| 00:02.14 | scooby2 | i'm just curious if it means I need 8 AddQueueMember's in the login to join them to 8 queues |
| 00:03.42 | ManxPower | rasterix: I would be surpized if it worked with Asterisk |
| 00:03.50 | *** join/#asterisk RB2 (n=RB2@pool-71-255-92-53.nwrknj.east.verizon.net) |
| 00:06.15 | rasterix | i dont even know what it is... it reads like you can instruct the exchange to forward to a different number (providing you dont answer the call) |
| 00:07.09 | rasterix | it could be useful if we are below X free lines we forward calls for a particular ddi |
| 00:07.10 | ManxPower | I've only heard of "customer control of call forward no answer" |
| 00:07.25 | ManxPower | or remote access to call forwarding, that might actually work |
| 00:08.06 | rasterix | basically i want to selectively call forward BEFORE all our channels are full |
| 00:08.43 | ManxPower | rasterix: why not just have a fall forward busy setup to go to an ITSP |
| 00:09.12 | *** join/#asterisk luckyaba (n=lucky@ip70-177-7-204.sb.sd.cox.net) |
| 00:09.43 | rasterix | im probably being stupid... but i dont want to forward calls on some ddis |
| 00:10.20 | ManxPower | rasterix: do the easy part, start sending calls out to an ITSP before the lines are maxed out for OUTGOING calls. |
| 00:11.08 | rasterix | we will have calls coming in on one ddi from a tv campaign and im worried that its going to swamp all our our channels |
| 00:11.41 | ManxPower | rasterix: you had better have a failover plan in place. Do you have a good relationship with your telco? |
| 00:11.49 | rasterix | its bt |
| 00:11.51 | rasterix | so no |
| 00:11.58 | ManxPower | You poor sod. |
| 00:12.03 | rasterix | :( |
| 00:12.58 | ManxPower | rasterix: how many lines do you have with BT right now? |
| 00:13.12 | rasterix | just one isdn30 |
| 00:13.15 | rasterix | 30 channels |
| 00:13.35 | ManxPower | When was the last time you talked with your BT account manager? |
| 00:14.24 | rasterix | when they failed to apply our one plan discounts AND the line went down and the BT emergency call centre failed |
| 00:14.37 | *** part/#asterisk wolfelectronic (n=wolfelec@91.112.227.150) |
| 00:15.27 | ManxPower | Call them, tell them you want to find out about any way to get calls to roll over to a different number if your PRI is full. |
| 00:15.37 | ManxPower | Is your internet connection decent? |
| 00:15.40 | rasterix | yeah |
| 00:15.47 | rasterix | im going to call them |
| 00:16.06 | rasterix | i just wanted to have a clue before i do |
| 00:16.07 | ManxPower | I'd have everything in place to send all outgoing calls out via IP at a moment's notice. |
| 00:16.23 | rasterix | i dont care about outgoing |
| 00:16.45 | rasterix | these ads will hit over a 20 min period |
| 00:16.57 | ManxPower | Outgoing calls can use lines that could be handling incoming calls. |
| 00:17.10 | rasterix | we basically only hanle incoming |
| 00:17.13 | rasterix | handle* |
| 00:17.20 | rasterix | if outgoing goes down |
| 00:17.26 | rasterix | it wont be the end of the world |
| 00:17.42 | rasterix | or even the end of my employment |
| 00:17.54 | ManxPower | still, it would be easy to do and would show you know what you are doing. |
| 00:17.57 | rasterix | i already sent email of doom covering my ass |
| 00:18.26 | ManxPower | I seem to remember BT getting into VoIP. Maybe they have a combo offering. |
| 00:18.38 | rasterix | your right |
| 00:18.44 | rasterix | i should look into that |
| 00:18.50 | ManxPower | I doubt it, but you should ask. |
| 00:18.58 | rasterix | but it wont solve the main problem |
| 00:19.14 | *** join/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 00:19.14 | *** mode/#asterisk [+o mog] by ChanServ |
| 00:19.20 | ManxPower | Telcos tend to have offerings geared for radio station promotions. |
| 00:19.59 | rasterix | i will speak to them tomorrow... maybe bt will suprise me and offer a solution |
| 00:20.04 | rasterix | < doubts it |
| 00:20.07 | ManxPower | rasterix: iAsk them if BT could take overflow calls send them to you via IP with the same destination number. They will look at you funny, and then change the subject. |
| 00:20.31 | rasterix | lol |
| 00:21.01 | rasterix | ill ask them if they do pizza at the same time |
| 00:21.34 | ManxPower | rasterix: competing phone companies might very well offer such a service. Look into them when the crisis is over. |
| 00:22.08 | *** join/#asterisk Defraz (n=T0tal@24-117-156-21.cpe.cableone.net) |
| 00:22.36 | rasterix | at times like this i wish i wasnt a telecoms noob |
| 00:22.59 | rasterix | thanks to asterisk for getting me involved in this shit >< |
| 00:23.32 | rasterix | I used to just laff wen the phone systems screwed up... now its my problem *sigh* |
| 00:23.56 | jaytee | the double-edged sword of convergence :-) |
| 00:24.40 | rasterix | still not to panic i have 4 days to find a solution |
| 00:26.12 | ManxPower | rasterix: I am available for remote (or even local if you want to fly me from the USA) design, consulting, and setup. |
| 00:26.15 | ManxPower | ~manxpower |
| 00:26.15 | jbot | [manxpower] NOT an employee of Digium. He is looking for a training/teaching job in networking and/or Asterisk. Currently doing Asterisk and WAN consulting. Contact: eric@fnords.org |
| 00:27.13 | rasterix | cheaper just to ask u on here :) |
| 00:27.21 | ManxPower | However, I am not very familiar with BT services |
| 00:28.09 | ManxPower | rasterix: I meant something like being on standby during the first ads |
| 00:29.18 | rasterix | does anyone offer an overflow service that just ivrs name and number for call back? |
| 00:30.48 | *** join/#asterisk thansen (n=thansen@c-67-171-115-155.hsd1.ut.comcast.net) |
| 00:31.58 | SwK | anyone around huntsville have an extra single port T1 board they wanna part with |
| 00:34.38 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 00:36.05 | CGMChris | Any new joiners in the room that can help me solve my problem -> Call Queue's do not ring indefinately, but instead go directly to voicemail if even a single agent is offline or otherwise unavailable. |
| 00:36.21 | rene- | well |
| 00:36.35 | rene- | your call to Queue should specify a timeout |
| 00:37.04 | rene- | there are two timeouts you need to be aware of |
| 00:37.05 | zerko | anyone here use dedicated servers in an actual datacenter? |
| 00:37.21 | SwK | zerko, lotta people do |
| 00:37.42 | rene- | time out to offer the call again to other callers and main timeout for the call to be delivered to an agent |
| 00:38.01 | CGMChris | timeout = 15, wrapuptime = 0, strategy = ringall, joinempty = yes, leavewhenempty = no, maxlen = 0 |
| 00:38.05 | rene- | also you have to check your join/leave emtpy sruff |
| 00:38.20 | rene- | What about your Queue command |
| 00:38.31 | rene- | how are u calling it? and are u using Local Channels as agents? |
| 00:38.43 | CGMChris | there is absolutely no delay... it goes immediately to voicemail. Used Gui 2.0 |
| 00:38.53 | rene- | hmm |
| 00:38.54 | rene- | gui |
| 00:39.18 | rene- | dont know about that sorry |
| 00:39.20 | ManxPower | Ah yes, GUI |
| 00:39.22 | CGMChris | if all agents are available, logged in, not on the phone, it works and queues the call. otherwise, fails. |
| 00:39.31 | ManxPower | ~zeeek |
| 00:39.31 | jbot | zeeek is, like, someone who once said "learning asterisk using a GUI is like learning sex through masturbation. You'll never get to the good stuff." |
| 00:39.40 | jaytee | GUI, that sticky stuff in your underwear after a night of "vivid" dreams |
| 00:40.09 | CGMChris | I can edit the config files myself, but up to this point, the gui has been faster. |
| 00:40.35 | ManxPower | There's a secret channel for GUI users. |
| 00:40.37 | rene- | yeah but then you dont understand what is it doing |
| 00:41.15 | CGMChris | I have been reading the book all along and making the corresponding changes using the GUI, where possible. |
| 00:41.21 | CGMChris | exten = 5000,1,Queue(5000) |
| 00:41.26 | CGMChris | my queue name is 5000 |
| 00:41.26 | rene- | ok |
| 00:41.38 | rene- | you are not specifying any timeout |
| 00:41.45 | rene- | for the call in the queue |
| 00:41.53 | rene- | go to your cli and type show application queue |
| 00:41.57 | rene- | read about the timeout option |
| 00:42.06 | CGMChris | k, thanks. |
| 00:42.19 | rene- | sure |
| 00:46.07 | CGMChris | exten = 5000,1,Queue(5000,,,60) <- same thing |
| 00:46.29 | CGMChris | immediately goes to first UNavailable agents voicemail |
| 00:47.42 | *** join/#asterisk xuser (i=jaood@unaffiliated/xuser) |
| 00:48.27 | rene- | hmm |
| 00:48.55 | rene- | is gui generating local channels? |
| 00:48.59 | rene- | for the agents |
| 00:49.21 | rene- | sorry got to go, will be here tomorrow 9 AM central |
| 00:49.28 | rene- | 9 30 hehe |
| 00:49.31 | CGMChris | k, thanks for help. :) |
| 00:50.34 | LARefugee | sip behind nat: do you really have to forward so many ports like they say at nerd vittles? I don't forward any. |
| 00:51.22 | rasterix | cgmchris |
| 00:51.33 | rasterix | set call-limit=1? |
| 00:51.39 | CGMChris | LARefugee: I am using 5060 and 10k thru 20k, but then again, I dont even have incoming calling setup yet :-) but outgoing works. |
| 00:51.45 | rasterix | no ignore that |
| 00:51.51 | rasterix | i need to pay more attention |
| 00:52.22 | CGMChris | rasterix: I appreciate the effort. call-limit won't help me if an agent is offline though. :( |
| 00:53.20 | *** join/#asterisk ftp3 (n=none@pool-96-225-238-78.ptldor.fios.verizon.net) |
| 00:53.25 | rasterix | i dont know the gui |
| 00:53.42 | rasterix | people might find it easier to help you if you paste-bin your .confs |
| 00:53.45 | CGMChris | I can use commands and edit config files... I will do whatever it takes to make it work properly. |
| 00:54.10 | LARefugee | CGMChris: I spent hours getting a vonage softphone account to work the way I wanted. I don't forward one single port. |
| 00:55.14 | CGMChris | LARefugee: I will have to try removing ports and see what happens. It is my understanding that if your equipment is using SPUN and not SIP that it *should* find its way to your device without port forwarding. |
| 00:55.35 | CGMChris | Then again, I just read that today -- so I could be completely wrong. |
| 00:56.28 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-66aff8c74bab303d) |
| 00:58.49 | LARefugee | CGMChris: I do however have a QOS setup on my router running tomato. The only weird thing I have to do is shut down asterisk one minute before my router is auto-rebooted and then restart it (using a script and cron) otherwise the registration is lost and I can't get incoming calls. Only restarting asterisk will work to get back registration. |
| 00:59.36 | rasterix | < out |
| 01:00.18 | LARefugee | I don't know maybe it works for me because I don't usually get more than one call at a time on that softphone account. |
| 01:01.59 | LARefugee | away taking a break from the computer |
| 01:03.53 | Swabby | How do you get phones to auto configure themselves? |
| 01:04.40 | *** part/#asterisk `paul (n=paul@125.252.70.126) |
| 01:05.26 | jaytee | Swabby, press 5#6* and then pray really, really hard. |
| 01:07.09 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 01:07.30 | *** join/#asterisk trnzmeta (n=bleh@secure27.lnk.telstra.net) |
| 01:10.08 | *** join/#asterisk emist_ (n=emist@unaffiliated/emist) |
| 01:15.13 | [TK]D-Fender | Swabby: And don't forget the sacrificial goat. These things always require blood. |
| 01:16.43 | ManxPower | A sacrificial ferret might be a good idea too. |
| 01:17.31 | Swabby | LOL |
| 01:17.43 | Swabby | no seriously...is there auto configure option if you set your ducks right? |
| 01:22.07 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 01:22.07 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0, 1.4.22 (2008/10/02), *-Addons 1.6.0 (2008/10/02), 1.4.7 (2008/06/04), dahdi-linux-complete 2.0.0+2.0.0 (2008/10/02), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-commits #asterisk-bugs #asterisk-dev |
| 01:23.01 | *** join/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net) |
| 01:25.02 | CGMChris | Swabby... The closest you're going to get is: Print up an instruction sheet on how to do it for your users and hand them out. Maybe an instructional video too. Be sure to NOT list any contact information (Name, Address, Phone, Email, etc.) The phones will either "autoconfigure" or they will get a new system. |
| 01:26.49 | *** join/#asterisk uTx (n=unix@modemcable074.229-82-70.mc.videotron.ca) |
| 01:26.50 | Swabby | so like ip address, buttons can't be pushed? |
| 01:27.06 | Swabby | scratch that..i know ips can be pushed using dhcp and mac |
| 01:27.07 | [TK]D-Fender | Swabby: Auto configure what exactly? Do you think * is psychic and can do anything you expect of it? |
| 01:27.21 | Swabby | tk: no i was thinking like buttons and such |
| 01:27.22 | CGMChris | Swabby: The phones should come from the factory with DHCP enabled. |
| 01:27.47 | CGMChris | Swabby: Cisco phones offer a central management interface that can be used to do distributed configurations. |
| 01:27.49 | [TK]D-Fender | Swabby: Increasingly vague |
| 01:28.11 | [TK]D-Fender | Swabby: Perhaps you should actuall say what models you're expecting * to know how to "configure" for you... |
| 01:28.22 | Swabby | TK: like send configuration such as speed dial, "msg" indicator buttons, etc.... I have Grandstream GXP 2000 |
| 01:28.42 | Swabby | and i know they're cheap phones.....maybe i'm expecting alot |
| 01:29.13 | CGMChris | Do you live in an area where illegal mexican labor is accessible? This may be a prime method of autoconfigure. |
| 01:29.21 | Swabby | my biggest problem is this... i built my * network on a SEPERATE network....and i am having challenges getting tne * network talking to the REAL network...i'm assuming i need to do some routing via linux |
| 01:29.27 | Swabby | CGMChris: hahahaha |
| 01:30.09 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 01:30.51 | Swabby | TK: Here's my other problem..i got two analog lines but my dialplan won't let me set rules for BOTH lines..it only allows me to apply the rule to one or the other |
| 01:30.57 | Swabby | i'm assuming i have something in my configuration jacked up |
| 01:31.31 | [TK]D-Fender | Swabby: Indeed you are. It is not *'s job to configure your phones for you, not to configure your mail server, Samba shares, X-Server display settings, *NIX loging accounts, or anything else for that matter |
| 01:31.59 | [TK]D-Fender | Swabby: And you're not showing us anything either. |
| 01:32.25 | [TK]D-Fender | Swabby: You configure *, not the other way around. |
| 01:32.49 | Swabby | i will try to get the server back over here and jump on one day this week.. |
| 01:33.00 | Swabby | its not here right now |
| 01:33.26 | jaytee | damn, I was watching the "debate" and almost missed the sacrificial ferret |
| 01:33.41 | Swabby | me and a friend did this as a learning project for Big Brother Big Sister |
| 01:33.48 | Swabby | but we're afraid we are biting more than we can chew |
| 01:34.00 | Swabby | i'm hoping i can get the system stable and working good so it needs little maintenance |
| 01:34.12 | Swabby | and stuff like vm pws can easily be reset through the gui (it's asterisknow) |
| 01:34.31 | Swabby | i'm also hoping my issues thus far has just been due to my lack of configuration... |
| 01:34.57 | Swabby | TK: If i am having problems defining dial plans for BOTH analong lines what conf file should i concentrate on? |
| 01:34.59 | [TK]D-Fender | Swabby: Indeed not having configured things is a common reason for them not working. |
| 01:36.07 | [TK]D-Fender | Swabby: And you continue to talk about "phones" and "lines" without give precise MAKES & MODELS. Tell you what... You guess what my engine probelm is and we'll see how long it takes you to find out I'm working on a WW1 rotary aircraft engine, ok? |
| 01:36.30 | Swabby | TK: You have a good point |
| 01:36.41 | Swabby | can i run down my architecture with you? |
| 01:36.48 | [TK]D-Fender | Swabby: I have many, those were freebies, the rest arre $4.95/min |
| 01:37.01 | [TK]D-Fender | Swabby: I'm sure it can be done in 2 senteces or less. |
| 01:37.07 | Swabby | obviously we're not fixing anything tonight--i'm not in front of the system but maybe you cna give me some inkling tips |
| 01:37.09 | Swabby | okay..here goes |
| 01:37.26 | riddlebox | anyone else having problems with fxs ports on a tdm after upgrading to 1.2.22.1? Mine stopped working all together.. |
| 01:38.07 | Swabby | I have a box with AsteriskNOW instaled ( i used the express option) I have a TDM 400, three analog lines coming in (we have two coming into the phone) only three ports are live on the card the fourth one does not have a module installed on it) |
| 01:38.43 | Swabby | I have two ANALOG lines plugged into ports 2 and 4. MY goal right now is to use my Grandstream GXP 2000 phones to make/recieve calls on the analog lines and ALL of them to ring on incoming calls |
| 01:39.04 | Swabby | I am NOT using any VoIP providers at this time |
| 01:39.12 | [TK]D-Fender | riddlebox: that's old, even for 1.2 |
| 01:40.44 | [TK]D-Fender | riddlebox: and what package ver is that eeven in refernce to? Zaptel? Asterisk? What combination are you running now? do you have ANY details for us? |
| 01:40.46 | Swabby | on the TDM400 there's four ports. Three of them have FXO modules |
| 01:41.15 | jaytee | WW1 rotary? hmmm, Sopwith Camel? |
| 01:42.54 | [TK]D-Fender | jaytee: Nope, Fokker |
| 01:43.21 | [TK]D-Fender | Swabby: (we have two coming into the phone) <--- HUH?! |
| 01:43.48 | Swabby | TK: sorry...we have two of the three analog lines terminating on the TDM 400 card to provide outbound/inbound to the * system |
| 01:44.25 | [TK]D-Fender | Swabby: Ok, 3 FXO on a TDM400 + Some GXP-2000's. As much as need have been said |
| 01:44.57 | Swabby | Si |
| 01:45.00 | Swabby | no softphones right now |
| 01:45.16 | Swabby | my switch doesn't support DHCP and i don't have dhcp installed..so i hard coded ips on all the phones |
| 01:45.27 | riddlebox | crap [TK]D-Fender i meant 1.4.22.1 |
| 01:45.38 | [TK]D-Fender | Swabby: What was this talk about GXP's then? |
| 01:45.38 | Swabby | server = 192.168.1.2* phones are 192.168.0.25, 192.168.1.225, 192.168.1.226, etc. |
| 01:45.51 | Swabby | TK: GXPs i consider hard phones |
| 01:45.55 | [TK]D-Fender | Swabby: And I don't know of any switches that support DHCP.... thats a neat trick... |
| 01:46.18 | Swabby | TK: I think it's a cisco switch that is suppose to be bundled with an enterprise cisco system.... |
| 01:46.34 | Swabby | TK: Thats' why it doesn't support dhcp..we got it donated through techsoup... |
| 01:46.44 | Swabby | maybe i should install dhcp in linux.. |
| 01:46.58 | [TK]D-Fender | Swabby:Holy crap a 20$ router will give you DHCP. I'm sure I could hack my ANALOG WATCH to do it if I tried jsut a little |
| 01:47.05 | riddlebox | [TK]D-Fender, man I dont even know what the heck I was typing, it was 1.4.22 |
| 01:47.14 | Swabby | TK: lol |
| 01:47.29 | [TK]D-Fender | riddlebox: Try again when all the neurons are firing in the proper order & direction :) |
| 01:47.47 | Swabby | aside from my ghetto networking |
| 01:47.55 | [TK]D-Fender | Swabby: And yes, your * box might as well be a DHCP server for the 2 minutes that should take |
| 01:48.03 | riddlebox | [TK]D-Fender, I know, I am a little crazy these days we leave on friday to go to the Dominican Republic to get married |
| 01:48.21 | Swabby | TK: the biggest isuse there has been getting the actual files over to the server....ususally i use yum to install stuff like that..but that box has NO internet. |
| 01:48.46 | Swabby | in result of my once again ghetto network |
| 01:48.49 | riddlebox | [TK]D-Fender, but really I installed 1.4.22 and lost my fxs ports on the tdm card, the fxo still worked though |
| 01:49.11 | Swabby | i guess if i install dhcp i would have the correct time on my phones too right? |
| 01:49.21 | [TK]D-Fender | Swabby: Next time install a real distro that has all this stuff handy at the start. |
| 01:49.33 | Swabby | TK: i am thinking about doing a hand install |
| 01:49.42 | Swabby | installing fedora and then putting asterisk on top |
| 01:49.45 | [TK]D-Fender | Swabby: DHCP doesn't hand out time, it can only point to a time server |
| 01:49.47 | jaytee | CentOS FTW!!! |
| 01:50.16 | Swabby | TK: Here's my biggest issues though..tell me your thoughts...maybe i should just wipe everything |
| 01:50.30 | Swabby | if i go into the dial plans it won't let me select more than just one provider... |
| 01:50.36 | [TK]D-Fender | Swabby: So to clarify, you don't HAVE hard phones, only softphones and were CONSIDERING GXP's? |
| 01:50.40 | Swabby | but the book shows you cna select all of them |
| 01:50.47 | Swabby | tk: no, no..i only have GXP |
| 01:51.08 | [TK]D-Fender | Swabby: What dialplan's? What are you talking about? a PHONE'S internal dialplan? *'s dialplan? |
| 01:51.11 | Swabby | all the gxps have ips in the 192.168.1.* range... |
| 01:51.20 | Swabby | TK: * dial plan.. |
| 01:51.36 | Swabby | in asterisknow there's a gui that shows which line to go outbound on |
| 01:52.03 | Swabby | and what to do with the #'s..like drop the 9, etc |
| 01:52.04 | [TK]D-Fender | Swabby: *'s dialplan coun route your calls any which way you want. use line 2 if its tuesdays night, its raining, and the Cubs won their last hom game by a margin of 3 runs or more. |
| 01:52.31 | Swabby | TK: problem is..it won't let me define both lines. |
| 01:52.55 | Swabby | like if i have a rule for local..it won't let me allow it on both line 2 and 3 |
| 01:53.24 | Swabby | TK: do you recommend i just wipe this bull hunkie clean, install a linux os and then install my asterisk? |
| 01:53.38 | [TK]D-Fender | Swabby: I am going to use some really harsh language now, I just thought I wanr you. Its not directed at you but at a piece used in your approach so don't take this personally, but more as a warning on what you should probably be doing in your attempts to learn and configure your system. Ok? |
| 01:53.47 | [TK]D-Fender | I'd warn* |
| 01:53.55 | Swabby | other question..can i get asterisk to work without a tdm card (For testing if i want to play with this in vmware) |
| 01:53.59 | Swabby | break it down :) |
| 01:54.07 | Swabby | hit me with it |
| 01:54.12 | [TK]D-Fender | Swabby: FUCK THE FUCKING GUI |
| 01:54.22 | Swabby | i like it |
| 01:54.30 | [TK]D-Fender | </breathout> |
| 01:54.31 | Swabby | not the gui |
| 01:54.36 | Swabby | your idea |
| 01:54.40 | [TK]D-Fender | There, much better |
| 01:54.48 | Swabby | now..let me ask though |
| 01:54.53 | Swabby | if i do a manual install.. |
| 01:54.59 | Swabby | is there ANY gui that allows people to reset vm pw and stuff |
| 01:55.03 | Swabby | or is it all cli? |
| 01:55.08 | [TK]D-Fender | Swabby: Asterisk-GUI is a work in progress, and a HALF-complete attempt at best that forces you to to quite a bit yourself |
| 01:55.37 | [TK]D-Fender | Swabby: It is a retard toaster PBX generator (ATTEMPT), and its concept of "dialplan" will get you LOST> |
| 01:55.47 | riddlebox | [TK]D-Fender, have you tried the 2.0 version? |
| 01:55.52 | [TK]D-Fender | Swabby: This is a massive impediment to your learning anything |
| 01:56.08 | Swabby | tk: so if you were me...you would build a linux box, get my routing in place, and then install asterisk? |
| 01:56.18 | [TK]D-Fender | Swabby: Yes. |
| 01:56.21 | Swabby | ok |
| 01:56.27 | *** join/#asterisk MrNaz (n=naz@124-168-123-152.dyn.iinet.net.au) |
| 01:56.35 | Swabby | and i've done linux before...i've configured apache, mysql, etc |
| 01:56.47 | Swabby | so i feel fairly comfortable i could compile it and install it |
| 01:56.53 | Swabby | i tried freepbx before and it was jacking stuff up too |
| 01:57.06 | Swabby | with the gui it's hard to determine what's going on at times because all the guts are behind the scenes |
| 01:57.13 | Swabby | and your not responsible for configuring them! |
| 01:57.19 | [TK]D-Fender | Swabby: Its idea of "lines" and "routes", and all that other crap is like a holdover from other toaster config generators and won't be "flexible" becasuee they really CAN'T. Its a GUI. It has to work in a limited number of ways off stupid web config |
| 01:57.56 | Swabby | TK: will it autoconfig my TDM Card? or will i have to build a config? |
| 01:57.56 | jaytee | would someone please bind my hands and feet and then ask me to tapdance while juggling eggs? |
| 01:57.58 | [TK]D-Fender | Swabby: And with *NOW you are manually configuring a bit of it and the GUI is fucking with you the moment you turn your back |
| 01:58.12 | [TK]D-Fender | Swabby: Configuring a TDM card is 5 minutes work <- |
| 01:58.15 | jaytee | it likes to do that |
| 01:58.25 | Swabby | and it HAS been fucking with me...litterally..i had stuff working well.. |
| 01:58.29 | Swabby | and one reboot all the phones are unavailable |
| 01:58.32 | Swabby | crazy shit..ya know? |
| 01:58.39 | jaytee | poof! it's like magic! |
| 01:58.50 | Swabby | okay...tk..can i ask a quick networking question? |
| 01:59.17 | [TK]D-Fender | Swabby: Shoot |
| 01:59.22 | Swabby | SO i got this * network 192.168.2* and my REAL network (with PCs and the internet) are 192.168.2* so i'm assuming i need to change the ip range of the asterisk network if i EVER want this shit to talk to each other |
| 01:59.43 | Swabby | to like 192.168.3* |
| 02:00.00 | Swabby | and then use a how to for linux to setup iptables to forward my traffic accordingly between nic 1 and nic 2... ? |
| 02:00.07 | [TK]D-Fender | Swabby: Well if you want 2 separate subnets yeah clearly they'd have to be different |
| 02:00.25 | Swabby | Do you recommend two seperate subnets for QoS? |
| 02:00.29 | [TK]D-Fender | Swabby: Nothing to do in iptables to forward between the two. |
| 02:00.42 | Swabby | since my switches are not smart... |
| 02:00.51 | [TK]D-Fender | Swabby: We're talking phones on a switch. It really doesn't matter much. |
| 02:00.58 | Swabby | TK: Where would i perform the forwarding? like the cmd name so i can search out a help |
| 02:01.14 | jaytee | this debate is so lame and boring, they keep asking both of them questions about the economy, healthcare and crap like that. I wanna hear questions like, "Senator McCain, while you were a POW in North Vietnam did you ever have sex with another man?" and "Senator Obama, if you're elected President will you push to legalize marijuana and prostitution?" |
| 02:01.28 | [TK]D-Fender | Swabby: "echo 1 > /proc/sys/net/ipv4/ip_forward". The End. |
| 02:01.49 | Swabby | my main reasons for getting the networks merged are this 1. configuration (Right now i have a PC next to the box just so i can get to the web gui!) 2> SIP Clients if i ever want this and 3> Updates... oh..4> voip providers if i ever go there |
| 02:02.10 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
| 02:02.31 | Swabby | ipforward..got it |
| 02:02.32 | [TK]D-Fender | Swabby: Now of course your default route should be the * box so that it would even get traffic for the other subnet, or listed as a router by another router, etc |
| 02:02.49 | Swabby | tk: makes sense |
| 02:03.06 | Swabby | i guess #1 don't make a difference since all my config would be done in the shell... |
| 02:03.19 | [TK]D-Fender | Swabby: You should have a proper grasp of networking to be using VoIP in any capacity with * |
| 02:03.54 | [TK]D-Fender | Swabby: And you're talking like You can't run X on your server and config it FROM ther server itself. |
| 02:04.16 | [TK]D-Fender | Swabby: But as far as that goes, see my first harsh comment. |
| 02:04.42 | Swabby | well..if i do what we been discusing i'm not worried about X windows..i wouldn't even install it |
| 02:04.54 | Swabby | What distro would you recommend? |
| 02:05.05 | Swabby | does fedora play well with it? |
| 02:05.10 | voxter | fedora? |
| 02:05.11 | [TK]D-Fender | CentOS |
| 02:05.20 | Swabby | CentOS.. k |
| 02:05.23 | voxter | fedora is like the experimental bleeding edge 'about to be rhel' |
| 02:05.30 | voxter | why would anyone run fedora on any server |
| 02:05.50 | Swabby | voxter: i thought it was still fairly popular..i must be behind :) |
| 02:05.56 | voxter | oh its popular |
| 02:06.10 | voxter | its just asinine to trust a server to it, imho |
| 02:06.14 | Swabby | now..one last major question.. |
| 02:06.35 | Swabby | i install centos, get my updates, dl asterisk get everything compiled, config my cards, etc....what do you recommend i follow to make sure i cover all my basis? |
| 02:06.44 | Swabby | i'm assuming i will setup my lines, dialplans, my users, etc... |
| 02:06.55 | Swabby | are there sample config files? |
| 02:07.23 | LARefugee | Swabby: sample config files yes! The best. I refer to them all the time. |
| 02:07.47 | [TK]D-Fender | Swabby: Use a computer with internet access and install and learn from there. all you need is a box to drop * on and a soft-phone. |
| 02:08.48 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 02:08.56 | v4mp | my provider use fedora on most there servers i cant stand it so i use centos |
| 02:09.16 | v4mp | but as im here i have this problem still... i have it setup so the agent has to login which works fine but when theres an incoming call its trying to find the agent thats logged in but its not ringing the agents phone so the agent cant aswer any idea what this could be ? |
| 02:11.32 | Swabby | tk: can i install it in vmware with a softphone? |
| 02:11.39 | *** join/#asterisk foamball (n=eisert@user-0c6sp2h.cable.mindspring.com) |
| 02:11.54 | jaytee | wow, with such complete and detailed information about your problem it's easy to see that your magic.conf file is missing the wanker=yes statement. |
| 02:12.28 | [TK]D-Fender | SwabbySwabbyWho needs VMWARE? |
| 02:12.36 | Swabby | jaytee: i actually just added wanker=yes to mine, recompiled and it works great |
| 02:12.46 | Swabby | TK: if i want to test on my windows box... |
| 02:12.48 | v4mp | o_O |
| 02:13.00 | Swabby | tk: btw...i know i've been a minor pain in the ass...i appreciate your help |
| 02:13.40 | jaytee | minor pain in the ass? hell, I'm on my second tube of Prep H and I was just listening |
| 02:13.50 | v4mp | jaytee, if that was to me i have checked over with many sites and seems that my setup is correct |
| 02:14.10 | Swabby | jay: LOL |
| 02:14.38 | jaytee | v4mp, is the agent's phone registered? is the agent logged in? |
| 02:14.45 | v4mp | yes |
| 02:14.54 | Swabby | obama says pakistan weird |
| 02:14.56 | [TK]D-Fender | Swabby: Just Install a softphone on SOME friggen system. It ain't Raw-Cat Science. |
| 02:15.24 | jaytee | v4mp, how about a pastebin of queues.conf and agents.conf? |
| 02:15.30 | v4mp | ok 1 sec |
| 02:15.40 | Swabby | raw cat hah |
| 02:18.46 | Swabby | will centos+asterisk work ok under vmware? |
| 02:19.18 | *** join/#asterisk brimstone (n=brimston@pdpc/sponsor/digium/brimstone) |
| 02:19.32 | [TK]D-Fender | Swabby: ok/fine/sure/whatever |
| 02:19.44 | [TK]D-Fender | Swabby: Oh... if you forget about the TDM card |
| 02:19.45 | Swabby | just for testing |
| 02:20.02 | Swabby | if i wanted to just play with stuff.... |
| 02:20.03 | [TK]D-Fender | Swabby: Fine if you talking about a basic machine to setup * on ASIDE fro hardware |
| 02:20.10 | brimstone | anyone seen gcc fail like this before? http://pastebin.ca/1222366 |
| 02:20.10 | Swabby | yeah...i'm not going live or anything |
| 02:20.23 | [TK]D-Fender | Swabby: It'll do |
| 02:20.41 | Swabby | you recommend a dedicated box though huh |
| 02:20.46 | *** join/#asterisk techman97 (n=myweiner@97-91-103-181.dhcp.roch.mn.charter.com) |
| 02:20.54 | Swabby | could i still test like call routes and stuff? |
| 02:21.00 | *** join/#asterisk voxter (n=voxter@S01060016b6b53c0c.vc.shawcable.net) |
| 02:21.06 | Qwell | brimstone: holy crap |
| 02:22.01 | Qwell | brimstone: what did you do? O.o |
| 02:22.19 | brimstone | Qwell: i didn't mean to |
| 02:22.23 | Qwell | ! |
| 02:22.26 | *** join/#asterisk hi365_m (n=hi365@213.151.52.225) |
| 02:22.36 | brimstone | i just disabled a bunch of stuff and told it to compiler |
| 02:22.41 | Qwell | it says it's a hardware/OS issue.. |
| 02:22.43 | brimstone | i just need this box for sip/iax meetmes |
| 02:22.45 | Qwell | I'd be inclined to believe it |
| 02:22.50 | brimstone | figures, pos box |
| 02:23.37 | brimstone | so... ug |
| 02:23.44 | [TK]D-Fender | Swabby: all that terminology goes right out the door. |
| 02:24.03 | [TK]D-Fender | Swabby: The only difference between a test & the real thing is a DIAL statemtn |
| 02:24.04 | Swabby | doesn't make any sense under regular asterisk then |
| 02:24.07 | [TK]D-Fender | statement* |
| 02:24.28 | Swabby | got it |
| 02:25.06 | [TK]D-Fender | Swabby: From now on, every call is just a CALL. *gets a call, it goes through the dialplan having matched the exten that was dialed. You do "whatever" with it. |
| 02:25.13 | [TK]D-Fender | Swabby: "route" = BS term. |
| 02:25.21 | Swabby | got it |
| 02:25.38 | Swabby | can i simulate calls coming in going out on a adhoc system with no nodes? |
| 02:25.44 | Swabby | i mean no cards or anything |
| 02:25.57 | Swabby | i guess with multiple softphones i could huh.. |
| 02:26.04 | [TK]D-Fender | Swabby: Every call is jsut a call. |
| 02:26.44 | brimstone | Qwell: if i run make install again, it works fine, holy cow |
| 02:26.48 | v4mp | my queues.conf and angent.conf can be found here... http://v4mpire.pastebin.com/d40173317 |
| 02:26.57 | Qwell | memtest? |
| 02:27.01 | [TK]D-Fender | Swabby: Makes no difference what you do with them. You can set it all up to the point where you'd want to connect your caller to another device and just playback a sound file saying "would have called bob" or do SayDigits(1234567) or whatever,. |
| 02:27.11 | Swabby | got it... |
| 02:27.22 | Swabby | i could even say if dialed 911 play "your fucked" |
| 02:27.24 | Swabby | literally... |
| 02:27.25 | brimstone | Qwell: i dunno, just gonna chalk it up as a win |
| 02:27.30 | Swabby | if i had my configuration right |
| 02:27.59 | [TK]D-Fender | zSwabbyRemember what I said about the Cubs earlier.... |
| 02:28.11 | [TK]D-Fender | Swabby: just about anything. |
| 02:28.14 | Swabby | does regular asterisk allow you to record messages using the phones ..like ivr prompts or do you have to record the wav files on your own |
| 02:28.29 | [TK]D-Fender | Swabby: "core show application record" |
| 02:28.40 | Qwell | Swabby: tip for the future... |
| 02:28.49 | [TK]D-Fender | Swabby: Go sit down with the book. Go read ALL of *'s dialplan applications and go do something |
| 02:28.58 | Qwell | if the question begins "Does Asterisk allow you to ...", the answer is almost always "yes" |
| 02:29.22 | Swabby | Qwell: Lol..it seems like a pretty dynamic system |
| 02:29.23 | jaytee | v4mp, what do you get at the CLI when you type agent show? |
| 02:29.24 | Qwell | somebody feel free to prove otherwise |
| 02:29.27 | Swabby | i was reading the case studies... |
| 02:29.44 | Swabby | there was a guy in the book talking about how he gave his kids a math quiz on it |
| 02:29.46 | Swabby | thats really cool stuff |
| 02:29.48 | v4mp | 1003 (Gary Syms) available at '1@incoming_calls' (musiconhold is 'default') |
| 02:29.48 | v4mp | 1 agents configured [1 online , 0 offline] |
| 02:30.36 | *** join/#asterisk MrNaz (n=naz@210-84-39-63.dyn.iinet.net.au) |
| 02:31.04 | Swabby | i mean i guess you could literally QUERY a database and play it on the phone line huh |
| 02:31.04 | Swabby | heh |
| 02:31.29 | jaytee | v4mp, and what do you get on the CLI when you try to call the queue? |
| 02:31.42 | brimstone | Qwell: ttfn |
| 02:31.44 | *** part/#asterisk brimstone (n=brimston@pdpc/sponsor/digium/brimstone) |
| 02:32.33 | [TK]D-Fender | Swabby: My * used to make me COFFEE. How hard do you think a quiz would be? |
| 02:33.22 | jaytee | mine used to make me coffee but then it tried to force me to switch to decaf so I got out my Qparted CD and wiped the bitch and started over. |
| 02:33.24 | [TK]D-Fender | Swabby: Phone in to disable your kid's internet access. |
| 02:33.37 | [TK]D-Fender | jaytee: For the best really.... damn HAL |
| 02:34.05 | jaytee | never get in the way of a tech guy and his caffiene |
| 02:34.11 | v4mp | http://v4mpire.pastebin.com/d5b30e5ec |
| 02:34.30 | v4mp | its also not saying the wait time etc to caller either |
| 02:35.16 | jaytee | I saw one guy's blog about setting up MythTV with Asterisk and he could dial into his server and key in codes that would kick off recordings of whatever channel he wanted to record. |
| 02:36.05 | phix | jaytee: awesome |
| 02:36.07 | [TK]D-Fender | jaytee: Yeah I think I heard about that. I had a dial-in diagnostic script once. Would let you ping, etc. |
| 02:36.22 | phix | jaytee: I just SSH from my phone to my server to do that :P |
| 02:37.37 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 02:38.11 | jaytee | v4mp, I'm not the brightest one in here but what doesn't make sense about your config is that you've got Gary Syms logged in but the calls go to a Local channel and not a SIP phone that Gary's logged in from. |
| 02:38.50 | v4mp | oh ? o_O |
| 02:39.01 | v4mp | mm |
| 02:39.05 | v4mp | *mm |
| 02:39.07 | v4mp | bah lol |
| 02:39.08 | Swabby | what modules do i use to make coffee with it? |
| 02:39.13 | jaytee | I'd ask to see your extensions.conf but now that Chris Matthews is doing the post-debate analysis I'm gonna have to go vomit. |
| 02:39.36 | [TK]D-Fender | Swabby: X-10 |
| 02:39.43 | jaytee | Swabby, res_coffee.so, app_brew.so and app_latte.so |
| 02:39.56 | Swabby | jaytee: lol... :) |
| 02:40.00 | jaytee | but be warned, app_latte.so is very unstable |
| 02:40.13 | Swabby | tk: no shit though..did you really get x10 to make coffee based on asterisk? |
| 02:40.44 | jaytee | [TK]D-Fender, seriously though, * with X-10 just makes the mind reel with the possible applications don't it :-) |
| 02:41.14 | ManxPower | It does? |
| 02:41.18 | [TK]D-Fender | Swabby: Yup. Dial-a-coffee |
| 02:41.29 | v4mp | jaytee, the agent logs in via internal extension |
| 02:41.38 | Swabby | TK: LOL |
| 02:41.39 | ManxPower | Oh! Sorry. I mean "Oh yes infinite things!" |
| 02:42.03 | ManxPower | [TK]D-Fender: dial a maid to put the grounds and water in the coffee maker before calling into asterisk to start the coffee maker. |
| 02:42.36 | v4mp | haha |
| 02:42.39 | [TK]D-Fender | ManxPower: if I had a maid... she wouldn't be making me coffee ;) |
| 02:42.41 | drmessano | http://www.beigerecords.com/cory/pizza_party/ |
| 02:42.46 | drmessano | app_pizza FTW |
| 02:43.22 | jaytee | ManxPower, well, you could use it to turn on or off your house lighting while away, kickoff an automated feeder for your pet, probably even tie it in with the HVAC system in the house. |
| 02:43.23 | [TK]D-Fender | drmessano: Wait... ordering a pizza with a PHONE? Have you insaned?! |
| 02:44.29 | drmessano | You still use Asterisk for phone calls? |
| 02:44.37 | drmessano | I did away with those months ago |
| 02:44.43 | [TK]D-Fender | You still use pizza for food? |
| 02:45.04 | drmessano | There's an alternative? |
| 02:45.06 | [TK]D-Fender | still needs one more box to get his monitor to the perfect height |
| 02:45.06 | jaytee | "Pizza In A Cup" |
| 02:45.28 | jaytee | "It's these cans!!!! He hates these cans!!!!" |
| 02:45.28 | [TK]D-Fender | jaytee: 2 pepperoni, 1 c.... er.... nevermind |
| 02:45.30 | ManxPower | jaytee: that sort of stuff is for people with lives so boring they have to worry about such things. |
| 02:45.55 | jaytee | ManxPower, so...... you've been talking to one of my 3 friends? |
| 02:46.03 | [TK]D-Fender | ManxPower: Boring people don't have worries. |
| 02:46.03 | drmessano | System(pizza -p -m 1 medium thin) |
| 02:46.14 | [TK]D-Fender | ManxPower: Boring is when you have nothing left to do. |
| 02:46.35 | ManxPower | jaytee: I admit it might be nice to be able to turn on the heat or a/c an hour before I arrive home. |
| 02:46.41 | [TK]D-Fender | ManxPower: Which is a sad semblance of the state I'm in right now. |
| 02:46.41 | jaytee | The Lumenvox Voice Recognition software for * comes with a sample application that is a Pizza ordering system |
| 02:47.10 | [TK]D-Fender | jaytee: Wow.. now its just like ordering pizza from a real human being only its wrong 26% of the time! |
| 02:47.12 | jaytee | ManxPower, yeah but they have programmable thermostats that make that much easier, less time consuming for setup and probably less costly. |
| 02:47.13 | drmessano | Using asterisk to run an application that orders pizza via CLI is the ultimate "Suck it, word-speakers", IMHO |
| 02:47.18 | ManxPower | Me: I'd like pepperoni and green olives Pizza VR: You have ordered anchovies and saussage! |
| 02:48.05 | jaytee | [TK]D-Fender, the 26% failure rate is due to bad programming of the grammars and error correction. I've had very good results with Lumenvox actually. |
| 02:48.31 | [TK]D-Fender | jaytee: I've have better results from pizza delivery call takers :) |
| 02:48.38 | jaytee | me too |
| 02:48.41 | ManxPower | Voice recog is nice until you try calling in from a noisy environment... like trying to figure out why your credit card was declined while you are in the airport. |
| 02:48.43 | drmessano | If I used * to run apps that replaced people paid to answer the phone to handle such trivial tasks, I would feel like I was using their own tools against them, and stickin-it-to-the-man |
| 02:49.13 | drmessano | Digg users would make me their +22 of the hour |
| 02:49.39 | jaytee | ManxPower, yeah background noise is a problem with most VR systems. You can adjust for that or if just have a parallel system mode that's DTMF only. |
| 02:49.39 | ManxPower | quickly patents the IVR Score, a measurement of how much better or worse the IVR is as compared to someone in India earning $5/day |
| 02:50.00 | drmessano | "Wut, I R front-pageon of the hour? I WUN THE INTARWEBS RIGHT NOW FOR A MINUTE!!!!11!!" |
| 02:50.35 | v4mp | jaytee, would you like to see my extensions.conf ? |
| 02:50.48 | drmessano | v4mp: I dont think anyone would |
| 02:50.51 | jaytee | v4mp, um......no, not really |
| 02:51.01 | ManxPower | v4mp: Egads man! Cover that thing up! |
| 02:51.17 | v4mp | o_O |
| 02:51.30 | drmessano | v4mp: You're fired. Dont bother turning in your keys, the locks have already been changed. Security is on it's way. |
| 02:51.47 | v4mp | haha |
| 02:52.30 | v4mp | would the Local for where its mean to be sending the calls to be anything to do with the Agent login ? |
| 02:52.57 | jaytee | Local channels are not phones |
| 02:53.06 | *** join/#asterisk sucituanbo (n=free@c-24-21-121-148.hsd1.wa.comcast.net) |
| 02:53.16 | v4mp | but theres nothing refering to local in extensions |
| 02:53.45 | [TK]D-Fender | v4mp: that CLI snippet didn't show any agent status, dialout atempts, or much at all |
| 02:53.53 | ManxPower | v4mp: Local/ means "dial this extension in this context just as though it was a call, but don't require any SIP or zap or anything like that. |
| 02:53.59 | ManxPower | think of it as a Loopback channel |
| 02:54.31 | v4mp | the agent logs in through internal and the call comes in through incoming_calls |
| 02:55.14 | v4mp | tk what kind of dialout attempts where you expecting ? |
| 02:56.27 | [TK]D-Fender | v4mp: Show us your agent status, queue status, etc. |
| 02:56.33 | *** join/#asterisk setkeh_ (n=setkeh@CPE-124-180-146-148.vic.bigpond.net.au) |
| 02:58.05 | v4mp | i dont understand what your after |
| 02:58.06 | setkeh_ | hey guys what are these values suposed to look like amaflags = AstAccountAMAFlags |
| 03:00.49 | ManxPower | setkeh_: it is mainly cosmetic for all but the largest organizations |
| 03:01.31 | setkeh_ | max_______, so i dont have to touch it ?? |
| 03:01.52 | setkeh_ | ManxPower, sorry typo |
| 03:02.21 | *** join/#asterisk cesal (n=jcesarlp@200.106.8.189) |
| 03:02.36 | cesal | hardwire hi |
| 03:02.41 | cesal | can u help me |
| 03:02.48 | cesal | i have install centos |
| 03:02.51 | cesal | 5.2 |
| 03:02.58 | cesal | now i want to install astersk |
| 03:03.03 | cesal | can u help me? |
| 03:03.07 | ElCheapo | download source |
| 03:03.12 | ElCheapo | ./configure && make && make install |
| 03:03.17 | [TK]D-Fender | cesal: Instructions are in the source tarball. |
| 03:03.28 | cesal | where? |
| 03:03.44 | [TK]D-Fender | cesal: There are all sorts of great instructions jsut sitting there waiting to be read by you |
| 03:03.57 | cesal | please can u give a link? |
| 03:04.03 | [TK]D-Fender | cesal: Instructions are in the source tarball. <- what part of this is not as obvious as it sounds? |
| 03:04.20 | cesal | i am new to linux also |
| 03:04.26 | cesal | that why i ask |
| 03:04.29 | [TK]D-Fender | TARBALL. as in ".tar.gz" As in www.asterisk.org and go download the SOURCE |
| 03:04.40 | [TK]D-Fender | cesal: And I guess while you're at it : |
| 03:04.43 | [TK]D-Fender | ~book |
| 03:04.44 | jbot | somebody said book was 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 --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 03:04.45 | [TK]D-Fender | ^^^ |
| 03:04.54 | LARefugee | cesal: you have a lot to learn.. Get some help from a friend familiar with Linux. |
| 03:05.16 | ManxPower | What do people dislike about CentOS anyway? |
| 03:05.17 | LARefugee | ~alsa |
| 03:05.18 | jbot | somebody said alsa was the Advanced Linux Sound Architecture, or at http://www.alsa-project.org/, or available only for Linux (if you intend your code to be portable, ask me about OSS) |
| 03:05.31 | LARefugee | ~oss |
| 03:05.31 | jbot | oss is probably a portable sound interface available in 11 different Un*x systems. The propietary package is available at http://www.opensound.com/. However, free systems like GNU/Linux or *BSD include free GPL/BSD re-implementations. The support in Linux is being gradually removed in favor of ALSA. an acronym that stands for "Open Source Software" |
| 03:05.51 | LiNeTuX_Home | ManxPower: its repos aren't as complete as Debian's. |
| 03:05.57 | drmessano | ROFL |
| 03:06.10 | drmessano | propietary ? Is that like propecia? |
| 03:06.12 | ManxPower | is that all? |
| 03:06.15 | v4mp | tk, unless u meant agent show and queue show ? |
| 03:06.22 | phix | hmmm, this cant be good --> /dev/main-new/srv: Updating bad block inode. |
| 03:06.33 | phix | /dev/main-new/srv: Duplicate or bad block in use! |
| 03:06.41 | phix | /dev/main-new/srv: Multiply-claimed block(s) in inode 9355355: 37427455 |
| 03:06.44 | ManxPower | phix: It sucks to be you. |
| 03:06.48 | LiNeTuX_Home | ManxPower: that's my major gripe. But I use a lot of distros. BSD, Linux... whatever works. |
| 03:07.04 | drmessano | Yeah, it sucks to be us right now too, with all the paste-spam |
| 03:07.05 | phix | ManxPower: they are brand new drives in RAID1 |
| 03:07.11 | ManxPower | LiNeTuX_Home: I'm looking at switching distros. |
| 03:07.20 | LiNeTuX_Home | ManxPower: From? |
| 03:07.26 | phix | 2nd time I have put one through waranty |
| 03:07.33 | ManxPower | phix: I had a new drive fail in 4 weeks and the replacement failed in 2 weeks. Not heat related. |
| 03:07.43 | phix | oh and the computer runs asterisk, that is why I am typing that in here ;P |
| 03:07.49 | ManxPower | LiNeTuX_Home: Mandrake/Mandriva |
| 03:07.53 | LiNeTuX_Home | phix: Could be power supply related |
| 03:07.54 | drmessano | Must be more of those awesome SATA server drives |
| 03:07.59 | phix | I get a pci bus master errors from my TDM card too |
| 03:08.11 | phix | made my /var/log fill up pretty quick |
| 03:08.23 | LiNeTuX_Home | ManxPower: I never did stick with Mandrake for long. I just couldn't get used to where they put things. |
| 03:08.44 | ManxPower | LiNeTuX_Home: I can't get used to not having urpmi when I'm on other distros |
| 03:08.54 | phix | LiNeTuX_Home: how so? new 660watt powersupply, two 750Gb Samsung SATA drives, AMD quad core omething |
| 03:09.36 | LiNeTuX_Home | phix: I'm not saying the size is a problem... sometimes you get a dud, sometimes it's just not good quality... you might have a bad +5 or +12v rail |
| 03:09.46 | phix | LiNeTuX_Home: powersupply connected to ups |
| 03:10.14 | v4mp | this is what i get from queue show and agent show |
| 03:10.15 | v4mp | http://v4mpire.pastebin.com/d108bbb23 |
| 03:10.16 | LiNeTuX_Home | ManxPower: apt-get might be better - I haven't used Mandrake in forever to compare, tho |
| 03:10.29 | *** join/#asterisk admin0 (n=admin@bb121-7-191-70.singnet.com.sg) |
| 03:10.31 | LiNeTuX_Home | phix: UPS doesn't have anything to do with it |
| 03:10.47 | ManxPower | LiNeTuX_Home: urpmi resolves the dependencies and can be used for updates, local or remote storage for RPMs |
| 03:10.54 | drmessano | Hmmm |
| 03:10.55 | phix | in6: +4.08 V (min = +0.00 V, max = +4.08 V) |
| 03:10.56 | phix | hmmmmm |
| 03:11.00 | LiNeTuX_Home | phix: And Samsung isn't exactly known for having quality HDD products |
| 03:11.14 | phix | LiNeTuX_Home: great LCDs though :) |
| 03:11.23 | setkeh_ | what is the fedora directory server??? |
| 03:11.26 | phix | how much harder could a HDD be :) |
| 03:11.29 | LiNeTuX_Home | ManxPower: same thing as "yum" for RedHat/Fedora/Cent or "apt-get" for Debian/Ubuntu |
| 03:11.36 | ManxPower | I want to stick to an RPM based distro, dealing with clients that may be skittish about open source, CentOS / Redhat Enterprise seem to be good to know. |
| 03:12.01 | phix | :/ |
| 03:12.05 | phix | I prefer debs |
| 03:12.08 | ManxPower | LiNeTuX_Home: I started using Mandriva in 1999 |
| 03:12.16 | [TK]D-Fender | v4mp: maybe you should look at that while you actually have a CALL in your queue... |
| 03:12.17 | LiNeTuX_Home | ManxPower: yum is good... not as good as apt-get, but it's really good. |
| 03:12.28 | v4mp | ok |
| 03:12.37 | drmessano | I knew of a manufacturer of broadcast equipment that preferred them over Western Digital, Maxtor, and Seagate due to the speed and size of the buffer compared to the size that the others were packaging with with drives of the same size |
| 03:12.44 | jaytee | ManxPower, I've had very good results with CentOS and RHEL 5 and I was coming from a Debian/Ubuntu background. |
| 03:12.47 | phix | sudo apt-get --purge remove redhat; sudo apt-get install debian-or-ubuntu |
| 03:12.54 | drmessano | But it was one particular line, IIRC |
| 03:13.11 | drmessano | Seems like the lesser line was total crap in comparison |
| 03:14.01 | phix | drmessano: yeah, Seagates 120 - 200GB SATAs where complete shit, there 320+ are great |
| 03:14.08 | phix | there = their |
| 03:14.42 | ManxPower | jaytee: how are the docs? |
| 03:14.47 | LiNeTuX_Home | drmessano: I'd rather have a drive in a server last a long time. If I want speed, it's not going to be SATA :) |
| 03:14.48 | v4mp | tk this is with a call |
| 03:14.50 | v4mp | http://v4mpire.pastebin.com/d406669eb |
| 03:15.17 | phix | it is still going :( --> /dev/main-new/srv: Multiply-claimed block(s) in inode 15138831: 60580492 |
| 03:15.20 | jaytee | ManxPower, very good from RHEL and everything I reference from there works the same on CentOS |
| 03:15.25 | phix | so this means physcial problems with the disc? |
| 03:15.26 | drmessano | LiNeTuX_Home: You certainly missed the convo earlier.. heh |
| 03:15.47 | LiNeTuX_Home | A lot of folks are bitching about the perpendicular recording drives not lasting, either... I haven't had one of the 32 I have in a SATA iSCSI box go out on me, however |
| 03:16.17 | LiNeTuX_Home | drmessano: I guess so :) Storage is one of my areas of expertise :) |
| 03:16.18 | [TK]D-Fender | v4mp: Agent/1003 (Busy) has taken no calls yet <-- |
| 03:16.23 | [TK]D-Fender | v4BUSY! |
| 03:16.35 | v4mp | but i dont see how |
| 03:16.42 | [TK]D-Fender | v4mp: Stop phoneing FROM your agent into the queue |
| 03:16.48 | *** join/#asterisk xuser (i=jaood@unaffiliated/xuser) |
| 03:16.51 | LiNeTuX_Home | I think CentOS is very, very good for Asterisk. Apparently so does Digium and most others. |
| 03:16.51 | v4mp | im not |
| 03:16.57 | LARefugee | ~nat |
| 03:16.58 | jbot | nat is, like, Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 03:17.02 | [TK]D-Fender | 2. Local/1@incoming_calls-3239,2 (wait: 0:06, prio: 0) <- Funny this looks like your AGENT |
| 03:17.13 | v4mp | i know i dont get why |
| 03:17.22 | v4mp | i didn't even touch the agent softphone |
| 03:17.22 | scooby2 | does anyone post config examples online or are they too personal? |
| 03:17.27 | jaytee | Agent/1003 (Not in use) has taken no calls yet <--- that looks odd. Not in use? |
| 03:18.08 | jaytee | scooby2, I don't post mine because I get all embarrassed when ManxPower and [TK]D-Fender start laughing. :-) |
| 03:18.14 | scooby2 | lol |
| 03:18.42 | jaytee | plus I'm afraid I might give one of them a seizure just trying to follow my convoluted "logic" |
| 03:19.04 | v4mp | jaytee, where did you see that ? |
| 03:19.18 | [TK]D-Fender | v4mp: http://v4mpire.pastebin.com/m7a1ffe4e |
| 03:19.29 | jaytee | on your previous pastebin before the very last one that [TK]D-Fender commented on |
| 03:19.56 | LiNeTuX_Home | is pleasantly surprised that Budweiser actually put out a decent product with their "American Ale" |
| 03:20.16 | scooby2 | LiNeTuX_Home: you must be lying |
| 03:20.28 | [TK]D-Fender | v4mp: 2. Local/1@incoming_calls-3239,2 (wait: 0:06, prio: 0) this is the same piece of dialplan listed as a CALLER, and as the device for contacting your agent. because it is IN the queue it can't be called OUT. |
| 03:20.28 | v4mp | hmm |
| 03:20.42 | [TK]D-Fender | v4mp: Now stop with the BS circular tests. |
| 03:20.54 | scooby2 | Budweiser and decent in the same sentence? |
| 03:21.00 | v4mp | i dont know what i've done wrong |
| 03:21.02 | jaytee | I felt pretty good today though, had a meeting with my boss and an outside consultant/integrator that does Asterisk and they both complimented my setup and were pretty impressed that I'd done it all from just the book, google and coming in here bugging the hell out of ManxPower and [TK]D-Fender |
| 03:21.11 | [TK]D-Fender | scooby2: You know what American beer & sex in a canoe have in common? |
| 03:21.23 | scooby2 | what? |
| 03:21.31 | LiNeTuX_Home | scooby2: Heh. No, for Bud, it's actually not bad. It's not a great Ale, but it's a solid "C". |
| 03:21.52 | *** join/#asterisk genii (n=user@206-248-139-132.dsl.teksavvy.com) |
| 03:21.52 | [TK]D-Fender | scooby2: They're both fucking close to water |
| 03:21.52 | scooby2 | lol |
| 03:21.52 | *** part/#asterisk genii (n=user@206-248-139-132.dsl.teksavvy.com) |
| 03:21.52 | LiNeTuX_Home | For Bud, that's decent. |
| 03:22.09 | jaytee | I like Sam Adams. Most American beer is "weak tea" for the most part |
| 03:22.23 | jaytee | I love their Hefewiezen. |
| 03:22.31 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 03:22.45 | scooby2 | sam adams is as close to heaven as we get in America |
| 03:22.49 | LiNeTuX_Home | Bud's Ale is pretty close to Sam's Boston Ale. |
| 03:22.51 | v4mp | tk and was you saying that if i tryed to call my home phone from the agents phone it wouldn't work ? |
| 03:23.30 | [TK]D-Fender | Micro-Brewery is the only way to go. |
| 03:23.41 | v4mp | :@ y the hell is * still inviting the call after i ended it with the agents phone :/ |
| 03:23.49 | [TK]D-Fender | v4mp: 2. Local/1@incoming_calls-3239,2 (wait: 0:06, prio: 0) <- I'm saying this sure as hell shouldn't be a CALLER when thats the device used by an AGENT |
| 03:23.59 | jaytee | People are shocked that I compare Miller Genuine Draft to piss, but since I've had survival training in the military in both arctic and desert environments I've actually tasted both. |
| 03:24.02 | v4mp | i know |
| 03:24.10 | [TK]D-Fender | v4mp: so do something else. |
| 03:24.20 | v4mp | but im not putting it there so i dont know what i've done wrong to change it |
| 03:24.26 | LiNeTuX_Home | jaytee: heh |
| 03:24.40 | LiNeTuX_Home | loves me some Dogfish Head |
| 03:24.42 | [TK]D-Fender | v4mp: You have 2 CALLERS in your queue and you don't even know WHY? |
| 03:24.57 | v4mp | no |
| 03:25.02 | jaytee | yes |
| 03:25.12 | v4mp | i would if i was calling in from agents phone which im not |
| 03:25.14 | [TK]D-Fender | v4mp: then maybe you should PB the ENTIRE PROCESS. |
| 03:25.27 | hardwire | cesal: hi.. and no.. I'm not good with centos :( |
| 03:25.31 | [TK]D-Fender | v4mp: You are looking at tiny little pieces when the big picture shows all. |
| 03:26.05 | [TK]D-Fender | v4mp: Step back and look at the building on fire before wondering why one of the room's door handlles seems hot |
| 03:26.21 | jaytee | [TK]D-Fender, did you notice that I mentioned the time earlier today around 4:30 right when someone you were helping decided to fuck up the testing by adding something new to the mix? :-) |
| 03:26.58 | v4mp | tk that doesn't help |
| 03:27.03 | [TK]D-Fender | jaytee: Yeah, I'm used to scatter-brained schmucks who wouldn't know a controlled test if it ran up and bit them in the face |
| 03:27.05 | hardwire | [TK]D-Fender: hot handles may be a sign of your impending assassination. |
| 03:27.14 | hardwire | you know? |
| 03:27.21 | hardwire | I SEEN IT IN A MOVIE#! |
| 03:27.32 | [TK]D-Fender | v4mp: PB the entire damn process of loggin in, placing a call into the queue with dumps before & after. |
| 03:28.10 | v4mp | agent logging in to ? because the login exten doesn't log me out lol |
| 03:28.37 | v4mp | *logout |
| 03:28.46 | *** join/#asterisk DaSkreech (n=skreech@katapult/ninja/daskreech) |
| 03:29.07 | [TK]D-Fender | v4mp: get off your ass a pastebin the empty queue and then the caller going in! |
| 03:30.05 | v4mp | i have already pasted an empty queue |
| 03:30.14 | v4mp | just dont the caller going in |
| 03:30.45 | [TK]D-Fender | v4mp: You're right. Will skip examining the evidence and move right along to passing a VERDICT |
| 03:30.52 | drmessano | [TK]D-Fender: Can't we just talk about what I was gonna pastebin? |
| 03:30.53 | [TK]D-Fender | GUILTY! |
| 03:30.55 | [TK]D-Fender | gavels. |
| 03:31.08 | *** join/#asterisk admin0 (n=admin@bb121-7-191-70.singnet.com.sg) |
| 03:31.13 | [TK]D-Fender | drmessano: Only if you pretend you don't know what I'd answer with ;) |
| 03:31.25 | v4mp | the call with the 2nd 1 jumping in |
| 03:31.26 | v4mp | http://v4mpire.pastebin.com/dc04b6b0 |
| 03:32.41 | LiNeTuX_Home | Why does it have to get so late so fast? I still have stuff I want to do. Ugh. |
| 03:33.01 | MrNaz | what's a good sip softphone for windows |
| 03:33.03 | MrNaz | ? |
| 03:33.08 | [TK]D-Fender | v4mp: the exten you use to call your agent is the exten you use to ENTER THE #&$ING QUEUE! You are sending them in CIRCLES! |
| 03:33.31 | [TK]D-Fender | v4mp: -- Executing [1@incoming_calls:5] Queue("Local/1@incoming_calls-d19a,2", "rep-sales") in new stack <- circular crap! |
| 03:33.49 | [TK]D-Fender | v4mp: Quue leads to calling dialplan that NESTS another damn queue |
| 03:34.08 | [TK]D-Fender | v4mp: No wonder it looked like 2 callers. the queue is calling ITSELF |
| 03:34.20 | v4mp | i dont see where the problem lies unless its where the agent is set to take the call |
| 03:34.28 | [TK]D-Fender | v4mp: Go pay attention where you actually pointed your agentlogin bit to go. |
| 03:34.35 | [TK]D-Fender | v4mp: its your AGENT! |
| 03:34.43 | [TK]D-Fender | Your agent is an exten that calls Queue AGAIN! |
| 03:35.03 | [TK]D-Fender | v4mp: 1003 (Gary Syms) logged in on Local/1@incoming_calls-3239,1 is idle (musiconhold is 'default') |
| 03:35.04 | v4mp | to incoming_calls as i thought thats where i would need to have it listen on |
| 03:35.26 | [TK]D-Fender | v4mp: go look at the dialplan that is executing |
| 03:35.32 | jaytee | MrNaz, X-Lite is probably the most popular |
| 03:36.02 | v4mp | the call comes in on incoming_calls so if the agent shouldn't be listening to that i dont see how it would get the call |
| 03:36.05 | [TK]D-Fender | v4mp: -- Executing [1@incoming_calls:5] Queue("Local/1@incoming_calls-d19a,2", "rep-sales") in new stack <-- this is the same exten you use to call your "agent" |
| 03:37.16 | v4mp | maybe |
| 03:37.21 | jaytee | no, definitely |
| 03:37.23 | [TK]D-Fender | no, not "mabye" |
| 03:37.30 | v4mp | exten=> 2001,1,AgentCallbackLogin(||${CALLERIDNUM}@incoming_calls) should be exten=> 2001,1,AgentCallbackLogin(||${CALLERIDNUM}@phones) |
| 03:37.35 | v4mp | ? |
| 03:37.46 | jaytee | the first thing after the open paren for Queue should be the friggin queue name, not the agent. |
| 03:38.03 | [TK]D-Fender | v4mp: Its writing it right in your face and we all see the exten you told your agent to be conacted at and we all see the code as its executing. Same exten. Same context |
| 03:38.35 | v4mp | and the only thing i see its down to is what i just asked |
| 03:38.37 | [TK]D-Fender | v4mp: 1003 (Gary Syms) logged in on Local/1@incoming_calls-3239,1 is idle (musiconhold is 'default') <-- this happened. |
| 03:38.51 | [TK]D-Fender | v4mp: FIX IT, and then learn to use DIFFERENT CONTEXTS |
| 03:39.18 | [TK]D-Fender | v4mp: You shouldn't have agent call-outs in a context with extens other than those to directly dial an agent's device |
| 03:39.26 | *** join/#asterisk mazpe (n=mazpe@adsl-065-006-163-191.sip.mia.bellsouth.net) |
| 03:39.43 | v4mp | this isn't getting me nowhere is it... im asking if i should be using phones as thats what context the agents account to log into the server is |
| 03:40.04 | v4mp | tk, the only line really to do with agents is the login line and the agents.conf |
| 03:40.35 | [TK]D-Fender | v4mp: Yes, and we SEE where your login line POITNS TO. This is clearly bad. |
| 03:40.50 | [TK]D-Fender | v4mp: You should NOT have pointed your agent call-out context as that one. |
| 03:41.27 | v4mp | so it should be phones ? as thats what the main login to connect to the server is used with |
| 03:41.45 | [TK]D-Fender | v4mp: 1003 (Gary Syms) logged in on Local/1@incoming_calls-3239,1 is idle (musiconhold is 'default') <-- He ended up here do to values not being what you thought they were and picking THAT context. the context choice alone is horrendous. Next you are using a CID var deprecated in 1.2 <- |
| 03:41.55 | [TK]D-Fender | v4mp: Make anothier context |
| 03:43.25 | v4mp | and what should be in the context ? |
| 03:44.19 | [TK]D-Fender | v4mp: When you log in you tell it where to send to calls to. make something useful. |
| 03:47.21 | v4mp | i still dont get what i have to do with the context because its setup so u have to use the [internal] context to be able to make a call to the server |
| 03:47.32 | *** part/#asterisk DaSkreech (n=skreech@katapult/ninja/daskreech) |
| 03:48.48 | *** join/#asterisk AlexTO (n=alex@75.149.245.109) |
| 03:48.51 | Swabby | tk: i can't get my coffee maker to turn on when i press "java" on my phone |
| 03:48.52 | Swabby | j/k |
| 03:49.52 | jaytee | Swabby, it's not "java", it's "brew#" |
| 03:49.55 | v4mp | w8 |
| 03:50.11 | LiNeTuX_Home | IceTea |
| 03:50.18 | v4mp | it should be [context] then an extention to access the agents phone correct ? |
| 03:51.21 | v4mp | how would i make an agent logout extension ? |
| 03:51.27 | [TK]D-Fender | v4mp: It calls your agent via the dialplan <----- What are you not getting here? It goes to an exten and somehow you think your agen'ts phone is going to magically ring. |
| 03:52.32 | [TK]D-Fender | v4mp: v4mp v4mp Well its not magic. It goes to that exten and you'd better put something PRACTICAL in there. |
| 03:52.39 | [TK]D-Fender | damn A-C repeat |
| 03:53.11 | v4mp | ok tk how do i logout from agent as the extension setup i have is wrong |
| 03:53.24 | [TK]D-Fender | v4mp: Fix your login to go somewhere decent and log them in |
| 03:53.40 | v4mp | so i dont need to log the agent out ? |
| 03:53.54 | [TK]D-Fender | v4mp: and pastebin that act as well as you complete dialplan so we can see where you think "appropriate" is. |
| 03:54.16 | [TK]D-Fender | v4mp: If you log them somewhere else, its as good as out & back in, no? |
| 03:54.57 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 03:54.57 | v4mp | exten=> 2001,1,AgentCallbackLogin(||${CALLERIDNUM}@gv) << thats what i use to log them in as agent |
| 03:56.26 | [TK]D-Fender | v4mp: Stop shoing useless individual lines and PB all of it |
| 03:56.40 | v4mp | which i already have done |
| 03:56.45 | [TK]D-Fender | v4mp: And as I told you that var is DEAD! Stop using it. |
| 03:57.09 | v4mp | the CID ? |
| 03:57.41 | [TK]D-Fender | v4mp: CALLER ID <-. Aand don't think for a second I trust that your CID is right on the device logging in or that you have any sane exten to match in that context until you show me |
| 04:03.13 | v4mp | http://v4mpire.pastebin.com/dbdeed75 |
| 04:04.41 | drmessano | [C-Net] Google launches AdSense for Games <-- Adsense for sex coming in 2009?? |
| 04:05.06 | trnzmeta | anyone in us able to test a free call number for me |
| 04:05.12 | trnzmeta | I'm calling from australia and it's not working |
| 04:05.18 | trnzmeta | so I want to check within US |
| 04:05.27 | drmessano | What kind of free call number? |
| 04:05.39 | trnzmeta | alarm signal processor |
| 04:05.47 | trnzmeta | 866-630-6619 |
| 04:06.05 | trnzmeta | it comes up as engaged for me |
| 04:06.35 | jaytee | I get busy when I dial it |
| 04:06.48 | drmessano | "your call did not go thru" |
| 04:07.01 | drmessano | I think their dotcom bubble burst |
| 04:07.36 | trnzmeta | whom? google? |
| 04:08.15 | drmessano | That number is for Google? |
| 04:08.31 | drmessano | Im confused |
| 04:08.45 | trnzmeta | <drmessano> I think their dotcom bubble burst --> re: this |
| 04:08.57 | trnzmeta | thanks for calling the number guys |
| 04:09.06 | drmessano | wow |
| 04:09.11 | jaytee | no worries mate |
| 04:09.28 | drmessano | You just posted the number of some place |
| 04:09.31 | drmessano | We called it |
| 04:09.32 | trnzmeta | I can tell the guys in US to fix their telco stuff first |
| 04:09.39 | drmessano | Get either busy or some other message |
| 04:09.53 | drmessano | So I said I think their dotcom bubble burst |
| 04:10.12 | drmessano | As in |
| 04:10.17 | jaytee | that's one hell of a delayed reaction to the dot-com meltdown then |
| 04:10.26 | drmessano | Fair Dinkum they've gone tits up, shrimp on the barbie |
| 04:10.40 | jaytee | groans |
| 04:10.40 | drmessano | dingo got me baby |
| 04:10.51 | trnzmeta | dinky die, true blue yank over here mate |
| 04:10.54 | jaytee | it's "dingo ate my baby" |
| 04:11.18 | drmessano | err |
| 04:11.19 | drmessano | no |
| 04:11.22 | jaytee | you're a yank? how'd ya end up in Oz? |
| 04:11.27 | drmessano | "dingo got me baby" |
| 04:11.59 | trnzmeta | nah not a yank, I'm just taking the piss |
| 04:12.08 | drmessano | bloody oath |
| 04:12.34 | v4mp | lokl |
| 04:12.45 | drmessano | Sounds like a one pot pisser, mate |
| 04:13.00 | trnzmeta | taking the piss = taking the mickey |
| 04:13.09 | trnzmeta | =teasing |
| 04:13.24 | v4mp | they will know that |
| 04:13.25 | drmessano | lock him in the dunny closet, fair dinkum |
| 04:13.27 | jaytee | "it's a guy! guy dressed up like a sheila! and you all knew ya pack o' bastards!" |
| 04:14.12 | v4mp | think i'll stick to british sayings |
| 04:14.59 | drmessano | Like a bloody yank stuck in a roundabout |
| 04:15.58 | trnzmeta | does it matter if that number I gave out is canadian? |
| 04:16.00 | vader-- | Any of you guys deal with a consulting company? I am looking for ideas of how they charge or have service/maintenance contracts with companies? |
| 04:16.06 | drmessano | I still haven't found a copy of Kumbang on DVD |
| 04:16.29 | drmessano | I talked to a canadian today |
| 04:16.44 | drmessano | Called Xerox support.. kept mentioning something about a boat |
| 04:17.15 | drmessano | "No, not a boat, a copier" |
| 04:17.38 | LARefugee | Re sip behind nat, typical home user, do you really need to forward so many darn ports from your router to your * server (behind the router)? |
| 04:18.35 | drmessano | You can limit it to what you plan to really use in rtp.conf.. but the port number is irrelevant.. same listener |
| 04:18.36 | [TK]D-Fender | vader--: Go call a telecom consulting company and ask for one. |
| 04:18.41 | drmessano | It's just a number |
| 04:18.48 | [TK]D-Fender | vader--: You've been in here daily withthis same quesition. |
| 04:20.14 | vader-- | tkd na this is different |
| 04:20.23 | vader-- | i am looking for ideas on how consultants charge |
| 04:20.24 | v4mp | tk, did you check out my config ? |
| 04:20.37 | vader-- | like X amount of hours of service per month for X amount of dollars |
| 04:20.42 | vader-- | or per incident charges, etc. |
| 04:21.11 | LARefugee | drmessano: hmmmm. b4 I respond I'll take a look at rtp.conf |
| 04:21.15 | vader-- | like 15 Hours a month for 1200$ retainer |
| 04:21.16 | jaytee | and we're all experts at what other consultants charge of course |
| 04:21.29 | vader-- | and then 100$ for every hour after those |
| 04:21.34 | vader-- | use them or loose them, etc. |
| 04:22.31 | drmessano | LARefugee: There is no need to have 10000 ports open, technically.. but you're not fixing anything, making anything more secure, etc, by changing that number... |
| 04:22.35 | jaytee | vader, most of them will charge the standard rate of lebenty-leven, buck 380 for 60K milliseconds |
| 04:22.51 | [TK]D-Fender | vader--: Go ask some other local telecom guy then |
| 04:24.09 | jaytee | vader, just take whatever your local plumber charges per hour and subtract 10%. |
| 04:24.35 | [TK]D-Fender | v4mp: You are still failing to follow even direct instructions and show no initiative. I am losing all will to continue wasting my time with this. |
| 04:24.52 | v4mp | whats wrong now ? |
| 04:24.56 | drmessano | vader--: I charge $337.50 an hour, rounded up to the next hour, and I charge $7.83 for each question the customer asks, with the answering fee being rolled into my hourly charge. If there's pie charts or line graphs involved, we get into exponential cosine's of the hourly-past-response-quotient |
| 04:25.12 | jaytee | rofl |
| 04:25.48 | LARefugee | drmessano: Some howtos recommend forwarding a lot of ports if the * server is behind a nat. Others not at all. I don't. I'm just trying to figure out what's the best approach and why? |
| 04:25.50 | [TK]D-Fender | v4mp: My last statement says it all. |
| 04:25.53 | drmessano | Usually when I roll in the door of a new consulting gig, I affectionately tell the customer, "Better grab your calculator, bitch" |
| 04:26.43 | v4mp | [TK]D-Fender, you said to make a new context which i did and logged the agent in with that... and that works it calls that agents phone like it should so i dont see where the problem is |
| 04:27.04 | drmessano | I like to keep them in the loop on the entire span of charges. It's always awesome when they ask me to elaborate, and as i've been gradually feeding them charges to add, i'll tell them to "ok, now multiply by two".. It |
| 04:27.08 | [TK]D-Fender | v4mp: I asked for a lot more and I am not dealing with this 1 piece at a time. |
| 04:27.10 | drmessano | <PROTECTED> |
| 04:27.30 | drmessano | "Multiply by two, for what?" |
| 04:27.54 | v4mp | [TK]D-Fender, last you asked for was the extensions.conf which is what i gave you |
| 04:27.57 | drmessano | If they don't, the door works the same in both directions.. They can go hire a Skype consultant |
| 04:28.09 | LARefugee | drmessano: Again I'm talking typical home installation with * server behind nat and using one or two sip based services. |
| 04:28.11 | drmessano | Which, BTW.. is awesome job #11 |
| 04:28.31 | drmessano | LARefugee: Open 50 ports and pretend you're more secure |
| 04:28.48 | drmessano | 10000-10050 |
| 04:28.55 | drmessano | Fix them in rtp.conf |
| 04:29.01 | [TK]D-Fender | v4mp: Keep reading back. You are on a journey with this and stopping at EVERY inch and going "are we there yet?" |
| 04:29.20 | v4mp | oh [TK]D-Fender "v4mp: and pastebin that act as well as you complete dialplan so we can see where you think "appropriate" is" that you wanted a call log aswell? |
| 04:29.38 | drmessano | [TK]D-Fender: From the look of that pastebin, I think he just got out to pee at a gas station and found they had no toilet paper |
| 04:29.59 | v4mp | o_O |
| 04:30.32 | drmessano | Hmm.. Skype consulting |
| 04:30.40 | drmessano | That sounds lucrative |
| 04:30.43 | LARefugee | drmessano: I'm not concerned with security at this point. I don't forward *any* ports and it works for me but I'm wondering if I'm limiting myself. Like can I get more than one inbound call on my softphone account. |
| 04:31.06 | jaytee | looking at his extensions.conf file I'm just glad he doesn't work for a company I have to call for tech support since he has no extension 3 |
| 04:31.14 | v4mp | i dont see what you guys are complaining about which is wrong all see that you wanted is to see my extesions.conf as it is now and working with what i have |
| 04:31.46 | [TK]D-Fender | [23:53]<[TK]D-Fender>v4mp: Fix your login to go somewhere decent and log them in |
| 04:31.48 | [TK]D-Fender | [23:53]<v4mp>so i dont need to log the agent out ? |
| 04:31.50 | [TK]D-Fender | [23:53]<[TK]D-Fender>v4mp: and pastebin that act as well as you complete dialplan so we can see where you think "appropriate" is. |
| 04:32.09 | v4mp | act meaning what tho ? |
| 04:32.17 | [TK]D-Fender | v4mp: No, I told to go fix, and then go TRY IT, and then **PASTEBIN THE WHOLE BLOODY MESS ** |
| 04:32.48 | drmessano | Damn |
| 04:33.09 | drmessano | Someone is one step away from being taken out of someones top8 |
| 04:33.13 | [TK]D-Fender | "log them in" <- what part about this isn't clear? What part about pastebinning the attempt? Why aren't you testing your queue? How much more hand holding do you need? |
| 04:33.23 | v4mp | jaytee, extension 3 is irrelvent at present tho getting 1 working first because the extension 3 would be str8 forward as will be going to same place |
| 04:33.33 | v4mp | [TK]D-Fender, i said i tested it and it worked... |
| 04:33.57 | jaytee | good, problem solved. he tested it and it worked. we can all go home now |
| 04:34.00 | [TK]D-Fender | v4mp: You asked what I was waiting for... and it was the pastebin, not your vieled claim. |
| 04:34.10 | v4mp | although i do have an extension 3 but not using it yet just playing sound files with it atm to see what rthere is in sound folder |
| 04:34.30 | v4mp | [TK]D-Fender, fine im trying to avoid extra calls when not needed when it works cuz it costs me |
| 04:34.52 | drmessano | wow |
| 04:34.56 | [TK]D-Fender | v4mp: thats what softphones are fo, this shouldn't cost you anything |
| 04:35.04 | LARefugee | [TK]D-Fender: You know I got chan_alsa to work, right? |
| 04:35.21 | [TK]D-Fender | v4mp: You are simply failing to think outside of "call in via my ITSP" |
| 04:35.32 | [TK]D-Fender | LARefugee: Yes I did. Congrats |
| 04:36.01 | drmessano | I still dont see why theres all this "Pastebin your config" crap. I should be able to describe the problem with the careless abandon of a complete newcomer to asterisk and get an effective solution to my problem |
| 04:36.08 | v4mp | [TK]D-Fender, im using a softphone for the agent phone i have no setup to use another on this system as the sound setup is crap |
| 04:36.12 | [TK]D-Fender | v4mp: And since you claim things work you could have PB'd that like requested. |
| 04:36.28 | [TK]D-Fender | v4mp: A million other ways to test. |
| 04:36.29 | LARefugee | [TK]D-Fender: Thanks. But you were kind of right. It wasn't worth much. Very feature poor compared to chan_oss and chan_console. |
| 04:36.48 | v4mp | http://v4mpire.pastebin.com/d5507dfca |
| 04:36.50 | jaytee | drmessano, :-) |
| 04:36.56 | v4mp | [TK]D-Fender, of which i dont know |
| 04:37.47 | [TK]D-Fender | v4mp: How about "set up another softphone" |
| 04:37.52 | drmessano | jaytee: I have this one line, and it has a colon, ok |
| 04:38.13 | LARefugee | [TK]D-Fender: but I can brag about my experiences on some blog. Like the one I can never get around to starting. |
| 04:38.22 | v4mp | [TK]D-Fender, i didn't find many of much use for mac only found x-lite |
| 04:38.24 | drmessano | jaytee: and a sqiuggly, one of those number signs, and a 103.. followed by some linux and a . |
| 04:38.27 | v4mp | which is what im using for the agent |
| 04:38.30 | drmessano | jaytee: Why cant I call it? |
| 04:38.37 | jaytee | ahahahahaha |
| 04:38.50 | [TK]D-Fender | v4mp: Use need only be enough to RING. |
| 04:38.57 | drmessano | jaytee: Why cant you just HELP me?? |
| 04:39.31 | v4mp | [TK]D-Fender, i couldn't get 1 working with my provider as they claimed to have sip settings which couldn't be found in the menus/prefs |
| 04:39.47 | v4mp | drmessano, thats OTT and far from how it is |
| 04:39.59 | drmessano | OTT? |
| 04:40.05 | v4mp | yes |
| 04:40.11 | drmessano | What is OTT? |
| 04:40.13 | jaytee | "why can't I call this phone?" exten !=> _?~q4,1,Dial(Skype2Flash/444) |
| 04:40.15 | [TK]D-Fender | v4mp: Provider? What does setting up 1 stupid softphone direct with * have to do with a PROVIDER? |
| 04:40.17 | v4mp | over the top |
| 04:40.28 | drmessano | I LOVED that MOVIE |
| 04:40.30 | drmessano | ZOMG |
| 04:40.49 | *** join/#asterisk sah-work (n=Bawbatos@adsl-76-211-254-206.dsl.pltn13.sbcglobal.net) |
| 04:40.51 | drmessano | Stallone + tractor trailer + arm wrestling = FTFW |
| 04:41.03 | v4mp | [TK]D-Fender, because the call would have to come from external to the * to be able to access incoming_calls menu |
| 04:41.19 | drmessano | http://www.imdb.com/title/tt0093692/ |
| 04:41.21 | jaytee | my fav Stallone movie will always be Demolition Man |
| 04:41.29 | [TK]D-Fender | v4mp: Says who? |
| 04:41.46 | drmessano | v4mp thinks his dialplan is as cool as OTT.. whatever |
| 04:41.51 | [TK]D-Fender | v4mp: YOU point them to the context you want. YOU put extens in there to do what you want. |
| 04:42.10 | drmessano | Crash your dialplan into a freakin mansion with an 18-wheeler |
| 04:42.11 | v4mp | well true but either way would be sip connection ? |
| 04:42.17 | [TK]D-Fender | v4And? |
| 04:42.33 | v4mp | so less hassle to just connect to provider |
| 04:42.47 | v4mp | would still be free |
| 04:42.54 | jaytee | v4mp, you can have a context that your phone is in have an include=incoming_calls and that'll let you call it internally. |
| 04:43.00 | [TK]D-Fender | v4mp: But NOOOOO that would cost you and you can't think of alternatives. Sure seems like a simple one to me. |
| 04:43.31 | *** join/#asterisk emist_ (n=emist@unaffiliated/emist) |
| 04:43.48 | v4mp | [TK]D-Fender, well thats not the problem the problem is i dont know of another sip softphone for mac to use along side x-lite so can use it to make calls to * |
| 04:44.24 | jaytee | so just run Windows on Parallels and load X-lite on that as well |
| 04:44.38 | jaytee | jeez, these Mac fanbois need so much handholdin |
| 04:44.45 | v4mp | haha would probz kill the system its an old imac |
| 04:44.58 | [TK]D-Fender | v4mp: And I'm sure you looked real hard. And that IAX couldn't be an option either. Or anything else. I hear excuses, and very weak ones at that. No initiate, no imagination. This is a very upsetting chain of events |
| 04:45.08 | v4mp | fanbois ? lol i dont think much of them which is why i barely use it |
| 04:45.22 | jaytee | so blow it away and install a linux distro that works on PowerlessPC chips like an old version of Ubuntu and then load Ekiga |
| 04:45.35 | v4mp | jaytee, this client is linux |
| 04:45.45 | jaytee | on a Mac? |
| 04:45.48 | v4mp | no |
| 04:45.51 | v4mp | mac is behind me |
| 04:45.57 | [TK]D-Fender | v4mp: http://www.google.ca/search?hl=en&q=macos+free+sip+client&btnG=Google+Search&meta= |
| 04:46.11 | [TK]D-Fender | v4mp: only a pile listed on the first page and links to pages with TONS. |
| 04:46.12 | jaytee | so you're running X-lite for linux? |
| 04:46.21 | v4mp | jaytee, no osx |
| 04:46.25 | [TK]D-Fender | v4mp: Well yippy-kai-yay |
| 04:47.47 | jaytee | then where the fuck is the linux client again? nevermind, I don't really care anymore. I'm picturing a room with 3 computers in it and you can barely walk without tripping over empty beer cans, pizza boxes and Hustler magazines. |
| 04:47.48 | *** join/#asterisk WilliamUIUC (n=will@87.sub-75-206-187.myvzw.com) |
| 04:48.20 | v4mp | haha |
| 04:48.34 | v4mp | jaytee, i did say earlier why i dont have 1 on here |
| 04:48.40 | *** part/#asterisk WilliamUIUC (n=will@87.sub-75-206-187.myvzw.com) |
| 04:49.06 | drmessano | HAHAH!!! |
| 04:49.37 | drmessano | Perfect.. right down to Hustler |
| 04:50.03 | jaytee | v4mp, I'm too lazy to scroll back that far |
| 04:50.15 | v4mp | why would i need hustler i have a wife |
| 04:50.27 | drmessano | Now I know you're full of it |
| 04:50.45 | v4mp | what cuz u dont think i have a wife ? |
| 04:50.46 | jaytee | v4mp, if you really need to ask that question I'm betting you haven't been married for more than 5 years |
| 04:51.34 | drmessano | "I dont need porn, I have a wife" <-- No married man would ever #1 imply sex after marriage, #2 denounce porn |
| 04:51.43 | v4mp | no i haven't |
| 04:52.17 | jaytee | plus it flys in the face of logic. most couples in America are oppossed to "same sex" marriage but they've been having the same sex for years and years. |
| 04:52.19 | drmessano | Im gonna have to stick with Jaytee on this one, not just because I am an strangely attracted to him either |
| 04:52.33 | jaytee | blushes |
| 04:52.43 | drmessano | But you did leave one thing out |
| 04:53.02 | tzanger | heh |
| 04:53.37 | v4mp | oh ? |
| 04:53.39 | drmessano | The patchwork floor where the holes for the toilet and drainpipes used to be, and the capped off pipe stubs under the computer desk where the water lines came into the room when it used to be the spare bathroom |
| 04:54.03 | jaytee | and that constant dripping noise he can't quite locate |
| 04:54.06 | drmessano | Same light fixture on the wall |
| 04:54.10 | drmessano | HAW! |
| 04:54.28 | jaytee | plop! ........... plop! |
| 04:54.52 | v4mp | i find it highly amusing that you try to run people down with no success |
| 04:55.10 | drmessano | Piss poor single bulb bathroom light on the wall that makes a 1 on the screen look like a ! or a |, and even a [, ], [, l, ok, most of the other vertical characters on the keyboard |
| 04:55.38 | drmessano | and the smell that no amount of pizza or farting can seem to cover up |
| 04:56.05 | drmessano | Oh v4mp, laugh a little.. geez |
| 04:56.07 | jaytee | hahahahaha |
| 04:56.19 | v4mp | .. |
| 04:56.36 | v4mp | whats there to laugh about not alot other than what you try to do fails |
| 04:56.41 | drmessano | I know when you laugh that bum leg on your folding chair starts to creek, but calm down |
| 04:57.23 | drmessano | For all you know, I weigh 500 lbs and havent left the house except by flatbed truck in over 10 years |
| 04:58.14 | drmessano | "Not without my pizza" |
| 04:59.05 | LARefugee | G'night all |
| 04:59.10 | drmessano | Hmm |
| 04:59.17 | [TK]D-Fender | ~iwmwb |
| 04:59.18 | jbot | I WANT MY WEEKEND BACK! |
| 04:59.19 | drmessano | I dont see where OSLEC runs on 1.6 |
| 04:59.55 | jaytee | could that be deliberate? |
| 04:59.57 | drmessano | [TK]D-Fender: I would pastebin you a weekend, but that would be all too tragic of an irony |
| 05:03.23 | drmessano | jaytee: dunno |
| 05:03.57 | drmessano | jaytee: I'm also inaccurate.. OSLEC doesn't work with DAHDI.. Asterisk is irrelevant |
| 05:04.36 | jaytee | seems to be quite a bit of confusion as to what does work with DAHDI |
| 05:04.48 | Swabby | i are tired |
| 05:04.51 | Swabby | sleep33 time |
| 05:04.52 | Swabby | lol |
| 05:04.55 | jaytee | nite |
| 05:05.07 | v4mp | hmm dont think theres anything else i need to do for time being |
| 05:05.57 | v4mp | would i need to setup Agent group for more than 1 person to answer the calls or can i just add several agents and have them login the same way ? |
| 05:06.02 | jaytee | no? I usually like to read when things quiet down a little. in fact I highly recommend this detective novel, it's spellbinding |
| 05:06.05 | jaytee | ~book |
| 05:06.06 | jbot | extra, extra, read all about it, book is 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 --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 05:07.11 | v4mp | right time to get some sleep up again in a few hours ¬_¬ |
| 05:07.16 | v4mp | cheers for the help |
| 05:08.05 | jaytee | don't forget to shut off the bathroom....err, office light |
| 05:08.23 | Swabby | speaking of bathroom |
| 05:08.28 | Swabby | maybe i should take a poop before bed |
| 05:08.28 | v4mp | haha more like living room light |
| 05:08.50 | jaytee | Swabby!!!! Hey, thanks for sharing that! |
| 05:09.04 | jaytee | cuz I was starting to miss jeev |
| 05:09.09 | drmessano | jaytee: If he shuts the light off, his "servers" will go down |
| 05:09.15 | Swabby | nice |
| 05:09.34 | Swabby | obama says pakistan weird |
| 05:09.36 | Swabby | i was watching the recap |
| 05:09.38 | v4mp | i dont run ay servers at home |
| 05:09.45 | jaytee | someone else said that earlier |
| 05:10.03 | jaytee | about Obama pronouncing pakistan weird |
| 05:10.08 | Swabby | o |
| 05:10.18 | drmessano | PAK - E - STAN |
| 05:10.24 | Swabby | i like him though |
| 05:10.32 | Swabby | mccain is like an old worn out puppet |
| 05:10.42 | drmessano | I could hate him, and McCain/Palin still scare me |
| 05:11.16 | Swabby | mccain is too fucking old |
| 05:11.58 | drmessano | Jan 21, 2009 - After McCain's inauguration.. pulls a sock puppet out of his sock drawer.. "Ok, LBJ, time to go finish what we started in '64" |
| 05:14.11 | drmessano | I get all the Obama hatemail chain e-mails from a friend of a friend |
| 05:14.35 | drmessano | I take them right to snopes ---> False |
| 05:14.47 | drmessano | Yet people circulate that crap and believe it |
| 05:15.01 | Swabby | whos initials is lbj |
| 05:15.33 | drmessano | Lyndon Baines Johnson |
| 05:15.36 | Swabby | o |
| 05:16.25 | drmessano | LBJ and the CIA had JFK murdered to further the war in Vietnam |
| 05:16.56 | jaytee | yep, at least that's what I've suspected, or he was a reluctant accomplice |
| 05:18.16 | drmessano | I also believe that Trixbox is an elaborate conspiracy by Cisco to discredit Asterisk |
| 05:18.47 | jaytee | wow, as wild as that claim is it actually makes logical sense |
| 05:19.47 | drmessano | Well, the oil companies DO kill people that invent cars powered by water |
| 05:20.02 | the_5th_wheel | has anyone here played with the sangoma BRI cards? |
| 05:20.29 | jaytee | nope, just Legos and Lincoln Logs |
| 05:20.49 | jaytee | oh! and Transformers too |
| 05:22.00 | drmessano | I'm not allowed to talk about BRI |
| 05:22.07 | the_5th_wheel | why? |
| 05:22.42 | drmessano | Ever since I streaked the BRI developers conference in '96 with "BRI BOMB" painted in shoepolish on my chest and buttocks |
| 05:22.58 | drmessano | Court order.. sorry |
| 05:23.07 | the_5th_wheel | goes to find a metal brush to clean his brain |
| 05:24.09 | drmessano | Yeah, that made me a little nauseous too |
| 05:25.29 | *** join/#asterisk CrashSys (i=Kumba@azrael.crashsys.com) |
| 05:25.39 | CrashSys | Can a macro have an H extension? |
| 05:26.37 | jaytee | h |
| 05:27.58 | CrashSys | Plan B... test and see :) |
| 05:29.23 | jaytee | ok, time for me to get some sleep |
| 05:29.25 | jaytee | nite all |
| 05:31.06 | drmessano | Anyone know what the proper codec name for G726 variations is in the SPA9xx Linksys stuff? |
| 05:31.42 | drmessano | I see G726r32 in one place and G726-32 another |
| 05:34.44 | drmessano | Duh nevermind |
| 05:38.38 | *** join/#asterisk thansen (n=thansen@c-67-171-115-155.hsd1.ut.comcast.net) |
| 05:39.08 | *** join/#asterisk emist (n=emist@unaffiliated/emist) |
| 05:41.08 | CrashSys | Does Snom support GSM codecs? |
| 05:41.41 | thansen | I'm upgrading from zaptel to dahdi...does dahdi come with an init script to load the modules? |
| 05:41.57 | CrashSys | did you remember to install mahme? |
| 05:42.08 | thansen | ? |
| 05:42.13 | CrashSys | :) |
| 05:42.23 | thansen | hoped that was a joke :) |
| 05:42.29 | CrashSys | it twas |
| 05:42.35 | CrashSys | but you had to think about it ;) |
| 05:42.42 | thansen | it's true |
| 05:43.21 | CrashSys | We should all think of good acronyms for a phonetically "mommy" sounding name and have Digium fork LibPRI or make sangoma rename Wanpipe... |
| 05:44.08 | thansen | LOL |
| 05:44.41 | CrashSys | Then we can all talk about compiling Dahdi with "mommy" |
| 05:45.15 | thansen | wow |
| 05:45.45 | CrashSys | Vote for me in '08... I have a plan! |
| 05:46.58 | *** join/#asterisk scardinal (n=supreme@90.184.100.170) |
| 05:49.18 | *** join/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net) |
| 06:07.14 | *** join/#asterisk coolthreads (n=shane@203-97-238-71.cable.telstraclear.net) |
| 06:07.58 | coolthreads | my phone calls sound clear but as for when I play a sound file it sounds crap |
| 06:08.11 | coolthreads | any ideas |
| 06:08.27 | sfire | tried multiple players? |
| 06:09.21 | coolthreads | when I use the playback function the answering system sounds crappy |
| 06:09.35 | sfire | hmmm |
| 06:09.50 | kaldemar | coolthreads: what codecs (both call and sound files) are you using and how do you play sound files? |
| 06:10.36 | coolthreads | calls use g711u but the audio files are gsm |
| 06:11.52 | *** join/#asterisk nicox_ (n=nicox@vie-nas-ge-0-2.onenet.at) |
| 06:12.36 | kaldemar | coolthreads: did you compile with gcc > 4.1 ? |
| 06:13.11 | coolthreads | yup I am sure I did |
| 06:13.12 | *** join/#asterisk emist_ (n=emist@unaffiliated/emist) |
| 06:13.33 | kaldemar | there's a known issue with the gsm codec and gcc 4.2. don't know about 4.3. |
| 06:13.55 | kaldemar | wav sound files should work fine for you. |
| 06:14.45 | kaldemar | or you could recompile with gcc 4.1... |
| 06:15.18 | coolthreads | Umm okay I might try again and see how that goes |
| 06:28.28 | coolthreads | what version of asterisk is everyone running? |
| 06:30.19 | sfire | AsteriskNOW-1.0.2.1 |
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| 06:33.55 | CrashSys | 1.4.21.2 is our current stable |
| 06:34.00 | CrashSys | haven't evaluated 1.4.22 |
| 06:36.43 | coolthreads | Prefer to run asterisk through config files rather than gui, okay will try 1.4.21.2? |
| 06:38.50 | coolthreads | only issue I seem to be having is playing back my sound files other than that, my inbounds and outbounds sound great. |
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| 06:49.43 | phix | how do I stop this from happening? -> TDM PCI Master abort |
| 06:51.48 | coolthreads | phix: were u refering to my message? wasnt sure |
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| 07:01.18 | *** join/#asterisk Chris-NB (n=chris@nfw.ecos.at) |
| 07:01.21 | Chris-NB | hi |
| 07:01.52 | Chris-NB | I'm running a callcenter on asterisk 1.4.13 |
| 07:02.02 | Chris-NB | all agents are called via local channel |
| 07:02.10 | CrashSys | sounds like ViciDial |
| 07:02.39 | Chris-NB | sometimes I get this strange behavior. no calls are sent to agents from two (out of 10) specific queues |
| 07:02.53 | Chris-NB | if I do queue show, I can see this: Local/50018@intern (paused) (Not in use) has taken no calls yet |
| 07:03.06 | Chris-NB | but I've no idea how that agent gets 'paused' |
| 07:03.19 | Chris-NB | is this done via PauseQueueMember ? |
| 07:03.38 | Chris-NB | or, can this be done via PauseQueueMember? |
| 07:03.50 | Chris-NB | but! I'm not using this app. |
| 07:04.00 | Chris-NB | I've no clue how this happens |
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| 07:11.48 | Chris-NB | anyone seen this behavior? |
| 07:26.18 | SwK | Chris-NB, sounds like you have auto-pausing turned on... |
| 07:26.34 | SwK | I think that patch was added into the main line stuff anyway... |
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| 07:52.46 | *** join/#asterisk ana_micho (n=sniper_v@87.236.144.38) |
| 07:53.14 | ana_micho | hi I need to recompile asterisk and regenerate all config files I did make clean ...make ..make install..make samples..but the old config files are still there |
| 07:53.45 | mort_gib | ana_micho: remove the first mv -f /etc/asterisk /etc/asterisk.org |
| 07:54.14 | mort_gib | -Remember to remove modules also <- Important |
| 07:54.17 | *** join/#asterisk Slashman (n=Slash@ariane.fimasys.fr) |
| 07:54.31 | ana_micho | mort_gib, did that after that I did execute the same commands ..but ther are no files in /etc/asterisk after recompile |
| 07:56.16 | mort_gib | Uhm, are you sure make samples went ok?? |
| 07:56.42 | mort_gib | did you notice where the files were copied to?? |
| 07:56.55 | ana_micho | let me pastebin make samples |
| 07:57.25 | mort_gib | ok |
| 07:58.53 | ana_micho | mort_gib, http://pastebin.com/m5beeefc3 |
| 08:00.22 | mort_gib | That looks fine, but you are saying that the files are not created?? |
| 08:00.47 | ana_micho | mort_gib, yeeak check ls /etc/asterisk in the lower part of the pastbin |
| 08:00.54 | ana_micho | mort_gib, executing everything as root |
| 08:00.58 | mort_gib | Han on |
| 08:01.19 | mort_gib | -So you can't start * |
| 08:01.44 | mort_gib | Why do you need the samples anyway?? |
| 08:02.29 | ana_micho | mort_gib, I can start it |
| 08:02.39 | mort_gib | Without config files?? |
| 08:02.40 | ana_micho | mort_gib, but itsays mamanger.conf not found...modules.conf not found |
| 08:03.02 | mort_gib | And Extensions.conf sip.conf not found??? |
| 08:03.15 | ana_micho | mort_gib, no not found |
| 08:03.50 | mort_gib | I would, remove the /usr/src/asterisk* dir(s) download again and recompile |
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| 08:07.39 | spike008t | hi all |
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| 08:23.59 | CrashSys | Anyone ever messed with grandstream provisioning? |
| 08:24.38 | CrashSys | or more specifically how the AES provision file encryption works? |
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| 08:30.35 | steliosk | CrashSys : working on Grandstream provisioning but not on encryption (yet) |
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| 08:33.06 | CrashSys | I just want the encryption part |
| 08:33.12 | CrashSys | the provisioning I have figured out |
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| 09:03.33 | flohack | Is there a way to find out at which line in the dialplan a specific channel currently is? |
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| 09:04.58 | pputman | I've been having some issues with the hardware being recognized by the asterisk gui with latest 1.4 using dahdi, has anyone had the same problem, or know if 1.6 works better? |
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| 09:16.01 | Chris-NB | SwK, autp-pausing? how can I enable/disable that? |
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| 09:33.33 | flohack | Is there a way to find out at which line in the dialplan a specific channel currently is? |
| 09:34.37 | kaldemar | by line do you mean extension or priority? |
| 09:37.46 | kaldemar | core show channels will show you, anyway. |
| 09:44.11 | sky_blue | since installing app_conference i now get ast_translator_build_path: No translator path from unknown to unknown........... i'm guessing this is a codec problem? |
| 09:45.50 | flohack | kaldemar: priority, show channels just tells me the extension / macro |
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| 09:50.49 | kaldemar | flohack: under location you should see <exten>@<context>:<priority> |
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| 10:16.56 | flohack | kaldemar: You are right, the problem is just that the channel i've been looking at has a truncated Location string. |
| 10:17.07 | flohack | kaldemar: Is there a way to display the full string? |
| 10:19.31 | kaldemar | flohack: show channel <channel> |
| 10:19.54 | flohack | ok, thanks a lot! |
| 10:28.14 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 10:28.18 | dominic1 | Hello guys |
| 10:29.13 | dominic1 | short question: Do you know if there is any implementation in sip to get to show the caller the callerid(name) of the called person? We had a avaya system before and there we was able to see which person we called.... |
| 10:40.47 | *** join/#asterisk emist_ (n=emist@unaffiliated/emist) |
| 10:56.09 | flohack | dominic1: There was a discussion on the asterisk-ML about this topic a few days ago |
| 11:00.44 | flohack | dominic1: The thread was: [asterisk-users] How to add Callee's name into Dial command ? Friday Oct, 3 11:21:23 |
| 11:16.45 | *** join/#asterisk DarkRift (n=dark@65.92.171.219) |
| 11:26.48 | tzafrir_laptop | yikes ,res jabber is NOSY |
| 11:27.10 | tzafrir_laptop | NOISY, that is |
| 11:34.59 | *** join/#asterisk Squeeb (i=squeeb@eggwee.co.uk) |
| 11:35.50 | Squeeb | Hi, I need to cut customer's off after 20 minutes due to regulations. But instead of just abruptly ending the call while they're talking to an agent, I want the IVR to tell them they need to dial back to continue the conversation and then hangup |
| 11:36.05 | Squeeb | I've been using set(TIMEOUT(absolute)=XXXXXX) |
| 11:36.13 | Squeeb | but obviously that just hangs up the channel when the timeout is reached |
| 11:36.18 | Squeeb | any idea how I can do this? |
| 11:38.28 | Squeeb | Anyone here? |
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| 12:08.10 | *** mode/#asterisk [+o russellb] by ChanServ |
| 12:09.00 | *** join/#asterisk metfan2007 (n=jc@189.180.216.169) |
| 12:09.06 | metfan2007 | hi all! |
| 12:09.07 | riddlebox | when I upgraded to asterisk-1.4.22 I could no longer use my fxs ports on my tdm, even though the fxo port worked. has anyone else run into this? |
| 12:09.40 | metfan2007 | I'm receiving the messages posted in http://pastebin.ca/1222656 while trying to connect a PRI |
| 12:09.47 | metfan2007 | what does that means? |
| 12:10.00 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-76bae557e2eff08f) |
| 12:10.26 | pluesch0r | how should i debug a sip hardware client (some linksys phone) that's simply not able to connect? i'm doing sip debug on the asterisk console with a verbosity of 20 .. no luck in seeing any packets. |
| 12:10.45 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 12:11.06 | tzafrir_laptop | riddlebox, all channels are defined in the same config file? |
| 12:13.02 | *** join/#asterisk TommyBJ (n=noosjent@ipu.sandefjord.kommune.no) |
| 12:14.49 | metfan2007 | any? |
| 12:15.20 | Squeeb | If I have monitor-type = mixmonitor defined in queues.conf, should I be presented with only 1 file with both halves of the conversation mixed to mono? |
| 12:15.25 | Squeeb | I keep getting 2 files, in and out |
| 12:15.29 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:16.25 | TommyBJ | I'm having some trouble with res_config_mysql app. I'm trying to adopt from version 1.2 to 1.4 as "live" as possible. The 1.2 works fine, but the 1.4 does not show any users. There is an established connection between the asterisk server and the mysql server. |
| 12:17.38 | TommyBJ | Any ideas? |
| 12:20.13 | riddlebox | tzafrir_laptop, yes all are using the same config |
| 12:20.51 | tzafrir_laptop | the FXO ports are before or after the FXS ports? |
| 12:21.03 | riddlebox | yes fxo is port 1, fxs 2,3 |
| 12:21.03 | dominic1 | flohack: thank you very much |
| 12:22.01 | tzafrir_laptop | and port 1 actually works? for calls? |
| 12:22.09 | Squeeb | Hmm |
| 12:22.10 | Squeeb | wtf |
| 12:22.19 | Squeeb | does monitor-type = mixmonitor actually work? |
| 12:22.26 | Squeeb | because I keep getting split files |
| 12:22.48 | riddlebox | tzafrir_laptop, yup, and actually if a call comes in, I have the fxs ports in the ring group and they ring and I can talk, but I cannot dial out of them, or even check voicemail |
| 12:23.05 | riddlebox | tzafrir_laptop, I downgraded to 1.4.21.2 and they work fine again |
| 12:24.14 | tzafrir_laptop | do you get any error from loading chan_dahdi.so ? |
| 12:24.30 | tzafrir_laptop | do you have /etc/asterisk/chan_dahdi.conf ? |
| 12:25.01 | riddlebox | tzafrir_laptop, so I am supposed to move to dahdi in 1.4.22? |
| 12:25.14 | riddlebox | I know they moved all references to zap to dahdi |
| 12:25.26 | tzafrir_laptop | riddlebox, I just asked |
| 12:26.21 | riddlebox | tzafrir_laptop, nope nothing with dahdi in it |
| 12:26.28 | metfan2007 | I'm receiving the messages posted in http://pastebin.ca/1222656 while trying to connect a PRI, any idea what does that means? |
| 12:28.00 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 12:31.48 | Squeeb | Grr this is getting annoying |
| 12:32.07 | Squeeb | how come monitor-type = MixMonitor still creates two seperate files? |
| 12:33.27 | tzafrir_laptop | riddlebox, do you get any error (in 1.4.22) when running 'dahdi restart' ? |
| 12:33.46 | [TK]D-Fender | Squeeb: And if your use mixmonitor for a normal call,what happens? |
| 12:33.58 | Squeeb | not tried |
| 12:34.20 | riddlebox | I didnt check that, I just downgraded to 1.4.21.2, because the fiance likes to use the cordless phones |
| 12:34.36 | [TK]D-Fender | Squeeb: Go try |
| 12:35.10 | Squeeb | How do I define which directory MixMonitor records to? I found savecallsin inside agents.conf |
| 12:35.13 | Squeeb | is that correct |
| 12:36.33 | [TK]D-Fender | Squeeb: In a very clear folder in the astlibdir as specified in asterisk.conf unless you provide it an absolute path. |
| 12:36.46 | Squeeb | aah |
| 12:37.08 | Squeeb | I think it's because I've been using the record option in agents/conf |
| 12:37.13 | riddlebox | tzafrir_laptop, I guess I will have to play with it when she isnt around |
| 12:37.23 | TommyBJ | I'm having some trouble with res_config_mysql app. I'm trying to adopt from version 1.2 to 1.4 as "live" as possible. The 1.2 works fine, but the 1.4 does not show any users. There is an established connection between the asterisk server and the mysql server... any ideas? |
| 12:37.36 | TommyBJ | Anything would be useful at this point ;) |
| 12:38.24 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:38.24 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 12:39.54 | riddlebox | hey tzafrir_laptop I was wondering, what is the zaptel patch for oslec? I noticed there isnt one for the latest zaptel, I just used the last versions patch |
| 12:40.25 | tzafrir_laptop | shouldn't it work for latest zaptel? have you tried oslec 0.0.2 ? |
| 12:40.43 | riddlebox | I always use svn |
| 12:44.32 | Squeeb | Hmm |
| 12:44.54 | Squeeb | can I define a path for MixMonitor to record to in queues.conf? |
| 12:45.09 | Squeeb | because at the moment it just uses the mixmonitor default locatio |
| 12:46.50 | [TK]D-Fender | Squeeb: Have you resolved the non-mixing issue yet? |
| 12:46.57 | Squeeb | yea |
| 12:47.16 | [TK]D-Fender | Squeeb: show us your config |
| 12:47.22 | Squeeb | I was using the record feature in agents/conf |
| 12:49.55 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 12:53.59 | petererer | TommyBJ, I'd guess you'd need to check the table schemas -- fields may have been added or changed. |
| 12:54.15 | tzafrir_laptop | The patch there should work for latest zaptel, IIRC |
| 12:57.56 | Blackvel | anyone using LDAPGet with openldap? |
| 12:59.44 | Madkiss | just a quick question; I want something in my dialplan that is a "catch-all" rule for everything that is not explicitly covered by an extension |
| 12:59.52 | tzafrir_laptop | riddlebox, "chan_dahdi.c" warning/error in the logs? |
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| 13:02.17 | *** join/#asterisk bpgoldsb (n=bpgoldsb@gleim-gw.atlantic.net) |
| 13:02.43 | bpgoldsb | Can you call a macro with no arguements, or should you just use a context for that? |
| 13:03.17 | russellb | yeah, you can do that. |
| 13:03.41 | riddlebox | tzafrir_laptop, nothing in /var/log/asterisk/messages |
| 13:03.46 | bpgoldsb | Is either more efficient or 'right'? |
| 13:04.11 | *** join/#asterisk Linuturk (n=Linuturk@fluxbuntu/developer/Linuturk) |
| 13:04.21 | Carlos_PHX | bpgoldsb: My rule is to use a macro where I want to go do something and return. |
| 13:05.03 | [TK]D-Fender | Carlos_PHX: No need for arguments. Of course your could have just used Gosub, but whatever... |
| 13:05.16 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 13:05.43 | [TK]D-Fender | Madkiss: Fo anything but zaptel, not possible. |
| 13:07.36 | flohack | Has someone ever witnessed hangs within the dialplan Set(DB(..)) application? |
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| 13:08.20 | flohack | It looks a bit like a deadlock, as it only happens on higher load (about 10 concurrent calls) |
| 13:09.26 | *** join/#asterisk DarnoQ (i=DarnoQ@chello089076117071.chello.pl) |
| 13:10.45 | *** join/#asterisk bbryant (n=Brett_Br@adsl-159-32-97.flo.bellsouth.net) |
| 13:11.01 | flohack | As you can see here: http://pastebin.com/m517795eb That for example 7723 has one call up with the set-lastcall macro and at the same time the queue tries to get through and the caller has a ringing channel to the macro queue-pausewrapup. |
| 13:11.02 | *** join/#asterisk arpu (n=arpu@chello080109017114.12.14.vie.surfer.at) |
| 13:12.11 | flohack | At this point asterisk seems to be stuck, as the ringing call does net get through to the user (SIP usage limit) and the call to the macro set-lastcall does not hangup |
| 13:13.48 | *** join/#asterisk mltlnx (n=mltlnx@pool-96-224-6-243.nycmny.east.verizon.net) |
| 13:16.10 | flohack | The macro set-lastcall is executed by the dial command (M parameter) when the queue calls the extension |
| 13:16.33 | *** join/#asterisk ibm2 (n=Administ@196.203.192.179) |
| 13:17.48 | *** join/#asterisk FinboySlick (n=FinboySl@207.134.8.4) |
| 13:19.03 | ibm2 | hi , please there's any one know how to install video in asterisk |
| 13:19.16 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 13:20.24 | [TK]D-Fender | ibm2: Its documented on the WIKI. Go lookup it and give ti a try |
| 13:20.50 | bpgoldsb | If you're calling a macro without an arguement, you just call it is '¯oname();' I guess? |
| 13:20.52 | FinboySlick | I'm trying to diagnose why voicemail won't send email. The settings look right but I don't even see anything resembling an attempt to send an email in the logs. |
| 13:21.18 | *** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk) |
| 13:21.54 | [TK]D-Fender | FinboySlick: And clearly we see this too... |
| 13:23.00 | FinboySlick | [TK]D-Fender: Hehe, still a meanie... So, you want my voicemail.conf? |
| 13:23.17 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 13:23.32 | [TK]D-Fender | finDo you want help? If so you'd best be showing us something useful to do so with |
| 13:25.14 | FinboySlick | [TK]D-Fender: Of course. one moment while I strip coments. |
| 13:26.16 | knobo | is there any documentation on how a table should look like in mysql for realtime configuration? |
| 13:26.19 | knobo | somewhere? |
| 13:26.29 | Squeeb | Yea, I saw it earlier .. let me see if I can dig it out |
| 13:26.34 | Squeeb | infact I remember |
| 13:26.41 | Squeeb | it's in the O'Reilly book on asterisk |
| 13:26.46 | Squeeb | under the AMI section |
| 13:26.51 | [TK]D-Fender | knobo: Looked in the BOOK lately? |
| 13:27.01 | Squeeb | [TK]D-Fender: beat you to it :P |
| 13:27.02 | flohack | knobo: On the wiki too: http://www.voip-info.org/wiki-Asterisk+RealTime+Voicemail |
| 13:27.43 | [TK]D-Fender | Squeeb: Congratulations, you can claim a plushie of your choice from the top rack now... |
| 13:27.55 | Squeeb | yay |
| 13:28.25 | flohack | Any ideas concerning my deadlock problem with Set(DB(..)) as posted above? |
| 13:31.48 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 13:33.29 | CGMChris | Can anyone tell me how to send multiple extension numbers to the same physical phone? |
| 13:35.54 | FinboySlick | [TK]D-Fender: http://www.pastebin.ca/1222715 would be voicemail.conf |
| 13:37.49 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 13:41.29 | rwaite | lord this provider is crap |
| 13:42.05 | Squeeb | What would be the best method of limiting a call to a maximum of twenty minutes. |
| 13:42.20 | Squeeb | I've thought about using set(TIMEOUT(absolute)=whatever) |
| 13:42.46 | Squeeb | but that bins the call as soon as the time is up, how would I at least have a good bye message before the hangup? |
| 13:42.46 | rwaite | what's the best way to eat? |
| 13:42.56 | rwaite | i've thought about putting food in my mouth and swallowing it |
| 13:42.57 | Squeeb | Eh |
| 13:43.16 | rwaite | Squeeb: probably the h extension |
| 13:43.27 | ibm2 | please can anyone tell me how i can install video in asterisk |
| 13:43.27 | Squeeb | hmm |
| 13:43.34 | Squeeb | ibm2: it's on the wiki |
| 13:43.55 | ibm2 | is not clear in wiki |
| 13:43.56 | rwaite | but i'm not *absolutely* sure that the h extension will be called in that case |
| 13:44.08 | Squeeb | yea, that's what I thought |
| 13:44.24 | Squeeb | also, if it's called AFTER the call is terminated |
| 13:44.33 | Squeeb | etc.. |
| 13:44.34 | rwaite | that's true |
| 13:44.40 | rwaite | hmm. |
| 13:45.02 | kaldemar | Squeeb: look at dial app's L() and g |
| 13:45.03 | Squeeb | Unless you know of a way of having asterisk "butt in" to a conversation with a warning message just before the absolute timeout |
| 13:45.23 | kaldemar | core show application dial |
| 13:45.44 | Squeeb | kaldemar: I saw those yea, but the calls are placed from the Queue, to agents.. can't see how to add the L option to the Dial app called by queue. |
| 13:45.45 | rwaite | you can put a timeout in the dial command |
| 13:46.15 | rwaite | nevermind im an idiot |
| 13:46.52 | kaldemar | Dial(Local/queueexten@queuecontext,...) |
| 13:47.07 | kaldemar | ugly as hell, but... |
| 13:48.38 | Squeeb | erg |
| 13:48.39 | Squeeb | yea |
| 13:49.29 | Squeeb | thing is, I'm already passing a 2 minute timeout to Queue() so if nobody answers, it apologises and calls another number |
| 13:50.03 | Squeeb | IE: exten => 6500,n,Queue(${EXTEN}||||120) |
| 13:56.37 | bpgoldsb | Does the 2nd edition orilley book cover AEL/AEL2? |
| 13:57.41 | russellb | i don't think so |
| 13:57.47 | bpgoldsb | :( |
| 13:58.19 | bpgoldsb | Know one that does? Or a good resource? |
| 13:58.22 | [TK]D-Fender | FinboySlick: Maybe you'll think of showing me your mail Q's, and note that you'll never get the recording in any e-mail it'll send out regardless |
| 13:59.08 | FinboySlick | [TK]D-Fender: The last bit is intentional. I just want notification. As for the first... It's empty. |
| 13:59.31 | *** join/#asterisk NoCarrier (n=NoCarrie@unaffiliated/badpacket) |
| 14:00.34 | FinboySlick | [TK]D-Fender: I'm really just looking for asterisk to tell me when it tried to send an email. Where that mail gets jammed afterward I can deal with. |
| 14:02.11 | CGMChris | Still trying to figure out why call queues automatically go to the first unavailable agents voicemail... if and only if agents have voicemail. Shouldnt an agent be able to have their own voicemail and still help answer calls from a queue? |
| 14:02.30 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 14:03.52 | *** join/#asterisk seanmh (n=seanmh@nat/digium/x-45d3d88522359deb) |
| 14:04.21 | jaytee | anyone here tried HUD Lite with standard Asterisk? |
| 14:05.40 | *** join/#asterisk chrisq (n=chrisq@parrot.kotelett.no) |
| 14:05.42 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 14:06.11 | chrisq | anyone had any luck implementing queues-with-callback-members.txt? |
| 14:06.23 | [TK]D-Fender | FinboySlick: * doesn't send you an e-mail... your sendmail equivalent script does, and you should look in there for anything queued up |
| 14:06.57 | [TK]D-Fender | CGMChris: If it hits VM its because you sent it to an exten that HAS voicemail which is something you should never do |
| 14:07.04 | FinboySlick | [TK]D-Fender: I know that. I want asterisk to give me a hint, like: /usr/bin/sendmail: command not found |
| 14:07.22 | [TK]D-Fender | FinboySlick: Is there a sendmail binary? |
| 14:07.32 | TommyBJ | Asterisk keeps telling me that there are no D-Channels assigned... where do I assign these? |
| 14:07.35 | FinboySlick | [TK]D-Fender: I'm giving an example. This isn't the problem. |
| 14:07.39 | *** join/#asterisk mog (n=mog@nat/digium/x-0ba250c05479f136) |
| 14:07.39 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:08.15 | FinboySlick | But yeah, my sendmail is functioning properly. |
| 14:08.32 | [TK]D-Fender | FinboySlick: Go look at the queue |
| 14:08.37 | codefreeze-lap | bpgoldsb: I try to keep the voip-info wiki page on AEL2 up to date... see http://voip-info.org/wiki/view/Asterisk+AEL2 |
| 14:08.40 | [TK]D-Fender | FinboySlick: look at its logs, etc |
| 14:09.01 | cryptnix | wireless sip phone ... battery & clarity. any thoughts? |
| 14:09.03 | bpgoldsb | codefreeze-lap: I've been using that, and it's pretty good. I just need more examples and whatnot. |
| 14:11.06 | [TK]D-Fender | ~wifivoip |
| 14:11.06 | jbot | [~wifivoip] Wi-Fi (802.11a/b/g) VoIP phones on the market may possess any of the following drawbacks : poor battery life, limited range, lack of call features (many lack transfer, etc), poor or no NAT traversal, shoddy construction, etc. Because of this they are usually not recommended. Some alternatives are : VoIP ATA + cordless, DECT w/ VoIP base, etc |
| 14:11.10 | codefreeze-lap | bpgoldsb: there is a link to some further examples in there... lets see.... http://voip-info.org/wiki/view/AEL+Example+Snippets |
| 14:11.59 | cryptnix | hrm... then the question is ... phone base stations that have range boosters |
| 14:12.02 | cryptnix | of some type. |
| 14:12.23 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 14:12.37 | FinboySlick | [TK]D-Fender: I found noting in its logs or my mailq... hence my looking for info from asterisk. As far as I can tell, sendmail never even gets called. |
| 14:12.45 | chrisq | codefreeze-lap: is the ael-queue example in doc/ still valid ael? |
| 14:13.01 | *** join/#asterisk ming_zym (n=ming_zym@220.181.34.146) |
| 14:13.06 | chrisq | i'm trying to use it on 1.4.19, not without problems |
| 14:13.18 | [TK]D-Fender | cryptnix: DECT is good for range,quality, & battery unless the product is particularly bad. The tech itself is fine |
| 14:13.35 | [TK]D-Fender | FinboySlick: There is nothing in *. * niether knows or cares once it calls sendmail. |
| 14:14.02 | FinboySlick | Ah, but does it call sendmail at all in my specific case, that is my question. |
| 14:14.05 | codefreeze-lap | Or, carry a 19-24 db directional antenna with you at all time, connected to your phone with a fat cable, you can get 2 ft mesh dish antennas or big yagi's! |
| 14:14.24 | FinboySlick | Can it tell me that it did call sendmail, or at least tried to. |
| 14:14.39 | [TK]D-Fender | FinboySlick: go see. |
| 14:16.12 | adr3nalin3 | Does echo cancellation do anything for speaker phone? A lot of my users complain about the echo when the far end has them on speaker. Would echo cancellation do anything or should I just tell them to stfu? |
| 14:18.11 | *** join/#asterisk shazaum (n=shazaum@unaffiliated/shazaum) |
| 14:18.29 | [TK]D-Fender | adr3nalin3 :What phone? Whats on the other end of the call? Is it on while on SP mode? |
| 14:20.50 | CGMChris | D-Fender: So, in other words, if any agent associated with a call queue has their own voicemail, and is busy/offline, the queue will always pull the caller OUT of the queue and redirect to that persons voicemail? It just seems counter-intuitive that it would work that way. |
| 14:21.40 | [TK]D-Fender | CGMChris: No, I'm saying that if YOU say that an Agent is to be called via the dialplan that you'd better make sure that exten doesn't lead to VM or anything else that will answer the call <- |
| 14:21.44 | cryptnix | wow. |
| 14:21.55 | cryptnix | no hack jobs son! |
| 14:22.02 | *** join/#asterisk XnOSX (n=XnOSX@212.145.173.80) |
| 14:22.11 | CGMChris | D-Fender: I am not even talking about a dial plan... This is the default behavior for internal calls. |
| 14:22.42 | [TK]D-Fender | CGMChris: This IS dialplan. |
| 14:23.02 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 14:23.16 | [TK]D-Fender | CGMChris: "if any agent associated with a call queue has their own voicemail" <- this is dialplan. VM doesn't happen out of thin air. Pastebin is yro friend |
| 14:23.24 | [TK]D-Fender | your* |
| 14:23.52 | *** join/#asterisk tvirus (i=TheVirus@c-69-243-46-240.hsd1.md.comcast.net) |
| 14:23.56 | CGMChris | D-Fender: So, what is the solution to my problem? The developers suggested that I need to create 2 extenions per phone...1 with and 1 without voicemail, and use only the one without voicemail for the queue... but I cannot figure out how to do this. |
| 14:24.13 | [TK]D-Fender | CGMChris: You should be showing us your queu processing calls and your dialplan including the parts used by the queue |
| 14:24.26 | adr3nalin3 | [TK]D-Fender: It is a snom 320, SIP mode, not sure whats on the other end. My guess is it really depends on the phone. |
| 14:24.39 | CGMChris | D-Fender: http://pastebin.com/d3fccffe0 |
| 14:24.40 | [TK]D-Fender | CGMChris: Go make another dialplan context and put extens in there that will dial your PHONES. |
| 14:24.41 | tvirus | I'm trying to prevent dialing when 2 SIP channels are in use, but this isn't working: http://rafb.net/p/Xq3mNT39.html Do I need to define SIPGROUP as a global variable in extensions.conf or something? |
| 14:24.55 | *** join/#asterisk NoCarrier (n=NoCarrie@unaffiliated/badpacket) |
| 14:24.59 | Squeeb | Dial(Local/queueexten@queuecontext,...) ... kaldemar I'm not sure I understand that |
| 14:25.12 | [TK]D-Fender | adr3nalin3 :What TECH is the call going though? Is it to the PSTN? How? |
| 14:25.13 | Squeeb | queuecontext? |
| 14:25.28 | adr3nalin3 | [TK]D-Fender: PRI |
| 14:25.29 | FinboySlick | [TK]D-Fender: To make my question very clear, where does asterisk log its attempts to call sendmail and under which conditions, if any? |
| 14:25.51 | *** join/#asterisk ManxPower (n=manxpowe@96.sub-75-249-89.myvzw.com) |
| 14:25.54 | kaldemar | Squeeb: your dialplan consists of contexts that have extensions in them. in that, queuecontext is just a context in your dialplan. |
| 14:26.02 | [TK]D-Fender | FinboySlick: There is nothing in *. * niether knows or cares once it calls sendmail. <----------- |
| 14:26.32 | Squeeb | aah right |
| 14:26.41 | FinboySlick | [TK]D-Fender: *once* it calls sendmail... Yes. But I want to know *IF* it calls sendmail. |
| 14:26.56 | [TK]D-Fender | FinboySlick: Go replace Sendmail and see. |
| 14:27.25 | adr3nalin3 | FinboySlick: tail -f /var/log/mail or /var/log/maillog |
| 14:28.00 | adr3nalin3 | while you leave a vm. that will tell you. It also might tell you if you have routing problems to your smtp server. |
| 14:29.31 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.95.130) |
| 14:30.02 | FinboySlick | adr3nalin3: The queue is not local, I use ssmtp as a relay to another mail server. but okay, I'll manage. |
| 14:31.44 | CGMChris | D-Fender: If I create a seperate dialplan/context for queues only, I can no longer test the queue from internal phones by dialing the queues extension (5000). How do I test the system? |
| 14:32.25 | [TK]D-Fender | CGMChris: Says who? |
| 14:32.46 | [TK]D-Fender | CGMChris: this is the context the queue will dail out into to contact your AGENTS |
| 14:33.10 | [TK]D-Fender | CGMChris: It serves no other purpose |
| 14:36.21 | *** join/#asterisk mags2 (n=mags2@ampulex.whoi.edu) |
| 14:37.03 | CGMChris | So, to make sure I understand... I am creating a new context, adding agents to it... and these agents should not have voicemail? |
| 14:39.27 | [TK]D-Fender | CGMChris: You are creating a new contex, add EXTENSIONS in it that will dial the appropriate DEVICE only, and no, it should not lead to VM or anything else that would answer the call |
| 14:42.38 | bpgoldsb | Can someone give me a one-liner explaining when to use a macro vs when to use a 'goto to a seperate context' |
| 14:42.50 | [TK]D-Fender | bpgoldsb: Use a macro when you want to come BACK |
| 14:43.09 | [TK]D-Fender | (or pass arguments |
| 14:43.12 | tvirus | Hmm, this group stuff still isn't working with a global variable. |
| 14:43.17 | ManxPower | bpgoldsb: Macros are going away. I suggest you use gosub if you can. |
| 14:43.36 | bpgoldsb | ManxPower: I thought GoSub was removed? |
| 14:43.41 | bpgoldsb | I get warnings every time I use it |
| 14:44.04 | ManxPower | Anyone know if Macros are officially deprecated? |
| 14:44.16 | ManxPower | bpgoldsb: when do you use it? |
| 14:44.20 | [TK]D-Fender | ManxPower: they are in 1.6 |
| 14:44.34 | ManxPower | [TK]D-Fender: chances are they will be removed in 1.8? |
| 14:45.04 | ManxPower | bpgoldsb: now if you are talking about usage in AEL, that is a different story -- one which was discussed on the mailing list this week |
| 14:45.05 | bpgoldsb | ManxPower: I'm updating some older, 1.2 style dialplan for 1.6/AEL2 |
| 14:45.09 | [TK]D-Fender | ManxPower: I don't know exactly hbow the deprecation cycle works due to X.Y.Z changing on "Z" now |
| 14:45.42 | ManxPower | bpgoldsb: AEL incorrectly throws a warning when using gosubs, search the mailing list archives for this week. |
| 14:46.06 | ManxPower | [TK]D-Fender: also Macros are everywhere so maybe there will be an extra release cycle or two before it's actually removed. |
| 14:46.12 | *** join/#asterisk sam_albuquerque (n=sam@gauntlet.oregan.net) |
| 14:46.18 | bpgoldsb | ManxPower: Will do, thanks. I just got started a few days ago, so I'm not really up to speed on all the latest. I appreciate the help. |
| 14:46.18 | [TK]D-Fender | ManxPower: I strongly suspect that. |
| 14:46.29 | CGMChris | D-Fender: Is there an example or documentation on how to link multiple extensions to a single device? I am struggling with that part. |
| 14:46.50 | kaldemar | from UPGRADE.txt: "However, since Macro() has been around for a long time and so many dialplans depend heavily on it, for the sake of backwards compatibility it will not be removed." |
| 14:46.52 | [TK]D-Fender | CGMChris: There is no such thing as linking 2 extensions. |
| 14:47.06 | [TK]D-Fender | CGMChris: Each extension does whatever you tell it. |
| 14:47.11 | ManxPower | bpgoldsb: I think the Subject of the thread is "AEL and swap from macros to contexts" |
| 14:47.17 | CGMChris | Extension 1001, extension 5000, etc. |
| 14:47.24 | CGMChris | Multiple extensions to the same device. |
| 14:47.25 | *** part/#asterisk vitalstatistix (n=sam@gauntlet.oregan.net) |
| 14:47.40 | [TK]D-Fender | CGMChris: there is no concept of "association". At all. |
| 14:47.53 | [TK]D-Fender | CGMChris: You jsut make another EXTENSION. |
| 14:48.07 | ManxPower | CGMChris: exten => 1001,1,Dial(SIP/device-1) and exten => 5000,1,Dial(SIP/device-2) |
| 14:48.19 | [TK]D-Fender | ManxPower: Not the same device |
| 14:48.21 | ManxPower | or did you do something SILLY and name your devices and extensions the same? |
| 14:48.30 | ManxPower | [TK]D-Fender: perhaps I need more coffee. |
| 14:48.41 | ManxPower | GMChris: exten => 1001,1,Dial(SIP/device-1) and exten => 5000,1,Dial(SIP/device-1) |
| 14:48.52 | [TK]D-Fender | ManxPower: You don't need this so early in the morning? (greif, not coffe... THAT you clearly do need) |
| 14:48.54 | NovceGuru | edits extensions.conf |
| 14:48.57 | *** join/#asterisk pluesch0r (n=pluesch0@91.186.158.42) |
| 14:49.34 | NovceGuru | ManxPower: there are a LOT of howtos that have you do that |
| 14:49.43 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-7764df401a8401a8) |
| 14:49.44 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:49.57 | ManxPower | [TK]D-Fender: I suspect is problem is the classic "I named my devices and extensions the same thing and now I'm getting confused when I try to do anything unusual in my dialplan" thing. |
| 14:50.22 | ManxPower | NovceGuru: It is STILL a bad idea. |
| 14:50.24 | [TK]D-Fender | ManxPower: Trust me, EVERYTHING is getting jumbled up in those requests. |
| 14:50.51 | ManxPower | [TK]D-Fender: Huh? |
| 14:50.57 | *** join/#asterisk easycrypt (n=savek@ip-186.emscb.ruhr-uni-bochum.de) |
| 14:51.01 | CGMChris | Well, I got everything setup exactly how I want EXCEPT this... now I am converting from using the GUI to editing the conf files, and IT named the devices and extensions the same. I guess its time to echo "" > extensions.conf |
| 14:51.28 | ManxPower | CGMChris: Now you start to understand why we don't like GUIs. |
| 14:51.35 | NovceGuru | ManxPower: I wasn't disagreeing :) |
| 14:51.41 | ManxPower | I think it is irresponsible for Asterisk GUI to do that. |
| 14:52.05 | [TK]D-Fender | CGMChris: Trash your configs and start over. |
| 14:52.10 | CGMChris | It's irresponsible to get this far and not be able to properly configure a call queue with the GUI too. I object! :) |
| 14:52.26 | CGMChris | K, starting over, will be back in a few days. |
| 14:52.30 | *** join/#asterisk MindTheGap (n=MindTheG@201.80.60.227) |
| 14:52.51 | [TK]D-Fender | CGMChris: Objection duly noted. Unless you're ready, don't bother asking how much it was due... |
| 14:52.56 | ManxPower | CGMChris: make a backup copy of your configs. Asterisk GUI seems to be one of the best of a really crappy class os software. |
| 14:53.22 | CGMChris | I will backup the configs... there are alot of features/examples in it. |
| 14:53.22 | *** part/#asterisk pluesch0r (n=pluesch0@91.186.158.42) |
| 14:53.41 | ManxPower | I need to find a place to download the CentOS ISO. |
| 14:53.41 | MindTheGap | i have made a small change in the manager interface "manager.c" how do i compile just the parts that need it? without wecompiling the whole asterisk? |
| 14:55.10 | Squeeb | kaldemar: |
| 14:55.11 | Squeeb | thanks |
| 14:55.12 | Squeeb | exten => s,n,Dial(Local/6500@queues||L(1200000:60000)) |
| 14:55.14 | Squeeb | this worked |
| 14:55.33 | Blackvel | any way to check on linux if a script has newline characters at end of line? |
| 14:55.41 | Blackvel | can vi display them? |
| 14:55.54 | Blackvel | my script and ldapadd is complaining about missing newlines |
| 14:59.36 | *** join/#asterisk mltlnx (n=mltlnx@nmd.sbx07238.newyony.wayport.net) |
| 15:00.48 | NovceGuru | can you just add a newline, or echo -e "\n" or something |
| 15:01.26 | NovceGuru | (echo -e "\n" >> file) |
| 15:02.20 | Blackvel | for each line? |
| 15:02.49 | ManxPower | Blackvel: edit the file, add a blank line at the end. |
| 15:04.03 | Blackvel | no way to show the special characters in any linux editor? |
| 15:05.29 | *** join/#asterisk kanelbullar (n=kanelbul@193.126.30.193) |
| 15:05.38 | Blackvel | hm did that |
| 15:05.43 | Blackvel | ldapadd still complains |
| 15:05.54 | ManxPower | Blackvel: you would have to check the docs for YOUR editor, but it would be faster to just add a blank line at the end. |
| 15:05.59 | Blackvel | I am on my way to throw it out of the window |
| 15:06.08 | Blackvel | did that... |
| 15:06.18 | ManxPower | it's LDAP of course it's not going to work. 8-| |
| 15:06.57 | Blackvel | I just understand nada....working hours on this very simple ldap tree to pump my contacts into |
| 15:07.12 | Blackvel | someone helped me on #openldap |
| 15:07.27 | Blackvel | and now it doesn'T accept the ldif file |
| 15:07.38 | Blackvel | why must all be that complicated |
| 15:07.55 | Blackvel | just because snom MISSED to have a phone which accepts many phone entries... |
| 15:08.03 | ManxPower | Blackvel: you are experiencing what I experience everytime I try to use LDAP |
| 15:08.17 | Blackvel | look, I mean I am not dumb |
| 15:08.31 | Blackvel | have running asterisk, running ivr, running ISDN BRI connection |
| 15:09.06 | Blackvel | but it takes me hours just to create a simple structure (and even that is not working now) or to find out what all that crap of dc=, o=, ou= stuff means |
| 15:09.16 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:09.33 | Blackvel | I do not even want to count what money I could have earned in these days |
| 15:09.54 | *** join/#asterisk mltlnx (n=mltlnx@nmd.sbx07238.newyony.wayport.net) |
| 15:10.00 | Blackvel | could have been SOO easy |
| 15:10.05 | Blackvel | outlook plugin |
| 15:10.11 | Blackvel | pumping address book to snom phone |
| 15:10.13 | Blackvel | voilà |
| 15:11.19 | Blackvel | if LDAPGet crashes when I have my data in the ldap server, I swear, I will throw it out of window |
| 15:11.45 | Blackvel | how do you guys make it running with 1000-5000 phone entries? |
| 15:11.51 | Blackvel | aint there any easy solution? |
| 15:12.21 | smth | <PROTECTED> |
| 15:12.36 | ManxPower | Blackvel: I don't. Address books are up to the user |
| 15:12.46 | Blackvel | and how do they do? |
| 15:13.18 | Blackvel | I mean I have phonesuite.de which syncronizes my laptop with asterisk incoming call already |
| 15:13.18 | [TK]D-Fender | smth: You have no dtmfmode set |
| 15:13.22 | Blackvel | but its not always the case to have the computer running |
| 15:15.45 | smth | [TK]D-Fender, I set it at a peer configuration. and I also tried add dtmfmode set at general. same results. |
| 15:15.51 | *** join/#asterisk tcseke (n=chatzill@217.20.134.239) |
| 15:16.14 | [TK]D-Fender | smth: is not set in general, and your peer is irrelevant |
| 15:18.13 | *** join/#asterisk tobias (n=tobias@user-0c8hj2l.cable.mindspring.com) |
| 15:18.49 | Blackvel | the peer is irrelevant? |
| 15:18.55 | smth | <PROTECTED> |
| 15:19.18 | Blackvel | smth: dtmf not working? for IVR? |
| 15:19.23 | [TK]D-Fender | smth: it isn't matching your peer and you should be apying attention to your sip debug |
| 15:19.30 | Blackvel | pstn - patton GW - asterisk |
| 15:20.11 | Blackvel | had to change the peer in asterisk from inbound to rfc (patton used rfc dtmf). but of course it matched the correct peer |
| 15:20.22 | smth | Blackvel, yeah, just inband dtmf for incoming call not working |
| 15:20.41 | ManxPower | smith: to make debugging easier set context=INVALID in [general] then put the correct context= in for the peer. If an incoming call gets sent to INVALID context, then the incoming call is not matching the peer and so uses the settings in [general] |
| 15:21.19 | ManxPower | smth: You understand that gateways don't just magically know what Asterisk's DTMF most is set to, right? |
| 15:21.31 | [TK]D-Fender | smth: And inband on GSm is CRAZY |
| 15:22.08 | *** join/#asterisk badcfe (i=christia@morra.di.er.kjip.no) |
| 15:22.32 | ManxPower | We all know inband with anything other than ulaw or alaw just won't work |
| 15:22.35 | badcfe | i have a deny and permit rule under [general] in sip.conf but its not taken into account by * |
| 15:22.46 | badcfe | should it be under [authentication]? |
| 15:22.54 | ManxPower | badcfe: no it won't be, that's for peers. Was sip.conf.sample wrong? |
| 15:23.06 | *** join/#asterisk bram247 (n=bram@96.28.114.46) |
| 15:23.12 | smth | but I made the outbound call from asterisk .it works fine whatever inband,2833 or info. |
| 15:23.21 | badcfe | ManxPower: prolly the sample is fine. its me who blowed it up. thanks |
| 15:23.24 | ManxPower | smth: That does not change facts. |
| 15:23.28 | Blackvel | smth: can't find disallow or allow lines in your sip.conf |
| 15:23.46 | ManxPower | badcfe: access controls are on a per sip.conf entry basis, not [general] |
| 15:24.24 | [TK]D-Fender | smth: And when you dial out you are also probably using a peer where at least you SET a dtmfmode in the first place.. |
| 15:24.24 | smth | I use a/ulaw. and tried reclaimd them in sip.conf . same thing |
| 15:24.27 | badcfe | ManxPower: hmm. so i cant say "that if something is not recognised specifically it must come from this and that ip address"? |
| 15:24.38 | [TK]D-Fender | smth: You're inbound sample has "failure" written all over it from the start |
| 15:25.21 | [TK]D-Fender | badcfe: just send your calls under [general] to an empty context |
| 15:25.46 | ManxPower | badcfe: not that I'm aware of. |
| 15:26.08 | ManxPower | smrepastebin your config after your changes. |
| 15:27.19 | *** join/#asterisk LARefugee (n=chatzill@c-76-104-191-194.hsd1.wa.comcast.net) |
| 15:27.36 | smth | <PROTECTED> |
| 15:28.15 | [TK]D-Fender | smth: No peer was getting MATCHED on the call you showed me! |
| 15:28.52 | *** join/#asterisk Thorn (n=thorn@unaffiliated/thorn) |
| 15:29.50 | *** join/#asterisk _mm_ (n=mmclain@cpe-67-49-233-178.dc.res.rr.com) |
| 15:31.11 | smth | [TK]D-Fender,the peer'account showed in sip.conf with a did number which is showed in inbound extension. |
| 15:32.37 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 15:33.36 | [TK]D-Fender | smth: -- Executing [6474768313@inbound:1] Goto("SIP/76.74.139.50-08205948", "testdtmf|s|1") in new stack <---- ... NO |
| 15:35.02 | *** join/#asterisk boolean12 (n=boolean1@38.100.94.60) |
| 15:35.34 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:35.58 | *** join/#asterisk boolean12 (n=boolean1@38.100.94.60) |
| 15:36.37 | *** join/#asterisk steliosk (n=Stelios@91.140.124.241) |
| 15:36.37 | ManxPower | [TK]D-Fender: shouldn't "SIP/76.74.139.50-08205948" be something like "SIP/gateway_inband-incoming-08205948"? if the incoming call matched the peer? |
| 15:36.57 | *** join/#asterisk MrNaz (n=naz@210-84-39-63.dyn.iinet.net.au) |
| 15:37.12 | [TK]D-Fender | ManxPower Yes, and I don't know how many more times I have to repeat the same thing...... |
| 15:37.36 | outtolunc | at least 3 more times <G> |
| 15:37.41 | ManxPower | [TK]D-Fender: he's not listening don't waste your time on him. |
| 15:38.06 | [TK]D-Fender | ManxPower: thats what I tried saving you from earlier. Guess the physician oughtta.... |
| 15:38.26 | ManxPower | *nod* |
| 15:38.32 | Blackvel | guys |
| 15:38.51 | Blackvel | what linux editor (vi options ??? ) can display all characters e.g newlines in a file? |
| 15:39.00 | Blackvel | come on you know it |
| 15:39.01 | boolean12 | nano, joe, pico |
| 15:39.06 | Blackvel | you use it on a dialy basis :) |
| 15:39.54 | ManxPower | Blackvel: what in the world makes you think we need to view newlines in a file? |
| 15:40.03 | ManxPower | do read the damn frickn docs for VI |
| 15:40.50 | boolean12 | brb |
| 15:41.06 | smth | sorry.[TK]D-Fender, I ve just something else around. I get what you mean. I am checking . thanks anyway! |
| 15:41.20 | ManxPower | smth: what is your native language? |
| 15:41.34 | smth | japanese |
| 15:41.47 | Blackvel | ManxPower: doing that for days for openldap...I am fed up reading docs with no return |
| 15:41.55 | Blackvel | anyways...used nano (didn' |
| 15:42.01 | Blackvel | didn't display)...but its fixed now |
| 15:42.06 | ManxPower | Blackvel: I'll be happy to put you on my /ignore list. |
| 15:42.30 | ManxPower | [TK]D-Fender: seems like a monday, doesn't it? |
| 15:42.46 | smth | ManxPower, that's why sometime I cannot response quickly. |
| 15:44.04 | jaytee | ManxPower, here's a link for CentOS download mirrors. http://mirror.centos.org/centos/5/isos/ |
| 15:44.46 | ManxPower | jaytee: I meant a place for my computer to be that does not have metered service (my home service has a 5 GB/month cap. |
| 15:44.56 | FinboySlick | [TK]D-Fender: I really don't want to be an annoyance with this, and I understand that there are lots of other factors that are possibly the cause of my problem. But still, is there a way to get some sort of confimation from asterisk itself that it did try to call sendmail? All I've tried so far points to no. I guess I could strace, but I'd prefer some sort of log file. |
| 15:45.27 | jaytee | ManxPower, ah i see. 5GB a month cap? that's pretty stingy of your ISP |
| 15:45.30 | ManxPower | FinboySlick: setting debug to 99 does not show it running sendmail? |
| 15:45.43 | ManxPower | jaytee: Verizon Wireless EVDO |
| 15:45.44 | Blackvel | ManxPower: oh, thank you very much for being that nice to me. didn't even know that "not interested to help you" can be rephrased as "I'll be happy to put you on my /ignore list." |
| 15:45.45 | [TK]D-Fender | FinboySlick: I told you what to do but you seem to have a reading problem. I'm not going to beit either of us over the head for this any longer |
| 15:46.12 | ManxPower | Blackvel: you said you don't want to read any more docs. |
| 15:46.12 | FinboySlick | ManxPower: Well, that looks like an answer. set core debug 99? |
| 15:46.35 | ManxPower | FinboySlick: what happens when you try it. |
| 15:47.19 | FinboySlick | ManxPower: I'll give it a go. With verbose, I saw that it wrote the message to disk, but not that it called sendmail to tell me it was there. I'm trying now. |
| 15:47.27 | ManxPower | jaytee: I have 4 options for internet service where I live: dialup, satellite or Verizon EVDO, T-1 |
| 15:48.17 | jaytee | ManxPower, wow no broadband at all? that sucks |
| 15:48.20 | Blackvel | again my bad English problem... I should have more specifically said: I am fed up searching for informations in docs which are not there, not understandable or whatever...look I already lost TOO much time :(. But it's working now...thanks to boolean12 |
| 15:48.48 | ManxPower | jaytee: not everywhere has broadband |
| 15:49.12 | [TK]D-Fender | beat* |
| 15:49.24 | rwaite | is running fxotune something that should always be done on a new server with a tdm400p? |
| 15:49.29 | ManxPower | last time I downloaded an ISO I was at some nearby coffee shop. |
| 15:49.34 | jaytee | ManxPower, yeah most carriers won't run it into rural areas because the profits just aren't big enough. |
| 15:49.41 | ManxPower | rwaite: it would not hurt to run it |
| 15:50.13 | mort_gib | ManxPower: I get HSDPA only in my house |
| 15:50.15 | rwaite | cause ive had some severe echo issues and i ran it and it seems to have helped, but i dont remember seeing anything about it until a few days ago on th voip-info wiki. just wondering |
| 15:50.21 | ManxPower | jaytee: I'm 11 miles from the CO (phone office) and the calls are mux'd on a SLIC 96 device about 3 miles from where I live. |
| 15:50.40 | ManxPower | mort_gib: verizon is the only carrier with ANY cell service where I live |
| 15:50.54 | FinboySlick | ManxPower: I tried the "core set debug 99" thing. Once again I see it write the wav file to disk, but I have no indication that sendmail or any other command was called. Don't I have to enable some other flag and restart asterisk to get full debug info? |
| 15:51.03 | mort_gib | I get Vodafone :-( Just for the record -THEY SUCK! |
| 15:51.14 | ManxPower | FinboySlick: put a copy of your voicemail.conf on pastebin.ca |
| 15:51.33 | rwaite | hmm. |
| 15:52.13 | *** join/#asterisk voxter (n=voxter@S01060016b6b53c0c.vc.shawcable.net) |
| 15:52.24 | FinboySlick | ManxPower: http://www.pastebin.ca/1222820 |
| 15:52.38 | jaytee | ManxPower, well I messed with Satellite internet access at my last job and at that time the speeds were horrible. |
| 15:53.11 | ManxPower | jaytee: I had satellite internet for a while. dialup was actually more responsive than the satellite for SSH |
| 15:53.57 | *** join/#asterisk freakazoid0223 (n=mattc@pool-68-238-186-228.phil.east.verizon.net) |
| 15:54.14 | jaytee | ManxPower, I can't even remember the name of the system we were testing at my old job back in 2000. The downlink wasn't bad but the uplink was glacial and dodgy. |
| 15:54.40 | *** join/#asterisk pluesch0r (n=pluesch0@91.186.158.42) |
| 15:54.43 | ManxPower | jaytee: it's the high latency (600ms - 5000ms) |
| 15:54.49 | jaytee | ManxPower, yep |
| 15:55.29 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 15:55.41 | jaytee | damn, my boss has a major hemorrhoid flareup going on today. He's been in a closed door meeting yelling at one of my coworkers for an hour now. |
| 15:56.20 | pluesch0r | evening! i've got asterisk running with sip. i've also got voicemail service working with sending recorded voicemails via email working. |
| 15:56.27 | jaytee | which sucks because what he's pissed at is the software vendor's fault and not my coworker. |
| 15:56.42 | pluesch0r | what i want to do now is offer each user the possibility to listen to voicemails when calling *98 or whatever .. how do i achieve that? |
| 15:57.41 | [TK]D-Fender | pluesch0r: "core show application voicemailmain" |
| 15:57.44 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 15:57.53 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:58.11 | pluesch0r | uhm |
| 15:58.28 | sky_blue | hi all, i've been having dtmf issues with a certain sip provider and just used smth's testdtmf. # key is initiating a transfer... tried dtmf rfc, inband, auto & info. what next? |
| 15:58.43 | *** part/#asterisk ManxPower (n=manxpowe@96.sub-75-249-89.myvzw.com) |
| 15:58.56 | FinboySlick | pluesch0r: I'd say you create an extension that calls VoiceMailMain() |
| 15:59.24 | pluesch0r | FinboySlick: yeah .. but how do i achieve *89 to be callable by everyone and how do i invoke VoiceMailMain with the correct parameters? |
| 15:59.33 | FinboySlick | pluesch0r: Here's how I do it: exten => 500,1,VoiceMailMain(${CALLERID(num)}|s) That's for the trusted internal context though. |
| 15:59.56 | pluesch0r | thanks. :) |
| 15:59.58 | [TK]D-Fender | pluesch0r: Go read its instructions for the parameters, and the exten is just an exten. |
| 16:00.16 | pluesch0r | [TK]D-Fender: i just didn't know about CALLERID. thanks. :) |
| 16:00.28 | [TK]D-Fender | pluesch0r: Not needed. |
| 16:00.32 | *** join/#asterisk korihor (n=korihor@201.211.168.130) |
| 16:00.48 | [TK]D-Fender | pluesch0r: And it assumes it matches the box # anyways. |
| 16:00.58 | FinboySlick | pluesch0r: Yeah, I put it there for convenience so that people get straight to *their* mailbox when they call 500. |
| 16:01.08 | pluesch0r | okay |
| 16:01.28 | FinboySlick | Otherwise it'll ask for the mailbox number and then the password. |
| 16:01.49 | FinboySlick | Which is what I do when people call from outside. |
| 16:02.06 | pluesch0r | i see. |
| 16:02.12 | v4mp | what would i need to add for the called to be forwarded to an agent group ? |
| 16:02.14 | pluesch0r | it's still asking, even though i'm calling from internal. :) |
| 16:02.48 | v4mp | or a link to help with it better ? |
| 16:02.53 | v4mp | ~book |
| 16:02.54 | jbot | rumour has it, book is 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 --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 16:03.00 | FinboySlick | pluesch0r: Well, chances are your callerid isn't set properly, or your mailbox isn't named after the callerid. Try replacing the callerid variable by the name of the mailbox. |
| 16:03.40 | rene- | hello |
| 16:03.53 | rene- | any experienced user of asterisk answering machine detection? |
| 16:03.59 | pluesch0r | FinboySlick: how do i debug what callerid is set? |
| 16:04.10 | sky_blue | anybody know why # key is initiating xfer instead of accepting pin code on meetme() |
| 16:04.31 | *** join/#asterisk Thorn (n=thorn@unaffiliated/thorn) |
| 16:04.59 | FinboySlick | pluesch0r: You could make an extension that tells it to you when you call. |
| 16:05.34 | FinboySlick | pluesch0r: exten => 700,n,SayNumber(${CALLERID(num)}) |
| 16:05.39 | pluesch0r | whoa. |
| 16:05.44 | pluesch0r | thanks. hah. |
| 16:06.24 | pluesch0r | interesting. "zero" |
| 16:06.51 | rwaite | cero? |
| 16:08.07 | Linuturk | can anyone recommend a good, solid state, appliance type device to use in a simple asterisk setup in my satellite offices? I'd like to be able to connect to our main asterisk box over IAX as well |
| 16:08.08 | pluesch0r | okay .. setting it in sip.conf worked. ;) |
| 16:08.23 | Linuturk | taking 3 analog lines at each location |
| 16:08.52 | Linuturk | I'm guessing something with 1 pci slot should do, running off of CF card |
| 16:09.07 | Linuturk | these things are going to be sitting in a closet, next to the water heaters |
| 16:09.17 | Linuturk | so, rugged is desired |
| 16:09.42 | Linuturk | anyone have experience with a particular piece of hardware they can recommend? |
| 16:10.06 | FinboySlick | Linuturk: Alix boards are solid little PC things. I'm sure you could get asterisk running on those. Do you need to use analog phones? |
| 16:10.08 | MrNaz | yea |
| 16:10.28 | Linuturk | we've got all IP based phones internal, but we have analog lines coming in |
| 16:10.47 | mort_gib | Linuturk I use Soekris 5501 boards |
| 16:11.22 | mort_gib | Which means that I can use a HDD if my clients start complaining about Voicemail storage |
| 16:11.58 | Linuturk | well, there are only like 4 or 5 phones in these offices |
| 16:12.04 | FinboySlick | Linuturk: I don't think they make USB fxo modules so ALIX is out ;) |
| 16:12.09 | mort_gib | The Case that Wim (www.kd85.com) sells will fit a Sangoma A200 card, that can take up to 8 incoming calls |
| 16:12.41 | mort_gib | You could try to get hld of a few Soekris 4801, but they are at end of life... |
| 16:13.27 | Linuturk | mort_gib, well, your 5501 boards look good |
| 16:13.43 | Linuturk | handle 3 concurrent calls? |
| 16:13.52 | mort_gib | What codec?? |
| 16:13.55 | Linuturk | ulaw |
| 16:14.17 | mort_gib | Dunno, depends. Some 8-12 I think |
| 16:15.03 | mort_gib | I started a conference, one Zap and 5 SIP running G729 and CPU went to 78% |
| 16:15.18 | mort_gib | But then, you DON'T have to use G729 internally |
| 16:16.19 | Linuturk | well, these guys are running 512 Mb ram, with a p4 2.8 in them right now |
| 16:16.25 | Linuturk | at least, that's one of them |
| 16:16.49 | mort_gib | You can get a rack mount fo rthat board too... |
| 16:17.26 | Linuturk | same specs on both satellite offices |
| 16:17.36 | mort_gib | Linuturk: You know what, get the BEST spec you can, NOBODY will thank you for going cheap! |
| 16:17.42 | *** join/#asterisk mercutioviz (n=chatzill@freeswitch/developer/msc) |
| 16:17.46 | Linuturk | yeah, I was just thinking that |
| 16:17.47 | mort_gib | That said, the Soekris boards ARE nice |
| 16:18.12 | Linuturk | I just want to replace these towers with something smaller, so they can be taken off the floor, even wall mounted |
| 16:18.14 | mort_gib | I use them for Firewalls running OpenBSD, where the QOS comes in REALLY handy! |
| 16:18.23 | *** part/#asterisk mercutioviz (n=chatzill@freeswitch/developer/msc) |
| 16:18.34 | Linuturk | right now, they are sitting next to a water heater, on the floor |
| 16:18.37 | Linuturk | not ideal |
| 16:19.10 | mort_gib | -No, not ideal. As I mentioned, there is a rack mount out for the 5501 |
| 16:21.46 | Linuturk | well, I figure a few of these 5501's will do the trick |
| 16:22.14 | Linuturk | I can use the existing cards from the current servers |
| 16:23.11 | Linuturk | thanks mort_gib :) |
| 16:23.30 | *** join/#asterisk emist (n=emist@unaffiliated/emist) |
| 16:23.34 | mort_gib | Your elcome |
| 16:23.40 | mort_gib | elcome=welcome |
| 16:23.49 | pluesch0r | hmm .. where do i set/enable that dialing-tone sound when calling an extension? |
| 16:23.57 | pluesch0r | that extension is set to call some sip number .. |
| 16:24.07 | pluesch0r | is musiconhold the correct file? |
| 16:26.08 | *** join/#asterisk implicit- (n=bayan@unaffiliated/implicit) |
| 16:28.26 | v4mp | how would i send the call queue to an agent group ? i know how for a single agent but cant find out how to change that to a group |
| 16:28.54 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 16:29.13 | bpgoldsb | Is there a way to do elseif { } in AEL2? |
| 16:31.22 | *** join/#asterisk implicit- (n=bayan@unaffiliated/implicit) |
| 16:32.06 | *** join/#asterisk tigger27 (n=clegault@69.15.99.2) |
| 16:32.35 | tigger27 | I knew some folks in here, so I thought I'd give this a shot, you know how people are on IRC, just lurking about |
| 16:32.42 | tigger27 | wrong channel |
| 16:32.59 | *** part/#asterisk pluesch0r (n=pluesch0@91.186.158.42) |
| 16:33.04 | tigger27 | feels stupid now |
| 16:34.23 | tigger27 | So, obviously I have a question or I wouldn't be showing up in a channel that I am never in |
| 16:35.05 | tigger27 | I was wondering if anyone has time to direct me to some documentation that would help me in getting channel id information via the manager api |
| 16:35.28 | tigger27 | I would like to do something like show channelid for extension 101 |
| 16:35.51 | tigger27 | I can't seem to find anything about this |
| 16:36.05 | tigger27 | I find lots of useful commands if I already know the channel id |
| 16:37.20 | *** join/#asterisk angryuser (n=Miranda@51.210.20.81.dynamic.adsl.abo.nordnet.fr) |
| 16:38.01 | jameswf | does callerid=no prevent an fxs channel from sending (passing) caller id |
| 16:38.17 | Qwell | fxs sends callerid? |
| 16:38.26 | jameswf | pass to cid box |
| 16:38.34 | Qwell | oh |
| 16:40.21 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 16:40.26 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 16:40.42 | v4mp | Qwell, whats the extension line i would need to send the queue to an agent group ? |
| 16:40.58 | Qwell | "agent group"? |
| 16:41.04 | tigger27 | jameswf: it is supposed to not, correct |
| 16:41.11 | Qwell | there's no such thing, and I suspect if you asked previously, you were already told that |
| 16:41.27 | v4mp | Qwell, then why is it on the configs ? |
| 16:41.48 | v4mp | Group memberships for agents (may change in mid-file) |
| 16:41.54 | tigger27 | I think I saw a bug registered in 1.4.10 that shows its broken though |
| 16:42.32 | v4mp | then in queues.conf |
| 16:42.45 | v4mp | member => Agent/@1 ; Any agent in group 1 |
| 16:43.02 | Qwell | so what's the question? |
| 16:43.31 | v4mp | how to i add it to the extension so incoming calls in the queue ring the agents |
| 16:43.45 | Qwell | what is the name of your queue? |
| 16:45.02 | mort_gib | I have an issue with the incoming callerid on a A20-0 Sangoma card |
| 16:45.27 | mort_gib | That delays the call being picked up by * |
| 16:45.59 | v4mp | Qwell, 1 as the others are numbered so i thought it had to be a number but either way thats fine for now |
| 16:46.12 | Qwell | your queue is named "1"? |
| 16:46.17 | v4mp | yes |
| 16:46.22 | Qwell | so then Queue(1) |
| 16:46.27 | v4mp | oh w8 no |
| 16:46.39 | bpgoldsb | Is there a way to do elseif in AEL other than this http://pastebin.com/m4e3eea48 |
| 16:46.59 | v4mp | Qwell, i got that part sorted thats fine |
| 16:47.00 | v4mp | but |
| 16:47.27 | v4mp | when an agent logs in.. need the queue to ring the agents group so they call can be answered |
| 16:49.07 | v4mp | Qwell, this is the part uim having trouble with http://v4mpire.pastebin.com/d342ace76 |
| 16:49.12 | Blackvel | can snom 370 ldap integration use the ldap entries for outbound calls or just for inbound calls? |
| 16:49.58 | codefreeze-lap | v4mp: you can do if(x) {} else if (y) {} and skip all the {}'s.... |
| 16:50.33 | v4mp | o_O |
| 16:53.10 | codefreeze-lap | v4mp: or if you get a lot of that, and it adapts properly, you could also use a switch... |
| 16:54.00 | v4mp | that doesn't matter to me atm main thing is getting it to work properly as it is |
| 16:54.19 | Qwell | codefreeze-lap: I think you means bpgoldsb |
| 16:54.21 | Qwell | meant* |
| 16:55.22 | codefreeze-lap | uh, yeah, sorry v4mp, I **did** mean bpgoldsb. Thanks, Qwell. Us old guys get confused easily... |
| 16:56.16 | codefreeze-lap | I was wondering about the o_O, now I know... lol |
| 16:57.31 | codefreeze-lap | bpgoldsb: read back and everything I said to v4mp, I meant to say to you! |
| 16:57.54 | bpgoldsb | codefreeze-lap: Thanks :) |
| 16:58.28 | v4mp | haha |
| 16:58.28 | codefreeze-lap | goes to wash off glasses |
| 17:04.42 | *** join/#asterisk paul0 (n=paulo@201-25-243-191.fnsce701.dsl.brasiltelecom.net.br) |
| 17:04.57 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 17:05.10 | paul0 | hi, i would like to know what is the minimun latency to make good voip calls |
| 17:05.15 | sky_blue | i've redefined blindxfer => # in features.conf to blindxfer => ## however # still xfers. anything i'm missing? |
| 17:05.57 | paul0 | i've ordered 10 sipdiscount credit, but the quality isn't very good, the delay is quite annoying |
| 17:06.03 | *** join/#asterisk MrNaz (n=naz@210-84-35-223.dyn.iinet.net.au) |
| 17:08.41 | *** join/#asterisk vonkleist (n=vonkleis@189.155.116.80) |
| 17:09.29 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:11.49 | angryuser | paul0 : 300 ms max, if more people will see the difference |
| 17:17.13 | *** join/#asterisk luckyaba (n=lucky@ip70-177-7-204.sb.sd.cox.net) |
| 17:17.44 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.95.130) |
| 17:25.48 | bpgoldsb | is there a way to specify an optional arguement for a Macro in AEL? |
| 17:26.14 | bpgoldsb | i.e. macro foo ( REQUIRED_ARG, OPTIONAL_ARG ) { ... |
| 17:26.29 | codefreeze-lap | bpgoldsb: no, not really. I guess you leave the arg as a null string, and act accordingly.... |
| 17:26.55 | bpgoldsb | Our old 1.2 dialplan was doing it |
| 17:27.02 | bpgoldsb | I guess I need to rework it in AEL2 |
| 17:27.14 | codefreeze-lap | double checks to make sure he's addressing the right guy.... yep! |
| 17:27.44 | codefreeze-lap | bpgoldsb: you will SO much better like dealing with AEL2... |
| 17:28.44 | *** join/#asterisk ManxPower (n=manxpowe@96.sub-75-249-89.myvzw.com) |
| 17:29.06 | ManxPower | http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=180297472167 (yes, I'm selling this) |
| 17:30.44 | Qwell | echo cancel canceler? |
| 17:30.50 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 17:30.52 | Qwell | so it cancels echo cans? |
| 17:31.36 | [TK]D-Fender | Qwell: Don't not use double negatives like that neither! |
| 17:32.09 | anonymouz666 | ManxPower: hot food! |
| 17:32.36 | ManxPower | Qwell: wanted both words in the title |
| 17:32.49 | *** join/#asterisk sacitec (n=tobi@201.144.211.82) |
| 17:32.51 | voxter | wow thats quite the chassis for one card. |
| 17:33.08 | ManxPower | voxter: smaller chassis are very expensive and hard to find |
| 17:33.31 | ManxPower | The next one will be wired for 4 T-1s w/4 cards |
| 17:33.33 | bpgoldsb | codefreeze-lap: Why is that? |
| 17:34.49 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 17:35.38 | *** join/#asterisk rene- (n=renemend@200.34.66.137) |
| 17:37.42 | *** join/#asterisk CGMChris (n=chris@mail.cgmyes.com) |
| 17:38.05 | *** join/#asterisk zippytech (n=ron@75.149.24.162) |
| 17:38.15 | CGMChris | ManxPower: Still there? |
| 17:38.18 | zippytech | how can i increase the ring amount on an fxo_ks ZAP 1 channel extention |
| 17:38.59 | ManxPower | CGMChris: more or less |
| 17:39.08 | *** join/#asterisk Greek-Boy (n=email@41.221.58.13) |
| 17:39.16 | ManxPower | zippytech: the last field in the Dial line specifies the timeout |
| 17:39.28 | zippytech | cool thanks |
| 17:40.31 | ManxPower | zippytech: spend more time reading the book. |
| 17:40.33 | ManxPower | ~book |
| 17:40.33 | jbot | i heard book is 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 --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 17:42.37 | rwaite | anyone here know off the top of their heads how many sip trunks using g729 a cable internet connection could reliably handle? |
| 17:43.09 | rwaite | i'm expecting like .. 5 maybe 6 concurrently? |
| 17:43.26 | CGMChris | ManxPower: I have reconfigured from the ground up from empty extensions.conf and sip.conf. My question, is even if I have a 2 seperate extensions ring a specific device, I still have to set hasvoicemail = [yes|no] at the device level, don't I ? |
| 17:44.07 | ManxPower | CGMChris: I don't believe Asterisk support a hasvoicemail= option. I'd have to check sip.conf.sample to be sure. |
| 17:44.08 | bpgoldsb | orielly provides the book for free as a pdf? |
| 17:44.09 | bpgoldsb | Neat. |
| 17:44.49 | Qwell | ManxPower: users.conf |
| 17:45.03 | ManxPower | Qwell: Ah. Only GUIs use users.conf |
| 17:45.13 | ManxPower | CGMChris: I cannot help you with users.conf stuff. |
| 17:45.49 | ManxPower | Actually, I can help you with users.conf but I'd need a credit card and a couple of hours of research forst. |
| 17:45.51 | ManxPower | first too. |
| 17:46.25 | *** join/#asterisk matsk (n=Mats@host-90-235-59-179.mobileonline.telia.com) |
| 17:46.41 | [TK]D-Fender | users.conf is a steamy hot pile of manure. |
| 17:46.53 | jaytee | with sprinkles! |
| 17:47.01 | [TK]D-Fender | ...with sprinkles |
| 17:47.10 | Qwell | [TK]D-Fender: You should buy me new tires. |
| 17:47.14 | CGMChris | ManxPower: hasvoicemail = yes, this was in my sample config files that came with asterisk. Is there a mailinglist or room where I can find these so called gui 'developers' ? |
| 17:47.21 | *** join/#asterisk propellerhead (n=yogurt2u@200.41.65.114) |
| 17:47.47 | jaytee | wow |
| 17:48.39 | jaytee | I actually made my own propellerhead beanie with a motor that would make the propeller spin whenever I closed a switch. |
| 17:49.04 | sfire | jaytee: congrats.. you have mastered very basic electronics |
| 17:49.05 | bpgoldsb | I'm trying to do a 'Set(faxdir=/var/spool/asterisk/foo.fax);' in AEL. Worked in 1.2 dialplan, but in 1.6 it complains about the '/' characters. Whats the best way to handle this? |
| 17:49.35 | jaytee | sfire, basic electronics for me was back in 1975 at Keesler AFB in Biloxi, MS. |
| 17:49.41 | sfire | hehehehe |
| 17:50.29 | bpgoldsb | actually, that isn't the problem. wth. |
| 17:51.05 | bpgoldsb | I'm retarded. |
| 17:51.53 | [TK]D-Fender | bpgoldsb: 11 steps to go! |
| 17:52.00 | bpgoldsb | :) |
| 17:52.06 | bpgoldsb | Is one of them drinking? |
| 17:52.09 | bpgoldsb | Cause I need a drink |
| 17:54.22 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:57.40 | *** part/#asterisk matsk (n=Mats@host-90-235-59-179.mobileonline.telia.com) |
| 17:59.44 | rwaite | hmm. now els in this area so we have to get seperate trunks for incoming vs outgoing |
| 17:59.52 | rwaite | s/now/no/ |
| 18:00.02 | rwaite | nifty |
| 18:01.47 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 18:05.57 | adr3nalin3 | Could someone point me in the right direction for getting a line light to light up on a snom phone when someone is parked? |
| 18:06.30 | FinboySlick | For the record, I found the source of my problem. It would still have been a lot easier to see /usr/sbin/sendmail: Permission denied in some sort of log file though. |
| 18:08.50 | ManxPower | FinboySlick: debug did not show it? |
| 18:09.42 | FinboySlick | ManxPower: Well, not 'core set debug 99', I vaguely remember having to re-start asterisk with a certain flag to get full debug however. |
| 18:11.53 | FinboySlick | ManxPower: For further reference, are you familiar with anything of the sort? |
| 18:12.06 | FinboySlick | Or should 'core set verbose' be enough? |
| 18:12.14 | FinboySlick | I mean core set debug |
| 18:12.54 | mvanbaak | FinboySlick: you have to create a hint for the parkspot |
| 18:13.50 | FinboySlick | mvanbaak: Um... Are you sure that's related? |
| 18:14.03 | mvanbaak | sorry |
| 18:14.14 | mvanbaak | off-by-one error |
| 18:14.29 | mvanbaak | adr3nalin3: you have to create a hint for the parkspot |
| 18:14.30 | *** join/#asterisk saftsack (n=saftsack@ip-77-25-229-88.web.vodafone.de) |
| 18:14.31 | mvanbaak | there ;) |
| 18:14.34 | FinboySlick | Hehe, gotta check your pointer increments! |
| 18:18.35 | adr3nalin3 | mvanbaak: ok thank you. |
| 18:19.54 | *** join/#asterisk John_Clay (n=J@unaffiliated/johnclay) |
| 18:20.25 | John_Clay | Just thought I'd pop in and say thanks to jaytee and [TK]D-Fender for your help. I got it working with TrixBox :) |
| 18:20.42 | jaytee | got what working? |
| 18:21.01 | Qwell | so... |
| 18:21.03 | mvanbaak | rm -rf |
| 18:21.04 | Qwell | why are you thanking them? |
| 18:21.13 | John_Clay | cause they helped me out the other day |
| 18:21.13 | Qwell | You clearly ignored any advice they gave you |
| 18:21.19 | John_Clay | answering machine :P |
| 18:21.38 | mvanbaak | TrixBox == evil |
| 18:21.40 | jaytee | sets a reminder in Outlook to sign up for the free Alzhiemer's testing |
| 18:21.49 | John_Clay | hah |
| 18:22.08 | jaytee | silly wabbit, trixbox is for kids |
| 18:22.29 | John_Clay | meh, it works with little fuss. |
| 18:22.47 | John_Clay | and with that, I'm gone. |
| 18:22.47 | *** part/#asterisk John_Clay (n=J@unaffiliated/johnclay) |
| 18:22.48 | Qwell | if you redefine "works" |
| 18:23.42 | mvanbaak | they must be running php_runkit.so |
| 18:23.54 | jaytee | so these assclowns in our education department dumped this video transcoding project on me that they need by 3pm. Why? seems I'm one of the few people that know this trick called "thinking". Problem is it's transcoding at a glacial pace and at 19K frames done and 91700 frames total I don't think they're gonna get what they want in time. |
| 18:25.06 | *** join/#asterisk CrashSys (i=Kumba@azrael.crashsys.com) |
| 18:27.11 | *** join/#asterisk pacmanfan (n=thepacma@12-218-136-3.client.mchsi.com) |
| 18:28.18 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:28.18 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 18:30.19 | hardwire | so if a call comes in from a sip provider.. hits an IVR.. then xfers to a SIP phone.. is there a SIP REDIRECT involved if configured to do so? |
| 18:30.35 | hardwire | I'd love for the IVR machine to not be in the media path any more |
| 18:31.03 | [TK]D-Fender | hardwire: "canreinvite=yes" Only if NAT will not FUBAR you |
| 18:31.26 | hardwire | [TK]D-Fender: so it actually stops audio transmission while they negotiate? |
| 18:34.01 | *** join/#asterisk nosbig (n=nosbig@cpe-75-185-59-24.columbus.res.rr.com) |
| 18:35.20 | *** join/#asterisk gsiener (n=gsiener@209.169.48.66) |
| 18:35.20 | anonymouz666 | WARNING[14711]: chan_sip.c:15234 handle_response: Forbidden - maybe wrong password on authentication for BYE |
| 18:35.24 | anonymouz666 | asterisk 1.6 is dangerous |
| 18:35.56 | FinboySlick | thinks he's going to try and play with chan_mobile :) |
| 18:36.01 | *** join/#asterisk iulius (n=iulius@mail1.technologieshq.com) |
| 18:36.05 | anonymouz666 | chan_mobile is good to waste time |
| 18:36.21 | FinboySlick | It might actually be useful in my case. |
| 18:36.28 | v4mp | guys how would i go about changing this http://v4mpire.pastebin.com/d342ace76 so it aill actually sent the queued calls to the agent group i have found its not SIP/1 so wht would i put there so it calls the agent group ? |
| 18:36.42 | anonymouz666 | until you figure out that can't be used to rely anything on that |
| 18:36.44 | *** join/#asterisk mltlnx (n=mltlnx@cpe-24-193-11-14.nyc.res.rr.com) |
| 18:37.45 | gsiener | Anyone here use Gizmo? Incoming calls were working for a few days but have somehow ceased. I'm using the settings from here http://tinyurl.com/astgiz |
| 18:38.46 | *** join/#asterisk propellerhead (n=yogurt2u@host130.190-226-46.telecom.net.ar) |
| 18:39.17 | hardwire | punches twisted |
| 18:41.53 | *** join/#asterisk Carlos_PHX (n=Carlos@64.129.235.200) |
| 18:43.13 | *** join/#asterisk Carlos_PHX (n=Carlos@64.129.235.200) |
| 18:43.15 | lmadsen | anonymouz666: many of the oldest issues are related to chan_misdn and chan_mobile as I found out yesterday :( |
| 18:43.23 | *** join/#asterisk Firass-VC22 (n=firass@restek-ws-0.vikcomm.wwu.edu) |
| 18:44.06 | *** join/#asterisk pyite (n=pyite@c-76-21-105-195.hsd1.ca.comcast.net) |
| 18:48.50 | sfire | gsiener: I use it |
| 18:49.52 | gsiener | sfire: how should I be troubleshooting? I've got the CLI up, and used to see the sip call coming in. Now when I dial from another gizmo account I see nothing... |
| 18:50.02 | gsiener | sfire: I can make outgoing calls via gizmo |
| 18:50.19 | sky_blue | when dialing into to my meetme conf room entering the pin number followed by # is initiating blind xfer, anybody got any suggestions? |
| 18:50.32 | sfire | gsiener: do you have a router? |
| 18:50.53 | gsiener | sfire: for which piece? |
| 18:51.10 | gsiener | * server has a public static ip |
| 18:51.15 | sfire | gsiener: on your internet connection |
| 18:52.01 | gsiener | I'm trying to connect via my laptop (osx), on the same network as the server |
| 18:52.03 | sky_blue | other than adjusting features.conf..... i've made blind xfer #2 in there |
| 18:52.10 | gsiener | but I am NATed, behind router |
| 18:52.28 | hardwire | nated is a 5 letter word |
| 18:52.28 | sfire | you might have to put in NAT stuff like I had to.. I could make outgoing but couldn't get incoming |
| 18:52.35 | sfire | it was all because of my NAT/Router |
| 18:52.41 | hardwire | sky_blue: get your stuff fixed? |
| 18:53.25 | *** join/#asterisk gsiener (n=gsiener@209.169.48.66) |
| 18:53.30 | sky_blue | hardwire: nearly... very close, it's the # key trying to blind xfer on entering the pin number..... |
| 18:54.00 | hardwire | sky_blue: whats the dialplan like when the sip trunk dials your meetme number |
| 18:54.08 | sky_blue | hardwire: i just can't disable it, i've been playing with the dtmf and features.conf |
| 18:54.10 | hardwire | do you dial in, then dial the meetme extension? |
| 18:54.48 | hardwire | sky_blue: also.. is it initiating a transfer on YOUR pbx.. or on your SIP providers PBX |
| 18:54.49 | sky_blue | no dial in straight to the conf room, may set up more later... |
| 18:55.51 | hardwire | if your sip provider is using asterisk then they may have some Dial flags initiated that are messing with you |
| 18:56.02 | sky_blue | hardwire: that my friend is an excellent question, as i don't have the problem on my sipgate account |
| 18:56.10 | hardwire | afaik Dial is the main app that "initiates" features lilke that |
| 18:56.16 | hardwire | so calling into a meetme shouldn't initiate those features |
| 18:56.31 | hardwire | sky_blue: when in doubt.. blame the ITSP |
| 18:56.44 | hardwire | paypal me.. $200 or so for my services |
| 18:56.45 | gsiener | sfire: sorry for dropping out |
| 18:57.00 | sfire | gsiener: sounds like the same problems I had... PM me |
| 18:57.07 | sfire | too much chatter here |
| 18:57.15 | hardwire | are you calling me fat? |
| 18:58.29 | sky_blue | :-D sounds like you've used ours before, their service is appalling !! |
| 18:58.54 | iulius | We have a fancy fax machine that needs to receive some DTMF digits before processing an incoming fax. Is there a way to Dial() the internal fax machine, but play digits (using senddtmf() I suppose) before connecting to the calling party? |
| 18:59.31 | lmadsen | iulius: core show application dial |
| 18:59.33 | sky_blue | hardwire, that would explain an afwul lot, for example why i wasn't seeing those logs for xfer on my cli |
| 18:59.45 | lmadsen | iulius: I believe the option you want is D() |
| 19:00.54 | *** join/#asterisk Vale-ICS (n=vale@boyne.demon.co.uk) |
| 19:01.38 | iulius | Yeah, that's what I needed. Thanks! |
| 19:01.52 | lmadsen | wonders why we even have documentation :) |
| 19:05.40 | *** join/#asterisk chigital (n=chigital@tmo-117-26.customers.d1-online.com) |
| 19:05.49 | hardwire | sky_blue: I'm gonna LOL if that's the case |
| 19:06.16 | *** join/#asterisk gsiener (n=gsiener@209.169.48.66) |
| 19:08.43 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:09.03 | rwaite | can someone help me here, i'm thinking of going with bandwidth.com for outbound sip trunks and they have a "intra-state rates calling specific to lata" |
| 19:09.16 | rwaite | does this mean that "intra-state" really means inside my lata? |
| 19:09.18 | mvanbaak | hhmm, with the new xml documentation it's very easy to just nuke all the documentation |
| 19:09.34 | hardwire | rwaite: that's the way things are tariffed mang |
| 19:09.45 | hardwire | rwaite: your guvment appreciates yur munny |
| 19:09.51 | rwaite | well i am in ohio and there are a lot of latas here |
| 19:09.57 | mvanbaak | sed -i 's/*** DOCUMENTATION/*** NO_DOCUMENTATION/g' *.c |
| 19:10.24 | rwaite | the intra-state calls are unlimited (the ones specifying the lata) but the inter-state are metered |
| 19:10.24 | hardwire | rwaite: if I were to terminate traffic in Alaska from another rate center in Alaska I have to report that to the authorities that be |
| 19:10.43 | hardwire | they are probably referring to AK, HI |
| 19:10.45 | rwaite | so are they saying all calls terminating outside my lata are going to be metered? |
| 19:10.59 | hardwire | rwaite: did you ask them yet? |
| 19:11.13 | rwaite | not yet, i wanted to ask you guys since you're not motivated to screw me |
| 19:11.15 | rwaite | (i hope) |
| 19:12.59 | Kobaz | hmmm |
| 19:13.49 | Kobaz | i'm not sure what happened here, but i think it's the latest polycom firmware... on my callerid it says "sip:2504@192.168.1.1" instead of just 2504 |
| 19:14.09 | Kobaz | anyone know offhand what the issue could be |
| 19:14.20 | *** join/#asterisk apocn (n=apo@unaffiliated/apocn) |
| 19:15.33 | *** join/#asterisk dklima (n=daniel@border.positivo.com.br) |
| 19:15.36 | *** part/#asterisk dklima (n=daniel@border.positivo.com.br) |
| 19:16.42 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.95.130) |
| 19:16.47 | apocn | Hello all, I'm trying to get the minivm-1.4 using the command svn checkout http://svn.digium.com/view/asterisk/team/oej/minivoicemail-1.4/ but I get the error 301 Moved Permanently. What can I do? |
| 19:16.57 | rwaite | Kobaz: what firmware version |
| 19:18.34 | *** join/#asterisk ew01f (n=chatzill@201.170.36.149) |
| 19:19.12 | sky_blue | hardwire: i'm 100% certain now that is the case, in the uk BT's test number is 17070, you can get a ringback test, quiet line test etc... when i first set this up i dialled 123 (speaking clock) and 17070 just to see what happened. 17070 took me to my ITSP's own version of their test facility... say caller id, echo test, and moh.... now i realise they are using * !!!! Thanks for all you help.. dashing off a curt email to their |
| 19:20.04 | *** join/#asterisk coil (i=coil@unaffiliated/coil) |
| 19:20.21 | Kobaz | rwaite: boot rom 4 |
| 19:20.25 | Kobaz | rwaite: spip 3.2 |
| 19:20.45 | rwaite | spip 3.2 is out? |
| 19:20.48 | rwaite | hot damn |
| 19:20.49 | Kobaz | er no wait, i got that backwares |
| 19:20.53 | Kobaz | i used to have boot rom 3.2 |
| 19:21.05 | Kobaz | i have boot rom 4.1 and spip 3.0.3 |
| 19:21.11 | rwaite | afaik the latest spip is 3.0.3RevB |
| 19:21.18 | rwaite | aight |
| 19:21.22 | Kobaz | yeah that's what i have |
| 19:21.39 | rwaite | ok, do the sip peers have a define cid in sip.conf |
| 19:22.49 | [TK]D-Fender | 3.1.0 |
| 19:22.50 | Kobaz | yeah |
| 19:23.05 | Kobaz | [2509] |
| 19:23.05 | Kobaz | callerid=Conf Room <2509> |
| 19:23.06 | Kobaz | etc |
| 19:23.12 | rwaite | Hmm. |
| 19:23.19 | rwaite | Are all the phones polycom? |
| 19:23.27 | Kobaz | not all |
| 19:23.39 | rwaite | on the non polycom phones, what does the cid show up as |
| 19:23.51 | Kobaz | regular |
| 19:23.55 | Kobaz | like, just the 4 digit exten |
| 19:24.09 | rwaite | okay so only the polycom handsets are showing the wonky cid |
| 19:24.17 | Kobaz | some grandstream, one analog ata, some mixture of different polycom phones |
| 19:24.25 | Kobaz | the polycom 320's show callerid right |
| 19:24.35 | Kobaz | the 501's and 550's dont |
| 19:25.02 | rwaite | hmm. have you looked over the administration guide to see if there are any options? |
| 19:25.18 | Kobaz | i've been digging through the sip.cfg template |
| 19:25.24 | Kobaz | yeah i didn't check that out yet |
| 19:27.06 | *** join/#asterisk shriven (n=shriven@rdu.crosscomm.net) |
| 19:27.16 | *** join/#asterisk `paul (n=paul@125.252.70.126) |
| 19:27.31 | shriven | hello. I am trying to get information from an ldap attribute into my dialplan, does anyone know how I could do this? |
| 19:27.58 | shriven | I would use ldapget but it doesn't compile on 1.6.0. |
| 19:29.28 | [TK]D-Fender | rwaite: It'll show the IP if the other call originated from a different domain or subnet |
| 19:29.44 | coil | hi how do i sip |
| 19:29.52 | `paul | i followed the tutorial on src/doc of asterisk on logging in agents and adding them to queue via VMauthenticate but the problem with VMauthenticate is it hangs up after i entered the password.... so i couldnt add them dynamically to a queue... help with VMauthenticate pls... how can i make it not hangup after entering the password? |
| 19:29.56 | rwaite | [TK]D-Fender: oh really? i did not know that |
| 19:29.59 | shriven | I also want to be able to reference an ldap attribute value in my sip.conf/users.conf |
| 19:30.18 | [TK]D-Fender | coil: We assume you are already capable of breathing, eating and drinking before you show up here. |
| 19:30.39 | Blackvel | shriven: playing around with ldapget. I think it will only work with 1.4. sip.conf/users.conf needs different ldap integration as I read this week |
| 19:31.02 | [TK]D-Fender | `paul: it does not hangup on exiting the app |
| 19:31.10 | shriven | blackvel: hmmm That was my fear, that it would only work with 1.4. : ( |
| 19:31.27 | shriven | does anyone know of another way to do this? |
| 19:31.35 | `paul | hmmmm.... how come i hear a busy tone after pressing #? |
| 19:31.41 | [TK]D-Fender | shriven: AGI <- |
| 19:31.53 | mags2 | anyone know more about the licensing on the freeplay moh files? the license files in the moh directory say digium is allowed to distribute for use with asterisk, but I'd like to be clear on the actual end use of the files. there were some discussions on the listserv and a couple bugs filed, but still no definite answer. |
| 19:31.57 | [TK]D-Fender | `paul: SHOW US. We're only psychic on TUESDAYS |
| 19:32.29 | Blackvel | shriven: what ldap server do you use? openldap? there are several api's out there incl. php/perl |
| 19:32.36 | mags2 | like, I could understand not being allowed to include the files in a commercial asterisk-based 3rd party product, but what about just daily use of them by an institution or company? |
| 19:32.44 | coil | what are good howto sites? |
| 19:34.14 | [TK]D-Fender | mags2: It says you are allowed to use them internally for whatever you want |
| 19:34.31 | [TK]D-Fender | ~book |
| 19:34.31 | jbot | [book] 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 --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 19:34.33 | [TK]D-Fender | coil: ^^^ |
| 19:34.57 | coil | kthx |
| 19:35.19 | mags2 | [TK]D-Fender: ok cool that is what I interpreted but it did not seem clear. |
| 19:35.49 | mags2 | [TK]D-Fender: although by internally, would that include when people call in from outside world and get that moh? |
| 19:35.49 | *** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 19:36.01 | [TK]D-Fender | mags2: Yes |
| 19:36.13 | mags2 | [TK]D-Fender: cool, thank you |
| 19:36.24 | [TK]D-Fender | mags2: That'd be INSANE if you weren't |
| 19:37.00 | `paul | D-Fender: |
| 19:37.02 | `paul | <PROTECTED> |
| 19:37.02 | `paul | <PROTECTED> |
| 19:37.02 | `paul | [Oct 8 19:36:18] NOTICE[27229]: chan_sip.c:15094 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 70 |
| 19:37.02 | `paul | <PROTECTED> |
| 19:37.02 | `paul | <PROTECTED> |
| 19:37.30 | *** join/#asterisk eric2 (n=nobody@sudbury-cable-69-60-228-64.unitz.ca) |
| 19:37.37 | mags2 | [TK]D-Fender: haha ok. I just wanted to be 100% positive and again the language in the files could be more explicit. thanks |
| 19:37.45 | [TK]D-Fender | `paul: PASTEBIN |
| 19:37.54 | shriven | blackvel: it is apple's open directory, which uses openldap. would the mentioned APIs allow me to get that data into the sip users/dialplan somehow? |
| 19:38.22 | [TK]D-Fender | `paul: VMAuthenticate does not hangup. YOU did that with your priority 3 "-- Executing [119@telemed:3] Hangup("SIP/40000-b7669c70", "") in new stack" |
| 19:38.59 | eric2 | what's with <ZOMBIE> appearing in the channel for call logs??? |
| 19:39.16 | Blackvel | shriven: for dialplan for sure |
| 19:39.26 | shriven | hmmm |
| 19:39.35 | Blackvel | shriven: as [TK]D-Fender told with AGI (e.g perl which connects with perl api openldap) |
| 19:39.54 | shriven | hmmm |
| 19:40.10 | Blackvel | shriven: for sip/users I don't know, will depend on the module you use |
| 19:40.34 | shriven | blackvel: what do you mean by which module? |
| 19:40.37 | `paul | hmmm |
| 19:40.40 | Blackvel | the ldap module |
| 19:40.46 | Blackvel | its a different one than ldapget |
| 19:40.56 | tvirus | Is it possible to only add SIP connections from a specific IP to a group to check against trunk usage? The reason I ask is because our SIP phones appear as a channel and are added to the group and it messes up when it dials agents in a queue. |
| 19:41.12 | `paul | D-fender: i put NoOp(test); right after VMauthenticate but it doesnt show |
| 19:42.25 | shriven | blackvel, [TK]D-Fender: ok I guess I'll have to go read up on AGI. Thanks for your assistance. |
| 19:43.12 | *** join/#asterisk pluesch0r (n=pluesch0@vie-078-142-131-201.dsl.sil.at) |
| 19:43.35 | Blackvel | shriven: isn't http://www.voip-info.org/wiki/view/LDAP and ldap realtime the thing you want to go for sip.conf, users, etc.? |
| 19:43.41 | pluesch0r | evening guys ... sip-wise .. what do i need to configure if one of the client ips has no PTR record? |
| 19:43.54 | Blackvel | and agi/perl api for extensions |
| 19:44.05 | Blackvel | seems its beta with 1.6 |
| 19:44.33 | pluesch0r | i'm getting 488 not acceptable here errors on the client side, process_sdp: Unable to lookup host in c= line, 'IN IP4 xxx.xxx.xxx.xxx' errors on the server. |
| 19:44.38 | Blackvel | shriven: btw...do you know any tool which integrates outlook 2003 out of the box for export with openldap? |
| 19:44.42 | shriven | blackvel: unfortunately no, that is for storing the entire config file in ldap. I only want to pull like 4 attributes from my ldap directory into the existing sip.conf etc |
| 19:45.13 | Blackvel | and that is possible with any ldap module for asterisk? |
| 19:45.29 | *** join/#asterisk rasterix (n=IceChat7@80.177.176.254) |
| 19:45.32 | shriven | ldapget did that for dialplans for sure |
| 19:45.39 | Blackvel | yes, dailplans |
| 19:45.43 | shriven | I have not yet discovered if I can do that in the other conf files |
| 19:45.44 | Blackvel | but not sip.conf |
| 19:46.00 | Blackvel | how should that work? ldapget is an application |
| 19:46.06 | shriven | right, that's what I really want. I want to specify my softphones based on ldap |
| 19:46.11 | shriven | well |
| 19:46.41 | *** part/#asterisk `paul (n=paul@125.252.70.126) |
| 19:47.29 | shriven | the way I envision it working is similar to how the extensions.conf file works... something like callerid=ldap_app(dc=fullname,dc=user,dc=people,dc=domain,dc=com |
| 19:47.31 | shriven | ) |
| 19:47.33 | shriven | something like that |
| 19:47.50 | rasterix | hi does anyone have any experience of call divert on an isdn30 (British Telecom)? I spoke to our account manager today who advised it would not be possible to divert since we only have 14 of the 30 channels enabled the caller will simply here engaged |
| 19:47.59 | shriven | where the value looked up would be used as the value for that setting |
| 19:48.12 | rasterix | hear* |
| 19:48.16 | shriven | blackvel: what do you mean by integrating outlook 2003 for export? |
| 19:49.13 | *** join/#asterisk Knightfal (n=jjj@66.178.134.235) |
| 19:49.17 | pluesch0r | Blackvel: sorry if i'm completely off path, FYI .. outlook 2003 seems to have huge problems fetching data off of slapd. |
| 19:49.26 | pluesch0r | i only managed to tie slapd to outlook express. |
| 19:49.56 | pluesch0r | outlook seems to make some freak-query that isn't compatible with slapd. |
| 19:50.05 | Blackvel | want to export >500 contacts from outlook to openldap |
| 19:50.14 | Knightfal | Guys I keep getting -- Channel 0/1, span 3 got hangup request, cause 102 |
| 19:50.20 | Blackvel | but i am not convinced that I can get it working with csv2ldif |
| 19:50.31 | Knightfal | I read about it a bit but am not quite sure how to resolve it |
| 19:50.31 | Blackvel | just want to run it inside outlook as an plugin |
| 19:50.35 | pluesch0r | anybody able to help me with the reverse lookup stuff? |
| 19:50.39 | Blackvel | hit export to openldap, and that's it |
| 19:51.04 | shriven | blackvel: seems a bit off topic here, maybe you should ask that in an ldap channel? But no, I don't know if there is an easy way to do that. : ( |
| 19:52.51 | *** join/#asterisk moy (n=moy@nat/ibm/x-9fb1c7d5699866ea) |
| 19:53.49 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 19:55.56 | apocn | Is there a way to automatically calling a person (SIP) and when this person answers then call an agent and establish the communication? |
| 19:56.29 | Knightfal | also my CLI shows that asterisk is constantly Parsing '/etc/asterisk/manager.conf Please help |
| 19:56.30 | apocn | I've tried using the originate command, and it works from extension to extension (locally) but not when I use SIP |
| 19:56.48 | lmadsen | apocn: yes -- use a callfile or the Asterisk Manager Interface |
| 19:57.27 | apocn | I've tried both, but it works in the other direction (first the agent, then the client) and if I specify the opposite, both are called at the same time. |
| 19:57.49 | apocn | I only want the agent's phone to ring when the person already answered. isnt it possible? |
| 19:58.42 | *** join/#asterisk waverly360 (n=waverly@ns2.dalcon.com) |
| 19:59.27 | smth | [TK]D-Fender, still about inband dtmf. after reprodued the issue ,i put the configuration and console message at http://pastebin.com/m3b900864 and I did not find the 'match issue' between incomming call and a peer setting as you mentioned before. but inband dtmf was still working. |
| 19:59.42 | waverly360 | Hey guys, can timing affect call parking? Or perhaps a better question is, can a pri with timing issues cause asterisk to start losing it's mind? |
| 20:00.08 | smth | [TK]D-Fender, sorry inband dtmf was still not working |
| 20:08.23 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:08.47 | bpgoldsb | is there any disadvantage to doing 'goto foo|${BAR}|1;' versus '&foo(bar);' |
| 20:09.00 | bpgoldsb | (using AEL/AEL2) |
| 20:10.24 | [TK]D-Fender | smth: What is that peer connecting to? |
| 20:11.47 | eric2 | what's the proper way to detect a hangup on a zap channel as I need to execute a shell script upon doing so...? |
| 20:13.06 | codefreeze-lap | bpgoldsb: interesting... to do &foo, you'd have to define a macro called foo, etc. I wouldn't call a macro if you never plan on returning... otherwise, ... whatever's best... |
| 20:13.48 | bpgoldsb | codefreeze-lap: I guess I'm trying to clarify when a macro is appropriate and when a context works best. Out old code had superfluous use of macros. |
| 20:14.31 | bpgoldsb | moved most stuff to goto+context, but I'm now realizing I should still be able to use goto+context when there's arguements. Before, I immediately jumped to using a macro if I had an arguement |
| 20:16.51 | *** join/#asterisk lasko (n=chatzill@70.102.15.210) |
| 20:17.09 | *** part/#asterisk lasko (n=chatzill@70.102.15.210) |
| 20:17.52 | *** join/#asterisk mateo_au (n=chatzill@c122-106-221-182.belrs3.nsw.optusnet.com.au) |
| 20:17.54 | codefreeze-lap | bpgoldsb: macros are for sets of dialplan you'd like to use multiple times, from different spots... It'd be bad form to implement such via goto's. Goto's are good for... well... there's lots of good reasons for using them, and lots of bad ones, too. |
| 20:18.23 | bpgoldsb | I fear my uses are bad ones, then |
| 20:18.31 | pluesch0r | nobody able to help me with my reverse lookup pita? :) |
| 20:23.15 | codefreeze-lap | bpgoldsb: uh, not seeing your code, but seeing stuff like "goto with arguments" -- I'd say, keep using the macros. Whatever you do, ask yourself, will this make it easier to understand? to read? to maintain? If I die, will my replacement speak ill of the dead? |
| 20:23.59 | bpgoldsb | Thanks for the advice. I'm currently speaking ill of the dead. |
| 20:24.53 | pluesch0r | codefreeze-lap: then again .. why should i give a fuck when i'm dead? :) |
| 20:25.18 | pluesch0r | i mean .. i won't even be able to give a fuck .. since i'm dead. ;) |
| 20:26.25 | [TK]D-Fender | ok, checkouyt time. Back later |
| 20:26.36 | codefreeze-lap | pluesch0r: Forget death, then. The important thing to ask yourself, then, is what if I don't die, and have to maintain this stuff for the rest of my life? |
| 20:26.58 | pluesch0r | bummer. |
| 20:27.14 | ManxPower | codefreeze-lap: what bpgoldsb seems to not understand is that the warning he is getting is a bug and Gosub is not actually going away |
| 20:27.50 | ManxPower | He should have figured this out when he read the mailing list messages I referred him to this morning |
| 20:28.24 | bpgoldsb | ManxPower: No, thats the the issue. |
| 20:28.33 | bpgoldsb | I've gotten those fixed. |
| 20:28.49 | ManxPower | bpgoldsb: you've never programmed before have you? |
| 20:29.13 | bpgoldsb | Not beyond 300-400 line python scripts |
| 20:29.36 | ManxPower | I would have thought you would have understood goto and gosub then. |
| 20:30.19 | ManxPower | bpgoldsb: AEL/AEL2 compiles your code into standard dialplan stuff and then runs that. AEL/AEL2 is not parsed at call time, only load time when it is converted. |
| 20:30.35 | *** join/#asterisk icel (n=dan@75.150.16.110) |
| 20:30.52 | bpgoldsb | I'm trying to wrap my head around 900ish lines of dialplan code having never worked with asterisk in a real capacity. I have a lot of general confusion :) |
| 20:31.05 | bpgoldsb | and by wrap my head around, I mean wrap my head around and port to AEL |
| 20:31.21 | ManxPower | bpgoldsb: learn asterisk first |
| 20:32.03 | bpgoldsb | I learn best by example :( |
| 20:33.18 | icel | is there any reason * configs (zaptel.conf/zapata.conf) that work with a digium TE212P wouldn't work with a TE405P? |
| 20:33.35 | *** join/#asterisk mog (n=mog@nat/digium/x-b7817a99c416335b) |
| 20:33.35 | *** mode/#asterisk [+o mog] by ChanServ |
| 20:33.43 | codefreeze-lap | bpgoldsb: Well, then the next few days/weeks should be educational. Read, search, and test things out, and you'll quickly come up to speed. And when you hit a wall, someone around here hopefully can fill in the missing facts. |
| 20:33.49 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 20:33.51 | *** join/#asterisk l2trace99 (n=jr@75.112.133.235) |
| 20:34.07 | bpgoldsb | Thats what I'm doing :) |
| 20:34.16 | smth | <PROTECTED> |
| 20:35.03 | ManxPower | icel: none |
| 20:35.37 | ManxPower | zaptel.conf is not an Asterisk config file. |
| 20:35.37 | icel | ManxPower: hmm, not what I wanted to hear since mine isn't working! |
| 20:35.37 | icel | true |
| 20:35.47 | ManxPower | icel: my guess is you are loading the wrong kernel driver |
| 20:36.00 | icel | ManxPower: same driver isnt it? wct4xxp? |
| 20:36.14 | ManxPower | icel: read the zaptel README |
| 20:36.20 | icel | ManxPower: yessir |
| 20:36.41 | ManxPower | It tells you exactly what card requires which driver |
| 20:36.52 | smth | <PROTECTED> |
| 20:37.06 | icel | ManxPower: Same driver |
| 20:37.13 | ManxPower | icel: then you solved that question |
| 20:37.30 | ManxPower | does ztcfg -vvv give you any errors? |
| 20:38.18 | ManxPower | What specific error/problem are you experiencing? |
| 20:39.45 | icel | one sec |
| 20:44.01 | *** join/#asterisk Kate-o (n=vajda1ka@unaffiliated/kateo/x-334924) |
| 20:45.29 | *** join/#asterisk nicox (n=nicox@212-183-42-113.adsl.highway.telekom.at) |
| 20:48.29 | *** join/#asterisk spokra (n=spokra@blockhead.sea0.speakeasy.net) |
| 20:50.10 | *** join/#asterisk angryuser (n=Miranda@51.210.20.81.dynamic.adsl.abo.nordnet.fr) |
| 20:51.58 | icel | ManxPower: Thanks for the help, I somehow managed to solve my problem...fancy that |
| 20:52.48 | *** join/#asterisk tali81 (i=434e2716@gateway/web/ajax/mibbit.com/x-2df063b0309b144b) |
| 20:53.30 | hardwire | nominates voipsupply.com as the worlds worst best redesign. |
| 20:53.54 | tali81 | What could be causing intermetiant scratchy noise when people call into my asterisk system and hit an ivr. The static is none on some calls but on those you cant even hear the prompts thru it, and there is now QoS limits blocking the trunking speeds |
| 20:55.01 | tali81 | i mean there is no not now |
| 20:56.01 | hardwire | sky_blue: so what was the problem? |
| 20:56.30 | rwaite | anyone here had any experiences with broadvoice.com, good or bad? |
| 20:56.55 | hardwire | http://www.voipsupply.com/1pfail |
| 20:56.55 | hardwire | niec |
| 20:58.35 | smth | hardwire, any idea about that inband dtmf does work on incoming call of asterisk. see http://pastebin.com/m26550730 |
| 20:58.43 | l2trace99 | rwaite: I used them some time ago, It was ok , but I had issues with faxes. I don't think they supported T38 at the time |
| 20:59.20 | hardwire | smth: explain incoming. |
| 20:59.58 | smth | inbound |
| 21:00.07 | hardwire | explain inbound. |
| 21:00.33 | smth | calls to asterisk |
| 21:00.41 | hardwire | via? |
| 21:01.02 | smth | carrier |
| 21:01.16 | *** join/#asterisk emist (n=emist@unaffiliated/emist) |
| 21:01.37 | hardwire | smth: calls inbound to your asterisk box over what technology? |
| 21:02.14 | smth | hardwire, can you see the link i pasted. |
| 21:02.20 | hardwire | indeed |
| 21:02.54 | hardwire | smth: so calls coming in via sip? |
| 21:03.11 | rwaite | l2trace99: other than that, pretty reliable? |
| 21:03.34 | hardwire | smth: does les.net support inband DTMF on their trunks? |
| 21:03.45 | jmacz | Hi everyone, I'm having some random crashes with * 1.2.26.2 due to segfaults and have just gotten a core_dump file |
| 21:04.00 | smth | should be. |
| 21:04.01 | hardwire | I'm guessing they don't.. have you called their support? |
| 21:04.10 | hardwire | 1-888-399-VOIP |
| 21:04.45 | *** join/#asterisk NirS (n=NirS@80.250.159.240) |
| 21:04.53 | smth | thx. |
| 21:05.05 | hardwire | np. |
| 21:05.45 | jmacz | My question is if may I run gdb using the asterisk binary (/usr/sbin/asterisk) in other box rathen than the one with the segfault problem |
| 21:05.51 | jmacz | ? |
| 21:07.17 | jmacz | (gdb -c /tmp/core.asterisk asterisk-binary-in-other-box) |
| 21:07.46 | *** join/#asterisk Danskmand (n=danskman@p4FD3D80A.dip.t-dialin.net) |
| 21:07.59 | bkw_ | jmacz: you need to run it with the binary that the core came from |
| 21:08.01 | l2trace99 | rwaite: i wouldn't put a hospital behind it , but yes |
| 21:08.03 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 21:08.06 | hardwire | jmacz: everything would be off. |
| 21:08.35 | hardwire | rwaite: limited channels per device.. kinda difficult to use IMHO |
| 21:08.39 | hardwire | I like teliax pay as you go |
| 21:08.49 | hardwire | they continually impress me |
| 21:08.49 | *** join/#asterisk gio_ita (n=nobody@host218-200-static.81-94-b.business.telecomitalia.it) |
| 21:09.01 | l2trace99 | anyone know how a can check if a channel is in use before calling chanspy on it ? |
| 21:09.27 | rwaite | core show channels? |
| 21:09.38 | l2trace99 | in the the dialplan |
| 21:09.44 | rwaite | i like bandwidth.com so far, but inter-state is expensive |
| 21:10.11 | rwaite | perl agi script to run asterisk -rx 'core show channels' and parse the output? |
| 21:10.21 | hardwire | rwaite: only 2 concurrent channels at a time with most broadvoice packages. |
| 21:10.37 | rwaite | hardwire: hmm. that's unacceptable for my business. |
| 21:10.38 | jmacz | bkw_, I'm not very used to gdb, and didn't want to mess with a production System, so I copied the asterisk-binary to may ws and processed it, It's this ok? |
| 21:10.52 | *** join/#asterisk spokra (n=spokra@blockhead.sea0.speakeasy.net) |
| 21:10.55 | hardwire | rwaite: orange you glad it wasn't a banana? |
| 21:11.20 | rwaite | sometimes i wish we could just buy a few more analog lines and another tdm card and be done with it |
| 21:11.21 | gio_ita | Hi can anyone help me with console/dsp. Operatore answer incoming call, put it on hold, dial intercom an next need to dial some extension and join caller with this extension but frequently, due some operation error, caller is bridge on the Console/dsp because the channel remain in hold |
| 21:11.30 | hardwire | rwaite: the devil uses POTS |
| 21:11.32 | rwaite | the only tricky thing is long distance |
| 21:11.54 | rwaite | hardwire: but the sip trunk situation is such a pita |
| 21:12.06 | hardwire | rwaite: check out a few more providers |
| 21:12.09 | rwaite | its either that or a pri, and thats why we started this to begin with, to get away from the pri |
| 21:12.11 | hardwire | there are some diamonds in the rough. |
| 21:12.25 | hardwire | rwaite: where are you? |
| 21:12.28 | jmacz | bkw_, the binary is the same one that runs in the box with problems but I'm using it locally in my pc |
| 21:12.34 | rwaite | hardwire: any suggestions? im looking over the list on voip-info and most are horrible from the looks of it |
| 21:12.39 | hardwire | teliax.com |
| 21:12.39 | rwaite | akron, ohio |
| 21:12.46 | hardwire | pay-as-you-go |
| 21:12.58 | hardwire | or you can buy extra channels with the business plans |
| 21:13.02 | hardwire | I don't work for em |
| 21:13.05 | nicox | where do you need a provider? |
| 21:13.08 | hardwire | I just really appreciate them/ |
| 21:13.25 | rwaite | im looking for unlimited |
| 21:13.57 | nicox | outgoing or incoming calls? and where? |
| 21:14.14 | rwaite | outgoing only in akron, ohio |
| 21:14.14 | hardwire | rwaite: pay-as-you-go is unlimited |
| 21:14.24 | hardwire | err.. near infinite :) |
| 21:14.28 | jmacz | hardwire, what do you mean with it'd be off? |
| 21:14.30 | rwaite | flat-rate unlimited |
| 21:14.44 | rwaite | that's what we have now, but they are flaky |
| 21:14.44 | hardwire | jmacz: the binary has functions in specific places in it's compiled form |
| 21:14.56 | hardwire | gdb uses the core and the binary to make sense of it all |
| 21:15.01 | hardwire | they are complimentary. |
| 21:15.09 | hardwire | if the binaries are identical, go for it. |
| 21:15.17 | hardwire | but you may fudge up some library references |
| 21:15.31 | hardwire | rwaite: w/ who? |
| 21:15.45 | rwaite | this tc systems/vip voip place |
| 21:16.26 | jmacz | hardwire, that's right, I saw a lot of warnings but gdb provided a valid (afaict) output anyway |
| 21:16.47 | hardwire | jmacz: so whats the question? |
| 21:16.54 | jmacz | hardwire, so that's why I was hesitating on trusting that |
| 21:17.02 | hardwire | jmacz: I wouldn't trust it :) |
| 21:17.03 | *** join/#asterisk mchou (n=mchou@unaffiliated/mchou) |
| 21:17.29 | jmacz | hardwire, if it was OK to copy an asterisk-binary tu run gdb with that binary and a core file ni my PC (not in my server) |
| 21:18.24 | hardwire | you just don't want to install gdb on the server? |
| 21:18.26 | jmacz | hardwire, actually, the question should be: which are the implications of debugging a core_dump on a running system? May it cause a crash? |
| 21:18.40 | hardwire | no |
| 21:18.44 | hardwire | it doesn't execute anything |
| 21:18.48 | jmacz | hardwire, I don't want to cause another segfault just by debugging (kind of paranoid here) |
| 21:18.57 | hardwire | it doesn't do anything |
| 21:19.09 | jmacz | hardwire, thank you very much |
| 21:19.31 | jmacz | guess I have a lot to learn about gdb |
| 21:19.37 | hardwire | me too |
| 21:19.42 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 21:19.54 | bkw_ | jmacz: you won't segfault running gdb |
| 21:19.55 | *** join/#asterisk propellerhead (n=yogurt2u@host209.200-82-99.telecom.net.ar) |
| 21:20.11 | hardwire | http://www.sangoma.com/products_and_solutions/hardware/netborder_express_gateway_cards.html |
| 21:20.15 | hardwire | that's the weirdest thing ever. |
| 21:20.24 | jmacz | bkw_, got it. Thank you for answering :-) |
| 21:20.27 | ManxPower | People sure are proud of 25 pair amphenol cables |
| 21:21.29 | ManxPower | I can find them retail cheaper than on eBay |
| 21:22.20 | hardwire | ManxPower: I totally thought about using some of those the other day |
| 21:22.38 | ManxPower | hardwire: I have some tellabs chassis with those types of connectors on it. |
| 21:22.46 | hardwire | ManxPower: yar |
| 21:23.00 | hardwire | I was gonna use it on a breakout for r-45 (2 pair) |
| 21:23.02 | hardwire | j |
| 21:23.09 | hardwire | simply because.. I didn't want to deal |
| 21:23.28 | hardwire | but I ended up running several pairs of cat5e and just doing it right :) |
| 21:25.37 | ManxPower | *nod* Cat3 (cable and RJ connectors/jacks) are fine for T-1s |
| 21:28.20 | *** part/#asterisk nicox (n=nicox@212-183-42-113.adsl.highway.telekom.at) |
| 21:30.10 | bpgoldsb | in the 1.2 dialplan, you could specify extensions inside a macro. That can't happen in AEL2, can it? |
| 21:30.27 | hardwire | ManxPower: concidering how they are brought into the building.. I'd think a few isolated water streams would work well enough as conductors for a T-1 |
| 21:32.15 | bpgoldsb | Basically I'm trying to port this macro to AEL: http://pastebin.com/m4d0a33c4 |
| 21:34.46 | smth | hardwire, les.net pretty sure inband dtmf works. so I just have no idea about why it does not work with my asterisk. |
| 21:35.19 | hardwire | smth: did you tell them it works fine with other sip providers? |
| 21:35.21 | hardwire | I mean srsly. |
| 21:35.34 | hardwire | err |
| 21:35.35 | hardwire | wait |
| 21:35.46 | hardwire | I'm replying to the wrong person :) |
| 21:35.51 | codefreeze-lap | bpgoldsb: you can use the "catch" statement for extens in macros... |
| 21:36.05 | hardwire | smth: they are only pretty sure? |
| 21:36.51 | smth | yeah |
| 21:38.06 | *** join/#asterisk pcrane (n=pcrane@202.49.106.158) |
| 21:38.07 | hardwire | what other dtmf options have you tried? |
| 21:39.01 | smth | actually i ever used different dids which carried by different providers. the inband dtmf dons not work at inbond call to asterisk. |
| 21:39.27 | bpgoldsb | codefreeze-lap: so 'catch o { foo }'? |
| 21:39.59 | smth | so I could not doubt the carrier at first. something needs to be figured out on asterisk I think. |
| 21:40.17 | hardwire | smth: in your pastebin |
| 21:40.22 | hardwire | you have dtmf=inband |
| 21:40.38 | hardwire | I assumed you knew that. |
| 21:41.19 | codefreeze-lap | bpgoldsb: hmmm, looking at that macro.... When you jump out of a macro, you'll never get there. The macro terminates and returns instead. Gosubs behave different, because there's no interpreter running on top of a Gosub, like there is with macros... and as to the catch o {foo}, yes, that should work... |
| 21:42.10 | bpgoldsb | codefreeze-lap: You're my bestest friend. |
| 21:42.32 | smth | anyidea about how detect the inband dtmf since you can not do it by using capturetool like wireshark? |
| 21:42.42 | codefreeze-lap | bpgoldsb: you're welcome! |
| 21:45.08 | hardwire | smth: have you tried setting dtmf in your sip.conf to anything other than inband? |
| 21:45.13 | *** join/#asterisk NirS (n=NirS@80.250.159.240) |
| 21:45.17 | hardwire | cause right now its set to only process inband DTMF |
| 21:45.34 | *** join/#asterisk beek (n=klinebl@65.211.106.242) |
| 21:46.34 | codefreeze-lap | bpgoldsb: In all our conversations, there is a chance for some confusion. In AEL, there is a macro definition and call. AEL compiles into extensions.conf sort of format, sort of, but it's really just in-mem structs. In extensions.conf, you can call the macro app, and the gosub app, etc. AEL used to compile macro defs and calls into calls to Macro(), etc. Now it compiles into Gosub() app calls. |
| 21:47.04 | smth | yeah, I did. rfc2833 /info both work . |
| 21:47.19 | hardwire | smth: if they work.. is there a problem? |
| 21:47.24 | bpgoldsb | So when I'm calling '&foo(bar);' it's really calling GoSub, not Macro? |
| 21:47.33 | bpgoldsb | Is that what you're trying to explain to me? |
| 21:47.45 | codefreeze-lap | bpgoldsb: so, don't get the AEL macro def/call mixed up with the extensions.conf Macro() app. |
| 21:48.05 | smth | problem is I need inband dtmf work too but it does not. ;) |
| 21:48.15 | *** join/#asterisk Tako-san (n=Tako-san@24.108.192.144) |
| 21:48.29 | hardwire | smth: why? |
| 21:48.51 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 21:48.51 | hardwire | smth: it's probably out of your control fooberry. |
| 21:49.44 | hardwire | smth: if les.net strips DTMF when it hits their PRI then sends it out of band.. or receives out of band DTMF and never generates a tone for you, then you will never have an INBAND tone to work with. |
| 21:49.55 | *** join/#asterisk Linuturk (n=Linuturk@fluxbuntu/developer/Linuturk) |
| 21:50.10 | hardwire | smth: all I'm saying is you probably don't have DTMF on your audio. |
| 21:50.15 | hardwire | even if you try to force it. |
| 21:50.17 | smth | hardwire, I should know if asterisk really work with inband dtmf at sip channel. some customer may use inband dtmf. |
| 21:50.41 | hardwire | smth: both endpoints need to support it correctly. on all networks they would ever negotiate with. |
| 21:50.50 | *** join/#asterisk Nashe (n=ehsan@TOROON12-1176044102.sdsl.bell.ca) |
| 21:50.56 | *** join/#asterisk MikeJ (n=MikeJ@freeswitch/developer/mikej) |
| 21:51.06 | hardwire | if you're trying to record DTMF, turn DTMF debugging on in asterisk and rely on the logs |
| 21:51.11 | smth | hardwire, I can hear the dial tone .is it mean inband dtmf were in the media. |
| 21:51.13 | hardwire | like.. record/monitor/archive |
| 21:51.39 | hardwire | smth: possibly. afaik asterisk handles inband dtmf in sip just fine |
| 21:51.45 | hardwire | hence why there is an option for it at all. |
| 21:51.52 | hardwire | what country are you in? |
| 21:52.01 | smth | canada |
| 21:52.04 | smth | toronto |
| 21:52.06 | hardwire | that's your problem right there. |
| 21:52.09 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 21:52.31 | hardwire | points and lols @ smth.. |
| 21:52.40 | MikeJ | in canada they put an a' after each dtmf digit ? |
| 21:52.51 | hardwire | exactly. |
| 21:53.00 | voxter | my extension is 102 eh. |
| 21:53.08 | *** join/#asterisk mahlon (i=mahlon@martini.nu) |
| 21:53.16 | voxter | (im allowed to make fun, im canadian too) |
| 21:53.20 | hardwire | did you set canuck=true? |
| 21:53.27 | smth | hardwire,can you explain more detail? |
| 21:53.36 | hardwire | smth: no.. probably not. |
| 21:53.39 | voxter | this guy must be french. |
| 21:53.41 | hardwire | I'd have a sit down with the provider. |
| 21:54.03 | MikeJ | I was joking btw |
| 21:54.08 | hardwire | lies |
| 21:54.20 | MikeJ | I'm 1/2 canadian.. I'm allowed |
| 21:54.40 | smth | dont do that so far until problem got solved .;) |
| 21:54.42 | hardwire | how does a 1/2 canadian say "about"? |
| 21:54.48 | eric2 | aboot |
| 21:54.51 | zerko | aboot |
| 21:54.55 | zerko | yeah, lol |
| 21:54.57 | MikeJ | depends where I am |
| 21:55.09 | eric2 | but that's not really the way we say it |
| 21:55.12 | hardwire | and where they buried the survivors. |
| 21:55.13 | smth | where are guys |
| 21:55.15 | jer | it's not aboot, it's aboat -- i know, I'm 100% Canadian |
| 21:55.16 | zerko | I had to say it one time before I typed it |
| 21:55.17 | eric2 | that's more british |
| 21:55.26 | voxter | eastern canadians say aboat way more. |
| 21:55.30 | jer | silly US-ians can't hear the Canadian raising and think it's aboot or aboat |
| 21:55.39 | zerko | aboat sounds more like it, yes |
| 21:55.40 | jer | just like they can't hear the difference between rider and writer |
| 21:55.46 | hardwire | smth: Alaska |
| 21:55.51 | hardwire | I'm your neighbor.. sorta. |
| 21:56.02 | MikeJ | no one can hear the difference in these lame narrowband codecs |
| 21:56.07 | voxter | jer: you should hear some people from far east say bar or car. ha! |
| 21:56.08 | MikeJ | thats what wideband is for |
| 21:56.08 | jer | habité aux l'Ontario |
| 21:56.10 | hardwire | smth: pump up the debugging and see what it says |
| 21:56.14 | jer | MikeJ, lol |
| 21:56.17 | Nashe | Hi All, I am having this problem with my newly installed "Elastix" I am using two Trunks for outbound 1. through SIP proxy which |
| 21:56.18 | jer | true =] |
| 21:56.32 | jer | voxter, stay where yer at, i'll come where ya too |
| 21:56.42 | jer | voxter, i'm quite familiar with newfoundland english anyway |
| 21:56.47 | hardwire | smth: btw.. what codec? |
| 21:56.47 | voxter | haha |
| 21:56.57 | smth | a/u law |
| 21:57.10 | hardwire | smth: ah |
| 21:57.11 | jer | newfoundland english, acadian french, québécois, and canadian english ... all very interesting |
| 21:57.15 | *** join/#asterisk Yourname` (i=Yourname@unaffiliated/yourname/x-837320) |
| 21:57.23 | hardwire | smth: I don't think I've ever used a sip trunk that offered out of band and in band at once. |
| 21:57.35 | hardwire | somebody here should call me crazy, or back me up. |
| 21:57.36 | hardwire | that would be nice. |
| 21:57.49 | ManxPower | There is no such thing as a "sip trunk" |
| 21:58.02 | ManxPower | So yes, you are crazy. |
| 21:58.05 | Yourname` | Hi, there's an AMI user who basically wants to test queue member login/logout. How can I add him in such a way that he doesn't get inundated with AMI spits out that he doesn't really need? |
| 21:59.01 | jer | ManxPower, sure there is, i have all my sip traffic in one vlan trunked out a specific switch port... that's my "sip trunk" =] |
| 21:59.17 | Nashe | Hi All, I am having this problem with my newly installed "Elastix" I am using two Trunks for outbound 1. through SIP proxy which registers to a SIP server.. 2. send sip calls to CISCO AS5400 (without registeration) to PSTN.... first trunk work perfectly but the second one faces One Way Audio problem... Elastix -> Cisco voice works while Cisco -> Elastix does not. My Elastix box is behind NAT where I am usin |
| 21:59.30 | smth | you know what , when you create the sip account of les.net , you can choose what dtmf format you like to use. so the account will be work on only one dtmf way. |
| 21:59.36 | *** join/#asterisk shinao1 (n=shinao1@83.229.85.138) |
| 22:00.48 | ManxPower | jer: that would be an ethernet trunk |
| 22:00.57 | jer | ManxPower, i was being facetious |
| 22:01.05 | jer | and it's actually a vlan trunk |
| 22:01.35 | ManxPower | if anyone has recommendations for purchasing serial cables and PBX amphenol cables online please /msg me. |
| 22:01.51 | Nashe | I am having this problem with my newly installed "Elastix" I am using two Trunks for outbound 1. through SIP proxy which registers to a SIP server.. 2. send sip calls to CISCO AS5400 (without registeration) to PSTN.... first trunk work perfectly but the second one faces One Way Audio problem... Elastix -> Cisco voice works while Cisco -> Elastix does not. My Elastix box is behind NAT where I am using DMZ op |
| 22:02.36 | ManxPower | Nashe: you would have better luck on the forum for your asterisk distro |
| 22:03.17 | MikeJ | ManxPower: did you turn off the ec in here? |
| 22:03.56 | ManxPower | MikeJ: no, Nashe is just an asshole for posting the same thing twice in a short time |
| 22:04.04 | Nashe | I did not find any help about this on any forms... thats why I am here... After all Asterisk is what is handling Asterisk calls |
| 22:04.52 | ManxPower | Nashe: that is like going to a redhat channel asking debian questions -- it's all linux afterall. |
| 22:05.31 | Nashe | ManxPower: thank you for your compliments and great hospitality to a new user.... If you have noticed I added some more information to what I said... beside... when no one answers... then you might repeat what you said assuming that no one listened to that |
| 22:05.55 | ManxPower | Nashe: Best of luck. |
| 22:07.20 | Nashe | OK let me re-word it... In my Asterisk Box... I am having OneWay Audio problem when calling to/from CISCO Gateway, however calls through SIP trunk works fine... My Asterisk Box is behind NAT |
| 22:07.57 | MikeJ | externip? |
| 22:08.56 | Yourname` | Ok, so since no one knows that question. How can I tell via AMI if an agent is logged in or not or what his status is? QueueStatus? |
| 22:09.44 | ManxPower | Nashe: Good. You should read the !sipnat document and set your externip= and localnet= and set canreinvite=no in sip.conf. |
| 22:09.46 | ManxPower | ~sipnat |
| 22:09.46 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 22:10.31 | Nashe | i already set canreinvite to NO |
| 22:10.34 | ManxPower | The wiki page on NAT solutions is practically useless |
| 22:10.49 | ManxPower | Nashe: that's not going to do any good without the other two options properly set |
| 22:11.22 | ManxPower | Nashe: is your cisco doing the NAT? |
| 22:11.34 | angryuser | ManxPower : wiki page is fine, worked for me at least phone---nat------nat-----box |
| 22:11.39 | Nashe | so these two again are sip.conf parameters? |
| 22:11.55 | Nashe | No CISCO is not doing NAT... CISCO is on a PUBLIC IP |
| 22:12.01 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
| 22:12.13 | Nashe | and also not behind any NAT / Firewall |
| 22:12.14 | ManxPower | Nashe: so again they are mentioned in the ~sipnat document. Yes, all three of those items go in sip.conf, as documented by the ~sipnat document. |
| 22:12.36 | ManxPower | What is doing the NAT? |
| 22:13.23 | Nashe | My Asterisk is connected on a DSL connection provided by Bell Canada... |
| 22:13.51 | Nashe | Bell's Speedstream modem is doing NAT and I am using its DMZ option to forward all ports to my asterisk Box |
| 22:14.49 | Nashe | So Asterisk is behind the NAT but CISCO AS5400 is on Public Internet |
| 22:15.37 | angryuser | Nashe: you dont need to forward all ports, only defined in rtp.conf (rtp) and 5060 (tcp) |
| 22:15.52 | voxter | 5060 is udp not tcp. |
| 22:16.23 | Nashe | in my case it will be .... phone --- Asterisk -- NAT --- CISCO |
| 22:16.24 | *** join/#asterisk vale-ICS (n=vale@boyne.demon.co.uk) |
| 22:16.48 | MikeJ | I thought asterisk supported tcp sip now |
| 22:16.56 | angryuser | voxter : very true ;) |
| 22:17.26 | voxter | MikeJ: 1.6 does i think, but i can pretty much assume thats not the case here. |
| 22:18.50 | Nashe | Actually my phones are also connected to some other Public SIP Proxies... that is why I cannot completely forward a particular port.. this DMZ option works well by letting my phone connect to public SIP proxy and also make calls through this local Asterisk Box.... |
| 22:18.58 | MikeJ | is the tcp stuff stable and compliant? I heard some not so good stuff about it initially |
| 22:19.16 | *** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 22:19.29 | angryuser | hey btw, after upgrading from 1.4.19.1 to 1.4.22 my zap port stop working after a certain amount of time (3-4 hrs) downgraded back, all is fine |
| 22:19.41 | angryuser | zap port's* |
| 22:19.51 | Nashe | as I mentioned earlier... my Asterisk box also when it makes calls when I register the trunk to the SIP proxy... that works perfect... but when I make calls out without registeration... to CISCO... thats where I get the problem |
| 22:21.36 | _ShrikE | MikeJ: I have had pretty good success with Asterisk 1.6 TCP talking to Exchange 2007 UM. |
| 22:22.15 | MikeJ | heh.. thats a very bad one to test against.. ms's compliance is conplete crap |
| 22:23.03 | _ShrikE | Very True. |
| 22:23.05 | angryuser | MikeJ : yes but well spreaded crap sometimes become standart ;) |
| 22:23.16 | MikeJ | heh |
| 22:23.22 | angryuser | MikeJ : and you need to adapt |
| 22:23.29 | *** join/#asterisk CrazyTux (n=brandon@65-60-108-170.static-ip.telepacific.net) |
| 22:23.36 | MikeJ | I don't |
| 22:23.37 | MikeJ | :P |
| 22:23.49 | angryuser | MikeJ : lucky you |
| 22:27.28 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
| 22:27.56 | ManxPower | You don't need to forward ANY ports unless asterisk is acting as a sip server for clients outside the nat |
| 22:28.04 | ManxPower | which it sounds like it is |
| 22:30.09 | *** join/#asterisk Linuturk (n=Linuturk@fluxbuntu/developer/Linuturk) |
| 22:34.17 | *** join/#asterisk Carlos_PHX (n=Carlos@71.36.172.121) |
| 22:45.09 | Nashe | thanks MaxPower |
| 22:45.24 | *** join/#asterisk Mw3_ (n=mw3@ip59934bd1.rubicom.hu) |
| 22:45.47 | Nashe | nat=yes externip and localnet solved the problem |
| 22:45.51 | Nashe | it is working alrgiht now |
| 22:49.31 | *** join/#asterisk mchou (n=mchou@unaffiliated/mchou) |
| 22:49.37 | Nashe | No more One Way Audio |
| 22:51.09 | ManxPower | nat=yes is only required if you have remote clients behind their own nat |
| 22:52.06 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 22:55.06 | *** join/#asterisk knarfly (n=vtserije@c-75-74-155-198.hsd1.fl.comcast.net) |
| 22:58.06 | *** join/#asterisk CGMChris (n=chris@mail.cgmyes.com) |
| 22:58.32 | Nashe | so it means even if I dont set nat=yes it shall work alright |
| 22:58.34 | Nashe | let me try |
| 22:58.35 | CGMChris | I am back, and with a GUI-free config. New problem: Outgoing SIP calls connect, but no sound... silent from both ends. Thoughts? |
| 22:59.35 | Nashe | thats right... it is working even without nat=yes |
| 22:59.47 | Greek-Boy | any of u guys here call center gurus? |
| 23:00.15 | Nashe | but the document you mentioned used it... so I was not sure if I need it or not as the document was to cover both scenarios |
| 23:00.21 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 23:00.22 | Nashe | anyways thank you for your help |
| 23:01.48 | hardwire | smth: get it figured out? |
| 23:03.21 | *** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 23:03.57 | *** part/#asterisk Firass-VC22 (n=firass@restek-ws-0.vikcomm.wwu.edu) |
| 23:04.42 | ManxPower | CGMChris: To answer your earlier questions the place to find the AsteriskGUI people is, oddly, on the #asteriskGUI channel. I'm sure there's a mailinglist for it soewhere too. |
| 23:04.45 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 23:04.53 | ManxPower | CGMChris: is NAT involved. |
| 23:07.00 | CGMChris | ManxPower: Nat is involed. No need for asteriskgui... #1, the room is so slow its pointless to even use it, and #2, I am no longer using the gui |
| 23:07.14 | *** part/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-7764df401a8401a8) |
| 23:07.44 | CGMChris | And I believe I found my problem.... asterisk -r goes NUTS when I make a call. I will review the errors. Thanks. |
| 23:07.53 | ManxPower | CGMChris: I often wonder why people try to use the GUI --- obviously support sucks. |
| 23:08.12 | ManxPower | CGMChris: "asterisk -rvvv" is usually more helpful. |
| 23:08.41 | CGMChris | asked to transmit type x, while native format is Y.... problem with my allow= config. |
| 23:10.50 | *** join/#asterisk Yourname` (i=Yourname@unaffiliated/yourname/x-837320) |
| 23:11.09 | CGMChris | This is getting much easier.... I think the GUI should still be alpha. Honestly, now that I understand both the conf files and the GUI, the conf files are easier AND faster. |
| 23:11.37 | CGMChris | There is, however, a huge learning curve... I cant even talk to the rest of my staff about what I know until they spend a few days learning. |
| 23:13.09 | De_Mon | is 1.6.0 considered production ready? |
| 23:13.33 | ManxPower | De_Mon: Can you think of even a single piece of software that was production ready on a .0 release? |
| 23:13.51 | CGMChris | I read the gui only works with 1.4.x |
| 23:14.12 | De_Mon | ManxPower no, but I also can't think of many releases that take 2-4 years |
| 23:15.07 | ManxPower | De_Mon: I don't know of any Asterisk releases that took more than a year to become stable. 8-| |
| 23:15.51 | De_Mon | sounds like you intend to stay away from 1.6 for a while |
| 23:17.18 | ManxPower | De_Mon: not really. I skipped 1.4 on all 6 or 8 servers I support. |
| 23:17.18 | bpgoldsb | De_Mon: From the people I spoke with at Digium, off the record, it's stable enough to use. |
| 23:17.27 | ManxPower | 1.6 is my last, best hope for Asterisk |
| 23:17.40 | De_Mon | ManxPower you're on 1.2? |
| 23:17.52 | bpgoldsb | De_Mon: And is being used in stable installations in many places. |
| 23:18.16 | ManxPower | you really can't call it 1.6 anymore. The new release method means major changes (including new features) could happen between 1.6.0 and 1.6.1 and 1.6.2, etc. |
| 23:18.24 | ManxPower | De_Mon: My customers use 1.2 |
| 23:18.53 | ManxPower | for 1.2 I did not install it on my systems until Digium installed it on their production PBX |
| 23:18.55 | De_Mon | yeah I know, it's going to make for some real interesting packages in debian i'm sure |
| 23:19.12 | voxter | De_Mon: they said at astricon that they waited this long to release 1.6.0 so that it WAS production ready |
| 23:19.27 | De_Mon | bpgoldsb good to hear |
| 23:19.51 | ManxPower | I may do the same for 1.6 |
| 23:19.58 | bpgoldsb | De_Mon: If you're using Debian, I suggest compiling from source. |
| 23:20.01 | bpgoldsb | It's pretty painless |
| 23:20.39 | ManxPower | bpgoldsb: what specific 1.6 installations? |
| 23:20.45 | De_Mon | voxter yeah I heard, but talk is cheap |
| 23:21.11 | bpgoldsb | ManxPower: I didn't ask for specifics, this was from a Digium Dev though |
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| 23:21.31 | ManxPower | bpgoldsb: without sources cited it is just a rumor |
| 23:21.39 | bpgoldsb | It was directly from a Dev |
| 23:21.57 | ManxPower | bpgoldsb: then he/she needs to site his/her source. |
| 23:21.59 | De_Mon | bpgoldsb I was hoping to spot tzafrir_laptop hanging around to see if there are any working debian build scripts for 1.6, I hate maintaining source installs |
| 23:22.23 | bpgoldsb | ManxPower: I wasn't writing a paper on it, someones word is good enough for me |
| 23:22.29 | bpgoldsb | Like I said, it was an off-the-record comment |
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| 23:23.57 | CGMChris | ManxPower: Maybe you can answer this, as the book doesnt touch on it much, but why would you want to use N instead of X in a pattern for matching outgoing calls? It it only possible to have a number with certain characters in the US or other countries? |
| 23:24.23 | ManxPower | because N=any digit except for 1 or 0 |
| 23:24.30 | ManxPower | X=any digit |
| 23:24.46 | De_Mon | the book doesn't touch on the difference between N and X??? |
| 23:24.50 | CGMChris | I understand that, but in the examples they use 1NXXNXXXXXX... meaning, the telco doesnt release numbers that dont match? |
| 23:25.21 | ManxPower | CGMChris: meaning there will NEVER be a 0 or 1 in that place in a NANPA (use/canada, a few others) phone number |
| 23:25.37 | CGMChris | hmm, ok, thats what I wanted to verify. |
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| 23:27.32 | ManxPower | CGMChris: http://nanpa.com/ |
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| 23:29.18 | CGMChris | ManxPower: I'll just take your word for it. I have far more important things to worry about than N'x and X's. |
| 23:31.19 | ManxPower | Of you want to manage a PBX you need to learn all you can about Telecom, Networiking (specifically UDP, NAT, and SIP), and Linux. |
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| 23:32.33 | CGMChris | I got Linux, UDP, and NAT down. SIP and * are the new elements in my equation. By the time I get my system figured out, I will have an excellent template for deploying cookie-cutter solutions to clients. |
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| 23:33.46 | Darthclue | evening all. anyone here tried setting up asterisk to direct connect to att uverse? |
| 23:37.46 | v4mp | guys how would i go about changing this http://v4mpire.pastebin.com/d342ace76 so it sends the queued calls to the agent group i have found its not SIP/1 so wht would i put there so it calls the agent group ? the agent group name is 1 |
| 23:38.55 | v4mp | the agents aren't loging in to server as an agent they are logging in as user then using agentcallbacklogin to login as an agent with a diff user to the user to login o server |
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| 23:52.38 | ManxPower | AT&T created the NANP in 1947. It currently contains 19 North American countries. (I didn't know there were 19 countries in NA) |
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| 23:53.11 | dexpdx | is 1.6 considered stable or beta? |
| 23:58.10 | hardwire | anybody seen x86 hardware with switched PoE ports? |
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