32.51 | [TK]D-Fender | jer, you ALLOW 80 minute VM's? |
| 23:33.18 | *** part/#asterisk JimmyDee (n=James@97-86-161-253.static.stls.mo.charter.com) |
| 23:34.28 | jer | [TK]D-Fender, well, in my previous setup, i had only allowed 3 minutes; and well, 3 minutes wasn't long enough to be in on a meeting, and record the conversation so i didn't have to type madly |
| 23:34.40 | jer | so i upped it to something reasonably long |
| 23:35.06 | [TK]D-Fender | jer, you shouldn't be using VM to record things like that. |
| 23:35.23 | jer | what would you suggest? |
| 23:35.25 | [TK]D-Fender | jer, "show application record" <--------- |
| 23:35.39 | jer | ah |
| 23:39.10 | *** join/#asterisk becks` (n=flux_@185-157.5-85.cust.bluewin.ch) |
| 23:39.40 | becks` | hi.. somebody knows how to run the conformance test cases from ETSI (written in ttcn-3)? |
| 23:42.58 | Siya | anyone here ever used Polycom PVX with Asterisk? |
| 23:57.26 | *** join/#asterisk adker (n=chatzill@74-37-21-237.br1.glv.ny.frontiernet.net) |