| 00:09.25 | _Corey_ | adeel: Yeah, I've found some other stuff that seems to confirm that Warning is going to be in a response to a request like an INVITE, so it'd have to be in the OK or BYE or whatever... |
| 00:09.55 | _Corey_ | adeel: Hacking asterisk is a possibility to prove it or doing something with SER perhaps |
| 00:10.03 | adeeln | _Corey_, interesting...it makes sense |
| 00:10.14 | adeeln | _Corey_, sipsak can't do full dialog sessions? |
| 00:10.33 | _Corey_ | Well, the problem is injecting your message instead of the one Asterisk is sending |
| 00:11.03 | _Corey_ | my sipsak experience is somewhat limited though, so maybe someone else could offer guidance |
| 00:11.05 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 00:11.21 | adeeln | yeah, it'll definitely require hacking asterisk to a degree to get this functionality into asterisk...but i think that might be a larger endeavor to undertake at this moment |
| 00:11.23 | _Corey_ | I use it typically to blast notifys at phones and such |
| 00:11.33 | _Corey_ | lol, i'm late for dinner |
| 00:11.46 | adeeln | haha, well that's what happens when you get sidetracked, right? |
| 00:12.05 | _Corey_ | it was a worthy attempt |
| 00:12.21 | _Corey_ | the fiance is going to be unhappy if i don't get going though |
| 00:12.24 | _Corey_ | 'night |
| 00:12.30 | adeeln | I'll try fiddling with it some after i finish some other work |
| 00:12.31 | adeeln | g'nite |
| 00:12.57 | _Corey_ | I usually have this open when I'm at my desk, so let me know if you come up with something tomorrow :) |
| 00:13.04 | adeeln | will do |
| 00:13.15 | _Corey_ | cya |
| 00:18.03 | *** join/#asterisk joobie (~joobie@mx01.anric.com.au) |
| 00:20.57 | *** join/#asterisk digilink (~digilink@70.90.228.197) |
| 00:21.13 | *** join/#asterisk chopp (~chopp@unaffiliated/chopp) |
| 00:21.47 | *** join/#asterisk digilink (~digilink@70.90.228.197) |
| 00:29.53 | *** join/#asterisk Mhaddog_Mac (~anonymous@z65-50-116-17.ips.direcpath.com) |
| 00:33.59 | *** join/#asterisk tash (~tash@ks-76-7-1-196.sta.embarqhsd.net) |
| 00:35.12 | tash | say I don't want to move calls files into the Asterisk outgoing directory until a later date ... is there a preferred method of saving the call file info (say, in Mysql) for later? and, an easy way to retrieve? |
| 00:36.31 | lanning | I would put it in another directory and have an at job mv into place |
| 00:36.31 | tash | am I looking for Asterisk cdr for mysql maybe? |
| 00:37.00 | tash | lanning: yeah, I can easily do that ... but I think I'd like to have a record in the DB of each call as well |
| 00:37.17 | tash | and, I wondered if maybe storing it in the DB would be fairly easy |
| 00:37.49 | lanning | well a record in the db, would be cdr to mysql |
| 00:38.03 | lanning | as for storing the call files, that will probably be custom |
| 00:38.07 | *** join/#asterisk coppice (~chatzilla@60.157.17.210.dyn.pacific.net.hk) |
| 00:38.21 | tash | can you pull info from cdr to create call files on the fly? |
| 00:38.32 | lanning | with a cron job, yes |
| 00:38.47 | tash | ok, so I guess I need to look into cdr |
| 00:39.20 | lanning | just remember cdr has nothing to do with call files |
| 00:39.24 | tash | at a simple level, what is cdr? I know it stands for call detail record (I think?) ... is it just text that gets created when you do *something*? |
| 00:39.56 | lanning | look at master.csv (the default cdr) |
| 00:40.16 | lanning | it is just a log of calls made, after the fact. |
| 00:40.21 | tash | oh, gotcha |
| 00:40.44 | lanning | from, to, length, time, yada yada yada |
| 00:41.14 | tash | would I even have a master.csv by default? out of the box? |
| 00:41.20 | lanning | yes |
| 00:41.22 | tash | I haven't installed any cdr stuff |
| 00:41.25 | tash | what dir? |
| 00:41.30 | tash | I'm in /etc/asterisk right now |
| 00:41.47 | lanning | /var/log/asterisk/cdr I think (it's been a while) |
| 00:42.35 | lanning | http://www.voip-info.org/wiki/view/Asterisk+cdr+csv |
| 00:43.07 | IsUp | /var/log/asterisk/cdr-csv and cdr-custom |
| 00:43.08 | tash | <PROTECTED> |
| 00:43.14 | tash | and read that URL you pasted ... |
| 00:43.17 | tash | thanks lanning |
| 00:43.44 | IsUp | tash: check cdr-csv, there must be Master.csv |
| 00:44.00 | IsUp | but you should set 'enable=yes' in cdr.conf |
| 00:46.06 | tash | enable=yes has a ; in front of it |
| 00:46.15 | tash | which I assume is asterisk conf comment? |
| 00:46.27 | tash | so, remove the ;, save, and restart asterisk? |
| 00:46.38 | lanning | yup |
| 00:46.48 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 00:47.02 | IsUp | Default is "yes" |
| 00:47.24 | tash | k, will try that ... will the Master.csv be created automatically after I do those steps, or do I need to make a call go through first? |
| 00:47.28 | IsUp | do 'cdr status' on asterisk console |
| 00:47.35 | IsUp | you need to place a call first |
| 00:48.18 | tash | cdr status |
| 00:48.18 | tash | No such command 'cdr status' |
| 00:48.28 | tash | i'm in asterisk console (asterisk -r) |
| 00:48.46 | tash | but yeah, haven't had a call go through since I restarted asterisk |
| 00:48.53 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 00:49.07 | IsUp | doesnt matter |
| 00:49.09 | lanning | sounds like the cdr module is not loaded |
| 00:49.11 | IsUp | your CDR module not loaded i think |
| 00:49.35 | IsUp | try 'module load cdr_csv.so' |
| 00:49.39 | tash | yeah, I think you are right ... I didn't do anything special to do that yet |
| 00:50.22 | IsUp | actually you dont have to on default Asterisk installation |
| 00:50.26 | tash | from where, IsUp: |
| 00:50.28 | IsUp | *to do |
| 00:50.31 | IsUp | on asterisk console |
| 00:51.04 | tash | module load cdr_csv.so |
| 00:51.04 | tash | Unable to load module cdr_csv.so |
| 00:51.04 | tash | Command 'module load cdr_csv.so' failed. |
| 00:51.04 | tash | [Feb 9 18:50:14] WARNING[9423]: loader.c:777 load_resource: Module 'cdr_csv.so' already exists. |
| 00:51.33 | IsUp | tash: can you try 'cdr show status' |
| 00:51.59 | tash | ah, better |
| 00:52.22 | tash | Loggin: Enabled, Mode: Simple, Log unanswered calls: No, etc |
| 00:52.33 | tash | what's the benefit of having cdr enabled as opposed to not? |
| 00:52.41 | tash | aren't asterisk logs sufficient? |
| 00:52.44 | IsUp | and registered backends? can you see any output? |
| 00:53.08 | tash | * Registered Backends |
| 00:53.08 | tash | <PROTECTED> |
| 00:53.08 | tash | <PROTECTED> |
| 00:53.08 | tash | <PROTECTED> |
| 00:53.08 | tash | <PROTECTED> |
| 00:53.09 | tash | <PROTECTED> |
| 00:53.09 | tash | <PROTECTED> |
| 00:53.34 | IsUp | oops, paste long lines to pastebin please :> |
| 00:54.11 | IsUp | also it seems that your CDRs are stored on database i think |
| 00:54.23 | IsUp | and check /var/log/asterisk/cdr-custom again |
| 00:54.49 | tash | sorry about the long lines |
| 00:55.09 | tash | ls -la in cdr-custom dir shows nothing |
| 00:55.29 | tash | i'll push a call through later tonight and see what happens |
| 00:55.34 | IsUp | ok, who installed your Asterisk? :P |
| 00:55.39 | tash | I am cooking at the moment with the laptop in the kitchen :P |
| 00:55.53 | tash | me |
| 00:55.55 | tash | from source |
| 00:56.11 | tash | wait ... |
| 00:56.29 | coppice | I had a Pentium 4 laptop which got that hot |
| 00:57.02 | IsUp | i still have P4 laptop which burning my knees |
| 00:57.28 | coppice | The Pentium 4 BBQ edition |
| 00:58.39 | IsUp | haha |
| 00:58.41 | tash | IsUp: I host my server with Linode.com. And, they don't build their linux images with any modules, such as Dahdi...so I had to follow a how-to guied on getting asterisk installed.... |
| 00:58.55 | tash | b/c apt-get install asterisk didn't do the trick |
| 00:59.06 | tash | so, idk why the cdr thing isn't working as you expect it to |
| 00:59.13 | tash | I'll push a call through later and see ... |
| 00:59.41 | IsUp | ok, place a call and let me know how it goes |
| 01:00.25 | tash | word |
| 01:00.39 | tash | you on later? tomorrow? not sure what time I'll test ... well, crap let me just do it now |
| 01:00.59 | IsUp | lol |
| 01:01.03 | IsUp | u got MSN? |
| 01:01.23 | *** join/#asterisk ClintGoudie-Nice (~clint@174-23-208-126.slkc.qwest.net) |
| 01:01.46 | tash | just placed a call from my asterisk box to my cell |
| 01:01.53 | tash | no Master.csv in cdr-custom |
| 01:02.05 | tash | or cdr-csv |
| 01:02.23 | IsUp | and you answered that call, right? |
| 01:02.30 | tash | wouldn't this fine the file? find / -name 'Master.csv' |
| 01:02.30 | *** join/#asterisk Lantizia (~Lantizia@erebus.seaquake.net) |
| 01:02.40 | tash | yes, i got the call, answered it...pressed 1 to ack that I received the call, etc |
| 01:02.59 | IsUp | you dont have to run 'find' |
| 01:03.41 | IsUp | if you are able to provide access, i can take a look into it |
| 01:03.49 | IsUp | while you are making dinner |
| 01:04.13 | tash | nah, not that I don't trust you ... but I'm not an idiot either :P |
| 01:04.16 | tash | no offense |
| 01:04.45 | tash | i meant find from the linux console |
| 01:04.46 | IsUp | lol, no problem |
| 01:04.47 | tash | not asterisk |
| 01:04.59 | tash | I couldn't remember the find command syntax, but that seemed right |
| 01:05.16 | tash | yeah, it was right ... i just found another file on my computer |
| 01:05.25 | tash | weird |
| 01:06.32 | IsUp | brazzers_Master.avi |
| 01:06.55 | IsUp | tash: paste your cdr*.conf to pastebin |
| 01:07.00 | IsUp | and i can take a look |
| 01:07.09 | tash | k |
| 01:08.45 | tash | these are the only lines that don't have a ; in front of them |
| 01:08.47 | tash | http://pastie.org/1547280 |
| 01:11.01 | IsUp | and paste your cdr_custom.conf please |
| 01:11.26 | *** join/#asterisk digilink (~digilink@70.90.228.197) |
| 01:12.38 | tash | here is a pastie ... if you need pastebin I can do that, URL? |
| 01:12.54 | IsUp | pastebin, yes |
| 01:13.36 | *** join/#asterisk digilink (~digilink@70.90.228.197) |
| 01:19.15 | tash | http://pastebin.com/D1KHzN8t |
| 01:19.20 | tash | hope that is the right site |
| 01:22.31 | tash | IsUp: find anything? |
| 01:22.42 | *** join/#asterisk digilink (~digilink@irc.stephennet.net) |
| 01:24.48 | IsUp | tash: and paste 'cdr show status' output to pastebin |
| 01:25.18 | *** join/#asterisk b14ck (~b14ck@cpe-72-129-70-245.socal.res.rr.com) |
| 01:26.11 | tash | :) |
| 01:27.11 | tash | http://pastebin.com/4Jaq7aBY |
| 01:29.28 | IsUp | tash: whats your asterisk version? |
| 01:29.49 | tash | 1.6 |
| 01:29.53 | tash | I gotta run |
| 01:29.56 | tash | sorry ... |
| 01:29.59 | tash | be on later? |
| 01:30.57 | IsUp | i'll be here tomorrow |
| 01:31.27 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-116-17.ips.direcpath.com) |
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| 02:23.31 | *** join/#asterisk daxt (~daxt@112.135.82.197) |
| 02:24.54 | *** join/#asterisk Ta^3 (~tacvbo@evdomip-203-91-81.iusacell.net) |
| 02:27.17 | daxt | guys is there a way for users to dynamically register their sip devices within asterisk rather than manually editing sip.conf ? |
| 02:29.15 | p3nguin | You can use realtime, but someone still has to manually add the device to the database. |
| 02:29.30 | IsUp | hey p3nguin |
| 02:29.34 | p3nguin | hi |
| 02:29.53 | Kobaz | auto add user on registration is evil |
| 02:30.12 | Kobaz | eeeevviiiil |
| 02:30.16 | daxt | why is that ? |
| 02:30.25 | Kobaz | security hazard? |
| 02:30.43 | daxt | then how come other auto add user apps run smoothly ? |
| 02:30.51 | daxt | and what makes asterisk an exception of that ? |
| 02:31.25 | Kobaz | it's not a matter of running |
| 02:31.44 | nestAr | what other apps are we talking about |
| 02:31.50 | Kobaz | sure it's technicaly feasible to do it |
| 02:32.02 | Kobaz | but is it a "good idea" (tm) |
| 02:32.14 | daxt | u have to make it a "better idea" :D |
| 02:32.25 | nestAr | why would anyone want to give unfettered access to something like a PBX? |
| 02:32.38 | nestAr | sounds like a disaster in the making |
| 02:32.42 | daxt | not untettered access |
| 02:32.43 | daxt | :D |
| 02:32.59 | daxt | there has to be a certain limit |
| 02:33.02 | Kobaz | nestAr: he's probably thinking of something like polycom kirk's where you can have the handset create iself a user on the base station |
| 02:33.28 | Kobaz | nestAr: but then once you're done you can turn it off... but even if you're on the base station it doesn't mean you have access to anything else |
| 02:33.29 | daxt | Kobaz exactly ! |
| 02:33.40 | Kobaz | so the security risk is very minimal |
| 02:33.48 | Kobaz | daxt: here's the problem |
| 02:34.08 | daxt | database overload ? |
| 02:34.30 | Kobaz | daxt: yes, we can do it... yes, it's pretty easy to do... but if it's in the standard build, people are going to use it, and people are going to forget to turn it off, or not secure it properly, and they will get owned |
| 02:34.38 | Kobaz | the database is hardly the problem |
| 02:34.55 | Kobaz | more like risking hundreds of thousands of dollars in bills once your system gets owned |
| 02:35.04 | daxt | when the device is turned off , u automatically remove that device from sip db |
| 02:35.29 | daxt | but u have to limit that to certain number of devices |
| 02:35.37 | daxt | say about 25 |
| 02:35.40 | Kobaz | if your're going to limit it to a range of devices |
| 02:35.46 | Kobaz | then put those devices in your system to begin with |
| 02:35.50 | Kobaz | done and done |
| 02:36.17 | Kobaz | daxt: it's not the number of devices, it's minutes... and black market international minutes are in high demand |
| 02:36.32 | daxt | i see |
| 02:36.36 | tzanger | auto create account on register? insanity. |
| 02:36.51 | Kobaz | tzanger: there should be a bot entry for that |
| 02:36.59 | tzanger | leifmadsen can handle that |
| 02:37.10 | Kobaz | to the documentation mobile! |
| 02:37.14 | tzanger | lawl |
| 02:37.16 | nestAr | INSANITY! |
| 02:37.42 | Kobaz | daxt: if you think auto creating accounts on register is the solution, you're asking the wrong question |
| 02:38.17 | tzanger | sounds like lazy administration to me |
| 02:38.26 | daxt | kobaz , no no i need dynamic sip device registration :D |
| 02:38.33 | Kobaz | no you don't |
| 02:38.35 | tzanger | you want automatic creation, put a human in the loop |
| 02:38.40 | Kobaz | rethink your problem you are trying to solve |
| 02:38.49 | tzanger | anything unknown goes to a context that has a "please enter pin" type interface |
| 02:39.18 | tzanger | any number they call asks them to refer to a card that came with the device which is unique per device... they enter the pin and that autoconfs |
| 02:39.49 | Kobaz | daxt: and you have yet to tell us what that is... that you are trying to accomplish |
| 02:39.50 | tzanger | the card is of course generated beforehand and the pbx db has a list of cards not yet activated |
| 02:40.00 | tzanger | that gives you autoreg without the risk |
| 02:41.27 | Kobaz | if you say you are building a hosted voip service |
| 02:41.38 | Kobaz | i will have tzanger over there to slap you |
| 02:41.43 | daxt | kebaz , i have a wireless network where number of computeres that connect to it change with time , and i want them to use sip softphones to communicate with a central location. |
| 02:42.04 | daxt | in that scenario , manually registering the sip devices in the sip.conf is a pain |
| 02:42.54 | Kobaz | daxt: give people a website to go to, have them enter in their reg info, show a list of unauthenticated devices, and have them click on the one thats theirs |
| 02:43.14 | tzanger | I still like my personalized id card or sticker on the bottom of the device |
| 02:43.18 | Kobaz | daxt: otherwise someone is going to walk by your office, hop on the wireless and own your pbx |
| 02:43.19 | tzanger | 5 digit PIN is plenty of security |
| 02:43.34 | tzanger | daxt: kobaz is 100% correct, you don't want what you're suggesting |
| 02:43.51 | tzanger | or rather, whoever will be responsible for the bills will not want you doing that |
| 02:43.56 | daxt | hmmm , isn't there a secure way of getting done what i need ? |
| 02:44.01 | Kobaz | yes there is |
| 02:44.06 | Kobaz | but not automagically |
| 02:44.06 | tzanger | daxt: we gave you two ways |
| 02:44.26 | Kobaz | secure and auto creating users do not go together, ever |
| 02:44.39 | Kobaz | it's like having a key dispenser in the front door of your house |
| 02:44.49 | daxt | by the way my PBX will route sip softphone calls only. so there wont be any risks involving in it. |
| 02:44.55 | tzanger | it's much, much worse than that |
| 02:45.01 | Kobaz | when someone walks by, say the mailman, it launches a key at them, with a note saying 'please come in' |
| 02:45.03 | tzanger | daxt: so? |
| 02:45.15 | tzanger | what stops me from running a SIP client on a laptop and making free calls? |
| 02:45.37 | tzanger | what stops me from running a 1000 user SIP proxy and selling black VOIP minutes? |
| 02:45.41 | Kobaz | daxt: how do you know the device hoping on your network is allowed to make calls |
| 02:45.54 | IsUp | what stops me from frauding |
| 02:45.56 | Kobaz | hopping |
| 02:46.07 | Kobaz | tzanger: didn't like my mailman analogy? |
| 02:46.07 | daxt | i test them |
| 02:46.22 | tzanger | Kobaz: if you have a key to the house, you break into one house |
| 02:46.25 | Kobaz | daxt: if you spend the time to test them, you can spend the time to register them |
| 02:46.42 | tzanger | it's like leaving all of your money on the front porch with a "please don't steal me" sign |
| 02:46.46 | Kobaz | tzanger: yeah, it's like giving someone a key to a key factory |
| 02:46.56 | daxt | i only test once , and i have to manually register every time it connects , won't it be a pain ? :D |
| 02:47.06 | Kobaz | daxt: leave the entry in the system |
| 02:47.18 | Kobaz | daxt: write yourself a script where it takes 2 seconds to create a new account |
| 02:47.24 | *** join/#asterisk faction (~faction@c-24-61-120-11.hsd1.nh.comcast.net) |
| 02:47.36 | Kobaz | ./adduser 555-1212 deviceid-32234 |
| 02:47.37 | Kobaz | wtc |
| 02:47.39 | Kobaz | etc |
| 02:47.40 | Kobaz | done and done |
| 02:48.41 | daxt | wow kobaz , that is what i was looking for |
| 02:48.51 | daxt | and never thought of a thing like that :D |
| 02:49.02 | daxt | do i have to reload asterisk after that ? |
| 02:49.25 | tzanger | you can do a sip reload or if you're running realtime, you can do nothing at all, it's live. |
| 02:49.34 | Kobaz | daxt: you can even have your script reload for you |
| 02:49.37 | Kobaz | asterisk -rx "sip reload" |
| 02:49.54 | Kobaz | daxt: i told you... you have to rethink your problem you're trying to solve |
| 02:50.02 | daxt | ya i will |
| 02:50.11 | daxt | but i will have to get this done first :D |
| 02:51.18 | Kobaz | if you have that many people coming and going |
| 02:51.29 | Kobaz | you can do something liek wipe yesterdays users every morning |
| 02:51.33 | Kobaz | or whatever |
| 02:51.38 | daxt | ya |
| 02:51.46 | daxt | using a cron job |
| 02:51.59 | tzanger | nah put an expire time in the sip.conf entry and have a crontab erase anyone older than 24h or something |
| 02:55.06 | *** join/#asterisk ectospasm (ectospasm@188.72.223.139) |
| 02:55.29 | daxt | tzanger , thanks dude ! |
| 02:55.55 | tzanger | daxt: you're not setting up a web server ... the worst thing that can happen there is you become a kiddie porn hub or something |
| 02:56.03 | tzanger | you're running a pbx... telco charges ring up VERY fast |
| 02:56.18 | daxt | tzanger - this is an intranet :D |
| 02:56.51 | tzanger | daxt: how many times have you heard about people compromising networks? |
| 02:57.05 | daxt | none :( |
| 03:07.22 | *** join/#asterisk csnook (~chris@c-76-19-64-161.hsd1.ma.comcast.net) |
| 03:10.04 | *** part/#asterisk b14ck (~b14ck@cpe-72-129-70-245.socal.res.rr.com) |
| 03:11.33 | daxt | guys what is the best way to monitor asterisk server ? |
| 03:11.53 | daxt | such as checking real time log and debug messages ? |
| 03:17.19 | *** join/#asterisk poptix (poptix@poptix.net) |
| 03:17.56 | poptix | howdy folks, what's the recommended setup with FreePBX for multiple extensions (linked via a call group) to share a voicemail box? |
| 03:18.45 | poptix | i currently have x8000 x8001 x8002 in the x800 ring group, with voicemail configured on the x8000 extension |
| 03:18.59 | poptix | it all works, but when they're sent there it says they reached x8000 instead of x800 =( |
| 03:19.28 | Kobaz | this is #asterisk? |
| 03:20.01 | poptix | [09/2117] uiu Irssi: Join to #asterisk was synced in 7 secs |
| 03:20.05 | poptix | possibly. |
| 03:21.09 | poptix | can a ring group share the same # as an extension? |
| 03:21.15 | adeel | no |
| 03:22.08 | poptix | didn't think so |
| 03:22.18 | adeel | poptix, are you trying to have a shared voicemail among X number of devices? |
| 03:22.40 | poptix | adeel: correct, employees have sip clients and office phones, but need a single mailbox |
| 03:22.56 | adeel | poptix, are you using symlinks to achieve it? probably the simplest way |
| 03:22.59 | poptix | (or multiple phones) |
| 03:23.33 | poptix | adeel: i was hoping to find the 'correct' way through freepbx so the system is easily maintainable by other employees, if there isn't one that's fine though |
| 03:24.17 | adeel | poptix, afaik, symlinks is the only way to do it...you probably could hack together a dialplan that allows for it, but i think that'd be more complicated and prone to bugs |
| 03:24.44 | adeel | and maintainability is defined by the quality of your documentation |
| 03:26.59 | poptix | unfortunately maintainability here is defined by the willingness of an employee to read the internal wiki |
| 03:28.47 | adeel | that's usually the case...but having documentation and not reading it, is still better than having no documentation at all |
| 03:29.00 | poptix | so when =) |
| 03:29.04 | poptix | er |
| 03:29.07 | poptix | =) |
| 03:29.09 | poptix | where is the voicemail box # read from then? |
| 03:29.53 | poptix | ie, for the text-to-speech "you've reached voicemail box XXXX" |
| 03:30.20 | adeel | actually, i'm not entirely sure; i just remember reading about shared voicemail via symlinks of /var/spool/asterisk/voicemail/ |
| 03:30.23 | poptix | if i can't create extension 800 (due to the ring group) I can't create the voicemail box for it either |
| 03:30.27 | *** join/#asterisk Kumbang (~kumbang@180.245.137.5) |
| 03:30.48 | adeel | you could always just record a greeting message to get around the text-to-speech issue |
| 03:31.02 | poptix | fair enough |
| 03:31.17 | poptix | i don't think the symlinking will help since i'm directing them to the specific mailbox via the ring group |
| 03:31.50 | poptix | i'll just record a generic 'please leave a message' .wav and set it as the default greeting on all mailboxes |
| 03:31.51 | adeel | well it allows for the group to check the voicemail box irrespective of what phone they're calling from (within that group that is) |
| 03:32.21 | adeel | otherwise, only 1 phone would get the MWI |
| 03:32.21 | poptix | we're delivering voicemail via email instead of storing it |
| 03:32.25 | adeel | ahh |
| 03:32.28 | adeel | n/m then |
| 03:32.45 | poptix | yeah, too many people forget VM passwords and complain about the blinky lights |
| 03:32.48 | poptix | heh |
| 03:32.53 | adeel | heh |
| 03:36.19 | carrar | People still use voicemail? |
| 03:36.33 | poptix | i rarely even plug my phone in |
| 03:36.41 | poptix | but apparently some people find it useful |
| 03:37.26 | poptix | besides, it's much easier to transfer the net.kooks to the extension of my 'manager' where they can leave voicemail |
| 03:37.40 | poptix | (which goes to /dev/null) |
| 03:38.19 | adeel | haha, like it should |
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| 04:37.42 | daxt | guys i have following sip.conf entry "[danush] |
| 04:37.42 | daxt | type=friend |
| 04:37.42 | daxt | host=dynamic |
| 04:37.42 | daxt | secret=whocares |
| 04:37.43 | daxt | context=default" |
| 04:38.05 | daxt | after reloading asterisk i tried to connect my sip softphone and get "invalid" as a response |
| 04:38.45 | daxt | but when i change [danush] to [myself] , reload asterisk and connect sip softphone again , i get "registered" as a response |
| 04:38.57 | daxt | don't u guys think it as a weird behaviour ? |
| 04:40.35 | p3nguin | No. Your phone is configured to use "myself" as the user ID. |
| 04:42.00 | daxt | if i change "myself" to any other username , then asterisk shud accept that too |
| 04:43.03 | lanning | sip.conf needs to match your softphone config |
| 04:44.01 | p3nguin | The name in the square brackets is what the phone needs to have on it. Always run "sip reload" after changing and saving sip.conf. |
| 04:44.04 | lanning | if sip.conf has "[johndoe]" then the softphone user ID should be "johndoe" |
| 04:50.16 | *** join/#asterisk daxt (~daxt@112.135.75.90) |
| 04:56.08 | daxt | guys what is the best tool to monitor asterisk , all it's debug and log messages ? |
| 04:56.26 | *** join/#asterisk timahvo1 (~rogue@41.215.1.35) |
| 04:57.34 | tash | anyone have any experience creating/editing the lexicon.txt file for a voice such as Allison? |
| 04:58.06 | tash | I'm using cepstral's Allison along with Asterisk ... annoyed at how she pronounces some words |
| 04:58.38 | *** join/#asterisk smash- (smash@245.sub-72-102-164.myvzw.com) |
| 04:58.43 | smash- | Hey, can someone help me out |
| 04:58.57 | tash | smash: maybe ... what's up? |
| 04:59.00 | smash- | I have a old cisco pix and im having real problems trying to get remote extensions to register. |
| 04:59.35 | smash- | i'm trying to compile an email to the cto letting them know about problems i have "heard" about with cisco pix and SIP RTP & NAT |
| 05:00.00 | tash | smash: do you have the correct ports open and such? |
| 05:00.12 | smash- | i see that the "Cisco Fix UP" fixes |
| 05:00.13 | tash | udp 5060 would be the first thing that comes to mind |
| 05:00.16 | smash- | yah |
| 05:00.19 | smash- | i got all that setup |
| 05:00.34 | smash- | i have 3 sites at another company i work for running endian & juniper firewalls that work just great |
| 05:00.50 | smash- | this location in conversation about has a 8yr old Cisco Pix 501e i belive |
| 05:00.54 | smash- | 500 series |
| 05:01.18 | tash | I could be wrong, but I doubt the pix is the problem. Like I said, I could be wrong ... but cisco usually has solid products that can do this kind of thing |
| 05:01.32 | tash | Have you done any packet captures to see where traffic is getting to, etc? |
| 05:01.55 | drmessano | PIX 501 is notorious for having problems with SIP |
| 05:02.12 | tash | drmessano: really? then I don't know what I'm talking about smash :P |
| 05:02.19 | tash | I've never used a pix myself ... |
| 05:02.24 | smash- | the admins are fucking located in demark |
| 05:02.26 | drmessano | They're horrible |
| 05:02.31 | smash- | yes |
| 05:02.34 | smash- | drmessano |
| 05:02.44 | smash- | i know i have heard this over and over the past half a decade |
| 05:02.53 | smash- | and never used them |
| 05:02.56 | *** join/#asterisk lfod- (lfod@unaffiliated/lfod-) |
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| 05:03.06 | smash- | this location has a TDM wanpipe so all traffic inside the building is great |
| 05:03.13 | drmessano | I don't know there is a fix other than "dont use them where SIP is involved" |
| 05:03.22 | smash- | but if i get phones from san diego trying to register |
| 05:03.25 | smash- | fail fail fail |
| 05:03.29 | drmessano | Yep |
| 05:03.35 | smash- | the tcp dump is unavailable to me |
| 05:03.37 | tash | wtf, that's messed up |
| 05:03.43 | tash | about the fail, I mean |
| 05:03.48 | smash- | its because cisco wants to use Skinny |
| 05:04.03 | smash- | this was before everyone went crazy and cisco fixed with the ASA |
| 05:04.04 | drmessano | LOL |
| 05:04.21 | drmessano | Cisco just has never done a good job with SIP.. Period |
| 05:04.31 | smash- | drmessano can you possible google me a artical that iterates this |
| 05:04.32 | tash | smash: Deary CTO, buy a new firewall. |
| 05:04.37 | smash- | yes they are going to |
| 05:04.51 | smash- | the CFO came up to me and said please write an email with this info so we can get us a new firewall |
| 05:05.00 | smash- | there about to fly me to san diego to setup firewall there aswell |
| 05:05.10 | smash- | i know if i repplace the firewall it will be no problem |
| 05:05.14 | tash | get an F5 load balancer, they kick ass and handle sip well ... maybe not what you need, just sayin' ... they are slick |
| 05:05.15 | drmessano | smash-, you're gonna have to do your own googling.. I am not a proxy. Just years of experience knowing they don't work worth a crap with SIP, and years of seeing complaints in here about 501s and SIP |
| 05:05.28 | smash- | lol |
| 05:05.35 | smash- | well drmessano any keywords u could advise? |
| 05:05.47 | drmessano | 501 SIP sucks Asterisk register broken |
| 05:05.52 | thehar | lol |
| 05:05.54 | thehar | so true |
| 05:05.55 | smash- | kk |
| 05:05.58 | smash- | sounds goood |
| 05:06.01 | smash- | ill be back |
| 05:06.04 | smash- | drunk emails are fun |
| 05:06.11 | drmessano | Dear CEO, |
| 05:06.13 | smash- | i told them 11pm i got 2 hr |
| 05:06.48 | drmessano | UR 501 CISKHO FIREWAIL DOESNT WORK WITH TEH SIPS WE USE WITH TEH ASTERISKS. U NED A NEW FIARWALL, SIR, K? THXGODNITE |
| 05:06.51 | drmessano | Hows that? |
| 05:07.00 | tash | haha |
| 05:07.04 | thehar | <PROTECTED> |
| 05:07.17 | tash | you left out the belch at the end |
| 05:07.27 | drmessano | So true |
| 05:07.52 | drmessano | Ok, forgot the signature |
| 05:07.55 | drmessano | Signed, |
| 05:08.06 | drmessano | Me |
| 05:08.14 | smash- | hahaha |
| 05:08.34 | tash | "TEH SIPS" <---- cracks me up |
| 05:09.08 | tash | drmessano: you ever tinkered with TTS engines such as Cepstral? |
| 05:09.21 | drmessano | I have... Cepstral is pretty cool |
| 05:09.40 | tash | I'm annoyed at how Allison effs up some words ... like the word "oil" |
| 05:09.46 | tash | I wish I could make her more human-like |
| 05:10.01 | tash | I know you can edit the lexicon file, but I don't know how to get her to stress a syllable |
| 05:10.03 | drmessano | Oh yes.. I tend to modify the words a little to make it work, though you can train it a little |
| 05:10.15 | tash | how the hell do you train it? haha |
| 05:10.16 | drmessano | OY UL |
| 05:10.31 | tash | I think I have it as ... well, let me look in lexicon.txt real quick |
| 05:10.36 | drmessano | Well, I was referring to the Lexicon file, where you're supposed to tweak. I just change the words |
| 05:11.05 | tash | oil 0 oy1 l |
| 05:11.08 | tash | yeah, I am too |
| 05:11.15 | tash | I have OY L |
| 05:11.27 | tash | guess I should put OY UL and see what she sounds like |
| 05:11.40 | drmessano | Wouldnt that be OY EL? You want the U sound |
| 05:11.50 | mzb | yeah, I've got a script that does pronunciation changes ... eg, for "Alicia" |
| 05:11.53 | tash | yeah, I think you are right |
| 05:12.09 | drmessano | OY UHL maybe |
| 05:12.19 | drmessano | That may be ULE though |
| 05:12.24 | drmessano | or YULE |
| 05:12.32 | drmessano | OY UL may be closer |
| 05:12.43 | drmessano | OY YULE is the Aussie Yule Log |
| 05:13.42 | tash | hmm, unknown phone type in wrule "ul" <--- that was after I changed it to oy1 ul ... I wonder if I need to put a 1 after something else |
| 05:14.08 | tash | maybe I should use uh1 |
| 05:14.18 | tash | oy1 uh1 l |
| 05:15.07 | tash | mzb: I bet that script is a lifesaver |
| 05:15.44 | mzb | it does a few other things |
| 05:16.12 | mzb | it's not using festival, so I get it to do phrase-caching |
| 05:16.13 | tash | OY1 UH1 L seems to have worked well for "oil" |
| 05:16.30 | mzb | (I used it on a pretty slow machine back in the day) |
| 05:16.44 | tash | can you stress stress certain syllables in the lexicon? |
| 05:17.02 | tash | for instance, say I wanted to stress the "p" in backpack for some reason |
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| 05:25.04 | mzb | no, it's not that .... |
| 05:25.06 | mzb | ah, nm |
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| 05:34.25 | tash | weird, network issues |
| 05:38.54 | mzb | no, it's not that complicated .... it just substitutes some words for "translations" so they sound right |
| 05:39.09 | mzb | so I have a "pronunciation" rc file, in effect |
| 05:40.41 | mzb | eg's include: |
| 05:40.44 | mzb | [Aa][Dd][Mm][Ii][Nn] ad-min |
| 05:41.00 | mzb | [sS][qQ][lL] S Q L |
| 05:41.13 | mzb | [Aa][Ll][Ii][Cc][Ii][Aa] Ah-leese-ea |
| 05:41.27 | mzb | [Mm][Aa][Rr][Cc][Ii][Aa] Marsee-ah |
| 05:41.44 | mzb | crude, but good enough for me |
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| 05:55.06 | tash | mzb: cool |
| 05:55.32 | tash | do the brackets mean anything? or are they just separators to make reading the examples easier? |
| 05:55.50 | mzb | case statement in bash iirc |
| 05:56.10 | mzb | or it might have been sed |
| 05:56.17 | mzb | don't recall exactly |
| 05:56.20 | tash | what do you mean by case statement? sorry |
| 05:56.32 | mzb | nm ... probably sed regex |
| 05:57.36 | tash | ah |
| 05:57.39 | tash | gotcha |
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| 06:36.30 | recluze | hi, I'm trying to get a custom greeting (french) for ONE of my extensions. I have put the wav file in /var/spool/asterisk/voicemail/default/1000/unavail. Doesn't do anything. Can I get a hint? |
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| 07:22.16 | schmidts | good morning |
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| 07:30.04 | recluze | welcome :) |
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| 08:57.16 | viraptor | hi all, what's the variable for iax username / where would I find that kind of information? |
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| 09:01.35 | kaldemar | viraptor: iax.conf. as variable, do you mean how to access that data in the dialplan? |
| 09:02.06 | viraptor | kaldemar: exactly - I'd like to get the username which was used for authentication |
| 09:02.15 | viraptor | in the dialplan |
| 09:03.34 | kaldemar | don't you see it in the channel name? |
| 09:04.12 | viraptor | yes, but isn't there a way to get it without trying to parse the channel name? |
| 09:05.14 | kaldemar | not that i know of. SIP has a function called SIPCHANINFO, but there's no equivalent for IAX. |
| 09:06.46 | kaldemar | ah, you can use IAXPEER(CURRENTCHANNEL[,item]) |
| 09:06.51 | kaldemar | core show function IAXPEER |
| 09:07.33 | kaldemar | there is no entry for the authentication name, but at least it gives some information if you don't want to parse the channel name. |
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| 09:37.23 | justdave | I'm running asterisk 1.8.2.1, and trying to connect an external phone to it... the asterisk is behind a static NAT, with all of the appropriate stuff port forwarded. Been doing some packet tracing to figure out what's going on, and asterisk seems to be sending its internal IP in the SDP payload instead of the external one |
| 09:37.37 | justdave | despite externip being set, and the client not being in the netblock defined in localnet |
| 09:37.53 | justdave | (client entry has nat=yes) |
| 09:38.22 | justdave | anyone have any ideas? |
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| 09:51.27 | kaldemar | justdave: [general] should have nat=yes under it. |
| 09:51.58 | kaldemar | if the client has a public ip, no need for nat=yes definition for it. |
| 09:53.07 | justdave | adding that to [general] doesn't seem to change the behavior any. |
| 09:54.21 | kaldemar | pastebin your config and a sip trace |
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| 10:01.58 | justdave | hmm, now that's interesting. |
| 10:02.05 | justdave | gives the network guy a poke. |
| 10:02.26 | justdave | trying to get that packet trace from the asterisk server, it claims it's sending its external IP in the SDP |
| 10:02.32 | justdave | the tcpdump on my end shows the internal one. |
| 10:02.38 | justdave | firewall must be swapping it back on the way out |
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| 10:15.48 | justdave | yay, works now. |
| 10:16.04 | justdave | for the record, Juniper firewalls apparently have a broken SIP ALG in them |
| 10:16.35 | justdave | "deactivating the ALG apparently is not the same as disabling it" he says |
| 10:16.50 | justdave | it had been deactivated several hours ago |
| 10:17.42 | justdave | SRX240 anyway |
| 10:17.42 | kaldemar | usually any ALG will cause problems. |
| 10:18.00 | justdave | stupid thing was translating the IP addresses the wrong direction |
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| 10:47.11 | schmidts | kaldemar i have found one CPE where SIP ALG really works very nice |
| 10:48.16 | schmidts | kaldemar a Pirelli Broadsoft VDSL Router with OpenRG system on it. This router really does a very nice sip nating without any problems, but its not so easy if you want to use different qos classes for sip and rtp traffic ;) |
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| 13:41.46 | *** mode/#asterisk [+o BMJ] by ChanServ |
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| 14:07.44 | BMJ | Asterisk SCF Developer Call at 10:00 EST. Bridge opens at 9:45 EST. Info here: https://wiki.asterisk.org/wiki/display/TOP/Asterisk+SCF+Weekly+Developer+Conference+Call |
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| 14:44.56 | leifmadsen | BMJ: heh, I keep thinking I'm entering the conference number wrong, but it's just because the room isn't open yet :) |
| 14:45.28 | BMJ | leifmadsen: yeah. ;-) Maybe it should announce that? |
| 14:46.03 | leifmadsen | that's what I was thinking |
| 14:46.19 | leifmadsen | I opened a ticket today to get access to the server, so we can play with that either this afternoon before the next call or tomorrow |
| 14:46.36 | leifmadsen | heh, didn't announce whoever just came into the conference |
| 14:46.41 | leifmadsen | "...is in the conference" |
| 14:46.57 | leifmadsen | I learn more things each time I participate ;) |
| 14:46.58 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 14:47.28 | BMJ | leifmadsen: that was me. |
| 14:47.40 | leifmadsen | does the leader have the option of recording their name? |
| 14:52.49 | BMJ | leifmadsen: yes |
| 14:53.25 | BMJ | Asterisk SCF Developer Call at 10:00 EST. Bridge open. Info here: https://wiki.asterisk.org/wiki/display/TOP/Asterisk+SCF+Weekly+Developer+Conference+Call |
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| 15:34.02 | a_m_y | hi may I know the best call center application for asterisk. tnx in advance |
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| 15:34.19 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:34.58 | a_m_y | ~sipnat |
| 15:34.58 | infobot | [~sipnat] Quick guide on configuring Asterisk + 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 . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, canreinvite, externhost or externip, and localnet. |
| 15:35.41 | a_m_y | ~book |
| 15:35.41 | infobot | [~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 at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook, or http://ofps.oreilly.com/ |
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| 15:42.43 | *** join/#asterisk Dovid (Dovid@office.mypbxmanager.net) |
| 15:42.47 | Dovid | j #juniper |
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| 15:51.59 | *** join/#asterisk tash (~Tommy@66.64.111.226) |
| 15:53.28 | tash | will asterisk play .wav and/or .mp3? Or, do I have to use sox to conver to .gsm? |
| 15:54.08 | *** join/#asterisk Defraz (~Defraz@63.226.95.152) |
| 15:54.32 | Dovid | tash: fromthe dp you need wav |
| 15:54.37 | Dovid | 8khz Mono |
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| 15:54.44 | Dovid | for moh you can use either or |
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| 15:55.40 | b14ck | Question: is there anyway that any of you can think of that I could load a list of subroutines based on name, then call each one of them in order? EG: |
| 15:55.51 | b14ck | I've got 50 subroutines named `test_blah_xxx` |
| 15:56.00 | b14ck | I'd like to do `Gosub(test_*,s,1)` |
| 15:56.13 | b14ck | And have asterisk run Gosub(test_blah_001,s,1), Gosub(test_blah_002,s,1), etc. |
| 15:56.25 | b14ck | If that makes sense. |
| 15:56.40 | b14ck | would certainly save me a lot of copy and paste |
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| 15:59.32 | kaldemar | b14ck: make a loop with the while applications. |
| 15:59.52 | b14ck | kaldemar: I would do that, but in actuality my tests arent named numerically like I showed above (sorry) |
| 16:00.11 | b14ck | They have human readable names, eg: `test_this_sub_returns_0`, `test_soemthing_else` |
| 16:00.14 | b14ck | etc |
| 16:00.17 | b14ck | Otherwise I'd loop through them. |
| 16:00.31 | b14ck | Unless I'm missing something--and there is a way to get a list of all subroutines by name? :o |
| 16:01.57 | schmidts | bl4ck you can use an array |
| 16:02.07 | a_m_y | hi may I know the best call center application for asterisk. tnx in advance |
| 16:02.10 | b14ck | schmidts: Could you elaborate a bit? |
| 16:02.24 | *** join/#asterisk jaxyeh (~jaxyeh@c-69-250-52-161.hsd1.md.comcast.net) |
| 16:02.27 | tash | Dovid: what do you mean by 'dp'? |
| 16:02.29 | *** part/#asterisk jaxyeh (~jaxyeh@c-69-250-52-161.hsd1.md.comcast.net) |
| 16:02.51 | schmidts | you have to initialise it but then you can use the while option |
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| 16:03.00 | b14ck | oh, I see |
| 16:03.06 | b14ck | Well, that is definitely better. |
| 16:03.12 | b14ck | Thanks! |
| 16:03.13 | schmidts | like SET(testing[0] = test_something |
| 16:03.17 | b14ck | Yah |
| 16:03.21 | tash | Dovid: by the way, at the moment I am creating .wav with the swift command. Then, I'm using sox to conver to gsm: sox blah.wav -r 8000 blah.gsm |
| 16:03.23 | schmidts | Set(testin{1}=test_something_else ;) |
| 16:03.51 | b14ck | I suppose I can also refactor that abit if I just substitute in test_modulename |
| 16:03.54 | b14ck | heh |
| 16:03.55 | b14ck | Thanks, I'll use this method. |
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| 16:06.03 | schmidts | your welcome |
| 16:13.15 | leifmadsen | you're* :) |
| 16:15.12 | breardo | thanks leif.. that was getting to me |
| 16:15.23 | Kobaz | youharre |
| 16:20.42 | *** join/#asterisk asterisk-learner (~chatzilla@77.42.241.114) |
| 16:22.22 | leifmadsen | :D |
| 16:22.43 | leifmadsen | english isn't schmidts native language so he can be excused ;) |
| 16:22.55 | leifmadsen | I know a lot of people who can only speak english get that wrong |
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| 16:36.52 | Kobaz | heh |
| 16:37.18 | Katty | good morning asterisk |
| 16:37.24 | Kobaz | axeterisk |
| 16:38.30 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 16:39.12 | n3hxs | Morning Katty |
| 16:40.52 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 16:43.20 | tash | n3hxs: iirc, you are the person who recommended linode to me ... do you use them? (I could be wrong and maybe it was someone else) |
| 16:43.47 | n3hxs | maybe n3glv |
| 16:44.20 | tash | ah, probably ... the n3 threw me |
| 16:44.52 | tash | whoever it was, I put them down as my referree (spelling?) |
| 16:45.10 | tash | I'm on a different computer, so I can't lookup my #asterisk chat log to see who it was |
| 16:46.02 | Katty | hugs n3hxs |
| 16:46.50 | n3hxs | hugs Katty back.... and front |
| 16:47.51 | Katty | :P |
| 16:49.25 | n3hxs | didn't expect a kiss like that. |
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| 16:53.42 | b14ck | oh nice |
| 16:53.47 | b14ck | this works great :o |
| 16:53.56 | b14ck | UNIT TEST FOR ASTERISK DIALPLAN -- ARE WORKING! |
| 16:54.19 | Qwell | BLARGHSIFDJShFJKSDHF |
| 16:54.24 | Qwell | F YOU SNOW |
| 16:54.28 | Qwell | </rant> |
| 16:54.33 | b14ck | sup :o |
| 16:54.43 | Kobaz | nooooo, good snow |
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| 17:17.06 | mechbangirc | hi i am using asterisk 1.4.39-1 when i call somebody via SIP carrier my asterisk doesnot get his dtmf? any idea |
| 17:23.08 | leifmadsen | wow he waited a whole 3 mins |
| 17:23.37 | Kobaz | haha |
| 17:23.43 | Kobaz | i was gonna write a suggestion too |
| 17:24.48 | tash | as was I |
| 17:24.49 | tash | :P |
| 17:24.51 | tash | oh well |
| 17:25.07 | Qwell | I wasn't. |
| 17:25.28 | Katty | wasn't what |
| 17:25.37 | Katty | did you go the dr |
| 17:25.41 | Qwell | nope |
| 17:25.45 | Katty | whynot |
| 17:25.45 | *** join/#asterisk ska (~ska@cpe-66-68-100-194.austin.res.rr.com) |
| 17:25.48 | Qwell | snow |
| 17:25.58 | Katty | did you reschedule |
| 17:26.04 | Qwell | nope, closed all day |
| 17:26.15 | Katty | are you going to reschedule |
| 17:26.20 | Qwell | mmhmm |
| 17:26.26 | Katty | k lemme know when you go to dr again |
| 17:26.38 | Katty | i want updates |
| 17:26.45 | Kobaz | are you two dating now |
| 17:26.51 | Qwell | maybe |
| 17:27.09 | Kobaz | :( i was hoping katty would pick me |
| 17:27.17 | Katty | katty is enjoyin the single life |
| 17:27.52 | Katty | lunch time! |
| 17:27.53 | Katty | poofs |
| 17:29.52 | leifmadsen | single doesn't mean you aren't having fun |
| 17:29.53 | leifmadsen | :D |
| 17:30.02 | leifmadsen | (with Qwell) |
| 17:31.11 | Kobaz | heh |
| 17:33.09 | Faustov | heard a rumor attractive women are not into IT |
| 17:35.01 | n3hxs | I have seen one or two.... out of how many, maybe millions? |
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| 17:35.32 | Kobaz | maybe billions |
| 17:35.40 | n3hxs | Yeah, more like it. |
| 17:35.56 | Faustov | exceptio probat regulam |
| 17:36.01 | Qwell | leifmadsen: ^5 |
| 17:36.14 | drift- | anyone ever make polycom's use intercoms? |
| 17:36.46 | drift- | http://fonality.com/trixbox/forums/vendor-moderated-forums/polycom/paging-intercom-only-1-way-2-2-and-501s-550s |
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| 17:43.53 | leifmadsen | drift-: yes, you use the Alert-Info header. It has been documented several times on the internets |
| 17:44.03 | leifmadsen | and I think at http://ofps.oreilly.com |
| 17:44.10 | leifmadsen | try the Parking and Paging chapter |
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| 18:02.06 | jordanb | hi |
| 18:02.07 | adeel | heh, chapter 1 of that book seems to be akin to declaring war on AT&T and ma-bell |
| 18:02.26 | adeel | 'Telecommunications companies who choose to ignore Asterisk do so at their peril' ...them are some fighting words |
| 18:02.40 | jordanb | when I press a key on the keypad when dialed through my pbx, I get a loud click sound, followed by the tone, on the other end |
| 18:03.02 | jordanb | this is making it impossible to interact with phone systems that require dial tones because the click gets interpeted as a tone |
| 18:04.46 | madsara | Hey, is there a way to check line voltage on a dadhi interface from a commandline? |
| 18:04.49 | *** join/#asterisk erinspice (~erinspice@nat/digium/x-qykadftdwqcdfkhr) |
| 18:05.29 | madsara | Looks like there used to be a winpipemon command possibly? |
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| 18:11.43 | leifmadsen | :) |
| 18:11.56 | leifmadsen | adeel: no, we are just giving them fair warning :) |
| 18:12.20 | leifmadsen | wanpipemon? |
| 18:12.25 | leifmadsen | madsara: ^^^ |
| 18:12.30 | *** join/#asterisk neurosys (~neurosys@ip67-91-38-226.z38-91-67.customer.algx.net) |
| 18:12.32 | leifmadsen | that would be a wanpipe command (Sangoma hardware) |
| 18:12.40 | madsara | leifmadsen: Yeah, agreed. |
| 18:12.42 | leifmadsen | you might mean dadhi_mon |
| 18:12.48 | leifmadsen | or something like that |
| 18:12.53 | leifmadsen | not sure if such a thing exists |
| 18:12.58 | leifmadsen | check dahdi-tools |
| 18:13.03 | madsara | I can use dahdi_mon for audio levels, seemingly not voltage. |
| 18:13.07 | madsara | Yeah, I don't know if it does either. |
| 18:13.15 | *** join/#asterisk b14ck (~b14ck@cpe-72-129-70-245.socal.res.rr.com) |
| 18:13.30 | madsara | I've got another plan if that's the case. |
| 18:13.49 | adeel | afaik you can't check voltage levels; but i've never really tried |
| 18:13.53 | madsara | Think they've run this line all throughout the building and we're running out of REN basically. |
| 18:14.21 | adeel | simplest way is just to attach a multimeter and read the signal |
| 18:14.41 | madsara | Yeah. |
| 18:14.52 | madsara | That's what I'm getting ready to do. |
| 18:15.44 | madsara | Actually, I'm planning on plugging a dsl modem into each port it's terminated at, place a test call through the asterisk and see if it bridges the call or just goes to dead air (like it has been) |
| 18:16.11 | madsara | I think the issue is happening when the crew downstairs is configuring DSL CPE> |
| 18:16.47 | adeel | that won't tell you the root cause, or verify if it's a physical wiring/hardware issue |
| 18:17.08 | madsara | adeel: It will if I can replicate it 100% of the time. |
| 18:17.14 | adeel | true |
| 18:17.35 | madsara | Although DSL CPE is powered, so a regular phone is probably best for the test. |
| 18:18.05 | madsara | Right now it's intermittent, but I'm 95% sure it's while the line has junk plugged into oit. |
| 18:19.12 | *** part/#asterisk Olivier_54 (~Olivier_5@vaillant.famille-fontes.net) |
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| 18:40.51 | jordanb | I did some more testing, when I press the key on the keypad, I hear the click sound |
| 18:41.09 | jordanb | then for however long I hold the key, I hear nothing, then when I lift it up I hear the tone |
| 18:41.30 | *** join/#asterisk jkroon (~jkroon@dsl-241-227-29.telkomadsl.co.za) |
| 18:41.33 | jordanb | so it seems like asterisk is intercepting the tone, but doing it incompletly (hense the click) then waiting for it to end, then re-sending it |
| 18:42.18 | jordanb | I tried changing dtmfmode to inline on both the inbound and outbound sip connections, with no difference |
| 18:42.50 | jordanb | so it's not rfc2833-related it seems |
| 18:44.46 | fauxalliance | better enable rfc2833 on everything that speaks SIP jordanb..... |
| 18:46.13 | jordanb | fauxalliance: my situation is I"m calling from a touchtone phone and routing through asterisk |
| 18:46.24 | *** join/#asterisk Lipsum (~sengebret@77.40.154.242) |
| 18:46.27 | jordanb | so I call my number, it gets turned into sip by my provider |
| 18:46.35 | jordanb | I route through asterisk and go back out to where I'm calling |
| 18:46.42 | fauxalliance | therefore... use RFC2833 |
| 18:47.04 | fauxalliance | or try a different provider... some are known bad with DTMF |
| 18:47.13 | jordanb | fauxalliance: my sip connections are set to auto, I tried switching to inline to see if that would prevent asterisk from molesting the tones |
| 18:47.32 | jordanb | http://www.voip-info.org/wiki/view/Asterisk+DTMF <-- it looks liek this is my problem |
| 18:47.59 | jordanb | In order to avoid this behaviour in Asterisk, what you can do is to modify the frequency of the DTMF tones Asterisk expects, so that the true DTMF tones are not recognized and therefore let alone by the DSP module. <-- Fucking honestly!?!? That's how that is done?!? |
| 18:48.13 | Qwell | uhh |
| 18:48.15 | fauxalliance | jordanb, I can't see any problem there... perhaps if I were looking at sip debug... |
| 18:48.47 | Qwell | This is why we don't use voip-info. |
| 18:48.51 | Qwell | that. right. there. |
| 18:49.03 | fauxalliance | Qwell, i had no idea... |
| 18:49.09 | jordanb | fauxalliance: it says by default asterisk monitors the connection for tones, and then when it detects one it mutes the line, buffers the tone, and resends it when it ends |
| 18:49.14 | breardo | i'd be sweet if voip-info was current :) |
| 18:49.15 | jordanb | which is exactly the behavior I see |
| 18:49.25 | Qwell | breardo: or even slightly accurate? |
| 18:49.28 | fauxalliance | Qwell, however... my brother can hurt himself with a phonebook |
| 18:49.32 | jordanb | except it lets a tiny bit of the tone through, resulting in the click |
| 18:49.41 | Qwell | jordanb: don't use inband dtmf |
| 18:49.54 | *** part/#asterisk ska (~ska@cpe-66-68-100-194.austin.res.rr.com) |
| 18:50.06 | jordanb | how would I not use it? |
| 18:50.16 | Qwell | <fauxalliance> therefore... use RFC2833 |
| 18:50.26 | jordanb | Is there a way to type tones in from the asterisk console? I could do that. |
| 18:51.38 | fauxalliance | jordanb, sure.. right after you fix DTMF |
| 18:51.54 | jordanb | ... |
| 18:52.17 | jordanb | so my issue is this, I'm calling from a touchtone phone. I call my pbx, go through a little security, and then dial an outside line |
| 18:52.29 | jordanb | it comes into asterisk and back out via sip |
| 18:52.47 | jordanb | it's messing with the tones typed into the keypad of the phone, which is connect to POTS |
| 18:52.48 | fauxalliance | jordanb, SO USE DTMFMODE=RFC2833 ! |
| 18:52.52 | jordanb | I am using that |
| 18:52.59 | jordanb | for the sip portion |
| 18:53.09 | jordanb | I tried turning it off to see if it was causing my problem |
| 18:53.10 | jordanb | it's not |
| 18:53.18 | Qwell | is this an analog phone? |
| 18:53.19 | jordanb | what's causing it is asterisk's internal tone detection thing |
| 18:53.28 | jordanb | yes, this is an analog phone connected to POTS |
| 18:53.58 | Qwell | then yes, you'll hear a slight blip at the beginning of a dtmf |
| 18:54.15 | jordanb | ok how do I prevent that? |
| 18:55.08 | jordanb | this thing says I should break asterisk's tone detection thing by changing the frequencies in the configuration, which seems like a really bad way to do something |
| 18:55.18 | Qwell | no, that is just dumb and wrong |
| 18:55.33 | Katty | hi Qwell |
| 18:55.49 | Qwell | the audio is being sent out before asterisk has a chance to detect that it's dtmf. that's why there is a short blip |
| 18:56.03 | Qwell | tickletackles Katty |
| 18:56.08 | jordanb | so what I want is for asterisk to just not molest the tones at all |
| 18:56.21 | Katty | ^_^ |
| 18:56.41 | madsara | Oh, I hate when asterisk diddles my DTMF |
| 18:56.42 | *** join/#asterisk mayfield (~mayfaild@etwas.baptisthealthsystem.com) |
| 18:56.45 | Katty | Qwell: i went and got riddick dog food over lunch |
| 18:56.54 | Qwell | jordanb: why is this sub-10ms blip an issue? |
| 18:57.03 | Katty | Qwell: but i think it was a bad idea, cause now my fever seems to be coming back |
| 18:57.23 | jordanb | Qwell: I need to call into a teleconferencing application that a client uses every day. That thing detects the blip as a double keypress |
| 18:57.35 | jordanb | Qwell: It didn't used to so much, but it seems that they've made it more sensitive |
| 18:58.02 | Qwell | there's a minimum length to detect a tone. it's horribly broken if it's detecting that. |
| 18:58.16 | jordanb | it's fuzemeeting.com, just so we know who the guilty party is |
| 18:58.39 | Qwell | Are you sure that your provider isn't sending duplicate tones? some are rather bad at that. |
| 18:58.49 | Qwell | Katty: :( |
| 18:59.03 | jordanb | in the alternative, if there was some way for me to call into asterisk, and then connect via the console and type dial tones in there, that would be an acceptable workaround for me |
| 18:59.43 | jordanb | Qwell: No that's not it, I call through my asterisk to my cellphone to listen, the sequence is the blip sound, then silence while I hold the key, and then one fixed-length tone when I lift up. |
| 19:03.27 | Katty | eyes jordanb |
| 19:03.35 | Katty | jordanb: you're not in stl, are you? |
| 19:03.43 | jordanb | no, chicago |
| 19:03.47 | Katty | ah ok. |
| 19:03.51 | Katty | thought perhaps you were someone i knew |
| 19:03.53 | Katty | please disregard! |
| 19:04.08 | jordanb | and vitelity.com is my sip provider, but I can verify that they're not causing the problem by calling through to my cellphone |
| 19:04.11 | jordanb | ahh ok. :) |
| 19:04.25 | Katty | how's the weather up north |
| 19:04.30 | Katty | we got a couple inches down here |
| 19:04.37 | jordanb | very very cold. :< |
| 19:04.44 | Katty | yeah, bout 8ish here too )= |
| 19:04.53 | Katty | luckily it's supposed to warm upto the mid 50s this weekend |
| 19:04.56 | Katty | hang in there (= |
| 19:04.56 | jordanb | man |
| 19:05.01 | jordanb | heh yeah. |
| 19:05.17 | Katty | you ever been up in the sears tower after a huge snow? |
| 19:05.18 | jordanb | It'll probably warm up enough for the snow to melt, and then freeze back up to turn it into ice |
| 19:05.21 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 19:05.28 | Katty | oh god, don't say that :< |
| 19:05.32 | jordanb | I"ve not |
| 19:05.38 | Katty | i wanna see it sometime |
| 19:05.41 | Katty | i bet it's gorgeous |
| 19:05.48 | Katty | if you go, take pictures. |
| 19:05.54 | jordanb | sure thing |
| 19:06.04 | Katty | yay |
| 19:06.30 | Katty | anywho, what are you trying to do? |
| 19:06.35 | Katty | DISA something? |
| 19:07.10 | *** join/#asterisk Olivier_54 (~Olivier_5@vaillant.famille-fontes.net) |
| 19:07.38 | Katty | oh analog phone |
| 19:07.44 | Katty | eeesha, yeah. |
| 19:07.55 | *** join/#asterisk proute (~proute@ARouen-153-1-29-91.w90-17.abo.wanadoo.fr) |
| 19:09.07 | proute | Hello, I used asterisk 1.4.x with misdn. i used with asterisk 1.4 ata gateway to use fax via g711 passthrough. it's work well. Now I uprade with dahdi and asterisk 1.6.2. g711 passthrough don't work. Why? |
| 19:09.34 | Katty | hi proute |
| 19:09.45 | *** join/#asterisk marienz (~marienz@freenode/staff/marienz) |
| 19:10.11 | proute | I tried this patch: https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=17433 but nothing better |
| 19:11.08 | Katty | aww won't even say hi? )= |
| 19:11.52 | proute | hi |
| 19:11.53 | *** join/#asterisk luckman212 (~irc@pool-173-77-253-145.nycmny.fios.verizon.net) |
| 19:11.59 | Katty | :> |
| 19:12.12 | Katty | proute: other than your issue, how is your day going? |
| 19:12.24 | proute | bad and you? |
| 19:12.28 | Kobaz | heh |
| 19:12.32 | Kobaz | aww |
| 19:13.07 | Katty | proute: awww, i'm sorry to hear that )= my day is going too well either. achey, fever, runny nose, sinus infection, and my right ear pressure is all messed up :< |
| 19:13.25 | Katty | proute: i hope your day gets better tho (= |
| 19:13.55 | proute | yes, it's litle better for me |
| 19:13.59 | Katty | yay |
| 19:23.26 | BMJ | Asterisk Developer Call at 5:00 PM EST. Bridge open at 4:45 PM EST. Info here: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Weekly+Developer+Conference+Call |
| 19:31.49 | *** join/#asterisk mcrownover (~markcrown@remote.gawest.com) |
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| 19:47.22 | mechbangirc | how do i pass correct CID between two SIP interconnect asterisk boxes. XXXXXXXXXX---->[PSTN]--->[*-A]----->[*-B]----->[IP PHONE gets XXXXXXXXXX]? |
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| 20:03.23 | Katty | okay asterisk channel |
| 20:03.27 | Katty | i need your advice. |
| 20:03.33 | Katty | i got sick sunday, it is now thursday. |
| 20:03.44 | Katty | i'm still feeling awful, and still running a fever. |
| 20:03.48 | Katty | do i go to the dr, or wait a full week |
| 20:04.24 | tzanger | have you been getting any better? |
| 20:05.37 | Katty | a little, i think, but i've been using tylenol lots |
| 20:05.43 | Katty | it's hard to say |
| 20:05.52 | Katty | it was 102.7 last night, after 1 tylenol it was back down to normal |
| 20:06.15 | Tozz_ | its only 5 days.. it can take a little longer |
| 20:06.33 | Katty | k |
| 20:06.39 | _Corey_ | I've had a cold since Monday... damned season |
| 20:06.52 | Katty | i want summer :< |
| 20:07.01 | Katty | pretty green leaves. |
| 20:07.06 | Katty | and warm sunshine |
| 20:07.18 | _Corey_ | Seriously... I was in Miami last week and it was such a tease |
| 20:07.26 | breardo | katty.. its viral.. they cant do anything.. dont go |
| 20:07.33 | tzanger | yeah if the tylenol wasn't helping (and the fact that one does it tells me it's not terribly serious) I'd wait it out |
| 20:07.39 | breardo | at least, the vicious 2-weeker that went around here recently was viral.. |
| 20:07.45 | Katty | tzanger: okay. |
| 20:07.52 | Katty | don't say that. :< |
| 20:07.52 | breardo | dayquil helped a lot with it |
| 20:07.52 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 20:07.56 | Katty | i have rollerderby next friday |
| 20:08.11 | tzanger | heh I didn't know you rollerderbied |
| 20:08.13 | breardo | if its the same as what went around here, it just DRAINS your energy |
| 20:08.17 | Katty | tzanger: i'm an nso |
| 20:08.23 | tzanger | got any pics or vids of games? |
| 20:08.23 | Katty | tzanger: erm, official |
| 20:08.39 | tzanger | oh... the officials don't have as much fun I don't think |
| 20:09.03 | Katty | that's ok (= |
| 20:09.09 | Katty | i don't wind up with cracked tibia's either |
| 20:09.13 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 20:09.13 | Katty | tzanger: -> |
| 20:09.51 | Katty | tzanger: and we have just as much fun at the after party's and sponsor events. |
| 20:14.05 | *** join/#asterisk hard2get (~hard2get@86.122.227.192) |
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| 20:21.19 | hard2get | hello everyone |
| 20:21.44 | hard2get | could you help me a little? |
| 20:24.39 | hard2get | i have a tdm 844 p board and i createdan extension for an analog phone but i keep getting calls from that extension |
| 20:24.45 | hard2get | on the analog phone |
| 20:24.58 | hard2get | how can i get rid of that |
| 20:25.04 | *** join/#asterisk FLeiXiuS (~unknown@unaffiliated/fleixius) |
| 20:25.36 | FLeiXiuS | How do I define codec priorities in sip.conf? |
| 20:42.04 | leifmadsen | FLeiXiuS: just define them in the order you prefer |
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| 23:10.40 | burnout1540 | does anyone have experience with digium's Fax For Asterisk? or even just faxing over IP in general? |
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| 23:47.54 | eskp | lo |
| 23:48.37 | eskp | i know this is asterisk channel but i'm having some troubles with PoE phones over here, if anyone could help me out or point to another channel? |
| 23:49.09 | eskp | PoE phones seem to turn off randomly every now and then |
| 23:50.34 | Thorn | too much load? |
| 23:51.53 | Kobaz | mananged switch? you can show the voltages |
| 23:52.06 | Kobaz | bad cable? run it through a cable tester? bad switch? |
| 23:52.29 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 23:52.46 | eskp | well i dunno, last time it happened to one connection, i took the phone that turned off and replaced with another which worked :/ |
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