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00:27.01 | Micc | can the sip jitter buffer help faxes over g711 and credit card machines? |
00:29.08 | pabelanger | no, but more bandwidth might help |
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01:22.03 | pabelanger | Anybody have experience with Supermicro boxes? Looking for feedback |
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01:42.08 | Tech_Travis | pabelanger: I use a SM mb in my server and it's great for the last year. |
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02:13.21 | nickzxcv | hi, so when I run fxotune -i I see /dev/dahdi/4 absent: Device busy when it comes to my fxo interface |
02:13.36 | nickzxcv | but when I run lsof | grep dahdi it doesn't show any processes have it open |
02:17.58 | nickzxcv | i'm running fxotune as root, is there anything else i should look at to see why its busy or how fxotune can lock it? |
02:18.30 | nickzxcv | is there a lock file or something? |
02:18.49 | shido6 | odd whats lsmod say? |
02:19.34 | shido6 | im getting rusty now that I use crisco gateways instead of dahdi hardwar |
02:19.36 | shido6 | e |
02:20.39 | nickzxcv | this is on freebsd, http://pastebin.com/L9uFAbb3 |
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02:23.09 | shido6 | can u stop asterisk and reload dahdi modules? |
02:27.17 | nickzxcv | asterisk isn't running when i run fxotune, i just reloaded the modules |
02:27.59 | nickzxcv | before running dahdi_cfg it said /dev/dahdi/4 absent: Device not configured and then after it said /dev/dahdi/4 absent: Device busy |
02:28.43 | nickzxcv | this is what dahdi_scan says http://pastebin.com/syb6T9pW |
02:28.54 | nickzxcv | the fxs doesn't work because i didn't have a power plug for it in this system |
02:29.17 | nickzxcv | but i only want the fxo to work right now anyway, could the fxs still be a problem? |
02:30.04 | nickzxcv | this is the dahdi/system.conf http://pastebin.com/AvtuPfBB |
02:30.12 | nickzxcv | is there anything else i should set there for this? |
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02:36.49 | shido6 | is the fxo card plugged in? |
02:36.58 | shido6 | and powered? |
02:37.06 | nickzxcv | to the phone line you mean? or in the system? |
02:37.20 | shido6 | the hardware |
02:39.12 | nickzxcv | yes, it is (its in a colo now so i can't look at it but it was the last time i was there) |
02:39.32 | nickzxcv | and dahdi_scan detects it as i would expect |
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03:04.02 | sawgood | If I have voicemail set to go out via email (as a .wav file) ... should I be able to 'see' this process happening from the CLI (to confirm if VM to email is working)? |
03:04.58 | sawgood | I know the * box can send email (because I do that from outside of * using Mutt or mail) |
03:09.09 | ChannelZ | no it doesn't really say anything |
03:09.39 | sawgood | I'll check the mail logs then |
03:09.41 | sawgood | thanks |
03:11.45 | sawgood | should voicemail.conf have anything special in it to make this happen (other then what I've read on the Internet to send voicemails to emails) |
03:12.49 | ChannelZ | I guess that depends on what you've read |
03:12.56 | sawgood | exactly |
03:13.11 | ChannelZ | but 'mailcmd' is important, and the mailbox needs to be setup properly with the email address argument |
03:14.15 | sawgood | by 'mailcmd' do you mean from the CLI or just configuing mail in general on a Linux box? |
03:14.32 | ChannelZ | mailcmd in voicemail.conf |
03:14.40 | sawgood | got it .. looking now |
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03:41.26 | sawgood | tail -f /var/log/maillog shows this entry for every email trying to leave the box as a voicemail to email |
03:41.28 | sawgood | <PROTECTED> |
03:41.47 | sawgood | problem is ... I can send emails fine from the command line (using mutt or mail) |
03:41.59 | sawgood | sendmail is resolving hostnames just fine |
03:42.19 | sawgood | I'm not sure what Asterisk is doing different vs sending emails from the CLI in Linux |
03:43.39 | ChannelZ | they might just be calling 'mail' and not sendmail, who knows |
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05:03.43 | BeeBuu | how can i log all the call ,even some call were NOANSWER? |
05:04.14 | ChannelZ | well CDR should log that already I thought |
05:04.44 | BeeBuu | ChannelZ: i don't think so |
05:06.00 | BeeBuu | ChannelZ: i make cdr & sip and queue in realtime by mysql,but i found if the queue no answer,no log in CDR |
05:07.49 | ChannelZ | maybe that's a failing of realtime then, I'm seeing NO ANSWER's in my csv log |
05:07.58 | Kyosh | beebuu: cdr does it all for realtime |
05:08.08 | Kyosh | i have it all in my cdr's |
05:08.32 | Kyosh | answer, busy, congestion, no answer, failed |
05:08.35 | ChannelZ | missing field then? or something in the dialplan that cause calls not to ever end that way.. |
05:08.59 | BeeBuu | Kyoush:o,really,so why it doesn't happen to my? |
05:09.16 | Kyosh | does your cdr's record any records? |
05:10.16 | BeeBuu | Kyosh: yes,it only record the ANSWERed call. |
05:10.24 | Kyosh | only answered? |
05:10.47 | BeeBuu | yes,so bad for me. |
05:11.13 | Kyosh | is your mysql table set up correctly? |
05:11.45 | BeeBuu | if it's incorrect,how can it record the answered call? |
05:14.02 | Kyosh | dont ask, just check |
05:14.17 | Kyosh | all columns must be entered in the order required, not random order |
05:14.47 | BeeBuu | yes,i follow the article in voip-info.org |
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05:15.19 | Kyosh | http://www.voip-info.org/wiki/view/Asterisk+RealTime+PostgreSQL |
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05:17.59 | BeeBuu | Kyosh:thanks,i had done like that. |
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05:54.54 | Get_The_Fish | anyone else here testing 1.8? |
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06:02.14 | schmidts | good morning all |
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06:31.25 | d-b | Any recommendations for a an au -> voip connection and a eu voip i> au ? |
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06:51.58 | AndyRomano | any ideas? - got an analog telephone calling our e1-line - and all digits after the first get cutoff. if i call from a digital phone or mobile - all is working as expected - any ideas? |
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06:53.14 | ectospasm | AndyRomano: try relaxdtmf in chan_dahdi.conf |
06:53.56 | ectospasm | unfortunately that may allow the analog phones to work, but may break the voip and mobile phones |
06:55.48 | AndyRomano | k will try now ty |
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06:58.23 | AndyRomano | changing relaxdtmf to yes doesn't change anything :( |
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07:00.41 | ectospasm | is there any difference in DTMF debug between the successful and unsuccessful DTMF navigation input? |
07:01.13 | AndyRomano | didn't try to do so till now |
07:01.15 | ectospasm | AndyRomano: edit logger.conf, append "dtmf" to the "console" line |
07:03.19 | ectospasm | I need to go to bed, it's too late for me to be up. |
07:03.49 | AndyRomano | ty 4 your help and gn8 |
07:04.21 | AndyRomano | the extension is not dialed with dtmf - it is appended to the number direcly |
07:05.03 | AndyRomano | when i call the extension 40200 i get a "extenstion '4' in context â¦â¦ doesn't exist |
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07:21.54 | jkroon | hi guys, dahdiras docs indicates that pppd needs to be patched to be dahdi aware - but I'm unable to locate information regarding exactly what this means - any ideas? |
07:23.26 | TobSnyder | someone here who can give me some hints concerning configuration of isdn telephone sets |
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07:27.03 | DND | guys is there any way i can prevent the 503 error that unregisters my soft phone when i received a busy tone? |
07:34.04 | jkroon | 503 != busy. |
07:34.10 | jkroon | 480 == busy. |
07:34.26 | jkroon | check your logs to figure out what causes the 503 to be sent to begin with. |
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07:41.22 | DND | jkroon: based on one website: The call is going to be automatically disconnected. Try to call later. |
07:41.32 | DND | wil lcheck on logs |
07:42.49 | jkroon | yes, 5XX => internal type errors. you need to figure out what causes the error. My SIP ref indicates that 503 implies Service Unavailable. |
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07:53.07 | Diffen2 | Morning. Do any of you guys know an application that can show extension status if a call has been forwared using cfa or followme to a cellphone, that the extension show busy? I know that the extension are not busy when there are a call scenario like that but somehow broadsoft manage to get this working and i belive asterisk could do that too. |
07:54.28 | DND | jkroon: here's part of the log file: http://pastebin.ca/1936373 |
07:56.52 | jkroon | DND, that is a long log, and a very quick scan doesn't show anything out of the ordinary except i'm wary about those last few lines re chan_dahdi. you're getting this on SIP though? |
07:56.59 | jkroon | perhaps enable sip debug for that peer? |
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07:57.27 | jkroon | Diffen2, you could use call groups ... |
07:58.04 | Diffen2 | jkroon hmmm how do you mean? im quite new to asterisk |
07:58.56 | DND | jkroon: that's just one call i believe. but one line there is this:"Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 3") in new stack" |
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08:13.28 | TobSnyder | no users with ISDN telephone sets here? |
08:15.51 | TobSnyder | <PROTECTED> |
08:15.51 | TobSnyder | <PROTECTED> |
08:18.03 | jkroon | DND, google for "isdn cause codes", first hit, look for code 3. |
08:19.56 | jkroon | 3 => no route to destination. |
08:20.04 | jkroon | B number is most likely incorrectly formatted. |
08:21.22 | jkroon | Called g0/048004931 <= not familiar with your area, but that's 9 digits - i would expect at least 10, but if that was the problem I'd also expect ISDN cause code 28. |
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08:48.31 | DND | jkroon: how can i change cause 3 to cause 17? because it seems this is the one i need http://networking.ringofsaturn.com/RemoteAccess/isdncausecodes.php |
08:49.13 | jkroon | DND - are you sure the user is busy? |
08:49.27 | jkroon | then you need to fix the equipment that is sending the cause code of 3, not asterisk. |
08:50.00 | DND | the user complains that after ringing, it gives out busy tone if no one answered. which is correct. but after the busy tone, it unregisters the soft phone |
08:50.17 | DND | and gives 503 |
08:50.44 | jkroon | I would first enable sip debug for that exten, reproduce and investigate the actual SIP traffic before drawing conclusions. |
08:50.47 | DND | so the user have to wait a couple of seconds to have the softphone re-register |
08:51.22 | DND | a lot of us are using different kinds of softphone but most of them encountered the same problem |
08:51.45 | DND | how can i debug the extension? |
08:54.24 | jkroon | sip set debug peer ???? |
08:55.06 | BarthezZ | hmm, can asterisk hold multiple registrations on 1 peer? |
08:56.00 | BarthezZ | I'm having a fun challenge, provider suggests a $ 4 figures solution.. while it should be able with about 600 |
08:56.22 | DND | jkroon: ok got the problem basede on the log, he called 048004931. i found out that its a n 800 number which doesnt require area code. |
08:56.41 | jkroon | BarthezZ, no. |
08:56.48 | DND | so when he dialed 048004931, it gives hangup cause 3 |
08:56.54 | jkroon | however, i typically cheat my way around that. |
08:57.02 | BarthezZ | as clear as you can be jkroon |
08:57.18 | BarthezZ | yeah well i'm currently in college so didn't had the time to check :p |
08:57.32 | BarthezZ | was thinking about the workaround already |
08:57.36 | DND | then unregisters the phone |
08:58.05 | jkroon | BarthezZ, create 4 peers with the same name -1 ... -4 and then when it becomes time to dial you select one that doesn't have a call on it yet. |
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08:58.26 | BarthezZ | well, the situation is as follows: |
08:58.39 | DND | is there a way that if the user enters a wrong number or maybe generates cause 3, can it just hang up not un-register? |
08:58.44 | jkroon | not rocket science, but not trivial either. not just anybody will be able to do it. |
08:59.02 | jkroon | You can send a manual Hangup() with a different cause code. |
08:59.20 | DND | in extension.conf? |
08:59.47 | jkroon | jip |
08:59.48 | BarthezZ | I have 2 buildings, which require the same IP-DECT solution. But I can buy a 4 figured REALLY expensive multi cell solution, or 2 cheap wireless servers and assign them both handsets and "virtually" have the same extensions |
08:59.59 | BarthezZ | only handover etc won't be possible |
09:00.02 | jkroon | or just IAX/2 |
09:00.07 | jkroon | with trunking. |
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09:00.36 | BarthezZ | brb |
09:00.36 | tgreer | morning all |
09:00.54 | jkroon | now you're starting to get the idea ... different accounts, same "number", ie, exten => ??,n,Dial(SIP/user-1&SIP/user-2) type of thing :p |
09:01.57 | AndyRomano | bist eh nicht über das taucher-sackerl geflogen im keller? |
09:02.14 | AndyRomano | sry wrong window ;) |
09:03.33 | _schmidts | lol :D |
09:10.55 | TobSnyder | test |
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09:11.51 | ruben23 | guys any idea on this error log during installation----> http://pastebin.com/tTFNig3T |
09:13.19 | kaldemar | ruben23: "You do not appear to have the sources for the 2.6.32-24-server kernel installed." |
09:13.21 | AndyRomano | kernel-headers missing? |
09:13.57 | tgreer | looks like it |
09:14.06 | tgreer | and kernel-devel? |
09:14.37 | ruben23 | kaldemar: -----> http://pastebin.com/F3XBtND9 |
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09:21.40 | kaldemar | ruben23: what does your /lib/modules/2.6.32-24-server/build point to? |
09:23.22 | kaldemar | ruben23: i.e. "readlink /lib/modules/2.6.32-24-server/build" |
09:25.00 | ruben23 | kaldemar:where is lib modules directory..? |
09:26.09 | kaldemar | ruben23: that is the full path. / in the beginning means it's in root dir. |
09:26.31 | kaldemar | ruben23: readlink is a command that does the job for you. |
09:26.35 | ruben23 | kaldemar: im here now how do i check where did this point to. |
09:27.06 | ruben23 | build -> /usr/src/linux-headers-2.6.32-24-server |
09:27.19 | kaldemar | ruben23: does that directory exist? |
09:28.01 | ruben23 | it does not exist on /usr/src/ |
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09:28.58 | ruben23 | what could ido.. |
09:29.01 | kaldemar | ruben23: does /usr/src/linux-headers-2.6.32-24 exist? |
09:29.15 | ruben23 | it does not exist |
09:30.00 | Azrael808 | Hi guys, I'm looking at purchasing some additional phones for our Asterisk server. Our regular supplier doesn't have the best collection of phones, but two in particular caught my eye. |
09:30.23 | kaldemar | ruben23: apt-get install --reinstall linux-headers-2.6.32-24-server |
09:31.53 | Azrael808 | The Cisco SPA504G |
09:32.01 | ruben23 | kaldemar: i got it now,what do i do next. |
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09:32.36 | Azrael808 | and a Draytek VPH350 |
09:32.52 | Azrael808 | Does anyone have any experience with these two devices? should I avoid them like the plague? |
09:33.25 | kaldemar | ruben23: back to the "make all" step |
09:34.58 | ruben23 | i still got the error--> http://pastebin.com/YkjtdUuK |
09:35.08 | skyion | SIP 180 ringing is handled in signalling, where is the responsibility to actually generate the tone, is it up to the handset or else in the event of a PRI card is it up to the card? |
09:35.51 | jkroon | 180 is handled by the handset, 183 means premature media (handled by whichever device decided to use 183 instead of 180. |
09:36.24 | jkroon | also, if channel is already up you will generate tones locally (often), in which case it's most likely up to the indications subsytem to generate the tones. |
09:38.58 | kaldemar | ruben23: does /usr/src/linux-headers-2.6.32-24-server exist? |
09:40.52 | ruben23 | yes iy did exist now |
09:41.51 | kaldemar | yet make says "no such file or directory" for /usr/src/linux-headers-2.6.32-24-server |
09:41.54 | kaldemar | interesting. |
09:42.15 | ruben23 | <PROTECTED> |
09:45.24 | kaldemar | ruben23: pastebin output for "ls -ld /usr/src/linux*" |
09:46.40 | ruben23 | http://pastebin.com/6s4GAq3k |
09:48.51 | kaldemar | ruben23: apt-get install --reinstall linux-headers-2.6.32-24 |
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09:49.30 | kaldemar | ruben23: if you install something with apt-get, don't remove it by hand. |
09:50.49 | ruben23 | ok |
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09:52.43 | skyion | in the event of a SIP 180 response reaching an asterisk system and being destined for a BRI card, what would happen? |
09:56.38 | ruben23 | kaldemar: thanks you so mcuh |
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10:05.12 | Azrael808 | If no one can confirm or deny that those two phones are any good, does anyone have any particular handset recommendations? |
10:09.18 | _schmidts | @azrael we use the cisco spa504 |
10:09.36 | Azrael808 | Oh, cool |
10:10.07 | _schmidts | atleast we have around 2k of the spa941 out at our customers and the 504 is the new type of this phone |
10:10.35 | Azrael808 | Excellent, the phone on offer by our supplier is the 504G... I assume this is essentially the same thing. |
10:10.59 | _schmidts | yes, the old spa941/942/962 are EOL by now |
10:11.26 | AndyRomano | good to know ;) |
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10:12.34 | _schmidts | but whats real nice about the 504 you could use the side modul to use blf |
10:12.50 | Azrael808 | blf? |
10:13.14 | _schmidts | Busy Lamp Field to see the activity of other extensions if they are talking, ringing or idle |
10:13.20 | _schmidts | and also do a pickup on a ringing phone |
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10:15.45 | Azrael808 | Cool! That's pretty useful |
10:16.03 | hrhrhr | that may not work out of the box, however |
10:16.33 | Azrael808 | Hey, baby steps... I'll be happy just to get some additional handsets for users |
10:16.39 | Azrael808 | We can look at fancy features later! |
10:16.41 | Azrael808 | :) |
10:16.44 | hrhrhr | :D |
10:17.56 | _schmidts | what do you mean? the blf would work, but pickup is a little work to do ;) |
10:19.20 | hrhrhr | indeed |
10:19.34 | hrhrhr | blf with pickup worked out of the box with my test handsets |
10:19.39 | hrhrhr | when i went and bought snoms |
10:19.42 | hrhrhr | it just refused to work |
10:19.58 | hrhrhr | that was about 2 years ago tho |
10:20.02 | hrhrhr | maybe it's easier now |
10:20.31 | _schmidts | ;) |
10:20.33 | Azrael808 | Thanks so much for your advice guys. |
10:20.52 | Azrael808 | I'm off to spend some company money now! :) |
10:22.24 | ruben23 | hi guys i been installing this version but first time to see this ---> http://pastebin.com/RZWy6DEh |
10:24.03 | kaldemar | ruben23: did you have a previous install of asterisk on the machine? if so, you probably didn't do a make uninstall for it before installing the new one. |
10:24.57 | ruben23 | now this is the problem all are manually remove.. |
10:25.11 | ruben23 | :-( |
10:26.42 | ruben23 | kaldemar: do i got options for it..? to resolve this. |
10:26.44 | *** join/#asterisk BANSAL (~bansal@117.207.84.86) |
10:28.53 | kaldemar | ruben23: "rm /usr/lib/asterisk/modules/* && make install" |
10:34.23 | ruben23 | http://pastebin.com/GHQaR3wS |
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11:17.46 | hay | hi all... I would like to have asterisk running on FreeBSD and on an ISDN line (1 BRI)... what PCI ISDN cards are tested and working in this combination... of course, as cheap as possible :-) thx |
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11:56.08 | hrhrhr | hay: take a look at the b410p. dunno about fbsd support tho |
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12:31.01 | geemee | I made an assumption that creating a conference room would be the same almost as an extension, however I can dial the confernce room number from an IVR by dialing the "extension" of the conference room. Is this behaviour correct? |
12:31.18 | geemee | I cant dial the conference room sorry - typo |
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12:33.53 | TobSnyder | where can I change default numbers like 411 for phone and so on? |
12:34.43 | drmessano | default numbers? |
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12:37.46 | TobSnyder | feature codes |
12:38.13 | TobSnyder | *98 voicemail |
12:38.17 | TobSnyder | 411 phonebook |
12:38.20 | TobSnyder | and so on |
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12:43.26 | geemee | Different wording to my question: Hi there, When creating conferneces can these be direct dialed in IVRs? Example make a conference room 250 and have extension 230 and 240. At the IVR someone could dial extension 230 or dial 250 to get the conference room? |
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12:50.53 | hay | hrhrhr: thanks... I am sure that there are some fbsd + asterisk + isdn users here and if anyone could recommend some 1 BRI ISDN cards, that would be excellent |
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12:53.47 | TobSnyder | found it |
12:54.40 | TobSnyder | junghanns |
12:54.40 | *** part/#asterisk AndyRomano (~Adium@mail-gw.helwacht.at) |
12:54.54 | TobSnyder | seem to be supported very well ? |
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13:05.08 | _schmidts | does anyone know a way to send AOC infos via sip? |
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13:08.35 | _schmidts | i have found sirrix but it looks like this is only able to send it via pri |
13:09.24 | _schmidts | patton told me they support AOC via sip in a application/QSIG body |
13:11.59 | *** join/#asterisk n0tk (~n0tk@216.160.42.30) |
13:12.44 | _schmidts | oh its allready in 1.8 itself |
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13:16.04 | *** join/#asterisk t_dot_zilla (~chatzilla@firewall-a.buf.ny.i-evolve.net) |
13:16.27 | Katty | great maker, i am exhausted |
13:17.12 | t_dot_zilla | somehow we've been hacked, the hacker is calling a DID -> extension -> to outbound international call |
13:17.17 | t_dot_zilla | how can this happen? |
13:17.18 | *** join/#asterisk LordZ (~lordz@217.12.113.114) |
13:17.56 | upb | lol |
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13:19.20 | t_dot_zilla | has anyone heard of this happening? |
13:19.42 | ectospasm | t_dot_zilla: that's usually because you allow folks dialing into the IVR to get to an outside line |
13:19.57 | ectospasm | like including from-internal in from-pstn in extensions.* |
13:20.03 | t_dot_zilla | no is no IVR though, DID goes to extension |
13:20.31 | t_dot_zilla | once the extension is reached, that extension makes the outbound call |
13:20.39 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
13:20.49 | _schmidts | 302 redirect from this extension? |
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13:20.55 | ectospasm | t_dot_zilla: I'd have to see the CLI trace of the call |
13:20.55 | Katty | hi fender. |
13:20.57 | Katty | bender. |
13:21.02 | krion | anyone with a link on how to write extension ? |
13:21.04 | Katty | hugs fender bender |
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13:21.25 | beek | hugs Katty |
13:21.34 | Katty | :> |
13:21.36 | Katty | hugs beek |
13:21.42 | beek | How are you today katty? |
13:21.48 | [TK]D-Fender | krion: http://dictionary.reference.com/ |
13:21.54 | [TK]D-Fender | krion: I'm sure they'd know |
13:22.05 | beek | notes [TK]D-Fender is in great form today. |
13:22.08 | Katty | beek: exhausted, actually :< two nights in a row up till midnight. |
13:22.15 | ectospasm | bad, [TK]D-Fender |
13:22.16 | krion | sorry if you don't understand my frenchglish |
13:22.24 | beek | Katty: Drinking and carousing? |
13:22.28 | Katty | oooh it's ectospasm |
13:22.35 | Katty | beek: movie watching, actually |
13:22.54 | ectospasm | krion: http://astbook.asteriskdocs.org |
13:22.58 | *** join/#asterisk [Jasper] (~jverberk@82-171-125-58.ip.telfort.nl) |
13:23.09 | [TK]D-Fender | krion: Then you might be better off picking one language :) |
13:23.10 | krion | i mean, i use stuff like exten => 5,1,Goto(ivr_stuff,s,1) |
13:23.16 | [Jasper] | hej guys, is there anyone here who knows why status of my sip peer shows unreachable..but I can just call out |
13:23.25 | Katty | naps on beek's shoulder |
13:23.45 | krion | but i'm not sure on what's every row comma separated are meaning |
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13:24.01 | beek | Katty: we'd be snoozing on each others' shoulders. I'm a bit sleep deprived myself. |
13:24.23 | Katty | s'all good |
13:24.34 | Katty | this vault is helping, but caffeine is a lie |
13:24.44 | krion | ectospasm: thanks, hope i quickly find what i'm looking for |
13:25.19 | beek | I'm working on some serious french-press, freshly roasted and ground French Roast. Mmmmmmmmm. |
13:25.19 | [TK]D-Fender | krion: "core show application goto" <---------- |
13:25.21 | beardy | Have some cake. |
13:25.49 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
13:25.52 | Katty | it's too early for cake |
13:26.02 | Katty | unless it is cake from the matrix. |
13:26.02 | beardy | Lies! |
13:26.15 | [Jasper] | how can i see which sip peers are actually connected |
13:26.29 | [TK]D-Fender | krion: Go read your app's INSTRUCTIONS |
13:26.46 | [Jasper] | my phone is connected to asterisk..but in asterisks it reports unmonitored after I removed qualify= yes and other wise it shows unreachable... |
13:26.49 | [Jasper] | but I know the phone is connected since I can call out using the asterisk box |
13:26.50 | beardy | Katty: Well it is I guess. |
13:27.17 | Katty | beardy: do sweets make you sick on an empty stomach? |
13:27.26 | [TK]D-Fender | [Jasper]: You have a networking issue if qualify times out and makes them unreachable |
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13:29.07 | beardy | Katty: Hmm, no, I don't think so. Being very hungry can though I guess. How about you? |
13:29.49 | Katty | beardy: it induces teh ill |
13:30.04 | ectospasm | [Jasper]: is it a user, peer, or friend? |
13:30.12 | beardy | Katty: Coffee on an empty stomach then? :) |
13:30.22 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
13:30.41 | Katty | beardy: soda, atm...but foodables soon. |
13:31.26 | beardy | Katty: Afternoon coffee in around an hour here probably. |
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13:31.46 | [Jasper] | hmm ectospasm type=friend |
13:31.48 | Katty | afternoon coffee? |
13:31.57 | Katty | you mean tea? |
13:32.26 | beardy | Katty: No.. coffee :) |
13:32.27 | Katty | beardy: uk? |
13:32.35 | beardy | Katty: .se |
13:32.37 | ectospasm | [Jasper]: [TK]D-Fender was right then, it's either not registering (and you're allowing insecure calls), or there's a network issue |
13:32.57 | Katty | beardy: ohah, k'then |
13:33.26 | Katty | swedish accent is omnomnomnom. |
13:33.27 | [TK]D-Fender | [Jasper]: You should be looking at SIP DEBUG to see what is happening |
13:33.45 | beardy | Hehe |
13:33.56 | *** join/#asterisk theHub (~karl@69.177.93.21) |
13:33.56 | [Jasper] | hhmm let kme check with my laptop if it says unregistered too |
13:34.19 | Katty | what is the significance of Afternoon Tea |
13:34.20 | *** join/#asterisk adyn (~adyn@unaffiliated/adyn) |
13:34.40 | Katty | is it basically snack time? |
13:34.46 | [Jasper] | hmm [TK]D-Fender my laptop also fails |
13:34.52 | [Jasper] | what could be wron with the network then? |
13:34.56 | [Jasper] | nat? |
13:35.16 | [TK]D-Fender | [Jasper]: You tell us. Because... you haven't actually told us ANYTHING yet. |
13:35.45 | beardy | Katty: 5PM, or 17:00 in sane time.. late snack time I guess? |
13:35.50 | *** join/#asterisk mikkel (~mikkel@84-238-113-66.u.parknet.dk) |
13:36.13 | Katty | that sounds like dinner. |
13:36.58 | beardy | Katty: (I don't always have coffee this time, but used to it from "home", so.. sometimes. at 16:15 ) |
13:37.00 | Katty | beardy: breakfast, lunch, tea? |
13:37.40 | Katty | beardy: breakfast, second breakfast, elevenses, lunch, afternoon tea, dinner, supper |
13:37.40 | beardy | Katty: Yeah.. plus evening meal. |
13:37.59 | beardy | Katty: Haha, yes, more like that from where I come ;) |
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13:38.23 | Katty | beardy: what are the 3 typical meals you guys have called? |
13:38.57 | beardy | Katty: frukost, middag(lunch), kvällsmat |
13:39.22 | [Jasper] | [TK]D-Fender I have 2 phones installed and 1 phone number |
13:39.26 | Katty | beardy: and how do you pronounce those in english? |
13:39.36 | [Jasper] | 1 phone works and the telephone number works |
13:39.57 | beardy | Katty: middag is dinner, but literally means "mid day" so.. many call "supper" middag. |
13:40.22 | [Jasper] | but the other phone shows unreachable...the only thing I can see is that it's using a weird port for that phone: 21984 |
13:40.23 | beardy | Katty: You mean phonetically, or translated? |
13:40.28 | [Jasper] | while the rest is on 5060 |
13:40.35 | Katty | beardy: yes |
13:40.56 | Katty | beardy: phonetically |
13:40.56 | beardy | Katty: Which? :) |
13:41.36 | [TK]D-Fender | Katty: You don't have enough phlegm for it ;) |
13:41.45 | Katty | [TK]D-Fender: you hush. |
13:42.08 | beardy | Katty: frewcost, middaaag, qwellsmaaat, sort of ;) |
13:42.23 | [TK]D-Fender | [Jasper]: that description is useless. what KIND of "phone" and EXACTLY how are they networked in relation to * <--------- |
13:42.27 | Katty | beardy: the aaa as in cat? |
13:42.55 | beardy | Katty: More like the a in aww, or awful |
13:43.14 | [Jasper] | it's hard to explain [TK]D-Fender ...I got 1 working phone which is in my other house...that's connected via the modem being a p-2602-r-d1a modem....so basically sip on the modem connects it.. |
13:43.32 | Katty | beardy: you should post an audio recording |
13:43.39 | [Jasper] | at this location I'm BEHIND a modem supporting sip....but I'm not using the native sip of the modem...I'm trying to conect my laptop via port 5060 to the server |
13:43.49 | [Jasper] | my laptop or my mobile phone.. |
13:43.51 | Katty | beardy: and talk slowly, not like a german |
13:44.04 | [Jasper] | port 5060 is open...since I can connect to the server...could it be that the modem blocks traffic on port 5060 |
13:44.07 | Katty | and definately not like junky |
13:44.10 | [Jasper] | incoming traffic? |
13:44.15 | Katty | dear lord that man is hard to understand. |
13:44.38 | [TK]D-Fender | [09:43]<[Jasper]>at this location I'm BEHIND a modem supporting sip....but I'm not using the native sip of the modem...I'm trying to conect my laptop via port 5060 to the server <- this modem could screw you over |
13:44.49 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
13:44.49 | *** mode/#asterisk [+o putnopvut] by ChanServ |
13:44.55 | krion | what i want to do is playback something, if timeout, playback again, maximum 3 times |
13:45.03 | beardy | Katty: I might. Or when we talk sometime. |
13:45.03 | [Jasper] | i can try another port [TK]D-Fender |
13:45.13 | Katty | [TK]D-Fender: i think i'm going to learn russian after i get the french down a bit better |
13:45.23 | krion | could i use something like $i++ |
13:45.23 | Katty | beardy: mkay |
13:45.49 | [TK]D-Fender | krion: No. * != C |
13:45.55 | [Jasper] | [TK]D-Fender I used utorrent to check if the port was actually open |
13:46.03 | [Jasper] | and it reports it's open port 5060 on my laptop |
13:46.09 | [TK]D-Fender | krion: GotoIf, Set, etc, these are the tools you'll be using |
13:46.13 | Katty | [TK]D-Fender: tho perhaps spanish might be a better idea considering my location |
13:46.13 | [Jasper] | so I don't think the modem is messing with me |
13:46.19 | [Jasper] | could it be something related to NAT? |
13:46.41 | [TK]D-Fender | [Jasper]: Yes, what did you do to set * up to work from behind it? |
13:47.14 | [Jasper] | what do you mean with * ? |
13:47.32 | [Jasper] | the asterisk box itself is in a datacenter and has a dedicated ip to the internet...so no nat there |
13:47.36 | beardy | Asterisk |
13:48.00 | [Jasper] | the other phone connects via the modem..so no nat there either...but HERE I'm behind the modem...so only location where I use NAT I guess |
13:48.27 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
13:49.29 | Katty | ohai Chainsaw |
13:49.31 | [TK]D-Fender | [Jasper]: your peer setup should be "nat=yes", and "qualify=yes". You shuold not need forwarding on your home side. |
13:49.36 | Chainsaw | Hello Katty :) |
13:50.02 | [Jasper] | [TK]D-Fender hmm I see...that's why it uses the weird port also because of the NAT |
13:50.06 | [TK]D-Fender | [Jasper]: And you may have to change the SIP port and specify "port=XXXX" for what you use to bypass the router. |
13:50.15 | [TK]D-Fender | [Jasper]: Which you might NOT be able to do. |
13:50.37 | [Jasper] | yea hI can change the port |
13:50.43 | [Jasper] | but I wonder how this is gonna solve the problem.. |
13:50.59 | [TK]D-Fender | [09:45]<[Jasper]>and it reports it's open port 5060 on my laptop <-- make it NOT use this port |
13:52.03 | [Jasper] | but [TK]D-Fender with NAT on the client side...asterisks actually uses other ports to communicate with the client then 5060 right? |
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13:52.45 | [TK]D-Fender | [Jasper]: * uses what it is told to use. |
13:52.54 | [TK]D-Fender | [Jasper]: Run your sip client on a DIFFERENT PORT |
13:53.04 | [TK]D-Fender | [Jasper]: And inform * |
13:53.13 | [Jasper] | can I do this per sip peer? |
13:53.15 | [Jasper] | the port? |
13:53.25 | [TK]D-Fender | [Jasper]: Yes |
13:53.27 | [Jasper] | guess I can |
13:53.28 | [Jasper] | :p |
13:53.28 | [Jasper] | ok |
13:53.38 | krion | [TK]D-Fender: ok thanks, i'll try something... |
13:53.48 | [Jasper] | so let's say ...port 3000 then |
13:54.28 | [TK]D-Fender | [Jasper]: I'd say 5070 |
13:56.07 | beardy | [Jasper]: 5061, 5062, and so on. |
13:56.51 | *** join/#asterisk frigidzephyr (~rnewton@nat/digium/x-njqdjxerfyawptwt) |
13:57.28 | *** part/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
13:59.29 | [Jasper] | not working [TK]D-Fender |
13:59.48 | [Jasper] | I get registered sip 'jasper' at ***.***.***.*** port 22907 |
13:59.53 | [Jasper] | saved useragent bladiebla |
14:00.05 | [Jasper] | peer 'jasper' is now UNREACHABLE! last qualify: 0 |
14:01.10 | [TK]D-Fender | [Jasper]: PASTEBIN the attempts with SIP DEBUG enabled |
14:01.27 | [TK]D-Fender | [Jasper]: "not working" doesn't help |
14:03.44 | *** join/#asterisk sekil (~sekil@81.18.48.208) |
14:05.19 | *** join/#asterisk speedlnx (9709b650@gateway/web/freenode/ip.151.9.182.80) |
14:05.23 | speedlnx | hello |
14:06.28 | speedlnx | can someone tell me which are the alternative for a CTI solution for windows and leopard that work with asterisk? |
14:07.19 | [TK]D-Fender | speedlnx: what is a "CTI solution"? You you be a little more vague? |
14:08.12 | [Jasper] | [TK]D-Fender http://www.pastebin.org/834523 |
14:09.18 | speedlnx | something to call from outlook or from a web page... |
14:09.24 | Katty | hmm. stomach is eating me alive from the inside >.< |
14:10.24 | n3hxs | speedlnx We don't know what "something" is... |
14:10.50 | Katty | n3hxs: ohai! |
14:11.05 | n3hxs | Yo Katty |
14:11.11 | Katty | n3hxs: let's hug |
14:11.13 | [TK]D-Fender | speedlnx: Anything that calls via SIP should work. |
14:11.16 | speedlnx | mmm.. a mean a software that communicate with asterisk |
14:11.36 | n3hxs | (H) |
14:12.03 | Katty | hugs n3hxs |
14:12.26 | speedlnx | ok, i will try to be more specific |
14:12.33 | n3hxs | Wait, your wrinkling my flowered Carribean shirt! |
14:12.50 | Katty | you can't wrinkle those shirts |
14:13.06 | n3hxs | Oh, yeah, you are right. |
14:13.16 | Katty | jeesh, boys. |
14:13.16 | n3hxs | Try again. |
14:13.25 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
14:13.51 | n3hxs | speedlnx, do you want to dial from a number you see on a webpage? |
14:14.05 | speedlnx | yes |
14:14.06 | n3hxs | is guessing here...;) |
14:14.08 | speedlnx | and from outlook |
14:14.16 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
14:14.39 | Katty | hi puzzled |
14:14.43 | n3hxs | http://www.google.com/search?q=dial+asterisk+from+Outlook&ie=utf-8&oe=utf-8&aq=t&client=firefox-a&rlz=1R1GGGL_en___US356 |
14:14.49 | [TK]D-Fender | speedlnx: http://www.google.ca/#hl=en&source=hp&q=ms+outlook+SIP+dialer&aq=f&aqi=&aql=&oq=&gs_rfai=&fp=2dcae7cea7739227 |
14:14.59 | n3hxs | LOL |
14:15.28 | [TK]D-Fender | JFGI <----- :p |
14:15.50 | *** join/#asterisk slipkid08 (~actioncom@rrcs-97-77-102-225.sw.biz.rr.com) |
14:16.09 | puzzled | hi Katty |
14:16.21 | slipkid08 | hello everyone, I am having an issue figuring out how to install this thing and get it working, can someone please help me |
14:16.51 | speedlnx | the first link seems to work only with 1.2 ver of asterisk |
14:19.24 | slipkid08 | it is so hard to get a response from 1 out of 249 people |
14:19.25 | slipkid08 | wtf |
14:19.50 | n3hxs | Only 3 are watching you slipkid08 |
14:20.01 | slipkid08 | watching me? |
14:20.04 | n3hxs | And some are working for a living. |
14:20.07 | n3hxs | CIA etc |
14:20.12 | slipkid08 | oh I see |
14:20.19 | Katty | FBI, personally. |
14:20.22 | Katty | i'm watching you. |
14:20.25 | slipkid08 | hehe |
14:20.34 | Katty | no really. |
14:20.35 | slipkid08 | it's ok I don't do anything illegal anyway |
14:20.41 | kaldemar | ~ask |
14:20.41 | 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. |
14:20.46 | speedlnx | ok, i will look on google for something like "outlook SIP dialer" |
14:20.48 | speedlnx | thank you |
14:20.58 | n3hxs | Also, since you are one of the 249 people you could talk with yourself. |
14:21.07 | slipkid08 | true that |
14:21.09 | slipkid08 | :P |
14:21.13 | n3hxs | LOL |
14:21.15 | slipkid08 | I just need some help is all |
14:21.22 | [Jasper] | [TK]D-Fender did you see my pastebin? |
14:21.37 | Corydon76-dig | slipkid08: did you run ./configure ? |
14:22.03 | Katty | hi Corydon76-dig |
14:22.37 | slipkid08 | oh |
14:22.41 | slipkid08 | so it's terminal |
14:22.50 | slipkid08 | it doesn't have an auto install |
14:23.00 | Katty | auto install? |
14:23.00 | kaldemar | slipkid08: what exactly are you installing? |
14:23.01 | Corydon76-dig | Not from source, it's not |
14:23.17 | slipkid08 | Ok I am kind of lost |
14:23.22 | Corydon76-dig | If you're installing from a package, there could be a GUI, though |
14:23.39 | slipkid08 | this is a VOIP system that you install on your server, and use with a SIP client right? |
14:23.41 | kaldemar | slipkid08: asterisk from source? asterisk from a package? asterisknow from a cd? |
14:23.47 | Corydon76-dig | But you need to consult with your distribution's help channels |
14:23.53 | slipkid08 | gah |
14:24.15 | Corydon76-dig | slipkid08: Servers don't generally run a GUI |
14:24.27 | slipkid08 | whatever lol |
14:24.33 | slipkid08 | I think I am in way over my head |
14:24.36 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
14:24.41 | slipkid08 | I know servers don't run a gui |
14:24.55 | slipkid08 | but I think I have no inclination of what is going on here |
14:25.13 | slipkid08 | so I'll get out with my butt still attached |
14:25.13 | Corydon76-dig | So how were you planning on getting Asterisk installed, if not from the command line? |
14:25.20 | slipkid08 | well, not really sure |
14:25.31 | slipkid08 | I was going to try and upload the files to my MAMP |
14:25.39 | Corydon76-dig | mamp? |
14:25.45 | slipkid08 | I am not sure how the configuration is supposed to go |
14:26.04 | slipkid08 | http://www.google.com/url?sa=t&source=web&cd=1&ved=0CBkQFjAA&url=http%3A%2F%2Fwww.mamp.info%2F&rct=j&q=mamp&ei=de6ITOS8AcOAlAeDs_nFDg&usg=AFQjCNE3Eeh78-JAsRygzNjtPVbj-miJkQ&sig2=XavsfQBBOQy_CJ-dLzABMw&cad=rja |
14:26.10 | kaldemar | ~thebook |
14:26.11 | infobot | rumour has it, thebook is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://astbook.asteriskdocs.org |
14:27.25 | Corydon76-dig | While you can run Asterisk on a Mac, I would not recommend it at this point. We still have a lot to work out with the Mac port |
14:28.08 | slipkid08 | ohhhh ok |
14:28.14 | slipkid08 | well let me ask you this |
14:28.15 | [TK]D-Fender | [Jasper]: that is NOT * SIP DEBUG. 'help sip" <- go get the syntax for enabling it on your version |
14:28.24 | Corydon76-dig | Linux and FreeBSD are your best bets at this point in time |
14:28.26 | slipkid08 | if I VMware it |
14:28.30 | slipkid08 | with ubuntu |
14:28.34 | slipkid08 | would that work |
14:28.56 | [TK]D-Fender | slipkid08: * wcan work on jsut about any *NIX |
14:29.18 | slipkid08 | ok...talk to me in laymans terms |
14:29.19 | Corydon76-dig | Generally, yes |
14:29.27 | slipkid08 | I am way new to this whole thing |
14:29.33 | [TK]D-Fender | slipkid08: YES <- |
14:29.35 | Corydon76-dig | Those WERE layman's terms |
14:29.37 | slipkid08 | doesn't mean I won't get it, just have to be patient |
14:29.48 | slipkid08 | I wasn't talking to you lol |
14:30.00 | slipkid08 | ok |
14:30.11 | slipkid08 | so I run ubuntu on vmware |
14:30.17 | [TK]D-Fender | sliYes, it can work |
14:30.24 | [TK]D-Fender | slipkid08: Yes, it can work |
14:30.43 | slipkid08 | when I get in there, download the pack, and then run ./configure on the package |
14:31.02 | slipkid08 | correct so far |
14:31.03 | slipkid08 | ? |
14:31.07 | [TK]D-Fender | slipkid08: Just "sudo apt-get asterisk" or download the tarball and read the instructions |
14:31.17 | slipkid08 | ok cool |
14:31.19 | [TK]D-Fender | slipkid08: "sudo apt-get install asterisk" |
14:31.21 | [TK]D-Fender | (rather) |
14:31.23 | slipkid08 | that should get me started |
14:31.46 | slipkid08 | I didn't realize there wasn't a mac port yet |
14:32.03 | *** join/#asterisk p3nguin (gpz5GvdFkf@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
14:32.22 | slipkid08 | now may I ask some general question? |
14:32.25 | [TK]D-Fender | slipkid08: Mac's are for kiddies, not servers :p |
14:32.30 | p3nguin | ~ask |
14:32.31 | 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. |
14:33.11 | [TK]D-Fender | slikAsterisk doesn't have a "Go" button and is therefor not conducive for use on Aplle tech. |
14:33.33 | slipkid08 | Is this a skype type of thing (i.e. calling only computer-computer) or can it call land-based lines too |
14:33.42 | [TK]D-Fender | slipkid08: No. |
14:33.47 | slipkid08 | haha fender.. |
14:34.01 | [TK]D-Fender | slipkid08: Call likes has nothing to do with the PROTOCOL (which Skype is an example of) |
14:34.06 | slipkid08 | trust me, I am not really a mac fan |
14:34.10 | [TK]D-Fender | Calling lines* |
14:34.12 | [TK]D-Fender | gah |
14:34.19 | [TK]D-Fender | Insufficiently caffienated this morning |
14:34.23 | slipkid08 | lol brb |
14:34.36 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
14:36.50 | Katty | The_Boy_Wonder: what is the secret of your power |
14:37.15 | The_Boy_Wonder | Katty: pancakes |
14:37.17 | tzanger | it's the tights he wears |
14:37.19 | n3hxs | White powder in a little bottle. |
14:37.35 | Katty | The_Boy_Wonder: pancakes++ |
14:37.36 | tzanger | there's a lot of power hiding in that tiny spandex |
14:37.45 | Katty | ooh la la. |
14:38.04 | slipkid08 | OK, so is this a Vonage type of system that only calls landlines? |
14:38.13 | n3hxs | Ewwww |
14:38.24 | slipkid08 | and/or mobiles |
14:38.37 | Katty | slipkid08: you can use pots lines, t1/pri, or sip providors |
14:38.48 | slipkid08 | hm |
14:38.51 | Katty | slipkid08: vonage is kinda craptastic, fyi |
14:39.04 | slipkid08 | I don't like vonage |
14:39.22 | slipkid08 | what I am asking is the system, once installed, can make phone calls anywhere in the world? |
14:39.31 | slipkid08 | for the cost of your host? |
14:39.40 | Katty | slipkid08: that probably depends on the terms of your telco |
14:39.53 | ectospasm | Katty: or ITSP |
14:40.02 | Katty | ^- what he said. |
14:40.13 | slipkid08 | ok |
14:40.41 | Katty | slipkid08: we currently have a pri, with unlimited long distance in the usa, canada, and mexico |
14:40.52 | slipkid08 | that's pretty cool |
14:40.53 | ectospasm | although I suppose the two terms are interchangeable, but I always think of telcos as being layer-1 providers, for traditional POTS/PSTN connections |
14:40.58 | Katty | slipkid08: places like norway, cost more |
14:40.59 | slipkid08 | but pretty far over my head, too |
14:41.09 | slipkid08 | wow, yeah |
14:41.16 | Katty | slipkid08: think of asterisk like an over glorified answering machine. |
14:41.26 | slipkid08 | I could just use onebox? |
14:41.29 | ectospasm | Katty: that's dumbing it down a lot! |
14:41.30 | slipkid08 | or google voice |
14:41.43 | Katty | ectospasm: i'm on the level, yo |
14:41.49 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
14:41.51 | ectospasm | Google Voice is essentially a hosted PBX |
14:41.53 | Katty | hi tony |
14:41.55 | slipkid08 | right |
14:41.55 | ectospasm | ...or cloud PBX |
14:42.03 | slipkid08 | I like that |
14:42.04 | slipkid08 | lol |
14:42.17 | ectospasm | slipkid08: no, it's simple, and free, and relatively no hassle |
14:42.30 | Katty | google vchat is nice. |
14:42.37 | slipkid08 | so, basically I would be getting the same thing with google voice as I would going with asterisk? |
14:42.38 | ectospasm | and the voicemail transcription is awesome |
14:42.49 | ectospasm | slipkid08: yes, but you'd have more control over Asterisk |
14:43.00 | slipkid08 | I think it's cool that you guys are building this, but why reinvent the wheel? |
14:43.08 | ectospasm | slipkid08: if you don't like something about Google Voice, and it doesn't provide you a way to workaround it, tough |
14:43.16 | slipkid08 | true |
14:43.24 | Katty | slipkid08: why? |
14:43.32 | [Jasper] | [TK]D-Fender http://www.pastebin.org/834911 |
14:43.32 | Katty | slipkid08: because when someone calls me, i want to rickroll them. |
14:43.34 | [Jasper] | thats sip debug |
14:43.39 | Katty | slipkid08: and eject the cdrom drive when i dial extension 42 |
14:43.40 | slipkid08 | haha |
14:43.41 | ectospasm | slipkid08: Asterisk was developed concurrently with Grand Central (which Google bought and renamed Google Voice) |
14:43.58 | ectospasm | ...or maybe even beforehand |
14:43.58 | slipkid08 | oh I see |
14:44.09 | chuckf | slipkid08: asterisk is not geared for the home user/single user audience (though there are many installs for that range) |
14:44.25 | Katty | chuckf: i use it at home. |
14:44.25 | slipkid08 | so, when will there be like a GUI, so that way someone such as myself doesn't have to go to such great lengths for this |
14:44.32 | ectospasm | slipkid08: and don't think that one solution is appropriate for all |
14:44.36 | Katty | chuckf: but i agree, generally speaking i find it better for small-med size business |
14:44.48 | slipkid08 | I know what a pbx is |
14:44.58 | slipkid08 | press one to go here press two to go there |
14:45.01 | chuckf | Katty: as do I, but most people don't have/want/need a pbx at home |
14:45.12 | slipkid08 | it would be geared towards business |
14:45.13 | Katty | chuckf: i enjoy rickrolling the telemarketers |
14:45.19 | slipkid08 | I was wanting it for my band |
14:45.20 | slipkid08 | lol; |
14:45.25 | ectospasm | slipkid08: the beauty of Asterisk is that it connects any kind of telephony technology to any other kind of technology |
14:45.31 | slipkid08 | so that way people could call one number |
14:45.40 | chuckf | Katty: that is a good use for it |
14:45.41 | slipkid08 | and get all 5 of us |
14:45.44 | slipkid08 | or our manager |
14:45.46 | Katty | chuckf: mhmm |
14:45.52 | slipkid08 | that's okay |
14:46.01 | slipkid08 | it was a thought but I don't have time to work around it |
14:46.08 | slipkid08 | I'll just buy one, I think |
14:46.14 | slipkid08 | vpbx |
14:46.22 | Katty | slipkid08: asterisk will take awhile to learn, that's for sure. |
14:46.37 | Katty | slipkid08: if you're not willing to spend the time, it would probably be best to find something a bit more suited to your needs |
14:46.39 | slipkid08 | yeah it seems to have a hefty learning curve |
14:46.44 | ectospasm | yeah, it took me two weeks to figure out how to get a call routed in the dialplan |
14:46.57 | slipkid08 | it's not that I WOULDN'T spend the time |
14:47.03 | slipkid08 | the time is not there to spend, you know? |
14:47.07 | Katty | slipkid08: depending on what country your in |
14:47.08 | slipkid08 | 3 jobs and a kid and a band |
14:47.21 | Katty | slipkid08: you might be just as well off going to Sam's Club and buying a small out of the box phone system |
14:47.23 | ectospasm | slipkid08: yeah, Asterisk isn't for the faint of heart |
14:47.31 | p3nguin | If you read The Book, you should be able to make calls within a few minutes after you finish reading. |
14:47.36 | slipkid08 | Well, I need a VPBX |
14:47.43 | ectospasm | p3nguin: heh, the book is over 400 pages! |
14:47.46 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
14:47.50 | Katty | slipkid08: you might consider talkswitch. |
14:47.54 | slipkid08 | pretty much is what it boils down to |
14:47.58 | slipkid08 | what is talkswitch |
14:48.11 | chuckf | I got it at home to play with at first, then my cousin started doing shows in eurpoe. I got DIDs from there so she could call local there to the *, then dial family and friends here in the US |
14:48.15 | Katty | slipkid08: i would steer clear of samsung boxes. |
14:48.20 | slipkid08 | huh |
14:48.23 | ectospasm | but remember, Asterisk _isn't_ a PBX. It's a PBX _Toolkit_ |
14:48.23 | slipkid08 | that's nice |
14:48.34 | slipkid08 | see I can't make out that grey line |
14:48.39 | Katty | Some Assembly Required |
14:48.44 | slipkid08 | oh haha |
14:48.48 | slipkid08 | I see |
14:48.54 | slipkid08 | kinda like my son's trike |
14:49.01 | slipkid08 | that was in 45 pieces lol |
14:49.10 | Katty | holy snickerdoodles, 45 pieces? |
14:49.11 | tzanger | heh |
14:49.14 | ectospasm | gah, I'd hate Asterisk if I had to know assembly to configure it! (-; |
14:49.24 | slipkid08 | but now it has rockets on the back and cards in the spokes |
14:49.24 | slipkid08 | lol |
14:49.26 | tzanger | someone should come up with a set of IKEA-like instruction pdfs for asterisk |
14:49.33 | slipkid08 | someone should |
14:49.42 | Katty | i'll get right on it. |
14:49.53 | ectospasm | tzanger: http://astbook.asteriskdocs.org |
14:49.55 | slipkid08 | I don't mind figuring stuff out, but I just don't have the time or the energy |
14:50.01 | ectospasm | yeah, it's outdated, but still good info |
14:50.04 | tzanger | ectospasm: no, I am talking pictograms :-) |
14:50.12 | ectospasm | tzanger: um. |
14:50.15 | Katty | slipkid08: if you really want asterisk, i'm sure you could find a consultant |
14:50.25 | ectospasm | tzanger: configure dialplan using pictograms. I dare ya |
14:50.42 | slipkid08 | Yeah but I am not paying $150/hr to have someone tell me something I could find on my own |
14:50.46 | slipkid08 | if given time |
14:50.53 | ectospasm | slipkid08: or use a turnkey solution like Switchvox |
14:50.56 | Katty | k |
14:51.06 | Katty | okay what does TURNKEY mean |
14:51.08 | Katty | seriously. |
14:51.11 | slipkid08 | haha |
14:51.18 | file | you turn a key and a cookie appears |
14:51.23 | ectospasm | Katty: basically a fully functioning system, COTS/out of the box/etc. |
14:51.27 | [TK]D-Fender | Katty: Put the Key in. Turn. Product jsut starts working. |
14:51.32 | Katty | ohah, kthen |
14:51.37 | slipkid08 | lol |
14:51.42 | slipkid08 | I like turnkey solutions |
14:51.46 | Katty | i like file's description better |
14:51.48 | *** join/#asterisk b0gatyr (~b0gatyr@host-208-88-126-198.biznesshosting.net) |
14:51.49 | Katty | file: did you go to pax? |
14:51.53 | file | Katty, I did |
14:52.08 | Katty | file: my sister was there |
14:52.24 | file | Katty, a lot of people were there >_> |
14:52.28 | Katty | file: yesh. |
14:52.42 | file | which produced the PAX cellular deadzone |
14:52.49 | Katty | :< |
14:53.10 | p3nguin | $150/hour? I'd work on your Asterisk system for less than half that! |
14:53.22 | b0gatyr | morning everyone, can someone please help me see what's going on here..I'm trying to call an international number but I get "Got SIP response 486 "Busy here" back from ..." but if I try calling some other number it goes through just fine.. what can be wrong? |
14:54.18 | slipkid08 | I'll brb |
14:54.38 | slipkid08 | I like you guys I think I'm going to lurk here for a while and just watch your conversations |
14:55.00 | p3nguin | If you want someone to set up a system for you, let me know. |
14:55.16 | jamko | gaytr sounds like your provider does not have you set up for int. |
14:55.48 | slipkid08 | how about setup for a trixbox Pro |
14:55.55 | b0gatyr | jamko: but it works when I dial another int number |
14:56.05 | Katty | p3nguin: take beer. |
14:56.09 | b0gatyr | it's just that number |
14:56.16 | b0gatyr | and it's random |
14:56.24 | jamko | gayr: ask your provider why they won't place the call. |
14:56.28 | p3nguin | katty: I like beer, but the doctor told me I have to stop drinking beer. |
14:56.55 | n3hxs | Ask the doc about Tequila. |
14:56.56 | Katty | p3nguin: whyfor? |
14:57.17 | p3nguin | He's worried about my liver. |
14:57.23 | Katty | oh :< |
14:57.57 | [TK]D-Fender | slipkid08: That is a closed platform and isn't supported here |
14:58.03 | p3nguin | I had an enzyme that was elevated 30% over a normal level, and it worried him so he had me stop drinking beer and stop taking Advil. We'll retest in 90 days. |
14:58.15 | n3hxs | p3nguin just add onions... ;) |
14:58.16 | slipkid08 | [TK]D-Fender: really?; |
14:58.19 | [TK]D-Fender | slipkid08: The kind of thing retard kiddies use :p |
14:58.22 | Katty | p3nguin: i hope everything turns out okay. |
14:58.26 | p3nguin | me too! |
14:58.53 | p3nguin | If it stays elevated, he'll probably test me to find out if I have NAFLD. |
14:59.03 | p3nguin | (non-alcoholic fatty liver disease) |
14:59.03 | Katty | what's that? |
14:59.08 | slipkid08 | what about virtual pbx |
14:59.22 | Katty | p3nguin: and what exactly does that mean? |
14:59.33 | [TK]D-Fender | slipkid08: Trixbox pro runs a closed source proprietary GUI that configures * to its cookie cutter design and isn't made for you to get "creative". |
14:59.50 | [TK]D-Fender | [10:58]<slipkid08>what about virtual pbx <- that is a "concept", not a "thing" |
15:00.07 | *** part/#asterisk skyion (~bradc@siza.thusa.net) |
15:00.30 | p3nguin | I guess having a fatty liver isn't acceptable. I've heard of other people being diagnosed with it, and they were kinda bummed over it. |
15:00.38 | Katty | Nugget: <3 |
15:00.51 | Katty | p3nguin: hmm. |
15:01.01 | Katty | p3nguin: you basically have to change your diet, i'm guessing? |
15:01.09 | p3nguin | I would think so. |
15:01.13 | Katty | nods |
15:01.26 | *** join/#asterisk grayhame (~grayhame@74-94-250-169-Nashville.hfc.comcastbusiness.net) |
15:01.50 | p3nguin | But my glucose, cholesterol, thyroid, kidneys... all good. |
15:04.18 | Katty | excellent. |
15:05.56 | n3hxs | p3nguin hope you feel better soon. |
15:06.44 | p3nguin | I hope the re-test shows that it was my overdosage of Advil that elevated that liver enzyme. |
15:06.54 | Katty | nods |
15:06.57 | Katty | i hope so too. |
15:07.07 | p3nguin | I had no thought at all when I took 600mg the night before my test panel. |
15:07.19 | slipkid08 | I am not really concerned with getting "creative" I just want to make and receive calls from anywhere and to anywhere without paying more server fees |
15:07.21 | Katty | i'm guessing that's high? |
15:07.43 | p3nguin | The normal adult dosage is two 200mg pills. |
15:07.57 | Katty | so 1 pill is 100mg? |
15:08.04 | p3nguin | I often take three because I figured more was better. |
15:08.07 | Katty | or 1 200mg? |
15:08.23 | Katty | ah. |
15:08.27 | p3nguin | one is 200mg, and I often take three. |
15:08.37 | Katty | does 2 not kick it? |
15:09.04 | slipkid08 | brb |
15:09.30 | Katty | oh, advil is ibuprofen |
15:09.32 | p3nguin | I felt like the regular 400mg dosage wasn't enough. He wasn't very happy with me for that. He said prescription strength 800mg is NOT the same as taking four regulars, so I was unncessarily overdosing. |
15:09.41 | p3nguin | yeah it is. |
15:09.49 | Katty | ibuprofen doesn't work for me, for some reason |
15:09.55 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
15:10.03 | n3hxs | slipkid08 google the ITSPs and look at their rates. I used les.net for a year and their rates were fairly good. Then they raised the base rates for just having an account and I dropped it because I was only "playing" with the service. |
15:10.19 | Katty | they had me on that stuff back when the wisdom teeth came out, but it didnt' really phase the pain. have you considered a different type of pain killer? |
15:10.27 | p3nguin | And he said the enzyme being high could be because of my overdosage of fast-acting ibuprofen. |
15:10.34 | Katty | nods |
15:10.55 | Katty | you might consider asking him what type of pain killer to try if the average dose of ibuprofen isn't doin it for ya |
15:11.01 | p3nguin | I forgot that I had a headache the night before the testing. |
15:11.07 | Katty | they switched me over to tylenol. |
15:11.13 | p3nguin | He said I can take that. |
15:11.16 | *** join/#asterisk KavanS (~KavanS@unaffiliated/kavans) |
15:11.17 | Katty | would not recommend asprin.... |
15:11.38 | p3nguin | I guess aspirin is almost the same as ibuprofen. |
15:12.26 | Katty | i've no idea |
15:12.35 | Katty | but i know it is autotoxic |
15:12.38 | p3nguin | Acetaminophen is okay, but he said stop taking ibuprofen. |
15:12.44 | Katty | damages your hearing over time |
15:12.55 | p3nguin | My hearing is bad enough already. |
15:13.03 | Katty | and can cause tinnitus if taken excessively |
15:13.17 | p3nguin | Hmm, I sometimes have some ringing, too. |
15:14.10 | Katty | i've had tinnitus for nearly a year. thankfull it's not bad. |
15:14.19 | p3nguin | faint ringing? |
15:14.27 | Katty | but i can tell you that caffeine, being tired, and high bp/stress really make it a lot worse |
15:14.41 | Katty | yes. faint now. used to be worse. i hear it a lot more right before i go to sleep. |
15:14.46 | p3nguin | It must not be too loud... I couldn't hear it on the conference last week. :D |
15:14.50 | Katty | during the day i can't really hear it anymore unless i listen for it |
15:15.01 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
15:16.03 | Katty | p3nguin: you might mention the tinnitus next time you're in for testing |
15:16.12 | KavanS | acetaminophen damages your liver... |
15:16.19 | KavanS | meh, I try to stay away from all that shit...unless you really need it |
15:16.25 | Katty | KavanS: i agree. |
15:17.34 | Katty | KavanS: pain killers are a beautiful thing, tho. |
15:18.26 | *** join/#asterisk felipe_ (~felipe@my.nada.kth.se) |
15:21.54 | *** part/#asterisk slipkid08 (~actioncom@rrcs-97-77-102-225.sw.biz.rr.com) |
15:25.16 | *** join/#asterisk Trox (~Adium@85-126-174-18.work.xdsl-line.inode.at) |
15:29.57 | *** join/#asterisk mrbnet (~ryanbantz@74-95-100-233-Minnesota.hfc.comcastbusiness.net) |
15:30.00 | n3hxs | likes his TENS for back pain. |
15:30.16 | Katty | tens? |
15:31.25 | n3hxs | Transdermal eletronic nerve stimulator. |
15:31.30 | n3hxs | or something like that. |
15:31.37 | *** join/#asterisk dodavoice (~d@41-134-22-10.dsl.mweb.co.za) |
15:32.15 | dodavoice | Kobaz you around? |
15:32.38 | p3nguin | looks around the channel |
15:32.50 | dodavoice | hi p3nguin |
15:32.54 | p3nguin | hello |
15:33.07 | dodavoice | have you seen him around today? |
15:33.17 | p3nguin | ~seen kobaz |
15:33.25 | infobot | kobaz is currently on #asterisk-dev #asterisk. Has said a total of 161 messages. Is idling for 18h 20m 27s, last said: 'perl.. not pearl'. |
15:33.25 | dodavoice | yip |
15:33.44 | dodavoice | thx |
15:34.43 | p3nguin | Anything the other 250 people here can help with? |
15:35.14 | dadavoip | he said he would hava a look ata project some customizing of vicidial |
15:35.50 | dadavoip | p3nguin how did you get the info regarding Kobaz? |
15:35.59 | p3nguin | You saw me. |
15:36.42 | dadavoip | yip |
15:36.51 | dadavoip | you hacker |
15:36.54 | dadavoip | lol |
15:39.02 | p3nguin | http://whatthefuckshouldimakefordinner.com/ |
15:40.06 | *** join/#asterisk CunningPike (~CunningPi@204.239.12.183) |
15:40.39 | dadavoip | <PROTECTED> |
15:41.45 | p3nguin | nod |
15:42.02 | dadavoip | you can put some goto if times |
15:42.47 | dadavoip | different menus for diverent days, even have some turkey on for the holidays |
15:45.17 | *** join/#asterisk AlHafoudh (~AlHafoudh@adsl-dyn250.78-98-184.t-com.sk) |
15:51.40 | *** join/#asterisk Deeewayne (~dwayne@75-150-14-49-Atlanta.hfc.comcastbusiness.net) |
15:51.40 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
15:53.02 | *** join/#asterisk BANSAL (~bansal@117.199.113.144) |
15:54.55 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
15:54.55 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
15:55.10 | *** join/#asterisk _zoom_ (~user@41.218.41.70) |
15:56.05 | _zoom_ | hi, for those who has played with termination, have experienced any problem with NICs ? |
15:56.37 | asilva | Little help.. this message is caused by what action - [Sep 9 12:53:35] WARNING[23860]: chan_iax2.c:3339 __attempt_transmit: Max retries exceeded to host 200.145.112.7 on IAX2/contadundi-7879 (type = 6, subclass = 11, ts=1780002, seqno=17) |
15:56.37 | asilva | [Sep 9 12:53:40] WARNING[23855]: chan_iax2.c:3339 __attempt_transmit: Max retries exceeded to host 200.145.112.7 on IAX2/contadundi-7879 (type = 6, subclass = 2, ts=1785007, seqno=18) |
15:56.37 | asilva | [Sep 9 12:53:45] WARNING[23854]: chan_iax2.c:3339 __attempt_transmit: Max retries exceeded to host 200.145.112.7 on IAX2/contadundi-7879 (type = 6, subclass = 11, ts=1790002, seqno=19) |
15:56.37 | asilva | [Sep 9 12:53:55] WARNING[23858]: chan_iax2.c:3339 __attempt_transmit: Max retries exceeded to host 200.145.112.7 on IAX2/contadundi-7879 (type = 6, subclass = 11, ts=1800002, seqno=20) |
15:56.40 | asilva | [Sep 9 12:54:01] WARNING[23857]: chan_iax2.c:3339 __attempt_transmit: Max retries exceeded to host 200.145.112.7 on IAX2/contadundi-7879 (type = 6, subclass = 2, ts=1806007, seqno=21) |
15:56.43 | asilva | <PROTECTED> |
15:56.45 | asilva | <PROTECTED> |
15:56.47 | asilva | [Sep 9 12:54:05] WARNING[23853]: chan_iax2.c:3339 __attempt_transmit: Max retries exceeded to host 200.145.112.7 on IAX2/contadundi-7879 (type = 6, subclass = 11, ts=1810002, seqno=22) |
15:56.52 | asilva | pabx*CLI> |
15:56.54 | asilva | [Sep 9 12:54:15] WARNING[23859]: chan_iax2.c:3339 __attempt_transmit: Max retries exceeded to host 200.145.112.7 on IAX2/contadundi-7879 (type = 6, subclass = 11, ts=1820002, seqno=23) |
15:56.57 | asilva | [Sep 9 12:54:22] WARNING[23857]: chan_iax2.c:3339 __attempt_transmit: Max retries exceeded to host 200.145.112.7 on IAX2/contadundi-7879 (type = 6, subclass = 2, ts=1827007, seqno=24) |
15:57.00 | asilva | [Sep 9 12:54:25] WARNING[23851]: chan_iax2.c:3339 __attempt_transmit: Max retries exceeded to host 200.145.112.7 on IAX2/contadundi-7879 (type = 6, subclass = 11, ts=1830002, seqno=25) |
15:57.01 | *** mode/#asterisk [+q asilva!*@*] by russellb |
15:57.03 | russellb | ~pb |
15:57.03 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood 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/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
15:57.07 | p3nguin | Jesus... use a fucking pastebin. |
15:57.47 | p3nguin | I wonder if he'd like it if we all went into his living room and dump a big bucket of shit on his floor. |
15:58.22 | *** join/#asterisk coppice (~chatzilla@112.203.17.210.dyn.pacific.net.hk) |
16:02.06 | *** mode/#asterisk [-q asilva!*@*] by russellb |
16:02.23 | _zoom_ | hi, for those who has played with termination, have experienced any problem with NICs ? |
16:02.37 | [TK]D-Fender | _zoom_: Huh? |
16:02.39 | p3nguin | huh? |
16:02.59 | p3nguin | Haha, glad I wasn't the only one with that thought. |
16:03.36 | *** join/#asterisk DaCoD (~DaCoD@189.4.108.113) |
16:03.45 | [TK]D-Fender | _zoom_: http://tinyurl.com/y72t82u |
16:04.19 | frigidzephyr | _zoom_: you mean like hardware level issues with flakey NICs? |
16:04.36 | frigidzephyr | thats a pretty broad, vague question |
16:05.02 | DaCoD | hi all |
16:05.06 | DaCoD | Can someone tell me what this error means and how to fix |
16:05.15 | DaCoD | Ext: 1 Cause: Interworking, unspecified (127), class = Interworking (7) ] |
16:05.41 | DaCoD | asterisk 1.4.32, dahdi, isdn |
16:07.27 | frigidzephyr | "Cause No. 127 - SW56 disconnect (Internetworking, unspecified) This cause indicates that an interworking call (usually a call to SW56 service) has ended. May also be seen in the case of a non specific rejection by your long distance carrier (try again at a different rate) " |
16:07.53 | leifmadsen | Katty: pictures now exist! |
16:08.24 | Qwell | leifmadsen: of the new hoose? |
16:08.35 | leifmadsen | Qwell: yes! new counter tops specifically |
16:08.41 | Qwell | nice. link? |
16:08.44 | *** join/#asterisk zerohalo (~zerohalo@173-13-92-17-NewEngland.hfc.comcastbusiness.net) |
16:08.56 | Qwell | oh. |
16:08.59 | leifmadsen | Qwell: http://www.facebook.com/album.php?aid=159742&id=512680761&l=20abfce25a |
16:09.07 | Qwell | yes, I figured facebook |
16:09.13 | leifmadsen | heh, you don't need to be signed in though |
16:09.23 | Juggie | leifmadsen, geek |
16:09.30 | Qwell | those chairs are too much |
16:09.46 | leifmadsen | Qwell: the black ones? |
16:09.46 | Qwell | they remind me of those clear plastic spoons |
16:10.03 | leifmadsen | Qwell: I thought you liked spoons?! |
16:10.09 | leifmadsen | ba-dunk-chink |
16:10.14 | *** part/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
16:10.17 | leifmadsen | Juggie: well duh :) |
16:10.25 | leifmadsen | Juggie: slate is sealed now! |
16:10.27 | Juggie | wheres the furniture :P |
16:10.29 | Juggie | nice |
16:10.38 | leifmadsen | Juggie: still in the process of getting it into the house... couch comes tomorrow |
16:10.45 | leifmadsen | we're still working on furniture a bit, heh |
16:10.49 | leifmadsen | no rugs or coffee table yet |
16:10.56 | Juggie | no pictures :P |
16:11.06 | p3nguin | Is that a single-family home? |
16:11.18 | DaCoD | <frigidzephyr>: OK, And how can I change the rate ? |
16:11.34 | leifmadsen | p3nguin: townhouse, ya |
16:12.00 | p3nguin | Oh great... Another nick copy-and-paste person... |
16:12.26 | frigidzephyr | DaCoD: not sure, i would make sure you are formatting your dialed number correctly as the telco wants it |
16:12.30 | beardy | Does anyone know of a "Jp" person? |
16:12.58 | Qwell | beardy: "Jp" could mean lots of things. |
16:13.05 | frigidzephyr | DaCoD: compare a local number dialed, vs long distance, etc see if you get same error |
16:13.50 | frigidzephyr | DaCoD: I think I saw a guy with that error once, when had some kind of restriction set on his service, couldn't dial a certain area, or long distance or something |
16:14.02 | beardy | Qwell: Yeah.. I mean with Jp as nickname. Thought I'd check if he was around here.. made a helpful comment on an * related article I wrote.. |
16:14.12 | p3nguin | ~seen jp |
16:14.21 | infobot | jp <n=JP@m195e36d0.tmodns.net> was last seen on IRC in channel #norganna, 608d 12h 37m 8s ago, saying: 'hi'. |
16:14.30 | DaCoD | <frigidzephyr>: support operator, told me I should switch to National subescriber |
16:15.01 | frigidzephyr | DaCoD: look into the pridialplan setting in chan_dahdi.conf then |
16:15.15 | *** part/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
16:15.20 | frigidzephyr | DaCoD: try setting that to national, or unknown, |
16:15.31 | p3nguin | dacod: Instead of copying and pasting his nick, why not type a couple letters of his nick and then press the TAB KEY? fr <TAB> will work just fine. |
16:15.38 | DaCoD | I change in chan_dahdi.conf but don't work, change National subescriber to National |
16:16.02 | frigidzephyr | DaCoD: I don't understand what you mean there. |
16:16.32 | frigidzephyr | DaCoD: did you set pridialplan=national ? Are they maybe talking about the switchtype? you could try national and ni1 for that |
16:17.00 | p3nguin | Smells as if someone is making garlic bread... I should investigate. |
16:17.03 | DaCoD | http://paste.debian.net/88896/ |
16:17.18 | DaCoD | look my group config |
16:18.05 | _zoom_ | frigidzephyr: yeap |
16:18.09 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
16:18.55 | frigidzephyr | DaCoD: try changing pridialplan to national ? that will change the TON, i'm not sure what they mean by National Subscriber ? |
16:19.38 | frigidzephyr | maybe they mean the subscribers TON should be national |
16:20.06 | frigidzephyr | you may want to set pridialplan to dynamic and have it set based on the prefixes you are using |
16:20.53 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
16:21.22 | _zoom_ | frigidzephyr: could a normal PC handle 10K termination ? |
16:21.32 | _zoom_ | using asterisk |
16:22.35 | frigidzephyr | if you mean an average desktop PC, no, probably not, maybe 10K registrations, as long as the registrations are staggered or quite delayed. Definitely not 10K active sip calls |
16:22.58 | *** join/#asterisk war9407 (war@liquidswords.org) |
16:23.14 | leifmadsen | schmits in #asterisk-dev has been working on over 20k registrations |
16:23.17 | DaCoD | p3nguin: ok, I try on the next |
16:23.24 | p3nguin | dacod: THANKS! |
16:23.42 | leifmadsen | but it's still in development and requires code changes that are not yet merged into asterisk :) |
16:23.59 | p3nguin | So what is the current limit on registrations? |
16:24.03 | DaCoD | frigidzephyr: tks for your help |
16:24.17 | frigidzephyr | DaCoD: you're welcome |
16:24.43 | *** join/#asterisk pif (~ldm@zenon.apartia.fr) |
16:29.22 | *** join/#asterisk imox1234 (~imox1234@p4FC5C53A.dip0.t-ipconnect.de) |
16:29.41 | *** join/#asterisk myster (~myster@207.148.172.210) |
16:36.40 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
16:42.17 | Deeewayne | leifmadsen, nice house |
16:45.21 | *** join/#asterisk Mhaddog_ (~Mhaddog@adsl-32-170-232.mia.bellsouth.net) |
16:49.26 | *** join/#asterisk methodvon (~methodvon@108.18.246.223) |
16:50.56 | *** part/#asterisk _zoom_ (~user@41.218.41.70) |
16:52.32 | leifmadsen | Deeewayne: heh thanks :) |
16:53.30 | *** join/#asterisk sol (~sol@unaffiliated/sol) |
16:55.26 | asilva | back! |
16:55.45 | asilva | a little help here. - http://pastebin.com/yy3AcEG5 these messages cause my asterisk to FREEZE could it be like a DoS? |
16:55.56 | *** join/#asterisk ahowlader (Adnan@180.234.35.175) |
16:58.01 | Katty | drags in |
16:58.22 | Katty | i think my trainer is trying to kill me. |
16:58.26 | Qwell | drugs Katty |
16:58.32 | dadavoip | asilva do a top command and see what processes is taking uf resources(do you have iax trunk or extension? |
16:59.19 | Katty | Qwell: :< |
16:59.26 | Qwell | in a good way! |
16:59.33 | Katty | drugs are never good! |
16:59.38 | Katty | unless you are going into surgery |
16:59.38 | Qwell | flintstones chewables! |
16:59.42 | Katty | :>>> |
16:59.48 | Katty | those are acceptabuhls. |
17:00.10 | asilva | dadavoip, iax account used by dundi.. type=friend |
17:00.23 | Katty | Qwell: do you care about my fiber?! |
17:00.35 | asilva | dadavoip, i'll check the %% of cpu and asterisk process next time it happens |
17:00.35 | Qwell | There's no good way to answer that. |
17:00.56 | Katty | infobot: fiber? |
17:01.09 | Katty | infobot: fiber is You care about my fiber?! http://www.youtube.com/watch?v=DjyklC02MNM |
17:01.10 | infobot | ACTION stuffs is You care about my fiber?! http://www.youtube.com/watch?v=DjyklC02MNM with fiber. |
17:01.48 | Qwell | that response is...odd |
17:01.59 | Katty | yes. |
17:02.01 | Katty | infobot: fiber? |
17:02.04 | Katty | :< |
17:02.09 | Qwell | infobot: stuffs |
17:02.33 | Katty | infobot: you care about my fiber? |
17:02.40 | Katty | suspicious. |
17:03.01 | Letoric | can anybody tell me what variable I can put in a mixmonitor file name to make sure it is truly unique? |
17:03.04 | asilva | dadavoip, tkz for the tip! |
17:03.21 | Letoric | IE, is there a variable for the uniqueid that is assigned when the call is placed in cdr-csv/Master.csv |
17:03.32 | Qwell | ${UNIQUEID} |
17:03.47 | Letoric | that variable works in a dial plan? |
17:04.03 | p3nguin | Where else would it work? |
17:04.10 | *** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net) |
17:04.11 | Letoric | heh, in the csv files! ;) |
17:04.15 | Letoric | thank you |
17:04.54 | leifmadsen | ~fiber |
17:05.01 | leifmadsen | ~stuffs |
17:05.16 | Katty | leifmadsen: i think infobot got some wires crossed :< |
17:05.33 | *** join/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net) |
17:06.15 | Chainsaw | Katty: It is stressed. It needs a spa day. |
17:06.20 | *** join/#asterisk EndEng (~epierce@mail.endeavoreng.com) |
17:06.40 | *** join/#asterisk Ad-Hoc (~nimbus@193.92.83.151.dsl.dyn.forthnet.gr) |
17:07.04 | Katty | infobot and me both. |
17:07.04 | infobot | moi? |
17:07.11 | Katty | infobot: yes, you dear. |
17:07.47 | Katty | and possibly a manicure. |
17:11.04 | *** join/#asterisk thansen (~thansen@c-98-202-28-239.hsd1.ut.comcast.net) |
17:11.27 | Letoric | if I want to make SURE all calls are monitored regardless of how the call flows, do I have to create a monitor statement in each context? Is there a better way to ensure this? |
17:11.49 | [TK]D-Fender | Letoric: No |
17:12.03 | Letoric | I'm currently catching it, but it's high maintenance when I want to do something like change the file name, and setting a global variable for the file name doesn't seem to work, it doesn't find the uniqueid or callerid properly |
17:12.11 | [TK]D-Fender | Letoric: Recording is a dilaplan option. It goes where you put it. |
17:12.12 | drmessano | Tell your users to record all calls on a tape recorder or you'll punch them in the face? |
17:12.17 | drmessano | That actually works |
17:12.29 | p3nguin | Why would you try using a global for that? |
17:12.40 | Trox | what about using a macro for your Dials? |
17:12.43 | drmessano | [TK]D-Fender: So does a punch to the face, usually |
17:12.47 | Letoric | so that I wouldn't have to change it for every different spot we initiate monitoring |
17:13.02 | Letoric | I'm kind of new, I haven't jumped too much into macros, just the stdexten one |
17:13.22 | [TK]D-Fender | Letoric: That name isn't implict |
17:14.41 | *** join/#asterisk Blackvel (~blackvel@dslb-084-057-081-245.pools.arcor-ip.net) |
17:15.42 | Blackvel | hi all. how can help me about voip delays (not echos) on snom 370 and pstn/isdn + patton isdn media gateway 4634? |
17:15.46 | Blackvel | who can... |
17:17.18 | Blackvel | its just a matter of some millisecs...but still enough delay that a 1h business interview is difficult enough |
17:17.58 | Blackvel | noone of both parties knows how to interrupt the party...question / answer playing gets a mess when noone knows when to start/stop talking |
17:19.17 | Blackvel | i am running snom370 firmware 7.3.23 which improved that over older versions..but its still quite far away from what we all know from pstn/isdn |
17:19.53 | Blackvel | AND voip is only local at the office (ISDN pstn, 1meter to patton + 1meter to asterisk/snom |
17:27.12 | *** join/#asterisk nachoguy (~boster@nacho-tested.hrapproved.com) |
17:30.31 | *** join/#asterisk cusco (~trilili@62.28.152.206) |
17:30.32 | cusco | hi |
17:30.44 | Katty | ohai2u |
17:30.50 | cusco | asterisk for some reason is not listening om port 5060 |
17:30.55 | cusco | on external IP |
17:33.00 | *** join/#asterisk polk330 (4297116e@gateway/web/freenode/ip.66.151.17.110) |
17:33.21 | drmessano | How do you know? |
17:34.06 | drmessano | You can't just make a blanket statement like that, bro |
17:34.28 | drmessano | Don't tase your Asterisk box without proof |
17:34.37 | polk330 | Hey Guys, Useing AsteriskNow 2.8.0.2 Just wondering if there are any Front end typr thing for something like a Customer Interface. For someone that my company would sell VoIP to. Somewhere that thye can chage there Followme/VM etc |
17:34.54 | cusco | drmessano: telnet on port 5060 gives connection refused |
17:34.54 | Nugget | telnet is eeeeeeevil! |
17:35.06 | drmessano | cusco: It's UDP, of course it is |
17:35.06 | cusco | and I had permission to launch nc -l -p 5060 |
17:35.13 | cusco | it is UDP? |
17:35.18 | drmessano | 5060 UDP |
17:35.18 | Trox | cough - udp - cough |
17:35.36 | cusco | o.O |
17:35.37 | leifmadsen | p3nguin: regarding "how many registrations can asterisk handle right now?" see the [Code Review] Rate limit astdb->sync() calls" thread on asterisk-dev. Stefan Schmidt says 1.6.2.10 can do about 8k registrations, with the patch in that thread it goes to 15k, and with that patch plus another that will be on reviewboard soon, he gets over 20k peers without a problem. To quote him, "20k peers with qualify=yes would cause a l |
17:35.37 | leifmadsen | oad of 0,07 on my system, sending around 350 to 400 sip packets per second and around 2 mbit of bandwith." |
17:35.48 | cusco | sorry |
17:36.01 | *** join/#asterisk n3hxs (~HAMming@63.68.135.4) |
17:36.02 | Qwell | holy crap. 2mbit from qualifies? |
17:36.08 | leifmadsen | 20k registrations :D |
17:36.20 | cusco | well I can't find out what is wrong then, router has forwarding set OK... why can't I register from outside, and no error comes up on cli? |
17:36.23 | drmessano | polk330: You don't have AsteriskNOW 2.8.0.2, you have FreePBX 2.8.0.2, which isn't supported in this channel. In any case, there is no portal for customers, as it's not designed to run a VoIP business with. |
17:36.25 | leifmadsen | I'm guessing that is burst traffic |
17:36.43 | polk330 | My bad thanks, freepbx channel then? |
17:36.44 | polk330 | lol |
17:36.51 | drmessano | Yes, but the same answer applies |
17:36.56 | Trox | cusco: portforwarding for udp? |
17:37.01 | cusco | Trox: yes |
17:37.08 | drmessano | You can ask me in there and I can cut/paste my answer |
17:37.20 | polk330 | drmessano any suggestions? |
17:37.50 | [TK]D-Fender | polk330: No. You are using FREEPBX, and that is not meant for you to run a telco out of. |
17:38.01 | [TK]D-Fender | polk330: That is made for SMB PBX use. |
17:38.02 | cusco | how do I test udp connection? |
17:38.07 | polk330 | Ok, well i mean it has been heard of to be done |
17:38.15 | [TK]D-Fender | polk330: Not with FreePBX |
17:38.21 | polk330 | <PROTECTED> |
17:38.23 | [TK]D-Fender | polk330: * as a core sure. |
17:38.43 | drmessano | polk330: FreePBX is designed to setup/admin a PBX, not an ITSP.. You need another GOOEY, which you probably have to roll yourself |
17:38.55 | polk330 | Yeah, via PHP. |
17:39.01 | drmessano | or whatever |
17:39.15 | drmessano | .NET, something that rhymes with PAILS, whatever |
17:39.42 | polk330 | Would you have any idea where to start? |
17:39.47 | drmessano | Generally your Asterisk box wouldn't even host this GUI.. it would be done on some backend and perhaps a MYSQL connection to Asterisk |
17:40.02 | polk330 | Yeah for security reason |
17:40.27 | drmessano | polk330: No, because you dont run your customer frontend or billing on your call handler |
17:40.48 | drmessano | Why not just pass all the calls through the secretarys laptop? |
17:40.54 | polk330 | Lmfao, |
17:41.49 | drmessano | Do you realize what you're going to need to start an ITSP? |
17:42.26 | drmessano | It isn't just about having an Asterisk box and some ATA's or phones connected. It's the PSTN end of the cable where the money starts |
17:42.28 | leifmadsen | no one realizes what they need to start an ITSP :) |
17:42.32 | Qwell | drmessano: lies. |
17:42.39 | Qwell | ~next vonage |
17:42.50 | Qwell | ~vonage |
17:42.50 | infobot | it has been said that vonage is a bunch of monkeys |
17:42.54 | Qwell | shrugs |
17:42.59 | leifmadsen | ~nextvonage |
17:43.00 | polk330 | <PROTECTED> |
17:43.36 | drmessano | Ok, well you need to get your Asterisk knowledge well above "I just installed AsteriskNOW" before beginning to attempt it |
17:43.51 | drmessano | ~book |
17:43.51 | 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 |
17:43.55 | drmessano | Read that, 3 times |
17:44.10 | Trox | and terrorize jared a bit ;) |
17:44.22 | Qwell | adds an option 7 to the AsteriskNOW kickstart list. |
17:44.22 | polk330 | Ive installed AstersikNOW b4 |
17:44.31 | Qwell | "7) The Next Vonage" |
17:44.42 | drmessano | polk330: We are WAY beyond AsteriskNOW here, period |
17:44.43 | polk330 | Ive even set it up with a customer side, |
17:44.49 | polk330 | What are you at now? |
17:44.51 | polk330 | lmfao |
17:45.07 | b14ck | pro troll |
17:45.08 | drmessano | polk330: You're talking about IIS on an XP box, we're talking about Apache clusters in a datacenter |
17:45.25 | polk330 | Yeah? well you dont have to be a huge dick? |
17:45.32 | drmessano | I am? |
17:45.33 | *** join/#asterisk tzanger_ (~tzanger@gromit.mixdown.ca) |
17:45.43 | tzanger_ | stupid freenode. |
17:45.44 | b14ck | polk330, what's your goal? |
17:46.00 | *** join/#asterisk KermitTheFragger (~KermitThe@118-197.bbned.dsl.internl.net) |
17:46.09 | b14ck | polk330, I'll give you some practical advice. |
17:46.19 | polk330 | k |
17:46.20 | drmessano | He can sell you a big Trixbox |
17:46.21 | nachoguy | being a huge dick would be to tell you to take your tinkertoys and play elsewhere. He's (fairly kindly) telling you that you haven't thought the problem through |
17:46.27 | b14ck | doesn't use trixbox. |
17:46.43 | polk330 | Trixbox is shitty to. |
17:46.45 | [TK]D-Fender | Used to WORK for them if I recall.. |
17:46.55 | b14ck | I used to work for them. |
17:47.00 | b14ck | I'm not stupid though. |
17:47.01 | Qwell | [TK]D-Fender: he had the sense to QUIT |
17:47.01 | b14ck | =p |
17:47.34 | *** join/#asterisk cusco (~trilili@213.63.137.210) |
17:47.44 | [TK]D-Fender | [13:46]<b14ck>I used to work for them. [13:46]<b14ck>I'm not stupid though. <--- auto-contradiction FAIL :p |
17:47.59 | b14ck | To be fair, [TK]D-Fender, it was my first telephony-related job. |
17:48.11 | b14ck | My experience is in programming, and before working there I didn't know anything about telephony stuff. |
17:48.25 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
17:48.28 | tzanger | telephony's fun |
17:48.38 | tzanger | the realtime constraints of tdm networks doubly so |
17:48.52 | b14ck | Yah, telephony is ok =) |
17:49.00 | Qwell | phones suck |
17:49.09 | upb | b14ck: how do you rate the dialplan implementation/format ?:P |
17:49.14 | b14ck | upb, it sucks |
17:49.18 | b14ck | but so does AEL. |
17:49.25 | drmessano | polk330: You can call me names if you want, but you really need to understand the scale of what you're trying to do here.. Setting up a box designed for an office PBX, that controls the Asterisk dialplan to accomplish JUST that, is not the way to go here.. you need to be thinking at a much higher level.. Dedicated vanilla Asterisk install, likely a custom backend to glue it all together |
17:49.39 | b14ck | The best telephony libraries are that which come with twilio/tropo =p |
17:49.42 | upb | heh yeah, just wanted to know someones opinion who has design experience |
17:49.54 | b14ck | There's almost no room for proper design patterns with asterisk stuff :( |
17:50.17 | EndEng | should you asterisk box live in your DMZ or internal network? |
17:50.19 | *** join/#asterisk Martinblr (~Miranda@61.12.17.170) |
17:50.27 | b14ck | EndEng, internal network. |
17:50.39 | drmessano | ...or just stick with Asterisknow and send me your customer list when it hits around 200 and you can't bill or scale. Whateve. |
17:50.51 | Martinblr | I am trying to connect linksys SPA3102 box to my Asterisk and trying to send FAX but not able to send.. |
17:50.56 | drmessano | He's gone anyway :( |
17:51.11 | *** part/#asterisk ShaunR (~shaun@freenode/sponsor/NDChost.com) |
17:51.25 | Martinblr | the extension created in the Asterisk is configured in Linksys SPA3102 and fax machine is connected.. |
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18:11.27 | *** part/#asterisk Trox (~Adium@85-126-174-18.work.xdsl-line.inode.at) |
18:12.20 | voxter | Hey, any of you guys know of/use a tool that is preferably open source, that will take in a pcap stream (or just listen on an interface) and analyze sip/rtp to identify when there are periods of poor audio quality? |
18:12.41 | voxter | Im having an intermittent issue with audio quality degrading then going back to normal and I'd like to be able to pinpoint it happening |
18:16.14 | cusco | sipp (sip-tester package in debian) |
18:16.36 | cusco | its hard to handle tho |
18:17.28 | voxter | sipp can take an active stream from a live box and analyze RTP as well for low MOS? |
18:17.38 | voxter | I thought sipp only told you about SIP |
18:17.48 | cusco | ow sorry it is |
18:17.58 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.130.40.dsl.dyn.forthnet.gr) |
18:19.19 | coppice | a rolling SIPP gathers no MOS |
18:19.52 | *** join/#asterisk utahsaint (~utahsaint@mail.ntegratedsolutions.com) |
18:20.04 | asilva | <PROTECTED> |
18:23.42 | [TK]D-Fender | asilva: It means that no function with that name exists |
18:24.19 | [TK]D-Fender | asilva: Perhaps you should show us your failed call that generated that error. |
18:26.29 | asilva | here it goes |
18:26.30 | asilva | http://pastebin.com/CEkENJr0 |
18:29.43 | [TK]D-Fender | aliAnd the source line that coased it from your dialplan? |
18:31.10 | asilva | is under configuration |
18:31.19 | asilva | on the pastebin |
18:31.22 | asilva | line Set... |
18:31.35 | Letoric | Ok, I found this yesterday, but today it is evading my googlefoo! Can anybody point me to a good method for setting up a feature code that allows our team to activate a different greeting on our phone system when we're in a meeting? IE, an 'all our people are busy, if it's an emergency, dial 911, otherwise sit and wait!!' ;) |
18:32.15 | asilva | before the Set verbose message asterisk prints the error message |
18:32.15 | Letoric | sorry asilva, I was scrolled up, didn't mean to disrupt your conversation ;) |
18:32.29 | asilva | ohhh np!!! |
18:33.24 | cusco | asilva: means what it says. You are using IfTime but asterisk doesn't have that function registered |
18:33.45 | asilva | core show funcion command shows IFTIME there |
18:33.54 | [TK]D-Fender | asbeacuse that is in UPPERCASE |
18:34.06 | [TK]D-Fender | asilva: Which was your mistake. Functions are CASE SENSITIVE |
18:35.18 | asilva | lol that's right |
18:35.25 | asilva | solveddd |
18:35.39 | asilva | head full of things can't see a little thing.. |
18:36.40 | *** join/#asterisk nightwalk (~nightwalk@daimon.vixel.org) |
18:36.59 | Letoric | ok, now for mine ;) |
18:37.18 | Letoric | Ok, I found this yesterday, but today it is evading my googlefoo! Can anybody point me to a good method for setting up a feature code that allows our team to activate a different greeting on our phone system when we're in a meeting? IE, an 'all our people are busy, if it's an emergency, dial 911, otherwise sit and wait!!' ;) |
18:37.22 | Letoric | perty please! |
18:37.53 | Letoric | need to be able to activate and deactivate it by feature code |
18:40.31 | [TK]D-Fender | Letoric: "core show application gotoif", "core show function DB" <----------- |
18:41.05 | [TK]D-Fender | Letoric: Set a db value to indicate what state you should be in. Checkk that state where your call comes in and go somewhere else accordingly |
18:42.16 | Letoric | ok, I understand the basic theory, but may need a bit more help |
18:42.39 | Letoric | will try out my googlefoo and come back in a bit if I'm stuck. Thank you |
18:45.02 | Blackvel | what settings of a jitter buffer do you use on LAN (patton smartnode)? default seems to be 60ms |
18:46.35 | Blackvel | what is the default for snom phones? |
18:51.28 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
18:53.56 | Blackvel | hm. both patton GW 4634 and snom370 have alaw/ulaw with 20ms |
18:54.25 | Blackvel | i have no idea where i should "tune" for low latency / LAN |
18:56.30 | Blackvel | can reducing jitter buffer in patton gw from 60ms to 20ms improve delay/latency? |
18:58.02 | Chainsaw | looks for a jitter buffer setting in his 4634 config file and can't find any |
18:58.08 | Chainsaw | So I must be on whatever the default is... |
19:01.02 | Blackvel | 60ms is default |
19:01.21 | Blackvel | + adaptive |
19:01.38 | Blackvel | this delay/latency makes me crazy and any business talk useless |
19:01.47 | Qwell | Why do you have a jitterbuffer enabled, if it's a meter connection? |
19:01.54 | Qwell | you said it's just LAN, right? |
19:02.09 | Blackvel | snom370 - asterisk - patton gw - ISDN |
19:02.40 | Blackvel | there is no turn off...if i see it right...but i could set to a static one (from adaptive) |
19:02.55 | Blackvel | i doubt that will changes anything anyway |
19:03.39 | Blackvel | what could i be doing that packets get transmitted "faster"? |
19:03.42 | Blackvel | or shorter? |
19:04.04 | Blackvel | patton gw would allow 10ms setting |
19:04.11 | Blackvel | but snom only let me choose 20ms |
19:04.34 | Blackvel | again...dunno if that would change anything... |
19:08.27 | Blackvel | can it be that snom FW messes voice packets and doesn't get voice to the point? |
19:08.54 | Blackvel | do you all have the same problems within offices and smartnode/pstn/isdn gateways? |
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19:51.31 | *** join/#asterisk Knew (~Solipsism@unaffiliated/sabioso) |
19:51.33 | Knew | Hey |
19:51.42 | Naikrovek | hey |
19:51.58 | Knew | Anyone know of a PBX/IVR system that presents the customer (the caller) with a web interface? |
19:52.32 | Knew | IVR systems suck. Even the speech ones. |
19:53.00 | Naikrovek | i had an idea once where the phone you were calling from would present you the menu on its local interface, like softbuttons or something |
19:53.18 | Naikrovek | the communication is already digital <--> digital, why rely on DTMF |
19:53.22 | Knew | Has anyone seen a solution where the user makes their menu selections/enters extension or account info on the web site securely, then gets called by the menu system and patched through? |
19:53.22 | Naikrovek | but no one thought it was a good idea |
19:53.37 | Naikrovek | Knew: that is possible, but i don't know of a system that already does this |
19:53.48 | Knew | Naikrovek, do you think it's a good diea? |
19:53.49 | Knew | *idea |
19:53.57 | Naikrovek | yeah why not |
19:54.14 | Naikrovek | would make a lot of sense in customer service |
19:54.16 | Katty | hi Naikrovek |
19:54.21 | Naikrovek | hi katty :) |
19:54.28 | Deeewayne | knew, is your user an agent? |
19:54.35 | Knew | Right now with Chase, they have me entering my account information and PIN number over unencrypted, analog phone lines using DTMF. How secure |
19:54.37 | Katty | hugs Deeewayne |
19:54.44 | Kobaz | it's teh katty\ |
19:54.55 | Katty | mhmm |
19:54.55 | Deeewayne | bear hugs Katty |
19:55.09 | Knew | Deeewayne, what user? :p |
19:55.39 | Deeewayne | knew, re: 'user' in your 14:53:21 comment |
19:55.47 | Knew | I'm going to have about four developers working under me for a week and I'm thinking about what we want to do for a project. |
19:56.03 | Knew | Deeewayne, oh. The user is the caller. |
19:56.15 | *** join/#asterisk moa_ (~moa@65-19-228-168.vnet-inc.com) |
19:57.01 | Knew | I was thinking of developing a PBX/IVR solution that actually handles all extensions/menu navigation/account info entry over the web site, and then calls them. |
19:57.17 | Knew | Virtual queueing would of course exist automatically, too. |
19:57.48 | Knew | It would tell them on the web site how much longer to wait |
19:58.28 | Knew | What do you guys think? |
19:58.38 | nucce | May I have any ideas what too do with this problem? Ive got a wav file that I want into my asterisk.. So I did simple convert it into a alaw pcm64a file.. and when I call with a static phone it sounds great.. but when I call with a mobile phone it sounds crap.. It seems it try too play it too loud and makes the noise very bad.. Anyone have some tip howto solve this? |
19:59.05 | Naikrovek | Knew: what you are proposing is a great idea imho |
19:59.07 | nucce | I did try to lower the volume with sox but the result still aint any good. |
19:59.13 | *** join/#asterisk guilept (~guile@a213-22-53-84.cpe.netcabo.pt) |
19:59.47 | Naikrovek | nucce: turn down the volume on the mobile phone, and try it on more than one mobile phone |
19:59.52 | Naikrovek | it could be a GSM codec thing as well |
20:00.05 | nucce | I did try it on 3 different phones with 3 different operators. |
20:00.12 | Naikrovek | then it's GSM or something else |
20:00.23 | Naikrovek | the GSM audio codec is optimized for voice and nothing else |
20:00.47 | nucce | Well, other helplines got thoose audietunes and it sounds good from mobile phone |
20:01.22 | Naikrovek | nucce: yes but they pre-process the file. removing almost all low and high frequency sounds so that GSM doesn't chop them out and make them sound awful |
20:01.37 | Naikrovek | put it through audacity and turn down the bass and treble, and try again |
20:01.44 | Naikrovek | i think you'll find a much better outcome |
20:01.56 | nucce | thank you! that is what I call a good tip! |
20:02.02 | Naikrovek | well |
20:02.07 | Naikrovek | it's only a good tip if it's useful |
20:02.21 | Naikrovek | time will tell |
20:02.48 | nucce | I will try now.. :) |
20:03.26 | nucce | Im not so good with this audicity due |
20:03.28 | *** join/#asterisk adyn (~adyn@unaffiliated/adyn) |
20:03.31 | nucce | first time I download it now |
20:03.49 | hardwire | voxter: poke |
20:04.17 | voxter | sup |
20:04.41 | hardwire | do you guys operate as a hosted wholesale platform provider? |
20:05.03 | voxter | hardwire: we do now yep |
20:05.11 | hardwire | Interesting |
20:05.12 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
20:05.15 | *** join/#asterisk ccesario_ (~ccesario@187.75.139.188) |
20:05.19 | voxter | hardwire: north america coverage only for now, though |
20:05.54 | hardwire | I was thinking we'd use a hosted wholesale platform that somebody else maintains and manage our trunks through that platform as well as rates and LCR |
20:06.09 | hardwire | not specifically use a wholesale provider, but have somebody host the wholesale platform |
20:06.19 | voxter | hardwire: interesting, yeah we are just about finished rolling out a new API that allows us to give direct access to our provisioning platform for just that reason |
20:06.38 | voxter | hardwire: and we also have it set up to work for resellers as well, with multiple clients below them |
20:06.46 | hardwire | what about least cost routing and the like? |
20:07.01 | *** join/#asterisk guilept (~guile@a213-22-53-84.cpe.netcabo.pt) |
20:07.22 | guilept | hei |
20:07.31 | hardwire | hush |
20:07.45 | guilept | whats up? |
20:07.48 | hardwire | shhhhhhh |
20:08.39 | Qwell | we're mourning. |
20:13.20 | frigidzephyr | i read that as mounting |
20:13.32 | frigidzephyr | confused me for a minute |
20:13.51 | guilept | my condolences |
20:16.21 | *** part/#asterisk guilept (~guile@a213-22-53-84.cpe.netcabo.pt) |
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20:28.44 | ahowlader | hi tzafrir_laptop |
20:32.40 | *** join/#asterisk nightwalk (~nightwalk@daimon.vixel.org) |
20:36.30 | *** join/#asterisk Trox (~Adium@85-126-174-18.work.xdsl-line.inode.at) |
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20:39.58 | EndEng | is there a asterisk PPA for lucid? |
20:40.41 | Qwell | EndEng: #ubuntu |
20:40.53 | EndEng | yes ubuntu |
20:41.59 | Qwell | /join #ubuntu |
20:42.01 | Naikrovek | heh |
20:42.13 | tzafrir_laptop | ahowlader, hi |
20:42.41 | ahowlader | hello tzafrir_laptop |
20:42.50 | *** join/#asterisk xuser (~xuser@unaffiliated/xuser) |
20:43.01 | Qwell | LOOP DETECTED |
20:44.25 | *** join/#asterisk cjk (~cjk@vodsl-8936.vo.lu) |
20:44.30 | tzanger | Qwell: does the loop give a funky groove? |
20:44.44 | Letoric | ok Fender: I did some brief research. The samples I found on voip-info.org seem to reference out of date commands |
20:44.47 | Letoric | DBPut and DBGet |
20:45.07 | Letoric | how do I set and read the database with 1.6.2.11? |
20:46.03 | *** join/#asterisk unspin (~unspin@S010600031d02196a.vc.shawcable.net) |
20:46.24 | Naikrovek | continue your research and you will find that info, surely |
20:46.42 | Naikrovek | ([TK]D-Fender has stepped out) |
20:46.48 | Letoric | thanks |
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20:55.46 | Katty | ATTENTION |
20:55.51 | Katty | IT IS NOW 1 HR UNTIL 5PM, CST |
20:55.53 | Katty | that is all. |
20:56.08 | Kobaz | heh |
20:56.12 | Kobaz | it's 4:56 here |
20:56.19 | Katty | i hate you. |
20:56.26 | Katty | pouts. |
20:57.37 | Kobaz | heh |
20:57.56 | Letoric | Naikrovek - I don't quite follow what the wiki is saying about the database stuff |
20:58.19 | Letoric | I did find the updated commands |
20:58.30 | Letoric | but it's not quite clicking on how to mold that into what I need heh |
21:00.04 | Letoric | in the example, is family/key literal, or is that something like table and row? |
21:00.11 | Letoric | what is $foo supposed to be? |
21:00.25 | Letoric | err, ${foo} |
21:01.07 | Katty | Kobaz: i am so bored. |
21:02.06 | Kobaz | you can come visit |
21:03.10 | Katty | brt |
21:03.18 | Kobaz | brt? |
21:03.36 | Kobaz | bears remember tennis? |
21:03.45 | Qwell | bacon's really tasty |
21:04.33 | Kobaz | beavers racing through? |
21:06.12 | Katty | i like Qwell's better. |
21:06.29 | Qwell | that's wha... nevermind. |
21:06.42 | Katty | mhmm |
21:06.59 | hay | asking again since many joins and parts happened.. I would like to have asterisk running on FreeBSD and on an ISDN line (1 BRI)... what PCI ISDN cards are tested and working in this combination... of course, as cheap as possible :-) thx |
21:07.16 | Katty | hay: are you looking for a consultant? |
21:07.22 | Kobaz | ~cheap |
21:07.22 | infobot | cheap is probably a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
21:08.07 | hay | Katty: I suppose freenode IRC network isn't a right place of looking for a consultant... |
21:08.22 | Katty | hay: well it can be...i'm just not really sure waht you're asking for. |
21:08.48 | Katty | Qwell: and no comments about previous statement. kthx. |
21:09.02 | Kobaz | hay: your question could either mean "i want someone to set this up for me". or "i want a point in the right direction for getting this going" |
21:09.02 | Qwell | huh what? |
21:09.07 | hay | Katty: I am looking for some experiences of experienced users using FreeBSD and ISDN cards... as I mentioned before |
21:09.28 | Katty | hay: that's still not specific enough. |
21:09.28 | hay | Kobaz: second meaning was almost exactly what I wanted... |
21:09.32 | Katty | hay: what, precisely, do you want |
21:09.47 | Kobaz | Katty: someone to tell him which isdn cards work in freebsd |
21:10.05 | Katty | ah. |
21:10.20 | hay | Kobaz: and in Asterisk ... I hope that isn't too tough question... |
21:10.43 | beardy | Well, if you couldn't find out yet it can't be that easy. |
21:10.47 | Kobaz | hay: i'm not sure if we can handle that |
21:10.53 | Katty | hi beardy |
21:11.04 | Kobaz | it sounds hard |
21:11.07 | beardy | Hi Katty |
21:11.27 | Katty | you know what sounds good. |
21:11.32 | Qwell | bacon. |
21:11.34 | hay | if there are some docs anywhere I will be glad to read them... however, I wasn't able to find anything useful... |
21:11.36 | Katty | tacos. |
21:11.40 | Qwell | bacon tacos. |
21:11.42 | beardy | hay: There might be hardware compatability lists for FreeBSD, as there is for Linux, looked for that? |
21:11.47 | Katty | mmmm, no. |
21:11.49 | Katty | beef/turkey tacos |
21:11.56 | Qwell | I think I might have to go get tacos this weekend now |
21:11.58 | Katty | bison would also be acceptable. |
21:12.06 | beardy | Katty: Mm tacos |
21:12.12 | *** join/#asterisk jamicque (~jam@80.50.125.74) |
21:12.18 | Katty | those volcano tacos at taco bell are win. |
21:12.22 | Nugget | Qwell: any chance you've got a sample cisco phone config file for use with sccp/skinny? I want to take another pass at using that instead of sip and the internets are coming up empty |
21:12.25 | Qwell | wait, they still make those? |
21:12.27 | Katty | Nugget: ohai |
21:12.32 | jamicque | hi, can anyone recomend me a solution to make a CTI for Asterisk? |
21:12.33 | Qwell | Nugget: nope |
21:12.34 | Katty | Nugget: i was looking for you earlier. something about facebook. |
21:12.35 | Nugget | 79x1/79x5 series, not that old 79x0 shit |
21:12.40 | Katty | Nugget: but it seems to have slipped my memory |
21:12.42 | Nugget | ok, thanks |
21:12.46 | Nugget | huggles Katty |
21:12.54 | Katty | hugs Nugget, goes to refresh memory |
21:12.56 | hay | beardy: yes, our course... but combining that with asterisk and overall experience using it and getting some information from people that use it looked like something useful... |
21:13.27 | hay | but I suppose nobody is using freebsd with asterisk and ISDN at all? or are you all hiding it? :-) |
21:13.41 | Qwell | hay: Here's a tip. If you're having issues finding information about using FreeBSD.. Don't. |
21:13.56 | beardy | hay: Best of luck with it. Asking meta questions usually isn't the best approach though. |
21:14.01 | Katty | Nugget: oh yes, the tea party thing. |
21:14.22 | Kobaz | Qwell: hey... freebsd's not that bad |
21:14.23 | Katty | Nugget: did you find that off of reddit? |
21:14.27 | Kobaz | Qwell: it's got zfs |
21:14.32 | Kobaz | (which i use at home) |
21:14.34 | Qwell | Kobaz: you get to support him then ;) |
21:14.36 | Nugget | no |
21:14.41 | Katty | oh ah |
21:14.43 | Katty | digs up link |
21:14.45 | hay | Qwell: actually I am pretty satisfied with FreeBSD docs (a long time sysadmin of it), but asterisk's hardware lists seem to be hard to find for me :-) |
21:14.58 | Kobaz | Qwell: well i never set up asterisk or isdn cards in freebsd though... i couldn't even get asterisk to compile |
21:15.13 | Nugget | FreeBSD is doubleplusawesome, but if you want to run Asterisk just use Linux. I hate Linux just as much as the next guy, but it really is the path of least pain. |
21:15.14 | hay | overall... I am really surprised by the reactions of all of you here... I didn't want to hurt anybody's feelings at all... :-) |
21:15.27 | Kobaz | Nugget: why all the hate? |
21:15.32 | Katty | Nugget: http://i.imgur.com/1pWX5.jpg |
21:15.33 | beardy | Katty: So you're hungry again? :) |
21:15.41 | Nugget | I love Unix and Linux is a pretty shitty Unix. |
21:15.44 | hay | Nugget: excellent... that was an excellent answer... thanks :-) |
21:15.59 | Katty | beardy: are you kidding me? i have the metabolism of a ferret. |
21:16.13 | Kobaz | Katty: is that high or low? |
21:16.21 | Katty | Kobaz: that means i'm starving every 2 hours |
21:16.22 | beardy | wonders too |
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21:16.53 | Nugget | Katty: yeah, that's the pic that led me to the site. |
21:17.01 | Nugget | someone posted it somewhere I saw it, so I googled. |
21:17.50 | Kobaz | linux is supposed to get a native port of zfs soon |
21:17.57 | Kobaz | but not in the mainline kernel because of the stupid cddl |
21:18.05 | Nugget | s/cddl/gpl/ |
21:18.08 | Nugget | fixed that for you |
21:18.27 | Kobaz | but btrfs looks like a better fs anyway... it just needs to be finished |
21:18.36 | Nugget | the gpl is the license that was designed to be intentionally difficult to cooperate with other licenses. Not fair to blame the cddl for that. |
21:18.47 | Kobaz | Nugget: no it wasn't |
21:18.57 | Nugget | yes it was. stallman is quite clear on that. |
21:18.58 | Katty | Nugget: excellent. |
21:19.02 | Kobaz | Nugget: the cddl was designed like that... you have it backwards |
21:19.21 | Kobaz | stallman is clear on that he wanted a very specific definition of free |
21:19.27 | Kobaz | plenty of other licenses are compatible with the gpl |
21:19.32 | Nugget | no, the gpl did it first. stallman expressly said he viewed the gpl's intentional incompabatibility to be a key mechanism he wanted to use to drive developers to use the gpl. |
21:19.53 | Nugget | that was a design *goal* of the gpl -- to be hard to cooperate with other licenses |
21:20.00 | Nugget | there's plenty of writing on fsf.org that explains that |
21:21.07 | Kobaz | there's rumors that sun specifically picked cddl to keep zfs out of linux |
21:22.37 | Nugget | there's rumors that nasa faked the moon landing. I'm not sure that tells us muc. |
21:23.17 | Nugget | and, even if it's true, that doesn't change the fact that the gpl's deliberate incompatibility is what made the decision possible. |
21:23.39 | Nugget | it's trivial to make an open source license incompatible with the gpl -- witness the old form bsd "advertising clause" license |
21:24.22 | Nugget | (which pre-dates the gpl, obviously. I'm not saying it was crafted to be incompatible, that would be impossible. just that even the most innocuous constraints can exclude gpl cooperation) |
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21:40.03 | Kobaz | hmm |
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21:41.47 | DruRoland | anyone available to help with some, probably, basic questions regarding extensions? |
21:43.41 | Chainsaw | Quite likely. Please proceed with your question. |
21:44.18 | DruRoland | I've set up an outbound context that handles outbound NANPA calls, but I'm unable to set up another extension to forward calls elsewhere if they're in a 88XX format |
21:44.59 | DruRoland | basically, I want the pbx to outbound dial via a ITSP, but route over a different SIP provider for internal extensions between branch offices |
21:46.15 | voxter | anyone here use VQManage? |
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22:13.24 | LemensTS | i changed the ip of pbx.myserver.com ... is there anyway to make the devices reconnect to the new server? i could shut the old asterisk off or something |
22:16.52 | [TK]D-Fender | LemensTS: Perhaps you should check your device's respective manuals |
22:17.12 | [TK]D-Fender | LemensTS: Because no sever tells some random client to connect to it. That is backwards |
22:17.28 | pabelanger | If you are using DNS, it should resolve itself |
22:17.56 | Qwell | pabelanger: I see what you did there. |
22:19.46 | pabelanger | http://captionsearch.com/pix/2rub7eoeem.jpg |
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22:32.49 | DruRoland | Hmm, anyone gotten asterisk and televantage to play together? I'm trying to get the asterisk server to register as a SIP peer on the tv server |
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23:03.05 | [TK]D-Fender | DruRoland: Perhaps you could show us your actual problem |
23:03.08 | [TK]D-Fender | ~pb |
23:03.08 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood 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/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
23:03.09 | [TK]D-Fender | ^^^^^^^^^^^ |
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23:22.36 | DruRoland | http://pastebin.com/y9jCGsgG |
23:23.06 | [TK]D-Fender | DruRoland: Where is the FAILURE? |
23:23.27 | DruRoland | I haven't gotten far enough to have a failure yet |
23:23.37 | DruRoland | I'm still working on where to start |
23:23.50 | [TK]D-Fender | DruRoland: Then what have you shown me? |
23:24.02 | [TK]D-Fender | that isn't a "Start"? |
23:24.33 | DruRoland | True |
23:24.46 | [TK]D-Fender | DruRoland: I see no dialplan that would use your [televantage] peer. |
23:25.13 | DruRoland | yes, I'm not sure how to set that portion up |
23:25.14 | [TK]D-Fender | DruRoland: And that same peer points to [inbound] which you seem to show as being empty. |
23:25.37 | DruRoland | it is empty.. that's what flowroute has for their sample astrisk config |
23:25.47 | [TK]D-Fender | DruRoland: And you haven't told us which DIRECTION you are looking to work on |
23:25.59 | [TK]D-Fender | DruRoland: What does flowroute have to do with this? |
23:26.51 | DruRoland | I'm trying to send 88xx to the [televantage] peer, and all other numbers to the [flowroute] peer |
23:27.41 | DruRoland | so, to answer your previous question, this is all dealing with outbound calling |
23:28.06 | [TK]D-Fender | DruRoland: You have no dial() at all using that peer. |
23:28.12 | [TK]D-Fender | DruRoland: So go MAKE one. |
23:28.25 | [TK]D-Fender | dru you have failed to claify WHO is dialing the 88xx |
23:29.33 | DruRoland | The caller is a web app integrated with the asterisk server |
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23:36.54 | [TK]D-Fender | DruRoland: Which does not tell us how it communicates to * at all. Or what context it's calls fall into. |
23:38.58 | [TK]D-Fender | DruRoland: And you have no exten anywhere to match that patter, and none that use the [televantage] peer |
23:41.53 | LemensTS | what file has the cli output info for verbose, debug, warning, etc? |
23:41.58 | LemensTS | i thought it was console.conf but dont see it |
23:42.42 | LemensTS | wait nevermind its CONSOLE in logger.conf |
23:48.51 | [TK]D-Fender | BAI BAI |
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