| 00:00.11 | [TK]D-Fender | cracoucas83: Good for you, wrong for Asterisk. |
| 00:00.25 | Katty | someone go to redditroulette.appspot.com and talk to me |
| 00:00.33 | cracoucas83 | well goodbye |
| 00:01.05 | [TK]D-Fender | twit.... |
| 00:03.04 | *** join/#asterisk jks (jks@193.189.93.254) |
| 00:04.56 | *** join/#asterisk hluesea (~hulusikah@88.247.127.66) |
| 00:07.55 | rizwank | Is there a good extension/replacement for the Asterisk CLI - perhaps with tab completion, aliases (I know that's in 1.6.2), and more scriptability? |
| 00:08.18 | *** join/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 00:08.54 | bmoraca_work | the asterisk CLI is meant to be informative-only. what is it you're trying to do but can't? |
| 00:08.56 | rizwank | ok, I just saw the tab completions already in there. So I guess just aliases. |
| 00:09.37 | rizwank | ease debugging. (i.e. write a mini script that does a show channels then does a show channel for each of them.) |
| 00:09.45 | rizwank | i keep thinking of ipython for some reason |
| 00:10.29 | bmoraca_work | you should be able to write a fairly easy script for that...although, it shouldn't really be that necessary... "asterisk -rx" might be helpful for you |
| 00:10.58 | Katty | bored. |
| 00:11.10 | *** part/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 00:16.56 | Reemo | Thanks [TK]D-Fender :) works fine so far. |
| 00:18.30 | Reemo | How can i alter the message played by VoiceMail? |
| 00:21.12 | Reemo | (Without the VoiceMail Menu) |
| 00:22.38 | rizwank | I haven't been able to find any indication - is there an ability to modify asterisk logging to the log file based upon module - i.e. set chan_sip to DEBUG but chan_skype to VERBOSE? |
| 00:29.52 | bmoraca_work | rizwank: probably not, but that also sounds like a really bad idea anyway. you shouldn't need those to be logged to file. if you need to log them temporarily, just save the putty log while you're debugging. your files would be massive and the performance hit would be palpable |
| 00:30.28 | rizwank | it's a different methodoly, pretty pervasive in the Java world. allows for debugging of individual modules pretty well. |
| 00:30.33 | rizwank | *methodology |
| 00:32.36 | bmoraca_work | you missed what I said. i said that you would not want that information logged to the asterisk log files. observing it in the console is about all you ever need to do |
| 00:33.00 | *** join/#asterisk came0_ (~came0@34.124.188.72.cfl.res.rr.com) |
| 00:34.48 | dlynes | Anyone heard of a US credit union called 'Pen Fed'? |
| 00:37.01 | rizwank | I can think of why you would want it on a log file in enterprise situations, but it doesn't matter. |
| 00:38.32 | *** join/#asterisk rajiv (~rajiv@gentoo/developer/rajiv) |
| 00:39.27 | bmoraca_work | rizwank: the performance hit on the filesystem would severely limit the scalability of the system if you're logging DEBUG messages from chan_sip or verbose messages from almost anything else. |
| 00:40.01 | rizwank | that's fine. I didn't suggest that it should be a default setting. |
| 00:40.52 | bmoraca_work | rizwank: in an enterprise system, you don't need to be constantly monitoring. if issues are not reproducable, they're almost always user error, and therefore not visible in debug logs. the only thing you need to log are errors and warnings, because those are your real problems. your dialplan should be such that every call is nearly identical. |
| 00:42.13 | rizwank | I misspoke when when I said enterprise. Lets say you sell an application based upon asterisk, and almost everything works, but meetme performs oddly. And it's hard to reproduce. You'll want to verify meetme debug data without everything else. (You may want it, but it'd be spammy if it wasn't needed.) |
| 00:45.53 | Kobaz | bmoraca_work: i highly disagree |
| 00:45.55 | bmoraca_work | rizwank: i use the console for that if and when it happens (logged to my local machine if necessary)...additionally, iif meetme throws an error or a warning, it'll show up in my logs anyway. |
| 00:46.24 | Kobaz | i've found at least 3 problems so far that were *incredibly* hard to reproduce, to the point where it seemed completely random |
| 00:46.28 | rizwank | okay. That works for you, but it's inconceivable that a different pattern would work better for someone else? |
| 00:46.31 | Kobaz | and it was indeed bugs/limitations in asterisk |
| 00:46.55 | bmoraca_work | Kobaz: if you want to log 150 extra lines every call and 20-30 lines every few seconds for keepalives, be welcome. but file logging in a production system should be kept to an extreme minimum. |
| 00:47.09 | bmoraca_work | rizwank: not inconceivable...simply ill-advised. |
| 00:47.20 | Kobaz | i use almost full debug logging on all my production systems |
| 00:47.31 | Kobaz | it's helped catch I can't count how many problems |
| 00:48.02 | Kobaz | logging is better than no logging, and full logging is even better |
| 00:48.11 | bmoraca_work | Kobaz: my production systems don't HAVE problems that need to be caught. most issues are user issues that a quick glance at the console can diagnose. |
| 00:48.14 | Kobaz | i'm a big fan of logging everything you possibly can |
| 00:48.27 | bmoraca_work | Kobaz: i can't think of the last time i actually looked at an asterisk log. simply no point |
| 00:48.33 | Kobaz | bmoraca_work: correction... you haven't ran into any problems that needed catching yet |
| 00:48.41 | rizwank | exactly. |
| 00:48.44 | rizwank | http://confluence.atlassian.com/display/JIRA/Changing+JIRA%27s+log+output |
| 00:48.52 | rizwank | lots of modules, broken up, each with their own loggers |
| 00:48.53 | bmoraca_work | 5000 calls/day...i don't think it's going to happen. |
| 00:48.54 | rizwank | and a plugin system |
| 00:49.11 | Kobaz | bmoraca_work: i handle about that on one of my systems... lots of logs |
| 00:49.17 | rizwank | if a 3rd party dev wants you to send him logs of when his plugin messes up, how do you handle that? |
| 00:49.56 | rizwank | and the I/O that you're talking about isn't massive. It doesn't scale well, but that doesn't mean that it shouldn't be possible. |
| 00:49.58 | bmoraca_work | Kobaz: what made you think i was talking about more than one system? i supervise about 35 asterisk systems on a daily basis and don't have any problems with either and don't need to look at asterisk logs on any of them. |
| 00:49.59 | Kobaz | if someone calls up yelling about some problem.. you say... when did it happen... what extension were you sitting at... who called you... and when during the call did it happen |
| 00:50.32 | Kobaz | and within a few seconds you can find the spot in the logs relevant to the call, and usually you can find the problem |
| 00:50.35 | rizwank | Okay, you don't need it. Your dialplans are perfect, and you've never found a bug in a channel module, hardware driver, or asterisk. Bravo. |
| 00:50.36 | Kobaz | if it was user error or not |
| 00:50.57 | bmoraca_work | Kobaz: no, I log into the console and say "do it again". if they can't, it's not a problem. if they can, then I see exactly what the problem is. much more concise than wading through 100s of megs of data to find one line that may or may not be at the time they said (around 10:00am is rather imprecise) |
| 00:51.09 | Kobaz | that's bad |
| 00:51.26 | bmoraca_work | no, that's good, because i can resolve 99% of problems immediately, and without hours of legwork |
| 00:51.46 | Kobaz | "if they can't, it's not a problem" |
| 00:51.55 | Kobaz | i just think that's a bad mentality |
| 00:52.13 | bmoraca_work | is the truth. not reproducable = not a problem. this comes from experience and very good dialplan programming |
| 00:52.16 | Kobaz | i've seen problems crop up in asterisk where i say... make it happen again |
| 00:52.23 | Kobaz | this isn't dialplan i'm talking about |
| 00:52.28 | Kobaz | this is anything to do with asterisk |
| 00:52.54 | bmoraca_work | problems don't magically happen with a static system. if they do, you either fucked with something or someone else fucked with something. |
| 00:53.00 | Kobaz | yes they do |
| 00:53.04 | Kobaz | perfect example |
| 00:53.10 | bmoraca_work | but, whatever, i'm going home. arguing with novices sets my teeth on edge. |
| 00:53.17 | Kobaz | i was getting audio dropouts in random intervals |
| 00:53.31 | Kobaz | i started logging even more stuff |
| 00:53.45 | Kobaz | pci interrupts, disk activity, io throughputs, cpu load, everything |
| 00:53.49 | Kobaz | until i finally found the problem |
| 00:54.04 | bmoraca_work | and you saw you fucked up a config. big deal. don't fuck up configs and you don't have a problem. |
| 00:54.07 | Kobaz | nope |
| 00:54.11 | bmoraca_work | yep. |
| 00:54.13 | Kobaz | well, i found a limitation in asterisk |
| 00:54.13 | bmoraca_work | bye. |
| 00:54.24 | Kobaz | Monitor() runs in the same thread as audio bridging |
| 00:54.32 | Kobaz | and is heavily affected by high disk i/o |
| 00:54.55 | Kobaz | so there you have it |
| 00:54.57 | Kobaz | logging to the rescue |
| 00:55.21 | bmoraca_work | underestimating hardware requirements = fully preventable config problem. |
| 00:55.33 | rizwank | is there a pre-requisite requirement to core set debug 50 sip working? I set it and I'm not getting any sip debug messages in the log file, even though it's logging verbose. |
| 00:55.34 | Kobaz | you mean being an expert in the asterisk source |
| 00:55.49 | Kobaz | MixMonitor() does not suffer the same problem |
| 00:55.53 | rizwank | nice find Kobaz. |
| 00:55.57 | Kobaz | and in no where is this documented |
| 00:56.25 | Kobaz | MixMonitor() spawns a seperate thread for writing out the audio, and does not affect call bridging with high io |
| 00:56.28 | Kobaz | so |
| 00:56.38 | Kobaz | that's great that you've never had a problem |
| 00:56.56 | Kobaz | and i wish you lots of luck, that if you do have a problem, your lack of logging won't hinder you |
| 00:57.29 | Kobaz | rizwank: yeah, it took me about 5 months to find that problem |
| 00:57.53 | Kobaz | since it was one of those... 'can you reproduce it now?' 'no' sort of problems |
| 00:58.12 | *** join/#asterisk Caplain (~shayne@shayne.caplain.loves.boys.fbi.gov.silverelitez.org) |
| 00:58.43 | Kobaz | rizwank: you'll need to turn on debug logging too, verbose is seperate |
| 00:59.01 | Kobaz | verbose,debug,etc,etc... in logger.conf |
| 00:59.22 | rizwank | full => notice,warning,error,debug,verbose |
| 00:59.35 | rizwank | it works sometimes, other times it doesn't. restarting asterisk seems to help, oddly. |
| 00:59.39 | Kobaz | mm |
| 00:59.41 | Kobaz | dunno |
| 00:59.47 | Kobaz | anyways, busting out of the office |
| 00:59.50 | Kobaz | happy logging |
| 01:02.04 | rizwank | cheers. |
| 01:05.45 | rizwank | ah, needed sip set debug. |
| 01:05.46 | rizwank | on |
| 01:06.50 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 01:12.58 | *** join/#asterisk came0___ (~came0@34.124.188.72.cfl.res.rr.com) |
| 01:25.11 | *** join/#asterisk nny (~scott@64.203.239.83) |
| 01:25.38 | nny | hmm am I missing something here? If I create sym links to one mailbox for three users, shouldn't notifications etc work the same for all 3 devices? |
| 01:26.12 | nny | asterisk says that all 3 users have a vm, but only the phone who's box i have linked goes MWi after the message |
| 01:26.49 | nny | logging into the vm from the phone changes the status... |
| 01:26.51 | nny | hrrrm |
| 01:27.40 | *** join/#asterisk Kumbang (~kumbang@rusnas.paume.itb.ac.id) |
| 01:27.56 | nny | gonna mv the file back and forth and see how it acts |
| 01:29.20 | nny | odd |
| 01:29.34 | nny | maybe a setting in voicemail.conf I don't have time to check what the conf file could* have |
| 01:29.55 | nny | moved the file. asterisk says 0 msg, phone had mwi still |
| 01:30.00 | nny | has* |
| 01:36.44 | jaytee | The Night Chicago Died is the official Canadian national anthem |
| 01:36.57 | jaytee | O Canada is just a song they like to sing at hockey games |
| 01:37.23 | *** part/#asterisk korihor (~korihor@190.205.251.97) |
| 01:42.01 | nny | so trying to troubleshoot this MWi issue, am I to assume asterisk sends some kind of SIP packet to the phone when the MWI is supposed to change? |
| 01:43.09 | *** join/#asterisk voipmonk (~shido6@CPE002191f85581-CM001692568382.cpe.net.cable.rogers.com) |
| 01:43.52 | carrar | nny,something like RFC 3842 maybe? |
| 01:44.22 | drmessano | jaytee, I thought the national anthem of canada was "Bryan Adams Greatest Hits" oh, wait |
| 01:44.34 | carrar | Cuts like a knife |
| 01:44.40 | carrar | but it feels so right! |
| 01:44.40 | nny | carrar: hmm ok have to do a sip debug and see if asterisk sends a packet when the INBOX folder is populated with a message, right now phones seem to ignore it some how |
| 01:45.45 | nny | hmmm |
| 01:45.45 | nny | ok |
| 01:46.08 | nny | no packets ? |
| 01:46.49 | nny | http://pastebin.com/m7f690128 |
| 01:48.43 | nny | mv'd a file in, asterisk updates vm info, but don't see any packet lovin |
| 01:48.49 | nny | story of my life |
| 01:49.15 | staffmember | can someone help me out with something... I can receive calls to an extension.. but when i pick up that extension and start to dial, i get a busy signal |
| 01:49.21 | staffmember | anyone know what that may be about |
| 01:49.29 | carrar | Why are you using sym links? |
| 01:49.56 | carrar | Why not define all 3 phones with the same voicemail 100@default |
| 01:49.57 | nny | carrar: small company, no need for seperate voicemails for one user and 3 phones |
| 01:50.12 | nny | carrar: can try that too, didn't know the symlink would cause an issue |
| 01:50.24 | nny | carrar: saw it suggested somewhere, seemed simple :\ |
| 01:50.24 | carrar | symk is ass backwards ways |
| 01:50.44 | carrar | mailbox=100@default |
| 01:50.47 | carrar | for all 3 phones |
| 01:51.17 | nny | yeah also have to change voicemail dial structure and goto for how the user is dialed |
| 01:51.47 | nny | basically make sure if the user dials *7 it goes to the 100 box, and not ${EXTEN}@default |
| 01:51.56 | nny | i'll try that |
| 01:52.41 | voipmonk | yep |
| 01:52.55 | voipmonk | or use callerid instead of exten |
| 01:52.59 | *** join/#asterisk etnos (etnos@c-75-74-66-161.hsd1.fl.comcast.net) |
| 01:53.13 | voipmonk | or a special variable in sip.conf like setvar=callerid2=100 |
| 01:53.20 | voipmonk | and use ${callerid2} |
| 01:53.20 | *** join/#asterisk came0 (~came0@32.134.134.168) |
| 01:54.11 | etnos | has anyone ever used veritas netbackup? |
| 01:55.16 | nny | is mv a reliable way to test vm notification? Always seem to have worked before |
| 01:56.08 | nny | did: set mailbox=100 in sip.conf, mv .txt and .wav file to 100 INBOX in default, voice show users says "1 msg" no mwi :\ |
| 01:56.21 | nny | voicemail show users* |
| 01:57.37 | nny | asterisk restart and MWI lights up immediately |
| 01:58.06 | nny | and same if I empty the box and restart :\ |
| 01:58.24 | nny | is puzzled |
| 02:00.34 | *** join/#asterisk albertoandrade (~albertoan@189.34.121.136) |
| 02:01.29 | nny | well, using the voicemail system to delete it works at least |
| 02:01.30 | nny | game on! |
| 02:02.27 | nny | however vmail.cgi doesn't.. damn |
| 02:02.33 | Corydon76-dig | nny: If you mess with the mailbox outside of Asterisk, then you need to turning polling on in voicemail.conf |
| 02:02.42 | nny | Corydon76-dig: ahh my hero |
| 02:03.06 | nny | Corydon76-dig: is this the way it's always been? I have the webvmail ap installed in some places, didn't know that |
| 02:03.14 | Corydon76-dig | nny: the change was made to only sent mwi updates when Asterisk knows a change was made. This saves considerable CPU, when possible |
| 02:03.20 | nny | make ssense |
| 02:03.22 | nny | sense* |
| 02:03.38 | nny | although the web vmail thing not working is a bit of a suprise, considering the source :D |
| 02:03.51 | Corydon76-dig | I think the option is something like pollmailboxes=yes |
| 02:04.38 | nny | is the CPU savings exponential to the amount of mailboxes it has? |
| 02:04.46 | nny | i mean i would think so, just curious |
| 02:05.03 | Corydon76-dig | It doesn't trace like that |
| 02:05.56 | Corydon76-dig | There's a near-constant CPU load when it has to poll mailboxes every N seconds. And nearly 0, when it only sends MWI on a change |
| 02:06.32 | nny | just set it to 7 seconds, nothing special seems to have happened via top |
| 02:07.26 | Corydon76-dig | That's because disk access is defined as a long wait state, which does not go into load average calculations |
| 02:07.41 | nny | well good news is this has a 32gb ssd heh |
| 02:08.38 | Corydon76-dig | but it still has to scan the directories, which is somewhat expensive in terms of kernel time |
| 02:09.08 | Corydon76-dig | It's less expensive than it used to be, for app_voicemail, but it's still expensive |
| 02:09.15 | nny | running a pretty decent processor for our small systems, just trying to confirm it's not gonna go hungry hungry hippo on my cpu |
| 02:09.42 | nny | did top pXXXX where XXXX is my asterisk PID and it seems.. quiet |
| 02:09.45 | Corydon76-dig | Used to run a predictive filename algorithm; now it reads the entire directory once and uses a bitmap |
| 02:10.00 | nny | ahh so I need to make changes |
| 02:10.42 | Corydon76-dig | Well, SSD doesn't exactly make you wait, either |
| 02:10.46 | nny | indeed |
| 02:10.56 | nny | just making sure 7 seconds isn't gonna cause issues |
| 02:10.59 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 02:12.07 | nny | odd |
| 02:12.23 | nny | pollmailboxes=no pollfreq=7 |
| 02:12.27 | nny | oh crap |
| 02:12.28 | nny | nm |
| 02:12.30 | nny | sigh |
| 02:12.34 | nny | shoot me |
| 02:12.42 | nny | no means yes! |
| 02:13.00 | *** join/#asterisk SaiSoma (~SaiSoma@adsl-074-167-136-030.sip.mob.bellsouth.net) |
| 02:13.50 | nny | that works great |
| 02:13.56 | nny | Corydon76-dig: thank you |
| 02:14.01 | Corydon76-dig | nny: yw |
| 02:14.39 | nny | carrar: and voipmonk ty as well for the suggestion on mailbox vs symlinks, should have known better than to use the first google hit for asterisk shared mailbox and get a freepbx page :\ |
| 02:14.56 | nny | (i actually tried the mailbox option first, but thought I was doing it wrong lol) |
| 02:15.28 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 02:23.06 | nny | now to figure out why my second dahdi channel on my sangoma card is acting odd |
| 02:26.06 | nny | may be a dead channel heh |
| 02:27.57 | troy- | are there any Canadian LECs that sell origination? |
| 02:28.11 | voipmonk | back |
| 02:28.25 | troy- | there is a carrier called back? |
| 02:30.40 | nny | not sure if this is what you mean, but I know I have used les.net before |
| 02:30.48 | dlynes | les.net sucks, too |
| 02:30.58 | troy- | they are indeed terrible |
| 02:31.18 | troy- | dlynes: any suggestions? |
| 02:31.29 | dlynes | troy-, I use allstream and navigata |
| 02:31.49 | troy- | can you sell me service? |
| 02:31.52 | dlynes | troy-, they're top notch |
| 02:32.02 | dlynes | No, I can't...I don't own the companies |
| 02:32.02 | troy- | i'm just a hobby user that wants reliable service |
| 02:32.08 | troy- | ahh okay |
| 02:32.12 | dlynes | I just work there, and they don't sell to residential users |
| 02:32.24 | dlynes | One company only sells to offices |
| 02:32.29 | dlynes | The other only sells to care homes |
| 02:32.47 | troy- | dlynes: does it help that i have legal entities? |
| 02:32.58 | troy- | so i can sign up as a corp |
| 02:33.08 | dlynes | For companies that specialize in residential/hobby service, their quality is usually crap |
| 02:33.18 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 02:33.18 | dlynes | well, that's not why they don't do residential |
| 02:33.21 | troy- | exactly.. that is why i am in this position |
| 02:33.33 | dlynes | they don't do residential, because of economies of scale |
| 02:33.42 | dlynes | residential only need one line, and that's a money loser |
| 02:33.55 | troy- | i agree completely, figured maybe you could do a one off |
| 02:34.05 | dlynes | if i was the owner, possibly |
| 02:34.08 | dlynes | but i'm not |
| 02:34.13 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 02:34.13 | troy- | :/ |
| 02:34.28 | dlynes | both companies are in vancouver, too |
| 02:34.34 | troy- | i'm in Toronto |
| 02:34.43 | dlynes | Yeah...we do Toronto dids as well |
| 02:34.44 | troy- | but latency to vancouver isnt significant |
| 02:34.55 | dlynes | I'm in Hamilton |
| 02:35.02 | troy- | i live in hamilton too :P |
| 02:35.05 | *** join/#asterisk Caplain (shayne@shayne.caplain.loves.boys.fbi.gov.silverelitez.org) |
| 02:35.06 | dlynes | ah...weird |
| 02:35.09 | troy- | <-- McMaster student |
| 02:35.12 | dlynes | cool |
| 02:35.18 | dlynes | you looking for res for next year? |
| 02:35.27 | troy- | i'm actually transferring to Ryerson |
| 02:35.34 | dlynes | We're trying to find a group of 3 for our basement |
| 02:35.48 | voipmonk | what is McMaster? is that some kind of McDonald sales program? |
| 02:36.00 | troy- | haha i live in the orange condo building near university plaza |
| 02:36.08 | dlynes | voipmonk, one of the top universities in north america |
| 02:36.28 | dlynes | troy-, not familiar with it |
| 02:36.42 | dlynes | troy-, we've got a triplex student rental over on ward we're closing on the end of the month |
| 02:36.51 | voipmonk | how many universities are considered "top" ? |
| 02:37.04 | troy- | nice, if i was staying in hamilton.. :) |
| 02:37.32 | dlynes | troy-, well, i feel sorry for your wallet for next year |
| 02:37.41 | troy- | as do i :/ |
| 02:37.43 | dlynes | troy-, it's gonna be hurtin' from downtown toronto rent |
| 02:38.09 | troy- | i'm debating living at home but i'd really prefer not to |
| 02:38.11 | dlynes | It's about $1200/mo for a whole in the wall |
| 02:38.15 | dlynes | erm hole |
| 02:38.16 | troy- | haha yeah |
| 02:38.49 | dlynes | no amount of money in the world would convince me to move to toronto :) |
| 02:39.10 | troy- | well i was brought up here and lived in nyc for awhile |
| 02:39.14 | dlynes | ah |
| 02:39.15 | troy- | hamilton was a tough adjustment |
| 02:39.21 | dlynes | just got a job at mcmaster innovation park |
| 02:39.24 | dlynes | so I'm quite happy |
| 02:39.28 | troy- | nice |
| 02:39.37 | troy- | i just got a job too :P |
| 02:39.38 | dlynes | Yeah...working for a nice company, too |
| 02:39.48 | troy- | with a bank |
| 02:39.57 | dlynes | We're putting together some pretty cool technology |
| 02:40.11 | dlynes | So voip'll be taking a back seat |
| 02:41.12 | dlynes | I'm tired of troubleshooting shit I have no control over |
| 02:41.13 | troy- | heh arent we all |
| 02:41.13 | dlynes | so you were taking telecom at mac? |
| 02:41.25 | troy- | i'm in the commerce program right now |
| 02:41.32 | troy- | but switching to the comm / it management at ryerson |
| 02:41.57 | dlynes | ah |
| 02:42.12 | troy- | taking the reputation hit to work downtown for a big investment bank supporting trading systems |
| 02:42.26 | dlynes | ah...cool |
| 02:42.46 | troy- | but it sucks when you have a phone number on your resume that only works half the time |
| 02:42.47 | dlynes | TD, or Wood Gundy, or something? |
| 02:43.00 | dlynes | troy-, well, you're moving back to T.O. |
| 02:43.09 | dlynes | get a Fido City plan |
| 02:43.19 | troy- | well i want to keep a voip number to frontend the cell phone |
| 02:43.21 | dlynes | all you can eat |
| 02:43.29 | dlynes | or WindMobile for that matter |
| 02:43.31 | troy- | i have cisco 7941s all over |
| 02:43.35 | troy- | windmobile sucked really hard.. |
| 02:43.37 | dlynes | and you can get all you can eat internet as well |
| 02:43.42 | dlynes | really? |
| 02:43.51 | dlynes | What's wrong with WindMobile? |
| 02:44.04 | dlynes | I was thinking about switching to them in April when my fido contract expires |
| 02:44.18 | *** join/#asterisk came0 (~came0@167.83.189.72.cfl.res.rr.com) |
| 02:44.33 | troy- | yeah, busy trunk ingress/egrees, broken call forwarding, no blackberry tether, blocked ports, deadzones, etc. |
| 02:45.07 | dlynes | cs so fido and rogers don't have any of that? |
| 02:45.23 | troy- | this was far more pervasive |
| 02:45.42 | dlynes | ah |
| 02:48.35 | *** join/#asterisk OrNix (~ornix@l151-249-47.static.cn.ru) |
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| 03:34.07 | *** join/#asterisk SeriousMatters (~5773aad2@gateway/web/freenode/x-pbowujrobxcvrffq) |
| 03:34.27 | SeriousMatters | Hi all. |
| 03:34.55 | LemensTS | how do you get channel variables to carry over to a parked call pickup? |
| 03:35.08 | voipmonk | __ |
| 03:35.27 | *** join/#asterisk Kumbang (~kumbang@125.163.83.153) |
| 03:35.50 | voipmonk | set a variable |
| 03:35.55 | voipmonk | use the astdb or mysql |
| 03:38.43 | *** join/#asterisk `Sauron (sauron@dsl001-130-033.aus1.dsl.speakeasy.net) |
| 03:41.43 | SeriousMatters | is Asterisk1.6.x new release (2 weeks ago) available as rpm yet? |
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| 03:51.14 | m477au | anyone around that can help me out with a B410P under elastix? |
| 03:51.25 | m477au | incoming working perfectly, outgoing is the issue |
| 03:55.27 | *** join/#asterisk nightrid3r (kvirc@41.214.133.29) |
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| 04:17.22 | *** part/#asterisk LemensTS (~customgt@adsl-70-238-130-71.dsl.stlsmo.sbcglobal.net) |
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| 04:59.40 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 05:01.48 | Katty | HELLO THAR PEEPS |
| 05:03.35 | *** join/#asterisk xmitter (~xmitter@c-24-21-213-242.hsd1.or.comcast.net) |
| 05:04.01 | Katty | snobs. |
| 05:07.43 | styelz | :P |
| 05:08.00 | styelz | HI!!!! |
| 05:13.01 | SeriousMatters | I am pretty new to asterisk |
| 05:13.38 | p3nguin | looks at katty |
| 05:13.53 | Deeewayne | hugs Katty |
| 05:14.28 | SeriousMatters | a bug i am encountering is fixed by the most recent release, but not yet available as rpm package. therefore compiling asterisk for first time. |
| 05:14.47 | SeriousMatters | on a pentium 3 machine... ... its taking forever |
| 05:15.31 | p3nguin | I run a PIII 933MHz box, so I can understand why you feel that way. |
| 05:16.25 | p3nguin | I'm thinking it takes around 20 minutes for me. |
| 05:16.58 | SeriousMatters | mine is 600Hz ish. |
| 05:17.20 | p3nguin | So you're looking at 30+ minutes. |
| 05:17.22 | SeriousMatters | I guess it means 30 minutes for me. |
| 05:18.55 | SeriousMatters | the number of lines of .c 's already overflew my ssh client cache. |
| 05:19.48 | p3nguin | cache, or scroll buffer? |
| 05:20.45 | p3nguin | You can increase the number of lines of scrollback in PuTTY, if that's what you're using. |
| 05:21.09 | SeriousMatters | using ssh.com's |
| 05:21.41 | SeriousMatters | a free version they gave out years ago. |
| 05:22.21 | SeriousMatters | it has scp |
| 05:23.03 | SeriousMatters | anyway, just killing time while make is running. |
| 05:23.24 | p3nguin | I guess you know about checkinstall. |
| 05:24.02 | SeriousMatters | i don't actually. |
| 05:24.11 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-mrrqfqpihjiewxml) |
| 05:25.28 | p3nguin | Any sane admin prefers to use packages for software installation, but certain things need to be built from sources. checkinstall can be used to "record" the files into an RPM rather than just throwing them into the file system with make install. |
| 05:26.23 | SeriousMatters | I see, but its too late now. =P |
| 05:26.45 | p3nguin | Yeah, because you really should set the prefix and sysconfdir values when you configure it. ./configure --prefix=/usr --sysconfdir=/etc |
| 05:26.46 | SeriousMatters | thanks for the advice tho. I will remember to use checkinstall next time |
| 05:26.53 | *** join/#asterisk JAMMAN2110 (~james@unaffiliated/jamman2110) |
| 05:27.26 | p3nguin | You can still use checkinstall in lieu of make install, now... the files will just be in a non-standard location for packaged software. |
| 05:28.10 | p3nguin | Packaged things don't typically go in /usr/local, but most things built from source do. |
| 05:28.26 | funtoo_nbu | [Feb 15 07:20:37] ERROR[25119] chan_sip.c: Serious Network Trouble; __sip_xmit returns error for pkt data |
| 05:28.33 | funtoo_nbu | anyone know what this is? |
| 05:28.52 | funtoo_nbu | using Asterisk 1.6.1.12 |
| 05:30.55 | *** join/#asterisk Brack10 (~travis@68.186.68.71) |
| 05:31.35 | Brack10 | Hey. Maybe you guys will know this. Who has a better server infrastructure in the US, skype or Vonage? Is skype bearable in the US for calling regular phones? |
| 05:32.48 | Corydon76-dig | Information on those companies' server infrastructure is not available |
| 05:33.47 | Brack10 | so what's the best voip service for people in the US? I want to be able to call on Windows, Linux and iPhone (wifi) |
| 05:34.49 | Corydon76-dig | Then you want neither of those providers |
| 05:34.57 | Corydon76-dig | You want a SIP provider |
| 05:35.10 | SeriousMatters | gizmo? |
| 05:35.19 | funtoo_nbu | Corydon76-dig: any idea what my error is? |
| 05:35.56 | Corydon76-dig | funtoo_nbu: nope |
| 05:36.38 | p3nguin | Gizmo5 still isn't accepting new registrations. |
| 05:36.40 | Corydon76-dig | but that error looks pretty bad |
| 05:37.53 | SeriousMatters | oh, Gizmo isn't? I have no idea, I'm in UK. |
| 05:38.21 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-145.cablep.bezeqint.net) |
| 05:39.48 | funtoo_nbu | Corydon76-dig: aye, but asterisk works |
| 05:40.23 | funtoo_nbu | i have it all setup quite well, 6 sip phones provisioned all using sip locally and aswell as globally |
| 05:40.49 | funtoo_nbu | i do have some weird quirks which i cant explain, so i am trying to find out about some weird errors |
| 05:40.54 | funtoo_nbu | that is one of them :( |
| 05:41.00 | funtoo_nbu | google is not very helpful either |
| 05:41.02 | Corydon76-dig | Firewall, perhaps? |
| 05:41.28 | funtoo_nbu | mayhap |
| 05:41.37 | funtoo_nbu | asterisk is running on the firewall/nat box |
| 05:42.22 | funtoo_nbu | sucks that I cant find out much info except "this error was fixed in version X" |
| 05:42.24 | *** join/#asterisk alexsea7 (~Sheeju_Al@122.169.155.22) |
| 05:43.26 | *** part/#asterisk alexsea7 (~Sheeju_Al@122.169.155.22) |
| 05:43.57 | Corydon76-dig | funtoo_nbu: then upgrade to current |
| 05:44.20 | funtoo_nbu | that introduces more variables |
| 05:44.26 | funtoo_nbu | the system is up on a business location |
| 05:44.33 | funtoo_nbu | no phones = angry boss |
| 05:45.08 | Corydon76-dig | I don't mean bleeding edge current. I mean current 1.6.1 |
| 05:45.49 | funtoo_nbu | im on 1.6.1.12 |
| 05:45.57 | Corydon76-dig | Either SVN 1.6.1 or 1.6.1.15-rc2 |
| 05:46.48 | funtoo_nbu | who those? |
| 05:47.12 | funtoo_nbu | why* lol |
| 05:47.38 | Corydon76-dig | Because those have the latest and greatest bug fixes |
| 05:47.40 | *** part/#asterisk benngard (~benngard@90-230-92-67-no148.tbcn.telia.com) |
| 05:48.43 | funtoo_nbu | haha gentoo has the latest stable as net-misc/asterisk-1.2.37 |
| 05:48.56 | *** join/#asterisk xmitter (~xmitter@c-24-21-213-242.hsd1.or.comcast.net) |
| 05:49.15 | funtoo_nbu | where to find these releases? not on main page |
| 05:49.39 | Corydon76-dig | http://downloads.asterisk.org/pub/telephony/asterisk/ |
| 06:07.48 | yang | I am wondering has anyone an example of Queues ? |
| 06:09.18 | ChannelZ | stands in a line |
| 06:10.01 | SeriousMatters | ehh, like a bus stop? =P |
| 06:11.03 | SeriousMatters | is this a correct place to ask about openSER? |
| 06:21.17 | *** join/#asterisk Kumbang (~kumbang@rusnas.paume.itb.ac.id) |
| 06:23.19 | zamba | i'm getting messages like this in my syslog: rc_avpair_new: unknown attribute 1490026597 |
| 06:23.22 | zamba | they are from asterisk |
| 06:23.28 | zamba | what's causing that? |
| 06:23.29 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 06:26.08 | yang | I am getting such error with Queues - http://pastebin.com/mcb025ed |
| 06:30.48 | yang | SeriousMatters: #openser |
| 06:30.53 | yang | ~openser |
| 06:30.54 | infobot | [openser] an open source GPL project that aims to develop a robust and scalable SIP server. It is spawned from FhG FOKUS SIP Express Router (SER) and it promotes a development strategy open for contributors and contributions. From project's website http://www.voip-info.org/wiki/view/About+OpenSER, or now known as Kamailio |
| 06:31.34 | SeriousMatters | thanks, |
| 06:31.48 | SeriousMatters | I want to ask something about the usage. |
| 06:32.47 | *** join/#asterisk cuco (~Diego@local.xorcom.com) |
| 06:33.26 | *** join/#asterisk smooth_penguin (~smoove@59.95.23.32) |
| 06:33.46 | SeriousMatters | I need NAT traversal solution such as MediaProxy. |
| 06:40.38 | *** join/#asterisk ChannelZ (channelz@173.160.35.173) |
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| 07:01.18 | jaybinks | hey can someone with cisco voice gateway experience, give me 5 minutes ? |
| 07:05.17 | ChannelZ | yang: your queues.conf might not be getting parsed right with no [general] section |
| 07:06.39 | *** part/#asterisk Brack10 (~travis@68.186.68.71) |
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| 07:13.10 | carrar | Y*A*W*N |
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| 07:20.40 | *** join/#asterisk infinity1 (~brendon@web2.artsopolis.com) |
| 07:21.23 | infinity1 | if i were to use 100,000 minutes per month for termination, who would i call for pricing? is it possible to get less than 1.5 cents per minute? |
| 07:23.10 | *** join/#asterisk e-jones (~jkastner@nat/redhat/x-spofwxneuutewgor) |
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| 07:54.03 | awk | Hi, how many calls can one get on a 512KB line with g729 codec 16 calls? |
| 07:54.18 | jkroon | hi guys, I've got a problem where two SIP legs negotiates (I think) early media, and then asterisk (1.6.1.14) ends up leaking the RTP socket going towards the caller. SIP trace at http://pastebin.co.za/82853 |
| 07:54.36 | jkroon | well, for the initiating leg of the call anyway. |
| 07:58.48 | jkroon | would it be possible for someone to take a peek at that SIP trace and give me an indication of what could be wrong with my configuration? |
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| 08:28.38 | Antonjs | Hey all, I'm having some extremely strange issues with sip connections if anyone has any ideas |
| 08:28.47 | ChannelZ | You should fix them |
| 08:29.04 | Antonjs | I see SIP INVITE packets coming into the server via tcpdump and a port mirror on my switch |
| 08:29.19 | *** part/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 08:29.20 | Antonjs | but asterisk seems to see the packets a solid 15-20 seconds after the system's tcp stack does |
| 08:29.39 | Antonjs | after the SIP client has given up... |
| 08:36.10 | jkroon | Antonjs, are you sure asterisk isn't seeing the resend? |
| 08:36.41 | *** join/#asterisk Antonjs (~anton@mm-headoffice.clients.fishnet.net.nz) |
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| 09:17.37 | *** join/#asterisk pentanol (~pentanol@77.35.48.247) |
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| 09:31.01 | pentanol | hi, anybody use sip auth via radius into database? |
| 09:33.07 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
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| 09:41.07 | *** join/#asterisk Dovid (~annon@tony09-118-62.inter.net.il) |
| 09:41.39 | Dovid | what is the prefferd method for Privacy Signalling in Asterisk |
| 09:44.47 | kaldemar | Dovid: core show function CALLERPRES |
| 09:49.11 | *** join/#asterisk maxagaz (~maxagaz@soho2.i-xanadu.com) |
| 09:50.09 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 09:50.17 | *** join/#asterisk e-jones (~jkastner@nat/redhat/x-vyxpqvkxqvsdzvfg) |
| 09:51.21 | Dovid | kaldemar: I am filling out a form for an ITSP. options are Stripped A-Number, PAI and Privacy Flag, RPID and Privacy flag |
| 09:51.30 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 09:57.16 | kjs | anyone have any suggestions to make 4 sets ring at once? |
| 09:58.45 | kaldemar | kjs: Dial(Technology/Resource[&Technology2/Resource2[&...]) |
| 09:59.08 | kaldemar | see the app documentation for Dial() |
| 10:02.17 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 10:02.55 | tuxx- | core show application dial |
| 10:02.56 | tuxx- | \o/ |
| 10:10.19 | *** join/#asterisk e-jones (~jkastner@nat/redhat/x-tnhnkatotbngcrsy) |
| 10:13.39 | kjs | hum this syntax is incorrect? exten => 200,1,Dial(SIP/100[&SIP/101],10,t) |
| 10:14.04 | tuxx- | nope |
| 10:14.12 | tuxx- | Dial(SIP/100&SIP/101,10,t) |
| 10:14.15 | tuxx- | thats the correct syntax |
| 10:14.15 | tuxx- | ;P |
| 10:14.55 | kjs | ok nice |
| 10:15.01 | kjs | thats working :D |
| 10:15.14 | tuxx- | :) |
| 10:17.50 | kjs | If I want to record voice prompts with the handset, is this going to be a nightmare? |
| 10:19.07 | tuxx- | sound quality wise yes |
| 10:19.09 | tuxx- | :P |
| 10:19.44 | Chainsaw | kjs: If you *have* to, at least do it in a quiet room. |
| 10:20.04 | Chainsaw | kjs: Nothing's worse then *call center sounds* *awkward silence* Welcome to company X, please choose from the following options |
| 10:20.09 | tuxx- | :D |
| 10:20.11 | tuxx- | idd :) |
| 10:20.36 | Chainsaw | kjs: But seriously, use a decent mic in a proper quiet room and convert the .wav to .gsm |
| 10:20.48 | tuxx- | we usually first record them ourselfs, low quality, and when the soundfiles are perfected we send em to some audiorecording company |
| 10:20.51 | tuxx- | ;p |
| 10:25.00 | *** join/#asterisk icyValk77 (~icyValk77@gateway.ash.thebunker.net) |
| 10:25.27 | coppice | if you find a quiet room and someone with a nice voice its far easier to add an extra prompt here and there |
| 10:25.59 | *** join/#asterisk Dibri (~gavit@pop1.isgroup.sr) |
| 10:26.37 | Chainsaw | coppice has a point, you can just outsource it. Lot of pleasantly voiced people doing voice-over work & related recordings. |
| 10:28.58 | *** join/#asterisk soman (~somnath@stargate.starnet.fi) |
| 10:31.09 | Dovid | I stick to Alison |
| 10:31.41 | *** part/#asterisk Tech_Travis (~Administr@cpe-76-168-191-127.socal.res.rr.com) |
| 10:31.59 | Chainsaw | Dovid: The default voice? |
| 10:39.43 | Dovid | nope. i pay her for my IVR's |
| 10:41.40 | *** join/#asterisk Tim_Toady (~moi@77.49.178.169.dsl.dyn.forthnet.gr) |
| 10:47.32 | *** join/#asterisk nicknick (~administr@host213-123-201-13.in-addr.btopenworld.com) |
| 10:47.42 | kjs | Chainsaw: agreed, I will do the best job I can with the sound. |
| 10:47.56 | kjs | Infact a friend of mine records voice prompts. |
| 10:48.09 | kjs | in a studio, maybe ill call in a favour |
| 10:52.15 | *** join/#asterisk |Cybex| (~John@atwork-21.r-212.178.82.atwork.nl) |
| 10:53.22 | *** join/#asterisk |Cybex| (~John@atwork-21.r-212.178.82.atwork.nl) |
| 10:53.44 | Gugge | anyone know of a function/variable where i can read the full requesturi from an incoming sip call? |
| 10:55.20 | *** join/#asterisk basty (~basty@212.218.65.234) |
| 10:55.32 | basty | Hi |
| 10:55.47 | basty | anyone knows if there is any "snom pickup patch" available for asterisk 1.4.29 ? |
| 10:56.13 | jkroon | snom pickup patch? |
| 10:57.42 | basty | jkroon: i just updated from 1.2 to 1.4. In the 1.2 there used to be such a patch. The patch worked with the leds of the blf. Means if the blf was "blinking" you were able to press on that blf to pickup that conversation. |
| 10:58.14 | jkroon | works out of the box with 1.4, but blinking means that the extension is just ringing. |
| 10:58.39 | jkroon | you need to correctly configure your hints and stuff though, which I've done before but will need to go read up on again. |
| 10:59.08 | basty | jkroon: my blf is setup fine...the blf is working..but as soon as I press on a blf to pickup that conservation - nothing happens. |
| 11:00.10 | jkroon | you need to have an extension similar to this: exten => _**.,1,Pickup(${EXTEN:2}) |
| 11:04.54 | basty | jkroon: yeah I know - but I have also to change the function keys on my snom like "21|*8" to get that one working. The strange thing is, that with a "patch" it worked without changing the function keys. |
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| 11:36.33 | jkroon | basty, no, i've never done that. |
| 11:38.36 | *** join/#asterisk benngard (~benngard@213.88.138.230) |
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| 11:42.36 | basty | jkroon: okay thanks anyway |
| 12:04.19 | *** join/#asterisk m0t3jl (~m0t3jl@ip-40.galance.net) |
| 12:06.48 | m0t3jl | Hi, I would like to connect my Asterisk PBX to a 4 PSTN lines. I know I can buy an internal card for the PBX, but for me those are really expensive. Are there any like dedicated boxes which would act as a PSTN2SIP converters to which I could connect from my Asterisk PBX? Thanks a lot |
| 12:08.04 | Chainsaw | m0t3jl: Patton makes boxes like that, yes. |
| 12:08.25 | Chainsaw | m0t3jl: But if you find that the simple PCI cards are expensive, seeing the prices on those will probably be bad for your blood pressure. |
| 12:08.40 | *** join/#asterisk seanjohn (~john@www2.sheltoncomputers.com) |
| 12:08.49 | m0t3jl | Chainsaw, ... :( |
| 12:08.51 | seanjohn | what is wrong with this? exten => d,n,Set(USEROUTCID='"${CALLERID(name)}"<${CALLERID(number)}>') |
| 12:09.04 | seanjohn | exten => d,n,Set(USEROUTCID='"${CALLERID(name)}"<${CALLERID(number)}>') |
| 12:09.11 | Chainsaw | m0t3jl: So be specific. Tell us what you need, what general area you intend to source it from and what the upper limit on your budget is. |
| 12:09.28 | seanjohn | asterisk wants to be an ass and SKIP it in the dial plan |
| 12:09.49 | seanjohn | no gotoif or anything before it |
| 12:09.53 | Chainsaw | seanjohn: core set debug 10, core set verbose 10, make a test call. |
| 12:10.09 | Chainsaw | seanjohn: Pastebin the results, because then it will be forced to admit whether it ran it and whether anything went wrong. |
| 12:11.15 | seanjohn | nope |
| 12:11.17 | seanjohn | no difference |
| 12:11.45 | m0t3jl | Chainsaw, the thing is I need to replace our old analogue PBX. We are thinking about buying some new pure VoIP numbers and converting some of the PSTN numbers to pure VoIP, but we would like to keep those four PSTN lines so when there's some trouble with our Internt connection, we are still able to call and be called. |
| 12:11.53 | seanjohn | it says it goes from d 12 to d,13 where those that are named are NOT d,12 d,13 but the one i pasted is d,13 |
| 12:12.23 | Chainsaw | m0t3jl: Backups are always a good idea. I would probably end up going with a 4118 gateway in a 4 FXO (for your PSTN lines) 4 FXS (for any analog stuff like faxes) configuration. |
| 12:12.43 | Chainsaw | m0t3jl: Especially because you could then fail over straight from FXO to FXS if your SIP server is down or the world attempts to end in other ways. |
| 12:12.50 | seanjohn | the one before what I pasted is d,12 and the one after is d,14 but THOSE are showing as d,12 and d,13. The pasted doesn't even show up in dial plan |
| 12:13.12 | kaldemar | seanjohn: paste your whole extension and the output as requested |
| 12:13.13 | seanjohn | i'm about to go back to asterisk 1.4.26 |
| 12:13.19 | Chainsaw | seanjohn: If the ,n, is confusing you, just use explicit line numbers. |
| 12:13.44 | seanjohn | i swear to you this has to be a bug in 1.4.29 |
| 12:13.48 | seanjohn | hold on |
| 12:15.41 | m0t3jl | Chainsaw, I was thinking about buying four those cheap PSTN2SIP gateways and taking advantage of them ;) |
| 12:16.00 | Chainsaw | m0t3jl: Takes away the neat fallback solution I proposed. |
| 12:16.27 | Chainsaw | m0t3jl: But yes, that depends on the upper limit of your budget which you still haven't told me. |
| 12:18.16 | m0t3jl | Chainsaw, yep, it does take it ;) It's like this, there's actually no budget specified, but my boss does not want to pay that kind of money the internal card solution would cost ;) |
| 12:18.55 | Chainsaw | m0t3jl: You're evading my questions so I am unable to help you further. Sorry. |
| 12:19.20 | Antonjs | m0t3jl, we've had some luck with Soundwin gateways for analog termination stuff, but I'd be hesitant to actually recommend them |
| 12:19.37 | Antonjs | cheap, nasty, but seem to work |
| 12:20.12 | m0t3jl | Antonjs, ;) |
| 12:21.11 | seanjohn | how do you parse things? |
| 12:21.18 | seanjohn | \? |
| 12:22.17 | m0t3jl | Chainsaw, I wouldn't say it's ME who's avoiding... Maybe I am avoiding, but that's simply because I don't have the budget specified, it's like this: I invent something, go to my boss and when he does not like to pay for that, I have to reinvent it ;) |
| 12:23.22 | Chainsaw | m0t3jl: "My previous solution cost about X and that was voted out, so I'm looking for something <X, sourcing it from country Y... and actually we have a preferred supplier Z that we normally buy from" |
| 12:23.39 | Chainsaw | m0t3jl: Instead of "evade... boss at fault... my life sucks... evade" |
| 12:23.44 | *** join/#asterisk fskrotzki_ (~fskrotzki@cpe-74-74-245-250.rochester.res.rr.com) |
| 12:23.49 | Chainsaw | m0t3jl: If you can't do that, I can't help you. </story> |
| 12:23.53 | *** join/#asterisk Poincare (~jefffnode@213.219.184.23) |
| 12:24.13 | *** join/#asterisk corruptor (~corruptor@85.94.32.98) |
| 12:25.37 | m0t3jl | Chainsaw, what actually do you mean by sourcing something from country? |
| 12:25.44 | Chainsaw | gives up and walks off |
| 12:25.46 | dlynes | Does anyone happen to know what the causes of a corrupted t.38 conversation would be? |
| 12:26.02 | m0t3jl | Chainsaw, I am not familiar with that term ;) |
| 12:26.04 | Chainsaw | dlynes: How far do you get? G711 to T.38 switches? |
| 12:26.16 | dlynes | Chainsaw, t.38 all the way |
| 12:26.16 | Chainsaw | dlynes: That's normally the contentious part. |
| 12:26.23 | Antonjs | m0t3jl, where in the world are you |
| 12:26.33 | dlynes | Chainsaw, ulaw only until it negotiates t.38 |
| 12:26.34 | Antonjs | and where do you normally get your stuff from? |
| 12:26.47 | m0t3jl | Antonjs, ha ;) It's the Czech Republic ;) |
| 12:26.58 | Chainsaw | dlynes: Alright, and your codec renegotiation is successful. Okay, pulling out. |
| 12:27.03 | dlynes | Chainsaw, so about 1s of ulaw, and everything else is t.38 |
| 12:27.43 | corruptor | hello. how do you i think. Is it a bug? A talks to B using g711, A does attendended feature transfer to C and talks to C using g729. then A hangs up. Then A and C hear silence? |
| 12:27.48 | dlynes | Chainsaw, yeah...it successfully negotiates t.38...just has issues with the t.38 transmission |
| 12:28.14 | m0t3jl | Antonjs, well, for computer stuff I have a wholesale dealer, but since we are beginning with VoIP stuff, there is none for that ;) |
| 12:28.18 | corruptor | oh, B and C hear silence but they should talk |
| 12:28.33 | Antonjs | so I guess you're probably just ordering off whichever random online store happens to have what you want? |
| 12:28.40 | Antonjs | =P |
| 12:28.48 | Antonjs | know that feeling |
| 12:29.06 | dlynes | m0t3jl, there's plenty of wholesalers that sell voip gear |
| 12:29.26 | dlynes | m0t3jl, Anixter, White Radio, Ingram Micro, Synnex, ... |
| 12:29.31 | m0t3jl | dlynes, and I am not going against that ;) |
| 12:30.31 | dlynes | Seeing as how you're in Oregon, I think you should probably have an Anixter in Portland |
| 12:31.04 | dlynes | Don't know about the rest of the wholesalers though, because I don't know if they're Canadian only, or not |
| 12:32.27 | m0t3jl | Chainsaw, The solution would cost (roughly converted from CZK) $594. My boss thinks it's too much. ;) |
| 12:32.57 | Chainsaw | m0t3jl: Can I have that in CZK so I can convert to GBP? I don't use dollars :) |
| 12:32.57 | dlynes | oh...nvm...you're in cz |
| 12:33.08 | dlynes | I guess it's just your irc server in oregon |
| 12:33.25 | m0t3jl | Chainsaw, oh ;) |
| 12:33.29 | Dovid | You know what happens when you assume.... |
| 12:33.31 | dlynes | m0t3jl, Are you able to order from the UK? |
| 12:33.51 | Chainsaw | m0t3jl: But working in US dollars for now, the 4118 4FXO/4FXS solution I had in mind is $821.95 |
| 12:33.58 | dlynes | Dovid, yeah, yeah :) |
| 12:34.05 | Chainsaw | m0t3jl: So yes, you're probably looking at a few Linksys ATAs or something else cheap & horrid. |
| 12:34.10 | m0t3jl | dlynes, I think I am able to order from the UK, the thing is will they send it to the Czech Rep. ;) |
| 12:34.18 | dlynes | m0t3jl, that's what I meant :0 |
| 12:34.24 | Chainsaw | m0t3jl: Is it a career-endangering thing if this system fails? |
| 12:35.13 | m0t3jl | Chainsaw, it depends on the time it would take to get it back online, half a day would not be such a big deal |
| 12:35.35 | m0t3jl | Chainsaw, but that 4118 is an external box |
| 12:35.38 | Chainsaw | m0t3jl: *nod* Then I suppose consumer-grade gear wouldn't be that much of a problem. |
| 12:35.54 | Chainsaw | m0t3jl: Yes, that means you have options if your SIP server goes down. I have run these things for a while, sometimes they do that. |
| 12:36.36 | Chainsaw | m0t3jl: (In my case I have an ISDN gateway and an analog gateway as separate devices, if the SIP server is down the ISDN calls go straight through to dedicated analog handsets) |
| 12:36.55 | Chainsaw | m0t3jl: This one on my desk will ring in that case: http://www.vroon.org/ringring.jpg |
| 12:36.59 | Chainsaw | m0t3jl: And I'll know there's trouble. |
| 12:37.11 | Antonjs | having just broken out a 100$ fxo gateway as a backup for our failed voip server, I can second the idea ;) |
| 12:37.42 | m0t3jl | Chainsaw, nice thing ;) |
| 12:37.45 | Antonjs | seriously though, saving $100 on equipment now if you plan on using it as your main gateway is probably going to cost you 5 times that much down the track if it breaks |
| 12:41.22 | Chainsaw | A budget limit is a budget limit, I do appreciate that. But yes, I must urge you to consider the failure modes. |
| 12:41.45 | Chainsaw | If possible, have 4 ATAs with one FXO & one FXS each so you can run a call out to somewhere else then this single point of failu.. eh, SIP server. |
| 12:42.45 | *** join/#asterisk TimeRider (~steve@78.32.26.1) |
| 12:46.21 | m0t3jl | Chainsaw, one of my scenarios was to convert all except for fax lines to pure VoIP and have the phone company assign us new numbers to our current lines. My provider is able to redirect all calls going to a non registered VoIP number (meaning the Internet is down or Asterisk is down) to the analogue numbers (to which there would be some normal phones connected). So it would be possible to send/receive calls even when Asterisk or the Internet is down. |
| 12:47.04 | Chainsaw | m0t3jl: Ah, okay. You have a clueful telco at the other end. I'm in Britain, so I have to deal with BT. That's a luxury I don't have. |
| 12:47.21 | Chainsaw | m0t3jl: But that's good. As long as you've covered that. |
| 12:47.47 | m0t3jl | Chainsaw, which thing makes them clueful? ;) |
| 12:48.04 | Chainsaw | m0t3jl: Able to re-route calls, aware of VoIP... |
| 12:48.25 | m0t3jl | Chainsaw, one thing is the telco company, the other is the SIP provider ;) |
| 12:48.30 | *** join/#asterisk TimeRider (~steve@78.32.26.1) |
| 12:48.53 | m0t3jl | Chainsaw, the SIP provider is quiet reasonable, the telco ... well let's just say not that much ;) |
| 12:49.21 | Chainsaw | m0t3jl: Ah right, you have the SIP provider doing the redirect. Yes, that makes more sense now. |
| 12:50.09 | m0t3jl | Chainsaw, pitty the SIP provider can only redirect calls going to our SIP numbers to PSTN, not the other way around. |
| 12:50.14 | *** join/#asterisk cuco (~Diego@local.xorcom.com) |
| 12:50.15 | Chainsaw | *nod* |
| 12:50.57 | m0t3jl | Chainsaw, it would be great if their systems could reroute calls to the PSTN lines to our Asterisk when we are connected ;) |
| 12:51.28 | Chainsaw | m0t3jl: I wouldn't actually do that. You'll have the backup circuit conundrum. |
| 12:51.51 | Chainsaw | m0t3jl: As in, you never use it unless something has failed. It could have quietly died months earlier and you'd never know... because you don't use it. |
| 12:52.01 | Chainsaw | m0t3jl: Place an automated call to it from time to time. |
| 12:52.15 | m0t3jl | Chainsaw, the thing is I will most probably have to hide the caller's id when someone would actually call out from the PSTN lines. |
| 12:52.56 | *** join/#asterisk Alfio (~amunoz@75.112.88.200.m.sta.codetel.net.do) |
| 12:52.57 | m0t3jl | Chainsaw, it is for the called party not to be able to call back ;) |
| 12:53.12 | Alfio | hi, good morning everyone |
| 12:53.24 | m0t3jl | Alfio, hi, good 2pm ;) |
| 12:53.50 | Alfio | where i can find info in my asterisk server about dahdi errors |
| 12:54.13 | Alfio | m0t3jl hi here 8:52 am |
| 12:54.43 | *** join/#asterisk Akiraa (~Akira@92.81.163.127) |
| 12:54.47 | m0t3jl | Alfio, where is that? East coast of the USA? |
| 12:54.58 | Alfio | m0t3jl dominican republic |
| 12:55.07 | m0t3jl | Alfio, almost ;) |
| 12:55.45 | Alfio | i want to know where i can find info about red or yellow state of dahdi pci card |
| 12:55.54 | Alfio | i mean the logs of that |
| 12:57.01 | *** join/#asterisk benngard (~benngard@213.88.138.230) |
| 12:58.44 | *** join/#asterisk slight (~23hdci278@204.112.221.87.dynamic.jazztel.es) |
| 12:59.25 | slight | can anyone tell me how to strip the + (indicating an international call) from an incoming number so i can format it properly for my trunk? |
| 12:59.38 | slight | i've googled for half an hour without any joy :/ |
| 12:59.52 | dlynes | slight, ${EXTEN:1} |
| 13:00.07 | dlynes | slight, Have you tried that, yet? |
| 13:00.44 | slight | no i mean for the matching part |
| 13:00.56 | slight | "_+44[1-9]. =>" for example |
| 13:01.02 | slight | which doesn't work |
| 13:01.23 | dlynes | slight, Have you tried "_\+44[1-9]. =>"? |
| 13:01.27 | slight | yes |
| 13:01.31 | dlynes | hrm |
| 13:02.00 | dlynes | How do you dial a '+', anyways? |
| 13:02.26 | slight | contacts on my phone use + for intl prefix |
| 13:02.55 | dlynes | ah |
| 13:05.49 | *** join/#asterisk albertoandrade (~albertoan@189.34.121.136) |
| 13:06.12 | dlynes | slight, Try exten => _+X!,1, ... ${EXTEN:1} |
| 13:06.41 | dlynes | slight, does that work for you? |
| 13:07.13 | dlynes | slight, I just used the google search 'strip the plus sign on outbound calls using asterisk', and this solution was the third link down |
| 13:08.14 | slight | yes seen that, not been able to adapt it to use the 44 as well yet |
| 13:08.17 | slight | sec |
| 13:08.51 | dlynes | slight, exten => _+44X!,1,...? |
| 13:10.37 | slight | do you know what the exclamation mark is for? |
| 13:10.46 | dlynes | slight, no idea...never seen it before |
| 13:11.45 | slight | no it's still passing through the + |
| 13:11.51 | *** join/#asterisk sgimeno (~chatzilla@226.Red-80-33-64.staticIP.rima-tde.net) |
| 13:11.55 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 13:13.07 | dlynes | [TK]D-Fender, You happen to know what the '!' character is for in a dialplan pattern, or how to strip the '+' from an incoming exten? |
| 13:13.38 | [TK]D-Fender | dlynes: Its documented all over.... go brush up on this. |
| 13:13.47 | [TK]D-Fender | dlynes: So is stripping chars off a string... |
| 13:13.55 | slight | it's hard to google for symbols as it ignores them :) |
| 13:14.24 | dlynes | [TK]D-Fender, exten => _+44X.,1,Noop(${EXTEN:1}) passes through the '+' |
| 13:14.31 | dlynes | [TK]D-Fender, according to slight, anyways |
| 13:14.32 | *** join/#asterisk c0rnoTa (~c0rnoTa@nas-8.emserv.ru) |
| 13:14.32 | [TK]D-Fender | slight: "asterisk dialplan patterns" <- it ain't Raw-Cat Sigh Hence |
| 13:14.40 | slight | i know how to strip chars, the problem is matching the +, however i think i may have another problem in my dialplan that's causing this so i'll look into that first |
| 13:14.51 | c0rnoTa | good day, all |
| 13:15.07 | c0rnoTa | so sunny today |
| 13:15.33 | slight | ok thanks, no need to be rude though, i just didn't have the right search terms |
| 13:18.36 | [TK]D-Fender | slight: It's also in the BOOK. Checked it out by any chance? |
| 13:20.38 | slight | you've got a real chip on your shoulder haven't you? i spent a lot of time googling and couldn't find an answer so i asked for help, i'm very sorry if that offends you. |
| 13:21.50 | [TK]D-Fender | slight: Just checking to see if you knew that we do have a decent resource for this. You might not have. This isn't a chip on my shoulder. |
| 13:23.39 | [TK]D-Fender | slight: Now if things still aren't working right feel free to pastebin your dialplan and CLI output with appropriate channel debug and we'll see what's the matter. |
| 13:23.52 | [TK]D-Fender | ~pb |
| 13:23.53 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 13:25.02 | slight | it's ok thanks, i think i've found the problem. ubuntu's latest version of asterisk initially refused to use my extensions.conf, so i created an extensions.ael and it worked, but it seems another update has it now ignoring my extensions.ael and wanting extensions.conf. so editing my extensions.ael wasn't doing anything |
| 13:25.48 | *** join/#asterisk ManxPower-work (~EWieling@216.186.151.147) |
| 13:26.16 | [TK]D-Fender | slight: Only reason for that is if permissions are getting in the way. |
| 13:26.43 | slight | both are readable by the asterisk user |
| 13:26.58 | [TK]D-Fender | slight: IIRC * needs to be able to write to it as well |
| 13:27.00 | slight | and group |
| 13:27.10 | slight | both writable by asterisk user too |
| 13:28.55 | slight | ok, edited extensions.conf and it's working now. moved extensions.ael out of the way so it'll be clear if it happens again |
| 13:28.59 | slight | thanks for the help |
| 13:30.18 | *** join/#asterisk e4 (~e4@rrcs-76-79-59-194.west.biz.rr.com) |
| 13:34.51 | [TK]D-Fender | slight: You're welcome |
| 13:38.38 | *** join/#asterisk [sr] (~Unknowned@87.196.12.76) |
| 13:39.46 | *** join/#asterisk rossand (~aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 13:43.14 | plundra | Can you get the pretty coloring (like when started using -c), when attaching with asterisk -r? |
| 13:44.38 | [TK]D-Fender | plundra: I always have |
| 13:45.10 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 13:45.34 | plundra | [TK]D-Fender: Using no other arguments? The actual asterisk is started using -vvvf (by upstart) |
| 13:46.36 | [TK]D-Fender | plundra: I always use -gvvvvvvvvvvvvvc for my manual starts |
| 13:47.51 | plundra | Ok, well this isn't manual or anything. Do you have to start the actual server with -c, for later on attached -r's to have color too? |
| 13:48.56 | [TK]D-Fender | plundra: Dunno... go try :) |
| 13:49.22 | plundra | That would be later tonight then, people are using it :-) |
| 13:50.36 | plundra | Ah, well, yes, that is actually the case. |
| 13:51.48 | plundra | Hmm, I wonder if there are any downsides to start it with -c, it doesn't fork off or anything so should still be fine with upstart. But all that output is just thrown away. |
| 13:56.28 | *** join/#asterisk Dovid (~annon@tony09-118-62.inter.net.il) |
| 13:57.11 | *** join/#asterisk jaytee (~jforde@unaffiliated/jaytee) |
| 13:59.38 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 14:02.02 | *** join/#asterisk muiro (~muiro@unaffiliated/muiro) |
| 14:03.40 | *** join/#asterisk mikkel (~mikkel@84-238-113-66.u.parknet.dk) |
| 14:04.22 | Katty | good morning |
| 14:05.21 | jaytee | morning Katty *hugs* |
| 14:06.29 | jblack | Morn |
| 14:07.58 | *** join/#asterisk high-freq (~hfreq@99.188.122.87) |
| 14:08.42 | beek | morning Katty, jaytee, [TK]D-Fender, et al |
| 14:10.32 | jaytee | morning beek |
| 14:11.02 | jaytee | who is et al? some arab guy? |
| 14:11.25 | ManxPower-work | et tu, Al? |
| 14:11.43 | *** join/#asterisk voipmonk (~shido6@CPE002191f85581-CM001692568382.cpe.net.cable.rogers.com) |
| 14:13.01 | [TK]D-Fender | Beware the Ides of MArch... |
| 14:13.36 | jaytee | well, we got almost a month before we need to worry |
| 14:14.02 | *** join/#asterisk voipmonk (~shido6@CPE002191f85581-CM001692568382.cpe.net.cable.rogers.com) |
| 14:15.24 | jaytee | I'd go the Caligula route and bankrupt the government spending all the money on public baths and then make my horse a member of the Senate but since the government's almost bankrupt anyways and Congress is just a huge herd of horse's asses it would be a pointless gesture. |
| 14:16.18 | beek | jaytee: Ever see Bob Guccioni's (sp) movie Caligula? |
| 14:16.51 | *** join/#asterisk grey-monkey (~ericshell@75-148-103-190-Utah.hfc.comcastbusiness.net) |
| 14:16.58 | jaytee | beek, yes. it was "disturbing" to put it mildly but then I'd already studied the Roman Empire in college |
| 14:17.47 | beek | That movie came out when I was in college... I still have the Penthouse mag that was promoting the movie boxed up somewhere. |
| 14:17.55 | kjs | Any of you guys using gradwell for SIP / IAX ? |
| 14:17.59 | jblack | I never saw caligula. I thought it was a movie about some sort of perverse sex act |
| 14:18.05 | ManxPower-work | jaytee: I'm sure they would find today's nuclear family equally as disturbing. |
| 14:19.20 | jblack | Wow. That's a hell of a plot listed at http://en.wikipedia.org/wiki/Caligula_%28film%29 |
| 14:19.53 | jblack | This version contained significant sexual and violent content, including orgies, masturbation, fellatio, cunnilingus, anal fisting, homosexuality, transvestism, incest, rape, urination, as well as scenes of decapitation of prisoners by a giant "killing machine," infanticide, implied fratricide, and penile as well as testicular castration. |
| 14:20.02 | jblack | In a single 156 minute move? WTF, really? |
| 14:20.07 | beek | jblack: yes |
| 14:20.24 | jblack | On top of the plot? |
| 14:20.27 | jaytee | the problem with today's nuclear family is that it doesn't generate any useful energy, it just sucks it all out of the grid and into X-Boxes and Playstations while the kids suck down ground cow hooves baked into chips and drink gallons of colored corn syrup. |
| 14:20.30 | drmessano | Are those the release notes for 1.8? |
| 14:20.31 | jblack | Where did they find the time for all of that? |
| 14:20.44 | beek | drmessano: :D |
| 14:21.03 | jaytee | drmessano, LOL |
| 14:21.26 | jblack | drmessano++ |
| 14:21.43 | jaytee | drmessano#.NET |
| 14:21.51 | drmessano | lol |
| 14:22.34 | ManxPower-work | "OMG We have phone problems today!" "OMG get your lines fixed with your carrier and the probems should go away -- just like I've told you almost every day for the past 2 weeks" |
| 14:23.35 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 14:23.55 | plundra | Can I backtrack where exactly this is, "WARNING[14439]: ast_expr2.fl:433 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected '=', expecting $end; Input:\n=1\n^" (Pointing at the = :-P) |
| 14:24.38 | plundra | Aaah! Nevermind, I know :-) The variable I'm comparing with, unquoted, is empty. |
| 14:27.53 | Katty | hugs jaytee and beek |
| 14:28.14 | Katty | today is starting off badly |
| 14:28.45 | Chainsaw | Do you want to talk about it Katty? |
| 14:28.54 | Katty | you mean bitch about it? |
| 14:28.59 | Katty | no, wouldn't do any good. |
| 14:29.08 | Chainsaw | Hm, okay. Hope it gets better soon. |
| 14:29.13 | Katty | i'm sure it will (= |
| 14:29.25 | Chainsaw | Tea always helps for me. Perhaps coffee over there :) |
| 14:32.31 | Naikrovek | omfg why does my employer keep hiring people that don't know how to use a freaking computer |
| 14:32.35 | Naikrovek | they're supposed to be developers |
| 14:32.39 | Naikrovek | "how do i download" |
| 14:32.53 | Naikrovek | "it says right click, what does that mean" |
| 14:32.56 | Naikrovek | W T F |
| 14:33.22 | drmessano | They're Mac developers, duh |
| 14:33.36 | Naikrovek | they're electrical engineers with supposed degrees |
| 14:33.41 | Naikrovek | i wanna see these papers |
| 14:33.56 | Naikrovek | in reality they're friends of my boss and nothing more |
| 14:36.23 | *** join/#asterisk high-freq (~hfreq@99.188.122.87) |
| 14:37.16 | Katty | shoot them. |
| 14:37.41 | Chainsaw | Stage a high-voltage fault and send them in. |
| 14:37.49 | Katty | with a double barrel shotgun |
| 14:39.05 | Katty | anyone know of a website that shows the new releases that come out every tuesday? i figured blockbuster would have it, but they don't have the new ones posted yet |
| 14:39.22 | jaytee | Blockbuster is going under |
| 14:39.29 | jaytee | try Netflix or Amazon |
| 14:39.38 | Katty | err |
| 14:39.40 | Naikrovek | new releases of what? blu-ray? dvd? games? |
| 14:39.42 | Katty | I mean posts them |
| 14:39.44 | Katty | i just want a list |
| 14:40.06 | Katty | dvd releases |
| 14:40.09 | *** join/#asterisk SuPrSluG (~SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 14:40.32 | Naikrovek | katty: http://www.themovieinsider.com/dvd-releases/-/2010/ |
| 14:40.59 | Katty | looks |
| 14:41.54 | Katty | thanks. |
| 14:41.54 | jaytee | Katty, http://tinyurl.com/yjkjkd9 |
| 14:41.54 | Katty | bookmarks |
| 14:41.55 | Naikrovek | tinyurl is the herpes of the web |
| 14:42.29 | Naikrovek | like glitter is the herpes of crafts |
| 14:42.37 | Katty | hmm. not many good movies out this week. |
| 14:42.42 | Naikrovek | and how herpes is the herpes of sex |
| 14:43.12 | m0t3jl | Chainsaw, If I bought the 4118 gateway, would I be able to have 4 SIP accounts on it (for every one of the PSTN lines) so Asterisk could to the job of choosing lines to use, etc? |
| 14:43.25 | Chainsaw | m0t3jl: Yes, you can. |
| 14:43.30 | Chainsaw | m0t3jl: It registers once per extension. |
| 14:43.43 | Chainsaw | m0t3jl: And I have one extension per analog line. |
| 14:44.30 | Chainsaw | m0t3jl: I'm happy to get you some config examples as well, as they can be a bit daunting at first. The amount of things you can configure is mindblowing. |
| 14:44.54 | *** join/#asterisk Akiraa (~Akiraaaa@79.112.29.133) |
| 14:44.58 | m0t3jl | Chainsaw, ;) |
| 14:47.02 | m0t3jl | Chainsaw, can it handle a use case in which I would have 4 incomming analog lines (4 SIP accounts for Asterisk to connect to), four analogue internal phones and the 4118 would magically know that when Asterisk is not connected to the SIP accounts to redirect the calls to the analogue phones? |
| 14:47.17 | Chainsaw | m0t3jl: Yes, you can set it up that way. |
| 14:47.28 | *** join/#asterisk korihor (~korihor@190.205.251.97) |
| 14:47.32 | Chainsaw | m0t3jl: "Try this first, if that doesn't work, ring all these 4 lines until someone answers" |
| 14:47.51 | Chainsaw | m0t3jl: Or even "try the SIP server, ring some analog phones, cycle until something works" |
| 14:47.57 | m0t3jl | :) |
| 14:50.18 | m0t3jl | Chainsaw, I'll try to find Czech distributor ;) |
| 14:50.43 | Chainsaw | m0t3jl: Read the specs carefully, some have an external power supply while others take a mains feed. |
| 14:51.00 | Chainsaw | m0t3jl: The price difference is minimal and the convenience factor of not having a PSU brick is rather high. |
| 14:51.32 | staffmember | can someone help me out with something... I can receive calls to an extension.. but when i pick up that extension and start to dial, i get a busy signal |
| 14:51.38 | staffmember | using cisco 7975 phone |
| 14:51.57 | staffmember | when i pick up, the minute i press any number, it goes busy |
| 14:52.03 | staffmember | but i can receive calls at that extension |
| 14:52.22 | ManxPower-work | staffmember: the dialplan on the phone is messed up. |
| 14:52.25 | m0t3jl | Chainsaw, lol. If I hit search only in Czech on Google, it finds nothing :D |
| 14:52.42 | staffmember | is it a phone config issue, or is it a asterisk |
| 14:52.47 | staffmember | issue |
| 14:53.11 | ManxPower-work | staffmember: Well without any other information from you -- chances are it's a phone issue. |
| 14:53.42 | *** join/#asterisk garymc (~chatzilla@host81-139-136-16.in-addr.btopenworld.com) |
| 14:54.54 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:54.54 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:55.39 | rue_house | I'm starting to think people generally hate polycom sets |
| 14:55.47 | Naikrovek | um what? |
| 14:55.47 | Naikrovek | no |
| 14:55.52 | Naikrovek | i love them |
| 14:55.57 | Naikrovek | a lot of people in here love them |
| 14:56.00 | rue_house | your building the system tho |
| 14:56.04 | rue_house | k |
| 14:56.15 | Naikrovek | my users like them fine |
| 14:56.33 | rue_house | are you sure? |
| 14:56.33 | Naikrovek | there are complaints "I want cisco" to which I reply $500/phone please |
| 14:56.56 | Katty | polycom++ |
| 14:57.01 | rue_house | I mean in contrast to a keyed phonesystem |
| 14:57.08 | Katty | define Keyed |
| 14:57.12 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 14:57.16 | ManxPower-work | I think the only person that really hates Polycom is jmcdowell |
| 14:57.20 | rue_house | nortel 616 |
| 14:57.25 | Katty | that does not parse. |
| 14:57.27 | Katty | define keyed. |
| 14:57.27 | rue_house | panasonic tda30 |
| 14:57.52 | ManxPower-work | You are comparing Polycom phones to a Key System? |
| 14:57.57 | [TK]D-Fender | yup |
| 14:57.58 | Naikrovek | wtf is a keyed system |
| 14:58.04 | Katty | Naikrovek: dunno, he wont answer me |
| 14:58.05 | rue_house | users dont understand the diff much ya know |
| 14:58.09 | [TK]D-Fender | Naikrovek: Dumb PBX's like Norstar, etc |
| 14:58.10 | Katty | then tell them. |
| 14:58.14 | Naikrovek | ah |
| 14:58.17 | m0t3jl | Chainsaw, apparently I am unable to get the Patton brand in here. I can get Linksys, OneStream and Grandstream, maybe even Well. Are any of these as good as the Patton one? |
| 14:58.21 | Naikrovek | your users will understand most of what you teach them |
| 14:58.23 | Naikrovek | they're not supid |
| 14:58.25 | Naikrovek | stupid |
| 14:58.29 | Naikrovek | not THAT stupid anyway |
| 14:58.30 | rue_house | Katty, your not familiar with nortel 616? |
| 14:58.42 | Chainsaw | m0t3jl: Even if you ask them? |
| 14:58.45 | Katty | you say that as if you're in shock |
| 14:58.53 | ManxPower-work | Naikrovek: One major difference between a PBX and a Key System is that a key system has a 1-to-1 relationship between line buttons on the phones and the actual line. |
| 14:59.00 | Katty | eww |
| 14:59.07 | Naikrovek | yeah |
| 14:59.10 | Naikrovek | that's nastysause |
| 14:59.11 | Chainsaw | m0t3jl: There's about one supplier in the UK that admits to selling them, but others have gotten them in for me. |
| 14:59.14 | Naikrovek | nastysauce |
| 14:59.18 | [TK]D-Fender | ManxPower-work: Not exactly... just a common implementation |
| 14:59.41 | m0t3jl | Chainsaw, possibly. I would just like to know what would be a price for a similar one they have in actual stock ;) |
| 14:59.43 | ManxPower-work | Of course users are STUPID. They might be stupid because they can't learn or they can be stupid because they don't WANT to learn. The end result is the same. |
| 14:59.52 | Naikrovek | when you enter voip-land you lose the "line" concept and switch to concurrent calls |
| 14:59.55 | [TK]D-Fender | ManxPower-work: My norstar had plenty of phones with less buttons than lines. That's why I have multiple Intercom keys |
| 14:59.59 | rue_house | one of my users was at a car dealership in anohter town and noticed they had polycom phones, upon asking if they liked it, they said "we hate it, but were getting used to it" |
| 15:00.01 | Naikrovek | ManxPower-work: well i see your point, but stupid and ignorant are different |
| 15:00.21 | ManxPower-work | [TK]D-Fender: Nortel can be a plain key system or a hybrid system. |
| 15:00.21 | Naikrovek | stupid = can't learn, willfully ignorant is another matter |
| 15:00.24 | *** part/#asterisk doug (doug@breakout.telerama.com) |
| 15:00.37 | [TK]D-Fender | ManxPower-work: Hybrid? Do NOT even mention the BCM... |
| 15:00.41 | Naikrovek | :) |
| 15:00.43 | *** join/#asterisk AtLeT (~atletek@spletoknovsvet.spin.si) |
| 15:00.49 | AtLeT | hi |
| 15:00.54 | ManxPower-work | [TK]D-Fender: our MICS was hybrid. |
| 15:00.54 | Katty | hi |
| 15:00.55 | [TK]D-Fender | ManxPower-work: That's like duct-taping a GrandSuck phone onto a CICS |
| 15:00.57 | rue_house | my understanding of the BCM is that its a completel nightmare |
| 15:01.18 | AtLeT | I have some trubles with FollowMe on asterisk 1.6.1.x |
| 15:01.22 | [TK]D-Fender | rue_house: the term "flaming piece of shit" comes to mind. |
| 15:01.26 | Katty | sigh, only 9 am |
| 15:01.36 | Katty | it's going to be a long day |
| 15:01.39 | [TK]D-Fender | rue_house: Think a web-ified Norstar.... new tech, OLD concepts |
| 15:01.40 | rue_house | funny, I hear that from everyone who works with them |
| 15:01.56 | rue_house | its keyed voip? |
| 15:01.57 | AtLeT | can anbody help me? |
| 15:02.04 | Katty | AtLeT: how should we know |
| 15:02.08 | AtLeT | :) |
| 15:02.09 | Katty | AtLeT: you haven't asked a question yet |
| 15:02.11 | [TK]D-Fender | rue_house: And their bulky crap IP phones running proprietary protocols |
| 15:02.14 | rue_house | AtLeT, not till we know more about your problem |
| 15:02.16 | voipmonk | until you see the image of a phone and have to configure your system through that phone image |
| 15:02.17 | AtLeT | so, I setup followme |
| 15:02.32 | voipmonk | not image like firmware , image like a picture of a phone with buttons |
| 15:02.34 | AtLeT | when the call is execuet |
| 15:02.35 | rue_house | since when has nortel not been prop? |
| 15:02.39 | AtLeT | I canot pickup |
| 15:02.42 | m0t3jl | Chainsaw, what about the OneStream FX, 6x FXS, 6x FXO? |
| 15:02.53 | Chainsaw | m0t3jl: I haven't used them, so I can't vouch for them. |
| 15:02.56 | [TK]D-Fender | rue_house: is this the same client with that Hell-spawn TDM card? |
| 15:02.56 | AtLeT | the calling person only hears moh |
| 15:03.05 | AtLeT | I answer the phone |
| 15:03.06 | rue_house | AtLeT, it rings and you cant pick it up? |
| 15:03.12 | AtLeT | but I cannot hear nothing |
| 15:03.21 | AtLeT | yes, I answer the phone |
| 15:03.22 | Katty | cannot hear nothin |
| 15:03.23 | Katty | i like that |
| 15:03.27 | rue_house | are you using it thu a firewall? |
| 15:03.30 | AtLeT | but cannot pickup the person |
| 15:03.41 | AtLeT | no, it's not on nat |
| 15:03.45 | AtLeT | no firewall |
| 15:03.48 | rue_house | ok |
| 15:03.54 | Katty | AtLeT: what type of system do you have. |
| 15:03.59 | AtLeT | other thing are working fine |
| 15:04.08 | AtLeT | asterisk 1.6.1.x on ubuntu |
| 15:04.18 | rue_house | [TK]D-Fender, yes |
| 15:04.23 | AtLeT | is there any tutorial about followme |
| 15:04.35 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:04.35 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 15:04.36 | Katty | tons |
| 15:04.39 | ManxPower-work | Which GUI are you using? |
| 15:04.39 | Katty | just google it |
| 15:04.42 | rue_house | [TK]D-Fender, its my office witht eh hell-spawn digium card |
| 15:04.46 | Katty | hugs leifmadsen |
| 15:04.47 | AtLeT | I don't use any gui |
| 15:04.48 | m0t3jl | Chainsaw, have you only used the Patton ones? |
| 15:04.58 | AtLeT | I googled a lot |
| 15:04.59 | [TK]D-Fender | rue_house: Yeah, no wonder they hate it... you also tried faking SLA there, gave them a crap card that's been nothing but trouble for a year and a half |
| 15:05.05 | AtLeT | and don't find any usefull thing |
| 15:05.08 | leifmadsen | re-hugs Katty |
| 15:05.17 | rue_house | [TK]D-Fender, I steered them away from sla |
| 15:05.17 | Katty | AtLeT: then you're not doin it right |
| 15:05.22 | AtLeT | ;) |
| 15:05.39 | [TK]D-Fender | rue_house: A sufficiently dissatisfied customer will cost you more than the product you tried to sell them. |
| 15:05.41 | rue_house | we use SBA "single business appearance" |
| 15:05.52 | AtLeT | http://www.voip-info.org/wiki/view/Asterisk+cmd+FollowMe |
| 15:06.02 | AtLeT | this is how i did it |
| 15:06.08 | ManxPower-work | Heh! We had our first major Polycom issue --- Looks like Polycom has decided to go with the Cisco support model. |
| 15:06.14 | AtLeT | example 1 |
| 15:06.27 | *** part/#asterisk benngard (~benngard@213.88.138.230) |
| 15:06.39 | rue_house | [TK]D-Fender, the boss was thrilled I could have his phone N/A forward to his cell |
| 15:06.52 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 15:07.03 | ManxPower-work | AtLeT: Did you do a "core show application FollowMe" to confirm the options for YOUR follow me is the same on that old voip-info wiki site with all the outdated information? |
| 15:07.22 | AtLeT | no |
| 15:07.30 | ManxPower-work | Perhaps you should. |
| 15:07.43 | *** join/#asterisk superbeef (~lanej@74.84.194.4) |
| 15:07.50 | [TK]D-Fender | AtLeT: Don't point us to the sample you claimed to have followed. Show us YOUR code, and YOUR CLI output. |
| 15:07.52 | rue_house | [TK]D-Fender, if the machine(pc) is at fault for the echo, shoudl I see missed interrupts? |
| 15:07.53 | [TK]D-Fender | ~pb |
| 15:07.53 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 15:07.55 | [TK]D-Fender | ^^^^^^^^^^ |
| 15:08.08 | [TK]D-Fender | rue_house: just another sign |
| 15:08.18 | [TK]D-Fender | rue_house: I don't think we need to rehash this. |
| 15:08.37 | ManxPower-work | rue_house: um, echo is caused by the far end analog loop. Maybe you mean that echo cancelation? |
| 15:09.03 | m0t3jl | Chainsaw, what about Signamax? |
| 15:09.04 | rue_house | I'm wondering if a bad drive or soemthing might be causing processor delays |
| 15:09.24 | ManxPower-work | rue_house: "processor delays"? |
| 15:09.57 | rue_house | ManxPower-work, well say the processor gets locked up for a ms caught on a drive access interrupt |
| 15:10.11 | ManxPower-work | rue_house: very common. |
| 15:10.15 | Naikrovek | rue_house: you're overthinking it. echo is not caused that way |
| 15:10.17 | [TK]D-Fender | rue_house: If your card is shared with a HD, that's a.... |
| 15:10.19 | [TK]D-Fender | ~soso |
| 15:10.20 | infobot | [~soso] Shoot-On-Sight Offense |
| 15:10.29 | rue_house | heh |
| 15:10.37 | rue_house | I have to get going, running late |
| 15:10.46 | Naikrovek | late? |
| 15:10.47 | Naikrovek | where are you |
| 15:10.51 | ManxPower-work | when you have time to actually work on the issue, let us know. |
| 15:10.51 | Katty | for a very important date |
| 15:10.59 | Katty | no time to say hello! goodbye! i'm late i'm late i'm late! |
| 15:11.04 | AtLeT | http://pastebin.com/d68cde4ea |
| 15:11.08 | Naikrovek | ah yeah i supposed it could be for a dr. appt. or something |
| 15:11.11 | AtLeT | here is my confing |
| 15:11.12 | Naikrovek | or another country |
| 15:11.14 | Naikrovek | maybe mars |
| 15:11.42 | ManxPower-work | AtLeT: That is a totally invalid configuration |
| 15:11.53 | AtLeT | hm |
| 15:12.04 | AtLeT | what I did wrong? |
| 15:12.29 | ManxPower-work | For one thing I see no Macro call to run that macro you showed. |
| 15:12.51 | AtLeT | I didn't paste all the extensions.conf file |
| 15:12.54 | ManxPower-work | At best you did not post your complete dialplan, at worst your configuration is totally wrong. |
| 15:12.55 | AtLeT | only a macro line |
| 15:13.07 | ManxPower-work | AtLeT: without complete information nobody will be able to help you. |
| 15:13.15 | [TK]D-Fender | AtLeT: You aren't showing a complete enough picture for us to judge. You aren't showing the FAILED CALL. |
| 15:14.15 | Chainsaw | m0t3jl: I haven't used those either. |
| 15:14.24 | *** join/#asterisk came0 (~came0@rrcs-71-42-53-182.se.biz.rr.com) |
| 15:14.44 | rue_house | core set verbose 99 core set debug 99 pastebin |
| 15:15.19 | AtLeT | http://pastebin.com/d2e1f12c |
| 15:15.19 | ManxPower-work | Gah! Why can't "web technologies" be nice an simple like telecom? |
| 15:16.26 | ManxPower-work | AtLeT: I still see no CLI output. [TK]D-Fender does not ask for this stuff to be a jerk, he asks for this stuff because it's important to have. |
| 15:17.35 | AtLeT | http://pastebin.com/dcc5d542 |
| 15:17.47 | m0t3jl | Chainsaw, pitty |
| 15:17.49 | AtLeT | here is also cli now |
| 15:18.00 | ManxPower-work | Hopefully [TK]D-Fender is still available to help you. |
| 15:18.32 | Katty | i need a vacation |
| 15:18.48 | Katty | a week of sitting around, not being responsible for anything |
| 15:19.04 | Naikrovek | just a week? |
| 15:19.17 | Katty | i don't think i could get away with more than an week |
| 15:19.21 | Katty | the world would probalby come to an end |
| 15:19.32 | ManxPower-work | Katty: Some people make a career of that. |
| 15:19.33 | Naikrovek | it takes more than a week for me to wind down |
| 15:19.47 | Katty | WIND IT UP |
| 15:19.49 | [TK]D-Fender | AtLeT: Seems the call is getting instantly dropped. Enable SIP DEBUG and pastebin a new call <- |
| 15:20.12 | ManxPower-work | [TK]D-Fender: Think the telco is rejecting the call based on the callerid? |
| 15:20.37 | *** join/#asterisk RobH (~robh@2620:0:860:2:21e:c2ff:fe03:2465) |
| 15:20.43 | [TK]D-Fender | ManxPower-work: No, the 1st # contacted is a local SIP device. I'd bet on codecs or re-invite failure first |
| 15:20.44 | Katty | hi RobH |
| 15:22.17 | AtLeT | this is with enabled debug 99 |
| 15:22.28 | AtLeT | I pasted everything |
| 15:22.30 | ManxPower-work | AtLeT: debug 99 != sip debug! |
| 15:22.36 | [TK]D-Fender | altSIP DEBUG <------- |
| 15:22.39 | AtLeT | how I can enable sip debug |
| 15:22.43 | [TK]D-Fender | AtLeT: SIP DEBUG <------- |
| 15:22.43 | AtLeT | ok |
| 15:22.44 | Katty | oh boy. |
| 15:22.47 | AtLeT | ;) |
| 15:23.02 | Naikrovek | sip debug |
| 15:23.04 | [TK]D-Fender | AtLeT: "sip set debug on" |
| 15:23.08 | Naikrovek | that's it |
| 15:23.16 | Katty | interesting. |
| 15:23.26 | Katty | so if i put my elbows on the desk, and cover my ears |
| 15:23.34 | Katty | i can hear the vibration coming off the heater fan |
| 15:23.44 | Katty | duddddddddeee |
| 15:24.24 | Katty | awesome noise |
| 15:24.53 | AtLeT | hm I enabled |
| 15:25.12 | AtLeT | is there any way to redirect this to file |
| 15:25.21 | AtLeT | what the debug is putting out |
| 15:27.12 | *** join/#asterisk lordvadr (~something@jose-tc.ctc.biz) |
| 15:27.20 | *** join/#asterisk m_c_le (~marcello@dslb-088-074-131-094.pools.arcor-ip.net) |
| 15:28.23 | AtLeT | http://pastebin.com/d33b820d5 |
| 15:28.28 | AtLeT | I pasted as much I can |
| 15:30.52 | lordvadr | Some years ago when I bought my tdm2400, I hypothetically asked how many cards I could fit in a single box. The sales engineer said at some point, I'd hit an IRQ problem, but he didn't know what that limit was.... |
| 15:31.04 | lordvadr | I'm now being asked to look into the feasibility of an asterisk system to support on the order of 100 fxs ports and I was wondering if anybody has any idea or experience with 5 fully populated tdm2400 cards in a machine. |
| 15:31.30 | Chainsaw | lordvadr: Are the cards PCI? |
| 15:31.30 | Naikrovek | there are i think 15 IRQs in an x86 machine |
| 15:31.46 | Chainsaw | Naikrovek: But there's only 4 IRQ pins for a PCI bus. |
| 15:31.54 | lordvadr | Chainsaw: doesn't matter. PCI, pci-e. Either way, and depending on the board we use, probably a mix of both |
| 15:31.59 | Chainsaw | Naikrovek: So unless the device supports MSI, you're screwed earlier then you think. |
| 15:32.24 | lordvadr | Chainsaw: Does the tdm2400 support MSI? |
| 15:32.38 | Chainsaw | lordvadr: Now you're asking the right questions. |
| 15:32.44 | Chainsaw | lordvadr: I don't have the answer, but Digium should. |
| 15:32.49 | ManxPower-work | lordvadr: define "MSI". |
| 15:32.51 | Chainsaw | lordvadr: Also, does your current machine support MSI? |
| 15:32.56 | Chainsaw | ManxPower-work: Message Signalled Interrupts |
| 15:32.57 | florz | Chainsaw: there isn't "only 4 pins for a PCI bus", that depends on the board |
| 15:33.12 | lordvadr | ManxPower-work: Message Signaled Interrupts |
| 15:33.13 | ManxPower-work | Chainsaw: never heard of it. |
| 15:33.14 | Chainsaw | florz: If the board has multiple buses, sure. Rare though. |
| 15:33.41 | ManxPower-work | lordvadr: Let us know what you find out when you contact Digium. |
| 15:33.44 | lordvadr | I asked digium and the nice lady told me to ask the user base. I thought I might get a better answer here than on the ml. |
| 15:33.53 | florz | Chainsaw: well, not so much with board with many slots ... |
| 15:33.59 | florz | +s |
| 15:34.08 | Chainsaw | florz: A board with many PCI slots has multiple PCI buses. |
| 15:34.14 | Chainsaw | florz: Each of those buses has a set of 4 IRQ pins. |
| 15:34.35 | lordvadr | Chainsaw: At least for 32bit pci, I thought it was a shared bus |
| 15:35.03 | Chainsaw | lordvadr: You're going to have elaborate on that. |
| 15:35.04 | florz | Chainsaw: yeah, exactly - and usually you won't have much more than 4 slots per bus |
| 15:35.40 | Chainsaw | florz: Physical connectors on the board, no. Integrated devices leeching a PCI bus that you have slots on though... different story. |
| 15:35.53 | lordvadr | Chainsaw: Nothing really to elaborate on, I just thought all the IRQ pins and bandwidth were shared on the pci bus, but that's neither here nor there. I'm happy with PCI-e if that will eliminate the problem |
| 15:36.08 | lordvadr | If there's even a problem |
| 15:36.49 | Chainsaw | lordvadr: PCI-e will likely end up a better choice with that much hardware attached. It would help to know what mainboard you have, at any rate. |
| 15:36.56 | [TK]D-Fender | lordvadr: PCI = horrendous mistake |
| 15:36.58 | florz | Chainsaw: yeah, but those integrated devices usually don't cause all that many interrupts, and in particular not on all interrupts at once |
| 15:37.02 | lordvadr | Does anybody know who actually manufactures the cards? |
| 15:37.05 | [TK]D-Fender | lordvadr: 100 FXS is not the problem. |
| 15:37.12 | [TK]D-Fender | lordvadr: go get yourself some SIP gateways |
| 15:37.13 | Chainsaw | florz: It isn't about quantity, it's about timing. |
| 15:37.25 | ManxPower-work | I doubt you are going to be saturation the PCI bus with 1.544Mbps of data and 8,000 interrupts/second. |
| 15:37.36 | Naikrovek | yeah |
| 15:37.47 | lordvadr | TKD-Fender: Can't, need to service 100 analog lines for a nursing home |
| 15:37.54 | [TK]D-Fender | ManxPower-work: Think about the case that would be needed for those monstrosities |
| 15:38.11 | lordvadr | Chainsaw: No board picked yet |
| 15:38.11 | [TK]D-Fender | lodSIP GATEWAYS. I never said anything for TERMINAtiON |
| 15:38.35 | Chainsaw | ManxPower-work: Which are theoretical maximums under ideal circumstances that you will not be replicating. |
| 15:38.46 | ManxPower-work | if you need to service that many analog lines than I recommend a channel bank -- others will recommend a SIP multiport ATA/gateway |
| 15:39.05 | florz | Chainsaw: both, really (an occasional interrupt being services a bit too late usually won't matter much) |
| 15:39.06 | Chainsaw | ManxPower-work: Agreed. I prefer a separate device for fail-over reasons. |
| 15:39.15 | [TK]D-Fender | lordvadr: http://www.telephonydepot.com/Catalog/Analog-Gateways/Mediatrix-4124 |
| 15:39.21 | ManxPower-work | Chainsaw: I'm a fan of channelbanks. |
| 15:39.28 | *** part/#asterisk FloZix (~fmaisonni@93.6.184.151) |
| 15:39.40 | Chainsaw | ManxPower-work: *nod* Something to be said for either approach. Just a collection of TDM cards makes me nervous. |
| 15:40.30 | ManxPower-work | Chainsaw: trying to service 100 lines using the TDMxxP cards is just plain idiocy. |
| 15:40.33 | florz | but for the same latency you'll have to do just as many just as tightly timed transfers for a channel bank ... |
| 15:40.53 | florz | however, wiring might be easier with channel banks ... :-) |
| 15:41.02 | [TK]D-Fender | ManxPower-work: Even just one is nearly qualifying ;) |
| 15:41.41 | [TK]D-Fender | florz: My suggestion = far cleaner, cheaper, and more easily redundant |
| 15:41.43 | ManxPower-work | We might be doing a 2,000 phone analog install at some point soon. That will be interesting |
| 15:41.51 | ManxPower-work | (analog) |
| 15:41.52 | Kobaz | analog, heh |
| 15:41.53 | Naikrovek | wow |
| 15:42.00 | lordvadr | That many cards doesn't make me feel any better either, but the channel banks solve one problem (that might not even exist) and create others with the extra hardware and configuration |
| 15:42.05 | angryuser | sounds like a lot of fun |
| 15:42.15 | Kobaz | there's not much to configure with a channel bank |
| 15:42.23 | Kobaz | t1 signalling, and off you go |
| 15:42.31 | ManxPower-work | 1,000 room OLD hotel with maybe not even Cat3 in the walls. |
| 15:42.45 | Naikrovek | wow |
| 15:42.53 | Naikrovek | that'd be kinda fun |
| 15:42.58 | Naikrovek | at first |
| 15:43.20 | Kobaz | 24 port audiocodes analog gateways are pretty nice |
| 15:43.23 | lordvadr | ok, so now I need 5 pri ports in the server |
| 15:43.29 | lordvadr | and I guess it wouldn't be pri |
| 15:43.32 | ManxPower-work | We are having no end of problems with a 200 Soundpoint IP install. Polycom's support reminds me of Cisco support. |
| 15:43.41 | lordvadr | inband t1? How does that work with 24 channel? |
| 15:43.43 | Naikrovek | really |
| 15:43.47 | Naikrovek | ManxPower-work: what kind of problems |
| 15:43.59 | ManxPower-work | Naikrovek: LLDP bugs in 3.2.2 firmare |
| 15:44.05 | Naikrovek | ah |
| 15:44.09 | Kobaz | ManxPower-work: interesting |
| 15:44.18 | Kobaz | ManxPower-work: i'm having lddp issues at one site |
| 15:44.23 | ManxPower-work | I can't say for sure it is the Polycom or the Adtran switch, but I don't care. I just want it to work. |
| 15:44.29 | Kobaz | we're using adtran too |
| 15:44.32 | Naikrovek | don't polycom support CDP |
| 15:44.41 | ManxPower-work | The customer has been an asshole since day 1. |
| 15:44.45 | Naikrovek | heh |
| 15:44.54 | ManxPower-work | Naikrovek: yes, they do support CDP....unless they see LLDP on the network. |
| 15:45.00 | Naikrovek | ah |
| 15:45.19 | Kobaz | ManxPower-work: the polycom phones at this hospital we're working on.. they are on vlan3, and the pcs are on vlan1... the phones keep getting dhcp from vlan3 |
| 15:45.28 | ManxPower-work | Naikrovek: example: They told us were NOT allowed to send techs out on X date. When X date rolls around we get pissed off calls from the customer asking why a tech is not there. |
| 15:45.37 | Kobaz | er... the phones keep geting dhcp from vlan1 rather |
| 15:45.51 | ManxPower-work | Kobaz: that's what we've experienced. |
| 15:46.01 | Naikrovek | ManxPower-work: awesome |
| 15:46.06 | Kobaz | ManxPower-work: you aren't working on the same hospital are you? |
| 15:46.07 | *** join/#asterisk Chevelle (~chatzilla@190.8.150.148) |
| 15:46.11 | Naikrovek | heh |
| 15:46.34 | ManxPower-work | Kobaz: I have no idea who these people are. I don't deal with them directly. |
| 15:46.44 | [TK]D-Fender | lordvadr: SIP GATEWAY. Cheaper. Faster. Easier. Cleaner. |
| 15:46.52 | Kobaz | ManxPower-work: midtown nyc, trixbox server with like 150 polycom phones, and 4-5 adtran switches? |
| 15:46.54 | [TK]D-Fender | lordvadr: http://www.telephonydepot.com/Catalog/Analog-Gateways/Mediatrix-4124 |
| 15:46.58 | ManxPower-work | If I was dealing with them they would be a former customer. |
| 15:47.24 | Kobaz | we got called in to help them fix their install |
| 15:47.37 | ManxPower-work | Kobaz: somewhere in NY or NJ, 6 Adtran switches at least. |
| 15:47.48 | ManxPower-work | "ICD" |
| 15:47.51 | Kobaz | HAHA |
| 15:47.55 | Naikrovek | same? |
| 15:47.57 | Kobaz | HAHAHA |
| 15:48.01 | Naikrovek | how choice is that |
| 15:48.06 | Kobaz | i was just there last week |
| 15:48.07 | Naikrovek | lol |
| 15:48.23 | ManxPower-work | Ah, so you are the consultant that their consultant brought in to tell the customer we are idiots. |
| 15:48.37 | Kobaz | so it's the polycom phones? I thought we narrowed it down to the adtrans |
| 15:48.43 | lordvadr | TKD-Fender: I'm reading up on that box now. Thanks for pointing me in that direction. |
| 15:48.48 | ManxPower-work | Kobaz: I don't know for sure. |
| 15:48.55 | Kobaz | ManxPower-work: well noone is calling anyone idiots... it's a hardware/software problem somewhere |
| 15:49.13 | ManxPower-work | Kobaz: the customer was. |
| 15:49.26 | Naikrovek | this is awesome. |
| 15:49.28 | Naikrovek | reading this |
| 15:49.33 | Naikrovek | wow |
| 15:49.37 | Kobaz | we worked a full day trying to diagnose the problem |
| 15:49.51 | ManxPower-work | Gerald is my boss. |
| 15:49.54 | Kobaz | it's definitly vlan or lddp related |
| 15:49.57 | lordvadr | But that's an interesting question, with the TDM based channel banks, how is that signaled? Inband pri or just straight tdm? |
| 15:50.01 | Kobaz | ah yeah, i talked to gerald quite a bit |
| 15:50.25 | ManxPower-work | lordvadr: "CAS T-1", not PRI. |
| 15:50.39 | ManxPower-work | Kobaz: Polycom is giving us the runaround |
| 15:51.33 | Kobaz | That adtran switch shouldn't be passing vlan1 tagged packets through though |
| 15:51.41 | ManxPower-work | Kobaz: FreePBX, not Trixbox. Not my choice. |
| 15:52.07 | Kobaz | i only logged in for about 30 seconds, I saw something about trixbox. checked the dhcp server, and that was about it on the asterisk |
| 15:52.20 | *** join/#asterisk Deeewayne (~dwayne@75.76.254.162) |
| 15:52.20 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:52.43 | ManxPower-work | *nod* |
| 15:53.16 | Kobaz | very interesting |
| 15:53.43 | ManxPower-work | I thought maybe they created a loop in the network and that's why the tagged packets were appearing on VLAN 1 |
| 15:53.44 | Naikrovek | ithink between the two of you, you can get it worked out |
| 15:54.05 | Kobaz | Naikrovek: well... we had three seperate companies onsite... and it's still not figured out yet |
| 15:54.27 | ManxPower-work | This is our first install with LLDP switches and Polycoms that support LLDP |
| 15:54.38 | Kobaz | i thought it was a loop too, we elimited that by some physical testing |
| 15:54.39 | Naikrovek | swap out adtran for some other switch vendor and if the problem persists, it's a polycom issue. |
| 15:56.37 | [TK]D-Fender | lordvadr: This way no T1 card needed, you can proxy everything. lower system load, You can reconfigure ports individually without having to take downa all of your channels, etc. No tricky single POF |
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| 16:00.17 | *** join/#asterisk ChannelZ (channelz@burner.com) |
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| 16:02.58 | kjs | hum, how can I check to see if an asterisk trunk is working? |
| 16:03.02 | kjs | sip trunk even |
| 16:03.15 | *** join/#asterisk Slashman (~Slash@ariane.fimasys.com) |
| 16:03.23 | russellb | make a call? |
| 16:03.36 | ManxPower-work | ~trunk |
| 16:03.36 | infobot | extra, extra, read all about it, trunk is a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. |
| 16:03.41 | kjs | iv not sorted the dial plan yet |
| 16:03.54 | kjs | ah ok |
| 16:03.56 | p3nguin | manxpower-work: He's probably talking about Asterisk swimming trunks. |
| 16:03.58 | *** join/#asterisk nix8n82 (~AndChat@63.162.27.14) |
| 16:03.59 | ManxPower-work | kjs: you can't make a call without a dialplan |
| 16:04.07 | russellb | "SIP trunk" is an extremely common term in our industry, I'm sorry ManxPower-work :-) |
| 16:04.12 | ManxPower-work | p3nguin: maybe he's talking about a SIP Truck. |
| 16:04.33 | kjs | well, how can I see if the SIP provider is registered? |
| 16:04.44 | *** join/#asterisk xmitter (~xmitter@c-24-21-213-242.hsd1.or.comcast.net) |
| 16:04.44 | kjs | or will it only register if I try to make a call? |
| 16:04.57 | Kobaz | the thing is... sip trunk is becomming a valid term |
| 16:05.12 | Naikrovek | the more it's used, the more valid it is |
| 16:05.35 | Kobaz | almost every telco and tier1 that provides sip connectivity, says "sip trunking" somewhere on their website |
| 16:05.37 | Naikrovek | use defines language, not vice versa, despite my distaste of that |
| 16:07.15 | Naikrovek | but i'm with ManxPower-work in that it's a term that shouldn't be used |
| 16:07.16 | p3nguin | If they knew what trunking was, they wouldn't use the term. |
| 16:07.38 | Naikrovek | well |
| 16:07.42 | leifmadsen | anyone have a good one liner script for generating a secure password? |
| 16:08.08 | Naikrovek | if it's configured like a trunk, acts like a trunk, maybe it's okay to call it that, despite technical truthiness |
| 16:08.31 | Naikrovek | leifmadsen: i had one but wound up writing a C# program for it |
| 16:08.48 | [TK]D-Fender | kjs: "sip show registry |
| 16:08.50 | leifmadsen | ya, I've found a couple using dd and /dev/random |
| 16:09.01 | Naikrovek | heh |
| 16:09.03 | leifmadsen | I just want to give a simpl example for a document |
| 16:09.04 | ManxPower-work | Exactly how would a "trunk" config be different from a peer/user/friend config? |
| 16:09.07 | Naikrovek | you get unprintables that way though |
| 16:09.31 | leifmadsen | maybe I'll just point to a website |
| 16:09.32 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 16:10.48 | Naikrovek | ebook version should have the generator built in :) |
| 16:11.16 | ManxPower-work | leifmadsen: take random text, MD5SUM it, you have "random" |
| 16:11.38 | raden_work | *yawn* |
| 16:14.14 | *** part/#asterisk c0rnoTa (~c0rnoTa@nas-8.emserv.ru) |
| 16:17.08 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 16:18.29 | p3nguin | manxpower-work: http://imagebin.org/85038 |
| 16:18.43 | Naikrovek | haha |
| 16:19.27 | p3nguin | manxpower-work: http://imagebin.org/85039 |
| 16:19.27 | tuxx- | oneliner to create random passwords? |
| 16:19.28 | tuxx- | <PROTECTED> |
| 16:19.29 | tuxx- | ? |
| 16:19.29 | tuxx- | :D |
| 16:19.38 | p3nguin | tuxx-: uuoc |
| 16:19.48 | Naikrovek | md5sum of one character? |
| 16:19.57 | tuxx- | hehe |
| 16:19.58 | tuxx- | ;D |
| 16:20.07 | tuxx- | Naikrovek: 1 line ;) |
| 16:20.11 | Naikrovek | true |
| 16:20.36 | kjs | what would stop sip show peers |
| 16:20.39 | kjs | from working? |
| 16:20.49 | tuxx- | is your sip module loaded? :P |
| 16:20.55 | kjs | keep saying "no such command" |
| 16:21.02 | kjs | voip01*CLI> sip show peers |
| 16:21.04 | kjs | No such command 'sip show peers' (type 'core show help sip show' for other possible commands) |
| 16:21.21 | tuxx- | show modules like sip |
| 16:21.27 | tuxx- | if chan_sip.so loaded? |
| 16:21.27 | tuxx- | ;p |
| 16:21.29 | tuxx- | *is |
| 16:21.45 | kjs | show - No such command 'show' |
| 16:21.48 | tuxx- | hm |
| 16:21.49 | p3nguin | tuxx-: one line password generator: apg -a1 -n1 -m13 -x26 -MSNCL -E^[]{}\"? -s |
| 16:21.49 | tuxx- | core show? |
| 16:21.49 | tuxx- | :D |
| 16:21.56 | tuxx- | nice one p3nguin :) |
| 16:22.02 | kjs | No such command 'core show' |
| 16:22.03 | p3nguin | "module show like sip" |
| 16:22.10 | tuxx- | ah yes, im still on 1.4 xD |
| 16:22.23 | ManxPower-work | kjs: Your asterisk is seriously screwed up. |
| 16:22.34 | kjs | lol yeah, looks like it... |
| 16:22.38 | kjs | ffs. |
| 16:22.48 | kjs | ill resinstall. (sigh). |
| 16:22.51 | ManxPower-work | I suspect you are missing the [general] or other required sections of sip.conf. |
| 16:22.56 | tuxx- | hm, didnt know that binary, apg :) |
| 16:23.00 | ManxPower-work | kjs: What the hell do you mean "reinstall"? |
| 16:23.10 | kjs | hang on |
| 16:23.14 | ManxPower-work | reinstalling won't fix your problem. |
| 16:23.24 | tuxx- | windows attitude :X |
| 16:23.38 | p3nguin | apg (1) - generates several random passwords |
| 16:23.59 | tuxx- | already looked it up ;) |
| 16:24.01 | p3nguin | Windows attitude would be to "format" and reinstall. |
| 16:24.14 | Kobaz | mkpasswd |
| 16:24.17 | ManxPower-work | p3nguin: I think he's trying to use the sample config files. LOL! |
| 16:24.24 | p3nguin | STILL?! |
| 16:24.33 | p3nguin | That's what was wrong yesterday! |
| 16:24.38 | Kobaz | p3nguin: well yeah... that's the only way to fix windows |
| 16:24.42 | ManxPower-work | p3nguin: He's not too bright. |
| 16:25.28 | p3nguin | I hope you enjoyed the Asterisk trunks. :) |
| 16:26.05 | tuxx- | nice location for the next astricon |
| 16:26.38 | p3nguin | In an antique trunk? |
| 16:26.40 | tuxx- | gaylord nation resort |
| 16:26.41 | tuxx- | w00t |
| 16:26.41 | tuxx- | ;x |
| 16:26.43 | Kobaz | i missed last astricon... i'm definitly going to this years |
| 16:26.47 | tuxx- | *national |
| 16:26.52 | *** join/#asterisk chilicuil (~sistemas@unaffiliated/chilicuil) |
| 16:26.53 | Kobaz | paid for my the company of course |
| 16:26.58 | Kobaz | *by |
| 16:27.01 | tuxx- | :P |
| 16:27.11 | tuxx- | yeah, i hope i get a change to go again this year :) |
| 16:27.15 | tuxx- | *chance |
| 16:27.20 | tuxx- | typo galoree |
| 16:27.32 | *** join/#asterisk cusco (~trilili@2001:0:53aa:64c:1406:7aaa:a077:acde) |
| 16:27.34 | cusco | hello |
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| 16:44.28 | rubberneck | in 1.4 i could "meetme list <meeting> concise" Is there a new way to get the same output in 1.6? |
| 16:45.02 | Kobaz | rubberneck: you should be using the AMI for machine readable output |
| 16:46.03 | rubberneck | Kobaz: Then that is how Ill do it. thanks! |
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| 16:46.28 | Kobaz | most of the "list concise" stuff is going to go away in favor of the ami commands |
| 16:46.50 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 16:47.02 | ManxPower-work | Because it's so totally useless in the CLI. |
| 16:47.05 | ManxPower-work | *eyeroll* |
| 16:47.13 | rubberneck | Kobaz: is there a good place to look for available commands at the AMI? |
| 16:47.20 | Kobaz | one, because the list concise stuff tends to change and break people's programs... and number two...you don't get any verification from the cli commands |
| 16:47.25 | Kobaz | rubberneck: manager show commands |
| 16:47.57 | ManxPower-work | Kobaz: if the CLI commands didn't truncate fields it would not be an issue |
| 16:48.11 | Kobaz | there's also.... if you do asterisk -rx "foobar list concise"... you may get no data back... but did you get no data because there is none? or did you get no data because of an error |
| 16:48.36 | Kobaz | there is no error reporting with the cli list commands |
| 16:48.37 | ManxPower-work | Kobaz: the same applies to every single existing command in Asterisk. Lets remove them all! |
| 16:48.54 | Kobaz | ManxPower-work: no but... that's why you should use the ami so you get error reporting |
| 16:49.19 | ManxPower-work | I always thought "concise" was a funny name anyway. It's not concise output, it's EXPANDED non trunked output. To me that's "expanded", not "concise" |
| 16:49.27 | Kobaz | hehe |
| 16:49.57 | *** join/#asterisk nicknick (~administr@host213-123-201-13.in-addr.btopenworld.com) |
| 16:50.18 | [TK]D-Fender | ManxPower-work: It's "concise" for being non-spaced and delimited in a way to make aprsing easier |
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| 16:51.12 | ManxPower-work | [TK]D-Fender: Oh, I understand that. |
| 16:51.21 | ManxPower-work | I think think it was rather mean to remove the commands from the CLI. |
| 16:51.53 | Kobaz | i don't |
| 16:52.08 | Kobaz | it makes the platform more consistant, and easier to maintain |
| 16:53.10 | ManxPower-work | maybe someone will write a CLI for Asterisk that uses AMI. |
| 16:53.39 | Kobaz | that's probably the most ideal |
| 16:54.03 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-172.dslextreme.com) |
| 16:54.03 | Kobaz | and then the cli is completely independent, and now you can have custom interfaces, which would be great for log filtering |
| 16:54.09 | ManxPower-work | they need to remove the database * commands as well from the CLI. For the exact same reasons you cite. |
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| 17:00.45 | Katty | hmm. |
| 17:00.47 | Katty | bout that time |
| 17:02.56 | rubberneck | when im on the CLI i see "remote unix connection ..." whenever someone else connects. its annoying, so I looked at the source. There is a flag that can be set to turn it off, anyone know where to set that flag? |
| 17:03.22 | rubberneck | I just changed the logging level in the code and recompiled, but id rather not do it that way. |
| 17:04.02 | Qwell | rubberneck: asterisk.conf, probably |
| 17:04.07 | Qwell | displayconnects or something |
| 17:04.55 | Qwell | I may be thinking of manager connections though. *shrug* |
| 17:05.11 | rubberneck | Qwell: hideconenct there it is. thansk man. |
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| 17:11.02 | bmoraca_work | interesting... Congestion() throws a 503 Service Not Available over a SIP channel |
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| 17:11.47 | ManxPower-work | bmoraca_work: there's a Q.931<->SIP mapping RFC |
| 17:12.34 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 17:12.35 | bmoraca_work | i'd never really looked at it closely before. it makes sense, though, i suppose. |
| 17:12.45 | *** join/#asterisk Mhaddog (~Mhaddog@adsl-072-149-063-056.sip.bct.bellsouth.net) |
| 17:13.37 | ManxPower-work | You can use Hangup(XX) where X is the Q.931 cause code to send back custom cause codes |
| 17:13.48 | bmoraca_work | oh, nice |
| 17:14.50 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 17:15.10 | bmoraca_work | it's just interesting because 5xx codes tend to be error codes, whereas 4xx codes would be "unavailable" or "not found" type codes |
| 17:16.26 | *** join/#asterisk n3hxs (~HAMming@static-151-196-93-200.balt.east.verizon.net) |
| 17:22.04 | florz | no, 5xx are temporary errors, and 4xx are permanent errors |
| 17:22.17 | florz | but still, all errors |
| 17:25.28 | bmoraca_work | well, realistically, "service unavailable" describes what is going on in that case anyway, so it fits |
| 17:25.40 | bmoraca_work | i'd just never paid attention to the actual SIP code |
| 17:30.54 | etfonhomey | On an incoming call to an analog FXO port with fax detection enabled, would using Wait() be the best way of allowing it to detect the fax before ringing the receptionist phone? |
| 17:30.58 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 17:33.36 | *** join/#asterisk viq (~viq@unaffiliated/viq) |
| 17:35.31 | ManxPower-work | etfonhomey: It should, if you Answer the line before the wait. |
| 17:36.09 | p3nguin | What's the technical term for this type of coax/antenna connector? http://i50.tinypic.com/2gtalnt.jpg |
| 17:36.53 | *** join/#asterisk fifer (~fifer@67.208.108.228) |
| 17:38.02 | fifer | I'm setting up a macro to dial a phoreign phone system then dial an extension on that phone system all while just just providing a ring to the dialing phone untill the extension pickes up |
| 17:38.29 | fifer | I have done this kind of thing before over 2 years ago but dont have access to my old files and can not find any good examples |
| 17:38.38 | fifer | I'm on 1.4 |
| 17:39.18 | fifer | I have the macro setup and it dials the foreign system but the rest is eluding me |
| 17:41.19 | Naikrovek | p3nguin: that's a common connector, don't remember the name though |
| 17:42.24 | Qwell | p3nguin: The only coax connector name I know is BNC. Is that BNC? |
| 17:42.39 | p3nguin | naikrovek: I found a site that has a cable which connects to the jack, but they refer to the phone connection as "back plug." So that wasn't much help from them. |
| 17:42.57 | p3nguin | qwell: No, it's much smaller than BNC and doesn't twist to lock. |
| 17:42.57 | *** join/#asterisk garymc (~chatzilla@host86-164-36-128.range86-164.btcentralplus.com) |
| 17:42.59 | Naikrovek | Qwell: not BNC |
| 17:43.15 | Naikrovek | p3nguin: http://en.wikipedia.org/wiki/RF_connector |
| 17:43.23 | p3nguin | It resembles a miniature RCA jack. |
| 17:44.18 | Naikrovek | Hirose U.FL perhaps |
| 17:44.24 | etfonhomey | ManxPower-work, thanks for the info. |
| 17:44.39 | *** join/#asterisk soman (~somnath@dsl-jklbrasgw1-fe19fb00-113.dhcp.inet.fi) |
| 17:45.03 | Naikrovek | p3nguin: http://en.wikipedia.org/wiki/MMCX_connector ? |
| 17:45.42 | *** join/#asterisk Dandan (dandan@ip72-221-92-215.ri.ri.cox.net) |
| 17:45.48 | Dandan | hey all :) |
| 17:46.00 | Dandan | I need some help compiling ilibc |
| 17:46.11 | Dandan | I did run ./get... from contrib/scripts |
| 17:46.19 | Dandan | and it has downloaded whatever it needed |
| 17:46.43 | Dandan | now, I can't convince asterisk to compile that codec (it is gone from menuconfig)... |
| 17:46.43 | p3nguin | naikrovek: That's certainly closer than others, but it's not quite the same. The MMCX is found on some cell phones, though. |
| 17:46.49 | Dandan | any help would be appreciated... |
| 17:47.07 | Qwell | p3nguin: google the PN of the device.. you'll have much more luck |
| 17:47.14 | Naikrovek | p3nguin: can't tell from the pic of the MMCX if it has the rubber inside the connector, though this looks like a male and you have a female |
| 17:48.02 | Dandan | s/ilibc/ilbc/ |
| 17:48.09 | p3nguin | naikrovek: Yeah. The one I'm trying to match looks exactly like an RCA jack, but it very tiny. |
| 17:48.21 | Naikrovek | maybe it's micro-RCA or something |
| 17:49.36 | *** join/#asterisk ks3 (~ks3@74.203.195.1) |
| 17:50.06 | *** join/#asterisk ks3 (~ks3@74.203.195.1) |
| 17:56.08 | Dandan | is anyone using ilbc at all? |
| 17:56.15 | Dandan | or compiling it somehow? |
| 17:56.35 | Qwell | Dandan: I think you have to re-run configure after getting the ilbc source |
| 17:56.57 | *** join/#asterisk benngard (~benngard@213.88.138.230) |
| 17:57.07 | Dandan | hm, ok, I will do it once more :) |
| 17:59.58 | Dandan | qwell: valid offer, but it did not help :/ |
| 18:00.07 | Dandan | still missing from Codec Translators |
| 18:00.49 | fifer | Where can I find the most up to date info on the Dial() D flag for * 1.4 |
| 18:02.31 | benngard | core show app dial |
| 18:02.40 | fifer | forgot about that |
| 18:07.28 | fifer | hm, did not work |
| 18:07.33 | fifer | not recognized by c.li |
| 18:07.34 | fifer | cli |
| 18:07.43 | ManxPower-work | then you typed it wrong |
| 18:07.49 | ManxPower-work | core show application dial |
| 18:08.12 | fifer | that is now what he typed |
| 18:08.27 | fifer | thanks! |
| 18:08.47 | tzafrir_laptop | fifer, use the <tab> key to complete commands |
| 18:08.52 | ManxPower-work | he was being lazy |
| 18:08.56 | tzafrir_laptop | (and app names, and whatever) |
| 18:08.57 | fifer | np |
| 18:09.03 | fifer | I have been away from cli for 2 years |
| 18:09.05 | fifer | rusty |
| 18:10.15 | fifer | man, I missed this stuff! |
| 18:13.00 | *** join/#asterisk Slashman (~Slash@ariane.fimasys.com) |
| 18:16.56 | *** join/#asterisk mnt_real (~sinan@bas1-montreal43-1177754737.dsl.bell.ca) |
| 18:17.43 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 18:19.52 | Gido-E | fifer, if i may ask: what where you doing the past 2 years? |
| 18:21.06 | *** part/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 18:23.13 | fifer | Plone |
| 18:23.32 | Naikrovek | Dandan: i think you have to compile iLBC, install it, THEN ./configure in Asterisk source directory |
| 18:23.34 | Naikrovek | oh |
| 18:23.38 | Naikrovek | and add it via make menuconfig |
| 18:23.39 | *** join/#asterisk Godfather_ (~Godfather@129.Red-88-11-88.dynamicIP.rima-tde.net) |
| 18:24.01 | Godfather_ | o/ |
| 18:24.45 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 18:24.45 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 18:25.00 | Dandan | naik: I am sorry what? |
| 18:25.06 | Dandan | You can't "compile" it... |
| 18:25.17 | Naikrovek | Dandan: it's a library when you download it? |
| 18:25.24 | Naikrovek | or is it source |
| 18:25.27 | *** join/#asterisk mike3 (~niter3@CPE00131078a254-CM001ade840b26.cpe.net.cable.rogers.com) |
| 18:25.36 | Dandan | yeah, sources |
| 18:25.49 | Dandan | it is being downloaded and then placed "in appropriate places" |
| 18:25.49 | Naikrovek | so how can't you compile it then |
| 18:25.52 | ManxPower-work | Asterisk downloads ilibc automatically last time I checked. |
| 18:25.53 | Naikrovek | oh |
| 18:26.12 | Dandan | ManxPower: not so much :/ not in 1.6.2 vanilla |
| 18:26.14 | ManxPower-work | Not many people use iLIBC anymore, there are lower CPU options these days |
| 18:26.30 | Naikrovek | polycom 3.2 just added support for it |
| 18:26.40 | Naikrovek | but i use g729 where i can |
| 18:26.55 | Dandan | it is a lab setup, I am just doing it for testing. It is not a showstopper... |
| 18:30.20 | ManxPower-work | Dandan: Try G726 |
| 18:31.04 | Dandan | http://pastebin.ca/1798434 |
| 18:31.20 | Dandan | I got siren7/14, not I am missing ilbc only :-D |
| 18:31.34 | Dandan | as I said: lab setup, I am doing it just because I can... |
| 18:34.16 | *** join/#asterisk diatonic (~diatonic@mail.clearwater-research.com) |
| 18:35.07 | Katty | hi |
| 18:35.57 | Deeewayne | waves to Katty |
| 18:36.09 | Katty | my asterisk does not work /at all/ |
| 18:36.09 | titter | hi |
| 18:36.17 | trelane | grr |
| 18:36.20 | Katty | how to fix pls |
| 18:36.24 | Naikrovek | lol |
| 18:36.24 | Katty | ne1?????? |
| 18:36.27 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 18:36.32 | Naikrovek | omg we r useless |
| 18:36.37 | Dandan | hm, fix it by fixing it! |
| 18:36.40 | Naikrovek | like /part and stuff |
| 18:36.40 | trelane | I have an S101I and a bundle of AC/DC adapters... anyone happen to know which one it goes with? DC voltage, polarity, amperage? |
| 18:36.54 | Dandan | I left my crystal ball at home, can't help you Katty |
| 18:36.54 | trelane | this is what happens when I clean |
| 18:37.12 | Katty | what is crystal ball? how to apply to asterisk?? |
| 18:37.59 | trelane | 9v 1.2A tip positive |
| 18:38.10 | Katty | you guys are no fun :< |
| 18:38.11 | *** join/#asterisk mrbnet (~mrbnet@c-75-73-142-28.hsd1.mn.comcast.net) |
| 18:38.24 | mrbnet | you around? |
| 18:38.28 | Dandan | katty: sorry... |
| 18:38.32 | Dandan | we would need more details |
| 18:38.38 | mrbnet | oops! |
| 18:38.38 | Dandan | on why it doesn't work |
| 18:38.41 | Katty | huh? |
| 18:38.42 | Naikrovek | mrbnet: yes i am around |
| 18:38.47 | Katty | what's broken again? |
| 18:38.55 | Dandan | we are, we all are |
| 18:39.00 | Katty | k |
| 18:39.06 | Katty | WHERE IS DAVE |
| 18:39.06 | ManxPower-work | I'm sorry, Dave. I'm not here right now. |
| 18:39.09 | carrar | Katty did you format? |
| 18:39.15 | carrar | that usually fixes it |
| 18:39.18 | Dandan | format c: /s |
| 18:39.22 | Katty | eww |
| 18:39.25 | carrar | <PROTECTED> |
| 18:39.33 | Dandan | <PROTECTED> |
| 18:39.33 | Katty | keep your windows to yourself. |
| 18:39.39 | mrbnet | Naikrovek: just checking ;-) |
| 18:39.50 | Katty | goes hunting for dave |
| 18:39.55 | grey-monkey | Does anyone know a good open source way to record phone calls so that it doesn't take away from the asterisk machine processor? |
| 18:40.09 | Naikrovek | grey-monkey: how much recording you talking about |
| 18:40.10 | Qwell | grey-monkey: recording doesn't take much CPU |
| 18:40.17 | carrar | grey, use a softphone that records |
| 18:40.18 | Naikrovek | it takes almost no CPU |
| 18:40.24 | ecrane | grey-monkey: use another machine w. port mirroring |
| 18:40.34 | Qwell | There is file I/O to worry about, however. That's the big hit. |
| 18:40.36 | carrar | or use a inline capture device |
| 18:40.41 | Naikrovek | Qwell: yes |
| 18:40.48 | Katty | i kinda like isymphony for recording, actually. it makes it very easy on the users. |
| 18:40.52 | Qwell | Easily solved though. ramdisk works well |
| 18:41.00 | Katty | but i'm not sure how it really effects the CPU...probably does somewhat |
| 18:41.02 | bmoraca_work | Katty: isymphony just uses asterisk's monitor, though. |
| 18:41.16 | Katty | bmoraca_work: yeah but it still executes part of configuration files |
| 18:41.22 | Katty | bmoraca_work: so /technically/ it does use cpu powers |
| 18:41.27 | Katty | POWARS |
| 18:41.28 | bmoraca_work | Katty: but it is a nice interface for users to be able to record at whim |
| 18:41.34 | Katty | yesh. |
| 18:41.37 | Katty | they can push a button. |
| 18:41.39 | Katty | they're cool with that |
| 18:41.50 | bmoraca_work | Katty: right, but you still have to worry about the performance hit to file i/o. |
| 18:42.31 | Katty | mhmm |
| 18:42.31 | carrar | WORRY |
| 18:42.31 | Dandan | windows, eeeew.... |
| 18:42.31 | Katty | your mom |
| 18:42.31 | carrar | PICS PLS |
| 18:42.31 | Katty | so i checked out redditroulette last night |
| 18:42.31 | Katty | pretty awesome. |
| 18:42.36 | Katty | there are GIRLS there. it's amazing. |
| 18:42.38 | Qwell | glares at Katty |
| 18:42.42 | ecrane | my way uses no CPU power on the asterisk box ^^. Use port spanning/mirroring and have your switch direct the traffic to another device. |
| 18:42.48 | Katty | Qwell: perk up buttercup. |
| 18:43.03 | ManxPower-work | girls are overrated |
| 18:43.17 | Dandan | wifes too! |
| 18:43.21 | Dandan | *wives! |
| 18:43.22 | Katty | i know! |
| 18:43.39 | Naikrovek | wives rule |
| 18:43.42 | Naikrovek | mine does anyway |
| 18:43.43 | Naikrovek | love her |
| 18:43.45 | Katty | i totally am overrated. |
| 18:44.11 | Dandan | Naikrovek: you must have gotten married recently (relatively! :P) |
| 18:44.25 | Naikrovek | Dandan: 5 years ago and it's getting better and better |
| 18:44.32 | Dandan | katty: we like your attitude, but that's not good for your self-esteem |
| 18:44.34 | Naikrovek | i do predict a plateu and eventual fall |
| 18:44.36 | Dandan | Naikrovek: pfff :) |
| 18:45.08 | Dandan | they say that the first crisis creeps in around 7th year... |
| 18:45.42 | Katty | Dandan: yeah my self esteem is totally wacked. |
| 18:45.55 | Katty | Dandan: along with my sarcasml |
| 18:45.55 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
| 18:46.58 | Dandan | it's ok, the longer you stay here, the more out of wack it gets :-D |
| 18:47.57 | *** join/#asterisk Alagar (~Administr@122.164.35.103) |
| 18:48.09 | Katty | i blame Qwell. |
| 18:48.16 | titter | Katty: this should fix broken asterisk http://www.youtube.com/watch?v=pbVY5teBzlg |
| 18:48.21 | Dandan | take the blame Qwell! |
| 18:48.22 | Katty | orly. |
| 18:48.27 | Qwell | Dandan: used to it |
| 18:48.34 | Dandan | titter: SFW? |
| 18:48.38 | titter | yes |
| 18:48.45 | Katty | ooooh about transaxles |
| 18:48.48 | titter | rofl |
| 18:48.50 | Katty | oh hey look, it's mister rogers |
| 18:48.51 | titter | just have to listen to it |
| 18:49.01 | Katty | k |
| 18:49.13 | Katty | the what ^_- |
| 18:49.26 | titter | rofl |
| 18:49.41 | *** join/#asterisk clintc (~clintc@n128-227-126-137.xlate.ufl.edu) |
| 18:49.41 | Katty | SPURBING BEARINGS?! |
| 18:49.47 | fifer | I have a Dial() step that calls an outside number and then uses the D flag to call an extension on the far system once it has picked up. |
| 18:49.49 | Katty | WANE SHAFT?! |
| 18:50.07 | Katty | girdle spring? |
| 18:50.09 | Katty | haha |
| 18:50.14 | ManxPower-work | fifer: FXO, FXS, analog, T-1, PRI, SIP? |
| 18:50.25 | Katty | dinglearm? |
| 18:50.25 | fifer | This works fine but after the DTMF has been sent and the call is rining the internal far extension there is silence on the calling end |
| 18:50.58 | fifer | Is there a way to continue sending the user a ring while the call is ringing the far extension but then tetect the final pickup of the call? |
| 18:51.14 | ManxPower-work | Yes, but I'm not telling you until you answer my question. |
| 18:51.20 | fifer | The call is going out a sip trunk |
| 18:51.25 | fifer | lol |
| 18:51.33 | ManxPower-work | Yay! on the far end put "r" on the Dial line. |
| 18:51.45 | ManxPower-work | i.e. on the PBX you are dialing into. |
| 18:51.50 | Katty | FLUX MUSTER |
| 18:52.11 | fifer | ? I have no control over the far PBX? or do you mean at the end of my dial line? |
| 18:52.27 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 18:52.40 | *** join/#asterisk mr_ian (~mr_ian@S0106001b63f49383.du.shawcable.net) |
| 18:52.44 | ManxPower-work | fifer: As far as Asterisk is concerned, the call has been answered by the far end. Everything you want to do must be done on the far end. |
| 18:52.45 | Katty | titter: girdle spring was the best. |
| 18:53.02 | titter | Katty: lol |
| 18:53.16 | fifer | right, that makes sense, just was hoping there may be a trick |
| 18:53.24 | ManxPower-work | You can control the call before the answer, of course. |
| 18:53.53 | Katty | and now for something completely different!!! http://www.youtube.com/watch?v=ywH7KfBK8fw |
| 18:54.13 | Dandan | SFW? :) |
| 18:54.16 | Katty | yes. |
| 18:54.40 | fifer | I can probably mitigate this a little by adding some w's to the and of my DTMF string to account for at least one ring on the far extension but much more than that and I risk the far user picking up to a bit of silence |
| 18:55.13 | Katty | ^- pay special attention to 1:15 |
| 18:59.06 | cusco | 7w |
| 18:59.09 | cusco | hi |
| 18:59.12 | Dandan | 8u |
| 18:59.14 | Dandan | hello |
| 18:59.21 | cusco | heh |
| 18:59.34 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 18:59.42 | cusco | can I have sip.conf port = 5010 |
| 18:59.46 | Dandan | sure |
| 18:59.48 | cusco | and have a softphone working |
| 18:59.54 | p3nguin | But why would you want to? |
| 18:59.54 | cusco | normally with other asterisk |
| 19:00.09 | cusco | I want to test some asterisk configs and I am short on machines |
| 19:00.20 | cusco | so I want to use a operators machine |
| 19:00.24 | cusco | but it is running twinkle |
| 19:00.30 | cusco | and port 5060 is in use |
| 19:00.39 | p3nguin | Since no other SIP devices talk on 5010 unless they are specifically told to, that's not a grand idea. |
| 19:00.49 | p3nguin | Change twinkle to 5061 in twinkle's configuration. |
| 19:00.56 | cusco | a softphone can be told to speak on 5010 |
| 19:00.59 | cusco | like twinkle |
| 19:01.02 | cusco | that is all I need |
| 19:01.04 | Katty | http://www.youtube.com/watch?v=zbr2ao86ww0 <- just in case you haven't seen it. |
| 19:01.52 | p3nguin | cusco: Like I said, change the port IN TWINKLE and leave sip.conf alone. |
| 19:01.57 | cusco | Katty: is that joking with that wedding .. |
| 19:02.02 | cusco | yea it is |
| 19:02.27 | Kobaz | is there a way to turn on polycom's MWI without having it display '1 new message' on the idle screen |
| 19:02.29 | cusco | ok p3nguin I will try that |
| 19:02.42 | Katty | cusco: :> |
| 19:02.53 | Kobaz | i want to use mwi for a non-voicemail notification |
| 19:03.01 | Naikrovek | Kobaz: really |
| 19:03.05 | Naikrovek | what you gonna use it for |
| 19:03.12 | Kobaz | agent login/logout status |
| 19:03.21 | Naikrovek | nice |
| 19:03.33 | Kobaz | i can turn it off and on really easily with the minivm stuff |
| 19:03.39 | Kobaz | but the polycom part of it... it says '1 new message' |
| 19:03.46 | Naikrovek | but the message text shows up.. yeah |
| 19:04.02 | Kobaz | i can't find anything in the docs that would indicate you can control it |
| 19:04.06 | p3nguin | You want the light on the phone to be lit when logged in? |
| 19:04.11 | Kobaz | logged out |
| 19:05.50 | Kobaz | and this is for polycom 330's... so there's no microbrowser |
| 19:06.02 | Naikrovek | yes there is a microbrowser |
| 19:06.13 | Naikrovek | my 320s and 321s and 330s have microbrowsers |
| 19:06.13 | Kobaz | oh, when did they add it for the 330? |
| 19:06.21 | Naikrovek | i don't know when |
| 19:06.27 | Kobaz | they didn't used to |
| 19:06.36 | bmoraca_work | Kobaz: in sip.conf, there is a setting to turn off the display of the message. |
| 19:06.49 | Naikrovek | Kobaz: menu 4 2 |
| 19:06.52 | bmoraca_work | Kobaz: sip.conf as in the polycom config file, not the asterisk peer config file |
| 19:07.25 | Kobaz | yeah |
| 19:07.31 | bmoraca_work | which would make it sip.cfg, i suppose |
| 19:08.12 | [TK]D-Fender | Kobaz: IP 330 has the Microbrowser |
| 19:08.26 | ManxPower-work | [TK]D-Fender: a microbrowser with a 1 line display at least. |
| 19:08.33 | Kobaz | hmm |
| 19:08.37 | Kobaz | interesting |
| 19:09.30 | Kobaz | tests |
| 19:09.45 | Naikrovek | bmoraca_work: do you know the config item? |
| 19:09.55 | Kobaz | bmoraca_work: that would be cool... i haven't found any config for it though... i've looked for quite a while last week |
| 19:09.56 | bmoraca_work | i don't remember exactly |
| 19:10.10 | *** join/#asterisk mikaeel (~lenz@93.177.150.182) |
| 19:10.11 | [TK]D-Fender | ManxPower-work: 2 :) |
| 19:10.12 | bmoraca_work | there's a way to control how new messages display |
| 19:10.15 | bmoraca_work | and i don't remember |
| 19:10.21 | bmoraca_work | i'll see if i can find it |
| 19:10.26 | mikaeel | Hail People! |
| 19:10.37 | Kobaz | bitmap.IP_300.12.name="CtxMail" |
| 19:10.41 | Kobaz | is it that stuff? |
| 19:11.06 | Naikrovek | that's the icon for a registration with a voicemail i think |
| 19:11.16 | Naikrovek | for a line appearance with voice mail, rather |
| 19:11.24 | Kobaz | okay so |
| 19:11.33 | Kobaz | i added a microbrowser config item |
| 19:11.40 | Kobaz | now the phone boots, and says "MicroBrowser -" |
| 19:11.49 | Kobaz | but it's partially cut off, with a line through it |
| 19:12.01 | Kobaz | and it hasn't hit my web page that i set |
| 19:12.09 | *** join/#asterisk p4p4 (~P4p4@248.121.113.82.net.de.o2.com) |
| 19:13.51 | bmoraca_work | hrm |
| 19:13.54 | bmoraca_work | i'm not finding it |
| 19:14.01 | bmoraca_work | i swore it was there, though |
| 19:14.01 | mikaeel | anybody can thell me , asterisk supports the g711Alaw codec? |
| 19:14.04 | Naikrovek | Kobaz: i think they're XHTML sites only |
| 19:14.09 | Naikrovek | mikaeel: of course |
| 19:14.13 | p3nguin | mikaeel: Sure it does. |
| 19:14.21 | mikaeel | thanx Naikrovek |
| 19:14.24 | bmoraca_work | mikaeel: nope, it only supports speex |
| 19:14.32 | Naikrovek | mikaeel: supports a lot more than that, too |
| 19:14.35 | Kobaz | Naikrovek: yeah i know... i have a hello world in a little html file |
| 19:14.39 | p3nguin | mikaeel: core show codecs |
| 19:14.56 | Naikrovek | Kobaz: ah okay |
| 19:15.09 | mikaeel | great, thanx p3nguin! |
| 19:15.20 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 19:15.20 | Naikrovek | Kobaz: there's a microbrowser dev kit on polycom's support site i think |
| 19:15.32 | mikaeel | I connect to the avaya Definity via open H323 |
| 19:15.52 | p3nguin | mikaeel: And if you want to see what codecs your system can transcode, run core show translation. |
| 19:15.53 | mikaeel | but the applications does not work |
| 19:15.56 | ManxPower-work | Naikrovek: there was supposed to be, it was delayed until Mar 1. There is an Web Apps development guide that refers to the mythical Polycom SDK |
| 19:16.08 | Kobaz | k |
| 19:16.48 | mikaeel | p3nguin, now I recompile asterisk with new souce |
| 19:16.58 | mikaeel | try to do lather |
| 19:17.16 | mikaeel | but you know, inbound calls from avaya all failed |
| 19:17.37 | mikaeel | and outbound works well |
| 19:18.10 | p3nguin | mikaeel: Does core show translation show the codec that you want to use? |
| 19:18.30 | *** join/#asterisk rpm- (~rpm@S01060014f691c3c1.cg.shawcable.net) |
| 19:18.44 | mikaeel | I will ry lather, asterisk is down now :) |
| 19:19.23 | *** join/#asterisk chilicuil (~sistemas@unaffiliated/chilicuil) |
| 19:19.28 | mikaeel | whole day I try to make working this interkonnect with both systems |
| 19:22.52 | Dandan | hm, does anyone use Skype for * here? is it business stable? or justa gimmick? |
| 19:30.08 | benngard | hmm, trying to send a fax (fax - spa2012 - asterisk - spa2102 -fax).. getting ast_rtp_read: Unknown RTP codec 100 received, i think i read something about it but cant remember what/where :( |
| 19:32.02 | Dandan | http://www.asteriskguru.com/tutorials/unknown_codec_received.html |
| 19:32.05 | Dandan | benngard: ^^ |
| 19:33.19 | benngard | i know i read about it |
| 19:33.25 | Dandan | then fix it |
| 19:33.54 | Katty | you fix it |
| 19:34.15 | benngard | but it doesnt make sense, i cant find any settings in the spa2102 that shows t30 |
| 19:34.16 | Dandan | I can't, I am incarcerated. |
| 19:34.27 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 19:34.33 | Dandan | benngard: it should say t.38 though |
| 19:34.41 | Dandan | t.30 is an old way and unsupported... |
| 19:35.10 | *** join/#asterisk moy (~moy@74.12.129.100) |
| 19:36.54 | bmoraca_work | benngard: codec 100 is g729. are you sure your asterisk has g729 loaded? have you configured the SIP peers to allow g729? |
| 19:37.17 | p3nguin | You're not going to want to fax with g.729, anyway. |
| 19:37.44 | Dandan | lol, why not, if he can do it, that would be like winning an olympic medal! :) |
| 19:37.56 | benngard | bmoraca_work: my spa has ONLU g711 alaw |
| 19:38.00 | benngard | ONLY |
| 19:38.10 | *** join/#asterisk Tech_Travis (~tech_trav@208.179.137.131) |
| 19:38.14 | benngard | let me double check |
| 19:38.32 | bmoraca_work | benngard: well, it's trying to use g729, so you might want to check on that |
| 19:38.43 | benngard | thx, let me check |
| 19:39.21 | bmoraca_work | or asterisk is trying to use g729 to talk to it, which isn't supported. you have a codec mismatch. if you're trying to fax, you'll want to stick to g711 on all sides. |
| 19:39.46 | benngard | a number: Codec Order : (alaw:40) |
| 19:40.14 | bmoraca_work | check the configuration of both devices, then. |
| 19:41.01 | benngard | b number: Codec Order : (alaw:20) |
| 19:41.19 | benngard | let me turn ioff t38 and try |
| 19:41.19 | bmoraca_work | i'm not talking about your asterisk configuration, i'm talking about the actual device itself |
| 19:41.50 | benngard | Preferred Codec: g711a at both ends |
| 19:42.11 | benngard | Use Pref Codec Only: yes at both ends |
| 19:42.17 | *** join/#asterisk mike3 (~niter3@CPE00131078a254-CM001ade840b26.cpe.net.cable.rogers.com) |
| 19:42.26 | bmoraca_work | take a sip debug of the failed call, then. the bottom line is that someone is trying to use g729. a sip debug will tell you who is. |
| 19:42.50 | *** join/#asterisk jmacz (~jmacz@190.125.223.23) |
| 19:42.50 | benngard | sec plz |
| 19:42.55 | ManxPower-work | benngard: add allowguest=no in sip.conf [general] |
| 19:43.15 | benngard | oki |
| 19:43.22 | benngard | ManxPower-work: oki |
| 19:43.36 | ManxPower-work | then do a sip reload, and try it again. |
| 19:43.45 | benngard | will do |
| 19:44.03 | ManxPower-work | It won't solve the issue, but it will help point the way to the solution |
| 19:44.17 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 19:45.03 | mike3 | aurgh.. |
| 19:45.03 | mike3 | :) |
| 19:46.46 | *** join/#asterisk TimeRider (~steve@78.32.26.1) |
| 19:50.49 | *** join/#asterisk eppigy (~Dave@216-139-241-102.aus.us.siteprotect.com) |
| 19:50.51 | eppigy | hello |
| 19:50.52 | eppigy | i am dave |
| 19:51.06 | [TK]D-Fender | eppigy: THESE ARE THE DAVES I KNOW |
| 19:51.09 | eppigy | YES |
| 19:52.19 | *** join/#asterisk lanning (~lanning@208.87.235.224) |
| 19:52.30 | eppigy | Katty: PING |
| 19:55.32 | *** join/#asterisk binbash_ (~peter@ip4da53781.direct-adsl.nl) |
| 19:57.54 | Naikrovek | just download a polycom microbrowser sample applications kit and has them working on his ip 320 right now. it sucks |
| 19:59.01 | Naikrovek | tiny tiny tiny screen is useless |
| 20:06.46 | ManxPower-work | http://www.polycom.com/support/voice/spip_ssip_vvx_SDK_availability.html |
| 20:06.52 | [TK]D-Fender | Naikrovek: You should only du ultra-minimal things on an IP3Xx series |
| 20:07.14 | Naikrovek | no kidding |
| 20:07.47 | *** join/#asterisk rtilder (~rtilder@cpe-76-174-37-230.socal.res.rr.com) |
| 20:10.46 | benngard | how will a "g729 packet" look like whe u do a sip set debug on? |
| 20:11.17 | bmoraca_work | benngard: it doesn't "look" like anything. you'll see it being advertised by one or the other peer in the INVITE message |
| 20:11.20 | Qwell | RTP packets don't show up in SIP debug |
| 20:11.31 | Qwell | You'll need RTP debug (and it'll show up as an RTP packet...since that's what it is.) |
| 20:11.42 | ManxPower-work | benngard: What happened when you set allowguest=no in sip.conf [general] ? |
| 20:11.50 | *** join/#asterisk RobH (~robh@2620:0:860:2:21e:c2ff:fe03:2465) |
| 20:11.57 | benngard | ManxPower-work: the same |
| 20:12.30 | ManxPower-work | benngard: Have you put your sip.conf on pastebin.ca yet? If not, do so now, masking ONLY passwords |
| 20:12.32 | benngard | i am looking through the "dump" of a failed call cant find anything like g729 |
| 20:12.51 | bmoraca_work | benngard: pastebin it |
| 20:12.55 | benngard | will do |
| 20:13.02 | ManxPower-work | whoch codec is it finding? |
| 20:13.15 | ManxPower-work | 256, right? |
| 20:13.22 | bmoraca_work | 100 |
| 20:13.35 | ManxPower-work | There is no such codec. |
| 20:13.36 | benngard | sec, the stupid thing is when i turn off t38 it is working |
| 20:13.57 | ManxPower-work | g729 is codec number 256 |
| 20:14.13 | bmoraca_work | it's 100 in hex |
| 20:14.32 | ManxPower-work | no, it would be 0x100 in hex. |
| 20:14.34 | *** join/#asterisk cuco (~Diego@109.253.57.27) |
| 20:14.59 | bmoraca_work | in the invite debug, i believe it shows the bitmask for the codecs in hex |
| 20:16.23 | bmoraca_work | makes a lot more sense to me, at least, to see 104 as supporting both g729 and ulaw versus seing 260 |
| 20:16.53 | bmoraca_work | benngard: you don't need t38 when faxing over g711 |
| 20:17.22 | bmoraca_work | t38 is useful for faxing over compressed codecs (like g729) |
| 20:17.23 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 20:18.29 | ManxPower-work | looks like benngard is trying to do both FaxOverVoiceOverIP AND FaxOverT.38 at the same time. No wonder the poor sod is having problems. |
| 20:19.52 | *** join/#asterisk Mark22 (~mark@unaffiliated/mark21) |
| 20:20.03 | bmoraca_work | would still like to see a sip debug... |
| 20:20.27 | benngard | here comes my sip dot conf |
| 20:20.30 | [TK]D-Fender | bmoraca_work: this has been a really poor couple of days when it comes to people providing backup... |
| 20:20.31 | benngard | http://pastebin.com/m18a33719 |
| 20:20.44 | benngard | would u like to have the sip debug to? |
| 20:21.12 | Mark22 | Hello, I get the following warning (a part from the config file above the warning): http://yourpaste.net/4749/ on Asterisk 1.6.2.1 |
| 20:21.18 | ManxPower-work | benngard: which do you want to do? FaxOverVoiceOverIP(G711) or FaxOverT-38? Pick one. |
| 20:21.26 | Mark22 | what is wrong? or where should I look for the fix |
| 20:21.32 | benngard | fax over g711 |
| 20:21.52 | ManxPower-work | benngard: You don't have allowguest=no. The disable T.38 on your SIPUra |
| 20:22.17 | Dandan | Mark22: drop "" |
| 20:22.21 | ManxPower-work | benngard: remove the T.38 stuff if you are not going to use it. |
| 20:22.44 | benngard | but in the avaya world i come from u need to enable t38 for reliable fax even if u are using g711 |
| 20:22.59 | ManxPower-work | benngard: T.38 does not use G711 |
| 20:23.01 | Dandan | has digium been aquired by avaya? |
| 20:23.22 | ManxPower-work | benngard: Pick one! Do you want FaxOverG711 or FaxOver T.38. You cannot do both. |
| 20:23.44 | benngard | then i will add another bug to my really long avaya cm bug list |
| 20:24.02 | benngard | faxoverg711 |
| 20:24.22 | ManxPower-work | Then remove the T-38 stuff. |
| 20:24.33 | benngard | yes np |
| 20:24.34 | ManxPower-work | and put in that allowguest=no like you said you did. |
| 20:24.43 | Mark22 | Dandan: that fixed it, I now see why it did work in the example I did look at and not in our situation (there they have "" around all contexts) |
| 20:24.52 | Dandan | Mark22: :) |
| 20:25.37 | Kobaz | okay so I got the microbrowser stuff working nicely |
| 20:25.46 | benngard | ManxPower-work: i have a php/postgresql script that generate the sip.conf, did forget to add it there when i did an update so ofc my allowguest=no was removed :( |
| 20:26.12 | Kobaz | if you have the microbrowser going, then the 'x messages waiting' doesn't appear when you kick on mwi |
| 20:26.21 | Kobaz | so, next question... how do i program soft keys on polycoms |
| 20:26.42 | *** join/#asterisk Yudaisrael1984 (~Yuda@80.179.161.1.static.012.net.il) |
| 20:27.45 | Yudaisrael1984 | hey guys how u all doing ? i have a quick question i need to debug audio issues what commands do i have in order to start debugging? (basicly calls that come into the system that i want to connect outside of the system are not passing any audio. |
| 20:28.13 | Dandan | that might be issues with firewall or other rtp traffic... |
| 20:29.09 | ManxPower-work | Kobaz: stop. go. read the polycom admin and web dev guide. Then read it again. then again. |
| 20:29.38 | bmoraca_work | Yudaisrael1984: "sip set debug" will pretty much get you everything you'll need to know |
| 20:29.56 | Mark22 | another question, what ports should I open on a firewall that is located between asterisk and other SIP servers and SIP clients? Currently all ports are open and we want to close a few ports |
| 20:30.06 | Kobaz | ManxPower-work: i've been reading, and none of this is in there |
| 20:30.12 | [TK]D-Fender | ~sipnat |
| 20:30.13 | infobot | [~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 |
| 20:30.14 | [TK]D-Fender | ^^^^^^^^^^^^ |
| 20:30.17 | ManxPower-work | Kobaz: of course it is in there. |
| 20:30.22 | Naikrovek | Mark22: udp:5060, and udp:10000-20000 |
| 20:30.28 | Naikrovek | Mark22: and what [TK]D-Fender said |
| 20:30.47 | Yudaisrael1984 | problem is that there is too much chatter on sip set debug |
| 20:31.05 | ManxPower-work | Kobaz: Page 4-40 for 3.2.2 |
| 20:31.15 | Kobaz | ManxPower-work: there's info about how to set the physical soft key behavior |
| 20:31.20 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 20:31.20 | bmoraca_work | Yudaisrael1984: log to a file and sort it out from there. copy it from putty and pastebin it if you want us to be able to help you. |
| 20:31.24 | Kobaz | like. o can remap soft key 1 to transfer, or something |
| 20:31.51 | Kobaz | mmm |
| 20:31.54 | Mark22 | [TK]D-Fender: we don't use NAT, it is a transparant firewall :) |
| 20:31.55 | Yudaisrael1984 | i will try to do that and also try to clean up because its going to be very long |
| 20:32.08 | ManxPower-work | Polycom has at least three app notes on EFK too, with one example specific to Asterisk. |
| 20:32.17 | [TK]D-Fender | Mark22: Same ports |
| 20:32.37 | [TK]D-Fender | Mark22: And the answer wasn't only for you |
| 20:32.39 | bmoraca_work | Yudaisrael1984: your potential problem will be readily apparent. deleting things will only make it harder to identify what the potential problem might be, because you might delete something we want to see |
| 20:32.40 | Dandan | Naikrovek: why would you need 10k ports open? |
| 20:32.45 | Dandan | are you expecting 10k concurrent calls? |
| 20:32.53 | Mark22 | but UDP ports 10000-20000 should be enough? :) |
| 20:33.03 | p3nguin | dandan: 10000 ports is 5000 calls |
| 20:33.06 | [TK]D-Fender | Dandan: Could but upwards of 4 ports per call <- |
| 20:33.06 | Dandan | that is still way to many... |
| 20:33.06 | ManxPower-work | Dandan: Because Asterisk DEFAULTS to ports 10000-20000 |
| 20:33.10 | Dandan | *STREAMS* |
| 20:33.12 | Yudaisrael1984 | i know how to see which line is related to which call |
| 20:33.13 | [TK]D-Fender | be* |
| 20:33.16 | benngard | ManxPower-work: http://pastebin.com/m510145df better? |
| 20:33.24 | Yudaisrael1984 | its from an active system running right now 25 concurrent calls |
| 20:33.26 | *** join/#asterisk Z_God (~julius@wlan233081.mobiel.utwente.nl) |
| 20:33.37 | Yudaisrael1984 | its kind of ridiculas to send all the info |
| 20:33.38 | Dandan | Heh, I was thinking streams, wrote calls :/ |
| 20:33.39 | Dandan | my bad |
| 20:33.42 | Z_God | is it possible to enable speex wideband for the echo test in asterisk? |
| 20:33.46 | Dandan | [TK]D-Fender: how come? |
| 20:33.48 | Z_God | I'm using the twinkle softphone |
| 20:34.09 | Z_God | sip^ |
| 20:34.16 | ManxPower-work | benngard: now pastebin the output of a failed call. |
| 20:35.19 | benngard | ManxPower-work: with that config i dont get any failed calls |
| 20:35.24 | [TK]D-Fender | Dandan: 2 for audio eachw ay, sip to the device for OOB signalling, and heck lets assume video too.. thats upwards of 6 |
| 20:35.27 | ManxPower-work | benngard: there you go. |
| 20:35.41 | Dandan | [TK]D-Fender: you are right... with canreinvite=no that doubles... |
| 20:35.46 | Dandan | duh! |
| 20:37.48 | benngard | ManxPower-work: sorry to say but i know how to "break" your theory, just put in a big pile of paper in the fax, like 10.15 of them and i wil get a failed call |
| 20:38.16 | *** join/#asterisk binbash_ (~peter@ip4da53781.direct-adsl.nl) |
| 20:38.18 | benngard | i see it happen every day |
| 20:39.13 | ManxPower-work | benngard: I've never seen FaxOverVoiceOverIP ever actually work. There's always reliability issues. Almost anyone here will tell you the same thing. |
| 20:39.46 | *** part/#asterisk Mhaddog (~Mhaddog@adsl-072-149-063-056.sip.bct.bellsouth.net) |
| 20:39.59 | ManxPower-work | I work for a CLEC and we still install POTS lines for fax. |
| 20:40.04 | Z_God | is there actually any support for speex wideband right now? |
| 20:40.46 | ManxPower-work | I wish you the BEST of luck, benngard. |
| 20:41.09 | Dandan | nods... |
| 20:41.43 | bmoraca_work | i've had fax over voip calls go for 20 pages and not fail. i've also had them work 95% of the time (out of a very large sampleset). if you do it right, it shouldn't be a problem with g711 as the codec. |
| 20:42.12 | voipmonk | bmoraca_work: what isp do you use? |
| 20:42.30 | Mark22 | bmoraca_work: here it works correctly |
| 20:42.30 | ManxPower-work | bmoraca_work: where "problem" means "more than %5 failure rate"? |
| 20:42.34 | bmoraca_work | telepacific, but it's also worked over AT&T uVerse and various other providers |
| 20:42.42 | Kobaz | aww |
| 20:42.48 | leifmadsen | agreed, I've had fax over voip work for me as well, but don't have a large sample set |
| 20:42.57 | Kobaz | you need some sort of license for polycom efk, to provision soft keys |
| 20:43.05 | ManxPower-work | Kobaz: WRONG! |
| 20:43.06 | Naikrovek | no |
| 20:43.24 | Naikrovek | Kobaz: it's all there in the admin guide |
| 20:43.24 | Kobaz | 0710140805|efk |4|03|EFK not licensed. |
| 20:43.25 | Kobaz | 0710140805|efk |4|03|EFK not enabled. |
| 20:43.42 | voipmonk | shows Kobaz the old manuals ManxPower-work used to read.... |
| 20:43.46 | ManxPower-work | Kobaz: are you running 3.2.2 firmware? |
| 20:43.47 | bmoraca_work | ManxPower-work: meh. it's always about tradeoff. i tell the customer they have two options: pay a little to build fax into their PBX or pay a little bit more for a POTS line. i always let them know about the issues. most are OK with it |
| 20:44.21 | Kobaz | oh crap |
| 20:44.23 | Kobaz | whoops |
| 20:44.27 | ManxPower-work | bmoraca_work: we would lose money on fax over VoIP simply from the significant support issues. |
| 20:44.27 | voipmonk | do you use spandsp, bmoraca_work ? |
| 20:44.33 | Naikrovek | that's some OOLD log you're lookin at there also, kobaz |
| 20:44.41 | Naikrovek | oct 14 2007 |
| 20:44.44 | ManxPower-work | Naikrovek: think he was testing us? |
| 20:44.49 | Kobaz | hmm |
| 20:44.50 | Naikrovek | dunno |
| 20:44.57 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 20:45.13 | Kobaz | 0216153113|efk |*|01|Initial log entry. Current logging level 4 |
| 20:45.16 | Kobaz | whoops again |
| 20:45.19 | Kobaz | that's more recent |
| 20:45.36 | ManxPower-work | Kobaz: what specific issue are you trying to solve. |
| 20:45.46 | bmoraca_work | voipmonk: spandsp is only needed if you want asterisk to terminate the faxes. if you're just passing audio, it's not necessary. g711 is the same resolution used over POTS lines, so it's just a matter of jitter and latency at that point. if it's good, you won't have problems |
| 20:45.51 | Kobaz | adding a login/logout to the soft keys |
| 20:45.57 | Kobaz | i've never played with efk before |
| 20:46.00 | ManxPower-work | If you thought Asterisk was complicated, just wait until you begin to understand the polycom docs |
| 20:46.08 | voipmonk | bmoraca_work: is that a yes or a no? |
| 20:46.12 | Kobaz | yeah, i know polycom is nutty |
| 20:46.13 | Naikrovek | <3 polycom |
| 20:46.25 | bmoraca_work | voipmonk: it's a "yes" in that i use it when i need asterisk to terminate faxes. |
| 20:46.30 | Kobaz | i've been using polycom for 3 years now, and I barely know what they can do |
| 20:46.56 | Naikrovek | Kobaz: honestly they're better than any other phone i've used, and i've used granstream, snom, and cisco |
| 20:46.57 | ManxPower-work | I've spent the past 3 months doing custom configs for Polycoms. |
| 20:47.04 | voipmonk | bmoraca_work: do you use t.38 as well? |
| 20:47.10 | bmoraca_work | voipmonk: no. |
| 20:47.11 | Naikrovek | ManxPower-work: same! super configurable |
| 20:47.14 | Kobaz | grandstream is horrible |
| 20:47.24 | Kobaz | the audio quality is worse than a $9 walmart phone |
| 20:47.48 | Dandan | has over 400 endpoints: cisco 7940/60s and no problems with those... |
| 20:47.55 | voipmonk | what ITSP are you using with telepacific? |
| 20:48.10 | bmoraca_work | myself |
| 20:48.13 | voipmonk | bmoraca_work: what ITSP are you using with telepacific |
| 20:48.26 | Naikrovek | Kobaz: i was reading the timestamp wrong. they don't have years in them. MMDDhhmmss is the format |
| 20:48.33 | voipmonk | bmoraca_work: do you have a website where I can buy services? |
| 20:48.37 | Kobaz | Naikrovek: yeah |
| 20:49.27 | Kobaz | Naikrovek: the weird thing about grandstream, is that the speakerphone gives you better quality audio than the handsets |
| 20:49.27 | bmoraca_work | we only offer service in California. tax/regulation reasons. the CA PUC doesn't recognize VOIP as a utility, so I don't have to worry about collecting any taxes or getting registered as a CLEC |
| 20:49.55 | Kobaz | the handsets are always all buzzy |
| 20:50.43 | Naikrovek | on mine, i'd hang up the phone, the phone would ring, i'd pick up, and i'd be back on the same call talking to the same person |
| 20:50.46 | Naikrovek | they were useless |
| 20:50.54 | Kobaz | heh |
| 20:51.15 | voipmonk | bmoraca_work: do you have a website where I can buy services? |
| 20:51.24 | Kobaz | i have some gxp2000's that i use for testing... it's nice they have 4 lines |
| 20:51.40 | Kobaz | one died completely though... i started with three |
| 20:51.51 | Naikrovek | Kobaz: i have one of those collecting dust |
| 20:52.01 | Naikrovek | it will collect dust until the plastic rots away |
| 20:52.06 | Kobaz | heh |
| 20:52.46 | benngard | have a lot of audiocodecs boxes that he can sell verey cheap! |
| 20:52.47 | Dandan | that's what I meant to ask... any good (and cheap) residential VoIP provider? (callcentric/broadvoice excluded..) for residential only though, I would port my number in... |
| 20:53.03 | Dandan | benngard: talk to me with prices and models |
| 20:53.10 | Dandan | I can msg you my email if you need to |
| 20:53.24 | Kobaz | voicepulse is decent |
| 20:53.49 | Dandan | ok, will give them a try |
| 20:53.50 | Kobaz | jsut make sure you keep an eye on your balance... sometimes your autofill gets turned off, and they have a $20 dollar negative balance fee |
| 20:54.02 | Kobaz | although if you call them up, they will waive it when it's their fault |
| 20:54.12 | Dandan | lol, that's good |
| 20:54.22 | benngard | Dandan: i was kidding i wont sell them to the devil himself, the break down faster than i can finish a beer |
| 20:54.55 | Dandan | Kobaz: I assume, no issues with * whatsoever...? |
| 20:54.58 | *** join/#asterisk danj1980 (~dan@91.109.81.119) |
| 20:54.59 | danj1980 | ir |
| 20:54.59 | Dandan | benngard: lol :) |
| 20:55.03 | danj1980 | Hi |
| 20:55.07 | Kobaz | Dandan: it depends |
| 20:55.10 | Dandan | ? |
| 20:55.27 | Kobaz | if you could random crashes as issues... then... no |
| 20:55.28 | Naikrovek | wow |
| 20:55.35 | Kobaz | could=count |
| 20:55.40 | Naikrovek | been doing IT for 15 years, just got my first compsci question from a user |
| 20:55.40 | Dandan | kobaz: on their end? |
| 20:55.43 | Kobaz | oooh |
| 20:55.45 | Kobaz | voicepulse |
| 20:56.04 | Kobaz | no, no issues with voicepulse... other than they were down for about 4 hours, in mid 2009 |
| 20:56.10 | Kobaz | i thought you were asking about asterisk issues |
| 20:56.43 | Dandan | nah, I can deal with any issues on my end :) |
| 20:56.52 | danj1980 | I was wondering... I've seen Openser mentioned along side asterisk. Mainly when discussing redundancy. What does the benefit of using Openser and is it necessary? |
| 20:57.22 | Naikrovek | it's like an application-level load balancer |
| 20:57.33 | Naikrovek | as i understand it |
| 20:57.51 | danj1980 | So its only used when you have more than 1 asterisk server? |
| 20:58.07 | Naikrovek | you put openser in front of X number of asterisk servers and it'll distribute the connections between servers. when a server goes away it won't send connections there anymore |
| 20:58.09 | Dandan | not necessarily |
| 20:58.28 | Dandan | asterisk works only as ivr when connected with openser and openser does call routing... |
| 20:58.52 | Kobaz | and then when your openser server goes away, then you don't have anything |
| 20:59.09 | Kobaz | so if you want truely fully failsafe stuff, you need two or more of everything |
| 20:59.14 | Dandan | they you have two opensers |
| 20:59.33 | danj1980 | If you use openser for load balancing, would asterisk still generate the cdrs for billing? |
| 20:59.39 | danj1980 | or openser? |
| 20:59.40 | Kobaz | sure |
| 20:59.43 | Kobaz | either way |
| 20:59.51 | Kobaz | depends what you're doing |
| 20:59.52 | Naikrovek | danj1980: yeah, if they point to the same db |
| 21:00.01 | danj1980 | ah |
| 21:00.02 | danj1980 | ok |
| 21:01.09 | Naikrovek | danj1980: leifmadsen has a site where he has some presentations on high availablity and scalable asterisk setups |
| 21:01.09 | danj1980 | I found a network diagram online showing an asterisk setup with redundancy and it showed openser, so I was wondering how it worked. |
| 21:01.13 | danj1980 | thanks for the info |
| 21:01.37 | danj1980 | Naikrovek: do you have the address? i'd be interested in taking a look. |
| 21:01.50 | Naikrovek | liefmadsen.com? i go check |
| 21:01.54 | leifmadsen | www.leifmadsen.com should have links to them |
| 21:01.59 | Naikrovek | there ya go |
| 21:02.15 | leifmadsen | http://leifmadsen.com/node/5 |
| 21:02.34 | leifmadsen | also check http://leifmadsen.wordpress.com for some links to other sites and other blog posts too |
| 21:02.44 | Naikrovek | plug |
| 21:02.52 | danj1980 | Thanks |
| 21:02.55 | danj1980 | Appreciated. |
| 21:02.58 | Naikrovek | i need to work with someone who has done big asterisk installs |
| 21:03.10 | Naikrovek | i poked around on one over the weekend but i need to pick at someone's brain for a few weeks |
| 21:03.31 | leifmadsen | http://leifmadsen.wordpress.com/2009/11/18/astricon-2009-presentation-building-a-distributed-call-center/ <-- check the comments too |
| 21:03.46 | danj1980 | leifmadsen: how did you get to the point of charging $200 an hour? You must have quite a bit of experience! |
| 21:03.58 | leifmadsen | danj1980: I've used Asterisk for 7 years :) |
| 21:04.01 | danj1980 | Its impressive. |
| 21:04.15 | leifmadsen | I also co-authored a book on asterisk (along with jsmith and jim van meggelen) |
| 21:04.15 | danj1980 | Cool |
| 21:04.51 | danj1980 | Great book. thanks for making it available in pdf |
| 21:05.00 | Naikrovek | yes |
| 21:05.07 | Naikrovek | i also thank you all for that book |
| 21:05.19 | Naikrovek | will buy 3rd ed when it comes out |
| 21:05.22 | *** join/#asterisk Warp4 (~robert.wo@firewall-a.buf.ny.i-evolve.net) |
| 21:05.53 | Dandan | oh, leif, how is progress on tfot 3? you should update your blog/rss once in a while! |
| 21:06.10 | Naikrovek | he updated 6 days ago |
| 21:06.14 | Naikrovek | that's pretty recent |
| 21:06.21 | Dandan | (but nothing in regards to tfot) |
| 21:06.28 | Naikrovek | ah true |
| 21:06.34 | leifmadsen | Dandan: I just wrote a blog the other day :) Still working on it. Jim and I have a new method we're using to try and work on it every day as things have been going slow. |
| 21:06.36 | Naikrovek | he's holding on for the big reveal |
| 21:06.37 | Naikrovek | he's a showman |
| 21:06.44 | leifmadsen | well, nothing new to report on TFoT 3 yet :) |
| 21:06.58 | leifmadsen | "Yep, still working on it. Nope, haven't killed myself yet." |
| 21:06.58 | Dandan | yeah, he is, met him in Phoenix a couple of years ago :) |
| 21:07.27 | Naikrovek | i meet child tv stars because i take my kids for autographs |
| 21:07.33 | Naikrovek | i don't meet anyone who can teach me anything |
| 21:07.36 | Naikrovek | <-- idiot |
| 21:07.39 | Dandan | lol :> |
| 21:08.04 | Naikrovek | met like 3 kids from icarly |
| 21:08.08 | Naikrovek | eh who cares |
| 21:08.16 | Naikrovek | reads up on cisco vpn configuration |
| 21:08.28 | Naikrovek | s/vpn/vlan/ |
| 21:08.31 | Dandan | eazzzie... :) |
| 21:08.38 | Naikrovek | i know |
| 21:08.44 | Naikrovek | just wanna change the subj |
| 21:09.02 | Naikrovek | only so many millionaire teenagers i can take before i wanna die |
| 21:10.32 | bmoraca_work | wow...when did regexten get added? |
| 21:11.11 | danj1980 | I'm in the process of investing in my own hardware. Currently I use cloud hosting for my asterisk, but a power supply failed on the host and I was down for 7 hours (luckily it was sunday morning so it was quiet). |
| 21:11.20 | danj1980 | What do people prefer? Dell or HP? |
| 21:11.30 | Corydon76-dig | danj1980: Apple |
| 21:11.33 | leifmadsen | IBM |
| 21:11.39 | Dandan | Asus :> |
| 21:11.44 | leifmadsen | Lenovo |
| 21:11.55 | leifmadsen | Gigabyte |
| 21:11.58 | Corydon76-dig | Transmeta |
| 21:12.03 | Dandan | easy, we do not need no chinese spies here :P |
| 21:12.06 | leifmadsen | (Actually in all seriousness, I like Dell and HP) |
| 21:12.17 | Dandan | I am dell and penguincomputing guy :) |
| 21:12.20 | Naikrovek | ibm blades for server |
| 21:12.42 | danj1980 | How many servers do you need to make blade worth it? |
| 21:13.00 | Naikrovek | well i can fit 14 servers in 7u |
| 21:13.18 | *** join/#asterisk RobH (~robh@cpe-173-169-30-118.tampabay.res.rr.com) |
| 21:13.19 | Naikrovek | depends entirely on what scale you are working with |
| 21:13.21 | danj1980 | Dell sales said to me that blade is only worth it if you need to save space and want 30 to 50 servers. |
| 21:13.22 | Dandan | depneding on capacity planning |
| 21:13.33 | Naikrovek | 1 blade, blade server completely and totally not worth it |
| 21:13.43 | Naikrovek | 84 servers in a single rack? bladecenter |
| 21:13.48 | Naikrovek | worth it worth it worth it |
| 21:13.56 | danj1980 | Ok, I get that |
| 21:14.14 | Corydon76-dig | Naikrovek: you just need an 84u rack. ;-) |
| 21:14.26 | Naikrovek | if you only have 1 or two servers, get some 64-bit hardware and virtualize |
| 21:14.26 | danj1980 | Also, do people lease or purchase their servers? Sorry for the questions. Just wondering what route to take. |
| 21:14.45 | Naikrovek | danj1980: i know people that rent time on Amazon EC2 for their asterisk installs |
| 21:14.47 | bmoraca_work | Naikrovek: HP has a new server style out that fits two distinct, non-blade servers into a single 1U box |
| 21:14.51 | Naikrovek | i tested it a while back and it worked great |
| 21:14.57 | Naikrovek | bmoraca_work: nice |
| 21:15.12 | Corydon76-dig | buys servers once they come off lease... |
| 21:15.46 | Corydon76-dig | They're usually only 1-2 years old, one generation behind, and extremely inexpensive |
| 21:15.49 | danj1980 | Naikrovek: I use cloud hosting from another company. Much lower pricing compared to Amazon |
| 21:16.00 | Naikrovek | danj1980: yeah there are cheaper solutions, no doubt |
| 21:16.01 | Naikrovek | though |
| 21:16.08 | Naikrovek | they just added a pricing feature to comete |
| 21:16.11 | Naikrovek | comptete* |
| 21:16.14 | Naikrovek | ugh |
| 21:16.14 | danj1980 | Corydon76-dig: where do you find the ex-lease server? |
| 21:16.16 | Naikrovek | you know what i maen |
| 21:16.30 | Corydon76-dig | danj1980: they get sold in bulk to resellers |
| 21:16.49 | Naikrovek | danj1980: i was paying $0.025/hr the other day. normal price is $0.10 I think |
| 21:17.20 | danj1980 | Naikrovek: why the discount? |
| 21:17.26 | Naikrovek | $0.085 normal price, I was paying $0.029/hr |
| 21:17.37 | Naikrovek | danj1980: you can rent unused capacity at a much lower rate |
| 21:17.43 | Corydon76-dig | NautilusNet.com usually has post-lease servers for good prices |
| 21:18.01 | danj1980 | The problem with clouds is that you pay for bandwidth. |
| 21:18.05 | Naikrovek | it's all unused capacity, but they sell the big majority of their unused capacity at much lower |
| 21:18.09 | Naikrovek | danj1980: yes, and cpu time |
| 21:18.24 | Naikrovek | danj1980: but for some people, it is suitable. |
| 21:18.42 | *** join/#asterisk evil_gordita (~evilgordi@ip70-188-50-186.rn.hr.cox.net) |
| 21:18.46 | danj1980 | Its been great to start up with. But i'm not sure if its going to work out in the long term |
| 21:19.20 | danj1980 | I need to be able to say to myself that i've built a redundant asterisk platform with almost no single points for failure. |
| 21:19.37 | Corydon76-dig | danj1980: http://www.nautilusnet.com/pages/InventoryBrowse.aspx?Action=Edit&pK=117&InStock=true |
| 21:19.39 | Dandan | Corydon76-dig: where from do you buy them? |
| 21:19.46 | danj1980 | Corydon76-dig: thanks for the link, but im UK based |
| 21:19.58 | Naikrovek | danj1980: it can be done. but as you get closer to 100% redundant, the cost raises exponentially in time and money |
| 21:20.22 | bmoraca_work | Naikrovek: vSphere, baby! |
| 21:20.24 | danj1980 | I agree, but if you have a failure, clients and the public can remember it for years. |
| 21:20.27 | Naikrovek | bmoraca_work: oh yeah |
| 21:20.33 | Naikrovek | virtualization is the way 2 go |
| 21:20.41 | Corydon76-dig | danj1980: I'm sure there are resellers on that side of the pond, too |
| 21:20.55 | danj1980 | I'll do some googling |
| 21:21.11 | Naikrovek | wish they had more than 1g of ram those servers |
| 21:21.17 | Dandan | danj: well, good luck, that will be expensive to do... |
| 21:22.09 | bmoraca_work | Corydon76-dig: i've been buying old HP DL380s from Aventis. i like them because they warranty for 3 years and they usually come stocked with RAM and hard drives |
| 21:22.10 | danj1980 | I can probably manage it with 3 physical servers and virtualising. |
| 21:22.15 | Corydon76-dig | danj1980: don't forget to go with 2 locations, geographically spaced and on different uplinks |
| 21:22.37 | Dandan | danj1980: you can't... no virtualizing if you go for HA |
| 21:22.40 | danj1980 | Corydon76-dig: sure, but i'm going to have to leave that out for the moment. |
| 21:22.44 | Corydon76-dig | bmoraca_work: doesn't matter where, but I agree that the secondary market is an excellent place to get servers |
| 21:22.52 | Dandan | or rather: minimize it, as you are introducint another layer of possible failure |
| 21:23.00 | Dandan | *introducing |
| 21:23.01 | bmoraca_work | yep |
| 21:23.06 | danj1980 | My first investment is 2 openser servers and 2 asterisk servers. (maybe virtualised for now) |
| 21:23.30 | Corydon76-dig | danj1980: in fact, your best colo is going to be whereever your provider is located |
| 21:23.39 | bmoraca_work | Dandan: vSphere can use active/standby failover for sub-second failover. yes, calls will drop, but your downtime will be very minimal |
| 21:23.47 | Dandan | master-master for sql/drbd/lvm and then openser - redundant and asterisk - redundant, in two different locations and loadbalanced |
| 21:23.50 | Corydon76-dig | Cross-connect within the same building |
| 21:24.16 | *** join/#asterisk niekvlessert (~niek@5ED25657.cable.ziggo.nl) |
| 21:24.18 | Dandan | bmoraca_work: I know you can do it with canreinvite=yes and no downtime on opensers |
| 21:24.21 | danj1980 | 2 locations is a long way down the line. |
| 21:24.25 | niekvlessert | hello, how would you do an attended transfer (with Aastra SIP phones) when you want to be able to do some dialplan stuff when it happens? |
| 21:24.48 | danj1980 | Hopefully with 2 providers to 1 location, there shouldnt be a case where we need to switch to a 2nd location. |
| 21:24.52 | danj1980 | Unless there is a fire. |
| 21:24.53 | danj1980 | :-S |
| 21:25.29 | Corydon76-dig | danj1980: you cannot plan for every situation if you're only in a single location |
| 21:25.30 | bmoraca_work | Dandan: depends on the application. can't do reinvites if your peers are behind firewalls. vSphere is useful in that it's a very, very simplistic topology that will give you almost perfect HA without having to worry about having multiple servers and other stuff like DUNDi |
| 21:26.03 | Dandan | Corydon76-dig: agreed |
| 21:26.06 | danj1980 | Corydon76-dig: I totally agree, but I have limited funds. Im going to have to do it in stages. |
| 21:26.26 | *** join/#asterisk Dibri (~gavit@pop1.isgroup.sr) |
| 21:26.33 | Corydon76-dig | My upstream provider recently moved from NYC to Chicago, for electrical rate reasons |
| 21:27.01 | Dandan | then do not claim to be almost no points of failure :) |
| 21:27.38 | bmoraca_work | "Nine 9's of uptime"? |
| 21:27.42 | danj1980 | When did I claim that? That is my eventual goal in the short term future. |
| 21:27.49 | Dandan | Corydon76-dig: they will move way out west, WA electricity rate is like 1/7 of that of this here, in CT... |
| 21:27.56 | carrar | yeah |
| 21:28.05 | carrar | we have cheap power in central WA |
| 21:28.07 | *** join/#asterisk ZX81 (~Matt_Ridd@121-74-248-216.telstraclear.net) |
| 21:28.08 | danj1980 | What do you think first step in creating a redundant asterisk platform if I was going to do it in stages? |
| 21:28.14 | Dandan | 16:19 < danj1980> I need to be able to say to myself that i've built a |
| 21:28.15 | Dandan | <PROTECTED> |
| 21:28.15 | Dandan | <PROTECTED> |
| 21:28.16 | Corydon76-dig | Dandan: There's something to be said for a colo in Herndon, VA, though |
| 21:28.46 | *** part/#asterisk ZX81 (~Matt_Ridd@121-74-248-216.telstraclear.net) |
| 21:28.53 | Corydon76-dig | It's the closest there is to a "center of the Internet" |
| 21:29.00 | danj1980 | Dandan: correction, i should have said "I'd need to eventually be able to say to myself..." |
| 21:29.02 | carrar | and old vacated SAC buildings they could make datacenters |
| 21:29.11 | carrar | 6' thick walls |
| 21:29.12 | Corydon76-dig | MAE-East meets MAE-West |
| 21:29.14 | Dandan | danj1980: hehehe, that's all cool :) |
| 21:29.21 | danj1980 | Currently, i'm on cloud hosting, I dont have my own infrastructure. |
| 21:30.02 | Corydon76-dig | Heh, currently Comcast is my uplink, and my infrastructure is about 3 feet away from me |
| 21:30.21 | danj1980 | Corydon76-dig: what kind of bandwidth do you have? |
| 21:30.28 | Dandan | 20/6 probably :) |
| 21:30.29 | Corydon76-dig | Consumer grade |
| 21:30.30 | bmoraca_work | i reside in my CLEC's datacenter. my PRI uplinks are directly into their switches |
| 21:31.18 | Corydon76-dig | Only thing I'm really hosting are IVRs for some nonprofit toll-frees |
| 21:31.27 | *** join/#asterisk aces1up (~ri@ip70-173-52-152.lv.lv.cox.net) |
| 21:31.50 | danj1980 | I'm hosting a few virtual PBX's for a handful of companies, but I want to grow it. |
| 21:32.04 | aces1up | so i have a crap load of avaya 4412d phones, it says these are digital phones, does that mean they are ip phones? will they work with asterisk? |
| 21:32.23 | Corydon76-dig | aces1up: Highly unlikely |
| 21:32.32 | Corydon76-dig | on both counts |
| 21:32.47 | Dandan | benngard was our Avaya specialist :) |
| 21:32.57 | Corydon76-dig | digital phones is another way of saying overpriced proprietary crap |
| 21:33.00 | bmoraca_work | aces1up: if they're not IP phones, you may be able to use a Portico TVA to use them with Asterisk |
| 21:34.00 | aces1up | hrmm, well my boss just bought them off ebay, and as far as i can tell there is not "system" to plug them into.. with digital phones do they typically just work from the phones? or do they need a pbx of some sort to plug into? |
| 21:34.27 | bmoraca_work | aces1up: they'll almost always need a PBX |
| 21:34.28 | Dandan | there needs to be a mothership :) so they ca communicate with |
| 21:34.31 | Dandan | s/ca/can/ |
| 21:34.52 | aces1up | yeh, i thought so.. |
| 21:35.05 | aces1up | i don't see any mothership, hence they great price :) snicker... |
| 21:35.17 | aces1up | i think he got like 50 phones for 999.00 |
| 21:35.50 | bmoraca_work | aces1up: they aren't real IP phones, though they are designed to be used with the Avaya IP Office |
| 21:35.58 | benngard | Dandan: i am deep in an avaya with may atm213 |
| 21:36.23 | danj1980 | Ok guys, I have to sign out. Thanks for the discussion and advice! Much appreciated. |
| 21:37.05 | *** join/#asterisk doolittlework (~d@196.211.34.2) |
| 21:37.16 | aces1up | bmoraca, would they work with that citel device you were talking about? or would you think that alot of the functionality would break if they don't plug into the avaya ip office.. |
| 21:37.30 | aces1up | and how much does one of those avaya ip office pbx go for anyhow? |
| 21:37.46 | bmoraca_work | aces1up: nope, i just checked and they're not compatible with the Portico TVA |
| 21:37.50 | aces1up | anyway, i'm looking for solutions on how to use these phones in the best way possible. |
| 21:38.15 | aces1up | bmoraca hrmm well are they compatible with anything then? other that the avaya ip office? |
| 21:38.16 | bmoraca_work | aces1up: use them to smack your boss with for making a stupid decision like this without first having a purpose |
| 21:38.22 | Corydon76-dig | aces1up: put them on ebay and use the proceeds to buy Polycom IP phones |
| 21:38.23 | bmoraca_work | aces1up: no |
| 21:38.34 | aces1up | heh.. i thought so. |
| 21:38.50 | Katty | hmmmmmms. |
| 21:38.58 | aces1up | so he would have to buy the avaya ip office seperately to do something with these phones right? |
| 21:39.05 | bmoraca_work | yes |
| 21:39.23 | aces1up | i see.. hrmm have a good vendor to check pricing on that? |
| 21:39.31 | bmoraca_work | nope |
| 21:39.47 | *** join/#asterisk benngard (~benngard@213.88.138.230) |
| 21:39.47 | aces1up | bmoraca no way at all to connect these to asterisk with SOME sort of adapter? |
| 21:39.54 | bmoraca_work | nope |
| 21:40.03 | aces1up | really? why no options? |
| 21:40.16 | aces1up | cause its propriety digital? |
| 21:40.19 | bmoraca_work | they're designed to specifically work with a specific piece of hardware |
| 21:40.26 | aces1up | hrmm. |
| 21:40.46 | bmoraca_work | the Portico TVA can emulate the PBX for a variety of types of phones, including some Avaya phones, but not that particular one |
| 21:41.19 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
| 21:41.32 | aces1up | bmoraca and no other vendors do that type of emulation? |
| 21:42.00 | aces1up | had another site mention this system working with these phones. |
| 21:42.02 | aces1up | Merlin Magix telephone system |
| 21:44.05 | aces1up | crapola... found pricing.. |
| 21:44.07 | aces1up | $4,691.99 |
| 21:44.21 | aces1up | jeez.. |
| 21:45.15 | *** join/#asterisk mmattice (mmattice@unaffiliated/mmattice) |
| 21:45.35 | mmattice | where should I start looking for causes for "chan_dahdi.c:3360 pri_find_dchan: No D-channels available! Using Primary channel 24 as D-channel anyway! |
| 21:46.13 | mmattice | I have been able to call into this pri several times today, and no config has changed |
| 21:54.38 | *** join/#asterisk SeriousMatters (~Sirius@87.115.170.210) |
| 21:55.28 | Yudaisrael1984 | guys where can i get the ARI files from? |
| 21:57.50 | *** join/#asterisk hipitihop (~denis@203.132.229.187) |
| 21:58.16 | *** join/#asterisk caim (~user@unaffiliated/caim) |
| 21:58.55 | *** part/#asterisk etfonhomey (~etfonhome@74-143-192-74.static.insightbb.com) |
| 21:59.13 | bmoraca_work | Yudaisrael1984: ARI is part of FreePBX. try there. |
| 21:59.25 | Yudaisrael1984 | thanks |
| 22:00.26 | hipitihop | is it possible to integrate and control existing ATA voip hardware ? |
| 22:00.57 | *** join/#asterisk shin-shoryuken_ (~shin-shor@hephaestus.restechservices.net) |
| 22:01.48 | bmoraca_work | hipitihop: you're going to have to be more specific than that. |
| 22:02.38 | Kobaz | asterisk call queues are really limiting |
| 22:04.56 | hipitihop | bmoraca_work, I have a small Linksys PAP2T which does my current Voip at home and I was wondering if/how that could be integrated |
| 22:06.00 | bmoraca_work | hipitihop: it probably can't be. if it was provided by an ITSP, it is probably locked to that ITSP. if you want to be able to modify the settings of it, you'll probably have to talk to them. |
| 22:06.50 | hipitihop | bmoraca_work, no, it is an open one I bought and configured myself |
| 22:07.14 | bmoraca_work | then all you need to do is create a peer and set the credentials/proxy address. |
| 22:09.10 | hipitihop | bmoraca_work, sorry not up to all the lingo yet, but does that mean then asterisk would be in control of and aware of all activity on that ATA |
| 22:09.25 | bmoraca_work | not all, no. |
| 22:09.27 | *** join/#asterisk smooth_penguin (~smoove@59.95.36.80) |
| 22:09.33 | bmoraca_work | it sounds like you need to read "the book" |
| 22:09.36 | bmoraca_work | ~book |
| 22:09.37 | 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 |
| 22:10.18 | hipitihop | bmoraca_work, indeed I'll go and do that and stop wasting your time with noob questions. thanks for your help |
| 22:11.31 | Z_God | anyone here who can tell me something about asterisk's support for wideband audio codecs? |
| 22:12.37 | doolittlework | what does noob mean? |
| 22:13.13 | cusco | hi |
| 22:13.34 | cusco | how can I find out if the patch for issue 0015609 is in 1.6.2.3 rc2 ? |
| 22:14.07 | doolittlework | <PROTECTED> |
| 22:17.15 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 22:27.00 | bmoraca_work | Z_God: asterisk can transport most wideband codecs, but I believe that the only one it can actually terminate is speex. i don't believe that it can transcode any of the other ones. |
| 22:27.11 | bmoraca_work | someone else will correct me if i am wrong |
| 22:27.29 | Z_God | bmoraca_work: alright, I'm dealing with speexwb here now, but I'm running into issues with it |
| 22:29.52 | Z_God | this is the error asterisk keeps spamming: |
| 22:29.54 | Z_God | frame.c:1410 speex_samples: Not enough bits remaining after wide band for |
| 22:29.55 | Z_God | > speex samples |
| 22:30.14 | bmoraca_work | i'm not familiar enough with speex to be able to help you, sorry |
| 22:30.20 | *** join/#asterisk garymc (~chatzilla@host86-164-36-128.range86-164.btcentralplus.com) |
| 22:30.49 | Z_God | ok, anyone else saw this before? google only gives me results of the asterisk source code .. |
| 22:32.14 | *** join/#asterisk andinbg (~andilist@DSL01.83.171.155.238.ip-pool.NEFkom.net) |
| 22:32.28 | drmessano | Are you using Wideband on a device or PBX <> PBX? |
| 22:33.16 | andinbg | hi, can anybody give me some hints with a problem with SIP and AGI |
| 22:33.32 | drmessano | ?ask |
| 22:33.47 | drmessano | :( |
| 22:33.51 | drmessano | ~ask |
| 22:33.52 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 22:34.59 | raden_work | how can i show the time that asterisk is writing to the log files |
| 22:35.05 | raden_work | cause i dont believe its the system time |
| 22:35.49 | andinbg | ok, i have two asterisk-servers connected to the same sip-account (asterisks er clients). the first server answering should get the call. if a call arives, both servers start the agi-script and answers the call. (both servers get SIP INVITE. One gets ACK the other CANCEL. But the AGI-script does not recognize the cancel) |
| 22:36.46 | andinbg | whats even more strange: the server whcih receives CANCEL can get the Callers dtfm inputs. |
| 22:37.39 | benngard | ManxPower-work: just for your information, we did manage to negotiate from g711 to t38 on both sip and ooh323, sometimes it is a god idea to do that, will do some stress tests tomorrow |
| 22:37.42 | SeriousMatters | I make installed the new version 1.6.2.2 and got some errors |
| 22:38.41 | SeriousMatters | asterisk -r |
| 22:38.41 | SeriousMatters | Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 22:38.52 | SeriousMatters | tail /var/log/asterisk/full |
| 22:38.52 | SeriousMatters | tail: cannot open `/var/log/asterisk/full' for reading: No such file or directory |
| 22:38.53 | Kobaz | SeriousMatters: you have piqued my interest, and i would like to subscribe to your newsletter |
| 22:39.51 | SeriousMatters | service asterisk start works fine. but it just won't start at boot time. |
| 22:40.16 | SeriousMatters | Kobaz: what news letter? |
| 22:40.26 | Kobaz | it's a joke... heh |
| 22:40.43 | Kobaz | google it.. anyways |
| 22:40.48 | SeriousMatters | I tried. |
| 22:40.50 | Kobaz | something is wrong with your startup sequence then |
| 22:40.51 | raden_work | what does asterisk use to set the time in the LOGS ?>>>>>> |
| 22:41.14 | Kobaz | SeriousMatters: no... google the newsletter bit... not the asterisk issue |
| 22:41.36 | Kobaz | being unable to connect to asterisk is a sure indication that it's not running |
| 22:41.39 | SeriousMatters | everyone says tail /var/log/asterisk/full to see why. but there is no such file |
| 22:41.46 | andinbg | so, how should agi react on a SIP CANCEL |
| 22:41.48 | Kobaz | so, look at your logs. and figure out why it's not running |
| 22:41.50 | SeriousMatters | yes, its not running. |
| 22:42.29 | SeriousMatters | root# service asterisk start can force it to start. but I need to auto start at boot time |
| 22:43.07 | Kobaz | so fix your startup sequence |
| 22:43.25 | Kobaz | asterisk obviously needs something that's starting later on in the boot sequence |
| 22:43.29 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 22:43.33 | Kobaz | figure out what it is, and start it before asterisk does |
| 22:44.02 | Kobaz | but in theory, asterisk doesn't really depend on anything else... or rather, it shouldn't unless it's configured to |
| 22:44.11 | Kobaz | so something in your config is wacky also |
| 22:44.23 | Kobaz | but then again, you're not really giving us any details, so all I'm doing is guessing |
| 22:44.29 | SeriousMatters | where is my log anyway? |
| 22:44.55 | Kobaz | it's wherever you set up asterisk to log to |
| 22:45.08 | SeriousMatters | its a fresh install |
| 22:45.15 | raden_work | ok why are all my asterisk logs in GMT ??? |
| 22:45.25 | Kobaz | SeriousMatters: that doesn't tell me anything |
| 22:45.41 | SeriousMatters | I didn't set it to log anywhere |
| 22:45.55 | Kobaz | fresh install could mean, you coppied the sample configs and just hoped that it would work. or you have no configs at all |
| 22:46.09 | Kobaz | if you didn't set it up to log, why are you expecting to see a log file? |
| 22:46.27 | SeriousMatters | hm... I was expecting it to log by default |
| 22:46.33 | Kobaz | it doesn't do anything by default |
| 22:46.50 | *** join/#asterisk trikis2010 (~chatzilla@189.210.155.67) |
| 22:46.51 | *** join/#asterisk etfonhomey (~etfonhome@74-131-159-160.dhcp.insightbb.com) |
| 22:47.00 | Kobaz | just like your computer doesn't do anything by default, other than try to boot the first storage device it finds |
| 22:47.13 | Kobaz | and if that storage device is empty... well... nothing is going to happen |
| 22:47.20 | trikis2010 | Hi I just bought 20 licenses for g729 codec where do I find the KEYS |
| 22:47.30 | trikis2010 | or will they be emailed |
| 22:47.40 | Kobaz | trikis2010: talk to whoever you bought them from |
| 22:47.58 | Kobaz | SeriousMatters: read the asterisk book first |
| 22:47.59 | Kobaz | ~book |
| 22:48.00 | 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 |
| 22:48.08 | Kobaz | SeriousMatters: it's free.. on the pdf download |
| 22:48.20 | Kobaz | SeriousMatters: and then once you've read that... check out /etc/asterisk/logger.conf |
| 22:48.36 | etfonhomey | trikis2010, you'll get the key in an email which you use to register the licenses on your * box. |
| 22:48.42 | trikis2010 | I bought from the digium's online store |
| 22:49.29 | trikis2010 | does it takes to much? I have been waiting for about 1/2 hour |
| 22:49.30 | etfonhomey | trikis2010, I did the whole process last week. If you haven't received the email, check your spam filter. |
| 22:50.35 | SeriousMatters | well, I used a yum install version before and it logs default. |
| 22:50.41 | trikis2010 | etfonhomey, did it took a lot of time? |
| 22:51.03 | SeriousMatters | but it has a bug so I end up makeing from source of new version. |
| 22:51.06 | Kobaz | SeriousMatters: okay so the package maintainer populated some configs... find them, and make sure they are set up right |
| 22:52.08 | *** part/#asterisk andinbg (~andilist@DSL01.83.171.155.238.ip-pool.NEFkom.net) |
| 22:52.09 | Kobaz | SeriousMatters: there's really nothing i can tell you that will magically fix it... you'll have to dig up as much information you can find... it's a "help us... help you" type of thing |
| 22:52.25 | SeriousMatters | sure sure, of course |
| 22:53.43 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 22:54.37 | etfonhomey | trikis2010, it took one minute from when I received the order confirmation email to when I received the email with the registration key. |
| 22:55.31 | trikis2010 | eftonhomey the think is that on tha store sais placed but i havent receive the confirmation email yet |
| 22:56.31 | carrar | USER ERROR |
| 22:57.11 | etfonhomey | trikis2010, then either the emails aren't making it to you (wrong email address, spam filter, etc.) or you didn't complete the order process all the way. I would login to your digium account and check out your order history/status to confirm it actually went through. |
| 23:04.16 | *** join/#asterisk bmoraca (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 23:04.21 | bmoraca | got my ass kicked |
| 23:04.46 | trikis2010 | etfonhomey, at order history sais status placed but no email |
| 23:07.01 | carrar | trikis2010, how about you call Digium? |
| 23:08.25 | trikis2010 | carrar do the have a tolfree number or just + 1 256 428 6000 |
| 23:08.45 | carrar | I can google digium for you if you need me too |
| 23:08.58 | trikis2010 | carrar sorry |
| 23:09.36 | carrar | press your life alert button |
| 23:10.55 | *** join/#asterisk Dibri (~gavit@190.98.32.116) |
| 23:14.50 | *** join/#asterisk fifer (~fifer@67.208.108.228) |
| 23:15.17 | *** join/#asterisk pmhaddad (~Phil@71-13-218-72.dhcp.mrqt.mi.charter.com) |
| 23:15.20 | etfonhomey | trikis2010, let me look at my order status to see if it is different than "Placed". Maybe they are holding up your credit card or something. |
| 23:16.51 | etfonhomey | trikis2010, my Order Status is "SUBMITTED". So your order may not be complete yet. |
| 23:16.53 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 23:17.06 | trikis2010 | ok Ill wait then thanks a lot |
| 23:18.26 | etfonhomey | trikis2010, I will say that the response time for my order was immediate, but I am in the states. |
| 23:18.48 | trikis2010 | ok thanks a lot Im in Mexico |
| 23:20.19 | Yudaisrael1984 | another quick question is it possible to have freepbx and a2billing running on the same server while in freepbx its set up to give a2billing an extension that a2billing should use and only that and basicly set it up as a trunk in a2billing? |
| 23:20.21 | Yudaisrael1984 | is that possible |
| 23:21.17 | SeriousMatters | is there something you want from freepbx that a2b doesn't have? |
| 23:22.00 | Yudaisrael1984 | basicly having freepbx deciding on which route to take when dialing out and a2billing to do the authentication |
| 23:22.52 | Yudaisrael1984 | i plan on using a2billing as a call thru setup and i want to use a2billing to authenticate and to do all the billing for me and then i want free pbx to be in front of a2billing and that it should give a2billing a user as a trunk |
| 23:24.46 | Yudaisrael1984 | so in theory i want freepbx to create a user in sip.conf which isnt going outside the system but it would be configured on the same server though its to be used only by a2billing . did u understand? |
| 23:25.58 | SeriousMatters | I see, so you want to have a normal asterisk server but bill them. |
| 23:26.34 | Yudaisrael1984 | and to do other things |
| 23:26.35 | Yudaisrael1984 | yes |
| 23:27.12 | Yudaisrael1984 | the question of course is the same question if it was the opposite way |
| 23:27.31 | *** join/#asterisk LemensTS (~customgt@adsl-70-238-130-71.dsl.stlsmo.sbcglobal.net) |
| 23:27.50 | [TK]D-Fender | Yudaisrael1984: FreePBX is not supported here, it has its own channel. |
| 23:27.52 | [TK]D-Fender | ~freepbx |
| 23:27.53 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 23:28.02 | [TK]D-Fender | Yudaisrael1984: And a2billing isn't supported in either |
| 23:28.03 | Yudaisrael1984 | its not regarding freepbx |
| 23:28.09 | Yudaisrael1984 | its a general question |
| 23:28.11 | Yudaisrael1984 | in theory |
| 23:28.35 | Yudaisrael1984 | its more for asterisk programmers then the other channels |
| 23:28.44 | Yudaisrael1984 | its not about freepbx D-fender |
| 23:28.46 | LemensTS | phone A calls phone B. Phone B parks the call. Phone B picks up the parked call at 701. Is there anyway to figure out the username of phone A now? |
| 23:28.50 | [TK]D-Fender | [18:24]<Yudaisrael1984>so in theory i want freepbx to create a user in sip.conf which isnt going outside the system but it would be configured on the same server though its to be used only by a2billing . did u understand? <- this IS a FreePBX question |
| 23:28.58 | [TK]D-Fender | Yudaisrael1984: Configs are built by it |
| 23:29.04 | beta2k | I know trixbox itself isn't generally supported here but I'm hoping this is generic enough that someone might be able to chime in about it. I'm trying to enable g729 as a codec, I've added allow=g729 to sip_general_custom.conf (which is included by sip.conf) and set my sip ATA to use g729, yet it still shows as g711u on the ata as how the call is comming in (ringing atleast) |
| 23:29.06 | LemensTS | Ive tried sipchaninfo, it gives me phone B info |
| 23:29.30 | Yudaisrael1984 | D-Fender it is a theoretical question im not talking about the program itself of freepbx |
| 23:29.32 | [TK]D-Fender | beta2k: Go look at what your Extension is using |
| 23:30.10 | Yudaisrael1984 | the question is can asterisk use a trunk which is also a friend |
| 23:30.34 | [TK]D-Fender | lemenyudthat doesn't make any sense.... |
| 23:30.56 | beta2k | [TK]D-Fender: Which file should they be defined in? |
| 23:31.30 | LemensTS | TKD-Fender: phone A calls phone B. Phone B parks the call. Phone C picks up the parked call at 701. Is there anyway to figure out the username of phone A now? <===Phone A and Phone C are in the call |
| 23:32.51 | LemensTS | TKD-Fender: all the channel variables clear on park. |
| 23:33.51 | LemensTS | TKD-Fender: I thought about storing in mysql or setglobalvar, but i don't see where I would know what phone or what channel it was to call the data back... |
| 23:33.59 | [TK]D-Fender | beta2k: in your FreePBX EXTENSION config page |
| 23:34.22 | [TK]D-Fender | ~cpid |
| 23:34.23 | infobot | [~cpid] Called-Party ID is possible with * using patches on Mantis. See : http://bugs.digium.com/view.php?id=8824 |
| 23:34.24 | [TK]D-Fender | LemensTS: ^^^ |
| 23:36.03 | Yudaisrael1984 | Ok so i can explain differently i have user [test_A] [test_B] and [trunk_A]. [test_A] has set up to a context that has _XXXX,..... Dial(SIP/test_b) and [test_b] has in the context of _XXXX,.......... Dial(SIP/Trunk_A/ .......... IS THIS POSSIBLE |
| 23:36.59 | Yudaisrael1984 | sorry bout the caps at the end |
| 23:37.05 | *** join/#asterisk SaiSoma (~SaiSoma@adsl-074-167-136-030.sip.mob.bellsouth.net) |
| 23:38.43 | [TK]D-Fender | Yudaisrael1984: You can do anything you want to in your dialplan |
| 23:38.46 | Yudaisrael1984 | D-Fender is the question better this time? (it has nothing to do with a2billing , freepbx, or any of those its the programming of asterisk thats important here) |
| 23:38.56 | Yudaisrael1984 | but would it work |
| 23:39.22 | [TK]D-Fender | Yudaisrael1984: Devices point to contexts. Contexts have extensions. You put whatever kind of extensions that do whatever you in them |
| 23:41.03 | Yudaisrael1984 | but its a device pointing to itself as a different name |
| 23:41.19 | Yudaisrael1984 | so i dont see how that should work |
| 23:41.24 | [TK]D-Fender | Yudaisrael1984: A device cannot point to a DEVICE |
| 23:41.33 | beta2k | [TK]D-Fender: Doesn't show any options for setting codecs, that's why I assumed I needed to go edit a config :) |
| 23:41.34 | [TK]D-Fender | Yudaisrael1984: A device points to DIALPLAN |
| 23:41.49 | [TK]D-Fender | beta2k: Via the GUI in that magic little edit-box |
| 23:43.22 | etfonhomey | [TK]D-Fender, on an incoming call to an FXO port, is it best to use the Answer() application in the dialplan before using the Dial() to ring the receptionist phone? (http://pastebin.ca/1799181) |
| 23:45.37 | beta2k | [TK]D-Fender: Ok, I see disallow and allow text boxs under device options for the extension and I have them empty |
| 23:45.39 | [TK]D-Fender | etfonhomey: Depends. if you line will ring indefinately, then not answering will save on toll charges as you haven't answered it yet and theeey might give up waiting. |
| 23:45.55 | [TK]D-Fender | etfonhomey: Or it may fail out the the telco will take over before you get a chance. |
| 23:46.04 | [TK]D-Fender | etfonhomey: Generally I like KEEPING my calls... |
| 23:46.06 | beta2k | I just tried setting them to disallow all and allow g729, still it's ringing in ulaw |
| 23:46.22 | [TK]D-Fender | beta2k: and I unsurprisingly see nothing <- |
| 23:46.24 | [TK]D-Fender | ~pb |
| 23:46.25 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 23:46.27 | *** join/#asterisk Faithful (~Faithful@ns.linuxterminal.com) |
| 23:46.27 | [TK]D-Fender | ^^^^^ |
| 23:48.27 | *** part/#asterisk korihor (~korihor@190.205.251.97) |
| 23:49.36 | jaytee | http://buttersafe.com/2009/02/24/head-situations/ |
| 23:50.51 | beta2k | [TK]D-Fender: Sorry, but what do you want me to paste? I'm looking at a web page, not a config file :) |
| 23:51.59 | *** join/#asterisk came0 (~came0@rrcs-71-42-53-182.se.biz.rr.com) |
| 23:53.28 | [TK]D-Fender | beta2k: How about the CALL with SIP DEBUG so we can see who it's matching, and what is being negociated. |
| 23:53.55 | beta2k | NP :) |
| 23:54.04 | etfonhomey | [TK]D-Fender, have you ever seen this warning message: [Feb 16 18:21:22] WARNING[23437]: pbx.c:2988 pbx_substitute_variables_helper_full: Error in extension logic (missing '}') |
| 23:55.25 | *** join/#asterisk corretico (~laguilar@201.201.46.106) |
| 23:56.14 | etfonhomey | [TK]D-Fender, I get it after an incoming DAHDI call hangs up (whether it was answered or not). Here's the full log of a call to the FXO port where I hang up before any of the SIP endpoints answer: http://pastebin.ca/1799195 |
| 23:58.29 | etfonhomey | extensions.conf is at the bottom |
