| 00:00.17 | justinu|laptop | i have yes, but the power manage is so bad, and the wifi card gets even hotter |
| 00:00.21 | l-fy | file > also since ser is a mear sip proxy you can register users |
| 00:00.21 | justinu|laptop | so I don't use it often |
| 00:00.36 | justinu|laptop | everything works, it just runs hotter, and battery runs for 1/2 the time |
| 00:00.42 | l-fy | the usual way to use to use ser is to put it in front of an asterisk to protect it |
| 00:01.01 | justinu|laptop | the graphics system sucks... 1024x768 |
| 00:01.07 | file | l-fy: but that's not SER registering to another server... that's SER acting as a proxy, packet comes in packet goes out |
| 00:01.09 | justinu|laptop | but it's literally the size of a notebook |
| 00:01.15 | justinu|laptop | so I often travel without a charger or case |
| 00:01.17 | l-fy | file > you're right |
| 00:01.28 | l-fy | i'm gonna set it to forward the registration |
| 00:01.32 | mog | ? |
| 00:01.52 | purrdeta | where might I find a softphone that can connect to asterisk |
| 00:01.55 | l-fy | actually i need it just to convert my sip from udp to tcp and back |
| 00:02.04 | *** join/#asterisk DSpair (n=dphillip@74.133.167.232) |
| 00:02.14 | purrdeta | nvm |
| 00:02.15 | justinu|laptop | well, for day trips mog |
| 00:02.19 | justinu|laptop | not for overnight trips |
| 00:02.20 | l-fy | purrdeta > use yateclient, it works with h323, iax, and sip |
| 00:02.27 | mog | ahh |
| 00:02.36 | DSpair | Can someone recommend a VoIP service (i.e. Vonage, etc...) which is compatible with Asterisk? |
| 00:02.39 | justinu|laptop | i can get about 8h out of hte battery, which is often a full day of work |
| 00:02.48 | purrdeta | not vonage LOL |
| 00:02.55 | mog | is that the 4, 8 . or extended bat? |
| 00:03.01 | justinu|laptop | 8 cell |
| 00:03.10 | justinu|laptop | 4 is about 4h |
| 00:03.22 | justinu|laptop | i have both |
| 00:03.26 | mog | that is good to hear |
| 00:03.35 | mog | i am gonna get the 8 and the extended |
| 00:03.36 | justinu|laptop | if you're serious about ordering one, make sure you get tabook.pdf |
| 00:03.40 | justinu|laptop | if you don't have it already |
| 00:03.41 | mog | if / when i get it? |
| 00:03.48 | mog | tabook.pdf? |
| 00:04.02 | justinu|laptop | it's the de-facto ordering guide... tells you all features/specs of each thinkpad model |
| 00:04.08 | justinu|laptop | and there's lots of models |
| 00:04.13 | justinu|laptop | the website is useless |
| 00:04.25 | DSpair | The only two I have read about on-line are Bandwidth.com and AGN... Any other suggestions? |
| 00:04.27 | mog | true that |
| 00:04.39 | justinu|laptop | remind me tomorow and i'll get you a link if you can't find it |
| 00:04.55 | mog | i just did |
| 00:04.58 | mog | thanks though |
| 00:05.37 | mog | i wish i wasnt buying a house, otherwise id buy laptop tonight |
| 00:05.44 | justinu|laptop | i got 1704-69U model |
| 00:05.48 | justinu|laptop | x60s |
| 00:06.50 | justinu|laptop | anyways, the macbook is also quite nice |
| 00:07.00 | justinu|laptop | and finally runs cooler thanks to a fan firmware update |
| 00:07.57 | mog | ^_^ |
| 00:08.00 | mog | it hers |
| 00:08.05 | mog | she just has to run osx |
| 00:08.08 | justinu|laptop | its my wife's also |
| 00:08.10 | twisted[mobile] | hah |
| 00:08.17 | justinu|laptop | but i check it out every now and then |
| 00:08.47 | mog | only nonfree software in the house is on her computer |
| 00:09.01 | twisted[mobile] | that's bad |
| 00:09.12 | twisted[mobile] | that means she is a non-OSS supporter! |
| 00:09.17 | twisted[mobile] | bad!! |
| 00:09.24 | mog | she has loyalty to apple |
| 00:09.30 | twisted[mobile] | so do i |
| 00:09.34 | mog | althought she uses the linux boxen more |
| 00:09.40 | twisted[mobile] | just about everything on my powerbook however is free except the osx interface |
| 00:09.43 | mog | she still wants macosx for her main machine |
| 00:09.55 | mog | aka the majority |
| 00:09.56 | mog | ^_^ |
| 00:10.09 | mog | just messing with you |
| 00:10.33 | twisted[mobile] | and honestly, if it wasn't put together so nicely, i'd be running linux on it :P |
| 00:10.43 | twisted[mobile] | or straight darwin |
| 00:10.48 | twisted[mobile] | since it's got better hardware support |
| 00:11.01 | mog | linux runs just find on apple laptops now |
| 00:11.09 | justinu|laptop | os x on the intel core duo is fantastic |
| 00:11.13 | twisted[mobile] | linux ran just fine on older mac hardware too |
| 00:11.15 | l-fy | mog > i agree with you :) |
| 00:11.17 | mog | once the bcm43xx was reversed enginnered it was fine |
| 00:11.38 | l-fy | but than apple is slowly moving to yet another computer company |
| 00:12.00 | l-fy | file > help ? |
| 00:18.19 | mog | hey justinu|laptop how long have you had x60? |
| 00:18.58 | *** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 00:19.06 | justinu|laptop | since march |
| 00:19.49 | *** join/#asterisk bkw_ (n=bkw_@asterisk/friend-and-developer/bkw) |
| 00:20.27 | *** part/#asterisk bkw_ (n=bkw_@asterisk/friend-and-developer/bkw) |
| 00:21.59 | Cyt | Hi, I'm trying to use GotoIfTime but I guess I'm doing something wrong. I would like to direct the calls to the voicemail when it's after office hours. That's what Im doing http://pastebin.ca/174590 ! Could somebody give me a hand please? |
| 00:22.35 | twisted[mobile] | sorry, both of my hands are attached to arms. |
| 00:23.08 | Cyt | twisted[mobile]: I accept the arms too. hahaahah |
| 00:26.25 | Cyt | twisted[mobile]: could you take a look please? I guess Im missunderstanding something in that function |
| 00:27.41 | bungalow | hi all -- can anyone recommend tools for monitoring asterisk's performance/resource requirements? I've been using top, but a no load, and at high load where the audio starts disentigrating the top output is essentially the same... |
| 00:27.46 | [TK]D-Fender | Cyt : You running a night shift? |
| 00:28.33 | Cyt | TK: No! I would like to send the calls to the voicemail in the night shift |
| 00:29.20 | mog | bungalow, asterisk is great for doing that |
| 00:30.31 | [TK]D-Fender | Cyt : Your time was backwards, heres a better way |
| 00:30.37 | [TK]D-Fender | Cyt : exten => 344316501,1,GotoIfTime(9:00-16:59|*|*|*?2:3) |
| 00:31.51 | Cyt | [TK]D-Fender: hmm it's the other way around! I got! What 2:3 does? |
| 00:31.55 | [TK]D-Fender | eek, even pastebin.ca is damn slow today |
| 00:31.58 | bungalow | mog: humm.... is there anyway for me to tell when the audio is going to start breaking apart? just trying to figure out what my bottleneck is... couldn't get any clues from top... |
| 00:32.14 | [TK]D-Fender | Cyt : if it IS normal hours, goto step 2, otherwise step 3 |
| 00:32.28 | mog | you can listen to it |
| 00:32.30 | mog | chan_spy |
| 00:34.31 | bungalow | mog: ok... do you have any idea where asterisk is breaking down? how it's failing? I guess I'd like to know what I need to add more off to push the bounds of the system... |
| 00:34.49 | bungalow | ...add more of... |
| 00:35.10 | mog | it depends on what your doing |
| 00:36.16 | *** join/#asterisk lowlevel (n=Stuart@CPE0050ba71c82f-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 00:37.03 | bungalow | mog: my test script was running fake calls to generate conferences -- each conference has 2 SIP, 1 IAX... got to about 50 conferences when the audio started breaking apart... get's steadily worse |
| 00:37.16 | Cyt | [TK]D-Fender: Asterisk has an own clock or does it use the system clock? I just added your code to my dial plan and its 19:36 on the system and the call when thru... |
| 00:37.30 | mog | what are you using for timing, and what kind of machine |
| 00:38.08 | bungalow | mog: the whole time the CPU was about 95% idle, and the other stats basically pegged at 0 --- I'm using AMD Opteron 248 |
| 00:38.53 | JT | bungalow: what are you using for timing? |
| 00:38.55 | bungalow | mog: app_conference, show no zaptel timing --- but even outside of a conference the audio was disintegrating |
| 00:39.05 | bungalow | so no zaptel... |
| 00:39.22 | mog | cant support app_conference |
| 00:39.28 | mog | i only use stuff in main tree |
| 00:39.41 | mog | you should be able to get many more channels than that with app_meetme |
| 00:39.51 | *** join/#asterisk rokerr (i=rowan@CPE00095bdde585-CM0011ae91b176.cpe.net.cable.rogers.com) |
| 00:40.50 | bungalow | mog: not sure if that's the problem... again the audio degradation was the same in or out... |
| 00:41.50 | mog | understandable |
| 00:41.56 | mog | im just telling you what i know |
| 00:42.06 | mog | app_meetme could handle 300 users or so on a box like that |
| 00:42.10 | mog | possibly many more |
| 00:43.16 | bungalow | mog: wondering if I've got something configured wrong... app_conference, from what I've heard and read is supposed to over good performance, as well... |
| 00:43.56 | Cyt | [TK]D-Fender: is this right? http://paste-bin.com/516 (paste-bin.com it's faster) |
| 00:44.09 | mog | like i said before i only use things in the main tree so i dont know what problems you might be experiencing with it |
| 00:44.46 | file | Nugget: you slacker you2 |
| 00:44.49 | file | ! |
| 00:45.00 | [TK]D-Fender | Cyt : now pastebin the CLI output of your call attempts |
| 00:45.42 | rokerr | well the only digium canadian distrib apparently doesn't want my business. heh |
| 00:45.43 | bungalow | mog: ok, no problem... appreciate your help... |
| 00:45.43 | [TK]D-Fender | Polycom "auto answer" w/o ringing is EVIL! |
| 00:46.24 | [TK]D-Fender | And evenworse... calling with the MIC muted... = fre spying! |
| 00:46.40 | [TK]D-Fender | rokerr : Who? And what were you looking to buy? |
| 00:47.07 | Cyt | [TK]D-Fender: http://paste-bin.com/517 |
| 00:47.10 | rokerr | [TK]D-Fender, Williams global. I want the rtdm11b developer kit |
| 00:47.30 | rokerr | [TK]D-Fender, I guess I could just call them tomorrow.. but their website's not very useful |
| 00:47.31 | [TK]D-Fender | rokerr : Just 1? A normal consumer purchase? |
| 00:47.51 | [TK]D-Fender | rokerr : Williams is a wholesaler. they don't deal with the public. |
| 00:48.06 | [TK]D-Fender | rokerr : there are PLENTY of places you can get Digium cards from in Canada |
| 00:48.08 | rokerr | [TK]D-Fender, well Digium only sells those kits to distributors so..... |
| 00:48.29 | bungalow | JT: any ideas? |
| 00:48.29 | [TK]D-Fender | rokerr : thats just the 1FXO/FXS card right? |
| 00:48.33 | rokerr | [TK]D-Fender, yeah |
| 00:48.40 | [TK]D-Fender | rokerr : BS, you can get that anywhere. |
| 00:48.47 | JT | bungalow: does it use zt_dummy for timing? |
| 00:49.10 | bungalow | JT: no, it doesn't use an external timing source |
| 00:49.20 | [TK]D-Fender | rokerr: exten => 344316501,1,GotoIfTime(9:00-16:59|*|*|*?2:3) |
| 00:49.21 | JT | dummy....... |
| 00:49.25 | [TK]D-Fender | rokerr : exten => 344316501,1,GotoIfTime(9:00-16:59|*|*|*?2:3) |
| 00:49.31 | [TK]D-Fender | http://voipware.ca/en-ca/product_info.php?cPath=24&products_id=72 |
| 00:49.46 | [TK]D-Fender | DANGIT... Firefox was ignoring my Copies |
| 00:49.52 | [TK]D-Fender | 3rd times the charm! |
| 00:49.58 | rokerr | [TK]D-Fender, "anywhere" being a normal computer shop in torono? :) |
| 00:50.08 | rokerr | [TK]D-Fender, ahh good ol' firefox copy&paste |
| 00:50.24 | [TK]D-Fender | rokerr : What "normal' computer shop carries ANYTHING voip? |
| 00:50.25 | rokerr | [TK]D-Fender, thanks :) |
| 00:50.41 | rokerr | [TK]D-Fender, I don't know .. haven't shopped around for the gear before |
| 00:51.11 | [TK]D-Fender | rokerr : Nobody will stock it becuase it won't sell off the shelf. |
| 00:51.30 | JT | [TK]D-Fender: most "normal" computer shops carry something skype or linksys type boxes :P |
| 00:51.37 | rokerr | [TK]D-Fender, makes sense. same with nearly anything worth buying in Canada ;) |
| 00:51.38 | [TK]D-Fender | rokerr : But for items in normal distribution a computer store could easily purchase. |
| 00:51.39 | Cyt | [TK]D-Fender: was that for me? |
| 00:52.10 | [TK]D-Fender | JT : Yeah. Cheap-ass consumer-only junk. We are more like the pro-sumer class as its called. |
| 00:52.19 | [TK]D-Fender | cyt : no, paste error |
| 00:52.33 | Cyt | [TK]D-Fender: oh ok hahahaa |
| 00:52.59 | Cyt | [TK]D-Fender: I pasted the output here: http://paste-bin.com/517 |
| 00:53.03 | [TK]D-Fender | Cyt : look at your 2 pastebins. they don't match! |
| 00:53.20 | [TK]D-Fender | Cyt : the order of 3:2 vs 2:3 |
| 00:53.31 | l-fy | too many protocols make our life miserable |
| 00:53.38 | [TK]D-Fender | l-fy : You mean in general I take it, not "in here" |
| 00:53.51 | Cyt | [TK]D-Fender: oh I changed just to check if that would matter! but in both ways it goes straight to the call |
| 00:54.08 | l-fy | [TK]D-Fender > in general of course |
| 00:54.08 | wunderkin | the te110p should light up as soon as you turn on the pc right? like the 4 port card? |
| 00:54.18 | l-fy | i douby asterisk users are different than the others |
| 00:54.25 | l-fy | is just a statistics |
| 00:54.33 | l-fy | unrelated to software |
| 00:55.11 | l-fy | and anyway [TK]D-Fender after enjoying some ser i consider asterisk decent |
| 00:55.23 | rokerr | woah, voipware.ca is pretty sketchy |
| 00:55.41 | [TK]D-Fender | rokerr : Check the WIKi for a list of retailers |
| 00:55.53 | rokerr | http://www.voip-info.org/wiki/view/Asterisk+Canadian+Users |
| 00:57.35 | *** join/#asterisk IronMan2000 (n=kent@65.124.236.24) |
| 00:57.54 | Cyt | [TK]D-Fender: http://paste-bin.com/518 this is the original! but does the samething |
| 00:58.00 | [TK]D-Fender | Cyt : My bad.. I forget that gotoiftime does NOT have an "else" clause |
| 00:59.10 | Cyt | [TK]D-Fender: no problem! heheh so, it will do only like "go to priority 3" |
| 00:59.38 | *** join/#asterisk Amilcar_ (n=Email@201-25-83-240.cpece705.dsl.brasiltelecom.net.br) |
| 01:00.47 | [TK]D-Fender | Cyt : http://paste-bin.com/519 |
| 01:04.17 | shodan | any small business friendly wholesalers in the toronto/markham aera (that don't charge more than online retail stores o_O)? |
| 01:04.53 | rokerr | shodan, I was just looking at Syonex's site which is linked from the wiki |
| 01:05.40 | JT | if you don't want to pay more than an online store, why not use an online store? |
| 01:06.02 | *** join/#asterisk tengulre (n=tengulre@221.11.5.180) |
| 01:06.36 | shodan | I mean , wholesaler should cost less than online stores otherwise they're just retail stores with shitty service ;) |
| 01:07.05 | Cyt | [TK]D-Fender: thank you very much for your help once again! I really appreciate the help you've been giving me |
| 01:07.25 | [TK]D-Fender | Cyt :P Functional now? |
| 01:07.56 | Cyt | [TK]D-Fender: yes!!! Thank you, really. |
| 01:08.06 | JT | unless you're a reseller, why would a wholesaler sell to you, shodan? |
| 01:08.18 | [TK]D-Fender | Cyt : Quite welcome |
| 01:08.57 | shodan | because I'm low maintenance like a reseller , I buy all my computer gear from wholesale |
| 01:12.50 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 01:16.26 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 01:18.59 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.ITB.ac.id) |
| 01:33.45 | *** join/#asterisk tuck3r (n=tuck3r@unaffiliated/tuck3r) |
| 01:34.45 | *** join/#asterisk tengulre11 (n=tengulre@222.90.66.156) |
| 01:34.48 | tengulre11 | hi,all |
| 01:34.54 | tengulre11 | good morning! |
| 01:35.50 | *** join/#asterisk ESCulapio_ (n=The@230stb68.codetel.net.do) |
| 01:39.49 | *** join/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com) |
| 01:41.05 | shmaltz | tengulre11, good evening |
| 01:41.47 | tengulre11 | make a good dream! |
| 01:48.18 | Cyt | [TK]D-Fender: To play a message to the caller before go to the voicemail, should I use Playback function? |
| 01:54.32 | shmaltz | Cyt, or you could use Background |
| 01:55.34 | Cyt | shmaltz: perfect, let me try! Thank you!!! |
| 01:59.57 | tuck3r | what would be the proper way to do this? http://pastebin.ca/174608 so I could run commands after a my dial commands that only match one string |
| 02:00.44 | [TK]D-Fender | Cyt :what kind of message? You normally call Voicemail with "b" or"u" to playback either the busy or Unavailable message |
| 02:01.33 | [TK]D-Fender | tuck3r : Your 1st 3 lines dial nothin, and those "s" lines Do nothing. they will never be called. |
| 02:02.06 | tuck3r | [TK]D-Fender: yeah, i took the juicy info out ;) |
| 02:02.28 | [TK]D-Fender | tuck3r : yes, now its ripe for comment.... |
| 02:03.05 | tuck3r | but would that even work at all? |
| 02:03.27 | [TK]D-Fender | tuck3r : Well it does NOTHING.... |
| 02:03.38 | [TK]D-Fender | tuck3r : Completely broken comes to mind... |
| 02:03.58 | tuck3r | i will post my real code... |
| 02:04.58 | [TK]D-Fender | tuck3r : if you want proper commenty you'd be advised to.... |
| 02:05.27 | tuck3r | http://pastebin.ca/174611 |
| 02:05.46 | Cyt | tk: sorry! Ham... I have one voicemail message for busy, one for available and I would like a 3th one for "office closed". Is it possible? |
| 02:05.56 | Cyt | [TK]D-Fender: sorry! Ham... I have one voicemail message for busy, one for available and I would like a 3th one for "office closed". Is it possible? |
| 02:06.57 | [TK]D-Fender | Cyt : What is the goal of this box? |
| 02:07.13 | Cyt | [TK]D-Fender: what do you mean? |
| 02:07.19 | [TK]D-Fender | Cyt : the only way to do a third is to do it outside of the VM system. |
| 02:07.49 | [TK]D-Fender | Cyt : Why messages would you play before ending up in this box? What is its full purpose and when/why would you send a call there? |
| 02:07.50 | Cyt | [TK]D-Fender: yes. thats what I was thinking. Use background to play the message and then voicemail(s101).... |
| 02:08.31 | [TK]D-Fender | Cyt : You'd use Playback in that case, not Background. That is a valid idea if you really need 3 distinct messages |
| 02:08.57 | Cyt | [TK]D-Fender: I'm creating this for my house, so, I would have the unavailable (sorry, im not here), the busy (im on the phone, call back latter) and im sleeping now, call in the morning. |
| 02:09.20 | Cyt | [TK]D-Fender: so I would make playback say the message and then goes to voicemail(s..) |
| 02:10.12 | [TK]D-Fender | Cyt : jsut a tought, but I wouldn't want to presonally advertise that I'm on the phone. It makes it hard to ignore people without them knowing it :) |
| 02:11.15 | Cyt | [TK]D-Fender: Its just a test on a SIP line, just to learn asterisk... people would call my regular phone, not this one heheheh |
| 02:11.44 | IronMan2000 | e |
| 02:13.02 | [TK]D-Fender | Cyt : Like I said your methos is the right way to add a third and perhaps do all 3 and forgo the interal recordings completely. |
| 02:13.17 | [TK]D-Fender | Cyt : Only thing is you lose the easy tool to change those recordings. |
| 02:15.05 | Cyt | [TK]D-Fender: yes, I fully agree with you |
| 02:16.59 | *** join/#asterisk AndyCap (n=aoy@pdpc/supporter/sustaining/AndyCap) |
| 02:20.46 | *** join/#asterisk `Sauron (i=sauron@h-69-3-12-50.hstqtx02.covad.net) |
| 02:22.08 | Cyt | [TK]D-Fender: I will get some sleep, thank you very much for you help! cya |
| 02:22.21 | [TK]D-Fender | tuck3r : What calls "s" in there? Next, why does your first dial use an IP and the others a named device? |
| 02:22.29 | [TK]D-Fender | Cyt : np, good night |
| 02:23.01 | orlock | [TK]D-Fender: the incoming s context is actually hardcoded |
| 02:23.08 | tuck3r | [TK]D-Fender: the named device way decided to break on me |
| 02:24.16 | orlock | [TK]D-Fender: i had to read the source to figure out where "s" was coming form |
| 02:24.43 | tuck3r | C source makes my brain hurt |
| 02:26.33 | mog | whats wrong tuck3r |
| 02:26.37 | [TK]D-Fender | orlock : noting in that context hints as to where/why "s" would be called... |
| 02:26.55 | [TK]D-Fender | tuck3r : But I guess the question is, yeah, whats the matter? |
| 02:27.35 | [TK]D-Fender | orlock : And "s" is an * standard EXTENSION, not context. |
| 02:27.50 | tuck3r | if I have dial commands then "exten => s,n,Foo" will they be called? |
| 02:28.18 | [TK]D-Fender | tuck3r : Not THOSE. Those are not the next priority of the pattern match that starts them. |
| 02:29.27 | tuck3r | http://pastebin.ca/174625 |
| 02:29.47 | tuck3r | er shit |
| 02:30.03 | tuck3r | forgot to blank it all out :| |
| 02:30.48 | [TK]D-Fender | tuck3r : So far "s" is never called in there.... |
| 02:32.33 | tuck3r | is using Goto(s,1) the best way to do what I'm trying to do? |
| 02:32.39 | orlock | [TK]D-Fender: ahh, yeah, extension, not context, sorry |
| 02:33.06 | orlock | [TK]D-Fender: and the "s" extension is the default one.. in the source. |
| 02:33.56 | [TK]D-Fender | tuck3r : Here's a much nicer version for you - http://pastebin.ca/174628 |
| 02:34.18 | [TK]D-Fender | orlock : Yes, but what CALLS it in there? NOTHING. |
| 02:34.46 | [TK]D-Fender | tuck3r : I still have no idea WHAT you're trying to do yet. Let me re-read |
| 02:35.19 | [TK]D-Fender | tuck3r : What are in those "macros" you are calling? |
| 02:35.39 | tuck3r | they just playback a error and hangup |
| 02:35.40 | [TK]D-Fender | tuck3r : I think you should be showing more so we get a better idea of what you're trying to accomplish. |
| 02:35.55 | [TK]D-Fender | tuck3r : And do mask the private info. |
| 02:36.07 | tuck3r | yeah... oops |
| 02:36.18 | [TK]D-Fender | tuck3r : And take a close look at my rewrite |
| 02:37.43 | tuck3r | [TK]D-Fender: if the dial fails your code just fails |
| 02:38.05 | orlock | [TK]D-Fender: its the default sip extension |
| 02:38.12 | [TK]D-Fender | tuck3r : Correct, but your code never called "s" so there was no point to it. |
| 02:38.35 | [TK]D-Fender | tuck3r : I jsut trimmed what you di, and now we can add to it. Please show everything that gets called in your setup |
| 02:39.11 | wunderkin | exten => s,1,TakeOver(world) |
| 02:41.19 | tuck3r | http://pastebin.ca/174631 |
| 02:41.31 | tuck3r | ignore the commented out parts |
| 02:45.20 | IronMan2000 | w |
| 02:45.50 | [TK]D-Fender | tuck3r : basically those little macro's of yours only get called by one thing, right? |
| 02:46.31 | [TK]D-Fender | IronMan2000 : Are you capable of more than single-letter statements? :) |
| 02:46.40 | tuck3r | [TK]D-Fender: yeah, pointless. thats why I'm rewriting my outbound code or trying anyways |
| 02:52.25 | [TK]D-Fender | tuck3r : Hold one, working on it |
| 02:54.31 | [TK]D-Fender | tuck3r : You're lcuky I'm in such a generous mood today : http://pastebin.ca/174640 |
| 02:55.29 | tuck3r | [TK]D-Fender: ooh.. thanks :) |
| 02:55.51 | [TK]D-Fender | tuck3r : You need to keep an eye out for redundancy. |
| 02:56.02 | benjk | sounds painful |
| 02:56.17 | benjk | and not redundant at al either |
| 02:56.48 | benjk | unless you got 3 eyes, of course |
| 02:58.05 | tuck3r | [TK]D-Fender: your code is so clean :) |
| 02:58.29 | *** join/#asterisk Defraz (n=t0tal@fw.centrisys.com) |
| 03:00.00 | [TK]D-Fender | tuck3r : Its as much as it needs to be, nothing more. Abstract. |
| 03:00.31 | tuck3r | i just need to use the right amount of macros |
| 03:00.47 | [TK]D-Fender | tuck3r : when you have an outbound trunk, always set up a macro to dial it, NOT a pattern match. Use your pattern match to call that macro instead so that many patterns can taqke advantage of it. |
| 03:01.15 | [TK]D-Fender | tuck3r : Just look at everything that is duplicated and ask yourself what can/should be reduced. |
| 03:01.54 | [TK]D-Fender | I brought those little macro's into the main dial because NOTHING else called them. Being macros only added bloat. |
| 03:02.22 | [TK]D-Fender | tuck3r : and the "Deny" context was plenty obvious. Many matches to a single outcome. |
| 03:05.21 | IronMan2000 | does anyone else use 1-800-373-3411 to offer directory service to their customers? I have it when you dial 1411 or 411 it connects to 18003733411 to give customers dir assistance. |
| 03:06.36 | IronMan2000 | works pretty good. Just hope it does not create telemarketer calls to our customers. |
| 03:07.12 | tuck3r | ooh, interesting |
| 03:08.02 | IronMan2000 | it works, and it's free. |
| 03:09.01 | IronMan2000 | anyone here using Asterfax? I am thinking of setting it up. I really need a faxing solution. |
| 03:12.47 | [TK]D-Fender | IronMan2000 : Maybe better served by Hylafax. |
| 03:13.08 | [TK]D-Fender | e-mail -> fax is a kludgy idea. |
| 03:13.18 | [TK]D-Fender | the reverse is effecient however |
| 03:13.35 | IronMan2000 | losing business because of no good fax capabilities |
| 03:13.42 | *** join/#asterisk ahattar (n=aha@ool-43551487.dyn.optonline.net) |
| 03:13.53 | tuck3r | http://www.youtube.com/watch?v=PFyXdrpuZrE |
| 03:13.57 | tuck3r | ^^ lol |
| 03:17.21 | *** join/#asterisk Trazz (i=Trazz@c-71-57-36-106.hsd1.il.comcast.net) |
| 03:18.33 | *** part/#asterisk ahattar (n=aha@ool-43551487.dyn.optonline.net) |
| 03:19.50 | tengulre11 | tuck3r: lol, I very like it!! |
| 03:20.34 | [TK]D-Fender | Indeed.... cute |
| 03:21.00 | tengulre11 | :( |
| 03:21.09 | tuck3r | didn't really like the music video but the song is great |
| 03:22.25 | tengulre11 | http://www.megacomm.com.cn/music/001.mp3 |
| 03:24.00 | tuck3r | the great firewall doesn't block inbound connections? |
| 03:25.11 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 03:27.58 | tengulre11 | tuck3r: really ? the site in china. |
| 03:28.31 | tuck3r | no idea, I just hear a lot about the great firewall. it worked fine for me |
| 03:28.47 | tengulre11 | tuck3r: which IM are u using? |
| 03:29.07 | tuck3r | IRC client? or IM client? |
| 03:29.19 | tengulre11 | IM client! |
| 03:30.15 | tuck3r | uh, MSN, AIM, Yahoo, Gtalk, Jabber, .mac chat. think thats all |
| 03:30.25 | Trazz | anyone good with yum and errors ? |
| 03:30.56 | tengulre11 | :) |
| 03:31.17 | Trazz | i am getting some zaptel depency issues |
| 03:31.57 | Trazz | dependency issues |
| 03:32.07 | [TK]D-Fender | Trazz : Use the Source Luke! |
| 03:40.39 | *** join/#asterisk misc-- (n=blah@203.215.117.166) |
| 03:44.27 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 03:46.06 | *** join/#asterisk ManxPower (n=ManxPwer@71-8-11-111.dhcp.leds.al.charter.com) |
| 03:50.36 | *** join/#asterisk dmz (n=dmz@64.20.133.243) |
| 03:50.58 | dmz | hey y'all anyone here have a multi-menu ivr setup with asterisk? |
| 03:52.15 | dmz | for some reason, even though I have exten => i & t setup when it finishes playing (Background) it "Auto fallthrough"s out of the context and hangs up |
| 03:52.25 | dmz | my "i" & "t" exten settings are to goto the top of the menu again |
| 03:52.51 | dmz | I have set digittimeout to 5 & response timeout to 10 but as soon as it finishes playing Background(file) it drops right through and doesn't give any time to enter digits |
| 03:52.54 | dmz | any suggestions? |
| 03:53.18 | dmz | and what's up with topic, did something happen to rob?? |
| 03:53.30 | dmz | anyone alive? |
| 03:53.41 | [TK]D-Fender | dmz : You need to set "autofallthrough=no" in [general] |
| 03:54.04 | [TK]D-Fender | dmz : this is because of the creation of WaitExten in * 1.2 |
| 03:54.05 | *** join/#asterisk [RUsh] (n=nobody@212.57.157.11) |
| 03:54.06 | dmz | ah, let me try that |
| 03:54.34 | shodan | mahahahah weird al that never gets old ! :) |
| 03:54.51 | Frogdude | Weird Al's greatest hits was my first casette |
| 03:54.52 | Frogdude | :) |
| 03:55.20 | dmz | anyone know ref to lilo? last I spoke with him was about 4 months ago. is he ok? |
| 03:56.06 | dmz | D-Fender, thanks! |
| 03:56.23 | Frogdude | (14:25:34) christel: (notice) [Global Notice] On the 12th September Rob Levin, known to many as Freenode's lilo, was hit by a car while riding his bike. He suffered head injuries and passed away in hospital on the 16th. For more information please visit #freenode-announce |
| 03:56.25 | [TK]D-Fender | dmz : Quite welcome |
| 03:56.42 | shodan | "I'm fluent in javascript as well as klingon" mahhaah |
| 03:56.43 | dmz | one other question, can i keep track of how many times someone has looped through a menu so I can disconnect them on the 3rd or 4th time through the menu |
| 03:56.51 | dmz | Egad, dang that's sad |
| 03:56.55 | [TK]D-Fender | dmz : 1 sec |
| 03:57.08 | Frogdude | yeah |
| 03:57.23 | dmz | Frogdude, thanks :( |
| 03:57.34 | Frogdude | np |
| 03:58.17 | dmz | i remember sitting next to rob @ balug meetings before he left the bay area |
| 03:58.39 | Frogdude | I never knew him but it definitely is not a happy story |
| 03:58.56 | *** part/#asterisk jbroomeLUG (n=jbroome@dargo.trilug.org) |
| 03:59.16 | [TK]D-Fender | dmz : here's a sample : http://pastebin.ca/174698 |
| 04:00.10 | *** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg) |
| 04:00.11 | Frogdude | I'm using phpagi and am having some trouble originating a call |
| 04:00.16 | Frogdude | it connects to asterisk and disconnects fine |
| 04:00.54 | wunderkin | tuck3r, roflol ;) heheheh just watched the video |
| 04:00.55 | Frogdude | but when try to send request Originate, nothing shows up in the window or happens |
| 04:01.10 | tuck3r | wunderkin: heh |
| 04:01.12 | Frogdude | does anyone have a suggestion for what to look for? |
| 04:02.42 | Frogdude | it's working fine with a .call file with the same channel and extension but not with phpagi originate |
| 04:03.10 | dmz | D-Fender, danka!!! |
| 04:04.15 | *** join/#asterisk `Sauron (i=sauron@h-69-3-12-50.hstqtx02.covad.net) |
| 04:04.35 | dmz | D-Fender, nice! and it won't take much editing to fit it into my context setup |
| 04:04.53 | [TK]D-Fender | dmz : I know, very cut & paste worthy |
| 04:22.33 | *** join/#asterisk Phoenix^ (n=Phoenix@phoenixphire.org) |
| 04:22.48 | Phoenix^ | Hey all |
| 04:24.25 | Phoenix^ | I was wondering what card I would need for an asterisks server to allow me to connect to a POTS. I want to play with VoIP internally and use Astericks to go out via a POTS |
| 04:24.57 | [TK]D-Fender | Phoenix^ : X100P, TDM400P, A200,SPA-3000,SPA-3102. take your pick |
| 04:25.17 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 04:25.17 | *** mode/#asterisk [+o denon] by ChanServ |
| 04:26.43 | *** join/#asterisk DrkShdw (n=scorpio@unaffiliated/drkshdw) |
| 04:26.52 | Phoenix^ | thanks |
| 04:35.48 | *** join/#asterisk tengulre (n=tengulre@222.90.66.156) |
| 04:37.51 | CunningPike | Phoenix^: I would recommend an external gateway instead of a card - something like an SPA-3000 |
| 04:43.09 | [TK]D-Fender | Alright.... checkot time... later all |
| 04:43.24 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
| 04:47.46 | *** join/#asterisk DaPrivateer (i=Privatee@crimson.66fruit.com) |
| 04:57.32 | *** join/#asterisk Sakimustafa (n=yusuf@202.133.14.226) |
| 04:57.53 | Sakimustafa | hello |
| 04:58.09 | Sakimustafa | Good morning |
| 04:58.40 | Sakimustafa | Which distro of linux perfect for Asterisk |
| 04:59.06 | tuck3r | what ever you like |
| 04:59.22 | tuck3r | the best distro is the one you know... |
| 04:59.45 | Sakimustafa | RedHat 9.0 Or Fedora? |
| 04:59.57 | tuck3r | uh, RH9 is really old |
| 05:00.01 | tuck3r | so fedora |
| 05:02.13 | Sakimustafa | For test purpose Without digium or any card how I would deploy Asterisk? What I need for? |
| 05:02.33 | tuck3r | you don't need a zaptel card... |
| 05:02.36 | JT | voip? |
| 05:02.50 | Sakimustafa | Yes VOIP |
| 05:03.16 | tengulre | I like debian. |
| 05:03.39 | tuck3r | Sakimustafa: I know you read voip-info and TFOT before asking that question |
| 05:03.41 | Sakimustafa | With softphone and Asterisk will suffice? |
| 05:03.52 | JT | yes |
| 05:04.05 | rokerr | tuck3r, or centos if you want to use a redhat enterprise type distro |
| 05:04.25 | rokerr | or, the rpath asterisk virtual machine |
| 05:05.02 | tuck3r | Sakimustafa: http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 05:05.09 | tuck3r | ^^ read that whole book |
| 05:05.31 | stephane_ | jour |
| 05:05.52 | Sakimustafa | rokerr pls explain me Asterisk virtual machine.... |
| 05:06.16 | tuck3r | Sakimustafa: http://www.voip-info.org/wiki/view/RPath+-+Asterisk+Appliances |
| 05:06.32 | tuck3r | Please search google/voip-info |
| 05:09.30 | *** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.wa.comcast.net) |
| 05:09.51 | aao_pwner | im an asterisk noob and erm when i try to compile from the soruce it says this.. |
| 05:09.57 | aao_pwner | "Makefile", line 28: Missing dependency operator |
| 05:09.57 | aao_pwner | "Makefile", line 32: Need an operator |
| 05:09.57 | aao_pwner | "Makefile", line 35: Need an operator |
| 05:10.01 | aao_pwner | Error expanding embedded variable. |
| 05:10.17 | tuck3r | 1.2.12.1? |
| 05:10.25 | aao_pwner | yes |
| 05:10.41 | tuck3r | and you just ran make? |
| 05:10.42 | aao_pwner | [root@aaopwner /hd1/asterisk/asterisk-1.2.12.1]# make install |
| 05:10.44 | aao_pwner | yes |
| 05:10.47 | aao_pwner | oh wait |
| 05:10.48 | aao_pwner | ill try make |
| 05:10.49 | aao_pwner | lol |
| 05:10.52 | tuck3r | you already run make? |
| 05:10.53 | aao_pwner | does the same thing |
| 05:10.55 | tuck3r | yeah ;) |
| 05:10.55 | aao_pwner | yes |
| 05:11.05 | hads | What version of make? |
| 05:11.17 | aao_pwner | how can i tell... |
| 05:11.30 | hads | make -v |
| 05:11.47 | aao_pwner | well i have to do which make first |
| 05:11.49 | aao_pwner | lol |
| 05:11.57 | aao_pwner | errr |
| 05:11.59 | aao_pwner | didnt work |
| 05:12.02 | aao_pwner | http://pastebin.ca/174755 |
| 05:12.04 | aao_pwner | i got that though |
| 05:12.27 | tuck3r | bsd/solaris? |
| 05:12.42 | aao_pwner | [root@aaopwner /hd1/asterisk/asterisk-1.2.12.1]# uname -a |
| 05:12.42 | aao_pwner | FreeBSD aaopwner.com 6.1-BETA1 FreeBSD 6.1-BETA1 #0: Wed Feb 1 21:59:39 UTC 2006 root@opus.cse.buffalo.edu:/usr/obj/usr/src/sys/GENERIC i386 |
| 05:12.42 | aao_pwner | [root@aaopwner /hd1/asterisk/asterisk-1.2.12.1]# |
| 05:12.59 | tuck3r | use the BSD port |
| 05:13.10 | aao_pwner | i did |
| 05:13.11 | aao_pwner | and it erm.. |
| 05:13.14 | aao_pwner | screws around with stuff |
| 05:13.17 | aao_pwner | let me get you a lionk |
| 05:13.23 | aao_pwner | i probably spent the last hour trying to fix it with no luck |
| 05:13.32 | aao_pwner | http://pastebin.ca/174740 |
| 05:13.55 | tuck3r | zaptel-1.0 depends on shared library: newt.51 - not found |
| 05:13.59 | tuck3r | ^^^^ |
| 05:14.23 | aao_pwner | scroll down |
| 05:14.37 | aao_pwner | i try to install newt |
| 05:14.41 | aao_pwner | and it goess all buzzurk |
| 05:14.41 | aao_pwner | lol |
| 05:15.09 | tuck3r | well you should fix that.... |
| 05:15.19 | aao_pwner | thats what i tried to do |
| 05:15.35 | tuck3r | meh, use linux :P |
| 05:17.27 | aao_pwner | :( |
| 05:17.34 | aao_pwner | bsd is leet though lol |
| 05:18.00 | hads | The OS is as leet as you ;) |
| 05:18.06 | tuck3r | it's not leet if you can't use it |
| 05:18.45 | hads | You will find a lot more help here for Asterisk on Linux |
| 05:20.12 | aao_pwner | well...errr |
| 05:21.05 | *** join/#asterisk kumamoto (n=eryco@24-178-2-212.dhcp.ftwo.tx.charter.com) |
| 05:21.14 | rokerr | why are you running a beta and not release of 6.1? |
| 05:22.59 | aao_pwner | it said it was stable on the site |
| 05:23.02 | aao_pwner | i have no idea... |
| 05:23.07 | aao_pwner | it said 6.1 stablwe |
| 05:23.09 | aao_pwner | stable* |
| 05:23.14 | aao_pwner | but apperentally this is the beat |
| 05:23.15 | aao_pwner | beta* |
| 05:23.26 | aao_pwner | im gonna get 6.2 when it comes out (on my birthday :) ) |
| 05:27.58 | *** part/#asterisk Phoenix^ (n=Phoenix@phoenixphire.org) |
| 05:41.26 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 05:41.32 | blitzrage | anyone notice trunk doesn't build right now? |
| 05:50.03 | *** join/#asterisk tengulre (n=tengulre@222.90.66.4) |
| 05:51.59 | *** join/#asterisk parag_ast (n=parag_as@dxb-b18839.alshamil.net.ae) |
| 05:58.17 | *** join/#asterisk Rahail (n=dd@ip66-104-51-162.z51-104-66.customer.algx.net) |
| 05:59.38 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com) |
| 06:02.50 | *** join/#asterisk nn (n=joseph@pool-72-65-155-239.chrlwv.east.verizon.net) |
| 06:03.01 | nn | anyone do toll-free termination for 888 #'s? |
| 06:03.14 | nn | or would it be origination? :) |
| 06:03.36 | tuck3r | want to be caller or call out? |
| 06:03.52 | nn | callee |
| 06:04.16 | tuck3r | er it's late :P |
| 06:04.27 | *** join/#asterisk h3x0r (n=hex@ip68-224-236-92.lv.lv.cox.net) |
| 06:04.32 | nn | thats quite alright |
| 06:04.39 | nn | basically, i have a specific 888 # i want :) |
| 06:04.47 | tuck3r | origination= inbound termination= outbound |
| 06:05.01 | nn | kk |
| 06:05.11 | *** join/#asterisk teknoprep (n=teknopre@unaffiliated/teknoprep) |
| 06:05.19 | *** part/#asterisk hohum (n=dcorbe@69-175-203-11.chvlva.adelphia.net) |
| 06:06.52 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 06:07.14 | nn | tuck3r: any suggestions? |
| 06:07.31 | tuck3r | uh, iaxtel does them |
| 06:07.38 | tuck3r | or whatever there called now |
| 06:07.44 | tuck3r | their* |
| 06:07.52 | nn | and i can pick the #? |
| 06:07.59 | tuck3r | IIRC, yes |
| 06:07.59 | nn | pay via paypal?:) |
| 06:08.09 | CunningPike | tuck3r: they're ;) |
| 06:08.25 | tuck3r | CunningPike: you win :) |
| 06:08.28 | CunningPike | :D |
| 06:08.35 | tuck3r | nn: again I think so |
| 06:08.38 | h3x0r | i think more specifically termination and origination is talking about the perspective of whos paying for the call |
| 06:09.02 | nn | h3x0r: hell, let the bastard calling me pay:) |
| 06:09.16 | tuck3r | nn: not with 1800 #s |
| 06:09.19 | nn | ~ |
| 06:09.22 | nn | i know :( |
| 06:09.29 | nn | maybe my office needs a 9xx |
| 06:09.36 | tuck3r | hah |
| 06:10.06 | aao_pwner | my office has xxx |
| 06:10.07 | nn | i think that'll work for customer service |
| 06:10.10 | aao_pwner | but i dont know about an 9xx |
| 06:10.15 | nn | aao_pwner: we have XXX too! :) |
| 06:10.18 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 06:10.18 | *** mode/#asterisk [+o mog] by ChanServ |
| 06:10.21 | nn | a big selection, in fact. |
| 06:10.34 | nn | Toys, movies, anything XXX you could ever want. If we don't have it, we can get it :) |
| 06:11.04 | aao_pwner | nn, you have gay midget porn? |
| 06:11.08 | aao_pwner | in your office? |
| 06:11.09 | nn | aao_pwner: That's on isle 7 |
| 06:11.14 | aao_pwner | oh, ok |
| 06:11.37 | nn | Mike says he recommends the movie "I'm short, not small" |
| 06:12.06 | CunningPike | nn: aisle |
| 06:12.12 | CunningPike | Sorry - I'm a pedant |
| 06:12.24 | nn | CunningPike: no, they're floating above water :) |
| 06:14.21 | *** join/#asterisk af_ (n=af@ip-143-59.sn1.eutelia.it) |
| 06:14.56 | CunningPike | nn: heh |
| 06:16.30 | *** join/#asterisk aadilismail (n=aaaaaaaa@202.166.161.18) |
| 06:17.02 | aadilismail | hi guys |
| 06:18.21 | blitzrage | night all |
| 06:21.26 | *** join/#asterisk smackus2 (n=smackus2@c-67-169-248-217.hsd1.ut.comcast.net) |
| 06:22.21 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
| 06:23.10 | smackus2 | I am calling in and dialing digits for an extension... from one persons house it does not detect the first digit of the dialed extension, but on another it works correctly. |
| 06:23.22 | smackus2 | probably the persons phone, but could it be asterisk? |
| 06:23.43 | smackus2 | it is not the digit number that is not being detected... |
| 06:24.08 | smackus2 | for example, extension 1718, "-- Invalid extension '718' in context " |
| 06:25.33 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 06:28.00 | *** part/#asterisk nn (n=joseph@pool-72-65-155-239.chrlwv.east.verizon.net) |
| 06:30.48 | smackus2 | seems to be his phone.... |
| 06:30.49 | *** part/#asterisk smackus2 (n=smackus2@c-67-169-248-217.hsd1.ut.comcast.net) |
| 06:34.46 | *** join/#asterisk shtoom (n=godson@59.184.16.174) |
| 06:35.04 | *** join/#asterisk Sakimustafa (n=yusuf@202.133.14.226) |
| 06:36.08 | *** join/#asterisk rue_mohr (n=rue@h24-207-89-118.cst.dccnet.com) |
| 06:37.19 | rue_mohr | hello, I cant run asterisk, and I cant get my hands on a proper voice modem. however, I do have this modem that links to the TAD on a sound card, are there tools I can use to make this work? |
| 06:37.37 | tuck3r | no, get an ATA |
| 06:38.08 | rue_mohr | :/ I know this is close, vgetty commes with /bin/vm, but there seems to be no documentation for it |
| 06:38.40 | tuck3r | waist of time |
| 06:38.42 | rue_mohr | I just want an answering machine |
| 06:38.48 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 06:38.49 | tuck3r | then get a ATA |
| 06:39.02 | rue_mohr | ok, thanks for your oppinion... |
| 06:39.11 | *** join/#asterisk daysmen3 (n=primus@host86-143-5-29.range86-143.btcentralplus.com) |
| 06:41.19 | *** join/#asterisk kevvvy (n=kevvy@dsl-202-72-180-216.wa.westnet.com.au) |
| 06:41.58 | *** join/#asterisk shtoom (n=godson@59.184.16.174) |
| 06:49.00 | aao_pwner | LOL |
| 06:49.02 | aao_pwner | my problem was |
| 06:49.04 | aao_pwner | i was using |
| 06:49.13 | aao_pwner | "make," instead of "Gmake." |
| 06:49.14 | aao_pwner | LoL |
| 06:49.33 | tuck3r | funny BSD |
| 06:50.03 | aao_pwner | silly, silly bsd |
| 06:50.15 | Lyfe | well, gnu make is too restricitve, you can't ship it with bsd. |
| 06:50.31 | Lyfe | and since there's major portions of the system that depend on make.... |
| 06:50.35 | aao_pwner | err, this better not be one of thoes 500 year compiles |
| 06:50.39 | tuck3r | thats RMS |
| 06:50.50 | tuck3r | aao_pwner: it's about 10 mins |
| 06:50.59 | aao_pwner | ok goo |
| 06:51.00 | aao_pwner | good* |
| 06:51.01 | aao_pwner | lol |
| 06:51.09 | tuck3r | well if your computer doesn't suck |
| 06:51.11 | Lyfe | i've met RMS.. he's weird. |
| 06:51.17 | aao_pwner | it doesnt |
| 06:51.20 | aao_pwner | it has an opteron 170 |
| 06:51.23 | aao_pwner | and 4GB of ECC ram |
| 06:51.25 | aao_pwner | i think its decent |
| 06:51.27 | tuck3r | DAMN |
| 06:51.33 | aao_pwner | im 14 :) |
| 06:51.37 | aao_pwner | cry even more |
| 06:51.39 | Lyfe | yeah, that'll work nicely, and build nice and fast. :) |
| 06:51.42 | aao_pwner | its my personal server right next to me |
| 06:51.53 | Lyfe | spoiled. |
| 06:51.57 | aao_pwner | Lyfe i freelance |
| 06:51.59 | aao_pwner | make all the money my self |
| 06:52.10 | row | Is there a beginner's configuration howto that covers the basics of setting up a internal pbx ie just extensions + voice mail? |
| 06:52.14 | Lyfe | that works too.. you're too young to make that much money ;) |
| 06:52.41 | *** join/#asterisk aadilismail (n=aaaaaaaa@202.166.161.18) |
| 06:52.45 | Lyfe | course, prices have been droppin gon hardware for many many years, so maybe i'm just jealous that it's affordable at a young age. |
| 06:52.49 | aao_pwner | Lyfe indians will pay anything for 30 lines of C++ |
| 06:52.49 | tzafrir | Lyfe, there is never a license problem including gmake in bsd (as a mere package) |
| 06:52.49 | aao_pwner | lol |
| 06:53.29 | aao_pwner | uhh |
| 06:53.29 | aao_pwner | /usr/bin/ld: warning: libintl.so.6, needed by /usr/local/lib/libpopt.so, not found (try using -rpath or -rpath-link) |
| 06:53.32 | Lyfe | tzafrir: very true, however, i was commenting on why it's actually gmake in freebsd as opposed to just 'make'. |
| 06:53.34 | aao_pwner | /usr/bin/ld: warning: libintl.so.6, needed by /usr/local/lib/libpopt.so, not found (try using -rpath or -rpath-link) |
| 06:53.35 | aao_pwner | /usr/local/lib/libpopt.so: undefined reference to `libintl_dgettext' |
| 06:53.35 | aao_pwner | gmake[1]: *** [smsq] Error 1 |
| 06:53.35 | aao_pwner | gmake[1]: Leaving directory `/hd1/asterisk/asterisk-1.2.12.1/utils' |
| 06:53.35 | aao_pwner | gmake: *** [subdirs] Error 1 |
| 06:53.38 | aao_pwner | what's that about |
| 06:53.48 | Lyfe | ouch. lot of stuff to paste, don't do that. |
| 06:53.52 | Lyfe | you install newt? |
| 06:53.55 | tuck3r | the linker failed |
| 06:53.57 | aao_pwner | i tried |
| 06:54.02 | aao_pwner | how do i links it? |
| 06:54.10 | aao_pwner | well, i tried installing newt |
| 06:54.11 | Lyfe | pastebin.ca is your friend. |
| 06:54.13 | aao_pwner | and it failed horribly |
| 06:54.20 | Lyfe | really? you do it out of ports? |
| 06:54.20 | aadilismail | new to asterisk .... can anybody tell ... if i have been copied complete asterisk versions in my root . .then how can i instal it from there... |
| 06:54.33 | aao_pwner | make? |
| 06:54.34 | aao_pwner | lol |
| 06:54.38 | tuck3r | aadilismail: read TFOT |
| 06:54.43 | aadilismail | ok |
| 06:54.44 | tzafrir | Lyfe, that's why it should never be called make. It should be called $(MAKE) |
| 06:54.54 | tuck3r | aadilismail: http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 06:55.21 | aao_pwner | where does libintl.so.6, |
| 06:55.22 | aao_pwner | come from |
| 06:55.22 | aadilismail | thanx |
| 06:55.33 | Lyfe | tzafrir: heh. fair 'nuff.. it came from that someone was trying to run 'make' instead of 'gmake', where the makefile that comes with asterisk requires gnu make. |
| 06:55.41 | Lyfe | aao_pwner: gettext (i think) |
| 06:55.51 | aao_pwner | so i can just install gettext and i SHOULD be good to go? |
| 06:55.56 | aao_pwner | if you're right |
| 06:55.59 | Lyfe | i would imagine so. |
| 06:56.08 | Lyfe | you could always deinstall gettext if i'm wrong. :) |
| 06:58.07 | aao_pwner | well, it's apperentally not installed |
| 06:58.08 | aao_pwner | so its all cute |
| 06:58.09 | aao_pwner | :) |
| 06:58.35 | aao_pwner | god damn mailing lists are scary |
| 06:58.37 | aao_pwner | i email one thing |
| 06:58.38 | aao_pwner | and like |
| 06:58.40 | aao_pwner | 30 seconds later |
| 06:58.44 | aao_pwner | i get 50 responses |
| 06:58.52 | aao_pwner | from AU and China |
| 06:59.00 | aao_pwner | god the world is insanely... insane now |
| 06:59.04 | Lyfe | damn, that good, eh? i wish i got responses that fast on questions. |
| 06:59.09 | aao_pwner | lol |
| 06:59.18 | Lyfe | which.. is why i hate turning to mailing lists. |
| 07:00.31 | aao_pwner | but this one is apperentally fast |
| 07:00.33 | aao_pwner | ill get it for you |
| 07:00.38 | tuck3r | i just spent two hours trying to fix my outbound, restart asterisk and it all works |
| 07:00.57 | aao_pwner | http://lists.digium.com/mailman/listinfo/asterisk-bsd |
| 07:00.58 | aao_pwner | that one apperentally |
| 07:01.13 | Lyfe | tuck3r: don't you hate when you have it right, but it just doesn't want to get it in memory? :) |
| 07:01.33 | tuck3r | Lyfe: it seems to be a dynamic IP things |
| 07:01.40 | aao_pwner | i had to deinstall libiconv |
| 07:01.46 | aao_pwner | but it's installing it again :) |
| 07:02.00 | aao_pwner | tuck3r freenet6.com |
| 07:02.03 | aao_pwner | or *.net |
| 07:02.04 | aao_pwner | i forget |
| 07:02.19 | tuck3r | IPV6 tunnel? |
| 07:02.26 | aao_pwner | i dunno |
| 07:02.31 | aao_pwner | you could use IPv7 |
| 07:02.32 | Lyfe | sounds like one to me. |
| 07:02.36 | aao_pwner | v6* |
| 07:02.48 | tuck3r | aao_pwner: if my ISP would support it |
| 07:02.50 | aao_pwner | my friend has like 13 ip addresses on his server... |
| 07:03.00 | aao_pwner | my ISP is a meanie |
| 07:03.03 | aao_pwner | comcast is retarded.. |
| 07:03.11 | tuck3r | yeah, so is qwest |
| 07:03.11 | aao_pwner | i call them and tell them to unblock my outbound 80 |
| 07:03.13 | aao_pwner | and they say |
| 07:03.15 | aao_pwner | What is a port?! |
| 07:03.18 | aao_pwner | i said |
| 07:03.22 | row | lol |
| 07:03.24 | aao_pwner | oh god, get me someone who knows what their doing |
| 07:03.29 | aao_pwner | and they actually transferd me |
| 07:03.31 | orlock | "its like one of those pipes the inernet is made up of" |
| 07:03.36 | aao_pwner | LoL |
| 07:03.40 | orlock | hahah |
| 07:03.40 | aao_pwner | the pipe and the lane |
| 07:03.42 | tuck3r | orlock: TUBES!!!! |
| 07:03.47 | orlock | tubes, thats it! |
| 07:03.48 | aao_pwner | a pipe cleaner |
| 07:03.52 | aao_pwner | takes all the gunk out |
| 07:03.55 | aao_pwner | and the gunk are viruses |
| 07:03.55 | aao_pwner | lol |
| 07:04.09 | tuck3r | who used the tubes as a dumb truck? |
| 07:04.11 | aao_pwner | all your net pipes are belong to us |
| 07:06.07 | aao_pwner | ERR WTF |
| 07:06.24 | aao_pwner | http://pastebin.ca/174810 |
| 07:06.53 | tuck3r | do what it says? |
| 07:07.14 | *** join/#asterisk kamileon (n=kamileon@c-71-207-212-67.hsd1.al.comcast.net) |
| 07:07.44 | aao_pwner | trelane |
| 07:07.46 | aao_pwner | tuck3r* |
| 07:07.48 | aao_pwner | it says.. |
| 07:07.54 | aao_pwner | the package im trying to install is already there |
| 07:08.00 | aao_pwner | and when i goto pkg_delete mode |
| 07:08.02 | aao_pwner | it lol |
| 07:08.09 | tuck3r | make deinstall? |
| 07:08.16 | aao_pwner | i need the port dir for it |
| 07:08.56 | tuck3r | <PROTECTED> |
| 07:09.17 | aao_pwner | theres where im at right now |
| 07:09.27 | aao_pwner | an OLDER version |
| 07:09.29 | aao_pwner | lol |
| 07:09.49 | tuck3r | LOL |
| 07:12.28 | tuck3r | did you updated the ports tree |
| 07:12.51 | aao_pwner | how to? |
| 07:12.52 | aao_pwner | lol |
| 07:13.01 | tuck3r | uh |
| 07:13.31 | tuck3r | basic fbsd i thought |
| 07:13.42 | aao_pwner | well i am now... |
| 07:13.47 | aao_pwner | portupgrade ./ |
| 07:13.53 | aao_pwner | portupgrade -f |
| 07:14.08 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 07:14.38 | Rahail | hey any one use iptv |
| 07:15.05 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 07:15.09 | aao_pwner | LoL |
| 07:15.16 | aao_pwner | that's not a good thing when i look in my SSH client |
| 07:15.19 | aao_pwner | and i see shit happening |
| 07:15.22 | aao_pwner | and i forget what i ran last |
| 07:15.30 | tuck3r | Rahail: i use a form of it |
| 07:19.14 | E-bola | Morning |
| 07:20.49 | *** join/#asterisk Sasch (n=Admin@host102-30-static.107-82-b.business.telecomitalia.it) |
| 07:20.59 | Sasch | hi all |
| 07:21.01 | Sasch | I have one question .... |
| 07:21.20 | Sasch | i have a TDM400P with 1 analog telephone (FXS modules) |
| 07:21.51 | Sasch | but when i recivede a call and i want to transfer to another client ... i digit #10 for example |
| 07:21.56 | Sasch | but don't work .... why ?? |
| 07:22.23 | Sasch | is i digit in my telephone #10 when i don't any call ... the call to 10 go ... |
| 07:23.08 | aao_pwner | Y@Y |
| 07:23.10 | aao_pwner | i got it working |
| 07:28.00 | Rahail | tuck3r |
| 07:28.03 | Rahail | sorry i was away |
| 07:28.06 | tuck3r | yes |
| 07:28.20 | Rahail | I want start IPtv for my network |
| 07:28.28 | Rahail | can you guide me where i can get info from |
| 07:28.31 | Rahail | \? |
| 07:28.34 | Rahail | if you have time |
| 07:28.36 | tuck3r | Oh, I use it, not run it |
| 07:28.41 | Rahail | ops |
| 07:28.47 | Rahail | how was it ... |
| 07:29.01 | Rahail | how much diffrent from reuglar dish brodcasting |
| 07:29.08 | tuck3r | it's great but this is a mulit million dollar system |
| 07:29.21 | tuck3r | same as cable if I didn't know better |
| 07:29.39 | Rahail | I think there is open source software you can use |
| 07:29.41 | Rahail | :) |
| 07:29.57 | tuck3r | not that compares to this, but you could do a more simple setup |
| 07:30.01 | tuck3r | http://broadband.motorola.com/catalog/productdetail.asp?ProductID=368 |
| 07:30.08 | tuck3r | ^^ is the box that my ISP gave me |
| 07:30.48 | Rahail | who is your isp |
| 07:30.59 | tuck3r | Qwest aka USWest |
| 07:31.27 | Rahail | how much they charged you monthly |
| 07:31.47 | tuck3r | uh, around 75$ with internet |
| 07:32.10 | Rahail | how many channles any movie channel hbo cinmax strz etc |
| 07:32.30 | tuck3r | Rahail: I can get the movie channels but I don't |
| 07:33.05 | tuck3r | "ore than 230 all-digital channels, including basic and 45 digital music channels" |
| 07:33.15 | tuck3r | "ll your favorite premium movie and 66 Pay-Per-View channels (additional charge applies)" |
| 07:33.42 | Rahail | cool |
| 07:33.55 | tuck3r | er that was |
| 07:33.57 | tuck3r | More |
| 07:33.58 | tuck3r | and All |
| 07:34.14 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
| 07:34.32 | Rahail | let see what happend tomorrow |
| 07:34.44 | Rahail | some one going to provide me turnkey soulution for .... IPTV |
| 07:35.25 | tuck3r | Rahail: MS is doing one, this is a Moto setup I think |
| 07:36.43 | Rahail | MS? |
| 07:37.03 | tuck3r | microsoft |
| 07:37.20 | Rahail | oh yeah there expensive |
| 07:37.25 | tuck3r | yep |
| 07:38.15 | ingenius | Hi! |
| 07:39.13 | *** part/#asterisk rue_mohr (n=rue@h24-207-89-118.cst.dccnet.com) |
| 07:41.08 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
| 07:41.25 | *** join/#asterisk tRSS (n=tRSS@193.220.221.2) |
| 07:43.08 | *** join/#asterisk luchshiy (n=anonymou@d212-53-104-193.cust.tele2.ch) |
| 07:47.18 | *** join/#asterisk fix- (n=Bjorn@86.39.154.74) |
| 07:48.19 | fix- | using asterisk is it possible to record every participants voice seperate? |
| 07:48.27 | fix- | say you want to do a podcast interview |
| 07:49.01 | tuck3r | use monitor |
| 07:49.33 | fix- | k, google, don't know monitor :) |
| 07:50.04 | tuck3r | http://www.voip-info.org/wiki-Asterisk+cmd+monitor |
| 07:50.13 | tuck3r | fix-: voip-info then google ;) |
| 07:50.22 | fix- | thnx |
| 07:50.28 | *** join/#asterisk Ahrimanes (n=michael@81.7.159.2) |
| 07:50.44 | fix- | http://www.voip-info.org/wiki-Asterisk+cmd+monitor |
| 07:50.49 | fix- | this one, right? |
| 07:51.03 | tuck3r | yep |
| 07:51.36 | fix- | Okay, so I would have to use this command on every extension that joins in on the conference call? |
| 07:52.04 | tuck3r | i have no idea how to do it on a conf |
| 07:52.11 | fix- | :) |
| 07:53.10 | fix- | tuck3r, but you are sure that you can record separate? |
| 07:53.27 | tuck3r | that why monitor sucks (in most cases) |
| 07:53.39 | jeffjohnson | howdy |
| 07:53.48 | tuck3r | MixMonitor outputs one file |
| 07:54.11 | fix- | ah k, so monitor is multiple files |
| 07:54.15 | fix- | mixmonitor is one file |
| 07:54.18 | fix- | correct? |
| 07:54.24 | tuck3r | yes |
| 07:54.26 | fix- | k thnx |
| 07:54.27 | fix- | :) |
| 07:54.46 | tuck3r | before mixmonitor people used a program to combine them |
| 07:54.58 | jeffjohnson | i have several softphones with iax2 connected to asterisk, calls go out over a ZAP line. In the first ~10sec of a phone conversation the sound quality is bad... After 10sec it is quite good, what can be the reason for this? |
| 07:55.50 | sx-wks | echo learning ? |
| 07:55.59 | kevvvy | trying to configure zapata/zaptel to work with digium t410p 4 port isdn bri card. linux detects the card no problems. i noticed in var/log/asterisk/full i see this error: ERROR[4314] chan_zap.c: Unknown signalling method 'bri_cpe_ptmp' |
| 07:56.26 | kevvvy | any way to get it to "dis-unknown" ? :P |
| 07:56.30 | tuck3r | don't you have to install bristuff? |
| 07:57.20 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 07:57.26 | kevvvy | mmm let me install |
| 07:57.45 | tuck3r | don't take my word for it |
| 07:57.51 | tuck3r | but I think you do |
| 07:57.54 | kevvvy | hopefully it dosen't break anything heh |
| 07:58.40 | tuck3r | http://www.voip-info.org/wiki/view/BRI |
| 07:58.53 | *** join/#asterisk MstlyHrmls (n=mh@66.195.193.151) |
| 07:59.09 | Sasch | i have a analog telephone connet to tdm400p ... what is the sequence of key to make a transfer call ?? (# + number) ?? |
| 07:59.23 | tuck3r | what did you set it to? |
| 08:00.12 | jeffjohnson | sx-wks: is there a way to have also good quality in the beginning of a conversation? my customer dont like this .) |
| 08:00.21 | Sasch | <tuck3r> speak with me ? |
| 08:00.41 | tuck3r | yes, I don't think it's enabled by default |
| 08:01.42 | Sasch | hummmm because if i take telephone and i digit #10 the call start with 10 |
| 08:01.58 | sx-wks | jeffjohnson: learn the echo for good with zttune ? |
| 08:02.14 | Sasch | but if i response a call and after i digit #10 asterisk don't transfer a call ... |
| 08:02.50 | jeffjohnson | sx-wks: zttune? |
| 08:03.00 | sx-wks | jeffjohnson: check on the wiki |
| 08:03.06 | tuck3r | Sasch: yeah nevermind me |
| 08:03.36 | jeffjohnson | sx-wks: voip-info? which wiki u mean? |
| 08:04.02 | sx-wks | that one |
| 08:04.16 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 08:04.22 | jeffjohnson | sx-wks: no match for zttune, google has no match for zttune to |
| 08:04.24 | jeffjohnson | o |
| 08:04.36 | *** join/#asterisk Aurs (n=Aurs@host-81-191-123-189.bluecom.no) |
| 08:04.49 | *** join/#asterisk aadilismail (n=aaaaaaaa@202.166.161.18) |
| 08:05.59 | aadilismail | hi |
| 08:06.40 | Sasch | <tuck3r> do you have a link for a guide ... |
| 08:06.55 | tuck3r | http://www.voip-info.org/wiki/view/Asterisk+config+features.conf |
| 08:07.19 | jeffjohnson | sx-wks: ? |
| 08:07.55 | sx-wks | gah. voip-info is down at this time |
| 08:08.55 | jeffjohnson | sx-wks: voip-info isnt down .o |
| 08:09.07 | jeffjohnson | oh hum |
| 08:09.07 | tuck3r | yes it is |
| 08:09.13 | fish_ | re |
| 08:09.17 | jeffjohnson | 3min ago the site was up :) |
| 08:09.22 | sx-wks | lol |
| 08:09.31 | *** join/#asterisk daysmen3 (n=primus@host86-143-5-29.range86-143.btcentralplus.com) |
| 08:10.07 | sx-wks | jeffjohnson: there's a program that will learn the echo on the line. I know there's "tune" in the name. |
| 08:10.37 | tuck3r | and most likely starts with a z |
| 08:11.32 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 08:11.42 | jeffjohnson | sx-wks: fxotune? |
| 08:11.58 | tuck3r | guess not :| |
| 08:12.32 | jeffjohnson | sx-wks: http://72.14.221.104/search?q=cache:zyErR-OdkjsJ:www.voip-info.org/wiki/view/Asterisk%2Bfxotune+zap+echo+tune&hl=de&gl=de&ct=clnk&cd=1&client=firefox |
| 08:12.53 | sx-wks | yeah, that |
| 08:12.55 | jeffjohnson | sx-wks: but i have hfc quadbri and hfc pci cards ./ |
| 08:13.07 | jeffjohnson | sx-wks: dont work for me it is only for TDM400 cards |
| 08:13.09 | jeffjohnson | sx-wks: :/ |
| 08:13.11 | sx-wks | ah... can't help you there :D |
| 08:13.19 | jeffjohnson | sx-wks: thy anyway :) |
| 08:13.37 | sx-wks | if you're using ISDN you shouldn't have sound quality problems... |
| 08:14.40 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
| 08:15.56 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
| 08:20.11 | *** part/#asterisk jdub (n=jdub@home.waugh.id.au) |
| 08:27.02 | *** join/#asterisk alawguy (n=mike@85-124-36-164.dynamic.xdsl-line.inode.at) |
| 08:30.44 | tuck3r | anybody know what SLINEAR is? |
| 08:31.54 | *** join/#asterisk Guest^DJ (n=me@211.24.146.11) |
| 08:32.25 | Guest^DJ | hi all, can anyone tell me where can i find voip that provides callerID |
| 08:33.49 | tuck3r | all most all providers |
| 08:34.44 | Guest^DJ | please point me to 1 or 2 and it would be great if they provide trial call |
| 08:35.19 | tuck3r | uh, voipjet and voxee |
| 08:35.27 | tuck3r | and voxbone |
| 08:35.34 | Guest^DJ | tuck3r: thanks |
| 08:38.49 | tuck3r | w00t, my blackhole workd |
| 08:38.50 | tuck3r | s |
| 08:42.56 | tuck3r | nighty night |
| 08:45.23 | *** join/#asterisk HaMYaI (i=BugKhaM@202.8.86.164) |
| 08:46.37 | HaMYaI | twisted[mobile] : are you there? |
| 08:46.40 | *** join/#asterisk GiabboO (n=GiabboO@host-84-220-175-143.cust-adsl.tiscali.it) |
| 08:47.01 | GiabboO | hello, is there any italian user ? |
| 08:57.05 | tRSS | anyone here comfortable with gnudialer and the crm application that comes with it? |
| 09:01.18 | *** join/#asterisk Tili (n=tili@202.133.65.99) |
| 09:01.52 | *** join/#asterisk IOscanner (n=IOscanne@c-67-164-154-209.hsd1.tx.comcast.net) |
| 09:06.24 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 09:19.11 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 09:21.44 | *** join/#asterisk Gunnar (n=gunnar@101.Red-80-24-171.staticIP.rima-tde.net) |
| 09:25.45 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 09:31.35 | *** join/#asterisk Sloboda (n=slob@local.proggart.com) |
| 09:36.57 | *** part/#asterisk Sloboda (n=slob@local.proggart.com) |
| 09:38.20 | *** join/#asterisk adorah (n=admin@87.68.168.3.cable.012.net.il) |
| 09:39.10 | *** join/#asterisk docelmo (n=vircuser@55-65.126-70.tampabay.res.rr.com) |
| 09:41.05 | *** join/#asterisk af_ (n=af@ip-202-199.sn2.eutelia.it) |
| 09:50.10 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 09:53.33 | GiabboO | anybody here use asterisk on gentoo linux ? |
| 09:57.30 | RoyK | GiabboO: prolly quite a few |
| 10:00.36 | *** join/#asterisk evilmnky (n=evilmnky@216-106-185-169.ds1-static.mia1.net.ststelecom.com) |
| 10:00.39 | evilmnky | heh.. conf 360-207-0488 pin=1234 (testing asterisk) |
| 10:00.39 | *** join/#asterisk vgster (n=vgster@170.252.64.1) |
| 10:02.12 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 10:02.56 | Sasch | there are a person that use an analog telephone with asterisk ??? |
| 10:04.25 | *** join/#asterisk dyn (n=dyn@unaffiliated/dyn) |
| 10:04.25 | tzafrir | Sasch, many a person |
| 10:04.26 | dyn | hi |
| 10:04.43 | evilmnky | heh.. conf 360-207-0488 pin=1234 (testing asterisk) |
| 10:04.47 | dyn | anyone's using a Billion BiPAC PCI ISDN card here with asterisk? |
| 10:05.00 | dyn | i'd like to get a cheap isdn card which is correctly supported |
| 10:05.10 | dyn | so any hints/advices could help a lot |
| 10:05.11 | Sasch | <tzafrir> because i have a tdm400p with 1 fxs with an analog telephone |
| 10:05.27 | GiabboO | RoyK, I was long time without use asterisk, now i found on my portage some strange use to compile asterisk, mysqlfriends and vmdbmysql |
| 10:05.28 | tzafrir | Sasch, should work well |
| 10:05.36 | Sasch | <tzafrir> but when i would trasnfer a call to anolog telephone don't work |
| 10:06.02 | GiabboO | RoyK, mysqlfriends is a stable way to setup users on mysql ? the other is stable to voicemail ? |
| 10:06.23 | RoyK | GiabboO: mysqlfriends is an old and shabby hack for asterisk 1.0 |
| 10:06.29 | tzafrir | Sasch, for starters, to isolate from others problems you may have, try to call a simple echo-test extension (Echo() ) |
| 10:06.54 | Sasch | what is echo-test ?? |
| 10:06.56 | GiabboO | RoyK, so I really need update:) |
| 10:07.15 | RoyK | GiabboO: asterisk.org/download is a good start |
| 10:07.23 | Sasch | if you want i can link my dial plan |
| 10:07.35 | GiabboO | RoyK, I would like to use the portage system |
| 10:07.48 | tzafrir | Sasch, an asterisk application (that is: something that you can use as a "command" in the dialplan) that basically sends you back what it has recieved |
| 10:08.04 | RoyK | GiabboO: don't |
| 10:08.09 | GiabboO | ups |
| 10:08.11 | GiabboO | ok |
| 10:08.17 | tzafrir | anyway, got some work... |
| 10:08.19 | GiabboO | I was using the cvshead |
| 10:08.25 | GiabboO | more than 1year ago |
| 10:08.29 | GiabboO | i think 2 |
| 10:08.38 | Aurs | asterisk don't even use cvs anymore |
| 10:08.38 | RoyK | :) |
| 10:08.41 | Aurs | :P |
| 10:08.53 | GiabboO | oh |
| 10:08.56 | GiabboO | ^^ |
| 10:09.01 | RoyK | :) |
| 10:09.06 | RoyK | wb, GiabboO |
| 10:09.07 | GiabboO | I told you I really need to update |
| 10:09.09 | GiabboO | mysql :p |
| 10:09.12 | Aurs | hehe |
| 10:09.16 | GiabboO | lol |
| 10:09.23 | GiabboO | ty RoyK |
| 10:09.26 | GiabboO | :)) |
| 10:09.31 | GiabboO | ok cig time then ill work on it :D |
| 10:09.36 | GiabboO | lol |
| 10:09.45 | Sasch | this is my dial plan |
| 10:09.46 | Sasch | http://paste.uni.cc/10182 |
| 10:10.06 | GiabboO | ~# asterisk -V |
| 10:10.16 | GiabboO | Asterisk CVS-HEAD-03/21/05-16:41:05 |
| 10:10.16 | GiabboO | :P |
| 10:10.23 | GiabboO | BRB:) |
| 10:13.23 | *** join/#asterisk vazir (i=anton@82.198.21.17) |
| 10:13.42 | GiabboO | k, im back :) |
| 10:13.44 | docelmo | I remember 0.71 which is the first version I played with |
| 10:13.58 | docelmo | it wasnt too bad.. it worked kinda |
| 10:14.01 | GiabboO | I bet i need help to delete the cvs version before update to the current release |
| 10:14.27 | docelmo | I wouldnt goto the most current mabye a revision or 2 back |
| 10:14.39 | docelmo | Digium might have broke something in the most current you need |
| 10:14.43 | Sasch | <tzafrir> to transfer a call with analog telephone is key # ???? |
| 10:15.01 | *** part/#asterisk vazir (i=anton@82.198.21.17) |
| 10:15.20 | GiabboO | well ill backup my configs |
| 10:15.46 | GiabboO | getting http://ftp.digium.com/pub/asterisk/asterisk-1.2-current.tar.gz :> |
| 10:15.46 | docelmo | you will need to dump your modules directory and also the includes |
| 10:16.01 | docelmo | <PROTECTED> |
| 10:16.01 | GiabboO | I have no problem to configure it again |
| 10:16.11 | GiabboO | kk i check 1 sek |
| 10:16.32 | GiabboO | found. do I need to back it up ? |
| 10:16.51 | GiabboO | or drop it ? |
| 10:17.38 | dyn | anyone's using a Billion BiPAC PCI ISDN card here with asterisk? |
| 10:17.55 | docelmo | drop it |
| 10:18.10 | docelmo | if you're upgrading then those files will conflict |
| 10:18.16 | GiabboO | ok, did it, /usr/lib/asterisk and /usr/include/asterisk |
| 10:18.43 | docelmo | Now you need to install your new copy of asterisk. make; make install |
| 10:18.49 | docelmo | your config's should be ok |
| 10:19.23 | GiabboO | It was very dirty config, with extconf and mysql hack |
| 10:19.42 | docelmo | You should see my cluster |
| 10:19.42 | GiabboO | I think should be better to redo configuration from 0 |
| 10:19.48 | docelmo | :) |
| 10:19.51 | GiabboO | lol |
| 10:20.02 | docelmo | Most of asterisk is hacked to do things it wasnt designed for |
| 10:20.09 | GiabboO | eheheh |
| 10:20.17 | GiabboO | ok i feel better to hear this |
| 10:20.29 | docelmo | I wrote a LCR in C and billing engine from cdr_addons_mysql.c |
| 10:20.42 | GiabboO | making:> |
| 10:21.21 | GiabboO | I am moving my office so Ill asked my provider a PRI isdn line |
| 10:21.29 | GiabboO | (sorry for bad english:P) |
| 10:21.42 | GiabboO | so I also need to buy a PRI card, from Digium |
| 10:21.59 | docelmo | Italy.. E1 have fun with that.. :) |
| 10:22.13 | GiabboO | yes, is it complex ? |
| 10:22.32 | docelmo | I find E1's to be a pain but then again I am in the US where T1's are |
| 10:22.58 | GiabboO | I see, well i will buy that, E1 yes, to do active selection from outside |
| 10:23.15 | CtRiX | GiabboO, use sangoma cards. |
| 10:23.20 | GiabboO | This should be what I need http://www.digium.com/en/products/hardware/te110p.php |
| 10:23.28 | GiabboO | uh, CtRiX hi |
| 10:23.31 | docelmo | an E1 will give you 30 channels of inbound and outbound voice |
| 10:23.34 | GiabboO | could u link me something ? |
| 10:23.37 | docelmo | I wasnt gonna say it |
| 10:23.38 | docelmo | :) |
| 10:23.43 | docelmo | www.sangoma.com |
| 10:23.58 | GiabboO | is a good choice for my needs ? |
| 10:24.04 | CtRiX | digium stuff has problems with echo cancellation |
| 10:24.10 | GiabboO | uh |
| 10:24.24 | GiabboO | thanks god you told me this |
| 10:24.25 | CtRiX | of course that's a feature. They give you echo problems for free. |
| 10:24.28 | docelmo | Those cards are just as good as www.pbxeq.com |
| 10:24.30 | GiabboO | i was going to order it |
| 10:24.50 | GiabboO | 1 sec, ill check www.sangoma.com |
| 10:25.30 | CtRiX | pbxeq sells sangoma.... that's the same thing. it's a matter of price, i think |
| 10:26.11 | GiabboO | http://www.sangoma.com/datasheets/p_aft-et1-specs |
| 10:26.17 | GiabboO | this is what I need right ? |
| 10:26.19 | docelmo | Yes.. But I was keying more on the OpenVOX stuff |
| 10:26.24 | docelmo | I have one its not a bad board |
| 10:27.37 | GiabboO | found this |
| 10:27.39 | GiabboO | http://store.pbxeq.com/store/product.php?productid=16462&cat=0&page=1 |
| 10:28.12 | CtRiX | The D110P is 100% hardware compatible with TE110P. |
| 10:28.21 | CtRiX | i wonder if it's echo compatible :P |
| 10:29.06 | docelmo | http://store.pbxeq.com/store/product.php?productid=16152&cat=255&page=1 |
| 10:29.31 | docelmo | I have that card |
| 10:29.34 | docelmo | its a good board |
| 10:29.48 | docelmo | I dont have echo issues in my test lab |
| 10:30.27 | GiabboO | ahah |
| 10:30.41 | GiabboO | this make me a echodubt :P |
| 10:30.47 | *** join/#asterisk ys1797 (n=yuri@213.159.111.82) |
| 10:31.08 | GiabboO | where can I find good information about E1 card support for Italy ? |
| 10:31.20 | docelmo | You can ask them when you purchase it if you can return it if there are echo issues |
| 10:31.23 | GiabboO | I dont want echo problem :p |
| 10:31.39 | docelmo | a E1 is a E1 |
| 10:31.39 | GiabboO | i got not so much time to do tests :( |
| 10:31.52 | docelmo | they are all the same.. difference is how good your telco is |
| 10:32.01 | GiabboO | I see.. |
| 10:32.24 | docelmo | bbiaf |
| 10:32.34 | GiabboO | anyway, give me a final idea what and how to buy |
| 10:32.35 | RoyK | ...and ISUP is ISUP and ISDN is ISDN, but then there are several ways to do things and even here, in little norway, the telcos can't agree |
| 10:32.46 | GiabboO | digium seems to be expensive than others |
| 10:32.51 | CtRiX | there is not such a problem in italy. Only 1 telco selling isdn lines ! |
| 10:32.59 | GiabboO | yes:) |
| 10:33.08 | GiabboO | Telecom Italia < OWNERS |
| 10:33.18 | GiabboO | (and kinda suckers:P) |
| 10:33.24 | RoyK | GiabboO: digium is as expensive as sangoma, for instance, only sangoma makes what I'd call better hardware |
| 10:33.39 | RoyK | no BS about "non-supported server hardware" and suchlike |
| 10:33.42 | GiabboO | perfect, I would choice the best |
| 10:34.00 | RaYmAn-Bx | RoyK: and you dare say that in here? ;) |
| 10:34.06 | RoyK | RaYmAn-Bx: indeed |
| 10:34.09 | GiabboO | ahah |
| 10:34.17 | RoyK | both HP and IBM servers are unsupported by digium, but of course, that's because HP and IBM make bad servers.... |
| 10:34.28 | GiabboO | ops |
| 10:34.35 | RaYmAn-Bx | the worst! ;) |
| 10:34.41 | GiabboO | I am running Asterisk on ibm server |
| 10:34.46 | GiabboO | little eserver |
| 10:34.51 | RoyK | those are nice |
| 10:35.02 | RaYmAn-Bx | we only use ibm servers at work as well..we just got some nice bladecenters |
| 10:35.04 | GiabboO | yes imho |
| 10:35.20 | RoyK | 336s are nice |
| 10:35.29 | GiabboO | I dont remember what I have |
| 10:35.31 | GiabboO | i go check wait :D |
| 10:35.34 | RoyK | :) |
| 10:36.26 | docelmo | Well you have 3 options |
| 10:36.46 | GiabboO | Instellistation M pro |
| 10:36.52 | GiabboO | 6849-31G |
| 10:36.57 | docelmo | Most lean toward Sangoma as a better board but mainly your telco will determine how much of a headache you have setting it up |
| 10:37.04 | GiabboO | btw its old |
| 10:37.08 | GiabboO | but still works |
| 10:37.43 | GiabboO | ugh, I got problem with make command |
| 10:38.06 | GiabboO | chan_zap.c:62:2: #error "You need newer libpri" |
| 10:38.13 | GiabboO | then few warning and errors |
| 10:38.13 | docelmo | hehe |
| 10:38.21 | docelmo | Do you have ztdummy? |
| 10:38.33 | docelmo | or a Asterisk compatible TDM card? |
| 10:38.36 | GiabboO | I even dont know what is that |
| 10:38.39 | GiabboO | nope |
| 10:38.44 | GiabboO | I will buy it |
| 10:38.48 | docelmo | Do you use Meetme? |
| 10:38.55 | GiabboO | no |
| 10:39.05 | ys1797 | GiabboO: http://ftp.digium.com/pub/libpri/releases/libpri-1.2.3.tar.gz |
| 10:39.06 | GiabboO | I tried asterisk with a ISDN card |
| 10:39.13 | GiabboO | ty ys1797 |
| 10:41.17 | GiabboO | k seems working now |
| 10:41.20 | CtRiX | GiabboO, there are tons of reports of digium cards not working on IBM and HP servers |
| 10:41.56 | GiabboO | I should talk to some italian E1 users |
| 10:42.40 | docelmo | for what? |
| 10:42.41 | RoyK | CtRiX: there's even an official article about it on digium.com |
| 10:42.50 | GiabboO | to know what they use |
| 10:43.11 | docelmo | Minimize headache.. Sangoma or Openvo |
| 10:43.14 | docelmo | vox |
| 10:43.18 | GiabboO | ahah |
| 10:43.21 | RoyK | http://www.digium.com/en/docs/misc/compatibility_notes.php?redir=TDM2400P |
| 10:43.21 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 10:43.22 | GiabboO | thanks:) |
| 10:43.24 | docelmo | Digium boards are hated by some bios's |
| 10:43.32 | GiabboO | k |
| 10:43.37 | RoyK | well, dell boxes as well |
| 10:43.41 | docelmo | yes |
| 10:43.55 | docelmo | Digium's cards are hated by almost every dell I have tried |
| 10:43.59 | docelmo | I gave up |
| 10:44.07 | RoyK | docelmo: it all boils down to :%s/digium/Sangoma/gi |
| 10:44.15 | RoyK | imho |
| 10:44.28 | docelmo | I hate regex |
| 10:44.50 | RoyK | it's 'substitute digium by sangoma globally and case insensitive' |
| 10:44.51 | *** join/#asterisk kiko69 (n=keith@legoland.it.earthlink.net) |
| 10:44.54 | creativx | i like spandex |
| 10:45.01 | RoyK | spandex? |
| 10:45.07 | docelmo | hehe |
| 10:45.12 | docelmo | speex |
| 10:45.24 | docelmo | or spandsp |
| 10:45.50 | creativx | no. spandex |
| 10:45.52 | creativx | the fabric |
| 10:45.53 | CtRiX | anyway what is clear is that digium(tm) cards should be avoided unless a sysadmin wants something to fight against. |
| 10:45.54 | creativx | ;) |
| 10:46.27 | docelmo | Well I put one in a Tyan box and its fine.. |
| 10:46.33 | mut | anyone up much on IIS? i'm using iis5 and every approx 2 hours, all of the tcp connections start slowing down and eventually ya can't connect to it at all, but once you restart IIS it works again perfectly for approx 2 hours |
| 10:46.35 | RoyK | CtRiX: good thing the americans haven't woken up yet, or we'll be kicked out of here :P |
| 10:46.37 | docelmo | TE410p I think it was |
| 10:46.53 | docelmo | Im awake but running on UK time right now |
| 10:47.00 | RoyK | docelmo: show application iis? |
| 10:47.02 | creativx | mut: i run both 5 and 6 with no problems.. except on one box where it dies from time to time |
| 10:47.07 | mut | well |
| 10:47.14 | RoyK | iis? chan_iis? |
| 10:47.16 | mut | this started happening out of the blue last wednesday |
| 10:47.17 | CtRiX | RoyK, don't troll! |
| 10:47.22 | docelmo | mut this is not windows hell |
| 10:47.31 | creativx | windows and asterisk go hand in hand |
| 10:47.38 | RoyK | CtRiX: troll? |
| 10:47.40 | mut | its also not bash digium hardware either |
| 10:47.48 | CtRiX | RoyK :-) |
| 10:48.01 | docelmo | You cant bash when stating facts |
| 10:48.06 | RoyK | creativx: yes, like george bush and saddam hussein |
| 10:48.18 | GiabboO | ok |
| 10:48.21 | creativx | it it wasnt for windows there wouldnt be funding for me to buy the box that runs asterisk |
| 10:48.23 | creativx | how about that |
| 10:48.31 | GiabboO | I found Sangoma E1 card in italy |
| 10:48.36 | GiabboO | 490 € |
| 10:48.47 | RoyK | GiabboO: which one? |
| 10:48.53 | RoyK | A101 or something? |
| 10:48.55 | CtRiX | digium i think are more expensive |
| 10:49.03 | docelmo | Get A101U |
| 10:49.03 | CtRiX | at least here |
| 10:49.07 | RoyK | CtRiX: they cost about the same |
| 10:49.08 | GiabboO | A101 yea |
| 10:49.10 | RoyK | docelmo: what's U? |
| 10:49.14 | GiabboO | I also found other costructor |
| 10:49.15 | *** join/#asterisk angryuser (n=aster@i03v-213-44-169-43.d4.club-internet.fr) |
| 10:49.15 | GiabboO | s |
| 10:49.17 | RoyK | GiabboO: A101D is nice |
| 10:49.19 | docelmo | 2nd gen |
| 10:49.21 | GiabboO | RHINO |
| 10:49.26 | RoyK | more expensive but with HWEC |
| 10:49.31 | GiabboO | Varion |
| 10:49.44 | GiabboO | <PROTECTED> |
| 10:49.57 | GiabboO | RHINO is the cheapest |
| 10:50.22 | RoyK | GiabboO: expensive, but they do onboard channels switching and stuff which is really cool |
| 10:50.25 | GiabboO | hmm I can see only A101 without any U or D |
| 10:51.01 | docelmo | its probably the 1st gen |
| 10:51.06 | GiabboO | take a look http://www.wildix.it/product_info.php?products_id=27&cPath=50_8_16 |
| 10:51.07 | docelmo | I wouldnt touch that one |
| 10:51.13 | RoyK | GiabboO: D is like the A101 but with hardware echo cancel, which is far better than the one in craptel |
| 10:51.22 | RoyK | docelmo: huh? |
| 10:51.27 | docelmo | hehe they are using OsCommerce |
| 10:51.28 | RoyK | docelmo: we're running several of them in production |
| 10:51.33 | GiabboO | lol yea |
| 10:51.51 | RoyK | docelmo: it's not like the digium 0th generation crap |
| 10:52.02 | GiabboO | I would love to get the best release of that card |
| 10:52.15 | RoyK | get a A101D, then |
| 10:52.30 | *** join/#asterisk HaMYaI (i=BugKhaM@202.8.86.164) |
| 10:52.43 | docelmo | I cant read italian |
| 10:52.58 | GiabboO | <PROTECTED> |
| 10:53.26 | GiabboO | ill search for the D release |
| 10:53.35 | GiabboO | revision :p |
| 10:53.41 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 10:53.48 | backblue | hi, morning! |
| 10:53.53 | GiabboO | hello |
| 10:54.19 | RoyK | lol |
| 10:54.47 | RoyK | digium has a rather long list of unsupported hardware. sangoma just say "Comprise an integrated family of products that all work identically in all motherboards under all conditions of shared and unshared interrupts all the time. " |
| 10:55.05 | GiabboO | loves that :) |
| 10:55.08 | GiabboO | ehehe |
| 10:55.39 | HaMYaI | anyway to have a TDM400 as /proc/zaptel/1 and TE110P as /proc/zaptel/2 every time my server reboots? |
| 10:55.47 | backblue | sangoma it's the biggest hardware manufacture, in telephony market, since i remember, dont compare digium to sangoma please. |
| 10:55.48 | angryuser | great, i have that interruption stuff with digium |
| 10:55.54 | angryuser | *hate |
| 10:56.29 | HaMYaI | I had to restart zaptel every time to get it right |
| 10:56.55 | GiabboO | yes backblue, I am reading it in the websites who sell those cards |
| 10:57.00 | RoyK | HaMYaI: digium support might help |
| 10:57.02 | *** join/#asterisk pnlarsson (n=niklas@c83-248-0-248.bredband.comhem.se) |
| 10:57.10 | HaMYaI | RoyK: =) |
| 10:57.17 | GiabboO | bye RoyK |
| 10:57.21 | GiabboO | lo |
| 10:57.22 | backblue | HaMYaI: why should you have that? |
| 10:57.39 | *** join/#asterisk vgster (n=vgster@170.252.64.1) |
| 10:57.48 | HaMYaI | RoyK: but re-starting zaptel make it right, there should be an easy way to fix this |
| 10:57.57 | RoyK | yep... |
| 10:58.08 | RoyK | add an init script that does just that :P |
| 10:58.11 | CtRiX | removing zaptel ! |
| 10:58.13 | CtRiX | LOL! |
| 10:58.25 | backblue | HaMYaI: what does happend? everytime you restart your machine, you have to go restart zaptel after? |
| 10:58.26 | RoyK | sangoma is working on native support for opbx :) |
| 10:58.43 | CtRiX | very good news then |
| 10:58.58 | backblue | sangoma it's great, but it's expensive. |
| 10:58.59 | docelmo | RoyK at which time people will want to move it to asterisk |
| 10:59.04 | HaMYaI | RoyK: yeah, probably have to put zaptel at the end just before asterisk |
| 10:59.10 | RoyK | backblue: about the same as digium, and bettwe |
| 10:59.13 | docelmo | But will not work since asterisk has changed its API in 1.4 |
| 10:59.17 | HaMYaI | backblue: yes, sir |
| 10:59.25 | GiabboO | pff i cant find A101D in Italy |
| 10:59.25 | backblue | but sometimes, un-expensive cards, can be even more expensive than the others. |
| 10:59.37 | GiabboO | Asterisk Build Complete :)) |
| 10:59.44 | backblue | HaMYaI: some distro on redhat famaly? |
| 10:59.55 | backblue | family? |
| 10:59.55 | RoyK | GiabboO: i was wrong. i don't think there's a single-port card with EC. |
| 10:59.59 | HaMYaI | backblue: from dmesg, the cards were detected in varying orders |
| 10:59.59 | RoyK | I have a few A104s |
| 11:00.09 | RoyK | redhat famine... |
| 11:00.18 | HaMYaI | backblue: yes, FC5 |
| 11:00.20 | GiabboO | eh... |
| 11:00.31 | backblue | HaMYaI: ok, that's the timeone on the rc script |
| 11:00.45 | backblue | it's a bug, in some hardware. |
| 11:00.46 | GiabboO | ok ppl |
| 11:00.49 | GiabboO | GTG now |
| 11:00.52 | GiabboO | ill back later |
| 11:00.53 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 11:00.55 | HaMYaI | backblue: but that doesn't really make sense =( |
| 11:00.56 | GiabboO | thanks you all for support |
| 11:01.10 | backblue | i had the same issue, just go up with the timeout |
| 11:01.20 | backblue | HaMYaI: yes it does. |
| 11:01.35 | HaMYaI | backblue: changing orders in /etc/modprobe.d/zaptel, /etc/modprobe.conf I've tried all that |
| 11:01.39 | backblue | i dont remember, but, when i had that problem i have understand what was wrong, and it makes sense. |
| 11:01.56 | backblue | it's not orders i think. |
| 11:02.03 | backblue | what's your chipset on board? |
| 11:02.47 | backblue | GiabboO`away: send me some PASTA! :) |
| 11:04.05 | dyn | is it possible to use two pieces of 1BRI cards in the same box with asterisk? |
| 11:04.13 | HaMYaI | backblue: Intel(R) Pentium(R) D CPU 3.40GHz on P5LD2 SE |
| 11:04.50 | stoffell | dyn: you mean hfc BRI cards? |
| 11:04.52 | backblue | chipset |
| 11:05.00 | dyn | stoffell: exactly |
| 11:05.05 | backblue | dyn: yes. |
| 11:05.07 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
| 11:05.16 | backblue | dyn: i have machines with 5 single bri. |
| 11:05.25 | stoffell | dyn: should be possible.. but make sure you try it out.. |
| 11:05.25 | backblue | ups, 4 single bri, sorry. |
| 11:05.40 | dyn | excellent news |
| 11:05.44 | backblue | but does not make much sense, should be with 4s0 |
| 11:05.57 | backblue | but whatever, it does the job |
| 11:06.03 | dyn | i was a bit worried, the 2-ports bri card costs 10 times the price of the 1 port one |
| 11:06.17 | backblue | dyn: they are diferent. |
| 11:06.18 | *** join/#asterisk yogurt2ungue (n=yogurt2u@24-48-231-201.fibertel.com.ar) |
| 11:06.22 | stoffell | dyn: it's because they use 1 slot, 1 irq, etc.. it is much better.. |
| 11:06.35 | backblue | and have some other positive points |
| 11:06.35 | dyn | backblue: i [think I] get the difference with active and passive cards |
| 11:06.59 | backblue | like using only one irq, and only one pci, and has pcm bus, and led's. |
| 11:07.05 | dyn | stoffell: that's ok.. but still looks like an upgrade path buying an other 1port one after we succeed with the first |
| 11:07.30 | backblue | that's why i like working with bero |
| 11:07.53 | stoffell | dyn: correct. IMHO.. 1 is okay with single bri, but more.. you'd better use 4bri (beronet/junghanns/digium) |
| 11:08.10 | backblue | stoffell: they get expensive solution |
| 11:08.17 | dyn | *nods* |
| 11:08.27 | backblue | sometimes, 1s0 makes the movie work |
| 11:08.37 | backblue | or 2 x 1s0 |
| 11:08.55 | *** join/#asterisk jeremy_g (n=jerms@static-213-115-44-90.sme.bredbandsbolaget.se) |
| 11:08.58 | jeremy_g | h |
| 11:09.00 | backblue | i have quiete experience with bri |
| 11:09.18 | backblue | and they get much expensive |
| 11:09.24 | dyn | backblue: you have the billion bipac cards or something else? |
| 11:09.39 | backblue | dyn: yes, billion, we sell it. |
| 11:09.53 | dyn | backblue: the plain PCI v3.0? |
| 11:09.54 | backblue | but what matter's it's the chipset, and it's hfc. |
| 11:10.00 | dyn | sure thing |
| 11:10.17 | dyn | i dont yet have the card.. but what will I need to get it work? |
| 11:10.30 | backblue | gcc |
| 11:10.33 | stoffell | backblue: and where do you fetch those 1port HFC cards? |
| 11:10.36 | dyn | *snicker* |
| 11:10.37 | backblue | linux-headers |
| 11:10.40 | backblue | pci slot |
| 11:10.42 | backblue | :) |
| 11:10.55 | dyn | i mean, I installed asterisk-bristuff |
| 11:11.04 | dyn | but atm i'm a bit confused about zap*, capi, mISDN, etc |
| 11:11.15 | backblue | dyn: forget bristuff, install misdn. |
| 11:11.31 | *** join/#asterisk syn (i=syn@kenobi.sceen.net) |
| 11:11.45 | syn | hello |
| 11:12.08 | jeremy_g | hi syn |
| 11:12.08 | syn | i have problems understanding the behaviour of chan_ooh323 in some situation |
| 11:12.12 | syn | hi jeremy_g |
| 11:12.13 | backblue | stoffell: we sell it, sorry i cant give you more detail, because i dont know, i'm from technical |
| 11:12.29 | dyn | backblue: so i need to load misdn in the kernel and then asterisk will be able to use chan-misdn on top of it? |
| 11:12.36 | syn | i'm using asterisk with chan_ooh323 and gnugk, and when calling a registered alias, eveerything works fine |
| 11:12.43 | syn | the problem is when calling an unregistered alias |
| 11:12.46 | stoffell | backblue: ok, thanks :) |
| 11:12.47 | syn | there seems to be a loop |
| 11:13.04 | syn | apparently, gnugk doesn't know about the alias, so it sends it to the gateway (asterisk) |
| 11:13.07 | backblue | dyn: install libpri+zaptel+asterisk, then run the http://www.beronet.com/downloads/install-misdn-mqueue.tar.gz |
| 11:13.13 | syn | and asterisk asks the gatekeeper about aliases |
| 11:13.21 | syn | is it a known issue ? a configuration mistake ? |
| 11:13.30 | backblue | then follow the pdf on that website. |
| 11:13.36 | backblue | it's everything there |
| 11:13.42 | dyn | backblue: excellent, thank you |
| 11:14.16 | backblue | dyn: forget bristuff, it's bad planned, bad designed, and have a lot of weird bug's. |
| 11:14.44 | syn | yeah, bristuff sucks :) |
| 11:14.54 | dyn | backblue: okay.. you just see, it's easy to get confused with all the available options (and their layering - which uses which, etc) |
| 11:15.10 | backblue | misdn, it's totally rfc complaint, and you dont have to rebuild asterisk and zaptel to update a drivers. |
| 11:15.16 | backblue | like bristuff... |
| 11:16.05 | backblue | dyn: i know, been there, done that, quite time using this now, i can help in this simple questions. |
| 11:16.10 | dyn | :) |
| 11:16.17 | jeremy_g | syn:loop cure, no clue! |
| 11:16.28 | jeremy_g | :( |
| 11:16.29 | syn | jeremy_g: :( |
| 11:16.47 | backblue | i have not yet, been able to use digium bri card, but should be ok, since it uses misdn drivers :) |
| 11:17.05 | syn | my gnugk must be badly configured |
| 11:17.18 | dyn | okay.. let's assume I'll have a clean path once the card arrives, I'll turn on asking all the noob questions ;) |
| 11:17.21 | jeremy_g | has anyone tried osip - how we compare it with asterisk? |
| 11:17.29 | syn | but i could see than chan_ooh323 has serious memory leaks |
| 11:17.42 | jeremy_g | syn:how did u find that |
| 11:17.43 | syn | the loop makes asterisk crash in a few seconds |
| 11:17.50 | syn | with a memory allocation error |
| 11:17.58 | syn | i guees chan_ooh323 doesn't handle errors very well |
| 11:18.11 | jeremy_g | syn:ah! cud be the problem! |
| 11:18.12 | RoyK | syn: enable astmm and use 'show memory ....' commands to lookup where the leak is |
| 11:18.24 | syn | well, i have a very short deadline |
| 11:18.29 | *** join/#asterisk vgster (n=vgster@170.252.64.1) |
| 11:18.30 | syn | i can't spend time on the leaks |
| 11:18.39 | syn | the loop shouldn't happen, that's all :) |
| 11:18.46 | syn | and i guess it's the GK's fault |
| 11:19.26 | syn | btw, has anyone looked the res_sqlite addon sent to the BTS ? |
| 11:19.42 | stoffell | backblue: hm, misdn looks much better indeed (using latest asterisk is possible), know if it has good echo cancellation? |
| 11:19.47 | syn | i guess few people care about using sqlite 2 when a res_sqlite3 module is already available |
| 11:20.04 | syn | stoffell: it does |
| 11:20.10 | syn | it performs very well |
| 11:20.21 | syn | i regret the great lack of any documentation and comments about it |
| 11:20.36 | syn | (well, i'm wrong, there is some doc) |
| 11:20.38 | stoffell | syn: okay, cool |
| 11:20.48 | syn | stoffell: i've tested both visdn and misdn |
| 11:20.56 | syn | they both work fine, but mISDN performs better |
| 11:21.10 | stoffell | oh, cool.. gonna try misdn then for a change.. :) |
| 11:21.14 | syn | on the other way, it's more difficult to cross compile and embbed |
| 11:21.29 | syn | and it doesn't work find on big endian systems :( |
| 11:21.49 | syn | it also used to not work with PREEMPT kernels, and SMP of course |
| 11:21.53 | syn | this issue seems fixed now |
| 11:22.07 | stoffell | nice, so it catched up and passed bristuff then :) |
| 11:22.17 | syn | bristuff always sucked :/ |
| 11:22.40 | stoffell | syn: yeah.. bugs :) |
| 11:22.48 | syn | and bad design |
| 11:22.50 | dyn | mISDN is a kernel-level driver or userspace stuff? |
| 11:22.54 | syn | both |
| 11:22.57 | *** join/#asterisk daysmen3 (n=primus@host86-143-5-29.range86-143.btcentralplus.com) |
| 11:23.06 | dyn | and I have to patch my kernel to support mISDN, right? |
| 11:23.10 | syn | that's the purpose of mISDNuser |
| 11:23.12 | syn | no |
| 11:23.17 | syn | you just build modules |
| 11:23.21 | syn | but you need CAPI 2.0 support |
| 11:23.28 | dyn | syn: my kernel config only shows plain isdn and capi configuration options |
| 11:23.34 | syn | dyn: yes |
| 11:23.38 | dyn | at least grep -i misdn did not match :) |
| 11:23.40 | syn | at kernel compile time, it's normal |
| 11:23.53 | syn | you build mISDN after you have your kernel, using kernel headers, to build modules |
| 11:24.00 | syn | as you would do for e.g. nvidia driver |
| 11:24.00 | dyn | ah, gotcha |
| 11:24.03 | dyn | sure thing |
| 11:24.12 | syn | you can also patch the ernel |
| 11:24.15 | syn | but i wouldn't suggest it |
| 11:24.38 | dyn | hmn ubuntu aint have misdn packages, my gentoo desktop does |
| 11:26.38 | backblue | dyn: if you use debian family, you can use the debian script, that it's there too, it does almost everything. |
| 11:27.24 | dyn | backblue: i plan to install the card in an x86 server running ubuntu 6.06, yeah |
| 11:27.37 | dyn | backblue: which script are you referring to? |
| 11:27.52 | backblue | stoffell: it has echo cancelation, and they are developing a new echo cancelation, very much better that the digium one, in software |
| 11:28.03 | backblue | but it's in testing stage. |
| 11:28.15 | backblue | dyn: you have http://www.beronet.com/downloads/install-asterisk.tar.gz |
| 11:28.18 | backblue | just run make. |
| 11:28.35 | backblue | dyn: why you use ubuntu? just use debian on server. |
| 11:28.42 | stoffell | backblue: okay, will give it a try with the latest * version.. |
| 11:28.44 | backblue | or centos or something |
| 11:28.45 | dyn | backblue: *shrug* it's a debian afterall :) |
| 11:28.55 | backblue | stoffell: that script install asterisk too |
| 11:29.04 | backblue | it does everything |
| 11:29.19 | dyn | backblue: so i should use this tarball instead of installing asterisk from the debian repos? |
| 11:29.36 | backblue | dyn: yes i know, but dont make any sense using it, because it's oriented for desktop, debian it's there for years. |
| 11:29.43 | backblue | dyn: yes. |
| 11:29.49 | backblue | use that. |
| 11:30.04 | backblue | you will have 2 errors i think |
| 11:30.09 | backblue | that script has 2 bugs |
| 11:30.19 | backblue | the url for asterisk-sounds it's not ok, i think it has changed |
| 11:30.24 | dyn | backblue: i know, i've been using debian since early '97 |
| 11:30.26 | backblue | and it misses a package |
| 11:30.34 | backblue | but show me the error, when you get there |
| 11:30.46 | backblue | and the linux header package name, it's wrong too. |
| 11:30.54 | *** join/#asterisk inspired (n=mikael@85.221.7.59) |
| 11:30.56 | backblue | dyn: so continue using it! :P |
| 11:31.22 | backblue | we should dedicate our time, to improve what we know, not waste time, on something with a new face. |
| 11:32.01 | backblue | like my grandfather says, "the shit it's all the same, but the smell it's diferent" |
| 11:32.05 | dyn | haha |
| 11:32.19 | dyn | i'm not the type going for ubuntu's funky tools |
| 11:33.04 | backblue | well, i leave up with you |
| 11:33.05 | backblue | work to do now |
| 11:33.05 | dyn | sure |
| 11:37.16 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 11:40.19 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 11:40.57 | puzzled | morning |
| 11:45.55 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 11:47.25 | *** join/#asterisk budairc (n=chatzill@200.215.57.174) |
| 11:48.22 | *** join/#asterisk tparcina (n=tparcina@lns01-0556.dsl.iskon.hr) |
| 11:52.54 | *** join/#asterisk gardo (n=gardo@124.104.32.152) |
| 11:53.27 | jeremy_g | the book says, for 5 channels, a 486 would do fine for asterisk? but what if the channels are all sip i.e. no zaptel/zapata module loading, whats the performance penalty then? |
| 11:53.53 | E-bola | i run asterisk on a 900 mhz p3 |
| 11:53.57 | E-bola | with sip to sip |
| 11:54.00 | E-bola | theres no load at all |
| 11:54.01 | jeremy_g | how lighter are the sip channels over ethernet card as compared to zapata/zaptel |
| 11:54.14 | jeremy_g | kewl! |
| 11:54.21 | E-bola | with sip-sip its like a router |
| 11:54.35 | E-bola | unless u wanna start recording or making conferences or stuff like that |
| 11:54.39 | E-bola | there should be very little load |
| 11:54.51 | florz | jeremy_g: With that number of channels, all that counts is any recoding taking place |
| 11:54.54 | jeremy_g | E-bola:thats good news |
| 11:55.01 | gardo | how do i see what codecs are supported on my asterisk box? |
| 11:55.06 | jeremy_g | little load is good :) |
| 11:55.40 | florz | jeremy_g: a 486 won't be able to do any recoding other than ulaw <-> alaw |
| 11:55.41 | jeremy_g | gardo:type show codec or sth reasonable at asterisk console |
| 11:55.56 | jeremy_g | jeremy_g:yup |
| 11:56.16 | jeremy_g | how expensive in terms of performance is it to do an inband dtmf |
| 11:56.20 | *** part/#asterisk l-fy (n=l-fy@yate/developer/l-fy) |
| 11:56.46 | jeremy_g | sorry! silly me addressing myself |
| 11:56.54 | jeremy_g | :-> |
| 11:57.10 | gardo | jeremy_g: i mean show all the codecs supported on my asterisk box |
| 11:57.22 | jeremy_g | load modules;) |
| 11:57.30 | *** join/#asterisk xnon (i=xnon@200.82.223.42) |
| 11:57.34 | jeremy_g | check out the codec modules that exist and can be be loaded |
| 11:57.35 | xnon | hello friends |
| 11:57.42 | jeremy_g | hello friend :) |
| 11:58.04 | xnon | i cant understand this error: Sep 18 13:33:04 WARNING[9713]: chan_zap.c:10172 setup_zap: Ignoring |
| 11:58.32 | xnon | is a zaptel error compilation in the instalation |
| 11:58.41 | xnon | maybe? |
| 12:00.00 | Arno[Slack] | just a little question, I have Cisco phones using sccp (phones are : 796X, 794X and 7914) do you think I can use the skinny.conf to configure both lines AND buttons on thoses phones ? |
| 12:00.18 | jeremy_g | no clue |
| 12:00.30 | Arno[Slack] | :'( |
| 12:00.33 | jeremy_g | do they support it? |
| 12:00.37 | jeremy_g | :) |
| 12:00.56 | Arno[Slack] | i don't want to use chan_sccp and the sccp.conf file |
| 12:01.36 | Arno[Slack] | i'm very new with sccp and if i have all understood we can't do everything with the skinny.conf file |
| 12:02.21 | *** join/#asterisk binary- (n=marty@203-217-54-190.dyn.iinet.net.au) |
| 12:03.31 | tparcina | question, auto attendent answers a call, when calling person press number the call is ringing on particular extension. my problem is that caller doen't hear that extension is ringing. what to put in dialplan to make caller hear ringing? |
| 12:04.17 | tparcina | i have tried to put "r" on Dial when its calling extension, but it didn't help |
| 12:04.32 | backblue | tparcina: that it's done by the sip phone, or by asterisk doing maybe like playtones or something |
| 12:05.28 | tparcina | backblue: can you send me example? |
| 12:06.11 | tparcina | backblue: btw, how can it be done by sip phone? |
| 12:08.27 | backblue | well, when the call it's ringing, your phone gets a header saying it's ringing and can play some file you want, like your favority song or something, and that will be your ring tone. :) |
| 12:08.37 | *** join/#asterisk Cyt (n=danielcy@athedsl-141532.otenet.gr) |
| 12:09.01 | backblue | tparcina: you should look at voip-info they have examples of playtones, i cannot do it right now |
| 12:11.22 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:17.26 | *** join/#asterisk SaTLaN32 (n=satlan32@212.150.142.211) |
| 12:17.29 | SaTLaN32 | hi |
| 12:17.45 | SaTLaN32 | anyone know if Dialogic D/4PCI card can work on asterisk? |
| 12:17.52 | SaTLaN32 | 4 analog ports |
| 12:21.09 | jeremy_g | pop quiz :P how advnaced is your softphone? can you change the dtmf mode from iband to rfc2833 or sip info based? can u name such a soft phone that allows this? |
| 12:21.29 | tparcina | backblue: i'm looking on voip-info for playtones, but I don't understand how to dial, then play tones until other person answers, and when person answers to connect the phone call. |
| 12:22.24 | jeremy_g | tparcina:use macros |
| 12:22.38 | jeremy_g | it may not be the best way though |
| 12:23.27 | [TK]D-Fender | *yawn* |
| 12:24.03 | [TK]D-Fender | SaTLaN32: Some Dialogic support is expected with * 1.4 coming out (supposedly) next month. Till then : no |
| 12:24.26 | SaTLaN32 | this is a very old card... |
| 12:24.40 | SaTLaN32 | and i saw somewhere they should be supported |
| 12:25.37 | jeremy_g | i need a soft phone that supports various dtmf modes {rfc2833(both rtp + sip info), inband} |
| 12:25.40 | jeremy_g | plz suggest |
| 12:25.57 | SaTLaN32 | sjphone ? |
| 12:27.46 | RoyK | x-lite? |
| 12:28.58 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 12:28.58 | *** mode/#asterisk [+o denon] by ChanServ |
| 12:31.29 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.244) |
| 12:31.55 | tzafrir | support of dialogic cards is only with ABE |
| 12:34.15 | SaTLaN32 | ?? |
| 12:34.19 | SaTLaN32 | what is ABE |
| 12:34.40 | JT | asterisk business edition |
| 12:35.07 | [TK]D-Fender | LINCOLN! |
| 12:35.16 | SaTLaN32 | which cost money? |
| 12:35.16 | tparcina | jeremy_g: how would macros help me? |
| 12:36.06 | *** join/#asterisk Rhiliam (n=c37d6f4@CPE001310426d31-CM0012256ea75c.cpe.net.cable.rogers.com) |
| 12:36.27 | *** join/#asterisk MadRio (n=rio@80.255.176.161) |
| 12:36.48 | [TK]D-Fender | tparcina: "show application dial". Use the Macro parameter |
| 12:36.49 | MadRio | hello, can I speak with some asterisk developer? |
| 12:36.50 | tparcina | Fender know an answer for my problem, don't you? Auto attendent menu ansvers the call. caller press number and one extensions rings. problem is that caller doesn't hear ringing. |
| 12:37.34 | *** join/#asterisk angryuser (n=aster@i03v-213-44-169-43.d4.club-internet.fr) |
| 12:37.59 | [TK]D-Fender | MadRio: Just ask your question and see what kind of answer you get. |
| 12:40.08 | tparcina | Fender: this is from voip-info - M(x): Executes the macro (x) upon connect of the call (i.e. when the called party answers |
| 12:40.26 | Rhiliam | anyone familiar with Bell Canada caller id? |
| 12:40.32 | tparcina | if i got that right, that means that I can't play ringing thone with that one |
| 12:40.57 | MadRio | I need to write radius authentification under h323, we using ooh323, how can I make it? |
| 12:41.00 | [TK]D-Fender | tparcina: Why not? |
| 12:41.36 | Rhiliam | If I plug the PSTN line directly into a phone, I get callerid information, without fail. If I plug the line into an incoming FXO, most of the time the callerid is displayed as "asterisk" for the name and number - any ideas? |
| 12:41.40 | [TK]D-Fender | MadRio: I think you would be better off asking on the asterisk-dev mailing lists for that one. its probably a long discussion |
| 12:42.00 | MadRio | where can I get it? |
| 12:42.15 | [TK]D-Fender | Rhiliam: Pastebin your zapata.conf and please tell us what card you have |
| 12:42.49 | [TK]D-Fender | MadRio: http://lists.digium.com/mailman/listinfo/asterisk-dev |
| 12:43.29 | MadRio | thanx |
| 12:43.38 | Rhiliam | I have a digium TDM card |
| 12:44.11 | Rhiliam | not sure what pastebin is tho (new to irc) |
| 12:44.23 | [TK]D-Fender | Rhiliam: The model please.... |
| 12:44.24 | Druken | www.pastebin.ca |
| 12:44.25 | [TK]D-Fender | ~pb |
| 12:44.35 | jbot | [pb] a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 12:45.06 | tparcina | Fender: because macro will be executed when phone connects (when called party answers). there is no point of playing ringing tone when called person answered |
| 12:45.18 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
| 12:45.41 | [TK]D-Fender | tparcina: What point is there playing tones while the line is ringing then? When else would youplay them? |
| 12:46.21 | Rhiliam | Card is a TDM02B |
| 12:46.54 | Rhiliam | http://pastebin.ca/175011 |
| 12:47.37 | tparcina | Fender: you dind't understand. I need to play tones to person that is placing call so that he knows that other person phone is ringing. |
| 12:48.29 | Rhiliam | btw, asterisk version is 1.2.10 |
| 12:48.38 | [TK]D-Fender | Rhiliam: fixed version : http://pastebin.ca/175014 |
| 12:48.42 | *** join/#asterisk Seyr (n=Seyr@cpe-67-10-139-141.houston.res.rr.com) |
| 12:48.44 | Druken | tparcina: why tones? the ring not work anymore? |
| 12:49.35 | tparcina | Druken: Auto attendent menu ansvers the call. caller press number and one extensions rings. problem is that caller doesn't hear ringing. |
| 12:49.57 | [TK]D-Fender | tparcina: You shouldn't need to indicate progress on the line. Describe the hardware/protocols involved in this call. |
| 12:50.23 | *** join/#asterisk amir (n=amir@gentoo/developer/amir) |
| 12:50.25 | tparcina | I understand that caller can't hear the ringing because he has a call that is allready established |
| 12:50.49 | [TK]D-Fender | Rhiliam: Your 1st list didn't have "asreceived" thats what it was only SOME calls that worked, because it was the overflow to port 2. that was good. |
| 12:50.50 | amir | did someone already encounter this: udev_make_node: mknod(/dev/capi/1, 020660, 191, 29) failed: Operation not permitted ? |
| 12:50.56 | tparcina | but, i need to pass him somhow information that extension is ringigng. question is - how to do that? |
| 12:50.58 | *** join/#asterisk ambriento (n=melcon@200.192.160.100) |
| 12:51.12 | amir | i seem no longer able to correctly load capi... the folder /dev/capi/* stays empty ;( |
| 12:51.21 | [TK]D-Fender | tparcina: Try the "r" parameter in your Dial command |
| 12:51.32 | Druken | tparcina: when the voip phone rings, the caller should hear a ring, reguardless.... |
| 12:51.36 | tparcina | Fender: have tried, doesn't help |
| 12:52.10 | Rhiliam | Fender, line 3 & 4 are diffferent incoming PSTN lines. |
| 12:52.10 | tparcina | Drunken: well, caller doesn't hear anything |
| 12:52.22 | Rhiliam | sorry channel 3 & 4 |
| 12:52.46 | [TK]D-Fender | Rhiliam: I see that. just that channel 3 shouldn't get your anywahere for CID |
| 12:53.58 | Rhiliam | correct, channel 3 does not have callerid on the line, hence the reason I set it manually, channel 4 however does. I think I have the zapata.conf configured correctly, but don't understand why callerid on channel 4 is barely working. |
| 12:54.37 | Druken | Rhiliam: do you have a wait(2) BEFORE the answer? |
| 12:54.43 | Rhiliam | no |
| 12:54.54 | Druken | wow, problem solved.... |
| 12:54.55 | amir | a mknod in /dev/capi fails as well if done by hand ;( |
| 12:54.56 | [TK]D-Fender | Rhiliam: try setting your gains to 0.0 if that doesn't help, then try swapping your 2 modules. maybe your #4 is flakey. |
| 12:54.59 | amir | mknod: `/dev/capi/gna': Operation not permitted |
| 12:55.11 | amir | any capi specialist around? |
| 12:55.14 | Rhiliam | Ahh, I see :), Asterisk is answer the phone, before the callerid info comes across? |
| 12:55.26 | Druken | ding ding ding, we have a winner! |
| 12:55.52 | [TK]D-Fender | Rhiliam: Well add the "usercallerid=yes" as well, and it should by default wait until the 2nd ring before answering. |
| 12:56.10 | [TK]D-Fender | Druken: You don't need Wait(2) for CID.... |
| 12:56.29 | [TK]D-Fender | Druken: * waits automatically if its expecting CID. |
| 12:56.49 | Druken | [TK]D-Fender: well, i'm an old school kinda guy, and i use it on my system... doesn't affect it at all :) |
| 12:56.50 | [TK]D-Fender | Druken: And you'd have to tell * to ignore CID if you wanted to pickup immediately |
| 12:57.09 | *** join/#asterisk festr__ (n=festr@ns.regnet.cz) |
| 12:57.14 | Rhiliam | I did have usercallerid=yes as well, but it didn't change anything. I will add Wait(2) at the start of the context and then answer. |
| 12:57.52 | festr__ | hello, any asterisk developer here? i've found some bug in my latest svn 1.2 and need some help with debugging |
| 12:58.03 | Druken | 2 seconds never hurt anyone... cept a skydiver without a parachute during the last 100 FT |
| 12:58.10 | [TK]D-Fender | Rhiliam: Sure, and should that not do it I would then try swapping modules. Both of these however I would do after setting your gains back to 0 |
| 12:58.36 | [TK]D-Fender | Druken: Putting the "terminal" back in front of velocity where it belongs! |
| 12:59.01 | [TK]D-Fender | Druken: Some people just don't understand the gravity of their situations! |
| 12:59.03 | Druken | :P |
| 12:59.16 | Rhiliam | Thanks will give this a try - appreciate the help |
| 13:00.10 | [TK]D-Fender | Anyone out there have a good "BEEP" recording to use as a "paging" indication? |
| 13:00.13 | Druken | oh joy... i get to goto the college and gawk today... sweet! |
| 13:00.40 | Druken | what's wrong with beep ? |
| 13:00.42 | Druken | hehehe |
| 13:00.50 | jeremy_g | nopes, sjphones doesnt support any out of band dtmf :( |
| 13:01.03 | [TK]D-Fender | Druken: I need a seperate recording file I can load onto my phones. |
| 13:01.08 | amir | am i the only one seeing: udevd-event[21880]: udev_make_node: mknod(/dev/capi/14, 020660, 191, 14) failed: Operation not permitted ? |
| 13:01.20 | [TK]D-Fender | Druken: Has to be played before the auto-answer kicks in. |
| 13:01.51 | [TK]D-Fender | amir: Know virtually nothing I would still venture a gueess that the process calling it doesn't have UDEV rights... |
| 13:02.12 | [TK]D-Fender | amir: Or is in confict with something else. |
| 13:02.41 | Druken | [TK]D-Fender: i say i record yourself, "Answer the god damn phone people... Hello??" |
| 13:03.23 | amir | [TK]D-Fender: it happens if i do a 'modprobe capi' as root by hand as well |
| 13:03.29 | [TK]D-Fender | Druken: This is for paging... I have a silent ring one for just popping in unanounced and screaming at people ;) Double the fun when you do it muted just to spy ;) |
| 13:03.32 | jeremy_g | festr__:which release |
| 13:04.53 | tparcina | ok, playing ring tones - i give up on that. now i have another question. how to change language for agent? |
| 13:05.08 | Druken | [TK]D-Fender: it have to be a certain type of file? |
| 13:05.48 | tparcina | when agent has incoming call, queue reads to him how long that person has been in queue, but it allways read on english (en). is it possible to change language? |
| 13:06.11 | jeremy_g | tparcina:yup |
| 13:06.27 | tparcina | jeremy_g: well, how? |
| 13:06.55 | MadRio | <[TK]D-Fender> astrerisk-dev is some about 38 Mb!!!! may be u can advise some other way? |
| 13:07.29 | tparcina | please don't tell me to put my sound over sound in /var/lib/asteris/sounds :)) i prefer to kkep it where it should be under hr directora ;)) |
| 13:08.11 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:08.46 | tparcina | jeremy_g: are you there? |
| 13:08.46 | jeremy_g | tparcina:yup |
| 13:08.46 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 13:08.53 | tparcina | jeremy_g: well, how to change language for agent? |
| 13:08.55 | jeremy_g | i am so lost in my out of band dtmf testing |
| 13:09.16 | ambriento | he's thinking a g00d answer tparcina :) |
| 13:09.34 | tparcina | jeremy_g: what't problem do you have with dtmf? |
| 13:10.14 | jeremy_g | i need a soft phone that supports out of band rfc2833 dtmf |
| 13:10.26 | tparcina | ambriento: well, he is my best shut so far |
| 13:10.34 | tparcina | sj phone supports it |
| 13:11.07 | jeremy_g | tparcina: where exactly? audio-> dtmf..? |
| 13:11.08 | tparcina | jeremy_g: you have said that it's possible to change language for agent. can you please tell me how to change? |
| 13:11.13 | amir | damn... even crappy capiinfo works... it's just the devices missing ;(( |
| 13:11.44 | tparcina | jeremy_g: profile => edit => dtmf |
| 13:12.20 | jeremy_g | damn! :) so cute it was hiding over there |
| 13:12.28 | jeremy_g | now i must help u tparcina |
| 13:12.47 | *** join/#asterisk Dandre (n=Dandre@was59-3-82-236-48-30.fbx.proxad.net) |
| 13:12.48 | tparcina | jeremy_g: yes, you do :)) |
| 13:12.52 | Dandre | Hello all, |
| 13:13.02 | tparcina | hello dandre |
| 13:13.45 | [TK]D-Fender | MadRio: I didn't say to DOWNLOAD to list, just to post a message to it |
| 13:14.07 | tparcina | jeremy_g: in this century :))) |
| 13:14.14 | Dandre | I wonder to know if there is some configuration option to place a timeout between line hangup and listening for incomming calls on a tdm400 card |
| 13:14.24 | [TK]D-Fender | Druken: Any audio file I guess. I can convert it myself. |
| 13:14.54 | ambriento | hahahha |
| 13:15.28 | jeremy_g | :) |
| 13:15.39 | [TK]D-Fender | tparcina: If its not playing the file you want then its not in the right place with the right name. |
| 13:16.03 | jeremy_g | tparcina:there is this language directive: language () |
| 13:16.11 | tparcina | Fender, its playing the file, but on wrong language |
| 13:16.16 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 13:16.18 | jeremy_g | tparcina: language=en |
| 13:16.29 | *** join/#asterisk liamo (n=liam@86.43.74.94) |
| 13:16.32 | tparcina | jeremy_g: yes, but where to put that for agent? |
| 13:16.37 | MadRio | <[TK]D-Fender> thanx, I start reading it |
| 13:16.37 | [TK]D-Fender | tparcina: pastebin CLI output for the ENTIRE call, and show every file involved with the call. |
| 13:16.53 | tparcina | Fender, wait a min |
| 13:17.15 | jeremy_g | put in some relevant context |
| 13:17.16 | [TK]D-Fender | MadRio: I didn't say you should go searching in there randomly. I said you should just POST a message of your own introducting the project you'd like some help with |
| 13:18.00 | MadRio | tanx |
| 13:18.20 | jeremy_g | tparcina: e.g. in sip.conf [account1] language=no |
| 13:18.36 | jeremy_g | and u can also use it in extension.conf |
| 13:18.43 | *** join/#asterisk Ox0F0-0FF (n=pierre@200.216.238.226) |
| 13:18.51 | jeremy_g | um not 100% sure |
| 13:19.02 | ambriento | tparcina: what did you do so far? |
| 13:19.07 | tparcina | Fender: this is my problem - http://pastebin.ca/175040 - can't change en to hr. simply don't know where to do that for agent |
| 13:19.57 | tparcina | ambriento: about what? - "what did you do so far?" |
| 13:20.02 | [TK]D-Fender | tparcina: I said the WHOLE CALL! |
| 13:20.27 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 13:20.50 | *** join/#asterisk mercestes (n=merceste@216.54.143.242) |
| 13:21.06 | *** join/#asterisk Cyt (n=danielcy@athedsl-141532.otenet.gr) |
| 13:21.49 | tparcina | Fender: this is the whole call - http://pastebin.ca/175043 - sorry |
| 13:22.03 | ambriento | tparcina: about your issue. |
| 13:22.28 | *** join/#asterisk muppetmaster (n=jasongoe@81.184.73.169) |
| 13:22.43 | [TK]D-Fender | tparcina: That looks like part of a queue call.... |
| 13:23.41 | tparcina | ambriento: nothing, because I don't know where to put language=hr or SetLanguage(hr) |
| 13:23.42 | syn | it was a misconfiguration in gnugk |
| 13:23.45 | [TK]D-Fender | tparcina: I said the entire call including everything BEFORE you enter the queue. |
| 13:23.48 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 13:23.48 | syn | have a nice day :) |
| 13:23.52 | *** part/#asterisk syn (i=syn@kenobi.sceen.net) |
| 13:24.05 | [TK]D-Fender | tparcina: You put it BEFORE the queue is called obviously! |
| 13:24.12 | [TK]D-Fender | \ugh! |
| 13:24.35 | muppetmaster | Hello everyone |
| 13:24.37 | tparcina | Fender: exacly. phone call is coming to agent (from queue) and report of hold time is told on english. how to change it to another language? |
| 13:24.43 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 13:25.09 | tparcina | Fender: yes, but that it's changed for the person that is calling the queue not for the agent |
| 13:25.38 | tparcina | Fender: persons that are calling the queue hear other language messages - that works fine |
| 13:25.44 | [TK]D-Fender | tparcina: Last time : pastebin EVERYTHING related to the entire call. |
| 13:25.58 | mercestes | What's the language code for Klingon? |
| 13:26.05 | [TK]D-Fender | tparcina: Agents don't have a language |
| 13:26.19 | ambriento | thats his issue [TK]D-Fender |
| 13:26.32 | *** join/#asterisk De_Mon (n=de_mon@fl-69-69-144-77.dyn.embarqhsd.net) |
| 13:26.38 | Dandre | I wonder to know if there is some configuration option to place a timeout between line hangup and listening for incomming calls on a tdm400 card |
| 13:26.51 | tparcina | Fender: that is stupid. why do agents need to hear message on english? |
| 13:28.06 | jeremy_g | tparcina:agents can hear a message in something else as well |
| 13:28.29 | jeremy_g | tparcina:if all fails, set up an agi script and use a tts for ur language |
| 13:29.51 | tparcina | Fender: here is complitly complete call - http://pastebin.ca/175052 |
| 13:30.28 | jeremy_g | <PROTECTED> |
| 13:30.29 | jeremy_g | it differs from the current language of the channel. |
| 13:30.48 | tparcina | jeremy_g: what all (if all fials)? I haven't treied anything yet :)) |
| 13:30.59 | festr__ | jeremy_g: 1.2 latest svn |
| 13:31.22 | Dandre | I am faced to a problem that I don't know how to solve: |
| 13:31.29 | tparcina | jeremy_g: this with background miht work, let me check |
| 13:32.04 | jeremy_g | tparcina:i dont think so! |
| 13:32.16 | *** join/#asterisk UnderMine (n=paddy@host81-149-176-190.in-addr.btopenworld.com) |
| 13:32.19 | *** join/#asterisk RestLessGemini (n=rLg@202.61.49.248) |
| 13:32.23 | jeremy_g | tparcina:its only when u want to temporarily switch to a language other than the default i guess |
| 13:32.49 | tparcina | jeremy_g: you right. problem is that * is the one who makes phone call to agent. and I don't ahve dialplan for that so I can't define language |
| 13:33.11 | jeremy_g | :) [TK]D-Fender is za man |
| 13:33.19 | jeremy_g | as i said, he was right |
| 13:33.41 | festr__ | jeremy_g: i've problem with 100% system load with res_features.c i need some helo with debug this |
| 13:34.01 | [TK]D-Fender | tparcina: AGENTS don't get those messages! Its callers in the queue! |
| 13:34.22 | tparcina | Fneder: you got that wrong |
| 13:34.25 | Dandre | I have an inbound line connected to a tdm400 card and some sip phones. If I hangup a phone, the call seems to be active and this genrates another inbound call for about 1 second. I think that this is some polarity switch problem. Any help? |
| 13:34.31 | ambriento | [TK]D-Fender: yes they do. |
| 13:34.56 | ambriento | [TK]D-Fender: there's an option that tells agents how long the caller was waiting in line |
| 13:35.26 | tparcina | Fender: in queues.conf you have line - queue-reporthold = queue-reporthold, and that agent will know for how long caller has been in queue |
| 13:35.31 | ambriento | tparcina: do you have static members on that queue or are you using agentcallback? |
| 13:35.41 | festr__ | if anyone interested: as soon as i press DTMF which should activate feature in featuremap asterisk consume 100% system load. i've debugged thread and there is some fast loops with gettimeofday (difference between each call is about 60 microseconds) i attached it with gdb: http://pastebin.ca/175047 |
| 13:35.58 | tparcina | ambriento: static members |
| 13:36.22 | ambriento | and in your sip.conf do you have a laguage for them? |
| 13:36.40 | tparcina | in sip.conf in general section i have language=hr |
| 13:37.16 | tparcina | but it's for when that sip user is trying to establish the phone call, not when it's receiving phone call. and agent is receiving phone call from * |
| 13:37.49 | tparcina | I thing this is something that should be changed in asterisk code |
| 13:37.59 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 13:38.23 | tparcina | there should be option to define language for agents |
| 13:39.07 | *** join/#asterisk `Sauron (i=sauron@h-69-3-12-50.hstqtx02.covad.net) |
| 13:39.14 | ambriento | tparcina: if your using static members, your not using chan_agent |
| 13:39.29 | ambriento | you're* |
| 13:39.38 | [TK]D-Fender | tparcina: How about setting the language before the call ends up in queue, and how about pasting that agents SIP entry. I did say paste EVERYTHING related to that call. |
| 13:39.58 | tparcina | ambriento: a didn't understand this with can_agent |
| 13:40.38 | [TK]D-Fender | tparcina: Ok, I don't want to pick this apart the hard way. PM |
| 13:40.43 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 13:40.43 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:41.12 | Dandre | I am faced to a problem that I don't know how to solve: |
| 13:41.13 | Dandre | I have an inbound line connected to a tdm400 card and some sip phones. If I hangup a phone, the call seems to be active and this genrates another inbound call for about 1 second. I think that this is some polarity switch problem. |
| 13:41.15 | Dandre | Any idea to trace it down? |
| 13:43.21 | ambriento | Dandre: try pastebin the entire output |
| 13:44.04 | *** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk) |
| 13:44.06 | tparcina | Fender: why should I set the language before call ends in queue? I'm not trying to change language on call that is entering the queue (that language is set up well), I'm trying to change lagnguage for agent. |
| 13:44.12 | Dandre | which output ambriento ? |
| 13:44.43 | *** join/#asterisk f0urtyfive (i=f0urtyfi@c-67-165-5-232.hsd1.pa.comcast.net) |
| 13:44.49 | ambriento | Dandre: the CLI*> output |
| 13:44.57 | tparcina | Fender: here is sip.conf for that agent - http://pastebin.ca/175062 |
| 13:46.19 | *** join/#asterisk docelmo (n=vircuser@55-65.126-70.tampabay.res.rr.com) |
| 13:48.11 | tparcina | looks like I wont solve this one nither... :(( today, it just isn't my day |
| 13:51.38 | *** join/#asterisk MadRio (n=rio@80.255.176.161) |
| 13:53.21 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 13:53.54 | Dandre | ambriento: here it is: |
| 13:53.56 | Dandre | http://pastebin.ca/175070 |
| 13:54.08 | ambriento | Don't be so negative tparcina :) |
| 13:55.43 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 13:56.55 | tparcina | ambriento: there is one goot thing. it's 16:00, my working hour is over, I'm going home :))))) - best thing that has happend to me today :)) |
| 13:57.09 | tparcina | talk with you all some other day! |
| 13:57.14 | *** join/#asterisk edguy3 (n=edguy@host-208-115-200-88.patmedia.net) |
| 13:57.38 | ambriento | :) |
| 13:57.39 | *** join/#asterisk unixgeek (n=unixgeek@216-220-234-197.exploremaine.com) |
| 13:57.46 | *** join/#asterisk Gunnar (n=gunnar@101.Red-80-24-171.staticIP.rima-tde.net) |
| 13:58.06 | ambriento | I'll try to think in somthing to help you tparcina. |
| 14:00.16 | [TK]D-Fender | tparcina: Pastebin "ls -l /var/lib/asterisk/sounds" and "ls -l /var/lib/asterisk/sounds/hr" |
| 14:01.03 | tparcina | fender: don't bother, will solve that one tomorow |
| 14:01.09 | tparcina | thank you all, bye! |
| 14:01.14 | *** part/#asterisk tparcina (n=tparcina@lns01-0556.dsl.iskon.hr) |
| 14:03.34 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-246-145.buckeyecom.net) |
| 14:04.19 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 14:04.29 | gambolputty | Hi. I would like to know if Asterisk 1.2.6 is compatible with Addons 1.2.2. |
| 14:05.19 | ambriento | gambolputty: yes |
| 14:05.48 | gambolputty | ok |
| 14:06.11 | gambolputty | you use both together? |
| 14:06.47 | ambriento | gambolputty: yes. as the name says, its add-ons :) |
| 14:07.31 | ambriento | Dandre: Zap/6 is a FXO port? |
| 14:07.44 | *** join/#asterisk Rhiliam (n=c37d6f4@CPE001310426d31-CM0012256ea75c.cpe.net.cable.rogers.com) |
| 14:08.50 | Rhiliam | is there a way to automatically record an incoming conversation on a zap channel? Essentially, what I would like to do is if the callerid = xxxxxx then automatically record the call |
| 14:10.03 | Dandre | ambriento: part of my /etc/asterisk/zapata.conf : http://pastebin.ca/175083 |
| 14:10.51 | [TK]D-Fender | Rhiliam: Do a GotoIf on your inbound context based on CID and do whatever you like. |
| 14:12.48 | *** part/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 14:12.59 | *** join/#asterisk Gourou69 (n=test@par69-3-82-224-162-203.fbx.proxad.net) |
| 14:15.20 | UnderMine | Anyone any thoughts as to why inbound callerid not working on BRI channels? |
| 14:16.20 | RestLessGemini | hello room, does anyone succeeded in installing astribank-8 on Centos 4.3, kernel 2.6.9-34? |
| 14:16.58 | *** join/#asterisk foxxtrot (n=craig@c-67-185-55-194.hsd1.wa.comcast.net) |
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| 14:24.30 | *** join/#asterisk saftsack (n=saftsack@p54A7E055.dip.t-dialin.net) |
| 14:25.26 | Dandre | ambriento: yes Zap/6 is fxo |
| 14:26.13 | *** join/#asterisk FlatFoot (n=simon@80.88.192.113) |
| 14:26.43 | FlatFoot | good afternoon all |
| 14:27.22 | *** join/#asterisk riddlebox (n=blah@24-207-167-28.dhcp.stls.mo.charter.com) |
| 14:28.06 | riddlebox | hello, how far can the digium cards push dialtone? |
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| 14:34.32 | *** mode/#asterisk [+o twisted[asteria]] by ChanServ |
| 14:38.42 | ambriento | Dandre: i got it |
| 14:39.18 | UnderMine | Correct me if i'm wrong but shouldn't isdn to isdn not echo? |
| 14:39.26 | Dandre | you have found something wrong ? |
| 14:40.20 | *** join/#asterisk dasenjo (n=dasenjo@63.245.86.95) |
| 14:40.27 | dasenjo | Hi everybody! |
| 14:40.42 | coppice | UnderMine: depends. is the link pure ISDN, with a real ISDN phone at the far end? is speakerphone in use at the far end? |
| 14:41.56 | ambriento | Dandre: not really |
| 14:43.45 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
| 14:44.37 | UnderMine | coppice: SIP->ISDN2e->BT network->ISDN30->ISDN Phone |
| 14:45.16 | *** join/#asterisk manopulus (n=manopulu@cable-10-68.cgates.lt) |
| 14:45.48 | coppice | who hears the echo? |
| 14:45.52 | manopulus | hello, can i use Set(account,12345678) to change accountcode (it is related to billing, and i want to overwrite ) ? |
| 14:46.01 | *** join/#asterisk Juggie (n=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
| 14:46.13 | UnderMine | coppice: The SIP end with a 1/2 second delay |
| 14:47.20 | dasenjo | I was reading the voip-info.org terms of service and .. I did not like them at all |
| 14:47.48 | dasenjo | voip-info archives the main information about configuring and deploying an asterisk pbx |
| 14:47.49 | [TK]D-Fender | dasenjo: As in GPL? |
| 14:47.58 | *** join/#asterisk abelsgmx (n=abelsgmx@189.130.200.39) |
| 14:48.04 | *** join/#asterisk Ox0F0-0FF (n=pierre@200.216.238.226) |
| 14:48.05 | dasenjo | [TK]D-Fender, yes |
| 14:48.25 | dasenjo | a wiki with this info should be more GPL-friendly |
| 14:48.31 | dasenjo | don't you think? |
| 14:48.32 | [TK]D-Fender | dasenjo: Says who? Who owns Voip-Info.org? HMM? How much did you pay for*? What licence does it come with? What does it entitle you to? |
| 14:48.45 | *** join/#asterisk zaf (n=zaf@69.1.161.36) |
| 14:49.29 | [TK]D-Fender | dasenjo: I can write a book on advanced space propulsion knwing exactly NOTHING. Just because Yyou feel that my book means everythingis irrelevent. |
| 14:49.51 | [TK]D-Fender | dasenjo: So what's your real problem with it? |
| 14:50.09 | dasenjo | it is a great wiki |
| 14:50.42 | dasenjo | but the knowlegde on it was created by the community and now is owned by .. :? |
| 14:51.19 | abelsgmx | hi guys |
| 14:51.26 | [TK]D-Fender | dasenjo: Well you've just DEFINED what a WIKI is.... and owned my the owner of the site I'm guessing. |
| 14:52.15 | dasenjo | I don't have nothing particullary against voip-info or they owners .. but just .. it would be great to have a free (as in freedom) repository for the asterisk management and configuration |
| 14:52.16 | *** join/#asterisk asjdhsadjh2 (n=qrao@pool-71-245-225-219.bstnma.fios.verizon.net) |
| 14:52.22 | abelsgmx | some one know, where i can info about the basic linux demons to run Asterisk (excluding ZAP, Asterisk...) |
| 14:52.32 | asjdhsadjh2 | Didn't get a frame from channel: SIP/bard-2b72c2f |
| 14:52.39 | mercestes | For the Wiki to allow posters to retain ownership of the "knowledge" they provide could (would) make them liable for that intellectual property. |
| 14:52.41 | asjdhsadjh2 | Bridge stops bridging channels SIP/bard-2b72cf |
| 14:52.44 | dasenjo | I assumed another terms of use .. and realize of the truth yesterday .. |
| 14:52.47 | asjdhsadjh2 | what is this didn't get a frame from chan error? |
| 14:53.12 | mercestes | It's what is known as a "CYA" in America. |
| 14:53.17 | [TK]D-Fender | dasenjo: SOMEONE has to operate that site. What would you propose? |
| 14:53.28 | mercestes | abelsgmx: rc-update??? |
| 14:54.18 | dasenjo | [TK]D-Fender, maybe .. wiki.asterisk.org or asterisk-wiki.digium.com or something like that .. |
| 14:54.25 | dasenjo | [TK]D-Fender, searching .. |
| 14:54.45 | [TK]D-Fender | dasenjo: And who owns asterisk.org? ......... I don't think your getting any further with this idea.... |
| 14:55.03 | dasenjo | maybe not .. |
| 14:55.17 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 14:55.29 | [TK]D-Fender | dasenjo: What you are looking for does not exist in any form I can name an example of. |
| 14:55.32 | dasenjo | I'm going to write a mail, but I have to search a think better before |
| 14:56.02 | *** join/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 14:56.14 | dasenjo | [TK]D-Fender, do you know asteriskdocs.org? |
| 14:56.24 | [TK]D-Fender | dasenjo: Yeah, what about them? |
| 14:56.27 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 14:56.27 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:57.01 | dasenjo | they can be the free the free replacement for voip-info in the future .. |
| 14:57.20 | Qwell | dasenjo: except that isn't their goal... |
| 14:57.58 | mercestes | Maybe I'm retarded but..........what exactly is the issue here?? |
| 14:57.58 | *** join/#asterisk GiabboO (n=GiabboO@host-84-220-175-143.cust-adsl.tiscali.it) |
| 14:58.29 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 14:58.31 | dasenjo | Qwell, yeah .. maybe not asteriskdocs is not a wiki |
| 14:59.36 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.72.14) |
| 14:59.40 | [TK]D-Fender | dasenjo: And what makes them any "free-er"? |
| 15:01.25 | RoyK | ~sangoma |
| 15:01.27 | jbot | extra, extra, read all about it, sangoma is a company that makes PRI cards the way Digium should have done it in the first place.... |
| 15:01.29 | *** join/#asterisk corresponder (n=correspo@p54AD6C2D.dip.t-dialin.net) |
| 15:01.32 | corresponder | hi there |
| 15:02.02 | mercestes | *LMAO |
| 15:02.19 | corresponder | everytime i establish an isdn connetcion over asterisk i get a kernel panic ( i guess, the module of the avm card is the problem) is this a known bug? |
| 15:02.48 | Qwell | corresponder: No, but it seems like you've discovered the best ex-girlfriend script ever |
| 15:02.55 | corresponder | *g* |
| 15:03.05 | stoffell | corresponder: probably buggy driver? |
| 15:03.08 | RoyK | corresponder: rotfl |
| 15:03.12 | corresponder | misdn |
| 15:03.16 | corresponder | hmmm |
| 15:03.24 | RoyK | corresponder: trunk or 1.2? |
| 15:03.25 | corresponder | kernel 2.6.15 - 2.6.17 |
| 15:03.28 | corresponder | 1.2.12 |
| 15:03.41 | *** join/#asterisk mistral (n=mistral@jstevenson.plus.com) |
| 15:03.46 | mistral | would somebody be able to shed some light on why this wont work ? |
| 15:03.49 | mistral | exten => _0NXXXXXXXXX,1,Dial(CAPI/ISDN/${EXTEN}/B,,) |
| 15:03.59 | RoyK | corresponder: what sort of panic is it? |
| 15:04.08 | RoyK | erm |
| 15:04.11 | RoyK | mistral: not isdn |
| 15:04.16 | corresponder | im on the phone |
| 15:04.29 | mistral | royk: the item is labelled ISDN in capi.conf |
| 15:04.40 | RoyK | oh |
| 15:04.40 | RoyK | ok |
| 15:05.28 | asjdhsadjh2 | Didn't get a frame from channel: SIP/bard-2b72c2f |
| 15:05.30 | asjdhsadjh2 | why is that happening |
| 15:05.49 | mistral | royk: i was working with misdn on another box but gave up because i cannot get any audio in / out of the phone lines |
| 15:06.10 | mistral | any ideas what would cause that ? |
| 15:06.38 | mistral | umm or should i say. I can get asterisk <-> isdn eg voicemail / playbacks etc.. |
| 15:06.52 | mistral | but when i try to get sip or iax to tlak to misdn it messes up |
| 15:07.03 | RoyK | what sort of hardware? |
| 15:07.33 | mistral | Fritz |
| 15:07.54 | mistral | aka 0000:02:08.0 Network controller: AVM Audiovisuelles MKTG & Computer System GmbH A1 ISDN [Fritz] (rev 02) |
| 15:13.31 | gardo | how do i see what codecs are supported by my asterisk box? |
| 15:13.40 | Qwell | gardo: show translation |
| 15:13.42 | gardo | show codecs just lists the codecs |
| 15:14.09 | gardo | Qwell: thanks! |
| 15:14.50 | gardo | Qwell: g723 - - - - - - - - - - - |
| 15:15.00 | gardo | Qwell: that line means i don't have g723 support? |
| 15:15.03 | *** part/#asterisk RestLessGemini (n=rLg@202.61.49.248) |
| 15:15.13 | Qwell | right, because asterisk doesn't have a g723 codec |
| 15:15.39 | GiabboO | back |
| 15:15.41 | GiabboO | hi all |
| 15:15.45 | [TK]D-Fender | gardo: G723 is heavily patent encumbered. |
| 15:15.55 | GiabboO | RoyK, are you there ? |
| 15:15.59 | gardo | [TK]D-Fender: darn! |
| 15:16.04 | [TK]D-Fender | gardo: And therefor not eligible for redistribution in free format |
| 15:16.16 | gardo | [TK]D-Fender: thats what our provider is using |
| 15:16.22 | [TK]D-Fender | gardo: It is possible to acquire it for * though |
| 15:16.24 | RoyK | GiabboO: yes |
| 15:16.33 | Qwell | gardo: soon, Digium will have a transcoder card available, which can do g723 |
| 15:16.46 | RoyK | GiabboO: sorry. can't help anymore than |
| 15:16.53 | GiabboO | RoyK, ciao, do you use Asterisk + Mysql ? |
| 15:16.57 | RoyK | yes |
| 15:17.00 | GiabboO | Nice:) |
| 15:17.06 | GiabboO | I need help about that |
| 15:17.17 | gardo | Qwell: that means i need to tell my voip provider to have more codec support |
| 15:17.23 | RoyK | ~rules |
| 15:17.24 | jbot | somebody said rules was Ktron makes the rules! |
| 15:17.30 | stoffell | [TK]D-Fender: hiya.. uhm, with polycom's new sip firmware, is there any way to send text or a visualization to the phone via dialplan or shell/php scripting? |
| 15:17.30 | Qwell | gardo: correct |
| 15:17.39 | GiabboO | Royk, could you tell me wich files I need to configure mysql ? |
| 15:17.52 | RoyK | ~docs |
| 15:17.54 | jbot | from memory, docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 15:17.55 | RoyK | ~book |
| 15:17.56 | jbot | somebody said book was a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 15:17.56 | GiabboO | RoyK, my config dir is really messy |
| 15:17.57 | RoyK | ~rtfm |
| 15:17.58 | jbot | methinks rtfm is Read The F*cking Manual (TM). It is a suggestion to do your homework before posting a question. Sometimes used as RTFM $SPECIFIC_MANUAL to refer to a specific source of information. See also http://uncyclopedia.org/wiki/RTFM. |
| 15:18.15 | GiabboO | and actually my sip user cant register |
| 15:18.23 | sevard | damnit |
| 15:18.25 | sevard | what is that form of hierarchical note taking |
| 15:18.33 | GiabboO | I get this warning: config.c:920 find_engine: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available |
| 15:18.40 | [TK]D-Fender | stoffell: The only way would be to use a 6XX series MicroBrowser idle script on a moderately high refresh rate. But thats more Apache than anything else |
| 15:19.01 | RoyK | GiabboO: have you installed mysql-addons? |
| 15:19.05 | GiabboO | hmm |
| 15:19.06 | RoyK | might be worth a try |
| 15:19.07 | GiabboO | maybe not :p |
| 15:19.08 | GiabboO | :D |
| 15:19.10 | GiabboO | ty brb |
| 15:19.22 | *** join/#asterisk sevard (n=sev@c-67-188-173-23.hsd1.ca.comcast.net) |
| 15:19.30 | sevard | oops |
| 15:19.35 | sevard | does anyone know? |
| 15:19.46 | stoffell | [TK]D-Fender: hm, okay. still have to look at the 2.0.1, was hoping to find *something* for the non 6xx phones.. damn.. :) |
| 15:20.20 | [TK]D-Fender | stoffell: Nothing documented for implementing Messaging between * and Polycom yet. |
| 15:20.20 | stoffell | [TK]D-Fender: I can use the MWI, but.. then i can't indicate messages anymore :) |
| 15:20.33 | Juggie[bday] | GiabboO, you need asterisk-addons |
| 15:20.35 | corresponder | omg |
| 15:20.37 | [TK]D-Fender | stoffell: You could use a seperate reg for that... |
| 15:20.54 | *** join/#asterisk afrosheen (n=test@txprotoa2.august.net) |
| 15:21.01 | corresponder | RoyK, the last one was about IRQ using of the avm card |
| 15:21.04 | stoffell | [TK]D-Fender: ah, it's indicated differently also, then? |
| 15:21.12 | [TK]D-Fender | stoffell: What kind of information are you trying to get across? |
| 15:21.14 | GiabboO | Done |
| 15:21.19 | corresponder | is it easy to use the capi driver instead? |
| 15:21.22 | sevard | is it cornell? |
| 15:21.28 | [TK]D-Fender | stoffell: Each reg has its own MWI on the line-key |
| 15:21.33 | GiabboO | I remember with the old version, if i do sip show users, it wont show users stored in mysql, now is the same ? |
| 15:21.48 | stoffell | [TK]D-Fender: basically, anything is good. but I would like it to indicate if the phone is logged into a queue.. |
| 15:22.38 | [TK]D-Fender | stoffell: OH. challenging. There is a BRISTUFF patch that allows you to set a virtual device for the purpose of presence manipulation. But it conflicts with PRI things last I recall. |
| 15:22.44 | Juggie | grrr |
| 15:22.57 | Juggie | stupid freenode, drops my identify w/ services if i change nicks |
| 15:23.06 | [TK]D-Fender | stoffell: There is supposed to be a virtual device option built into * 1.4 which I read a while back. that would do it.... |
| 15:23.07 | Qwell | Juggie: link the nicks |
| 15:23.40 | mog | or use just one nick |
| 15:23.53 | stoffell | [TK]D-Fender: hmm, okay, making notes of that... |
| 15:24.05 | *** join/#asterisk viler (i=1000@200.114.70.228) |
| 15:24.12 | stoffell | [TK]D-Fender: basically, i don't mind *how* it's made visible, but a led or some text would do it.. |
| 15:24.16 | GiabboO | perfect |
| 15:24.20 | *** join/#asterisk mivck (i=1000@200.114.70.228) |
| 15:24.20 | GiabboO | my sip users can register now |
| 15:24.21 | GiabboO | :) |
| 15:24.29 | *** part/#asterisk mivck (i=1000@200.114.70.228) |
| 15:24.39 | [TK]D-Fender | stoffell: Like I said, 6XX is doable (I do that here already), just not any others. |
| 15:24.48 | GiabboO | I do I check if the iax users are loaded from mysql ? |
| 15:25.14 | stoffell | [TK]D-Fender: thanks, gotta do some searching on the virtual device thingy then.. :) thanks! |
| 15:25.23 | [TK]D-Fender | stoffell: You might be able to pull it off by doing file substitution for a 501 idle image, but thats a 50/50. Not sure if it only statically loads them. |
| 15:25.46 | [TK]D-Fender | stoffell: I am pretty sure it does, but thats as close as I can get... |
| 15:25.50 | stoffell | [TK]D-Fender: yeah, but that would cause some heavy traffic when using lots of phones, not? |
| 15:26.07 | [TK]D-Fender | stoffell: Possible which is why I think this entire idea ISN'T :) |
| 15:26.20 | stoffell | lol :) |
| 15:26.42 | stoffell | [TK]D-Fender: if I find *any* way, you'll be the first to know ;) |
| 15:27.01 | [TK]D-Fender | I have been rather successful this weekend as it is though : Got multiple forms of auto-answer working and idle images on the IP 501. |
| 15:27.21 | [TK]D-Fender | stoffell: So far you're on your way to doing ways I ALREADY know about, but you can remind me later ;) |
| 15:27.47 | stoffell | hm, that's nice also, the auto-answ.. |
| 15:28.02 | [TK]D-Fender | stoffell: Can't wait till I implement it here. Auto-spy! |
| 15:28.11 | stoffell | hehe :) if I make any progress, it's on the wiki and i'll probably inform u here :) |
| 15:28.29 | stoffell | [TK]D-Fender: hehe :) |
| 15:28.45 | [TK]D-Fender | *cough* I mean.... paging... yeah thats it! ;) |
| 15:30.00 | stoffell | :) |
| 15:30.19 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 15:30.57 | *** join/#asterisk chapeaurouge (n=chapeaur@80.92.83.35) |
| 15:31.12 | UnderMine | corresponder: I'm using an AVM C2 via capi it has issues but seams to work |
| 15:31.36 | *** join/#asterisk Ebola (n=Ebola@81-86-155-65.dsl.pipex.com) |
| 15:32.04 | afrosheen | I wish polycom would make paging easier on their phones with asterisk |
| 15:32.08 | GiabboO | RoyK, when Asterisk start it say "Binding iaxpeers to mysql/asterisk_db/iaxfriends", is this correct ? |
| 15:32.19 | UnderMine | most issues seem to be related to the old motherboard I'm using and getting freezes. |
| 15:33.01 | afrosheen | the last thing you want to do is mix bad hardware with asterisk |
| 15:33.10 | RoyK | GiabboO: possibly, maybe, perhaps, but i don't know and i can't talk right now |
| 15:33.12 | UnderMine | corresponder: system hangs (no kernel panic!) |
| 15:33.39 | GiabboO | kk |
| 15:34.25 | UnderMine | afrosheen: I agree hence new dedicated machine. Then hopefully the AVM C2 will work. |
| 15:34.41 | [TK]D-Fender | afrosheen: Its as easy as any other solution..... |
| 15:35.14 | *** join/#asterisk Defraz (n=t0tal@fw.centrisys.com) |
| 15:35.34 | afrosheen | [TK]D-Fender: we had problems with our testbed server with * since it was off-the-shelf...but put it on a 2850 and it was fine |
| 15:36.18 | mercestes | afrosheen: Whats hard about page(sip/peer)?? |
| 15:36.19 | [TK]D-Fender | afrosheen: Was just referring to your Polycom comment. As for hardware compatability, what cards are you referring to? |
| 15:36.23 | *** join/#asterisk my_name_is (n=j@nathan.epi.usf.edu) |
| 15:36.30 | afrosheen | [TK]D-Fender: ohhh, haha everyone is like 2 screens up |
| 15:36.31 | my_name_is | hi |
| 15:37.12 | my_name_is | is it possible to get data from astdb in the meetme.conf file? |
| 15:39.25 | Qwell | my_name_is: huh? |
| 15:39.44 | Qwell | d'oh |
| 15:40.42 | my_name_is | well in extensions.conf one can do a count=${DB(family/key)} does that work in meetme.conf also? |
| 15:40.46 | Qwell | no |
| 15:41.01 | Qwell | extensions.conf is "interpreted" on every call |
| 15:42.46 | *** join/#asterisk intralanman (n=lanman@pool-71-253-255-34.nrflva.east.verizon.net) |
| 15:44.09 | MrChimpy | hmm. I can't seem to get the length limit warnings working - defaults nor custom sample :( |
| 15:44.55 | *** join/#asterisk hfb (n=hfb@pool-71-108-114-33.lsanca.dsl-w.verizon.net) |
| 15:44.55 | GiabboO | IS there anyone that use app_dbodbc with asterisk current ? |
| 15:45.13 | Qwell | GiabboO: nah, I use func_odbc - it rocks |
| 15:45.19 | GiabboO | same commands ? |
| 15:45.22 | Qwell | no |
| 15:45.25 | GiabboO | ugh... |
| 15:45.34 | GiabboO | I have extension made long time ago with app_dbodbc |
| 15:45.35 | MrChimpy | Dial(${TRUNK}/${dialext},,L(${todiallimitms},${todiallwarnms},${todialrewarnms})) |
| 15:45.40 | Qwell | but it's far nicer than anything else I've seen |
| 15:45.43 | GiabboO | how can I do ? |
| 15:45.58 | MrChimpy | i've tried setting LIMIT_WARNING_FILE - doesn't work |
| 15:46.03 | GiabboO | do you think app_dbodbc will work with current release ? |
| 15:46.40 | Qwell | GiabboO: I don't know, I've never used it |
| 15:47.39 | ManxPower | my_name_is, no. meetme.conf only holds static settings |
| 15:48.10 | ManxPower | MrChimpy, what version of Asterisk? |
| 15:48.25 | ManxPower | MrChimpy, in one or 2 releases, the limits stuff was broken. I don't remember the versions |
| 15:49.01 | GiabboO | ill try now to compile app_dbodbc :p |
| 15:49.36 | ManxPower | ODBC stands for Oh Damn Binary Connection! |
| 15:49.38 | *** join/#asterisk Strom_C (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 15:49.46 | MrChimpy | 1.2.9.1 |
| 15:50.01 | Qwell | MrChimpy: no complaining about bugs, until you're on the latest version :p |
| 15:50.04 | ManxPower | MrChimpy, I'm pretty sure it fixed long before that. |
| 15:50.04 | Strom_C | holy god, is it monday morning already? :) |
| 15:50.12 | Qwell | Strom_C: It's Tuesday... |
| 15:50.13 | GiabboO | ahah ManxPower :) |
| 15:50.25 | Strom_C | oh noes |
| 15:50.35 | Qwell | mitcheloc: ping |
| 15:50.47 | Qwell | Strom_C: two weeks...we've gotta do something soon |
| 15:50.53 | Strom_C | yep |
| 15:50.56 | Strom_C | just tell me when |
| 15:51.22 | MrChimpy | and it happily plays a sample at call start using the limit mechanism |
| 15:53.40 | *** join/#asterisk Splas (n=jwb@206.252.198.101) |
| 15:54.35 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.72.99) |
| 16:01.38 | *** join/#asterisk P4C0 (n=ash@200.124.22.34) |
| 16:02.07 | P4C0 | hello guys, I have a little nat problem |
| 16:03.01 | afrosheen | oh crap, tell me he didn't say nat problem |
| 16:03.18 | GiabboO | :\ |
| 16:03.20 | *** join/#asterisk Corydon76-home (i=five@pdpc/supporter/sustaining/Corydon76-home) |
| 16:03.21 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 16:03.29 | GiabboO | broke pipe when i run asterisk with app_dbodbc |
| 16:03.49 | *** join/#asterisk Ebola (n=Ebola@user-54463be2.lns2-c12.dsl.pol.co.uk) |
| 16:04.11 | P4C0 | yup, but this one is a funny one... in my invite sip message my local ip address is nowhere... even so the reply of my voip provider comes with my local ip |
| 16:06.22 | *** join/#asterisk anthonyl (n=anthony@c-67-167-214-149.hsd1.il.comcast.net) |
| 16:10.13 | asjdhsadjh2 | P4C0, its ok to reference the private IP, but is it working? |
| 16:10.22 | *** join/#asterisk doolph (n=doolph@200.46.148.58) |
| 16:10.49 | asjdhsadjh2 | P4C0, u should try nat=route and qualify=yes in the user definition |
| 16:11.30 | P4C0 | asjdhsadjh2, no it's not... how can that be ok?? if I send for example an invite with my public ip all over it, and the server reply with 407 but with my local ip, then the call-id doesn't match ant the 407 is discarted by my asterisk |
| 16:12.07 | P4C0 | asjdhsadjh2, nat=route? instead of yes? |
| 16:12.19 | asjdhsadjh2 | yes and route should both work for u |
| 16:12.28 | doolph | hello anyone know why I get ***[JB LOG]*** fixed jitterbuffer created on channel Zap/1-1 |
| 16:12.37 | doolph | If I transfer this call it will hang up |
| 16:13.00 | P4C0 | asjdhsadjh2, I'll try it, thanks |
| 16:13.10 | Dandre | what is the difference between Loop, Ground an kewl signalisation? |
| 16:13.44 | GiabboO | I get this error loader.c:325 __load_resource: /usr/lib/asterisk/modules/app_dbodbc.so: undefined symbol: ast_direct_realtime |
| 16:13.49 | GiabboO | anyone know the solution ? |
| 16:14.20 | MrChimpy | whee. i got the call limiting stuff working |
| 16:14.40 | P4C0 | asjdhsadjh2, what does qualify really does? |
| 16:14.40 | MrChimpy | i can now forward a call and then have it moo EVERY FIVE SECONDS to the caller :) |
| 16:14.43 | doolph | <PROTECTED> |
| 16:14.44 | doolph | <PROTECTED> |
| 16:14.48 | doolph | help |
| 16:15.13 | asjdhsadjh2 | P4C0, qualify sends a NOTIFY to the client every so often to keep the NAT tunnel alive |
| 16:15.31 | P4C0 | asjdhsadjh2, ok, this is what happends, moment, I'll paste bin |
| 16:15.37 | *** join/#asterisk Mw3 (i=mw3@national.t-error.hu) |
| 16:18.13 | P4C0 | asjdhsadjh2, please take a look: http://rafb.net/paste/results/vwvkLx29.html |
| 16:22.54 | P4C0 | does someone knows why this happends: http://rafb.net/paste/results/vwvkLx29.html how is possible for my voip provider to fetch my local ip?? I'm not sending it... |
| 16:23.18 | De_Mon | did you say moo? |
| 16:26.29 | P4C0 | :( |
| 16:29.16 | GiabboO | anyone use app_dbodbc with Asterisk 1.2 ? o.O |
| 16:33.08 | *** join/#asterisk fall0ut (n=tim@c-69-180-250-113.hsd1.tn.comcast.net) |
| 16:33.10 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 16:34.48 | De_Mon | P4C0 the asterisk box is behind NAT? |
| 16:34.56 | P4C0 | De_Mon, yup |
| 16:35.15 | P4C0 | and my natter is a netgear, doing symetric nat |
| 16:35.52 | mut | http://tinyurl.com/ryek9 |
| 16:37.22 | P4C0 | De_Mon, I just email netgear support to see if maybe they can give me any tips... maybe the netgear is adding my private ip somewhere |
| 16:38.30 | [TK]D-Fender | P4C0: Pastebin the [general] sectiopn of your sip.conf |
| 16:38.46 | P4C0 | [TK]D-Fender, moment |
| 16:39.08 | *** join/#asterisk blaylock (n=seth@snap.helixsystems.com) |
| 16:39.22 | blaylock | hello all |
| 16:39.35 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 16:40.23 | blaylock | anyone have some good resources for line apperances in asterisk? |
| 16:40.32 | blaylock | google isnt turning up much so far |
| 16:40.45 | blaylock | starting to think there isnt much support for it |
| 16:41.05 | [TK]D-Fender | blaylock: Line appearances? Want to clarify exactly what you mean by that? |
| 16:41.14 | P4C0 | [TK]D-Fender, here it is: http://rafb.net/paste/results/YCjLir80.html |
| 16:42.10 | [TK]D-Fender | P4C0: Ok, no reason you should ever see your internal IP reflected back to you.... only thing I can think of is your router may be doing some for of SIP transform in-line. Check its settings |
| 16:42.17 | *** join/#asterisk ppyy (n=lala@222.185.17.94) |
| 16:43.01 | P4C0 | [TK]D-Fender, it have no seetings for that... it's a really cheap one... netgear fvs338, even so I just email them |
| 16:44.09 | fall0ut | http://kbserver.netgear.com/release_notes/D102929.asp |
| 16:44.13 | fall0ut | its got a SIP ALG in it |
| 16:45.33 | P4C0 | fall0ut, humm but there's no option to modify it.. |
| 16:45.54 | fall0ut | welcome to crappy firewalls |
| 16:46.05 | afrosheen | crappy firewalls 101 |
| 16:46.09 | fall0ut | Linksys/Netgear/D-Link have been throwing that in there |
| 16:46.15 | fall0ut | randomly in some products for the past year or two |
| 16:46.19 | fall0ut | with no options to turn it on/off/modify it |
| 16:46.30 | fall0ut | half of the time it's broken anyway |
| 16:46.46 | P4C0 | fall0ut, god... |
| 16:46.59 | fall0ut | PixOS/IOS lets you atleast turn it off |
| 16:47.16 | blaylock | line appearances = blinking lights to know when a line is in use |
| 16:47.21 | blaylock | like the old key systems use |
| 16:47.47 | [TK]D-Fender | blaylock: You can already do that with Presence. |
| 16:47.49 | P4C0 | what can I do? better yet, how its supposed to work? |
| 16:47.55 | blaylock | sort of what the buddy list does except visible on the phone itself |
| 16:48.09 | [TK]D-Fender | PIX does HORRIBLE things to NAT period..... |
| 16:48.12 | fall0ut | blaylock: I dunno if asterisk supports it |
| 16:48.17 | fall0ut | it'd be a SIP-B feature |
| 16:48.24 | fall0ut | or MGCP |
| 16:48.26 | [TK]D-Fender | blaylock: You can have it visible very easily. |
| 16:48.27 | blaylock | [TK]D-Fender, yeah but isnt Presence associated witht he buddy list? |
| 16:49.03 | blaylock | i mean i suppose it would depend on the type of phone and how many buttons are available |
| 16:49.09 | [TK]D-Fender | blaylock: Yes it is. You just need to have your buddies fall onto line keys, and you're good to go. Why do you think the IP 601/650 has an Attendant Module? |
| 16:49.29 | [TK]D-Fender | blaylock: Yes that statement is pretty self confirming... |
| 16:49.59 | fall0ut | so asterisk supports that now? |
| 16:50.06 | [TK]D-Fender | blaylock: http://www.polycom.com/products_services/0,1443,pw-34-182-12072,00.html |
| 16:50.16 | fall0ut | have they hacked the ACD login/out info from the polycoms -> sylantro/broadsoft yet for it? |
| 16:50.18 | blaylock | hmm |
| 16:50.30 | [TK]D-Fender | blaylock: There's a boatlod of indication, right there... |
| 16:50.41 | blaylock | cool thanks, Ill read up |
| 16:51.41 | [TK]D-Fender | fall0ut: bweschke already has an * branch for that. Its spotty, but somewhat functional. No idea what it'll take to go "mainstream" though.... |
| 16:51.58 | doolph | i hate this x100p thinggggggggg |
| 16:52.02 | fall0ut | thats cool |
| 16:52.10 | fall0ut | I haven't looked at this stuff in asterisk in > 2 yrs |
| 16:52.13 | backblue | Gigaset 2000T -> anyone know something about this, working with asterisk? |
| 16:52.40 | doolph | I am going to install asterisk and zaptel manually |
| 16:52.41 | fall0ut | I had been doing a lot of work w/ Sylantro for the past few |
| 16:52.48 | doolph | fuk rpms |
| 16:53.34 | blaylock | heh |
| 16:53.46 | blaylock | doolph, always good to compile and install asterisk anyway |
| 16:54.06 | blaylock | doolph, havent really heard many good things about asterisk rpms |
| 16:54.16 | doolph | what is jitterbuffer, it's hanging up my zap channels when I try to transfer a call |
| 16:54.23 | fall0ut | asterisk have any nice AA/Voicemail web front-ends yet? for just using it as VM/AA server... |
| 16:54.33 | Qwell | fall0ut: AA? |
| 16:54.38 | fall0ut | auto-attendant |
| 16:54.45 | *** join/#asterisk rnovotny22 (n=Bob@198.57.19.126) |
| 16:54.46 | doolph | freepbx ? |
| 16:55.23 | [TK]D-Fender | OUCHHH!!!!!!! IP 650 pricing!!!!! http://voipstore.atacomm.com/Shops/ViewItem.aspx/27934028032-54099021056.htm |
| 16:56.20 | doolph | too expensive? |
| 16:56.46 | mercestes | What's the diff between the 601 and the 650 besides those superfly racing stripes? |
| 16:58.23 | [TK]D-Fender | doolph: IP 601 +/- 240-250$. 650 = $315. Thats a big jump for the difference. Its too much relative to the 601 in my books. |
| 16:58.24 | afrosheen | ouch? LOL that's a bargain |
| 16:58.27 | fall0ut | probably not a whole lot yet |
| 16:58.43 | [TK]D-Fender | afrosheen: Perhaps in some sense, just a questio of relative value. |
| 16:58.44 | fall0ut | I bet in the future you start seeing more features going into 650 |
| 16:59.08 | [TK]D-Fender | 650 adds backlight, USB expansion, G.722, and a few other things. |
| 16:59.13 | afrosheen | yeah |
| 16:59.22 | afrosheen | usb = usb headsets, that'll be nice |
| 16:59.32 | Qwell | [TK]D-Fender: You pretty much called the 4xx.. I bet the 650 will go down a bit too |
| 16:59.41 | doolph | usb headset cost around $15 |
| 16:59.48 | [TK]D-Fender | Qwell: Called? |
| 16:59.55 | Qwell | the price, you were pretty much dead on |
| 17:00.04 | [TK]D-Fender | Qwell: In terms of my preempive price guesstimate? |
| 17:00.07 | Qwell | yes |
| 17:00.30 | [TK]D-Fender | Qwell: Yeah, well I never ventured a guess for this... I might have though $280 tops for best range |
| 17:00.31 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 17:00.45 | fall0ut | pricing on that will go down |
| 17:01.06 | Qwell | as much as they were downplaying the 601 at VON, I'm sure they'll drop the price of the whole line, to make the 650 match, heh |
| 17:01.23 | [TK]D-Fender | Qwell: the IP 430 was a far more reasonable item to guess. they added just enough to go wild on it. Though when you look at the real featureset (and imagin what add-ons are to come) its probably STILL a decent deal. |
| 17:01.40 | [TK]D-Fender | Qwell: You can be sure I'll be watching it like a hawk :) |
| 17:01.51 | afrosheen | it'd be nice to get color screens on them but it's just bling |
| 17:02.03 | Qwell | color touchscreen :D |
| 17:02.09 | afrosheen | hooyah |
| 17:02.14 | [TK]D-Fender | Qwell: And this weekend I worked out IP501 background images & auto-answer formats. Not much I haven't done on them now. |
| 17:02.33 | afrosheen | [TK]D-Fender: we did animations on ours a few weeks ago, turned out pretty cool |
| 17:02.34 | [TK]D-Fender | Qwell: Yeah, the Cisco sure as hell looks nice, but I'm STILL not spending that kind of cash! |
| 17:02.46 | [TK]D-Fender | afrosheen: Idle animations? |
| 17:02.47 | Qwell | [TK]D-Fender: it isn't a whole hell of a lot more than the 650 :D |
| 17:02.52 | afrosheen | [TK]D-Fender: yeah |
| 17:03.12 | fall0ut | anybody using 7941/61s yet? |
| 17:03.20 | [TK]D-Fender | afrosheen: Whats the transition like? I never read docs on it, but the frame delay seems somewhat obvious in nature... |
| 17:03.20 | afrosheen | Qwell: well the thing with the 6xx is, it's not for everyone..it's kind of an exec phone |
| 17:03.44 | afrosheen | [TK]D-Fender: not sure what you're referencing, I followed a howto on the wiki |
| 17:03.51 | [TK]D-Fender | afrosheen: I have my whole company on IP 600's :) Was a question of PoE at the time vs 501 + added cable). |
| 17:04.00 | [TK]D-Fender | afrosheen: For animating it? |
| 17:04.05 | afrosheen | [TK]D-Fender: yeah |
| 17:04.19 | [TK]D-Fender | afrosheen: how smooth is the transition between frames? Any effects? |
| 17:04.21 | afrosheen | our front desk phone has a 3d logo spinning on it :) |
| 17:04.33 | afrosheen | it's as smooth as number of frames / second\ |
| 17:04.43 | afrosheen | divided by the lcd refresh rate :) |
| 17:04.50 | doolph | I cannot reboot as usual |
| 17:04.52 | doolph | this sucks |
| 17:05.49 | ido | ugh...when will my wrt54g arrive damnit...i want to set up asterisk on it! :p |
| 17:06.11 | [TK]D-Fender | afrosheen: Which is? |
| 17:06.43 | doolph | anyone know how to shape upload bandwidth? |
| 17:08.44 | *** join/#asterisk teknoprep (n=tekon@unaffiliated/teknoprep) |
| 17:09.11 | *** join/#asterisk GingerDog (n=GingerDo@oak.palepurple.co.uk) |
| 17:09.34 | *** join/#asterisk robl^ (n=robl@dsl093-025-218.hou1.dsl.speakeasy.net) |
| 17:09.42 | GingerDog | hi; i've got an sql db with some phone numbers / names in etc, what's a good way of using that info to populate caller id with e.g. a name for incoming phone calls? |
| 17:10.59 | asjdhsadjh2 | use the mysql DB extension |
| 17:11.04 | [TK]D-Fender | GingerDog: Make an AGI to pull it up and set some channel variables |
| 17:11.22 | tzanger | woot woot woot |
| 17:11.41 | robl^ | GingerDog: are you using MySQL? |
| 17:11.58 | GingerDog | robl^: no; psql on a remote box (from asterisk) |
| 17:12.44 | robl^ | ahhh! I wrote an app that does cidnamelookup using MySQL. I never did port it to PostgreSQL. I should tough |
| 17:13.09 | GingerDog | robl^: src = ? |
| 17:14.01 | robl^ | GingerDog: http://no-no-badpuppy.com/?p=8 |
| 17:14.43 | Corydon-w | robl^: you can do that same thing with no rewriting with func_odbc |
| 17:14.59 | robl^ | it needs a little work for current versions.. but its not bad.. its just a re-work of the app_cidnamelookup.c |
| 17:15.39 | robl^ | Corydon-w: true.. but this was written before those APIs were stable. ;-) and it was a learning excervise... my first app in c. ;-) |
| 17:15.48 | Corydon-w | Heh |
| 17:16.11 | Corydon-w | Now that API is beyond stable... ;-) |
| 17:16.40 | teknoprep | how do i make extension 1003 mailbox the same as 1001 ? |
| 17:16.48 | teknoprep | can i use a symlink ? |
| 17:16.56 | robl^ | yeah! I need an Asterisk refresher course.. on all the new features / capabilities |
| 17:17.20 | festr__ | hello, anyone could help with debug res_features (ast 1.2 latest svn)? it is broken when featuremap activated (infinite loop with endless gettimeofday consuming all sys cpu) |
| 17:17.48 | [TK]D-Fender | WOW, SUPER CHEAP VIDEOPHONE... http://voipstore.atacomm.com/Shops/ViewItem.aspx/27934028032-53081680640.htm |
| 17:18.11 | [TK]D-Fender | teknoprep: And why would you do that? |
| 17:18.23 | Qwell | haha |
| 17:18.24 | Qwell | You Save $0.00 ( 0.0% ) |
| 17:18.26 | teknoprep | [TK]D-Fender, becuase its what i want to do |
| 17:18.53 | teknoprep | [TK]D-Fender, i have 3 phones .. 1 at home.. 1 on my computers.. 1 at work.. i want the voicemail to be all the same |
| 17:19.02 | robl^ | teknoprep: he's asking *why* because there may be other ways to solve the problem. |
| 17:19.23 | teknoprep | [TK]D-Fender, they are all in a ring group... ext 201 is actually a ring group that rings my 3 extensions |
| 17:19.24 | robl^ | give all 3 phones the same mailbox |
| 17:19.32 | teknoprep | thats what i want to do |
| 17:19.35 | teknoprep | lol |
| 17:20.12 | *** join/#asterisk X-Rob_ (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 17:20.16 | robl^ | teknoprep: create a mailbox 201. then use voicemail(201) for all 3 phones |
| 17:20.18 | [TK]D-Fender | teknoprep: So just only call 1 box. Where's the need for virtualizing boxes? |
| 17:20.48 | [TK]D-Fender | (here it comes....) |
| 17:20.56 | teknoprep | i want my phone to blink also when i have a voicemail |
| 17:21.15 | [TK]D-Fender | teknoprep: Also very easy. I use 1 box for 3 polycom phones at home. No issue. |
| 17:21.25 | [TK]D-Fender | teknoprep: So why do you think this will be hard? |
| 17:21.38 | teknoprep | [TK]D-Fender, i didn't say it would be hard.. i asked how to do it |
| 17:21.44 | robl^ | teknoprep: yes? you can have the MWI blink on multiple phones using a single mailbox |
| 17:22.04 | [TK]D-Fender | teknoprep: Well its your dialplan.. just only call that single box, and have "mailbox=(whatever) the same for all devices. |
| 17:22.04 | teknoprep | [TK]D-Fender, why all the smart ass comments? i asked a simple question |
| 17:22.13 | teknoprep | ahh |
| 17:22.13 | teknoprep | ty |
| 17:22.30 | teknoprep | i use trixbox.. but freepbx can't do this.. so i need to manually edit this crap |
| 17:22.39 | robl^ | teknoprep: are you using asterisk? or asterisk @ home / freepbx? |
| 17:22.44 | [TK]D-Fender | Druken: I caught one!!!! |
| 17:23.02 | [TK]D-Fender | teknoprep: See thats what I was 99.9999% sure of... |
| 17:23.14 | robl^ | teknoprep: ahh! yeah, using trixbox/freepbx makes it more difficuly |
| 17:23.19 | *** join/#asterisk Cyt (n=danielcy@athedsl-141532.otenet.gr) |
| 17:23.21 | [TK]D-Fender | teknoprep: You really should ditch that canned POS... |
| 17:23.31 | teknoprep | i don't have time right now to learn all of asterisk |
| 17:23.47 | teknoprep | plus for what i do.. it works like a charm and is quick to setup |
| 17:24.06 | robl^ | 10% of asterisk is 99% of what most people need |
| 17:24.12 | |Vulture| | anyone have a recommendation for a portable phone for a house? We use IP501s at the offices but they are a little much for a house |
| 17:24.20 | Qwell | |Vulture|: 301? |
| 17:24.41 | *** join/#asterisk Tr4d3r (n=Tr4der@200.71.51.147) |
| 17:24.51 | [TK]D-Fender | |Vulture|: Clarify "portable" |
| 17:25.36 | robl^ | |Vulture|: look at the Aastra 480i CT. It has a nice wireless handset in addition to a full feature desk phone |
| 17:25.45 | |Vulture| | well wireless would be nice.. but I havent heard anything good about the wireless phones right now |
| 17:25.49 | [TK]D-Fender | and 5% of the junk an * GUI throws into the configs is what its users really need. |
| 17:26.13 | Qwell | the other 95% makes it impossible to edit |
| 17:26.19 | [TK]D-Fender | |Vulture|: SIP WiFI phones SUCK. All of them. Get an ATA (SPA-2002) and a decent cordless analog phone for now. |
| 17:26.29 | Qwell | [TK]D-Fender: Have you tried the wip300? |
| 17:26.44 | [TK]D-Fender | Qwell: Maximum obfuscation acheived! |
| 17:26.47 | |Vulture| | [TK]D-Fender: thats what I was thinking I would have to do |
| 17:27.01 | [TK]D-Fender | Qwell: Actually..... I just ordered one :) |
| 17:27.06 | Qwell | [TK]D-Fender: order me one :p |
| 17:27.14 | [TK]D-Fender | |Vulture|: a lot cheaper too.... |
| 17:27.26 | Qwell | Cory said he'd give me the $120 price, but I can't afford it right now - damn this whole buying a house thing |
| 17:27.30 | [TK]D-Fender | Qwell: Sure... jsut gimme your CC info ;) |
| 17:27.32 | *** part/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 17:27.41 | Qwell | [TK]D-Fender: Will you take a radio shack card? :P |
| 17:27.47 | robl^ | The Aastra 480i is not WiFi.. but it is wireless.. its a wired ethernet connection for the base/desk phone.. the handset uses 5.8 ghz range.. just like a normal analog cordless phone |
| 17:28.11 | denon | [TK]D-Fender: you know, I really wish that wasnt true .. I want to replace a bunch of cordless phones at the offices |
| 17:28.27 | denon | but waiting for wireless sip phones get get reliable |
| 17:28.28 | Qwell | denon: Get a bunch of wip300s :D |
| 17:28.33 | [TK]D-Fender | denon: ATA's really aren't bad..... |
| 17:28.48 | denon | [TK]D-Fender: it's not that ..I run them all into a channel bank anyway |
| 17:28.51 | [TK]D-Fender | Qwell: I should take a pic of my IP 501 w/ * logo on it for you :) |
| 17:28.54 | denon | I want sip features .. hold/xfer/etc |
| 17:28.56 | Tr4d3r | #asterisk-la > Español |
| 17:29.02 | denon | the # crap blows |
| 17:29.07 | denon | or flash even |
| 17:29.07 | teknoprep | oh you can do what i wanted with freepbx |
| 17:29.13 | Qwell | <insert obligatory cisco 7920 joke> |
| 17:29.18 | [TK]D-Fender | denon: then your CB is the issue. SPA ATA's give you everything you could expect from a phone supporting 2 calls. |
| 17:29.24 | teknoprep | i never saw that config for mailbox on the list... just needed to put it at 1001@device |
| 17:29.28 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 17:29.51 | [TK]D-Fender | denon: Flash I can live with. Entirely. "#" transfers and mangling my dialplan is UNFORGIVABLE. |
| 17:30.25 | [TK]D-Fender | teknoprep: * does whatever you tell it to. jsut becareful that Trixbox doesn't wipee your changes with its next "commit" |
| 17:30.40 | teknoprep | [TK]D-Fender, yeah i saw that |
| 17:30.48 | teknoprep | [TK]D-Fender, but i just saw in freepbx you can do what i want also |
| 17:30.53 | teknoprep | [TK]D-Fender, so i just did it in freepbx |
| 17:31.06 | teknoprep | [TK]D-Fender, ty for the help |
| 17:31.20 | [TK]D-Fender | teknoprep: Congrats, np |
| 17:31.38 | teknoprep | now i have to work on setting up a feature code to do this... |
| 17:31.49 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 17:31.50 | teknoprep | exten => *888,1,DBput(ivr/mode=bizhours) and exten => *999,1,DBput(ivr/mode=afterhours) |
| 17:31.50 | teknoprep | GotoIf($[${ivrmode} = "bizhours"]?daytime-ivr|s|1:nighttime-ivr|s|1) |
| 17:33.05 | *** join/#asterisk m_o_d (n=kane@host.ltv.pl) |
| 17:33.22 | [TK]D-Fender | teknoprep: Well protecting yourself from the config-wipe may be the biggest part of that challenge. Also making sure that any IVR auto-renumbering doesn't break any custom features you are planning to make. |
| 17:33.44 | teknoprep | [TK]D-Fender, most trixbox/freepbx configs are include files |
| 17:34.00 | teknoprep | [TK]D-Fender, this is to help not wipe your configs |
| 17:34.23 | [TK]D-Fender | teknoprep: But can still break if the IVR numbering changes. (last I recall they were numbered and may get resequenced on a rebuild) |
| 17:34.35 | robl^ | teknoprep: correct.. but extra care must be taken to make sure you make your changes in the correct files. |
| 17:34.44 | *** join/#asterisk Ebola (n=Ebola@81-86-155-65.dsl.pipex.com) |
| 17:34.50 | teknoprep | agreed |
| 17:35.05 | [TK]D-Fender | teknoprep: Tahts with a few small assumptions in place, but I'll leave it at that. For what you want there isn't that much to learn.... |
| 17:36.00 | *** join/#asterisk anthonyl (n=anthony@c-67-167-214-149.hsd1.il.comcast.net) |
| 17:37.33 | *** join/#asterisk RoyK (n=roy@a217-118-45-74.bluecom.no) |
| 17:38.05 | teknoprep | i don't really care much about embedded systems |
| 17:38.06 | *** join/#asterisk rgsteele (n=chatzill@nat-pool.agora-net.com) |
| 17:38.14 | *** part/#asterisk rgsteele (n=chatzill@nat-pool.agora-net.com) |
| 17:39.32 | *** join/#asterisk _VoicePulse (n=contact@unaffiliated/voicepulse) |
| 17:40.24 | [TK]D-Fender | Holy Shit!@ Massive price drop on Citel Digital gateways! 1K for 24 ports! |
| 17:40.57 | *** join/#asterisk Elysius (n=Elysius@pool-71-108-163-36.lsanca.dsl-w.verizon.net) |
| 17:41.05 | [TK]D-Fender | http://voipstore.atacomm.com/Shops/ViewItem.aspx/27934028032-47071662592.htm |
| 17:42.19 | *** join/#asterisk Nand0 (n=Nando@unaffiliated/nand0) |
| 17:42.34 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 17:43.51 | *** join/#asterisk Trazz (i=Trazz@c-71-57-36-106.hsd1.il.comcast.net) |
| 17:43.58 | Elysius | Hello, everyone. I have a Digium Wildcard TDM400P with one FXO module and one FXS module. I want to make sure that I have the channels assigned correctly so that I know that the FXO card is able to receive calls. I have the fxsks=1 and fxofs=2 for the FXO and FXS modules respectively in '/etc/zaptel.conf'. I have also "modprobe zaptel ; modprobe wcfxo ; modprobe wcfxs ; ztcfg". Is there anything else I can check? I'm running |
| 17:44.27 | *** join/#asterisk dashu (n=dashu@p549C7EA6.dip.t-dialin.net) |
| 17:44.59 | RoyK | Elysius: a good start is calling digium |
| 17:45.08 | dashu | hi is there anyone that could help me with some asterisk isdn problems |
| 17:45.28 | Elysius | I wasn't able to find their telephone number on the www.digium.com web site, RoyK. I'll take another look for it. |
| 17:45.29 | RoyK | dasenjo: isdn what? misdn? capi? bristuff? zaptel? isdn4linux? |
| 17:45.39 | dashu | :o mmh i think capi |
| 17:45.47 | luke-jr_work | Elysius: or I could call you... |
| 17:46.00 | Qwell | Elysius: 877-linux-me |
| 17:46.18 | [TK]D-Fender | Elysius: You need to do "modprobe wctdm". wcfxo is for the X100, and ther is no wcfxs |
| 17:46.30 | dashu | i want that my asterisk servers hangs up when a specific guy is calling me :p |
| 17:46.37 | dashu | but he doesnt hangup :o |
| 17:46.47 | Elysius | I think I did a "modprobe wctdm", [TK]D-Fender. |
| 17:47.09 | [TK]D-Fender | dashu: Just check the caller ID and dump him off. |
| 17:48.01 | Elysius | The number worked, Qwell. Much appreciated. ':) |
| 17:48.33 | RoyK | Elysius: did you get an answer????? |
| 17:48.59 | RoyK | "well, that's a feature" |
| 17:49.27 | [TK]D-Fender | RoyK: And henceforth fatal bugs shall now be referred to as "SPECIAL features" |
| 17:49.51 | Elysius | I did get an answer, RoyK. |
| 17:50.41 | *** join/#asterisk Magicianx (n=magician@116-22.dr.cgocable.ca) |
| 17:51.04 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 17:51.51 | *** join/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 17:51.54 | dashu | can anyone help me with asterisk and isdn capi stuff ? :p im a student and mmh i have choosen the subject linux and makeing and asterisk isdn server is one part of it but i cant solve some problems that i got with it ^^ |
| 17:52.39 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 17:53.51 | afrosheen | any reason why chan_zap is missing from my install? |
| 17:54.51 | Juggie | because you didnt install zaptel before you installed asterisk |
| 17:56.09 | *** join/#asterisk stubert (i=stu@techtools.actusa.net) |
| 17:57.03 | afrosheen | ohhh..maybe..I have zaptel installed and working but yeah, all I have for chan_zap is chan_zap.c in the source folder |
| 17:57.22 | afrosheen | so asterisk has to see zap installed before it'll build that module? |
| 17:58.09 | Elysius | On the Digium Wildcard TDM400P I have the FXO (Red?) module pluged into the thing marked "J4" and "TEL1" and the FXS (Green?) module plugged into "J1" on "TEL2". Is 'fxsks = 1' and 'fxoks = 2' correct signalling wise for that hardware setup on the card? |
| 17:58.11 | afrosheen | how can I rebuild asterisk to *only* do the chan_zap.so build |
| 17:58.17 | luke-jr_work | I presume it needs zaptel headers to compile the module |
| 17:58.31 | luke-jr_work | afrosheen: don't 'make clean' ? |
| 17:58.32 | eKo1 | afrosheen: make install |
| 17:58.41 | afrosheen | lol ok I'll do a standard make |
| 17:58.54 | luke-jr_work | you'll probably need to reconfigure |
| 17:58.59 | eKo1 | make won't recompile things that don't need recompiling |
| 17:59.10 | luke-jr_work | eKo1: it never was compiled... |
| 18:00.06 | afrosheen | I've built asterisk 100 times in the past, can't believe I forgot order of operations like this :p |
| 18:00.34 | stubert | Elysius: http://www.digium.com/en/docs/TDM400P/tdm_inst.pdf |
| 18:00.52 | afrosheen | cool, running 'make' built the modules..and I don't want to make install, should I just copy the modules into /usr/lib/asterisk/modules or is there more to it |
| 18:01.05 | Elysius | Ah, RTFM. :p Okidokey. I'll go read it and doublecheck. |
| 18:01.18 | stubert | Elysius: one = chan one... |
| 18:01.23 | stubert | hope it helps |
| 18:02.09 | mistral | anyone know why a fritz card wont get the L2Link to come up ? |
| 18:02.10 | mistral | * Stack Addr:40000104 Port 1 Type TE Prot. PTP L2Link DOWN L1Link:UP Debug:2 |
| 18:02.17 | *** join/#asterisk razu_ (n=razu@87-119-182-130.tll.elisa.ee) |
| 18:02.30 | Elysius | It's just perplexing, stubert, because either I didn't config the 'zapata.conf' or 'extensions.conf' right.. since I think the card should be answering. |
| 18:05.49 | *** join/#asterisk doolph (n=doolph@201.218.125.92) |
| 18:06.17 | Elysius | Ok. Here's a sixty-four thousand dollar quetion. Will the FXO and FXS modules answer without a 3.5" Mylex power cable plugged into 'em? %-) |
| 18:06.52 | doolph | anyone know why after rebuild Zaptel Version 1.2.9.1 X100P stop working |
| 18:07.01 | *** join/#asterisk rokerr (n=rowan@38.99.162.187) |
| 18:07.14 | Elysius | Can't say I do, doolph |
| 18:07.26 | doolph | you do? |
| 18:07.40 | doolph | why |
| 18:08.16 | doolph | what's the trick |
| 18:08.27 | Elysius | I do not know is what I meant, doolph, why the 1.2.9.1 Zaptel build might have stopped working with the X100P |
| 18:08.43 | afrosheen | Elysius: one will, the other will not |
| 18:09.06 | afrosheen | Elysius: whichever module needs voltage to generate rings on analog phones will not work without it |
| 18:09.20 | stubert | Elysius: you need power for fxs cards, not fxo cards |
| 18:09.21 | doolph | ok I mean the x100p is not detected after install zaptel 1.2.9.1 |
| 18:09.23 | afrosheen | yep |
| 18:09.36 | *** join/#asterisk RoyK (n=roy@a217-118-45-74.bluecom.no) |
| 18:14.19 | *** join/#asterisk Adam06 (n=Adam@ip67-95-13-58.z13-95-67.customer.algx.net) |
| 18:14.29 | Adam06 | Hey, anyone here know Polycom sip.cfg configuation? |
| 18:14.38 | *** join/#asterisk _alex_mx_ (n=alex@200.78.229.18) |
| 18:15.04 | doolph | how do I load wcfxo? |
| 18:15.30 | [TK]D-Fender | Adam06: plenty of us, just ask |
| 18:15.31 | *** join/#asterisk adorah (n=admin@84.94.144.7.cable.012.net.il) |
| 18:16.00 | Adam06 | I'm having trouble getting Intercom working |
| 18:16.28 | Adam06 | I use Polycom 301's, |
| 18:17.03 | [TK]D-Fender | Adam06: Ok, clarify what you mean by "intercom" |
| 18:17.23 | Elysius | doolph 'modprobe wcfxo' on most Linux sboxen after a 'modprobe zaptel' if the 'make clean' and 'make install' from source worked. |
| 18:17.42 | niter3 | the thing that allows one person to talk on all phones. For instance : Bob please report to the office, Bob. |
| 18:17.47 | niter3 | [TK]D-Fender: |
| 18:17.56 | festr__ | anyone here using features.conf and applicationmap in asterisk 1.2? i've latest svn and think it is broken |
| 18:18.26 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 18:18.28 | [TK]D-Fender | niter3: I want to hear his intended meaning... he may not be thinking the same thing and I want to here the need clearly described. |
| 18:18.30 | *** join/#asterisk jbalcomb (n=jbalcomb@216.28.180.158) |
| 18:19.03 | Adam06 | The ability to hit 98105 and have the phone on ext 8105 auto pickup so I can just start talking |
| 18:19.04 | Adam06 | ring for maybe a second or something |
| 18:19.15 | [TK]D-Fender | Adam06: So a single targeted auto-answer then? |
| 18:19.18 | [TK]D-Fender | Adam06: http://www.voip-info.org/wiki/view/Polycom+auto-answer+config |
| 18:19.21 | Adam06 | http://rafb.net/paste/results/8PZHOy68.html < stuff in extensions.conf and sip.cfg |
| 18:19.24 | Adam06 | Yes. |
| 18:19.55 | Adam06 | I already went through the wiki, they dont really explain how to use it in the dialplan |
| 18:20.12 | Adam06 | I'm new to Asterisk, so I apologize if I'm doing something retarded :/ |
| 18:21.12 | Aurs | when I have a queue with strategy ringall, I get a lot of unanswered calls on my phones, because the queue starts a new call after the timeout value in the queue. does anyone have a solution for this? |
| 18:22.21 | [TK]D-Fender | Adam06: SIPAddHeader(Alert-Info: Ring Answer) |
| 18:22.22 | festr__ | Aurs: rise timeout value |
| 18:22.51 | Adam06 | replace Set(ALERT_INFO="Ring Answer") with SIPAddHeader(Alert-Info: Ring Answer) |
| 18:23.11 | [TK]D-Fender | Adam06: Though I'm not sure about the way you brok up your Aletinfo XML group in sip.cfg |
| 18:23.25 | [TK]D-Fender | Adam06: Yes |
| 18:24.05 | [TK]D-Fender | Adam06: and I suggest you merge those 3 lines into the same XML tag. |
| 18:24.39 | [TK]D-Fender | Adam06: (11-13) Multiple parms before closing |
| 18:24.59 | [TK]D-Fender | Adam06: I did this myself for the first time yesterday. |
| 18:25.03 | Adam06 | wait, do what? |
| 18:25.04 | Adam06 | lol |
| 18:25.29 | Adam06 | ok, How do I do that? |
| 18:25.36 | [TK]D-Fender | Adam06: Polycom auto-answer. I first tried & succeeded yesterday. |
| 18:26.19 | [TK]D-Fender | Adam06: <alertInfo voIpProt.SIP.alertInfo.1.value="" voIpProt.SIP.alertInfo.1.class="" voIpProt.SIP.alertInfo.2.value="Ring Answer" voIpProt.SIP.alertInfo.2.class="4" voIpProt.SIP.alertInfo.3.value="Internal" voIpProt.SIP.alertInfo.3.class="5" /> |
| 18:26.29 | Adam06 | always good to know. I just tested the SIPAddHeader and it didn't do the trick... sits and rings for 15 seconds and goes to VM |
| 18:26.43 | *** join/#asterisk Dibbler_ (n=Dibbler@dsl-217-155-254-174.zen.co.uk) |
| 18:27.03 | [TK]D-Fender | Adam06: I think its your broken tags. replace those 3 lines of <alertInfo> with the one I just gave you |
| 18:27.20 | [TK]D-Fender | Adam06: The last one may have overwritten the others. |
| 18:27.43 | mut | omfg microsoft phone system sucks |
| 18:27.46 | mut | the VAD is causing crazy static/choppiness |
| 18:27.58 | Adam06 | er, that was a snippet from sip.cfg... I can show you the entire alertInfo portion of sip.cfg if you need that |
| 18:28.45 | [TK]D-Fender | Adam06: Just substitute that and try |
| 18:30.26 | doolph | do I need to load wcfxo again after reboot? |
| 18:31.31 | Elysius | Ok. I checked my asterisk 1.2.9.1. install build with zaptel 1.2.9.1. and when I use 'zttool' I get one (1) "Wildcard TDM400P REV 1 Board 1" with "Span 1: 4 total, 2 configured, 0 active". Does that mean it worked? |
| 18:33.55 | doolph | how can I enable call waiting from another phone? |
| 18:36.59 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net) |
| 18:37.25 | tzafrir_laptop | anybody familiar with Asterisk + Lotus Notes? |
| 18:39.43 | afrosheen | oh god no not lotus notes |
| 18:40.16 | [TK]D-Fender | tzafrir : I have both. What do you have in mind? |
| 18:40.39 | tzafrir_laptop | someone is asking me... |
| 18:41.07 | tzafrir_laptop | (on the phone right now) |
| 18:42.07 | *** part/#asterisk Elysius (n=Elysius@pool-71-108-163-36.lsanca.dsl-w.verizon.net) |
| 18:46.42 | doolph | for what |
| 18:47.50 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 18:51.38 | *** join/#asterisk sting3r (n=sting3r@63.99.54.130) |
| 18:53.04 | *** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim) |
| 18:58.28 | *** join/#asterisk lowtone (i=lowtone@used.my.1337.h4x0r.skill.to.exploit.your-mom.info) |
| 18:58.39 | *** join/#asterisk hotroot (n=michael@pD9E96B89.dip.t-dialin.net) |
| 18:58.54 | lowtone | hello hello |
| 18:59.07 | *** join/#asterisk GerbilWrk (i=GerbilNu@65.88.144.41) |
| 19:00.00 | *** join/#asterisk ManxPower (n=ManxPwer@71-8-11-111.dhcp.leds.al.charter.com) |
| 19:00.26 | *** join/#asterisk bkw_ (n=bkw_@asterisk/friend-and-developer/bkw) |
| 19:01.56 | Aurs | festr__: no, I don't want to do that, because if I have a queue member that is on the phone, I want it to start calling at his phone when he hangs up. So I "have to" have a low timeout value |
| 19:03.45 | shodan | is it possible to receive fax with a linksys spa-3102 ? |
| 19:06.01 | mercestes | shodan: Is it an ATA? |
| 19:06.21 | mercestes | shodan: fax + ata is *BAD*. It's possible to recieve *A* fax...if you baby it. |
| 19:07.26 | mercestes | shodan: If you use g711ulaw with no echo cancelling and no jitter buffer under perfect network conditions on a LAN about 50% of the time if you have an exceptionally stable ATA Device. |
| 19:08.35 | GerbilWrk | does anyone have a way to monitor when Asterisk freezes up or goes down? |
| 19:10.38 | *** join/#asterisk Apturaa (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 19:10.41 | afrosheen | GerbilWrk: does asterisk freeze up or go down for you? |
| 19:10.47 | GerbilWrk | both |
| 19:10.54 | afrosheen | what kind of hardware? |
| 19:11.10 | GerbilWrk | running a t400p, 4port T1 card |
| 19:11.18 | afrosheen | server-wise, I mean |
| 19:11.26 | GerbilWrk | on an older Athlon XP system, with a gig of ram |
| 19:11.34 | GerbilWrk | don't have the specs in front of me |
| 19:11.42 | afrosheen | GerbilWrk: is it hand-built or commercial |
| 19:11.54 | GerbilWrk | hand built |
| 19:12.15 | GerbilWrk | it's been running great for a few months, just lately it's frozen up a few times |
| 19:12.26 | GerbilWrk | and twice since Friday it's stopped running completely |
| 19:12.28 | afrosheen | yeah we had random freeze-ups with an athlonxp box we built by hand a year or so ago before we switched to a dell server |
| 19:12.30 | GerbilWrk | asterisk that is |
| 19:12.39 | afrosheen | have you checked the power supply and run memtest on it |
| 19:13.34 | GerbilWrk | no, the os is still running great, it's just asterisk that stops responding, i can ssh into the box and do whatever, so i didn't suspect the mem or the ps |
| 19:13.48 | afrosheen | I'd be wary of the memory in that case |
| 19:14.11 | afrosheen | our * box would do random things like that as well, zero problems since we migrated it to perfect hardware :) |
| 19:14.35 | GerbilWrk | alright, i can test that |
| 19:14.48 | shodan | mercestes, yes it's an ata , I was kinda hoping it could be possible with t.38 or something |
| 19:14.53 | GerbilWrk | i'd planned to upgrade to more recent zaptel drivers and the newer asterisk release |
| 19:14.56 | GerbilWrk | still running 1.2.4 |
| 19:15.42 | afrosheen | GerbilWrk: nothing much wrong with 1.2.4, we were on that release forever :) |
| 19:16.09 | GerbilWrk | found a few issues so far, nothing dialplan changes hasn't fixed |
| 19:29.24 | *** join/#asterisk Adrian__ (n=foo@zux221-091-123.adsl.green.ch) |
| 19:32.43 | *** part/#asterisk bkw_ (n=bkw_@asterisk/friend-and-developer/bkw) |
| 19:34.00 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 19:34.25 | mercestes | shodan: t38 support in Asterisk is experimental and may not be in future revisions..>.> |
| 19:35.26 | *** join/#asterisk Gunnar (n=gunnar@101.Red-80-24-171.staticIP.rima-tde.net) |
| 19:36.11 | festr__ | Aurs: i dont understand if you have long timeout asap someone answer others stop ring? |
| 19:36.51 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 19:37.46 | Aurs | festr__: lets say all queue members are on the phone. with a high timeout, it can take unreasonable long time to answer the next call in the queue |
| 19:38.16 | shodan | mercestes, oh ok , then how about this , ATA picks up , asterisk listens , detect it's a fax because of the tone , then either using a x100p or hylafax with a real modem takes the call ? |
| 19:38.20 | festr__ | Aurs: aha i understand |
| 19:38.22 | Aurs | festr__: bcause the queue doesn't ring immediately after a member hangs up a call. it waits for the timeout. so I want the timeout to be low |
| 19:40.24 | CunningPike | I'm thinking that if I plug a Polycom IP300 into its power adapter and all that happens is that the handset makes a buzzing sound, it's fubar |
| 19:40.52 | afrosheen | CunningPike: I know this sounds stupid, but have you checked the cord that plugs the handset into the unit? |
| 19:41.09 | [TK]D-Fender | CunningPike: No display? I had that happen on an IP 600. Its crispy.... |
| 19:41.15 | afrosheen | CunningPike: polycoms are famous for LOOKING plugged in but are only halfway plugged in, you gotta hear that nice CLICK |
| 19:41.22 | *** join/#asterisk ToTo (n=ToTo@host149-109-dynamic.58-82-r.retail.telecomitalia.it) |
| 19:41.46 | CunningPike | [TK]D-Fender: No display - it's toast. I have a feeling someone used an IP500 power adapter on it........ |
| 19:42.19 | CunningPike | afrosheen: I hear ya - but that's not the problem this time |
| 19:42.23 | afrosheen | CunningPike: lol we did that before here, and one PoE plugged in once |
| 19:42.47 | CunningPike | afrosheen: 48V does funny things to a 12V device....... |
| 19:43.13 | CunningPike | Let's the factory smoke out |
| 19:43.54 | afrosheen | yeah and you know what, you can never get it back in |
| 19:44.05 | CunningPike | I know! WTF? |
| 19:44.09 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 19:45.21 | shodan | maybe it's just the power adapter that's busted ? |
| 19:45.36 | *** join/#asterisk PoWeRKiLL (n=powerkil@84.205.154.179) |
| 19:45.38 | shodan | I mean those wallwart are pretty crappy usually |
| 19:47.34 | [TK]D-Fender | CunningPike: Smoke refills are expensive... you should waste yours like that... |
| 19:48.12 | *** join/#asterisk mtaht4 (n=m@207.47.5.58.static.nextweb.net) |
| 19:48.47 | CunningPike | shodan: Nope - gives good voltage - same as another working one |
| 19:50.26 | shodan | did you test it under load ? some wallwart use SMPSs they sometime fail by having the voltage sagging under load |
| 19:53.08 | *** join/#asterisk Cyt (n=danielcy@athedsl-121035.otenet.gr) |
| 19:55.40 | profounded | anyone want to make a quick paypal $10 helping me setup an iptables rule? msg me |
| 19:56.01 | justinu|laptop | 10 bucks! i can eat lunch today! w00t |
| 19:56.07 | afrosheen | haha |
| 19:56.13 | Assid | hehee |
| 19:56.13 | profounded | id like to get to installing asterisk, but i need to get the iptables to forward packets to my other lan |
| 19:56.13 | Assid | rofl |
| 19:56.17 | shodan | you should probably ask in #iptables , then you should probably just ask |
| 19:56.55 | profounded | shodan, i appreicate, ill try that, but i offered if anyone wants to make a lil extra ;) hell i knew, id help |
| 19:56.59 | profounded | if i knew* |
| 19:57.01 | maciq | heheh |
| 19:57.02 | [TK]D-Fender | profounded: What other LAN? Got 2 NIC's? |
| 19:57.04 | shodan | iptables -t nat -A PREROUTING -p tcp --dport 1234 -j DNAT --to-destination 192.168.1.555:1235 |
| 19:57.09 | CunningPike | shodan: I tried the phone on a known working adapter - same thing |
| 19:57.24 | profounded | i want to forward packets from wlan0 to eth0 |
| 19:57.31 | shodan | CunningPike, oh too bad then , guess you get to see the insides of a polycom soon ;) |
| 19:57.34 | CunningPike | lol |
| 19:57.34 | maciq | profounded: forward, or NAT? |
| 19:57.53 | profounded | id assume NAT |
| 19:58.06 | profounded | i want the box to act as a router |
| 19:58.12 | maciq | well then iptables -t nat -A POSTROUTING -o wlan0 -j MASQUERADE |
| 19:58.19 | [TK]D-Fender | profounded: All you need to do is enable forwarding |
| 19:58.21 | Aurs | and echo "1" to ip_forward |
| 19:58.21 | profounded | then save the rules right? |
| 19:58.29 | [TK]D-Fender | profounded: Don't even need IPTABLES for that |
| 19:58.29 | maciq | echo 1 > /proc/sys/net/ipv4/ip_forward |
| 19:58.36 | [TK]D-Fender | see above :) |
| 19:58.44 | maciq | [TK]D-Fender: for NAT, he does |
| 19:58.51 | shodan | those are probably the 3 most knows networking commands hehehe |
| 19:58.58 | maciq | unless he uses fastnat, but thats not the case i guess |
| 19:59.20 | profounded | im going to use maciq's command? |
| 19:59.27 | [TK]D-Fender | maciq: is your ETH0 public? |
| 19:59.48 | profounded | my wlan0 is public btw if that was missed |
| 20:00.02 | maciq | [TK]D-Fender: hm? |
| 20:00.16 | [TK]D-Fender | profounded: OH, so you wlan is indeed your WAN link and ETH0 a private subnet? |
| 20:00.19 | *** join/#asterisk alawguy (n=mike@85-124-232-143.work.xdsl-line.inode.at) |
| 20:00.22 | profounded | yup |
| 20:00.24 | [TK]D-Fender | err... profounded |
| 20:00.28 | [TK]D-Fender | :) |
| 20:00.28 | maciq | thats what i thought ;] |
| 20:00.39 | profounded | ok so ill use that command :) |
| 20:00.54 | shodan | yummy , 0-day cheddar curds :) |
| 20:05.02 | *** join/#asterisk p0g0 (n=pogo@madwifi/support/p0g0) |
| 20:05.48 | dserban | How do I set the Callerid name when passing through a queued call? I have Set(CALLERID(name)='queuename')); And I have an error stating that CALLERID is not a defined function. Though it's something I yoinked via google, therefore the validity is probably true. Though I see that ${CALLERIDNAME} are valid variables etc... how do I go about making this work? |
| 20:06.57 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.41.209) |
| 20:07.19 | [TK]D-Fender | dserban: You on * 1.2.X? |
| 20:07.45 | Aurs | hmm |
| 20:07.57 | Aurs | dserban: you have a ) too much? |
| 20:08.09 | *** join/#asterisk jmacz (n=jmacz@190.24.97.87) |
| 20:09.10 | *** join/#asterisk gardo (n=gardo@124.104.32.152) |
| 20:10.03 | dserban | [TK]D-Fender: yes |
| 20:10.17 | GerbilWrk | dserban, you have one to many ) |
| 20:10.20 | dserban | Aurs: oops, that was me typing ;) not a paste. |
| 20:10.36 | dserban | Just re-typed it, but it's right in the code thingy, no extra )'s. |
| 20:11.09 | [TK]D-Fender | dserban: maybe the functions module didn't load for you. Have you recently upgraded from a 1.0 install? |
| 20:11.28 | GerbilWrk | try this, Set(CALLERID(name)=queruename) |
| 20:11.32 | dserban | [TK]D-Fender: Nope. Started on 1.2.10 |
| 20:11.48 | dserban | GerbilWrk: will do, brb |
| 20:12.17 | *** join/#asterisk kamileon (n=kamileon@c-71-207-212-67.hsd1.al.comcast.net) |
| 20:13.01 | Aurs | dserban: does "show function CALLERID" do anything for you? |
| 20:13.07 | dserban | macro agentqueue(queue,loc) { Background(menu/tlc-ann-1); Set(CALLERID(name)=${loc}1); |
| 20:13.18 | dserban | err oops that's my code verbatim from the config |
| 20:13.45 | dserban | Aurs: nope. Do I need to load a module? |
| 20:13.55 | dserban | Aurs: it says: |
| 20:14.06 | dserban | janus*CLI> show function CALLERID |
| 20:14.06 | dserban | No function by that name registered. |
| 20:14.13 | *** join/#asterisk ToTo (n=ToTo@host149-109-dynamic.58-82-r.retail.telecomitalia.it) |
| 20:14.19 | Aurs | and you're on version 1.2.10? |
| 20:14.26 | dserban | Aurs: 1.2.11 |
| 20:15.17 | Qwell | load func_callerid.so |
| 20:15.31 | dserban | :o |
| 20:15.54 | [TK]D-Fender | dserban: Your call format was perfectly valid. |
| 20:16.00 | dserban | Qwell: how do I get that to load by default? |
| 20:16.07 | Aurs | dserban: modules.conf |
| 20:16.10 | Qwell | dserban: don't disable autoload |
| 20:16.15 | Qwell | or load => func_callerid.so |
| 20:16.24 | dserban | Qwell: just followed what the asterisk book suggested. |
| 20:16.25 | *** join/#asterisk xtr (n=94752345@S0106000c41ed11e1.vf.shawcable.net) |
| 20:16.43 | *** join/#asterisk [hC] (n=hardcore@S01060004e21ea953.vc.shawcable.net) |
| 20:16.46 | dserban | Qwell: mmkay, it's a module. Didn't realize that. |
| 20:17.20 | *** part/#asterisk anthonyl (n=anthony@c-67-167-214-149.hsd1.il.comcast.net) |
| 20:17.32 | *** join/#asterisk fnordus (n=dnall@24.85.128.203) |
| 20:17.59 | dserban | Qwell: load => func_callerid.so? that's not the format I'm using in modules.conf...? |
| 20:18.21 | Qwell | Then what format are you using? |
| 20:18.45 | dserban | load=app_directory.so for example. |
| 20:18.51 | dserban | minus the > |
| 20:18.52 | Qwell | then use that format, it's fine |
| 20:18.59 | Qwell | => is "proper" |
| 20:19.14 | dserban | oh! 'tis? gah! |
| 20:19.26 | Qwell | doesn't matter a whole lot though |
| 20:19.54 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 20:20.56 | dserban | One more callerid question... I'm getting the number sent to the phones no problem. But the name never shows, what should I start checking to see if I've made a configuration mistake... is it a zap function or do I need to provide anything via the dialplan? |
| 20:21.21 | dserban | it's fine if it's aqueued call, but direct dials to extenstion, I'd like it to display the name as well as the number |
| 20:21.39 | *** part/#asterisk zaf (n=zaf@69.1.161.36) |
| 20:22.21 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 20:22.40 | Aurs | dserban: that can depend on the phones too. what phones are you using? |
| 20:22.58 | Aurs | some phones show only the callerid(name), others only the callerid(num) |
| 20:23.06 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-f96b91e797bd16a0) |
| 20:23.08 | Aurs | and the good ones show both |
| 20:23.56 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 20:24.18 | dserban | Aurs: they're spa-942's |
| 20:24.28 | Aurs | ok. they show both |
| 20:24.34 | dserban | Indeed. |
| 20:24.50 | Aurs | or atleast the 941 does, so I'd think that the 942 does too :) |
| 20:24.52 | *** join/#asterisk Druken (n=jdumais@out.clearnet.com) |
| 20:24.59 | dserban | Aurs: yep, same phone |
| 20:25.30 | Aurs | dserban: have you set callerid in sip.conf? |
| 20:25.39 | dserban | Btw, tool = greatest band ever.. *steps off the soapbox*, ahem, back to our regularly scheduled programming. |
| 20:25.43 | dserban | Aurs: yep. |
| 20:25.53 | Aurs | callerid="name" <123> |
| 20:26.01 | Aurs | like that, right |
| 20:26.15 | justinu | tool owns |
| 20:26.26 | Aurs | iron maiden owns |
| 20:27.06 | CunningPike | Aurs: Try it without quotes |
| 20:27.09 | dserban | Aurs: indeed yes, callerid="name" <123> |
| 20:27.22 | Aurs | CunningPike: why? |
| 20:27.27 | dserban | Aurs: iron maiden does inded kick ass |
| 20:27.33 | CunningPike | Aurs: Because it might work!! |
| 20:27.44 | Aurs | CunningPike: callerid="name" <123> works? |
| 20:27.52 | sevard | iron maiden is shite. |
| 20:28.20 | justinu | i think i'm totally fubar'd |
| 20:28.28 | justinu | <PROTECTED> |
| 20:28.28 | justinu | fsck.ext3: Bad magic number in super-block while trying to re-open /dev/VolGroup00/LogVol00 |
| 20:28.28 | justinu | Warning... fsck.ext3 for device /dev/VolGroup00/LogVol00 exited with signal 11. |
| 20:28.30 | sevard | what was that kid on the internet that sang to all those iron maiden midis |
| 20:28.40 | CunningPike | justinu: Ouch |
| 20:28.44 | sevard | justinuu: ouch |
| 20:28.47 | sevard | personal jynx |
| 20:28.54 | justinu | any advices? :P |
| 20:29.00 | sevard | use reiser :) |
| 20:29.05 | CunningPike | justinu: What distro? |
| 20:29.07 | Aurs | justinu: cry your brave tears! |
| 20:29.13 | dserban | CunningPike: ;O |
| 20:29.21 | dserban | nah, it's that I see the callerid number |
| 20:29.27 | dserban | but I never see the callerid name... |
| 20:29.28 | Strom_C | dserban: standard format is to not have spaces between the quote and the angle bracket: callerid="Name Here"<3115552368> |
| 20:29.31 | dserban | it's the same as the number |
| 20:29.43 | sevard | Strom_C: IIRC that's depreciated |
| 20:29.47 | Aurs | dserban: have you enabled autoload in modules.conf? |
| 20:29.51 | dserban | Strom_C: :o tell that to the people editing the voip-info wiki ;) |
| 20:29.53 | CunningPike | dserban: Seriously, try it without quotes.... |
| 20:29.54 | Strom_C | sevard: what? |
| 20:30.01 | dserban | Aurs: no, I haven't |
| 20:30.06 | sevard | Oh, I thought you were asking about setting it in the dialplan. |
| 20:30.11 | Strom_C | no |
| 20:30.13 | Strom_C | conf files |
| 20:30.23 | sevard | How you been nigz |
| 20:30.29 | dserban | CunningPike: it's not what the problem is though :s |
| 20:30.29 | sevard | I'm looking for a payphone. |
| 20:30.35 | CunningPike | dserban: OK |
| 20:30.59 | justinu | the machine is still running, but filesystem is in readonly mode. |
| 20:31.18 | Aurs | dserban: also try to not have the same thing as name and num |
| 20:31.21 | CunningPike | justinu: Do a backup your data and rebuild |
| 20:31.27 | justinu | centos 4.2 |
| 20:31.29 | dserban | CunningPike: I mean, incoming calls on the PRI come through to the phones with numbers no problem. But no names. |
| 20:31.32 | sevard | CentOS :\ |
| 20:31.33 | CunningPike | s/backup/backup of/ |
| 20:31.34 | justinu | yeah, that's what i think too. |
| 20:31.38 | Aurs | that way, you can tell if it's showing name or num |
| 20:32.03 | CunningPike | dserban: Are you getting number instead of name? |
| 20:32.05 | dserban | Aurs: I missed what you were saying? |
| 20:32.19 | CunningPike | dserban: We get that on 90% of our calls - telco has no clue |
| 20:32.34 | Strom_C | dserban: does name show up on station-to-station calls? |
| 20:32.35 | Aurs | dserban: you said that you had the same name and num. in callerid. try to have different name and num, so you can tell if it's showing name or num |
| 20:32.38 | dserban | CunningPike: yes for incoming calls only, everything in sip.conf shows up correctly... ie, my name show up at the internal extension I'm dialing. |
| 20:32.48 | CunningPike | dserban: Same here |
| 20:32.55 | Aurs | ah, ok |
| 20:32.56 | Strom_C | dserban: do you have calling name provisioned on your PRI? |
| 20:33.10 | CunningPike | dserban: Where are you and who is your telco? |
| 20:33.23 | dserban | Strom_C: Yes, it worked with the norstar system that asterisk took over from |
| 20:33.41 | sevard | Who is your daddy and what does he do? |
| 20:33.42 | *** join/#asterisk Apturaa (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 20:33.44 | Strom_C | dserban: pastebin zapata.conf |
| 20:33.54 | dserban | CunningPike: in kelowna (we've talked about this before ;) and telus provides the line, but allstream provisions it |
| 20:34.05 | *** join/#asterisk qdk (n=qdk@0x535eae17.boanxx9.adsl-dhcp.tele.dk) |
| 20:34.19 | CunningPike | dserban: Ah - lol - we are exactly the same, then |
| 20:34.26 | Apturaa | hi dserban |
| 20:34.50 | dserban | CunningPike: :o |
| 20:34.55 | dserban | Apturaa: hullo |
| 20:35.40 | *** join/#asterisk syzygyBSD_ (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 20:35.46 | Defraz | Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) |
| 20:35.50 | Defraz | ooops |
| 20:35.53 | dserban | Strom_C: http://pastebin.ca/175484 |
| 20:35.53 | Apturaa | dserban useing * for bussiness or resale |
| 20:35.59 | Defraz | I recompiled the latest asterisks and that is what I am getting now |
| 20:36.05 | dserban | Apturaa: bidness |
| 20:36.23 | Defraz | I do see a file in the directory called asterisk.ctl and it works fine except I can't do an asterisk -r |
| 20:36.24 | Apturaa | In the stage if implementation? |
| 20:36.26 | Strom_C | dserban: and where's the rest of it? :) |
| 20:36.37 | dserban | Strom_C: oh heh, the channels one? |
| 20:36.39 | *** join/#asterisk afrosheen (n=cj@txprotoa2.august.net) |
| 20:36.40 | Strom_C | yes |
| 20:36.40 | Defraz | Did I miss a compile option or something |
| 20:36.42 | CunningPike | Defraz: Asterisk isn't starting - start it from the command line with -vvvgc and see what you get? |
| 20:36.43 | dserban | Apturaa: deployed |
| 20:36.44 | Strom_C | that would be kind of important |
| 20:36.54 | Defraz | oh it is started I have my ata regged to it. |
| 20:36.59 | dserban | Strom_C: :) NEVER! |
| 20:37.02 | dserban | ... hold on i'll paste ie |
| 20:37.04 | dserban | it* |
| 20:37.39 | Aurs | Defraz: ps aux | grep asterisk |
| 20:37.45 | shodan | what's the deal with that "TELEOPEN.NET PCI GSM Card" x100p compatible ? is there anywhere that actually sell this thing ? |
| 20:37.48 | dserban | Strom_C: http://pastebin.ca/175485 |
| 20:38.11 | dserban | Apturaa: Do you work for nortel? |
| 20:38.16 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 20:38.16 | Defraz | it says /usr/sbin/asterisk -c -v -g -p -U asterisk -G asterisk |
| 20:38.22 | Defraz | yea it is running. |
| 20:38.23 | Strom_C | dserban: try adding "callerid=asreceived" to channels 1-23 |
| 20:38.27 | Defraz | I can watch the logs and stuff |
| 20:38.37 | CunningPike | Defraz: Try sudo /usr/sbin/asterisk -r |
| 20:38.38 | Defraz | but I thought I would be able to slip in with asterisk -r |
| 20:38.45 | dserban | Strom_C: will do, had it there at one time, don't remember why I removed it. I'll try it. |
| 20:38.45 | Defraz | I am root |
| 20:38.49 | Defraz | but okay |
| 20:38.59 | Defraz | same thing |
| 20:39.30 | CunningPike | Defraz: That would only make a difference if you weren't logged in as root - which you shouldn't be, hint, hint ;) |
| 20:39.42 | Defraz | I know I know |
| 20:39.49 | CunningPike | :D |
| 20:39.57 | Defraz | <PROTECTED> |
| 20:40.18 | CunningPike | Defraz: What does ps -ef | grep asterisk show? |
| 20:40.55 | Defraz | where should I paste it. |
| 20:41.10 | Defraz | it grabbed the http procs running as astrisks |
| 20:41.49 | Defraz | kinda big |
| 20:42.11 | CunningPike | ~pb |
| 20:42.20 | jbot | well, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 20:42.24 | CunningPike | There ya go |
| 20:42.49 | afrosheen | was that poke really necessary |
| 20:43.31 | CunningPike | afrosheen: I didn't poke you :) |
| 20:43.37 | Apturaa | CunningPike what would cause Voicemail(umailboxnumber) to not work in a example of implementing a DID when internal for that same extention does work? |
| 20:44.06 | CunningPike | Apturaa: Not sure...... you got some CLI output to look at? |
| 20:44.59 | Defraz | okay CunningPike it is on pastebin.ca |
| 20:45.08 | riddlebox | how would I make an extension a "hot phone" so that as soon as it goes offhook it will dial a certain number? |
| 20:45.19 | Juggie | the phone has to support that. |
| 20:45.26 | CunningPike | Nope, not it |
| 20:45.30 | CunningPike | No..... |
| 20:45.33 | CunningPike | Not that...... |
| 20:45.44 | Defraz | http://pastebin.ca/175490 |
| 20:45.47 | CunningPike | :D |
| 20:45.54 | CunningPike | j/k, Defraz |
| 20:46.11 | Defraz | haha sorry just wanting to make it as simple as possible. |
| 20:47.02 | *** join/#asterisk viler (i=1000@200.114.70.228) |
| 20:47.11 | *** join/#asterisk mivck (i=1000@200.114.70.228) |
| 20:48.13 | Defraz | I don't get it. I have it running it works, amp connects to it to admin it. It reloads it cleared up a lot of issues I was having but now I can asterisk -r into it. |
| 20:48.34 | CunningPike | Defraz: Weird |
| 20:48.49 | Defraz | yea |
| 20:48.54 | Apturaa | Verbosity is at least 3 |
| 20:48.54 | Apturaa | <PROTECTED> |
| 20:48.54 | Apturaa | <PROTECTED> |
| 20:48.54 | Apturaa | <PROTECTED> |
| 20:48.54 | Apturaa | <PROTECTED> |
| 20:48.56 | Apturaa | <PROTECTED> |
| 20:48.58 | Apturaa | <PROTECTED> |
| 20:48.59 | CunningPike | Defraz: Have you modified safe_asterisk at all? |
| 20:49.00 | Apturaa | <PROTECTED> |
| 20:49.10 | Defraz | I did but after it stopped working |
| 20:49.18 | CunningPike | Defraz: Hmm |
| 20:49.21 | Defraz | I put the -c in there so it would console |
| 20:49.28 | Defraz | but I can take it out and it does the same thing |
| 20:49.45 | Apturaa | there you go CunningPike |
| 20:49.45 | CunningPike | Defraz: Ya - we have -c there, too |
| 20:49.51 | CunningPike | Apturaa: Pastebin your extensions.conf |
| 20:51.26 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 20:52.12 | Defraz | and what does this module do. |
| 20:52.13 | Defraz | app_trunkisavail.so |
| 20:52.33 | CunningPike | Apturaa: OK - that's the CLI output - now pastebin your extensions.conf - and let's keep it in the main channel ;) |
| 20:52.54 | eKo1 | Defraz: If you had to guess, what do you think it does? |
| 20:53.35 | Defraz | I thought it might handle the trunks and if one was down it would right it to the next. |
| 20:54.28 | Defraz | Maybe it is built into the new asterisk because it errored out on compilation of the new 1.2.12.1 |
| 20:54.44 | Defraz | so I backed it up and asterisk started |
| 20:57.05 | *** join/#asterisk freebsd_fan (n=ebola@200.32.71.142) |
| 20:58.27 | Apturaa | An anonymous reader writes "Sam Houston State University (SHSU) is moving 6,000 users off a Cisco VoIP platform to an open-source VoIP network based on Asterisk |
| 20:58.54 | Apturaa | got that from sinapps |
| 20:59.32 | CunningPike | Apturaa: Not anonymous - it was on an IT web site - hang on..... |
| 20:59.53 | CunningPike | http://www.networkworld.com/news/2006/091206-von-sam-houston.html?page=1 |
| 21:00.24 | Defraz | Yea for Texas |
| 21:00.24 | Apturaa | yea thats pretty impressive and probebly what the biggest example how asterisk will be deployed? |
| 21:00.47 | CunningPike | Apturaa: Not the biggest, but the biggest that are actually talking about it |
| 21:00.47 | Defraz | That is awesome. |
| 21:01.21 | Defraz | So you think it might be a permissions thing |
| 21:01.32 | CunningPike | Apturaa: Did you pb your extensions.conf? |
| 21:02.15 | dashu | how can i dial someone with asterisk :o |
| 21:02.31 | dashu | is there a command for the console ? |
| 21:02.32 | CunningPike | dashu: Pick up the phone........ ;) |
| 21:02.42 | dashu | :p without the phone |
| 21:02.42 | justinu | let your fingers do the walking |
| 21:02.50 | CunningPike | dashu: Ah - I believe there is an addon for that.... |
| 21:02.58 | dashu | :O really |
| 21:03.02 | CunningPike | dashu: Or FOP |
| 21:03.07 | afrosheen | yeah fop can create calls |
| 21:03.13 | CunningPike | ~fop |
| 21:03.21 | jbot | An XSL formatter written in Java that outputs PDF. URL: http://www.jtauber.com/fop/, or the Flash Operator Panel |
| 21:03.21 | dashu | ok cool ill look for this :p |
| 21:03.21 | Apturaa | CunningPike no because I am in the process of writting new extentions and dump @home dont have another server at the moment so once its done it will be just a strait forward plan |
| 21:04.43 | justinu | fsck is working now... lots of problems reported |
| 21:04.49 | justinu | wonder what the chances it'll let me at the data |
| 21:05.33 | m_o_d | hello |
| 21:05.48 | afrosheen | justnu: did you have a hd crash? |
| 21:06.28 | justinu | something like that |
| 21:06.33 | m_o_d | if i want test my network with sip PBX can i use to that asterisk ? |
| 21:06.44 | justinu | not sure if it's actual hardware failure, or a filesystem corruption |
| 21:06.54 | Apturaa | CunningPike http://pastebin.ca/175507 It works just does not goto vm |
| 21:07.05 | Apturaa | vm works with internal |
| 21:07.07 | afrosheen | m_o_d, if I understand your strangely formatted question, yes |
| 21:07.10 | Apturaa | and zap |
| 21:07.26 | Apturaa | mabey I should have a include |
| 21:08.37 | CunningPike | Apturaa: You're busy priority should be 5, not 105 |
| 21:08.51 | Apturaa | okay |
| 21:09.15 | CunningPike | Apturaa: Or the the other way round - at any rate, you need an n+101 for one, and an n+1 for the other |
| 21:09.45 | dashu | is it ok if i paste some output i get from asterisk and u look what might be wrong there ? |
| 21:09.50 | CunningPike | Apturaa: So, 5 and 105 respectively |
| 21:09.52 | *** join/#asterisk pollohawk (n=pollohaw@mmail.picksend.com) |
| 21:09.58 | CunningPike | dashu: pastebin |
| 21:10.00 | CunningPike | ~pb |
| 21:10.02 | jbot | pb is, like, a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 21:10.02 | m_o_d | afrosheen: i want create voip centralbut without calling world only for my subnet (lokal users) |
| 21:10.18 | CunningPike | m_o_d: Then asterisk is your man |
| 21:10.22 | dashu | :o ok i try that |
| 21:11.12 | Apturaa | btw its always may me wonder why n+101 that would make the next priority 106. |
| 21:11.29 | CunningPike | Apturaa: http://pastebin.ca/175512 |
| 21:11.43 | pollohawk | Has anyone got the UIP300 telephone to work with ooh323 |
| 21:11.58 | CunningPike | Apturaa: The 'n' in this case is 4 - it's the 'n' of the Dial() statement that matters |
| 21:12.23 | m_o_d | second question, i read about cdr to mysql db but i can't find information about storing users information in db so my question: can i do that? |
| 21:12.53 | dashu | http://pastebin.ca/175515 |
| 21:12.54 | Apturaa | i see |
| 21:13.02 | dashu | anyone knows what might be wrong ? |
| 21:13.29 | CunningPike | m_o_d: look up asterisk realtime |
| 21:13.46 | afrosheen | m_o_d, yeah asterisk is a perfect fit for that |
| 21:13.57 | ManxPower | Apturaa, Why don't you just look at std-exten macro in extensions.conf.sample? |
| 21:14.00 | ManxPower | dashu, I'm still waitring for pastebin to load |
| 21:14.02 | ManxPower | Sorry. I can't help you with CAPI |
| 21:14.13 | CunningPike | dashu: Me either - sorry |
| 21:14.46 | dashu | :) |
| 21:14.59 | dashu | anyone knows who can help me with capi stuff ? |
| 21:15.35 | dashu | anyone knows someone who can help me with capi stuff ? |
| 21:15.51 | *** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.wa.comcast.net) |
| 21:15.56 | aao_pwner | who has a good tutorial for asterisk |
| 21:16.03 | aao_pwner | preferibly step-by-step |
| 21:16.06 | *** join/#asterisk pdt (n=pdthome@c-68-53-40-50.hsd1.tn.comcast.net) |
| 21:16.37 | [TK]D-Fender | aao_pwner : No such thing really, but you would be best served by reading "The Book" |
| 21:16.38 | [TK]D-Fender | ~book |
| 21:16.40 | jbot | well, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 21:16.55 | Apturaa | pb? |
| 21:16.56 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 21:17.05 | Apturaa | ignore that. |
| 21:17.06 | intralanman | ~pb |
| 21:17.07 | jbot | pb is probably a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 21:17.10 | intralanman | too late |
| 21:17.43 | *** join/#asterisk andresmujica (n=andresmu@201.245.230.87) |
| 21:17.53 | andresmujica | hi, anyone out there knows how can i integrate an asterisk box with a quintum 32000 tenor gateway |
| 21:18.30 | riddlebox | andresmujica, haha I tried to get one working as well but couldnt |
| 21:19.05 | riddlebox | andresmujica, quintum just showed me how to get their stuff working with their own hardware even though they said it would work with asterisk out of the box |
| 21:19.25 | andresmujica | ohh!! |
| 21:19.35 | andresmujica | and you know if the 3200 supports SIP ? |
| 21:19.48 | andresmujica | or did you use the PRI interface? |
| 21:19.56 | riddlebox | andresmujica, they also told me it would work with avaya stuff and it didnt |
| 21:20.08 | andresmujica | oh, men.. |
| 21:20.16 | andresmujica | that' s serious then... |
| 21:20.26 | riddlebox | we used the small ones, ummm it was like a 4 port ata device |
| 21:20.38 | andresmujica | looking at the website they have somee mention about asterisk... |
| 21:20.51 | andresmujica | but i couldn' t find good technical documentation about my unit... |
| 21:20.56 | andresmujica | so im a bit lost... |
| 21:21.02 | riddlebox | the docs are horrible |
| 21:21.40 | *** join/#asterisk Elysius (n=Elysius@pool-71-108-163-36.lsanca.dsl-w.verizon.net) |
| 21:21.57 | andresmujica | i think so.. |
| 21:22.17 | riddlebox | andresmujica, my quintum salesman says they are working to get avaya certified |
| 21:22.17 | E-bola | heh |
| 21:22.24 | E-bola | i changed sip->pstn provider |
| 21:22.29 | E-bola | and all of a sudden all my echo problems are gone |
| 21:22.35 | E-bola | and imagine all the time ive wasted investigating :/ |
| 21:23.39 | Elysius | I got my Fedora Core (FC) 5 Linux with asterisk 1.2.9.1 and zaptel 1.2.9.1 to work; I had an extra "[channel(s)]" line in my '/etc/asterisk/zapata.conf' configuration. My issue that I am having is now when the FC5 Linux box is receiving an incoming call on the FXO module, it picks up then hangs up. Suggestions? |
| 21:24.59 | [TK]D-Fender | Elysius : pastebin your zaptel & zapata |
| 21:25.38 | Elysius | [TK]D-Fender: Will do in a few moments; have to go talk with the boss. :) |
| 21:27.10 | *** part/#asterisk rokerr (n=rowan@38.99.162.187) |
| 21:28.13 | *** join/#asterisk rollergrrl (n=0x3e44d@71-213-6-123.slkc.qwest.net) |
| 21:28.24 | *** part/#asterisk rollergrrl (n=0x3e44d@71-213-6-123.slkc.qwest.net) |
| 21:36.26 | *** join/#asterisk jeremy_g (n=j@83.233.40.109) |
| 21:37.37 | *** join/#asterisk `Sauron (i=sauron@h-69-3-12-50.hstqtx02.covad.net) |
| 21:38.50 | *** join/#asterisk profounded (n=pro@ool-44c4eae2.dyn.optonline.net) |
| 21:40.44 | twisted[asteria] | 000DBD4B14CC |
| 21:41.00 | twisted[asteria] | INM07326404 |
| 21:45.34 | *** join/#asterisk stargazer_gr (n=stargr@adsl98-12dynamic.salonica.acn.gr) |
| 21:46.06 | andresmujica | http://www.quintum.com/support/1G/kb/system/Tenor_Call_Routing.pdf something more tech about quintum |
| 21:46.44 | Apturaa | CunningPike for some reason now its giving a non zero on the cli and phone is not ringing. |
| 21:46.59 | andresmujica | http://www.quintum.com/support/xplatform/network/Using_SIP.pdf a litle about sip |
| 21:47.45 | *** part/#asterisk FuriousGeorge (n=FuriousG@ool-43536ea8.dyn.optonline.net) |
| 21:51.56 | *** join/#asterisk negatendo (n=negatend@70.90.116.93) |
| 21:55.01 | *** join/#asterisk carrar (i=tim@osburn.com) |
| 21:56.26 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 21:57.00 | eKo1 | ah! Quintums |
| 21:57.11 | eKo1 | *hides rather |
| 22:00.30 | *** join/#asterisk Druken (n=jdumais@CPE000854de4ec0-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 22:02.47 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 22:05.31 | *** join/#asterisk smackus (n=ckwall@63.149.122.93) |
| 22:06.29 | smackus | in my queue database I am seeing "agent dump". what action would this be referring to? |
| 22:07.00 | smackus | 2006-09-18 14:41:08 ENTERQUEUE NONE 2006-09-18 14:50:15 AGENTDUMP SIP/3584-00b6a850 2006-09-18 14:50:28 AGENTDUMP SIP/3584-00a6a230 2006-09-18 14:50:39 AGENTDUMP SIP/3584-00a9fa40 2006-09-18 14:50:47 AGENTDUMP SIP/3584-00a66d10 2006-09-18 14:50:55 AGENTDUMP |
| 22:08.58 | sevard | Is anyone pretty good with sed? |
| 22:09.19 | eKo1 | sevard: compared too? |
| 22:09.50 | sevard | compared to me ;/ |
| 22:10.11 | sevard | I'm really not getting this, I have a 15-30% understanding of regular expressions but really can't figure out "fields" in sed. My CSV is denoted field1,field2,field3 and I'm looking at adding extra fields after field2, so it'd be field1,field2,extra,extra2,field3 |
| 22:10.18 | Druken | sevard: aren't we all?...... what is sed btw? |
| 22:10.40 | Druken | ouch! |
| 22:11.44 | *** join/#asterisk h3x0r4t0r (n=hex@ip68-224-236-92.lv.lv.cox.net) |
| 22:11.49 | sevard | I've figured out how to edit the 'first' field in any csv file but I can't edit the second or third without still editing the first. |
| 22:12.51 | eKo1 | sevard: don't use sed, use a shell script |
| 22:13.30 | sevard | People usually use sed within their shell scripts, can you be more descriptive? |
| 22:14.16 | eKo1 | make a script that reads each field and reconstructs the cdr with the two extra fields. |
| 22:14.44 | sevard | I could use cut and put each field into a mysql database and reconstruct the csv from the database |
| 22:14.53 | sevard | but then i though, jesus, there's an easier way, sed must be able to do this |
| 22:15.00 | sevard | thought* |
| 22:15.27 | *** join/#asterisk luckasrg (n=lucasrg@204.97.196.20) |
| 22:15.37 | luckasrg | hello all |
| 22:15.40 | sevard | I know a guy who is fluent in perl could do this in about 10 seconds |
| 22:15.42 | sevard | grr. |
| 22:15.44 | sevard | hey lu |
| 22:15.46 | sevard | hey luckasrg |
| 22:16.26 | eKo1 | sevard: yeah, with perl, that is dead simple |
| 22:16.27 | luckasrg | hi i have been looking on the internet how can i set up two peers on asterisk as passtrougt |
| 22:16.43 | *** join/#asterisk QbY (n=Kelvin@cm-64-221-172-88.dhcp.southerncoastalcable.net) |
| 22:17.00 | luckasrg | meaning sending rtp directly from one to the other |
| 22:17.31 | luckasrg | only using the asterick as the signaling device how can i do this? |
| 22:18.53 | *** join/#asterisk myshenka (n=spamyous@82.153.170.213) |
| 22:23.15 | QbY | Is there anyway to get the channel name without the "unique" id associated with it? Ie. ${CHANNEL} returns me IAX2/1000-19, where I only need IAX2/1000 |
| 22:25.51 | stubert | QbY: Use cut |
| 22:26.00 | QbY | i just stumbled across that one.. thanks.. |
| 22:26.07 | stubert | sure |
| 22:26.24 | sevard | does anyone know the escape sequence of a carriage return |
| 22:26.49 | sevard | \r |
| 22:26.50 | sevard | stupid brain |
| 22:27.38 | *** join/#asterisk BigBadHoss (i=HydraIRC@68-191-130-114.static.dctr.al.charter.com) |
| 22:28.04 | *** part/#asterisk smackus (n=ckwall@63.149.122.93) |
| 22:28.16 | stubert | or \0x013 |
| 22:28.32 | BigBadHoss | can I use an old MLS-18D avaya phone for IP, or is is proprietary? |
| 22:29.07 | tzafrir_laptop | 0xD or 13 or \015 . Make up your mind, please |
| 22:29.21 | sevard | show offs |
| 22:30.09 | tzafrir_laptop | stubert, you see, because of your miss-marking, h3x0r has quit |
| 22:30.10 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 22:30.45 | *** join/#asterisk sbma44 (n=tomlee@dsl092-174-002.wdc2.dsl.speakeasy.net) |
| 22:30.54 | BigBadHoss | anyone know? |
| 22:31.45 | Druken | BigBadHoss: i belive avaya is proprietary |
| 22:31.56 | BigBadHoss | damn |
| 22:32.00 | BigBadHoss | ok thanks |
| 22:32.00 | sbma44 | hey folks. getting some very weird behavior on sip trunks. two diff vendors, same symptom: inbound calls show up in sip debug and console log (dialplan execution), but no audio. yet using a diff inbound SIP provider -- one that doesn't require registration (ipkall.com) -- it works fine |
| 22:32.28 | sbma44 | I've been wrestling with this for a while, and feel totally stumped. anyone have any ideas? |
| 22:32.37 | QbY | stubert: I'm trying.. exten => 1000,n,CUT(agentchannel=CHANNEL,-,-2) --- But still getting the whole thing.. |
| 22:32.39 | BigBadHoss | nat? |
| 22:33.55 | sbma44 | bigbadhoss: you responding to me? it's a possibility, but I don't understand how it would affect registered inbound calls but not the ipkall.com ones. esp. since sip show registry and sip show peers make it look like everything's fine |
| 22:35.19 | stubert | QbY: us Set(Chan=${CUT(CHANNEL,,1)}) |
| 22:36.03 | stubert | ${Chan} will be what you are looking for |
| 22:36.36 | QbY | Perfecto |
| 22:37.16 | *** part/#asterisk myshenka (n=spamyous@82.153.170.213) |
| 22:39.41 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 22:40.50 | BigBadHoss | set it up under a dmz and see what happens |
| 22:41.50 | *** join/#asterisk Corydon76-home (i=red@pdpc/supporter/sustaining/Corydon76-home) |
| 22:41.50 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 22:46.43 | *** join/#asterisk Givemelove (n=foo@208.57.229.162) |
| 22:51.37 | luckasrg | how can i bypass the asterisk for the rtp |
| 22:52.09 | luckasrg | only use asterisk as signaling? |
| 22:52.38 | tamp4x | camreinvite=yes |
| 22:52.58 | _DAW | and that only works under certain conditions |
| 22:53.35 | luckasrg | does it work for sip calls |
| 22:55.28 | luckasrg | on the same lan |
| 22:55.56 | _DAW | it can if it the same codec, no nat, etc... but asterisk is not a true proxy |
| 22:58.26 | luckasrg | will it work for a hosted pbx kind of setup |
| 23:01.45 | _DAW | it can |
| 23:02.04 | *** part/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 23:02.32 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 23:04.02 | stubert | anybody know what the best dtmfmode setting would be between 2 asterisk boxes connecting via sip and using a gsm codec? |
| 23:06.31 | *** join/#asterisk RoyK (n=roy@gprs-ggsn5-nat.mobil.telenor.no) |
| 23:07.59 | Strom_C | rfc2833...but really you should be using iax2 for inter-asterisk connections |
| 23:08.14 | tzafrir_laptop | stubert, anything that is not inline |
| 23:14.08 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 23:14.23 | *** part/#asterisk andresmujica (n=andresmu@201.245.230.87) |
| 23:18.34 | RoyK | <PROTECTED> |
| 23:19.54 | tzafrir_laptop | That is a strange thing to do |
| 23:30.23 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 23:31.45 | L|NUX | can some one tell me how can i know that PRI installed on Asterisk Server i am using Sangoma Card |
| 23:32.17 | stubert | tzafrir_laptop: thanks... |
| 23:34.19 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 23:34.44 | tzafrir_laptop | L|NUX, zttool? cat /proc/zaptel/* ? |
| 23:35.40 | L|NUX | ok |
| 23:35.40 | L|NUX | thanks |
| 23:36.01 | L|NUX | but i am using sangoma |
| 23:36.34 | CunningPike | L|NUX: But /dev/zap should still exist |
| 23:36.45 | L|NUX | shit |
| 23:36.45 | L|NUX | my net |
| 23:38.27 | L|NUX | ok |
| 23:42.10 | L|NUX | i do not have zttool command |
| 23:42.15 | L|NUX | however when i did cat |
| 23:42.18 | L|NUX | i got this |
| 23:42.18 | L|NUX | Span 1: ZTDUMMY/1 "ZTDUMMY/1 1" |
| 23:43.06 | *** part/#asterisk hotroot (n=michael@pD9E96B89.dip.t-dialin.net) |
| 23:46.00 | *** join/#asterisk RoyKa (n=roy@ti211210a080-4037.bb.online.no) |
| 23:47.53 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 23:50.04 | aao_pwner | uhh, for some reason zaptel wont compile, apperentally because erm what's that.. |
| 23:50.13 | aao_pwner | newt is being gay again |
| 23:50.45 | L|NUX | sorry was dc |
| 23:51.07 | L|NUX | tzafrir_laptop : i have sangoma card |
| 23:51.09 | L|NUX | tzafrir_laptop : not zaptel |
| 23:51.15 | L|NUX | not digium |
| 23:51.41 | tzafrir_laptop | it doesn't use zaptel drivers at all? |
| 23:51.55 | L|NUX | i belive it will not use zaptel |
| 23:52.00 | L|NUX | it use wanpipe |
| 23:55.15 | *** join/#asterisk stronginteractio (n=Jay@lgb-static-66.18.140.106.mpowercom.net) |