| 00:01.14 | *** join/#asterisk Braxus (n=braxus@netblock-68-183-228-52.dslextreme.com) |
| 00:05.55 | drmessano | Asterisk needs more games |
| 00:06.24 | drmessano | Like "Higher/Lower" |
| 00:07.40 | mocker | I want a Simon game. |
| 00:07.46 | mocker | I think it's simon anyway. |
| 00:07.56 | mocker | that had the lights that you pressed in an order and had to repeat them back. |
| 00:07.59 | drmessano | What about a 12 panel minesweeper |
| 00:08.04 | drmessano | One for each button |
| 00:08.09 | drmessano | Yes, love Simon |
| 00:08.18 | puck | Argh, crappy LinkSys PAP2 ATAs that have crashy webservers |
| 00:08.26 | drmessano | lol |
| 00:08.40 | drmessano | crashy? |
| 00:08.49 | drmessano | how does the web server on a PAP2 crash? |
| 00:08.58 | puck | It just stops responding. |
| 00:09.08 | drmessano | Ive not seen that |
| 00:09.11 | puck | I hit the IP, and then none of the links work, and I can't reload the page. |
| 00:09.19 | drmessano | VPN> |
| 00:09.20 | drmessano | ? |
| 00:09.26 | puck | oh, oh, oh, its coming, just slowly. |
| 00:09.29 | puck | Nope, local network. |
| 00:09.37 | puck | (almost local, routed through one box) |
| 00:09.39 | jblack | puck: I noticed that my spa8000 gets really laggy when it's doing a register. |
| 00:09.41 | drmessano | Ah |
| 00:09.58 | puck | jblack: hmmmm, could be it... |
| 00:09.58 | drmessano | Routing AT ALL messed up the GUI on the PAP2 |
| 00:10.05 | drmessano | It doesnt work across a simple VPN |
| 00:10.08 | puck | drmessano: Oh really? |
| 00:10.11 | drmessano | Yes |
| 00:10.15 | puck | that bites. |
| 00:10.16 | drmessano | Thats the GUI now |
| 00:10.21 | drmessano | Not the device in general |
| 00:10.57 | drmessano | I VNC into boxes on the local subnets and admin them that way |
| 00:11.03 | drmessano | Its SUX0RS |
| 00:11.07 | drmessano | But thats the only way |
| 00:11.29 | jblack | what are the common names from G711u, G711a, G726-16, G726-24, G726-32, G726-40, G723 ? |
| 00:11.36 | drmessano | Ever try to admin a WRT54G on a diff segment? |
| 00:11.41 | Qwell | common names? |
| 00:11.58 | jblack | Well, for example, I think g711u is ulaw, but I'm not sure. |
| 00:12.00 | mocker | ulaw alaw ? |
| 00:12.34 | Qwell | ulaw and alaw, and I think g726 is a form of ADPCM |
| 00:12.45 | Qwell | (but there is also an ADPCM format...which is not g726) |
| 00:12.59 | puck | Nope, never tried a WRT54G. All our people tend to reflash OpenWRT on them anyhow. |
| 00:13.11 | puck | No responce in the text browser. :( |
| 00:13.11 | drmessano | Is this for the Linksys? |
| 00:13.24 | jblack | Who, me? |
| 00:13.26 | drmessano | jblack: for the SPA800 |
| 00:13.29 | drmessano | jblack: for the SPA8000 |
| 00:13.32 | jblack | Yup |
| 00:13.38 | mocker | SPA80000 |
| 00:13.40 | mocker | WOO |
| 00:13.41 | drmessano | PCMU, PCMA |
| 00:13.58 | drmessano | G726-16, etc for G726 |
| 00:14.07 | Qwell | g723 is also known as "that patent encumbered heavy cpu usage codec" |
| 00:14.17 | drmessano | G729ab |
| 00:14.22 | drmessano | G723 |
| 00:14.27 | drmessano | Those are the defaults |
| 00:14.41 | jblack | so, G723 is one of the evil ones. |
| 00:14.47 | mocker | Is speex in there somewhere? |
| 00:14.49 | Qwell | g723 is a great codec |
| 00:14.54 | mocker | Or is it just known as speex. |
| 00:15.01 | tzanger | ilbc FTW |
| 00:15.08 | tzanger | or wait, LPC10 |
| 00:15.10 | Qwell | mocker: it's just speex. I don't know if speex is a "spec" |
| 00:15.20 | jblack | There is 711, 726, 729 and 723 |
| 00:15.23 | Qwell | tzanger: lpc10 using the JPAH swec |
| 00:15.28 | tzanger | haha |
| 00:15.34 | drmessano | jblack: did you follow the names? |
| 00:15.38 | Qwell | JPAH is the most awesome EC ever. |
| 00:15.39 | jblack | I thought 729 was the evil one. |
| 00:15.45 | Qwell | jblack: g729 too |
| 00:16.10 | Qwell | I mean...seriously... how could a codec named after *ME* not be awesome? |
| 00:16.23 | drmessano | I have my G729 encumbered boxes on Sealand.. Let them find me |
| 00:16.27 | jblack | Ok, So avoid 723 and 729. 711(u|a) Are ok, and 726-(various bit rates, I believe) should be ok. |
| 00:16.36 | Qwell | jblack: asterisk supports g726-32 |
| 00:16.52 | drmessano | g726-32 = g726? |
| 00:16.57 | drmessano | For an allow |
| 00:17.00 | Qwell | or g726-aal :p |
| 00:17.04 | Qwell | depends on the device... |
| 00:17.12 | drmessano | For asterisk |
| 00:17.14 | Qwell | it's all b0rked up |
| 00:17.26 | Qwell | yeah, allow=g726 or allow=g726-aal2 |
| 00:17.27 | drmessano | Whats the allow for g726..? |
| 00:17.29 | drmessano | Ah |
| 00:17.33 | drmessano | ty |
| 00:17.33 | Qwell | it depends on the device... |
| 00:17.39 | jblack | Things are working fine as is. I set the default to g711u |
| 00:17.41 | drmessano | I see what youre saying |
| 00:17.52 | Qwell | AAL2 or RFC3551 codeword packing |
| 00:18.08 | Qwell | some devices lie (are wrong) about what they actually support |
| 00:18.26 | Qwell | I think the SPA/PAP2 was one of the ones that did that wrong |
| 00:18.33 | drmessano | Hmm |
| 00:18.37 | jblack | I was hoping one of these magic numbers was gsm, since I'm already reducing to that on voip calls. |
| 00:18.40 | drmessano | So it would be -aal2? |
| 00:18.53 | Qwell | drmessano: I don't recall which way they were broken |
| 00:19.12 | drmessano | Ive not seen a linksys with GSM |
| 00:19.23 | drmessano | Unless I missed something on the SPA-3102 |
| 00:19.46 | Qwell | it's not all that common for hardware devices to do gsm |
| 00:19.56 | Qwell | they're already using g729, so...what's the point, ya know? |
| 00:20.03 | drmessano | GSM is a great codec, if you're only calling people using cordless phones in public toilets |
| 00:20.21 | jblack | How did you know? |
| 00:20.32 | drmessano | Im figuring you out, jblack ;) |
| 00:20.38 | Qwell | there are better codecs than gsm, both quality and size wise.. g729 is an example, but it's not free, which is why software uses gsm a lot more |
| 00:20.50 | drmessano | Ive noticed that |
| 00:21.11 | drmessano | Hardware seems to support G729 a lot, and software has GSM in lieu og G729 |
| 00:21.18 | drmessano | of* |
| 00:22.32 | jblack | As I understand things (yeah, limited understanding), the unencumbered options are basically either gsm or pcmu, which can be generalized as "bandwidth scrooge" and "I don't like to waste bandwidth, but my calls should be clear", respectively. |
| 00:23.01 | ManxPower | g.726 is becoming popular in both software phones and hardware |
| 00:23.01 | jblack | Though I guess there's LPC, which could be retermed as "I don't give a shit what you say" |
| 00:23.04 | puck | I just went up 4 floors to restart the PAP2, now links can see the page.... |
| 00:23.13 | drmessano | GSM - "Let me call you on the radio, it sounds better" |
| 00:23.44 | jblack | ManxPower: But isn't g726 encumbered? |
| 00:24.59 | *** join/#asterisk Winkie (n=urmom@general-ld-220.t-mobile.co.uk) |
| 00:28.49 | drmessano | My best friend, who is not as literate with IT as I am, has been dablling with VoIP basically "through" me, tells me today the radio consoles hes using at work can interface with SIP devices |
| 00:29.02 | drmessano | He says "Man, we can talk to this think with Asterisk!! COOL!" |
| 00:29.06 | drmessano | So that should be fun |
| 00:29.28 | Qwell | drmessano: see app_rpt |
| 00:29.37 | Qwell | radio hardware stuff |
| 00:29.46 | drmessano | Im familiar.. app_rpt is sweet |
| 00:30.41 | jblack | Has anyone seen [TK] lately? |
| 00:30.52 | jblack | Oh, he's here. |
| 00:30.54 | jblack | Cool |
| 00:31.40 | drmessano | Im pretty sure these consoles can act as a softphone on one hand, and also accept incoming calls to patch to a radio channel on the other side |
| 00:33.01 | *** join/#asterisk ZX81_ (n=ZX81@202.20.97.211) |
| 00:33.16 | drmessano | Could set up app_rpt on a box for radio <> asterisk, then an asterisk trunk to the console, then back to radio there |
| 00:34.12 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 00:34.53 | drmessano | I'd love to see the hams implement an asterisk based solution for the so called "Ham VoIP" they've been using |
| 00:35.21 | drmessano | using app_rpt |
| 00:35.50 | drmessano | But all these proprietary apps seems to have a foothold |
| 00:36.05 | Qwell | drmessano: Jim Dixon has been doing just that.. |
| 00:36.17 | Qwell | I think that's exactly why he wrote it |
| 00:36.44 | drmessano | yeah, I know it's being used.. but in horribly small numbers |
| 00:37.25 | drmessano | Whcih sucks, because its a WAY better solution |
| 00:40.13 | drmessano | I've toyed with designing a lower end interface card for asterisk myself and open sourcing the design |
| 00:40.42 | drmessano | For Radio, that is |
| 00:41.29 | drmessano | The Quad Radio PCI card is nice, just a tad bit expensive.. I saw one in action in Atlanta, and it does all it claims |
| 00:43.08 | Qwell | http://app-rpt.qrvc.com/usbsoundfob.html |
| 00:44.08 | drmessano | I was just reading that |
| 00:44.09 | drmessano | lol |
| 00:44.48 | drmessano | Any idea how common a CM-108 USB device is? |
| 00:45.02 | Qwell | no idea |
| 00:45.17 | drmessano | It could be as common as an Realtek NIC for all I know |
| 00:45.21 | drmessano | Just never heard of it |
| 00:47.10 | JT | drmessano: you can also use a TDM400P or a channel bank, and ARIBs |
| 00:49.15 | J4k3 | quad radio controller for voip patches? |
| 00:49.23 | J4k3 | how do you keep the radios from crosstalking like crazy? :P |
| 00:50.02 | drmessano | lol |
| 00:50.03 | J4k3 | afaik the only places you might effectively run autopatch is 2m (sketchy), 70cm and 1.2ghz. |
| 00:50.21 | drmessano | Why is 2m sketchy? |
| 00:50.32 | J4k3 | dtmf over 2m tends to piss people off? :) |
| 00:50.38 | drmessano | Naah |
| 00:50.40 | JT | easy, duplexors |
| 00:50.50 | drmessano | 2m is the most common place for autopatches |
| 00:50.59 | drmessano | The ones that exist anyway |
| 00:51.18 | drmessano | Really? |
| 00:51.31 | JT | commercial repeater sites have tonnes of different requencies on the same tower |
| 00:51.32 | drmessano | We had one on every repeater in town years ago |
| 00:51.37 | JT | there are heaps of autopatches on 2m |
| 00:52.13 | J4k3 | well, I think it my listening for autopatches on 2m may be limited. nobody cool hangs out on 2m :) |
| 00:52.21 | drmessano | Autpatches are also dead |
| 00:52.24 | drmessano | Yeah |
| 00:52.28 | drmessano | 2m is the CB band now |
| 00:52.30 | J4k3 | yep |
| 00:52.41 | drmessano | I put up a 440 machine and havent looked back |
| 00:53.03 | J4k3 | 2m is the only thing that effectively works here, but theres not enough hams around to bother. |
| 00:53.07 | *** join/#asterisk Netgeeks-laptop (n=chris@204.11.231.198.static.etheric.net) |
| 00:53.09 | mosty | i have a problem with accepting an iax call, asterisk says rejected <ip addres> who was trying to call <number>@ , but the context is empty |
| 00:53.11 | J4k3 | here being rural east texas (pine tree hell) |
| 00:53.17 | drmessano | lol |
| 00:53.20 | JT | drmessano: there is certainly no reason why you need to buy the quad radio interface card |
| 00:53.26 | mosty | but the iax account has a context set |
| 00:53.32 | JT | curious why you thought that was the only option |
| 00:54.07 | drmessano | Well, the way it was presented to me some time back was that there was either that card or purely non-prefab do it yourself |
| 00:54.16 | J4k3 | back in the early 90s in houston there was repeaters that seemed to only exist for autopatch.. all 70cm (but I never heard much on 2M in houston... either interference issues or my scanner was a pieceofcrap) |
| 00:54.21 | drmessano | Not even "Build from this", but "grab the iron and some parts and go to it" |
| 00:54.26 | J4k3 | well |
| 00:54.57 | J4k3 | it appears the only things you need is a GPIO-ish signal line, and an decent sound device. |
| 00:55.03 | *** join/#asterisk DrRighteous (n=DrRighte@ool-44c7ad06.dyn.optonline.net) |
| 00:55.11 | JT | drmessano: on the apt_rpt page, it shows the ARIB as well as the quad radio interface board |
| 00:55.21 | J4k3 | shouldn't require any more advanced gear than the average APRS site (just split tx/rx) |
| 00:55.40 | drmessano | I see that.. I got the impression the ARIB was non existant |
| 00:55.45 | DrRighteous | Anyone know the reboot button combo on a Cisco 7970G |
| 00:56.03 | JT | drmessano: weird, i have a pile or ARIB PCBs at home |
| 00:56.30 | drmessano | Well |
| 00:56.42 | drmessano | Not "non existant", but abandoned for the newer card |
| 00:57.13 | JT | the newer card is certainly more efficient |
| 00:57.20 | JT | especially if you have multiple repeaters |
| 00:57.56 | jblack | Aww. Callwithus doesn't seem to honor Set(Callerid |
| 00:59.20 | jblack | hmm. according to their faq, user error. |
| 00:59.29 | drmessano | I think I also assumed from where I click the order page for the blank PCB, it only lists the Quad card |
| 01:00.07 | lmadsen | a lot of ITSP's don't allow you to set your own CID |
| 01:01.06 | jblack | I've been lucky then. Teliax does. Callwithus claims to too, provided one can pass basic reading tests like "is able to read the faq". :) |
| 01:01.44 | JT | drmessano: i don't think you can build your own quad card |
| 01:01.58 | drmessano | Right |
| 01:02.06 | JT | in most countries you cannot set your callerid to a number you don't own |
| 01:02.17 | drmessano | but I click the link for ordering ARIB PCBs |
| 01:02.23 | drmessano | and it takes me to the order page for the Quad card |
| 01:02.30 | drmessano | No mention of the ARIB |
| 01:02.38 | drmessano | Hence my thought of it being extinct |
| 01:02.56 | jblack | These are numbers that route to me. The number that needs to be set depends upon what number I'm calling, though. |
| 01:03.29 | drmessano | You go to http://app-rpt.qrvc.com/ and they push the Quad card or the USB device hack |
| 01:04.05 | jblack | I know better than to misbehave with callerid, as tempting as it is. I'm 35, not 15. ;) |
| 01:04.08 | drmessano | USB audio kinda scares me |
| 01:04.09 | drmessano | lol |
| 01:04.31 | drmessano | "Polece"<911> |
| 01:05.02 | *** join/#asterisk Igbothom_3rd (n=Hilton@office.quarkgroup.com.au) |
| 01:05.02 | *** part/#asterisk DrRighteous (n=DrRighte@ool-44c7ad06.dyn.optonline.net) |
| 01:05.09 | JT | drmessano: maybe they've stopped selling the ARIB |
| 01:05.12 | JT | which would suck |
| 01:05.21 | JT | yeah usb radio, evil |
| 01:05.30 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 01:05.31 | jblack | I own a kid, a house, a car, and computer. The last thing I want is people in black suits showing up on my doorstep deciding that they're now theirs. |
| 01:05.47 | jblack | well, many computers. |
| 01:07.10 | drmessano | JT: Thats what it looks like |
| 01:07.50 | drmessano | surely a PCI sound card with a game port would work |
| 01:08.01 | drmessano | if the pieces existed lol |
| 01:08.14 | *** join/#asterisk kiscokid (n=ron@38.104.140.82) |
| 01:08.31 | drmessano | Well, no |
| 01:08.37 | drmessano | Game port wouldnt work |
| 01:13.33 | kiscokid | anyone built and used RxFax lately? |
| 01:17.13 | lmadsen | never used it |
| 01:17.18 | lmadsen | faxing is ol' sk00l :) |
| 01:19.17 | J4k3 | fax is for suckers |
| 01:19.24 | J4k3 | if you can't email it, I don't wanna see it |
| 01:19.48 | kiscokid | yeah, tell tht to the powers that be |
| 01:19.55 | kiscokid | *that |
| 01:20.21 | J4k3 | I have |
| 01:20.35 | J4k3 | I've been in business, >10M income in the last 15 years, and never owned a fax machine |
| 01:21.04 | kiscokid | they want to see signatures for some reason |
| 01:21.21 | puck | Faxed signatures are legally binding in most countries... |
| 01:21.23 | J4k3 | thats why god made fedex |
| 01:21.39 | kiscokid | fedex takes too long |
| 01:21.54 | J4k3 | they aren't effective. its too easy to get out the scotch tape and move a signature from one piece of paper to another. |
| 01:22.11 | J4k3 | its been shown to work just fine in court. you show a jury doubt, you're out. |
| 01:22.35 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkgroup.com.au) |
| 01:23.02 | *** join/#asterisk RoyK (n=roy@91.149.26.158) |
| 01:23.08 | kiscokid | I don't get paid to tell them how to run their business, just get paid to implement stuff they want |
| 01:26.48 | *** join/#asterisk Dovid (n=Dovid@bzq-79-179-118-177.red.bezeqint.net) |
| 01:26.50 | Dovid | hi |
| 01:26.55 | Dovid | i am trying to use exten => 1212,2,Set(DB(fwd/${CALLERID(num)} = ${CALLERID(num)}) |
| 01:27.11 | Dovid | how ever when getting the value out of asterisk it seems to have a space before the value |
| 01:27.18 | Dovid | so if the callerid is 304 |
| 01:27.32 | Dovid | the value ends up being " 304" |
| 01:27.41 | Dovid | with a preceding space. anyone ever see it ? |
| 01:37.22 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 01:37.28 | mosty | dovid: don't put spaces around the = |
| 01:38.26 | jblack | lmadsen: btw, for what it's worth, perhaps sometimes it would be good if more providers allowed setting callerid. Back when I used skype, they had set my callerid to "012345". Used to drive people crazy. :) |
| 01:38.46 | jblack | I couldn't do a thing to fix it. |
| 01:44.28 | *** part/#asterisk RoyK (n=roy@91.149.26.158) |
| 01:46.21 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-32977ffcf2d91f38) |
| 01:55.34 | *** join/#asterisk mog (n=mog@c-68-62-172-83.hsd1.al.comcast.net) |
| 01:55.34 | *** mode/#asterisk [+o mog] by ChanServ |
| 01:58.13 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 02:07.13 | *** join/#asterisk nybble (n=nybble@about/apple/performa/nybble) |
| 02:07.47 | *** join/#asterisk macros73 (n=cs@c-24-3-246-27.hsd1.mn.comcast.net) |
| 02:11.32 | kiscokid | I saw a discussion of faxing with Asterisk and they mentioned some hardware from Digium that related to faxing. Anyone know what that is? |
| 02:11.47 | *** join/#asterisk etfonhomey (n=chatzill@74-143-196-250.static.insightbb.com) |
| 02:12.09 | J4k3 | a modem? |
| 02:12.09 | J4k3 | hah |
| 02:14.01 | kiscokid | Is there any builtin fax detection in * when it answers a call? |
| 02:17.32 | [TK]D-Fender | kiscokid, Yes, go read up on *'s Standard Extensions |
| 02:17.54 | kiscokid | ok, thanks |
| 02:18.52 | *** join/#asterisk fuzzbawl (n=fuzzbawl@c-98-206-92-172.hsd1.in.comcast.net) |
| 02:19.07 | *** part/#asterisk fuzzbawl (n=fuzzbawl@c-98-206-92-172.hsd1.in.comcast.net) |
| 02:20.35 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 02:20.35 | *** mode/#asterisk [+o anthm] by ChanServ |
| 02:22.14 | *** join/#asterisk dacs (n=haiger@unaffiliated/dacs) |
| 02:22.36 | lmadsen | jblack: I built an ITSP, and I don't necessarily think implicitly allowing the client to set their CID is a good idea |
| 02:23.01 | lmadsen | kiscokid: hint -- it starts with 'f' |
| 02:24.10 | kiscokid | yeah, I found it |
| 02:24.18 | kiscokid | thanks |
| 02:25.00 | *** join/#asterisk tobias (n=tobias@user-0ce2hr3.cable.mindspring.com) |
| 02:25.13 | dacs | hipeople lmadsen |
| 02:26.00 | dacs | huh |
| 02:30.42 | dacs | when i set vebose to 7 "asterisk -rvvvvvvv" and i do sip set debug. by the time i scroll up to copy , it will already start rolling the new info |
| 02:31.23 | [TK]D-Fender | dacs, when you want to grab, hit EXIT |
| 02:31.32 | [TK]D-Fender | dacs, grab what you need, then reconnect |
| 02:32.42 | dacs | [TK]D-Fender: you mean do sip set debug, then when something i am intrested in comes up, i type exit ? |
| 02:32.54 | [TK]D-Fender | dacs, yes |
| 02:33.09 | [TK]D-Fender | dacs, make you test. once it fails... well STOP |
| 02:33.43 | [TK]D-Fender | dacs, or set some sort of hot-key to copy the entire scroll-back buffer live |
| 02:35.03 | dacs | [TK]D-Fender: i don't know how to do that, i am using putty |
| 02:35.22 | [TK]D-Fender | dacs, then method 1 it is. |
| 02:35.33 | dacs | [TK]D-Fender: yep |
| 02:36.35 | dacs | [TK]D-Fender: http://pastebin.ca/850253 |
| 02:38.47 | Olobola | are there any advantages of using perl over PHP for dial plans? I've read conflicting reports. This will be written for a potentially high volume system. |
| 02:39.23 | [TK]D-Fender | Olobola, AGI period is a load regardless of the language that you use |
| 02:39.47 | [TK]D-Fender | Olobola, AGI should be restricted to the minimum of what cannot be done in * standard logic |
| 02:40.18 | [TK]D-Fender | Olobola, And "high volume" is a very relative concept |
| 02:40.33 | JT | if you've got high volume |
| 02:40.36 | JT | use FastAGI |
| 02:40.45 | Olobola | ok. I need to connect to DB is the only issue, assuming this can't be done directly through a dialplan at this point |
| 02:40.51 | JT | in fact, if you use AGI, you should use FastAGI |
| 02:41.05 | JT | using plain AGI is illogical from a resources point of view |
| 02:41.08 | JT | Olobola: it can. |
| 02:41.46 | [TK]D-Fender | Olobola, * has access to ODBC and MySQL rather natively |
| 02:42.50 | Olobola | ok, thank you. It's just nice to be able to whip a little php script, easy. |
| 02:42.53 | Olobola | eventually it will need to be much more robust. |
| 02:43.08 | *** join/#asterisk ManxPower (n=manxpowe@wsip-68-228-11-14.br.br.cox.net) |
| 02:43.21 | JT | php sounds erroneous to use for telephony, but that's just me |
| 02:44.09 | J4k3 | pee ache pee |
| 02:44.22 | J4k3 | if you experience it, you might need to go to the clinic |
| 02:44.45 | jblack | It kind of makes sense to me. There's a multitude of neat integration between * and websites just begging to be taken advantage of. |
| 02:44.59 | *** part/#asterisk kiscokid (n=ron@38.104.140.82) |
| 02:45.02 | JT | php shouldn't exist, even for web sites ;) |
| 02:45.39 | J4k3 | it burns! it burns! |
| 02:45.47 | jblack | lol |
| 02:49.23 | *** join/#asterisk truz_`24 (n=truz_`24@74-138-37-35.dhcp.insightbb.com) |
| 02:49.40 | drmessano | php is sweet |
| 02:49.46 | drmessano | Yo mock the PHP? |
| 02:49.52 | drmessano | You* |
| 02:50.07 | J4k3 | php is simple |
| 02:50.08 | J4k3 | and kinda fun |
| 02:50.14 | J4k3 | but its easy to mock. |
| 02:50.18 | drmessano | lol |
| 02:50.30 | Braxus | PHP: training wheels without the bike. |
| 02:50.32 | drmessano | No, LOLCAT is easy to mock |
| 02:50.44 | J4k3 | 2girls1cup/ |
| 02:50.44 | J4k3 | ? |
| 02:51.04 | drmessano | tubgirl? wut? |
| 02:51.23 | J4k3 | the other day I had the occasion to point at a goat and say "goat, see?" |
| 02:51.38 | J4k3 | I felt pretty wrong about doing it, despite its complete innocence. |
| 02:52.17 | drmessano | ROFLLLL |
| 02:52.24 | drmessano | COmplete geek moment, I love it |
| 02:52.56 | *** join/#asterisk WilliamK (n=noc@static-71-170-144-3.dllstx.fios.verizon.net) |
| 02:53.21 | [TK]D-Fender | J4k3 + complete innocence.... lemme think on that :\ |
| 02:54.13 | *** join/#asterisk asr33 (n=asr33@dsl-207-112-82-152.tor.primus.ca) |
| 02:54.19 | drmessano | That would be like [TK]D-Fender pwning a total newb by accident |
| 02:54.26 | drmessano | Oh wait, Hi [TK]D-Fender |
| 02:54.28 | drmessano | :) |
| 02:55.07 | [TK]D-Fender | drmessano, No... I'm rather deliberate in that sort of thing :) |
| 02:55.39 | drmessano | My point exactly ;) |
| 02:55.43 | J4k3 | but I *am* going to your momma's house! |
| 02:57.42 | drmessano | If you cut a grandstream in half, does it become an F> or an <S device? |
| 02:58.02 | J4k3 | prolly a P( or an )S |
| 02:58.08 | drmessano | LOL |
| 02:58.29 | drmessano | PV and a VN |
| 03:00.24 | drmessano | Another reason I love Twitter: |
| 03:00.30 | drmessano | "Configuring X-Lite soft phone in Vista to work with Trixbox VOIP system at corporate" |
| 03:00.45 | drmessano | Sca.......ry........not sure which part |
| 03:01.49 | *** join/#asterisk iamthelostboy (n=nathan@24.244.144.130) |
| 03:02.43 | J4k3 | I need a real machine so I can actually use a softphone |
| 03:02.51 | J4k3 | this thing chunks no matter what |
| 03:04.53 | iamthelostboy | is there anyway, other than adding an external server, i can have a client behind NAT connecting to a server behind NAT? |
| 03:05.05 | iamthelostboy | i've been reading a little about it, and im slightly confused |
| 03:07.00 | [TK]D-Fender | iamthelostboy, No need for any external server, read up : |
| 03:07.00 | drmessano | I do it all the time |
| 03:07.01 | [TK]D-Fender | ~sipnat |
| 03:07.02 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 03:07.10 | drmessano | lol |
| 03:09.18 | *** part/#asterisk asr33 (n=asr33@dsl-207-112-82-152.tor.primus.ca) |
| 03:13.11 | jblack | Is there a way to get debug info for iax2? |
| 03:13.58 | [TK]D-Fender | jblack, "iax2 debug" |
| 03:14.29 | *** join/#asterisk kusznir (n=kusznir@isg-grad-02.eecs.wsu.edu) |
| 03:15.13 | WilliamK | zaptel broken on the latest svn update |
| 03:15.24 | kusznir | Hi all: I have a "strange" application for asterisk, and was looking for some advice for potential configuration of the CDR system. |
| 03:15.29 | iamthelostboy | thanks... |
| 03:15.42 | drmessano | Theres no such thing as a strange Asterisk application |
| 03:15.54 | iamthelostboy | thats pretty much the document i was looking for |
| 03:16.25 | drmessano | chan_plant is as weird as it gets |
| 03:16.28 | kusznir | I'm using asterisk to create a 2-phone system for use in a psycology/computer science experiment where we will have peopole perform "daily living activities" including useing the phone. The space will be instrumented so that the computer can try and follow their progress in completeing these activities. |
| 03:16.59 | jblack | no iax2 debug here |
| 03:17.22 | kusznir | To this end, I need a way to get when a call is started, ended, and what number it dialed into our "system". Ideally, asterisk will call a program (shell script/perl script) with the values, and that script will do the magic to get it into our main system. |
| 03:17.53 | kusznir | We can do a post-process .csv, but if asterisk has the ability to run the script "real-time" as "events" (phone calls) happen, that could be easier and would be preferred. |
| 03:19.11 | kusznir | To the best of my knowledge, asterisk can write to a .csv file (or other file format), or to an SQL db. Can it call a script with some CDR data as command line options (or env vars)? |
| 03:21.41 | [TK]D-Fender | kusznir : just about anything you'll think of, yes |
| 03:22.02 | *** join/#asterisk khronos (n=khronos@c-66-229-159-175.hsd1.fl.comcast.net) |
| 03:22.15 | kusznir | [TK]D-Fender: where do I find info on how to configure it to do so? |
| 03:22.47 | [TK]D-Fender | kusznir, you need to read up on how * works, and all of its dialplan applications. Then read up on AGI for the complex bits. |
| 03:22.52 | [TK]D-Fender | kusznir, for now : |
| 03:22.53 | [TK]D-Fender | ~book |
| 03:22.55 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 03:22.56 | [TK]D-Fender | ^^^^^^^^^^^^ |
| 03:23.20 | [TK]D-Fender | kusznir, its all in there. The application reference yuo can skip to right away for a tase of how you can do things |
| 03:24.26 | kusznir | Ahh...I've read the book. I've actually been running * for a few years now, just on very small scale, and always where I didn't care about accounting. The last time I looked at the book, I found it didn't seem to have much of the technical details on configuring *...more of a "this is what VoIP is and this is how to get a very basic VoIP system working"; for complex/advanced things, you were left on your own. |
| 03:24.37 | kusznir | Did they release a new version now? |
| 03:24.56 | [TK]D-Fender | kusznir, yes |
| 03:25.32 | kusznir | And for what I'm looking for, I don't use the CDR modules at all; just pure AstCGI? |
| 03:25.38 | lmadsen | kusznir: there is a 2nd edition now |
| 03:25.48 | [TK]D-Fender | kusznir, well you're not going to find a "here's how to make a Phone-based Casino". Some thing you have to just imagine yourself and realized the pieces that will help you do it |
| 03:26.05 | WilliamK | kinda interesting |
| 03:26.24 | [TK]D-Fender | kusznir, I don't see where CDR is integral to your plan yet.. |
| 03:26.35 | [TK]D-Fender | kusznir, why is that the point of interest? |
| 03:26.36 | mmlj4 | grrr... i can't get incoming calls with FWD... it tells me that the extension doesn't exist, and I keep telling it that it does too exist |
| 03:26.51 | [TK]D-Fender | mmlj4, and you do this and show us NOTHING :) |
| 03:27.16 | mmlj4 | hey, copy-n-paste from the wiki, all I did |
| 03:27.17 | [TK]D-Fender | mmlj4, "Your failure is complete" - The Emperor |
| 03:27.28 | jblack | To have one's very existance challenged by a computer... Ohhh, the humanity |
| 03:27.34 | [TK]D-Fender | mmlj4, Oh yeah, and the WIKI's never wrong. |
| 03:27.38 | [TK]D-Fender | ... |
| 03:27.41 | [TK]D-Fender | </sarcasm> |
| 03:27.50 | jblack | Even in the matrix, they let us exist as batteries! |
| 03:27.51 | mmlj4 | [TK]D-Fender: that's what I was thinking, but you never know... |
| 03:27.59 | [TK]D-Fender | mmlj4, So that aside, pastebin the whole mess including SIP debug of your failure and we'll take a look |
| 03:28.00 | [TK]D-Fender | ~pb |
| 03:28.00 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 03:28.01 | [TK]D-Fender | ^^^^^^^^^^^^ |
| 03:28.16 | mmlj4 | if i can't figure it out, sure |
| 03:28.44 | [TK]D-Fender | mmlj4, Sorry we're here to help. If all you want is sympathy try #barbrawalters , or #oprah |
| 03:28.47 | kusznir | [TK]D-Fender: We're working on artifical intellegence algoritms to detect a person's daily living activities, and ultimately help elderly/brain dammaged/etc people live independantly longer. By detecting ones habbits and "normal behaviors", we can detect if something has happened that might need assistance. |
| 03:28.48 | mmlj4 | i'm just using you guys to gripe to |
| 03:29.03 | mmlj4 | ;-) |
| 03:29.55 | [TK]D-Fender | kusznir, and I still have no idea of your focus on CDR at the END of a call and have no more idea of what * is doing at all |
| 03:30.00 | kusznir | Antoher application is detecting someone "freezing" in a task and providing a cue or prompt or something. Psycology research has shown that one's phone usage is actually a very useful indicator. So for this experiemnt, we're trying to detect "successful phone usage" and feed that to the AI system as one of many (30-50) inputs. |
| 03:30.56 | kusznir | In a most basic situation, we can use a traditional, phone-company POTS line and a "line use detector" and get the bare bones info. However, 1) we don't have a POTS line at the installation, and 2) we'd rather be more forward-looking and get "more details". |
| 03:31.02 | [TK]D-Fender | kusznir, again you give us no clue of *'s involvement in any of this... |
| 03:32.30 | kusznir | So, we're using Asterisk to provide a phone. It will be pre-programmed with a few specific phone numbers. The user will be told to look up a specific person from the phone book and call them. Asterisk will route this "real" phone number either to another person involved with the experiment, or a message. If the person fails to dial the correct number, we need to know. |
| 03:32.53 | kusznir | This shows wether the person is successfully able to 1) use a phone book; 2) use a phone; 3) communicate with others not physically present in the same room. |
| 03:33.07 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 03:33.07 | *** mode/#asterisk [+o russellb] by ChanServ |
| 03:33.29 | [TK]D-Fender | kusznir, basic CDR can tell you that easily enough. or you can make a catch-all extension and parse the entire event live. |
| 03:34.47 | jblack | would someone mind dialing iax:jblack@mercury.linuxguru.net and seeing if they get a voice prompt? Someone I'm doing dundi with is swearing my machine doesn't exist. :) |
| 03:35.09 | kusznir | [TK]D-Fender: I origionally thought using CDRs would be fine, and in a "worst-case", we'd import the CDR.csv after the experiment into our main event database so we can run our algorithms that detect these activities and there sucess level. Ideally, though, these events will be inserted into our main database more or less as they happen. |
| 03:35.44 | kusznir | So if I can get a basic CDR that instead of being written to a file, calls a system program with the same data that would have been written out, that would be ideal. |
| 03:36.52 | kusznir | While we are currently post-processing the experiment, we are working toward real-time processing, and at that point, we'll have to have the events inserted in real-time. All we really need to know is when a call was placed, how long it was, and ideally, what number was dialed. |
| 03:37.16 | iamthelostboy | in regards to the * -> Nat -> Nat -> Client.. is forwarding a lot of ports like is recommended going to disrupt the connection, if it is shared with a bunch of computers browsing etc? |
| 03:39.31 | _ShrikE | kusznir: why dont you log your cdr directly to the database? |
| 03:40.25 | *** join/#asterisk etfonhomey (n=chatzill@74-143-196-250.static.insightbb.com) |
| 03:41.16 | kusznir | 1) our eventual archetecure does not have the database as the center (real-time users of the data will subscribe to a messaging system to get the data); 2) the database format is quite fixed and specialized with multiple table references, and I didn't think asterisk could adapt into our existing schema (remember, we have ~50 different data sources, only one of which is asterisk) |
| 03:42.43 | _ShrikE | ahh |
| 03:43.02 | [TK]D-Fender | iamthelostboy, Follow the guide.... |
| 03:46.31 | *** join/#asterisk marcan (i=1337@host215-248.cvf.fit.edu) |
| 03:53.26 | *** join/#asterisk phigan (n=phigan@ip68-109-166-1.ph.ph.cox.net) |
| 03:55.50 | etfonhomey | [TK]D-Fender, how's it going? |
| 03:56.11 | [TK]D-Fender | etfonhomey, getting by, just jamming away right now |
| 03:58.59 | etfonhomey | [TK]D-Fender, reading about setting up DNS/NAT so remote users can get my * box without having to do it via an IPSec VPN tunnel. |
| 03:59.38 | ManxPower | etfonhomey: Most of us can do that in our sleep, assuming your router is not a POS |
| 03:59.42 | [TK]D-Fender | etfonhomey, You know the link.... |
| 04:00.10 | etfonhomey | [TK]D-Fender, you talking about ~sipnat? |
| 04:00.17 | [TK]D-Fender | etfonhomey, yup |
| 04:01.16 | etfonhomey | [TK]D-Fender, got that. Wanting to read about SRV records. |
| 04:03.44 | ManxPower | etfonhomey: no real need for SRV records unless you want "failover" or roaming between a network that is local to Asterisk and one that is outside the NAT |
| 04:04.05 | etfonhomey | [TK]D-Fender, completely unrelated to this channel. Know anything about cell phone signal boosters? |
| 04:04.19 | [TK]D-Fender | etfonhomey, mostly scams |
| 04:04.57 | etfonhomey | ManxPower, just need static NAT and my remote users need to know the external IP, correct? Static NAT for both SIP and the media? |
| 04:05.52 | ManxPower | etfonhomey: External IP or hostname |
| 04:06.14 | ManxPower | specifically port forwarding. Don't know what "static nat" is. |
| 04:07.29 | etfonhomey | ManxPower, gotcha. |
| 04:08.01 | etfonhomey | [TK]D-Fender, what about this stuff: http://www.wi-ex.com/ |
| 04:08.23 | [TK]D-Fender | etfonhomey, no clue |
| 04:08.33 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 04:08.34 | etfonhomey | [TK]D-Fender, the reason I ask is that the president of our company lives in a rural area that doesn't get very good cell phone coverage. |
| 04:09.10 | [TK]D-Fender | etfonhomey, don't forget to change the time for sunrie & sunset so that it fits his tastes while you're at it.... |
| 04:09.20 | etfonhomey | [TK]D-Fender, He wanted to me to see of there were viable options for boosting the signal. |
| 04:10.03 | etfonhomey | [TK]D-Fender, Hey, I don't mind these things, pay me a good salary and benefits, plus let me come and go to do consulting during the day. |
| 04:10.53 | iamthelostboy | etfonhomey: we had a gsm repeater put into our warehouse |
| 04:11.30 | etfonhomey | iamthelostboy, do you have a good signal outside of the warehouse? |
| 04:11.57 | iamthelostboy | we had terrible reception.. in our case it was too many strong signals.. we talked our provider into putting in a repeater, which turns out to be stronger than the rest of the singals, so our phones use that instead |
| 04:13.10 | etfonhomey | iamthelostboy, I have the case were there is very little signal and need a way to amplify it. |
| 04:13.21 | iamthelostboy | though it outputs stronger signals than a cellphone, so can connect to cell sites further away, if there is a weak signal |
| 04:14.16 | iamthelostboy | without really going searching... http://www.powertec.com.au/repeater.php |
| 04:14.48 | iamthelostboy | i would look at the brand of ours for you, though I'm not near it at the moment |
| 04:14.57 | etfonhomey | [TK]D-Fender, how do you do call forwarding with your Polycoms? Do you just do it at the phone? |
| 04:15.10 | [TK]D-Fender | etfonhomey, usuall |
| 04:15.30 | [TK]D-Fender | etfonhomey, unless I care about being able to remotely maintain it |
| 04:16.36 | etfonhomey | [TK]D-Fender, in call forwarding at the phone, does the media stream actually travel to the phone and then get redirected? |
| 04:17.05 | [TK]D-Fender | etfonhomey, no, just the sip invit. which gets redirected. Media doesn't land |
| 04:18.47 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 04:19.02 | etfonhomey | [TK]D-Fender, I'm using an ITSP currently. So, if I call forward to my cell from one of my Polycoms, does the media stream of a call go like this: PSTN -> ITSP -> Internet -> * -> Internet -> ITSP -> PSTN |
| 04:19.29 | grandpapadot | HI all. Anyone use static realtime with queues.conf in the db? How are member => whatever treated? Each row found added? I ask because all other values are name = value, members are member => value |
| 04:20.02 | grandpapadot | etfonhomey: Yes. |
| 04:22.56 | ManxPower | etfonhomey: defactowireless.com has a bunch of RF stuff, including (somewhere on their site) cell repeaters |
| 04:23.03 | ManxPower | <PROTECTED> |
| 04:23.12 | etfonhomey | ManxPower, thanks. |
| 04:24.26 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 04:24.47 | etfonhomey | [TK]D-Fender and ManxPower, you guys use * realtime in production? |
| 04:25.21 | ManxPower | etfonhomey: the "new products" link on the site |
| 04:25.43 | ManxPower | etfonhomey: My asterisk servers have less than 80 phones each, no need for a database |
| 04:26.46 | ManxPower | cell booster/releaters, not cheap: http://shop.defactowireless.com/s.nl/sc.2/category.2120/.f |
| 04:28.25 | etfonhomey | ManxPower, Thanks. Have you messed with any cell phone repeaters before? |
| 04:44.11 | dacs | [TK]D-Fender: do you have time to help me |
| 04:48.22 | *** join/#asterisk pepo-- (n=pepOSX--@190.78.221.19) |
| 04:49.52 | dacs | can someone explain the syntax :exten => 400,1,Dial(sip/Phone1) please |
| 04:50.15 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 04:50.15 | *** mode/#asterisk [+o anthm] by ChanServ |
| 04:50.44 | *** part/#asterisk iamthelostboy (n=nathan@24.244.144.130) |
| 04:51.51 | etfonhomey | dacs, did you make up with [TK]D-Fender? |
| 04:52.16 | dacs | etfonhomey: what do you mean?lol |
| 04:52.17 | drmessano | OH |
| 04:52.19 | drmessano | Drama? |
| 04:52.50 | etfonhomey | dacs, seemed to me like you were steaming a week or so ago... |
| 04:53.23 | dacs | etfonhomey: oh yeah, we got on the wrong foot, but its cool now |
| 04:54.23 | dacs | now i have a softphone and a phone connected to my ATA, both register , but i can't call any off them |
| 04:54.53 | etfonhomey | dacs, What's your SIP debug look like? |
| 04:55.19 | dacs | no http://pastebin.ca/850253 |
| 04:56.30 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 04:59.54 | dacs | etfonhomey: take a look http://pastebin.ca/850412 |
| 05:00.25 | *** join/#asterisk RB2 (n=RB2@pool-71-251-47-64.nwrknj.east.verizon.net) |
| 05:02.06 | etfonhomey | What kind of ATA? |
| 05:02.32 | dacs | etfonhomey: Cisco ATA186 |
| 05:10.07 | etfonhomey | So, did you dial 500? |
| 05:10.31 | dacs | yes |
| 05:10.43 | dacs | nothing just dead air |
| 05:10.59 | etfonhomey | You have no Dial() application in your internal context for that extension. |
| 05:13.15 | etfonhomey | Shouldn't you have a Dial(SIP/phone1) if you want that phone to ring? |
| 05:14.58 | *** join/#asterisk lzhang (n=lzhang@66-90-152-164.dyn.grandenetworks.net) |
| 05:15.19 | lzhang | hello, I just installed asterisk and set up a couple sip phones on the same lan |
| 05:15.40 | lzhang | is it normal to be hearing a .25 second lag in audio when talking between the two phones? |
| 05:16.05 | dacs | etfonhomey: now its Dialing after i add the Dial application, now i have to figure for the other phone to ring |
| 05:16.21 | JT | lzhang: your ears measured the 0.25 seconds? |
| 05:16.33 | lzhang | JT: it's a guess |
| 05:16.47 | lzhang | a noticeable lag not longer than half a second |
| 05:16.57 | dacs | and why did you choose .25 sec |
| 05:16.57 | JT | i'd say it's way off unless there's something wrong with your lan |
| 05:17.03 | JT | it's probably way less |
| 05:17.09 | lzhang | should it be nearly instantaneous? |
| 05:17.24 | JT | sure but you will still hear an echo if you use echo test |
| 05:17.29 | *** join/#asterisk mast3rpyr0 (n=Mast3rpy@cpe-65-186-222-73.insight.res.rr.com) |
| 05:17.50 | lzhang | I used one sip phone to call the other |
| 05:18.08 | lzhang | obviously the echo test will be delayed significantly |
| 05:18.10 | *** join/#asterisk admin0 (n=admin@116.90.228.34) |
| 05:18.14 | mast3rpyr0 | umm hey could i get some quick help |
| 05:18.17 | JT | why is it obvious? |
| 05:18.19 | [TK]D-Fender | dacs, Your dialplan puts you in an echo test. if you talk you don't ehar it echo'd back? |
| 05:18.36 | dacs | [TK]D-Fender: no |
| 05:18.41 | lzhang | because the point is to hear the audio come back, so the pbx should send it delayed instead of immediately as you are speaking |
| 05:19.09 | mast3rpyr0 | what goes in the service provider part in the AsteriskNOW gui? |
| 05:19.43 | lzhang | is there a point to the nitpicking about my guesstimate numbers? all I'm looking for is an idea of what I should be expecting |
| 05:20.00 | JT | lzhang: absolute rubbish |
| 05:20.07 | JT | lzhang: asterisk sends packets back immediately |
| 05:20.10 | dacs | [TK]D-Fender: now when i call from my ATA to my x-lite, it will ring the x-lite and i can hear my voice in my PC speackes |
| 05:20.16 | JT | as soon as the audio comes in, it spits it out |
| 05:20.27 | JT | lzhang: the point is to test the latency of the network, etc |
| 05:20.28 | *** join/#asterisk sergey (n=sergey@91.189.233.71) |
| 05:20.33 | JT | not to make an artificial echo |
| 05:20.43 | JT | don't make assumptions |
| 05:21.01 | lzhang | is that so |
| 05:21.51 | lzhang | when I tried the echo test I also heard the lag |
| 05:22.10 | lzhang | so maybe this indicates some sort of network issue, or configuration problem |
| 05:22.46 | JT | no |
| 05:22.53 | JT | it indicates normal operation |
| 05:22.59 | JT | unless it was significant |
| 05:23.56 | dacs | etfonhomey: [TK]D-Fender : thank you guys, now i can call my ATA from my X-lite and vis versa |
| 05:24.08 | dacs | time to continue reading |
| 05:24.17 | lzhang | that's my problem, what is a significant amount of delay |
| 05:24.18 | etfonhomey | dacs, I'm glad I could help. |
| 05:24.35 | lzhang | is .25 seconds significant |
| 05:24.45 | etfonhomey | dacs, because usually I don't. I'm the one asking the questions. :) |
| 05:25.48 | JT | lzhang: i'm not sure, i have no idea what the intenal lag of ip phones are |
| 05:26.03 | JT | .25 seconds of network lag on a lan would be bad |
| 05:26.15 | lzhang | might be the crappy test phones I was using |
| 05:27.44 | JT | what phones? |
| 05:29.44 | lzhang | cor-something; I'm not at the office and I can't recall the actual name at the moment |
| 05:29.47 | *** join/#asterisk L|NUX (i=L_NUX@unaffiliated/lnux/x-10290) |
| 05:30.10 | JT | heh |
| 05:30.27 | mast3rpyr0 | does anyone know what to put in for service provider in the initial setup? |
| 05:30.29 | lzhang | I'll probably put the phones and server on the same switch instead of running through our physical network, make sure it's not any of the connections |
| 05:30.45 | lzhang | see if that changes things |
| 05:31.05 | lzhang | probably also test with xlite |
| 05:31.23 | mast3rpyr0 | why would you need a service provider when your the server? |
| 05:31.27 | dacs | etfonhomey: lol, are you new too |
| 05:31.41 | etfonhomey | thrice new |
| 05:32.03 | dacs | etfonhomey: welcome to the club , hahaha |
| 05:32.59 | mast3rpyr0 | soo.. is service provider important? |
| 05:33.23 | dacs | mast3rpyr0: if you want to make calls ...YES |
| 05:33.38 | lzhang | mast3rpyr0: if you want to make outbound calls to other numbers |
| 05:33.45 | mast3rpyr0 | i though asterisk was the server tho.. i want to be my service provider... |
| 05:34.46 | dacs | mast3rpyr0: nope , asterisk, refered to here as (*) is a PBX |
| 05:34.55 | mast3rpyr0 | wtf.. |
| 05:34.55 | dacs | mast3rpyr0: you know what is PBX |
| 05:35.02 | mast3rpyr0 | slighly |
| 05:35.23 | dacs | mast3rpyr0: no no no, not (wtf) it is (*) |
| 05:35.42 | mast3rpyr0 | what is the point of this software if it doesnt give me free calls |
| 05:36.08 | dacs | mast3rpyr0: if you slighly know what a PBX, do you expect to be a Service Provider? |
| 05:36.28 | mast3rpyr0 | ya im trying to set up voip for the ipodtouch |
| 05:36.48 | dacs | mast3rpyr0: it make sense to us crazy people here |
| 05:37.02 | JT | mast3rpyr0: if you want to connect to the PSTN, you need a VoIP provider, or real phone lines. |
| 05:37.59 | mast3rpyr0 | they should say that in the description.. i just wasted the last 5 hours of my life with this |
| 05:38.19 | JT | what the hell are you talking about? |
| 05:38.27 | JT | whinge whinge whinge |
| 05:38.35 | JT | perhaps you should learn to read |
| 05:38.48 | *** part/#asterisk lzhang (n=lzhang@66-90-152-164.dyn.grandenetworks.net) |
| 05:38.58 | JT | asterisk allows you to make free calls between people connected to your asterisk server |
| 05:39.09 | JT | it is a hybrid ip/tdm/etc PBX |
| 05:39.10 | *** join/#asterisk lzhang (n=lzhang@66-90-152-164.dyn.grandenetworks.net) |
| 05:39.23 | outtolunc | it is *magic* <G> |
| 05:39.24 | dacs | mast3rpyr0: i see where you coming from , is it because it say "OpenSource PBX"? |
| 05:39.38 | mast3rpyr0 | it was free so i tried it lol |
| 05:39.49 | mast3rpyr0 | its not sip? |
| 05:39.57 | JT | it supports SIP |
| 05:40.02 | mast3rpyr0 | but its not... |
| 05:40.03 | dacs | it support SIP |
| 05:40.05 | JT | but it is not exclusively SIP |
| 05:40.06 | JT | what? |
| 05:40.10 | JT | it supports SIP |
| 05:40.13 | JT | what more do you want |
| 05:40.16 | mast3rpyr0 | to be it |
| 05:40.23 | JT | SIP is a common voice over ip protocol |
| 05:40.23 | dacs | mast3rpyr0: what is SIP? |
| 05:40.31 | mast3rpyr0 | what he said |
| 05:40.34 | JT | i think you need to learn what SIP is |
| 05:40.38 | dacs | mast3rpyr0: what does is mean |
| 05:40.44 | JT | mast3rpyr0: you clearly have no idea what sip is |
| 05:40.59 | mast3rpyr0 | session initiats protocol |
| 05:41.09 | JT | initiats.. |
| 05:41.14 | mast3rpyr0 | you know what i mean |
| 05:41.49 | mast3rpyr0 | what does a site liek freecall.com use then? |
| 05:41.49 | JT | you clearly don't know what it means if you think a voip pbx can |
| 05:41.54 | ManxPower | understanding the words and understanding the design and concepts are two different things |
| 05:41.54 | JT | you clearly don't know what it means if you think a voip pbx can "be sip" |
| 05:42.04 | *** join/#asterisk AndyGraybeal (n=andy@node173.36.251.72.1dial.com) |
| 05:42.04 | JT | who knows |
| 05:42.05 | dacs | mast3rpyr0: well the same way you used google, you can read the whole thing |
| 05:42.13 | JT | ~thebook |
| 05:42.14 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 05:42.22 | mast3rpyr0 | i read that |
| 05:42.33 | Qwell | no you didn't |
| 05:42.35 | lzhang | I read it 5 times |
| 05:42.46 | mast3rpyr0 | why the hell would you do that.. |
| 05:42.46 | lzhang | memorized it too |
| 05:43.02 | outtolunc | maybe he has no life <G> |
| 05:43.04 | ManxPower | a PBX can hav nothing to do with SIP and SIP can have nothing to do with a PBX and SIP and a PBX can work togather. |
| 05:43.05 | JT | mast3rpyr0: you would understand if you read it |
| 05:43.26 | dacs | JT: err sorry |
| 05:43.49 | dacs | JT is it possable that you can name is ~TheOnlyBook |
| 05:43.54 | dacs | lol |
| 05:44.00 | mast3rpyr0 | ok fine forget this, any idea where i can get a server |
| 05:44.06 | lzhang | frys |
| 05:44.10 | JT | dacs: that doesn't make sense |
| 05:44.11 | ManxPower | mast3rpyr0: is english your native language? |
| 05:44.11 | dacs | mast3rpyr0: DELL.com |
| 05:44.19 | mast3rpyr0 | .. |
| 05:44.20 | JT | mast3rpyr0: ibm.com |
| 05:44.22 | dacs | lzhang: lol |
| 05:44.24 | mast3rpyr0 | a voip server |
| 05:44.27 | Qwell | ebay |
| 05:44.34 | JT | mast3rpyr0: ibm.com then add asterisk |
| 05:44.35 | mast3rpyr0 | that doesnt cost $400 liek this one http://www.easylivecd.com/english/voip/ |
| 05:44.37 | dacs | mast3rpyr0: cisco.com then |
| 05:44.46 | dacs | they are cheap too |
| 05:44.59 | lzhang | mast3rpyr0: imagine that you have a tv |
| 05:44.59 | mast3rpyr0 | im running it from home on the computer i have |
| 05:45.03 | mast3rpyr0 | ... |
| 05:45.12 | lzhang | just because you have a tv doesn't mean you can watch anything |
| 05:45.14 | ManxPower | mast3rpyr0: Asterisk is not really a PBX. It is a toolkit that allows you to build a PBX. |
| 05:45.17 | lzhang | you still need to buy cable service |
| 05:45.22 | mast3rpyr0 | i know more than you think i do.. |
| 05:45.24 | ManxPower | If you want a turnkey PBX, you should not be on this channel |
| 05:45.34 | JT | sure, then what's the problem mast3rpyr0 ? |
| 05:45.44 | ManxPower | There are turnkey PBXs that use Asterisk, however. |
| 05:45.44 | JT | asterisk can be a voip server |
| 05:45.49 | mast3rpyr0 | i want this http://www.easylivecd.com/english/voip/ free thats the problem |
| 05:46.17 | mast3rpyr0 | not exactly that but same concept |
| 05:46.35 | ManxPower | mast3rpyr0: you want trixbox, asterisk gui, asterisk now, freepbx or some other Asterisk GUI |
| 05:46.41 | JT | mast3rpyr0: so you're too lazy to spend time working on asterisk and OpenSER yourself |
| 05:46.53 | drmessano | lol |
| 05:46.55 | mast3rpyr0 | will * do it?> |
| 05:46.59 | JT | yes |
| 05:47.00 | ManxPower | We help people here learn asterisk, not learn guis or other products. |
| 05:47.00 | drmessano | LOL |
| 05:47.07 | drmessano | It says it USES ASTERISK |
| 05:47.23 | Qwell | drmessano: in like the second paragraph |
| 05:47.25 | lzhang | so is everyone in this chan usually so cranky |
| 05:47.27 | Qwell | ... |
| 05:47.28 | drmessano | "It is based on the Open Standard SIP Express Router (SER) and Asterisk." |
| 05:47.38 | Qwell | do you really think he got that far? ;/ |
| 05:47.42 | ManxPower | lzhang: it depends on what questions are asked. |
| 05:47.47 | drmessano | NO SPEEDA DE ENGLIS? |
| 05:47.51 | JT | lzhang: when they act really annoying and demandingly |
| 05:47.57 | ManxPower | But we tear people to pieces that don't want to learn |
| 05:48.00 | *** join/#asterisk jblack (n=jblack@pool-71-181-138-192.sctnpa.east.verizon.net) |
| 05:48.16 | lzhang | ManxPower: that's the thing, I don't feel like that's something to be proud of |
| 05:48.17 | mast3rpyr0 | if this will do what i want it to do ill learn it |
| 05:48.50 | JT | lzhang: but it is |
| 05:48.57 | drmessano | If someone wont HELP the people in here to HELP them, theyre NOT worth it |
| 05:49.03 | JT | lzhang: we deal with enough idiots in our day jobs, we don't need this |
| 05:49.07 | lzhang | lol |
| 05:49.09 | ManxPower | lzhang: I didn't say we are "proud of it", but asking for a turnkey Asterisk solution on this channel is like asking to buy a fully functional car at an auto parts store. |
| 05:49.10 | drmessano | "But I dont wanna read" "So why should WE HELP?" |
| 05:49.22 | dacs | jblack: wb |
| 05:49.25 | jblack | Thanks. |
| 05:49.31 | ManxPower | You'd get laughed out of the store or referred to a place that sells pre-assembled cars. |
| 05:49.33 | jblack | Here's my lesson of the day. |
| 05:50.11 | jblack | Let's say you're doing iax2 with a provider that you register with. Everything is _great_. Then somebody tries to call you on iax2, and he _swears_ you have a firewall. "Nah" you say, becuase the provider works FINE! |
| 05:50.13 | ManxPower | Heck, we spent close to 18 months working with asterisk before we put our first asterisk server into production |
| 05:50.23 | drmessano | Especially if you find some damn webpage, come in here and ask "R AKERISK LIKE THIS? HEP ME PEESE" when it says "ASTERISK" in flashy purple letter with sparkles |
| 05:50.31 | ManxPower | jblack: you'd be wrong. |
| 05:50.43 | jblack | well... before one gets _too_ insistent, check carefully for RELATED,ESTABLISHED to see if perhaps provider is getting a hidden exception. |
| 05:50.51 | mast3rpyr0 | gah give me a minute im regiestering domains and talking to 2 other people and too much at once.. |
| 05:50.54 | [TK]D-Fender | Hey... I bought an AM radio.... anyone want to help me make a satelite? :) |
| 05:51.06 | ManxPower | jblack: the new incoming connection is neither related or established. |
| 05:51.17 | drmessano | [TK]D-Fender: CAN AKERISK DEW THAT? |
| 05:51.23 | jblack | [TK]D-Fender: YOu did the hard part, and someone else should do the easy part of providing the rocket? |
| 05:51.29 | dacs | [TK]D-Fender: yep, get a big Dish and connect it to the radio |
| 05:51.30 | lzhang | ManxPower: 18 months is a too long of a time, depending on how much you and your coworkers are being paid I think it mightve been better just to buy a solution form someone |
| 05:51.31 | [TK]D-Fender | drmessano, I'm quite sure dew would cause a short! |
| 05:51.39 | drmessano | lol |
| 05:51.45 | ManxPower | lzhang: I didn't say we did it full time. |
| 05:51.58 | drmessano | [TK]D-Fender: R U NO SAY AKERISK DO, NO? |
| 05:52.07 | ManxPower | The Asterisk servers we have provide many more features than the existing turnkey pbxs we have. |
| 05:52.16 | lzhang | I can has VOIP? |
| 05:52.17 | jblack | Manxpower: O RLY? K THX BYE |
| 05:52.23 | drmessano | ROFL |
| 05:52.28 | drmessano | YEs |
| 05:52.32 | Qwell | oh, so... when I get a car... |
| 05:52.34 | drmessano | I CAN HAZ AKERISK? |
| 05:52.37 | [TK]D-Fender | jblack, No No.. I have an AM radio... a receiver, not a transmitter, but how hard could it be..... oh yeah and it has to be VOIP and NUKE-ULAR! |
| 05:52.38 | Qwell | my license plate is gonna be ICANHAZ |
| 05:52.52 | jblack | Some day I'm gonna get T-shirt that has O RLY on the front and K THX BYE!!! on the back |
| 05:53.02 | [TK]D-Fender | jblack, already done... |
| 05:53.24 | jblack | No doubt. |
| 05:53.26 | [TK]D-Fender | http://www.threadless.com/submission/70564/O_RLY |
| 05:53.42 | ManxPower | *sigh* I just realized I forgot to bring my "No, I don't fix your computer" t-shirt with me on this trip. |
| 05:53.52 | [TK]D-Fender | http://www.zazzle.com/o_rly_shirt-235698002768613398 |
| 05:54.11 | drmessano | I want a shirt that has "Trixbox" on the front and a silk screened "Kick Me" sign on the back |
| 05:54.12 | jblack | That is pretty close. |
| 05:54.13 | [TK]D-Fender | ManxPower, I like the dual-purpose "I read your e-mail" sysadmin shirt :) |
| 05:54.24 | ManxPower | Why anyone would say "OK, Surround sound, thanks" is beyond me. |
| 05:54.32 | lzhang | the best part about using mac/linux is being able to say to windows users; shit, I don't know what's wrong with your comp man... I use linux myself |
| 05:54.44 | ManxPower | [TK]D-Fender: I have that as a bumper sticker. The users at my customers would take the t-shirt too literally. |
| 05:55.07 | ManxPower | [TK]D-Fender: Do you remember me saying that I fired a customer? |
| 05:55.09 | jblack | I had a problem with people thinking I was empathetic and nice at work... So I wore a t-shit for a couple days that said "Am I pretending to care enough, yet?" |
| 05:55.29 | Qwell | a couple days? |
| 05:55.31 | Qwell | in a row? |
| 05:55.33 | drmessano | No, the best part of using a Mac is the lifetime license to be an elitist ass**** on Digg.. the SECOND part is the bit about being better than Windows :) |
| 05:55.34 | jblack | solved that problem. The whiners left me alone after that. |
| 05:55.44 | lzhang | jblack: maybe cuz you smelled |
| 05:55.47 | Qwell | that wasn't why they stopped talking to you :p |
| 05:55.51 | ManxPower | I don't fix computers, I fix networks. 8-) |
| 05:56.00 | jblack | qwell: yeah. Clothing washers work overnight, and it made sure people noticed it. |
| 05:56.07 | lzhang | ManxPower: so you're saying you run cable |
| 05:56.13 | [TK]D-Fender | ManxPower, Yeah I think so... |
| 05:56.33 | drmessano | "I dont fix computers, I fix people who use computers" |
| 05:56.45 | lzhang | I'm thinking about fixing my dog |
| 05:56.48 | [TK]D-Fender | "Guns don't kill people.... *I* kill people" |
| 05:56.59 | drmessano | [TK]D-Fender: I believe that ;) |
| 05:57.11 | lzhang | from shoot'em up: Guns dont kill ppl, but they sure help |
| 05:57.20 | ManxPower | [TK]D-Fender: Their new consultant migrated their domain to his servers, set up all their accounts for e-mail and told the previous provider to kill the domain and accounts on their server. The consultant didn't realize the customer had all their e-mail hosted on an IMAP server at their previous provider. All the customer's e-mail was lost. |
| 05:57.29 | [TK]D-Fender | drmessano, I think its time we bury the hatchet... you do know what a hatchet is don't you? ;) |
| 05:57.30 | drmessano | [TK]D-Fender is the Dexter of #asterisk.. Just taking out the trash |
| 05:57.30 | [TK]D-Fender | :f |
| 05:57.59 | [TK]D-Fender | ManxPower, IMAP woohoo! |
| 05:58.07 | dacs | jblack: mine says "DON'T HIT KIDS,no seriously. THEY HAVE GUNS NOW" |
| 05:58.14 | *** join/#asterisk fuzzbawl (n=fuzzbawl@blackhole.cyberlinkint.com) |
| 05:58.14 | drmessano | LOL |
| 05:58.43 | JT | Dexter is the best |
| 05:58.48 | drmessano | Yes he is |
| 05:58.49 | ManxPower | I love IMAP. Keeps you from having to migrate their e-mail to their latest computer. |
| 05:59.00 | lzhang | seriously guys, is sip phone to asterisk to sip phone supposed to be almost instantaneous audio??? |
| 05:59.02 | jblack | You can still hit kids with guns. You've just got to make sure the first whack really counts. |
| 05:59.11 | lzhang | I hate imap because it's slow |
| 05:59.15 | ManxPower | Would that be Dexter the serial killer, or Dexter the 7-yr old boy genius? |
| 05:59.15 | drmessano | LOL |
| 05:59.18 | mast3rpyr0 | gah ok i think im gonna abandon this and save this project for someone who knows about phone protocols |
| 05:59.26 | drmessano | Not IMAP 3000 miles away |
| 05:59.30 | mast3rpyr0 | tahnks for your help and sarcastic remarks |
| 05:59.37 | ManxPower | lzhang: My users can't tell the difference. |
| 05:59.48 | jblack | mast3rpyr0: Aww. I feel bad. Want a non-sarcastic comment/ |
| 05:59.50 | Shaun2222 | anybody know a why a incomming call on a zap interface wouldnt be able to join a queue? |
| 05:59.51 | lzhang | what if the phones are in the same room |
| 05:59.56 | drmessano | KTHX4AKERISKBYE |
| 06:00.15 | ManxPower | lzhang: then you could notice a delay, much like if you have 2 cell phones talking to each other in the same room |
| 06:00.26 | lzhang | here's the real problem |
| 06:00.37 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 06:00.39 | drmessano | "K I WUNT SOMETIN LIEK AKERISK BUT USERS ASKERISK PBX, NO YES NO?" |
| 06:00.43 | lzhang | I have our old altigen system, and our new asterisk test box set up |
| 06:00.44 | ManxPower | But I've never heard of a delay being more than you would experience with a cell phone. |
| 06:00.55 | lzhang | the president of the company comes in and complains about the delay |
| 06:01.06 | lzhang | I try it on our altigen shite, and it is instantaneous |
| 06:01.08 | lzhang | same room |
| 06:01.15 | lzhang | noticeable lag with the asterisk setup |
| 06:01.23 | ManxPower | You can usually shrink the jitterbuffer to reduce the delay or turn on reinvites (but that can cause a short delay just at the start of the call)( |
| 06:01.25 | dacs | jblack: "SL_T, all i need is U" |
| 06:01.46 | jblack | "AKERISK SUX0RZ! I want my 1 minute of googling back!" |
| 06:01.48 | dacs | jblack: that will be a nice gift for girl friend b-day |
| 06:01.51 | drmessano | HAHAHHA |
| 06:01.54 | Shaun2222 | wouldnt there always be a delay from the overhead of compressing/decompressing the audio? |
| 06:01.56 | ManxPower | lzhang: I have no idea what an altigen is. Some form of PBX, I assume. |
| 06:02.11 | ManxPower | Shaun2222: why compress/decompress the audio for calls on the same lan? |
| 06:02.11 | jblack | dacs: Not bad. |
| 06:02.13 | lzhang | ManxPower: sounds like something I definitely want to try, thanks |
| 06:02.29 | drmessano | "I UNZAPED AKERSIK ON VISTA AND EXE NO CUM UP...... .... Y THIS??? " |
| 06:02.34 | ManxPower | We use ALL ulaw for phone<->asterisk calls. |
| 06:02.43 | lzhang | Shaun2222: this is running over a local network, maybe I should run the least compressing protocol? |
| 06:02.45 | ManxPower | Asterisk <-> Asterisk calls are usually GSM, I think. |
| 06:02.47 | Shaun2222 | ManxPower: i figured all the codecs did some audio compression/encoding... |
| 06:02.58 | *** join/#asterisk adker (n=chatzill@74-33-205-192.br1.glv.ny.frontiernet.net) |
| 06:03.03 | ManxPower | Shaun2222: ulaw/alaw is what the telcos use. |
| 06:03.14 | jblack | Compressing audio is a good thing. It makes my girlfriend sound skinny |
| 06:03.15 | Shaun2222 | good to know. |
| 06:03.17 | mast3rpyr0 | lol you guys must have no lives if your still going on about me.. |
| 06:03.18 | lzhang | the thing is I tried disallow all and allow ulaw in my sip.conf |
| 06:03.23 | drmessano | LOL jblack |
| 06:03.40 | lzhang | but watching the CLI I cant tell if it is actually running using ulaw |
| 06:03.47 | Shaun2222 | jblack: lol, that sucks ;) |
| 06:03.47 | mast3rpyr0 | go talk about your switches and fancy protocols that dont work like anything else |
| 06:03.48 | jblack | Well... if I had a fat girlfriend. Sadly, I don't even have _that_ |
| 06:03.48 | ManxPower | sip show channels |
| 06:03.54 | drmessano | mast3rpyr0, you're just another n00b who doesnt read.. we've moved on |
| 06:04.08 | lzhang | ok I will write this stuff down for tomorrow, thanks a lot for your help Manx |
| 06:04.30 | mast3rpyr0 | lol doesnt look liek it |
| 06:04.31 | lzhang | jblack: what are you talking about fat girlfriends are awesome |
| 06:04.32 | ManxPower | lzhang: feel flattered, I'm officially on asterisk support strike. |
| 06:04.45 | lzhang | how magnanimous of you |
| 06:04.56 | jblack | lzhang: Sure, if you need to test your car's struts. |
| 06:05.05 | mast3rpyr0 | but whatever, im getting outa freenode before i catch something |
| 06:05.07 | drmessano | Y R AKERISK NOT GUI? |
| 06:05.08 | ManxPower | lzhang: you may bask in my magnificence |
| 06:05.13 | jblack | Bad, if you need to take a right turn at speed. |
| 06:05.23 | lzhang | it's... so.. beautiful!! |
| 06:05.27 | ManxPower | 8-) |
| 06:05.32 | Shaun2222 | nice nick.... |
| 06:05.45 | jblack | Isn't there a h9k reference in the code somewhere? |
| 06:05.47 | ManxPower | oh, that's magnanimousnes |
| 06:06.00 | drmessano | Hmm |
| 06:06.05 | jblack | Yup. |
| 06:06.09 | drmessano | That was a lousy SQL statement |
| 06:06.12 | jblack | ./main/http.c:return ast_http_error(403, "Access Denied", NULL, "Sorry, I cannot let you do that, Dave."); |
| 06:06.23 | lzhang | ahaha |
| 06:06.51 | [TK]D-Fender | HAL : Hardware Abstraction Layer MY ASS!!!!!!! |
| 06:06.53 | drmessano | drop table dumbshitmast3rpyr0said; |
| 06:07.37 | Shaun2222 | hmm does the stupid linksys wifi phones not let you enter keys while on a call? |
| 06:07.56 | jblack | They couldn't be that dumb. |
| 06:07.59 | dacs | Shaun2222: why is that |
| 06:08.00 | lzhang | the life of a php developer trying to set up asterisk is an arduous one |
| 06:08.15 | Shaun2222 | dacs: it was a question... |
| 06:08.19 | [TK]D-Fender | jblack, Linksys Wifi has no Transfer, conference and a whole host of other useful things... |
| 06:08.23 | Shaun2222 | trying to go through a IVR using it. |
| 06:08.35 | lzhang | you could make the dtf with your voice |
| 06:08.40 | jblack | sure, but no dtmf? |
| 06:08.40 | dacs | Shaun2222: tellabs |
| 06:08.45 | [TK]D-Fender | Shaun2222, And the lack of DTMF is because you didn't set the mode right |
| 06:08.50 | Shaun2222 | [TK]D-Fender: this one has transfer and converence and line 2... |
| 06:08.50 | drmessano | lzhang: remember register_globals = very yes |
| 06:08.52 | jblack | That doesn't even rise to the level of a $3.99 walmart phone. |
| 06:09.11 | Shaun2222 | [TK]D-Fender: set the mode right where? |
| 06:09.17 | [TK]D-Fender | Shaun2222, sip.conf |
| 06:09.19 | lzhang | set register globals ON |
| 06:09.30 | Shaun2222 | register_globals is evil |
| 06:09.37 | drmessano | Yes it is |
| 06:09.42 | [TK]D-Fender | jblack, just a bad mode choice... |
| 06:09.42 | drmessano | VERY OFF NOW KTHBYE |
| 06:09.50 | lzhang | register globals is like bareback sex |
| 06:09.54 | lzhang | AWESOME but DANGEROUS |
| 06:10.05 | drmessano | Hmm |
| 06:10.31 | Shaun2222 | [TK]D-Fender: never had to do that before, but my other phones are polycoms... what do i need to set it to? |
| 06:10.33 | drmessano | I was thinking "What does her husband coming home have to do with PHP?" |
| 06:10.43 | drmessano | I got ya now |
| 06:11.37 | [TK]D-Fender | Shaun2222, I suggest AVT on the linksys rfc2833 on * |
| 06:11.40 | Shaun2222 | i wouldnt say register_globals being on is awsome... or like bareback sex... more like oh shit the condom broke |
| 06:11.52 | Shaun2222 | whats his bitch have :) |
| 06:11.59 | Shaun2222 | s/his/this/ |
| 06:12.14 | jblack | Yeah. turning on register_globals on is exactly like a condom. One small break, and you're fucked. |
| 06:12.16 | lzhang | for a second there I thought your were insulting my gf |
| 06:12.25 | drmessano | lol |
| 06:12.34 | lzhang | seriously |
| 06:13.16 | lzhang | ok I'm outs thanks for the help guys |
| 06:13.46 | jblack | what's this? Sun is outsourcing itself? |
| 06:15.30 | Shaun2222 | [TK]D-Fender: i need to set dtmf on the phone it self too? |
| 06:15.30 | jblack | Apparently, they're gonna virtualize the data center, saving on machines. |
| 06:15.55 | jblack | Perhaps then they'll virtualize the virtualized machines, and then virtualize that.. Thereby causing Sun to collapse in on themselves, causing a black hole |
| 06:16.24 | [TK]D-Fender | Shaun2222, I answered BOTH END on that one already... |
| 06:16.28 | jblack | dacs: Your authentication is wrong |
| 06:16.37 | jblack | SIP/2.0 407 Proxy Authentication Required |
| 06:17.13 | [TK]D-Fender | jblack, not necessarily |
| 06:17.14 | Shaun2222 | [TK]D-Fender: n/m i figured it out, i didnt need to change anything on asterisk... just needed to enable dtmf relay on the phone.... |
| 06:17.25 | [TK]D-Fender | dacs, pastebin please... |
| 06:17.40 | jblack | He put things up here: http://pastebin.ca/850412 |
| 06:18.55 | [TK]D-Fender | tath same ting? Thats jsut an echo test, the call is accepted and he's in the echo test. Whats the issue? |
| 06:19.01 | drmessano | jblack: Is that like installing windows in a VM on an Asterisk box to install a softphone so it can call itself? |
| 06:19.08 | jblack | Yeah, I may be wrong. His dialplan is executing |
| 06:19.18 | AndyGraybeal | what is a fun softphone for linux? |
| 06:19.22 | jblack | drmessano: No idea... but here's fun. |
| 06:19.33 | drmessano | Fun softphone? |
| 06:19.43 | AndyGraybeal | yea, like exciting! |
| 06:19.44 | jblack | Earlier today, I considered hooking up accoustic modems over a couple * phones, to see if I could get a routable network. |
| 06:19.44 | drmessano | WackyClownPhone 0.97beta is HILAAAARIOUS |
| 06:19.45 | AndyGraybeal | hehe |
| 06:20.00 | AndyGraybeal | yea, i'd imagine something like indiana jones phone would be awsome |
| 06:20.05 | dacs | jblack: i changed line 208 to exten => 500,1,Dial(SIP/phone1) |
| 06:20.08 | jblack | I could do *, over ppp, over * over ethernet. That would ROCK! |
| 06:20.17 | drmessano | ROFFFFL |
| 06:20.24 | drmessano | jblack.. thats funny |
| 06:20.39 | [TK]D-Fender | dacs, pastebin yuor extensions.conf ,"dialplan show", and the clie output of your next call |
| 06:20.41 | drmessano | Asterisk Online ? |
| 06:20.43 | Shaun2222 | how can i debug why i call cant join a queue... nothing in the console or full log is giving me any ideas |
| 06:20.43 | drmessano | lol |
| 06:21.03 | [TK]D-Fender | Shaun2222, show US <- |
| 06:22.35 | drmessano | [TK]D-Fender needs a bot, Called "Pastebin_It_Now", so he can P-A-<TAB> |
| 06:22.52 | [TK]D-Fender | drmessano, I have a bot already! |
| 06:22.54 | [TK]D-Fender | ~jbot |
| 06:22.55 | jbot | somebody said jbot was a hack!, or known to have only said one useful thing. a tool, or dating the fembots, or [TK]D-Fender's b*tch |
| 06:22.56 | Shaun2222 | [TK]D-Fender: http://pastebin.ca/850459 |
| 06:23.04 | drmessano | LOL |
| 06:23.13 | drmessano | ~[TK]D-Fender |
| 06:23.14 | jbot | [TK]D-Fender is the Zen Master of the blatantly obvious, and #asterisk 's resident Google-proxy. |
| 06:23.17 | [TK]D-Fender | ~botsnack |
| 06:23.17 | jbot | :), [TK]D-Fender |
| 06:23.26 | drmessano | Obviously your bot |
| 06:24.00 | drmessano | Im almost certain you're one too.. and I had no clue that Alice had gotten so advanced |
| 06:24.06 | drmessano | Must be running SVN or something |
| 06:24.08 | [TK]D-Fender | Shaun2222, "show queue sales" and pastebin queues.conf |
| 06:24.26 | drmessano | Yep, SVN |
| 06:24.33 | jblack | ohhhhh. |
| 06:24.36 | jblack | 0xb65fa173 in strlen () from /lib/tls/i686/cmov/libc.so.6 |
| 06:24.50 | *** join/#asterisk neoalex (n=chatzill@user-0ccengj.cable.mindspring.com) |
| 06:24.56 | JT | drmessano: it's riker's bot |
| 06:24.57 | jblack | I'd say bad config file. |
| 06:25.13 | neoalex | hi guys, does the metermaid patch work for asterisk 1.2.24 |
| 06:25.14 | Shaun2222 | [TK]D-Fender: http://pastebin.ca/850461 |
| 06:25.43 | [TK]D-Fender | Shaun2222, Your queue has no valid members <- |
| 06:25.52 | [TK]D-Fender | Shaun2222, it is a dea-end, no wonder it won't let people in |
| 06:25.56 | [TK]D-Fender | dead* |
| 06:26.08 | Shaun2222 | bahh.. forgot joinempty = yes |
| 06:26.12 | [TK]D-Fender | Shaun2222, local/306 (Invalid) has taken no calls yet <-- |
| 06:26.31 | [TK]D-Fender | Shaun2222, there is a difference between "joinempty" and "joinhopeless" you know... |
| 06:26.41 | jblack | holy crap. ekiga does 2163 opens before it segfaults. |
| 06:26.49 | Shaun2222 | wonder why it says local/306 is invalid |
| 06:27.04 | jblack | .gtk-bookmarks alone, no less than a dozen times |
| 06:27.04 | [TK]D-Fender | jblack, darn.. I was sure it'd hit 3000 easy! |
| 06:27.17 | [TK]D-Fender | Shaun2222, because you didn't specify...... a CONTEXT <- |
| 06:27.38 | Shaun2222 | i just changed it to local/306@default |
| 06:27.40 | Shaun2222 | same thing |
| 06:27.42 | drmessano | ZOMG, A CONTEST.. DID I WIN? |
| 06:27.56 | jblack | brb |
| 06:28.01 | [TK]D-Fender | Shaun2222, pastebin please... including dialplan..... |
| 06:28.03 | Shaun2222 | [default] has exten => _3XX,1,Dial(SIP/306) also.... |
| 06:28.22 | Shaun2222 | which is ghetto i know... it's a test |
| 06:28.37 | *** join/#asterisk jblack (n=jblack@pool-71-181-138-192.sctnpa.east.verizon.net) |
| 06:28.43 | jblack | Huh. There was nobody in #akerisk |
| 06:29.06 | [TK]D-Fender | Shaun2222, pastebin........ |
| 06:29.44 | Shaun2222 | http://pastebin.ca/850467 |
| 06:31.06 | dacs | [TK]D-Fender: info you requested |
| 06:31.40 | drmessano | AKERISK IS DEAD? HURRAH UP, SOME1 FORK IT |
| 06:31.56 | dacs | [TK]D-Fender: http://pastebin.ca/850468 |
| 06:33.28 | [TK]D-Fender | dacs, so whats the problem in there? |
| 06:33.36 | *** join/#asterisk jblack (n=jblack@pool-71-181-138-192.sctnpa.east.verizon.net) |
| 06:33.38 | [TK]D-Fender | Shaun2222, All of the backup please... |
| 06:33.49 | Shaun2222 | all of the backup? |
| 06:34.39 | drmessano | YAY.. I R PROJECT LEEDER IN #AKERISK |
| 06:35.13 | dacs | [TK]D-Fender: now i want to bring in my voip provider |
| 06:35.31 | Shaun2222 | fuckin a... |
| 06:35.32 | [TK]D-Fender | dacs, So you weren't showing me a *problem*? |
| 06:35.36 | Shaun2222 | [TK]D-Fender: n/m i figured it out |
| 06:35.47 | Shaun2222 | i hit # instead of @ with teh member context |
| 06:35.56 | [TK]D-Fender | Shaun2222, do be a lad and smite yourself justly! :p |
| 06:36.35 | Shaun2222 | [TK]D-Fender: ya i'll take care of it :) |
| 06:37.02 | dacs | [TK]D-Fender: you asked for the info |
| 06:37.26 | *** join/#asterisk jarod14 (n=jarod14@ns2.viatelecom.com) |
| 06:37.58 | [TK]D-Fender | dacs, I was told you actually had a problem... |
| 06:38.08 | AndyGraybeal | hahha the freshmaker.. |
| 06:38.54 | dacs | [TK]D-Fender: my problem is i want to get my provider in asterisk servcer now |
| 06:39.20 | [TK]D-Fender | dacs, that isn't a problem, thats a nameless wish list. |
| 06:39.54 | jblack | Where is dacs at? Did the previous problem get fixed? |
| 06:42.13 | [TK]D-Fender | jblack, There wasn't one apparently |
| 06:42.39 | [TK]D-Fender | dacs, Go read a guide on how to set them up. when in doubt also look at how other ITSPs are set up. |
| 06:44.15 | *** join/#asterisk Porks (n=Porks@201.62.79.12) |
| 06:51.26 | Shaun2222 | QUEUE doesnt happen to set a env variable thats set to the queue name now would it? |
| 06:52.21 | Shaun2222 | i need a way to get the queue name a call was received from to the member.. |
| 06:52.47 | Shaun2222 | right now i'm setting a __Var but that seams alittle ghetto. |
| 06:57.47 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 06:59.40 | drmessano | Ok, next its my turn to set up an ITSP |
| 06:59.46 | drmessano | Let me know when youre ready |
| 06:59.56 | drmessano | Not add one, but I want to be one.. |
| 07:00.00 | drmessano | On 768k DSL |
| 07:01.32 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 07:04.37 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 07:05.22 | *** join/#asterisk mtryfoss (n=mtryfoss@80.239.93.22) |
| 07:07.50 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 07:13.44 | *** join/#asterisk tones95 (n=tones95@c1r-hollywood.cclr.net) |
| 07:14.01 | tones95 | hey ya'all |
| 07:14.14 | tones95 | can I ask a question I am sure/hope you hear a million times a day |
| 07:14.34 | tones95 | 1 way audio, I think it's a nat issue, but don't have a non-nat machine to work with to verify at the moment |
| 07:16.28 | mtryfoss | nat=yes enabled ? |
| 07:18.09 | *** join/#asterisk Porks (n=Porks@201.62.79.12) |
| 07:21.13 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 07:21.41 | Shaun2222 | does a channel need to be answered in every context it's sent to? |
| 07:22.04 | mtryfoss | no |
| 07:23.35 | kaldemar | tones95: http://www.voip-info.org/wiki-Asterisk+sip+nat |
| 07:24.08 | tones95 | mtryfoss nat=yes enabled |
| 07:31.16 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 07:39.25 | *** join/#asterisk fukz (n=johns@p4FC2CDF6.dip.t-dialin.net) |
| 07:45.31 | *** join/#asterisk felipex (n=dsfdsf@213-140-21-233.fastres.net) |
| 07:47.30 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 07:49.28 | *** join/#asterisk Aurs (n=Aurs@1elt2pn.ip.hipercom.no) |
| 07:53.16 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 07:54.39 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
| 07:55.01 | *** join/#asterisk af_ (n=getsmart@88-149-241-54.dynamic.ngi.it) |
| 07:55.52 | *** join/#asterisk harpal (n=Harpal@124.125.79.212) |
| 07:55.55 | *** join/#asterisk chanko (n=slaven@adsl3ci253.tel.net.ba) |
| 08:02.42 | Shaun2222 | is it better to use | or , in the extensions.conf? |
| 08:02.51 | nixguy | Shaun2222: it doesent really matter |
| 08:02.57 | nixguy | when asterisk parses the config file |
| 08:03.08 | nixguy | it replaces all | with , |
| 08:03.11 | nixguy | or viceversa :) |
| 08:03.17 | Shaun2222 | ya i know they both work, but with how asterisk likes to depricate things... |
| 08:03.43 | Shaun2222 | i would rather just use what it wants and be proper |
| 08:04.15 | nixguy | |
| 08:04.23 | nixguy | sorry bout that |
| 08:04.32 | nixguy | im still a noob so i can only tell you what i've read |
| 08:04.45 | nixguy | and and in my oreiley guide book they use , |
| 08:04.49 | nixguy | in their examples |
| 08:05.12 | nixguy | eventhough they state that you can use | |
| 08:05.31 | Shaun2222 | alot of the examples i see use , too |
| 08:05.38 | Shaun2222 | i'll just use those i guess |
| 08:05.47 | nixguy | that will make the two of us! |
| 08:06.00 | nixguy | spreading ,,,, in asterisk land! |
| 08:06.02 | nixguy | ,,,, |
| 08:06.03 | nixguy | ,, |
| 08:06.29 | Shaun2222 | %s/|/,/g |
| 08:06.32 | Shaun2222 | that wasnt hard :) |
| 08:06.41 | *** join/#asterisk xbmodder_ (n=Sargun@atarack/staff/sargun) |
| 08:06.47 | xbmodder_ | I'm having one way audio issues |
| 08:06.57 | Shaun2222 | now watch my extensions go hay wire |
| 08:07.14 | xbmodder_ | With Gizmo |
| 08:07.51 | xbmodder_ | I can hear the Gizmo user, they cannot hear me |
| 08:08.00 | xbmodder_ | I am not behind a NAT |
| 08:08.05 | xbmodder_ | or firewall |
| 08:09.08 | Aurs | from sip.conf.sample: "; See doc/README.tos for a description of these parameters. |
| 08:09.21 | Aurs | but I can't find that file? |
| 08:13.16 | *** join/#asterisk Alexandre_fr (n=alex@nat04-bzr.dys1.com) |
| 08:20.37 | drmessano | xbmodder_: Your box is on a public IP? |
| 08:27.40 | *** part/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 08:28.17 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 08:29.15 | *** join/#asterisk Federico2 (n=fede@pdpc/supporter/base/Federico2) |
| 08:29.18 | Federico2 | hi there |
| 08:31.05 | *** join/#asterisk jarod14 (n=jarod14@viatel-nautilsiege-52-218.cnt.nerim.net) |
| 08:31.25 | *** join/#asterisk stimpie (n=stimpie@84-104-5-115.cable.quicknet.nl) |
| 08:31.59 | *** join/#asterisk mosty (n=mostyn@60-241-198-194.static.tpgi.com.au) |
| 08:32.20 | *** join/#asterisk tengulre (n=tengulre@124.42.50.9) |
| 08:32.23 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 08:34.08 | *** part/#asterisk jarod14 (n=jarod14@viatel-nautilsiege-52-218.cnt.nerim.net) |
| 08:42.10 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 08:57.34 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 09:03.24 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 09:10.35 | *** join/#asterisk qdk (n=qdk@85.235.253.139) |
| 09:15.27 | jblack | <PROTECTED> |
| 09:15.41 | *** join/#asterisk Sinar (n=bwoSinar@81.144.146.38) |
| 09:16.18 | Sinar | Anyone know how to prevent extra INVITE packets being sent after a call has been forwarded? Seeing extra invites AFTER native bridging is established, but if I turn off 'canreinvite' the native bridging doesn't happen |
| 09:16.31 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 09:18.32 | *** join/#asterisk na0mi (n=kieran@82.152.56.113) |
| 09:19.17 | *** join/#asterisk RoyK (n=roy@fw.fortel.no) |
| 09:20.30 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 09:22.40 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 09:23.04 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 09:23.27 | creativx | wphooo |
| 09:32.16 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 09:32.16 | *** mode/#asterisk [+o denon] by ChanServ |
| 09:34.51 | *** join/#asterisk DarkRift (n=dark@bas10-montreal02-1177584876.dsl.bell.ca) |
| 09:36.53 | Sinar | If i do anything to stop the extra invite packets, it seems the native bridging fails, and even though the call is ACK'd and answered ok, there's no audio |
| 09:40.18 | *** join/#asterisk admin0 (n=admin@116.90.228.34) |
| 09:44.46 | *** join/#asterisk vale-ICS (n=vale@62.49.27.29) |
| 09:48.32 | *** part/#asterisk chanko (n=slaven@adsl3ci253.tel.net.ba) |
| 09:51.59 | *** part/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 09:52.50 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 09:53.27 | *** join/#asterisk tengulre (n=tengulre@124.42.50.9) |
| 09:57.21 | *** join/#asterisk shtoom (n=godson@59.93.113.128) |
| 10:28.54 | *** join/#asterisk Victor_Yure (n=aaa@200.166.132.131) |
| 10:31.02 | dennis- | ~book |
| 10:31.02 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 10:36.02 | *** join/#asterisk admin0 (n=admin@116.90.228.34) |
| 10:38.14 | *** join/#asterisk _ys (i=yuri@91.151.196.254) |
| 10:40.20 | *** join/#asterisk Porks (n=Porks@201.62.79.12) |
| 10:43.44 | *** join/#asterisk fujin (n=aj@unaffiliated/fujin) |
| 10:52.57 | *** join/#asterisk khronos (n=khronos@c-66-229-159-175.hsd1.fl.comcast.net) |
| 10:53.01 | *** join/#asterisk justdave_ (n=dave@unaffiliated/justdave) |
| 11:11.33 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 11:13.07 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 11:15.16 | *** join/#asterisk Winkie (n=urmom@149.254.192.192) |
| 11:16.47 | *** join/#asterisk vrtk (n=bb@189.21.178.20) |
| 11:22.47 | *** join/#asterisk nighty^ (n=nighty@p7068-adsau17honb13-acca.tokyo.ocn.ne.jp) |
| 11:32.36 | yang | How do I solve Jan 11 12:31:44 WARNING[13511]: app_dial.c:972 dial_exec_full: privacy: can't create directory priv-callerintros: No such file or directory |
| 11:33.20 | *** part/#asterisk admin0 (n=admin@116.90.228.34) |
| 11:33.38 | *** join/#asterisk Tili (n=tili@cm48.gamma244.maxonline.com.sg) |
| 11:37.20 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 11:40.47 | *** join/#asterisk puga (n=rafael@189.5.213.166) |
| 11:45.48 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:50.15 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 11:50.26 | *** join/#asterisk coppice (n=chatzill@230.202.17.210.dyn.pacific.net.hk) |
| 12:02.42 | *** join/#asterisk Winkie (n=urmom@149.254.192.192) |
| 12:02.47 | *** join/#asterisk tobias (n=tobias@user-0ce2hr3.cable.mindspring.com) |
| 12:03.13 | *** join/#asterisk ez` (n=ez@c75.152.78-116.clta.globetrotter.net) |
| 12:03.55 | *** join/#asterisk Winkie (n=urmom@149.254.192.192) |
| 12:04.23 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 12:18.58 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 12:31.54 | *** join/#asterisk vetetix (n=vetetix@eclip3.ec-lille.fr) |
| 12:32.04 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.64.193) |
| 12:32.35 | *** join/#asterisk cjk (n=cjk@d90-129-39-85.cust.tele2.lu) |
| 12:32.54 | cjk | hi, is there an application to send mwi messages to a user? |
| 12:34.15 | ManxPower | cjk: no. What are you trying to do, send morse code? |
| 12:34.44 | *** join/#asterisk jim`` (n=jim@wildern.plus.com) |
| 12:34.48 | cjk | ManxPower, oh now, i just do not like the voicemail application in asterisk its far too complex. so i write my own in agi. now i want to send mwi |
| 12:36.35 | ManxPower | I don't know about in 1.4, but in 1.2 and earlier there is no application to do that, however, you can just create the .msg and .txt files in the user's mailbox, asterisk will automatically turn the MVI on or off. |
| 12:39.17 | *** join/#asterisk jochien1 (n=jochieng@217.194.147.193) |
| 12:39.33 | *** join/#asterisk choogaistir (n=choog@91.192.151.1) |
| 12:39.52 | jochien1 | how can i upgrade 1.2 to 1.4 on etch |
| 12:40.43 | choogaistir | guys, i have one fu**n question about call queue and calls transfer which not in queue_log. Why? ))))) |
| 12:41.06 | ManxPower | jochien1: we don't really support prepackaged asterisk here. |
| 12:41.28 | ManxPower | jochien1: read UPGRADE.txt in 1.2 and 1.4 source code, that should give you everything you need. |
| 12:42.46 | mtryfoss | is it recommended to have the irqbalance deamon enabled or disabled when using zaptel ? |
| 12:43.21 | jochien1 | ManxPower: i hope i keep my configs intact after the upgrade |
| 12:43.41 | ManxPower | jochien1: I doubt you can, as some features in 1.2 have been removed or replaced in 1.4 |
| 12:43.55 | ManxPower | Which is why I suggested you read UPGRADE.txt |
| 12:44.22 | jochien1 | !upgrade |
| 12:44.28 | jochien1 | ManxPower: ok |
| 12:46.09 | nixguy | does one want to upgrade to 1.4? |
| 12:46.14 | nixguy | is it "stable" enough? |
| 12:46.37 | nixguy | a colleauge of mine constantly compalains about the spaghetti code of asterisk, but im no programmer so i cant really say.. |
| 12:48.27 | Alexandre_fr | asterisk 1.4 is stable, but you have to be carefull when you upgrade because some featurs change |
| 12:49.46 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 12:49.49 | jochien1 | i only have basic configs n i dnt mind loosing any ;-( |
| 12:52.17 | jochien1 | does 1.2 support IM services |
| 12:52.35 | mvanbaak | asterisk does not do IM |
| 12:54.03 | *** join/#asterisk af_ (n=getsmart@88-149-241-54.dynamic.ngi.it) |
| 12:54.37 | jochien1 | mvanbaak: i meant SIP messaging |
| 12:55.37 | puppet | mvanbaak: doesnt asterisk support jabber? |
| 12:57.38 | mvanbaak | puppet: it does, but only for some stuff |
| 12:58.09 | jochien1 | <PROTECTED> |
| 12:58.18 | jochien1 | http://voip-info.linuxsys.com/wiki/view/Asterisk+SIP+Messaging.html |
| 13:22.10 | *** join/#asterisk Cart- (i=cart@tyristori.de) |
| 13:22.13 | Cart- | hello |
| 13:22.34 | Cart- | any idea why asterisk segfaults in ubuntu 7.10 when trying to start it? http://pastebin.com/m1593ae53 |
| 13:23.21 | *** join/#asterisk [gnubie] (n=[gnubie]@cm205.gamma183.maxonline.com.sg) |
| 13:24.08 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:27.21 | *** join/#asterisk ming_zym (n=ming_zym@124.14.232.96) |
| 13:28.41 | jblack | Awesome. Now I'm part of a dundi network. |
| 13:28.59 | creativx | crocodile dundi |
| 13:32.53 | jblack | Huh. My very first * call was December 24th, at 17:40pm. |
| 13:35.57 | jblack | I wonder why all my cdr logs say DOCUMENTATION |
| 13:38.24 | [TK]D-Fender | jblack: .... RTFM :) |
| 13:38.35 | [TK]D-Fender | </lol> |
| 13:38.43 | jblack | Heh. I should. :) |
| 13:39.16 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 13:39.29 | jblack | correction, I will |
| 13:39.40 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:40.02 | jblack | My bet is that it's a statement of being. I'm so full of wisdom, that any call I'm in qualifies as DOCUMENTATION |
| 13:40.29 | mvanbaak | lol yeah |
| 13:41.52 | jblack | ahhh. amaflags. |
| 13:42.28 | [TK]D-Fender | jblack: If they call them "wisdom teeth", then why aren't they smart enough to leave by themselves when its time? |
| 13:42.52 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 13:43.02 | jblack | Good question. |
| 13:43.28 | jblack | cmon book, tell me what automated message accounting flags does for me! |
| 13:46.07 | jblack | Imagine. George bush and Lewis Monisky are setting up trysts on my conference line. Before I know it, Starr is at my door with a warrent. |
| 13:46.42 | [TK]D-Fender | jblack: Lies... we both know "W" doesn'[t feel he NEEDS to issue warrants for stuf ;) |
| 13:47.18 | jblack | When it's his dirty laundry? Hell. I bet he thinks a warrent isn't good enough at all, when it comes to him. |
| 13:52.39 | jblack | But first, I'm going to start the first national "Start smoking" hotline. How to get past the first smoke. How to buy in bulk. What diseases smoking protects against. One stop shopping! |
| 13:57.25 | *** join/#asterisk jarod14 (n=jarod14@ns2.viatelecom.com) |
| 13:58.45 | *** join/#asterisk dundel (n=daniel@200.2.161.172) |
| 14:00.51 | *** join/#asterisk lirakis (n=lirakis@65.200.191.241) |
| 14:03.13 | *** join/#asterisk Winkie (n=urmom@149.254.192.192) |
| 14:04.05 | choogaistir | anybody knows why transfered calls from agents not storing in queue_log as TRANSFER ? only COPLETECALLER/AGENT |
| 14:06.16 | lirakis | choogaistir: what version of * |
| 14:06.59 | choogaistir | lirakis, Asterisk 1.4.11 |
| 14:07.32 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:09.48 | [TK]D-Fender | choogaistir: How are you transferring the call? |
| 14:09.50 | lirakis | choogaistir: and are calls transfered from the phone features? |
| 14:10.45 | choogaistir | [TK]D-Fender, lirakis, yep, cisco 7940 with sip firmware |
| 14:11.06 | [TK]D-Fender | choogaistir: You need to use features.conf DTMF transfers and the "tT" flags on app_queue |
| 14:11.46 | choogaistir | "tT" already set |
| 14:12.01 | choogaistir | and transfers work fine ))) |
| 14:12.36 | choogaistir | [TK]D-Fender, but any information about calls transfer not storing to queue_log |
| 14:13.19 | jochien1 | !classful routing |
| 14:13.44 | [TK]D-Fender | choogaistir: You CANNOT use the "transfer" button on your phone. It must be via DTMF |
| 14:15.47 | choogaistir | [TK]D-Fender, hm, thats... if i wanna to log transfers, i must transfer calls via dtmf, not by button? |
| 14:16.02 | [TK]D-Fender | choogaistir: 3rd times the charm it seems.... YES |
| 14:16.15 | choogaistir | wtf? ))))))) |
| 14:18.15 | choogaistir | [TK]D-Fender, where i can see it? google say nothing |
| 14:18.33 | lirakis | choogaistir: .. lol .. i mean.. how is * supposed to know what your phone is doing magically? |
| 14:20.49 | *** join/#asterisk fiXXXerMet (n=kjohnson@dsl092-156-002.wdc2.dsl.speakeasy.net) |
| 14:20.52 | choogaistir | lirakis, i dont know, simply when i pressing "transfer", call transferin` ))) |
| 14:21.11 | [TK]D-Fender | choogaistir: http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue |
| 14:21.27 | [TK]D-Fender | choogaistir: "Transfers of calls that are answered out of a queue must be done using Asterisk '#' transfers (enabled with the 't' option above). SIP transfers result in the Agent remaining affiliated with the call until its eventual termination, preventing that agent from being offered another call." |
| 14:23.08 | choogaistir | [TK]D-Fender, 10x |
| 14:26.25 | davevg-btwtech | is there a known limitation in AGI on setting a variable based on an asterisk function (specifically STAT)? |
| 14:27.14 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 14:29.05 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 14:30.59 | *** part/#asterisk jochien1 (n=jochieng@217.194.147.193) |
| 14:32.11 | *** join/#asterisk mog (n=mog@216.207.245.1) |
| 14:32.11 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:33.22 | *** join/#asterisk tobias (n=tobias@cpe-069-134-226-227.nc.res.rr.com) |
| 14:33.55 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:33.55 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:35.13 | [TK]D-Fender | davevg-btwtech: Well, what happens when you try? |
| 14:36.56 | davevg-btwtech | sets it to the string instead of interpolating the results.. ie: ${STAT(e,/var/lib/asterisk/sounds/c3/tts/test.ulaw)} instead of 1, 0 or null |
| 14:37.11 | [TK]D-Fender | davevg-btwtech: And when you try with any other function? |
| 14:37.55 | *** join/#asterisk dundel (n=daniel@200.2.161.172) |
| 14:38.23 | davevg-btwtech | i've tried SET VARIABLE and EXEC SET with both the same results |
| 14:39.01 | davevg-btwtech | i can work around it by creating a shell script with an exit code and using system |
| 14:39.34 | [TK]D-Fender | davevg-btwtech: Shell Script? You're already in AGI... what language doesn't let you do that from within its confines? |
| 14:39.49 | [TK]D-Fender | davevg-btwtech: And I asked about your use of a function BESIDES STAT |
| 14:39.52 | davevg-btwtech | using fastagi, the agi script is not run on the * server |
| 14:40.07 | [TK]D-Fender | davevg-btwtech: Make sense |
| 14:40.22 | *** join/#asterisk hmmhesays (n=Neg@24-119-176-74.cpe.cableone.net) |
| 14:45.00 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 14:51.11 | fiXXXerMet | downloads.digium.com doesn't seem to be responding. |
| 14:51.45 | *** part/#asterisk [gnubie] (n=[gnubie]@cm205.gamma183.maxonline.com.sg) |
| 14:52.42 | [TK]D-Fender | fiXXXerMet: What protocol? |
| 14:54.02 | *** part/#asterisk dacs (n=haiger@unaffiliated/dacs) |
| 14:54.17 | fiXXXerMet | [TK]D-Fender: http |
| 14:54.20 | *** join/#asterisk macros73 (n=cs@c-24-3-246-27.hsd1.pa.comcast.net) |
| 14:54.21 | fiXXXerMet | Ah, back up now. |
| 14:54.38 | *** join/#asterisk JunK-Y (n=junky@modemcable183.17-83-70.mc.videotron.ca) |
| 14:57.54 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 15:01.58 | fiXXXerMet | Question about using realtime with MySQL. I already have my system configured using static, flat files (the default way). When I setup extconfig.conf and res_mysql.conf, how do the settings get from the flat files, into the database? |
| 15:02.08 | fiXXXerMet | Also, need I manually create the database tables, or is there a script somewhere? |
| 15:02.38 | *** join/#asterisk Blinkiz (n=niklas@h-89-233-204-231.wholesale.rp80.se) |
| 15:04.15 | *** join/#asterisk mkl1525 (n=qwertz@89.246.169.200) |
| 15:04.57 | shido6 | you can write a perl script to do it quickly..... or u can enter them one at a time , muahaha |
| 15:05.16 | shido6 | i think there is a script you can run to jump start your development |
| 15:05.46 | fiXXXerMet | OK, so I need to hand-enter the values, that's fine. What about when I want to make changes - I'll directly update the database with sql, instead of changing the flat files, right? |
| 15:06.01 | shido6 | i dont think you can use both |
| 15:06.05 | lmadsen | basically the format of the static realtime tables is a column for each option |
| 15:06.10 | mkl1525 | Hi, tried to use ChanIsAvail to check if an agent is logged in but that seems not to work. should this work or is there any other way to check if an agent is logged in or not? |
| 15:06.11 | shido6 | unless you do a switch |
| 15:06.14 | *** join/#asterisk nephfl (n=none@wsip-68-110-130-57.ga.at.cox.net) |
| 15:06.17 | lmadsen | and I think 'name' is the [name] part |
| 15:06.27 | fiXXXerMet | okay |
| 15:06.42 | lmadsen | fiXXXerMet: yes, you'll update the DB, then reload the appropriate module (as if it were a flatfile) |
| 15:07.01 | nephfl | hello, I'm having some trouble with digitmaps and dialing back from caller ID...is there an easier way to get all this crap to work together? |
| 15:07.11 | lmadsen | mkl1525: chanisavail() will check to see if a channel is available |
| 15:07.17 | fiXXXerMet | I guess I'll end up writing a little web frontend for updating the values in the db |
| 15:07.19 | lmadsen | i.e. in SIP, if it is registered, that is all |
| 15:09.12 | mkl1525 | @lmadsen thanks thought that sip registration (I'm a phone and ready) would be similar for an agent (I'm an agent and wait) after login |
| 15:09.36 | lmadsen | no -- different things |
| 15:09.52 | lmadsen | ChanIsAvail() just checks to see if the channel would be available to accept a call |
| 15:09.56 | lmadsen | Agents are different things |
| 15:11.00 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 15:12.41 | nephfl | could you use soft keys and zap barge to make asterisk behave like a key system? |
| 15:14.08 | hmmhesays | ahh key systems |
| 15:14.48 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 15:18.20 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 15:18.27 | [TK]D-Fender | nephfl: thats what *'s 1.4 fake SLA is for.... |
| 15:19.28 | jameswf-home | I like my SLA wit a little cole in the front and a W in the back |
| 15:22.42 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 15:23.07 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 15:23.12 | ZaVoid | lol |
| 15:23.14 | ZaVoid | 370 active SIP channels |
| 15:23.17 | ZaVoid | LIES! |
| 15:23.24 | ZaVoid | DAMN asterisk lies! |
| 15:24.01 | jameswf-home | a misplaced decimal isnt a lie :) |
| 15:24.17 | jameswf-home | asterisk -rx "show warranty" |
| 15:24.27 | ZaVoid | its only off by a bout 180 channels |
| 15:24.30 | ZaVoid | lol |
| 15:24.54 | ZaVoid | <PROTECTED> |
| 15:25.09 | jameswf-home | well the specs allow up to 200 off so it is within the acceptible range |
| 15:25.35 | *** join/#asterisk florinel (n=florinel@ip66-104-156-2.z156-104-66.customer.algx.net) |
| 15:25.45 | ZaVoid | 200 off? |
| 15:25.48 | ZaVoid | oh channels haha |
| 15:26.14 | florinel | hello guys. does asterisknow beta 6 include all the addons? |
| 15:26.41 | ZaVoid | dunno bout asterisknow sorry |
| 15:26.55 | jameswf-home | I wonder is the microsoft folks raise an eyebrow when I log in ti their oem site on linux... |
| 15:26.58 | *** join/#asterisk AndyGraybeal (n=andy@node239.35.251.72.1dial.com) |
| 15:28.08 | *** join/#asterisk mikkel (n=mikkel@84.238.113.66) |
| 15:28.58 | florinel | is there a good addon for asterisk that displays stats and dialplan info? |
| 15:29.26 | hmmhesays | what exactly do you want to see? |
| 15:30.16 | mikkel | When accessing http://127.0.0.1:8088/asterisk/static/config/setup/install.html I get "Nothing to see here. Move along. Asterisk Server", anyone know what is could be ? Have done all in the README file. |
| 15:30.54 | mikkel | "make checkconfig" give all OK |
| 15:31.04 | jameswf-home | show dialplan xyz |
| 15:31.07 | jameswf-home | ummm |
| 15:32.07 | florinel | i want to see call stats and maybe server/network stats also |
| 15:32.17 | jameswf-home | step 1 you cut a hole in the box |
| 15:32.41 | hmmhesays | step 2 put your junk in the box |
| 15:32.59 | jameswf-home | step make her open the box |
| 15:33.11 | jameswf-home | ah hell |
| 15:34.39 | *** join/#asterisk inv_arp[work] (n=junya@c-76-26-24-197.hsd1.fl.comcast.net) |
| 15:34.47 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:38.48 | florinel | anyone using a third party statistics addon with asterisk? |
| 15:39.02 | jblack | you mean a cdr browser? |
| 15:39.18 | florinel | ya or adhearsion |
| 15:40.00 | jblack | I went looking for about 15 minutes. I didn't see anything blatently obvious, but I didn't exactly look carefully |
| 15:40.33 | florinel | tried cdr, but the install instructions are a bit vague |
| 15:41.04 | florinel | doesnt say if a db has to be created... i bring it up on a browser and i get 404 errors and stuff |
| 15:41.44 | florinel | there's no index.php and when i point to any other php filees for cdr firefox tries to save em instead of parsing them |
| 15:42.21 | hmmhesays | asterisk-stat-v2 is ok |
| 15:42.29 | hmmhesays | thats because your webserver is set up wrong |
| 15:42.58 | florinel | well...the asterisk gui works fine |
| 15:43.13 | florinel | im assumin it's using the same webserver |
| 15:43.14 | hmmhesays | asterisk gui doesn't use php |
| 15:43.25 | florinel | oh |
| 15:43.27 | hmmhesays | hence, your webserver is set up wrong |
| 15:44.31 | florinel | any idea on what id have to do? do i upload my asterisk-stat-v2 in //var/lib/asterisk/static-http? |
| 15:44.33 | hmmhesays | in fact doesn't asterisk-gui use its own webserver? |
| 15:44.51 | hmmhesays | if you use asterisk stat you need to install apache or something similar with php |
| 15:45.17 | florinel | interesting that this linux distro doesn't come with all that |
| 15:45.25 | hmmhesays | what distro? |
| 15:45.45 | florinel | welll it's asterisknow, i believe its built on either aslinux or centOS |
| 15:46.01 | hmmhesays | did asterisknow come with asterisk-stat-v2? |
| 15:46.08 | florinel | no... |
| 15:46.08 | fiXXXerMet | How do I manage things like variables and includes from the extensions.conf file, in a database (realtime)? |
| 15:46.13 | florinel | had to download it |
| 15:46.28 | hmmhesays | then you need to install apache or something similar if it isn't installed along with php |
| 15:46.39 | florinel | k. thanks |
| 15:46.55 | hmmhesays | yum install httpd |
| 15:47.05 | florinel | no yum in here |
| 15:47.10 | florinel | or apt-get |
| 15:47.13 | hmmhesays | its centos with no yum? |
| 15:47.18 | hmmhesays | bah install yum |
| 15:47.21 | *** join/#asterisk JaminCollins (n=jcollins@151.101.5.95) |
| 15:47.31 | florinel | i think it uses conary for installs |
| 15:47.46 | florinel | and not 100% it's centos |
| 15:47.55 | florinel | uname -a dont say anything |
| 15:49.08 | hmmhesays | ditch it |
| 15:49.24 | JaminCollins | I'm using cdr_odbc.conf to log cdr information to a MySQL DB and noticed in my testing that the first call of the day is not logged to the database table. I assume this is due to the connection being idle for longer than MySQL's timeout value (8 hours). However, is there a way to force asterisk to confirm the record was written or detect the insert failure and reconnect? |
| 15:49.28 | hmmhesays | install asterisk on cent or debian |
| 15:49.58 | hmmhesays | JaminCollins: whoa that is weird man |
| 15:50.05 | jameswf-home | Words you should never hear together: I am a linux newb and want to recompile from source |
| 15:50.27 | *** join/#asterisk pLr (n=bobo@unaffiliated/plr) |
| 15:50.46 | hmmhesays | Words you should hear together: I am a linux newb and I want to pay you to teach my the ways of the geek |
| 15:51.03 | JaminCollins | not really weird, just the insert being lost due to the connection being idle for too long... the next insert succeeds... so, it is reconnecting... but it's a problem to have a CDR insert lost |
| 15:51.49 | florinel | yea...the reason i use asterisknow is so that i have a quick install, so i can spend my time in setup |
| 15:52.00 | hmmhesays | I guess you could do a dirty hack and have a script make a connection every once in awhile |
| 15:52.06 | JaminCollins | I can increase the MySQL timeout, but that will only make the problem happen less frequently |
| 15:52.22 | hmmhesays | I would lower it to next to nothing and figure out a fix |
| 15:52.31 | hmmhesays | florinel yeah that is a bad idea |
| 15:52.43 | hmmhesays | you want to know what is going on in the background |
| 15:53.08 | jameswf-home | the real question is whats wrong iwith your server that makes it take solong to connect |
| 15:53.13 | hmmhesays | JaminCollins: in fact its probably not asterisk specific I bet you can find a similar problem on google |
| 15:53.22 | nephfl | so, there is no easy way to get all the digit maps call back from caller id to work consistently? just have to fiddle with it? |
| 15:53.24 | JaminCollins | afaics the proper fix would be having asterisk reconnect and resend the insert again |
| 15:53.26 | drmessano | Words you should never hear together: I installed Linux and I don't see it in the Start Menu |
| 15:53.48 | JaminCollins | the problem is all over google with the mysql timeout, the fix is to have your app detect and reconnect |
| 15:54.06 | jameswf-home | my linux has a start menu or umm a k menu thats in the same location |
| 15:54.09 | hmmhesays | does that problem exist with postgres? |
| 15:54.37 | JaminCollins | unless postgres doesn't close its idle connections I would think so |
| 15:54.46 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:55.07 | lirakis | drmessano: lol |
| 15:55.35 | lirakis | i wonder why on earth anyone installs X on a server |
| 15:55.38 | hmmhesays | I haven't had any problems with func_odbc or cdr_odbc and postgres |
| 15:55.45 | lirakis | .. maybe its not purely a server (shrug) |
| 15:55.55 | drmessano | Depends |
| 15:56.22 | drmessano | Is a server "A singularly tasked box for running mission critical daemons" |
| 15:56.23 | drmessano | or |
| 15:56.33 | JaminCollins | two questions... how long does the server ever go idle between calls and are you absolutely certain all calls are getting logged to the DB? |
| 15:57.09 | pLr | lirakis: i use X on servers to have pretty graphs shown so that i feel important |
| 15:57.16 | hmmhesays | I guess this server is nearly never idle |
| 15:57.19 | drmessano | "ZOMG I RUNNIN IRCD AND AKERISK AND APACHE AND WoW AND BIND AND DOPEWARS AND OH I INSTALL THAT TOO" |
| 15:57.20 | hmmhesays | for more than a couple hours |
| 15:57.32 | hmmhesays | but I'm definately going to test this now |
| 15:57.37 | drmessano | If the latter, X is probably fine |
| 15:57.42 | JaminCollins | problem will only happen with a default mysql configuration if it's idle for more than 8 hours |
| 15:57.55 | hmmhesays | JaminCollins, in the mean time why don't you just script up something and have cron connect every few hours? |
| 15:58.28 | drmessano | For some, a "Server" means "The old laptop in the closet with Ubuntu on it" |
| 15:58.29 | JaminCollins | not sure that would work... since it's not asterisk's connection |
| 15:58.40 | hmmhesays | cron a call file |
| 15:58.42 | lirakis | drmessano: dopewars.. hehe .. i remember playing that on my ti-81 calculator in highschool |
| 15:58.53 | hmmhesays | haha ti-83 here |
| 15:59.01 | drmessano | <-- Palm IIIc |
| 15:59.13 | AndyGraybeal | i always got kiilled in dopewars |
| 15:59.29 | hmmhesays | I remember when they finally cracked the ti assembly code |
| 15:59.31 | JaminCollins | hmmm, a call file connected to a hangup script... |
| 15:59.31 | AndyGraybeal | drmessano: your really too funny |
| 15:59.32 | hmmhesays | then the games got cool |
| 15:59.32 | drmessano | Best game ever.. it was good practice for my carrer |
| 15:59.37 | drmessano | Best game ever.. it was good practice for my career |
| 15:59.40 | drmessano | Eh, nm |
| 15:59.48 | hmmhesays | JaminCollins: sure |
| 15:59.54 | lirakis | hmmhesays: bill nagel = genius |
| 15:59.58 | lirakis | lol |
| 16:00.16 | drmessano | I remember "CRAP, ONE OF MY HO'S GOT SHOT" |
| 16:00.20 | drmessano | :( |
| 16:00.28 | drmessano | Tragic |
| 16:02.05 | khronos | Anybody have an Aastra 9133i they can give me the button layout on? |
| 16:03.04 | *** join/#asterisk UnixDog (n=unixdog@adsl-69-234-198-40.dsl.irvnca.pacbell.net) |
| 16:03.21 | UnixDog | 1.24.17 is a thorn |
| 16:04.19 | UnixDog | 1.4.17 that is |
| 16:05.07 | AndyGraybeal | iax is pronounced 'eeks' LOL!@ |
| 16:05.56 | nixguy | ok a simple que4stion, im trying to learn asterisk with some tutorials, and i just cant get background() to work |
| 16:06.04 | nixguy | whenever i try to enter a digit |
| 16:06.08 | nixguy | i get an invalid extension |
| 16:06.32 | nixguy | the log doesent even say i pressed a number |
| 16:06.35 | fiXXXerMet | nixguy: ~book has good examples of that |
| 16:06.40 | fiXXXerMet | ~book |
| 16:06.40 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 16:07.14 | fiXXXerMet | nixguy: Download the PDF and jump to page 127 |
| 16:07.26 | nixguy | i am on page 127 :) |
| 16:07.31 | nixguy | im doing everything right |
| 16:07.39 | nixguy | i think |
| 16:07.50 | nixguy | is the manual for 1.4 or 1.2? |
| 16:07.53 | nixguy | im using 1.2 |
| 16:08.08 | UnixDog | I thought 2ed was 1.4 |
| 16:08.12 | JaminCollins | the book is for 1.4 |
| 16:08.15 | UnixDog | 1st ed was 1.2 |
| 16:09.24 | nixguy | anyone with a link to the first edition? |
| 16:09.59 | [TK]D-Fender | nixguy: pastebin your dialplan and the CLI output of your failed attempt at verbose 10 |
| 16:10.01 | [TK]D-Fender | ~pb |
| 16:10.01 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 16:10.03 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^6 |
| 16:11.01 | nixguy | you want it all or is [incoming] enough? |
| 16:11.22 | [TK]D-Fender | nixguy: I'd better see everything thats used in your CLI attempt and anything it links to |
| 16:11.39 | JaminCollins | hmmhesays: originate Zap/pseudo application hangup |
| 16:11.47 | *** join/#asterisk macros73 (n=cs@c-24-3-246-27.hsd1.mn.comcast.net) |
| 16:11.59 | JaminCollins | that appears to do the trick |
| 16:12.23 | *** join/#asterisk jmesquita (n=jmesquit@201.7.117.114) |
| 16:13.01 | JaminCollins | momentary blip on the CDR just enough to cause it to write a record and the channel would never be used for an actual real call so we can filter it out of our reports |
| 16:13.04 | *** join/#asterisk h4lt (n=Gustavo@geness.funcitec.rct-sc.br) |
| 16:13.11 | h4lt | hello people |
| 16:13.14 | florinel | hey guys....i was reading the manual and its getting confusing. do i need both the [incoming] and the [employees] contexts? |
| 16:13.39 | JaminCollins | don't truly need either... it's all about what you want the system to do |
| 16:13.42 | lirakis | florinel: thats arbitrary |
| 16:13.45 | florinel | i just wanna setup a simple peer to peer sip call centre |
| 16:13.53 | lirakis | florinel: you can have whatever contexts you want |
| 16:13.54 | florinel | with 20 users or so |
| 16:14.04 | lirakis | florinel: uhh... then you need to read a lot more |
| 16:14.06 | lirakis | ~thebook |
| 16:14.07 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 16:14.13 | florinel | i know, but everytime i touch my gui it makes changes into thew extensions file |
| 16:14.21 | *** join/#asterisk jameswf (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 16:14.23 | lirakis | florinel: what gui? |
| 16:14.25 | florinel | and i have no idea what to include where anymore |
| 16:14.34 | florinel | the asterisk gui |
| 16:14.40 | florinel | the standard one |
| 16:15.24 | lirakis | florinel: .. yeah .. if you make changes in the gui .. it changes your config... thats the idea of the gui |
| 16:15.41 | florinel | i've read the book and i followed the samples |
| 16:16.06 | florinel | it always refers to this extension 123 which throws me off big time. users have 101, 102, 103...etc |
| 16:16.27 | UnixDog | I hate 3 digit user extensions |
| 16:16.31 | UnixDog | grrr |
| 16:16.35 | lirakis | florinel: .. re-read.. starting on pg. 119 |
| 16:16.54 | florinel | well..the manual referrs to 3 digit extensions. i actually use 2 digits |
| 16:17.51 | lirakis | florinel: you are going to go down in a giant fireball that smells like poo .. if you dont read.. b/c right now ... its clear you need to do some more legwork ... im not trying to be a jerk... but you need to grasp the basic concepts before you can move on. |
| 16:18.06 | lirakis | florinel: dont think about a call center right now.. |
| 16:18.07 | *** join/#asterisk variable_office (n=variable@cerberus.iswan.net) |
| 16:18.12 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 16:18.13 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:18.43 | florinel | hmm..ok |
| 16:18.58 | florinel | well..i got phones working...i just wanna clean up the extensions file |
| 16:19.33 | florinel | i just wasnt sure on what this 123 extension was for, thats all |
| 16:20.13 | lirakis | florinel: .. again .. thats "arbitrary" .. it is just an extension .. |
| 16:20.40 | lirakis | florinel: pastebin your extensions.conf |
| 16:20.45 | lirakis | florinel: the whole thing |
| 16:20.48 | lirakis | ~pb |
| 16:20.48 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 16:20.57 | florinel | i wish the book was a bit more consistent. up until pg 119 its talking about an internal context and after 119 about the incoming context |
| 16:21.26 | lirakis | florinel: .. you need to understand contexts !!!!! |
| 16:21.38 | florinel | lirakis: sorry i got u mad dude |
| 16:21.40 | florinel | ;) |
| 16:21.53 | florinel | do i need to download postbin for this? |
| 16:21.55 | lirakis | florinel: the names of contexts are arbitrary.. the are associated with peers/friends/users |
| 16:22.01 | variable_office | I am having problems with bad voice quality, I am trying to trace down the issue; the user is not behind nat, i am running 1.4.11; pinging the ata from the asterisk box doesn't result in high or jittery pings, about 10ms +- 2ms; any ideas on other things i should check? also, is there a good utility out there for diagnosing problems like this? or should i just keep using a combination of ping and ping -f ? |
| 16:22.05 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 16:22.16 | lirakis | florinel: did you even click one of the links? |
| 16:22.19 | florinel | lirakis: u want the file in private? |
| 16:22.19 | lirakis | ~pb |
| 16:22.20 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 16:22.23 | lirakis | omg |
| 16:22.31 | *** join/#asterisk angeldavid (n=angeldav@nelug/coreteam/pepo) |
| 16:23.28 | hmmhesays | there is an #asterisk-now channel |
| 16:23.47 | lirakis | florinel: internal is a context (that the book defines .. the name is arbitrary .. it could be [bob]) that its peers are members of... this means when a user picks up a phone.. their call will start processing in the [internal] context |
| 16:24.04 | florinel | i see |
| 16:24.57 | lirakis | florinel: [incoming] (again name doesnt matter .. could be [asdafs]) is what the book defines sets for the context of the peer that will be reciving inbound calls (from a provider) .. so inbound calls will start processing in that context |
| 16:25.06 | lirakis | florinel: now go read the book |
| 16:25.16 | lirakis | florinel: and play around on a smaller scale |
| 16:26.08 | florinel | will do |
| 16:26.18 | variable_office | i had read at one time that you needed to setup a false zaptel interface if you dont have any real zaptel interfaces so that the timings are right, do i need to do this maybe? |
| 16:26.20 | florinel | i got my extensions.conf in pastebin |
| 16:26.25 | florinel | wanna have a look at it? |
| 16:26.36 | *** join/#asterisk vetetix (n=vetetix@eclip3.ec-lille.fr) |
| 16:26.52 | lirakis | florinel: ... well you dont really have a specific question .. so unless you do.. no not really |
| 16:27.57 | florinel | well..i am wondering if things are correct in it |
| 16:28.12 | florinel | or if iam duplicating things |
| 16:28.20 | florinel | http://pastebin.com/m4daaf40e |
| 16:28.23 | florinel | have a look pls |
| 16:29.43 | florinel | my question is: my [phones] context containing all my sip users...are the includes in that correct? |
| 16:31.03 | florinel | lirakis: lemme know if u're havin a look or if i should just mind my business |
| 16:31.11 | lirakis | florinel: its pretty much an incoherent mess.. i suggest you join the #asterisk-gui channel |
| 16:31.29 | florinel | incoherent mess... hmm |
| 16:31.43 | florinel | actually i've seen uglier samples, but thanks |
| 16:31.44 | lirakis | florinel: i mean there is stuff in there you def. dont need... stuff that .. is not properly implemented (VM) |
| 16:32.04 | lirakis | florinel: i mean .. what is [phones] ?? |
| 16:32.06 | florinel | voice mail works great thos |
| 16:32.26 | florinel | in my sip.conf, each user has context=phones |
| 16:32.27 | lirakis | florinel: so .. you were planning on adding a line for every vm account you have? |
| 16:32.56 | florinel | well i use the vm macro, so a line per user would have been ok still.. |
| 16:33.51 | *** join/#asterisk `paul (n=aldee@125.252.68.68) |
| 16:34.01 | florinel | ill only have up tp 30 users |
| 16:34.02 | lirakis | florinel: its .. not good... i mean.. you dont use [incoming] at all from what i can see |
| 16:34.10 | lirakis | florinel: trust me.. its not the right way |
| 16:34.35 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 16:34.39 | *** join/#asterisk snuffie16 (n=chatzill@fw.receivia.com) |
| 16:34.42 | florinel | do u have a few tips? i mean i wanna reinstall and start fresh, but i fear that if i follow the book ill end up in the same spot |
| 16:34.47 | `paul | if a call enters a queue how can i know which agent(extension) received the call.... i need to append it to the voicelog file name |
| 16:36.01 | lirakis | florinel: voicemail(${CALLERID(num)}) |
| 16:36.19 | snuffie16 | anyone know where to get install instruction guide for 1.2? |
| 16:36.40 | lirakis | paul: queue_log .. possibly from dstchannel in cdr .. but thats not really reliable |
| 16:37.10 | *** join/#asterisk maldous (n=user@f28115.upc-f.chello.nl) |
| 16:37.11 | `paul | lirakis: how about the sip extension that answerd the call? |
| 16:37.16 | maldous | hiya. |
| 16:37.36 | lirakis | florinel: start fresh.. but the book isnt a howto implement "insert what you want here" ... its a guide .. to show you how things work.. so you can understand and implement them yourself |
| 16:38.01 | lirakis | florinel: and again.. start small ... youll shoot yourself in the foot 100 times if you try to implement a call center right now |
| 16:38.16 | *** join/#asterisk mascool (n=george@c-68-84-164-71.hsd1.mi.comcast.net) |
| 16:38.17 | maldous | does anyone know if there's a *free* iax/sip/skype solution available today? |
| 16:38.28 | florinel | lirakis: see something like exten => 11,1,Macro(voicemail,${TEST1}) cannot be replaced with what u suggested, because this is calling to a macro for the voicemail app |
| 16:38.55 | mascool | does anyone know how to fix this error: an_sip.c:8165 check_auth: Correct auth, but based on stale nonce received from |
| 16:39.04 | mascool | the phone does not register |
| 16:39.29 | florinel | lirakis: it is small, im just testing 2 softphones untill everything is good, then we'll have 20 or so users - which is still small |
| 16:40.01 | lirakis | paul: ... check eparam1 from queuelog |
| 16:40.51 | snuffie16 | is there a channel for 1.2? |
| 16:40.53 | florinel | lirakis: for a simple setup like mine...u need general, global obviously + incoming, outgoing, internal extensions. right? |
| 16:40.58 | florinel | i mean those are a must |
| 16:41.17 | florinel | lirakis: not extensions sry..i meant contexts |
| 16:41.52 | *** join/#asterisk tsearle (n=torrey@98.110-246-81.adsl-static.isp.belgacom.be) |
| 16:41.52 | lirakis | florinel: .. sigh .. exten => *97,1,VoicmailMain(${CALLERID(num)}@yourcontext) ... |
| 16:42.07 | lirakis | florinel: we are done .. go read .. |
| 16:42.52 | lirakis | florinel: and go to #asterisk-gui next time |
| 16:42.58 | mascool | anyone ? |
| 16:43.46 | *** join/#asterisk darkskiez (i=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) |
| 16:44.28 | florinel | lirakis: ok...let's make sure u dont start sweating |
| 16:44.29 | snuffie16 | can anyone steer me towards help with asterisk-1.2.9.1 ? |
| 16:44.36 | `paul | lirakis: so theres no way to append the extension in real time to the file name of the voice log? |
| 16:45.09 | florinel | lirakis: thanks for the huge efforts u've put in support in the last 10 minutes. Goes a long way |
| 16:45.52 | lirakis | `paul: are you on 1.2? |
| 16:46.28 | lirakis | mascool: post the right error.. thats a botched copy paste ... whats your network latency? |
| 16:46.54 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:47.06 | mascool | this is only happeneing to some phones, not all, behind the same nat, asterisk has public ip |
| 16:47.56 | fiXXXerMet | I am getting "No application 'MeetMe' for extension (tvicorp, 555, 1)" when I try to access my voicemail. The extension is under the [tvicorp] context and in meetme.conf I just have conf => 555 |
| 16:48.35 | lirakis | fiXXXerMet: why are you using meetme ... for vm? |
| 16:48.45 | fiXXXerMet | Oops |
| 16:48.51 | fiXXXerMet | When I try to access my conference rooms |
| 16:48.52 | *** join/#asterisk lzhang (n=lzhang@67.95.13.186) |
| 16:49.30 | lirakis | fiXXXerMet: pastebin extension.conf |
| 16:49.46 | lirakis | and meetme.conf |
| 16:49.52 | lzhang | JT: I've identified the echo issue as feedback from my handset on this "cortelco" ip phone |
| 16:51.35 | fiXXXerMet | lirakis: http://pastebin.com/m4f2aa8fc |
| 16:51.53 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 16:51.56 | fiXXXerMet | lirakis: Do I need to do MeetMe(@tvicorp) or something? |
| 16:52.06 | *** join/#asterisk zeppelin_ (n=zeppelin@ns.atendebem.com.br) |
| 16:52.58 | lirakis | fiXXXerMet: Meetme(555) |
| 16:53.41 | fiXXXerMet | Same thing. |
| 16:53.50 | fiXXXerMet | No application 'MeetMe' for extension (tvicorp, 555, 1) |
| 16:53.59 | lirakis | fiXXXerMet: Meetme(555|c) will announce the number of callers when you enter |
| 16:55.02 | lirakis | fiXXXerMet: did you compile meetme? |
| 16:55.15 | lirakis | fiXXXerMet: 1.4 doesnt do it by default |
| 16:55.29 | fiXXXerMet | That would be the problem. |
| 16:55.32 | lirakis | fiXXXerMet: type show modules |
| 16:55.35 | lirakis | at cli |
| 16:55.46 | *** join/#asterisk redback (n=kieran@82.152.56.113) |
| 16:55.51 | redback | ~book |
| 16:55.52 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 16:56.20 | maldous | does anyone know someone here who uses skype with asterisk? |
| 16:56.41 | fiXXXerMet | lirakis: Not there. Looks like I also need a "zaptel timing interface" |
| 16:57.20 | lirakis | fiXXXerMet: yeah you need ztdummy for conferences if you have no cards |
| 16:57.25 | fiXXXerMet | okay |
| 16:58.48 | lirakis | maldous: ive heard of Chanskype .. but i dont know much about it |
| 16:59.37 | jameswf | I need a res_getmeabeerwoman.so to use on my home system |
| 16:59.42 | maldous | lirakis: i'm starting to think there's no 'free' solution. |
| 16:59.48 | maldous | i see there's a bounty for it. |
| 17:00.10 | lirakis | jameswf: lol |
| 17:00.29 | *** join/#asterisk nclx (n=nclx@192.235.8.67.cfl.res.rr.com) |
| 17:01.02 | puppet | http://www.chanskype.com/ |
| 17:01.21 | *** join/#asterisk c4t3l (n=c4t3l@74.95.210.124) |
| 17:01.26 | lzhang | are there any echo cancellation plugins for sip -> asterisk -> sip calls? |
| 17:01.27 | [TK]D-Fender | ~SKYPE |
| 17:01.28 | jbot | [~skype] Skype is a free VoIP software and service using a closed client and propritary protocol. Only commercial channel drivers exist, all of which are poor implementations of convoluted hacks. In general, forget about using Skype with Asterisk... |
| 17:02.11 | lirakis | lzhang: http://www.voip-info.org/wiki/view/Asterisk+echo+cancellation |
| 17:02.12 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 17:02.12 | c4t3l | greetings, can anyone point me to a detailed explanation of the output of the "pri show span n" command |
| 17:02.27 | [TK]D-Fender | lzhang: No. |
| 17:02.30 | lzhang | lirakis: I'm looking at that page already, it seems to be all zaptel related stuff? |
| 17:03.02 | nclx | I have an asterisk server behind NAT (unfortunately no choice), it is registered via SIP to broadvoice.com, I can call internal phones no problem, if I try to call out through broadvoice it rings the phone, however when they answer neither party can hear audio, I know this is an RTP issue with NAT. What is recommended, I have set on my [sip.broadvoice.com] context in sip.conf: nat=yes externip=myexternnatip localnet=asteriskboxip, I forwarded 50 |
| 17:03.03 | lzhang | [TK]D-Fender: thanks, I guess my only recourse is to use a higher quality ip phone with less leakage? |
| 17:03.46 | [TK]D-Fender | lzhang: What do you have now? And what is on each of of this SIP -> SIP you're talking about? |
| 17:03.47 | puppet | god i dont want to write a realtime edit thing... anyone know any simple editthing for mysql where u can save queries? or wait i can use the windows thing, if i just rememebr the name |
| 17:03.49 | variable_office | what features of asterisk have a required pre-requisite of a zaptel channel? |
| 17:04.03 | [TK]D-Fender | nclx: Go read this now : |
| 17:04.04 | [TK]D-Fender | ~sipnat |
| 17:04.05 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 17:04.07 | [TK]D-Fender | ^^^^^^^^^^^^^ |
| 17:04.26 | [TK]D-Fender | variable_office: Meetme, Page, IAX2 Trunking |
| 17:04.29 | lzhang | [TK]D-Fender: 2 cortelco phones connected to an asterisk box all on the same network |
| 17:04.45 | [TK]D-Fender | lzhang: Coretelco = analog? |
| 17:04.50 | lzhang | no, they're IP phones |
| 17:05.05 | [TK]D-Fender | lzhang: Ok, then they suck. Tweak them if you can, replace them if you must |
| 17:05.06 | nclx | variable_office meetme will work, but the admin commands in the meeting will not |
| 17:05.28 | [TK]D-Fender | nclx: No, without a zaptel timing source Meetme will not work at all. |
| 17:05.29 | variable_office | ok, but sip shouldnt have a problem right? |
| 17:05.42 | c4t3l | are there any old-school telco dudes here? |
| 17:05.46 | nixguy | variable_office: zip works fine |
| 17:05.48 | [TK]D-Fender | c4t3l: plenty |
| 17:05.50 | nixguy | sip |
| 17:07.00 | c4t3l | I didnt mean to ask a newB question, I just want to learn more about what I'm seeing with the "pri show span n" commands |
| 17:07.07 | variable_office | I am having some problems with sip skipping and being overall crappy, but the network from the asterisk box to the user has no nat and the ping times are good 8-12ms; any ideas on what else I could check? |
| 17:07.19 | *** part/#asterisk florinel (n=florinel@ip66-104-156-2.z156-104-66.customer.algx.net) |
| 17:07.26 | c4t3l | I've got a telco disputing with my equipment saying that it is mis-configured |
| 17:08.13 | c4t3l | its my inderstanding that if telco is providing pri to my box then I set it on "pri_cpe" in zapata.conf |
| 17:09.15 | [TK]D-Fender | c4t3l: pastebin your zaptel & zapata |
| 17:09.26 | nclx | is it possible to specify multiple localnet's for a context? |
| 17:09.29 | c4t3l | hold plz |
| 17:09.34 | *** join/#asterisk |dennis| (n=dennis@200.32.233.84) |
| 17:10.36 | nclx | never mind I found that it is |
| 17:11.01 | [TK]D-Fender | nclx: You don't specif localnets for your PEERS (stop calling them "contexts"), this is for your * server only, and belong under [general] |
| 17:11.57 | c4t3l | [TK]D-Fender: plesae see http://pastebin.com/m170e6bd6 |
| 17:13.53 | maldous | [TK]D-Fender: thx |
| 17:14.02 | maldous | is there anyone here from digium? |
| 17:17.01 | c4t3l | Is there a good book someone could point me to to learn more about PRIs ISDN and the ilke? |
| 17:17.18 | lirakis | c4t3l: t1 survival guide |
| 17:17.23 | lirakis | c4t3l: oreilly |
| 17:17.30 | c4t3l | sweet, thanks |
| 17:18.16 | [TK]D-Fender | maldous: just ask your question |
| 17:18.24 | *** join/#asterisk Dr{Who} (n=mathewss@dev.null.nutech.com) |
| 17:19.00 | [TK]D-Fender | c4t3l: bchan=4-23 should be bchan=1-23 |
| 17:19.27 | [TK]D-Fender | c4t3l: and channel=>4-23 as channel=>1-23 |
| 17:19.39 | *** join/#asterisk Victor_Yure (n=aaa@200.166.132.131) |
| 17:20.07 | c4t3l | [TK]D-Fender: my telco requires that it be that way, they broke out their channels all screwy and it absolutely didn't work the "defualt" way |
| 17:20.40 | [TK]D-Fender | c4t3l: What signalling did they provision you for exactly? |
| 17:22.39 | c4t3l | the first four channels were broken out as single lines going to an Adtran 608 |
| 17:22.49 | pLr | does anyone have a cheap voip provider w/ callerid spoof on and unlimited minutes? |
| 17:24.03 | [TK]D-Fender | c4t3l: Waitasec... are you sure that adtran is spitting PRI back out to * and not CAS? |
| 17:24.17 | [TK]D-Fender | pLr: "Would you like fries with that, sir?" |
| 17:24.29 | Dr{Who} | easy one.. the term used to refer to a device that has sip/ethernet on one end and a fxs port on the other. |
| 17:24.32 | outtolunc | pull up to next window <G> |
| 17:24.47 | _ShrikE | Dr{Who}: ATA |
| 17:24.51 | Dr{Who} | thanks. |
| 17:25.17 | lirakis | hmm... im monitoring some agent calls.. and i hear what sounds like an AC hum .. and some "blipping" .. like little jitters here and there |
| 17:25.25 | c4t3l | [TK]D-Fender: the adtran 608 is splitting those first 4 to a 66 block the remaining to go to b channels |
| 17:26.18 | [TK]D-Fender | c4t3l: then you should not be using PRI signalling if they're all "b"'s |
| 17:26.33 | c4t3l | well there is a D on 24 |
| 17:28.37 | [TK]D-Fender | c4t3l: http://pastebin.com/m5ef8fcbc |
| 17:28.48 | [TK]D-Fender | c4t3l: not if they're all "b"'s' it isn't |
| 17:29.37 | c4t3l | hmm |
| 17:29.41 | [TK]D-Fender | c4t3l: I doubt it can split off 4 as CAS and still leave a D on 24 for the others... thats whacked... usually your 100% PRI or not. |
| 17:29.44 | *** join/#asterisk ddunavant (n=David@70-4-149-49.area3.spcsdns.net) |
| 17:30.17 | *** join/#asterisk jetlagmk2 (i=jetlag@pool-70-17-48-245.pskn.east.verizon.net) |
| 17:30.39 | c4t3l | the carrier is Logix communications out of Houston TX |
| 17:30.40 | outtolunc | i'd like to know how he got the signal back out.. i think the 608 only has a eth port and a v.35 'going out' |
| 17:30.55 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.64.193) |
| 17:31.13 | Dr{Who} | i have seen fxs refered to as station or subscriber.. what is more common? |
| 17:32.16 | outtolunc | unless he got the optional dsx-1 |
| 17:32.54 | nDuff | Dr{Who}: I've seen station more often. |
| 17:33.12 | c4t3l | there are 2 t-1s inbound, one for data the other for voice, but the splitout is very uncommon eh? |
| 17:33.43 | outtolunc | what is 'uncommon' is the *multiprotocol* pass through |
| 17:34.22 | outtolunc | meaning, once the 608 takes a PRI and breaks it out to B's it will then pass B channels (E&M) to the next deviec |
| 17:35.22 | Dr{Who} | thanks. |
| 17:36.01 | lirakis | does any one know of a "simple" jitter test tool |
| 17:36.05 | [TK]D-Fender | c4t3l: I've never seen one that takes in PRI/24, spitting out CAS/4 + PRI/19 |
| 17:36.43 | [TK]D-Fender | outtolunc: Nobody said E&M did they? Or is this your more intimate knowledge of this aprticular model? |
| 17:37.24 | outtolunc | just my dealing with 'other' multiprotocol channel banks |
| 17:37.28 | outtolunc | not this one |
| 17:43.54 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 17:45.36 | *** join/#asterisk drmessano-LT (n=nonya@207.230.140.240) |
| 17:46.09 | jameswf | em is ass |
| 17:46.42 | jameswf | em(o) the signalling method that should kill it's self |
| 17:47.48 | outtolunc | on the optional DSX module the choices (i see) are ESF or SF, and B8ZS or AMI, nothing for determining timing on that one.. so i would try setting the asterisk box to pri_net |
| 17:48.14 | outtolunc | (if using the ESF/B8ZS obviously) |
| 17:48.44 | outtolunc | if that doesn't work, just use SF/AMI and LS and set asterisk to E&M |
| 17:49.39 | jameswf | <cough> our channelbanks auto configure </cough> :) |
| 17:50.16 | outtolunc | see appendix D |
| 17:50.26 | outtolunc | http://www.adtran.com/adtranpx/Doc/0/SHBEMB7S9G44779QH37TH4M6B8/61200624L1-1B.pdf |
| 17:51.42 | *** join/#asterisk vrtk (n=bb@189.21.178.20) |
| 17:53.47 | UnixDog | I keep getting bus error core dumbs with asterisk 1.4.17 |
| 17:55.20 | kusznir | Is there some where I can go get/look at the asterisk sample config files? My installation didn't come with them. |
| 17:55.41 | nclx | Well I found my NAT problem, asterisk is set to use RTP on 10000:20000UDP, which was allowed through my firewall but apparently broadvoice is requesting RTP on ports up in the 28000 range, so I had to allow UDP through 10000:31000 through my firewall and now it works, thanks for the links [TK]D-Fender |
| 17:56.06 | nDuff | UnixDog: turn on core dumps. recompile with debug symbols if necessary. use gdb to get a stack trace. where's it happening? |
| 17:56.59 | UnixDog | well it gets to the cli sits there for 5 min the core dumbs with bus error |
| 17:57.11 | UnixDog | but ok will recompile |
| 17:59.13 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 18:00.07 | nDuff | UnixDog: well -- if you're getting a core dump, you can use that to do a backtrace in gdb already. Won't do much good if your build doesn't have debug symbols enabled, but can't hurt to check if you don't know if they're on or not. |
| 18:03.25 | lirakis | kusznir: make samples |
| 18:04.10 | lirakis | kusznir: pg 48 of |
| 18:04.13 | lirakis | ~the book |
| 18:04.30 | lirakis | ~book |
| 18:04.30 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 18:04.44 | drmessano-LT | JAHHH!!! KDE 4.0 was released.. time to go upgrade my PBX |
| 18:04.48 | drmessano-LT | j/k |
| 18:04.52 | lirakis | ha ha |
| 18:05.28 | mvanbaak | drmessano-LT: while you're at it, please fix pbx_kdeconsole.c |
| 18:05.35 | lirakis | i wish ubuntu standardized on kde instead of gnome .. but (shrug) |
| 18:05.44 | drmessano-LT | Kubuntu lol |
| 18:05.55 | *** join/#asterisk murdmath (n=vircuser@mail.kimballequipment.com) |
| 18:05.56 | lirakis | then we could all run ubuntu kde pbx's |
| 18:06.03 | drmessano-LT | YAY |
| 18:06.27 | drmessano-LT | Ubuntu + Asterisk + KDE = VoIP 2.0 |
| 18:06.33 | lirakis | ha ha |
| 18:06.35 | jameswf | I wonder when my distro will release it... the rc kinda sucked kde4 is verry vistaish |
| 18:07.18 | drmessano-LT | Ubuntu + Asterisk + KDE + Ron Paul on a Mac = Digg frontpage |
| 18:07.23 | lirakis | jameswf: your distro ? .. like ... "my island" ? |
| 18:07.49 | jameswf | like pen island |
| 18:07.52 | jameswf | :) |
| 18:08.25 | lirakis | kastarbu he he |
| 18:08.41 | drmessano-LT | muja muja to you too |
| 18:08.54 | lirakis | uhh.. i think its time for lunch soon.. my brain is obviously out of fuel for which to function good |
| 18:09.43 | jameswf | I keep trying to upgrade to internet 2.0 but it says some crap about dont be a moron |
| 18:10.56 | lirakis | heh heh |
| 18:11.20 | [TK]D-Fender | kusznir: "your installation"? How did * get installed on your system? |
| 18:11.35 | *** join/#asterisk mintee (n=mintee@72-165-177-90.dia.static.qwest.net) |
| 18:11.49 | mintee | what's a good SIP service provider for home use in the US? |
| 18:12.22 | hmmhesays | copper |
| 18:12.28 | hmmhesays | itsp's suck |
| 18:13.10 | mintee | itsp? |
| 18:26.31 | hmmhesays | internet telephony service provider |
| 18:26.35 | mocker | I use Vitelity and have had good lukc. |
| 18:26.50 | pLr | teliax is great |
| 18:26.51 | hmmhesays | mocker: I've had problems with there 1+ terminations this week |
| 18:27.35 | mocker | hmmhesays: I've been out of town this week, that's not good though. |
| 18:27.46 | mocker | Did teliax ever decide to offer e911? |
| 18:27.47 | hmmhesays | I changed to voipjet this week |
| 18:27.54 | hmmhesays | for my outbound routing |
| 18:28.24 | hmmhesays | I'm actually working on a script to dynamically change my providers based on a few routing metrics |
| 18:32.05 | lirakis | hmmhesays: like what kind of metrics? |
| 18:32.16 | *** part/#asterisk ddunavant (n=David@70-4-149-49.area3.spcsdns.net) |
| 18:35.08 | *** join/#asterisk ddunavant (n=David@70-4-149-49.area3.spcsdns.net) |
| 18:35.34 | jameswf | ~voipsex |
| 18:36.54 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 18:37.37 | jameswf | ~voipsex |
| 18:37.38 | jbot | i heard voipsex is NOTICE[6151]: File chan_sip.c, Line 5074 (handle_request):core dump |
| 18:38.06 | lirakis | oookay |
| 18:38.10 | Qwell | that's...wow |
| 18:38.11 | Qwell | just wow |
| 18:38.46 | mintee | pLr, it's interesting that teliax can't even create a website that works.. I donno how i feel about them being my itsp. |
| 18:40.37 | fiXXXerMet | After I install ztdummy, need I recompile asterisk? |
| 18:42.13 | [TK]D-Fender | fiXXXerMet: yes |
| 18:42.30 | fiXXXerMet | thanks. |
| 18:44.31 | *** join/#asterisk bhima (n=gopi@72.19.4.237) |
| 18:46.20 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 18:46.23 | bhima | Is it worth getting echo cancellation on a T1 card? Can I do that in software if I don't have a heavily loaded box? |
| 18:46.41 | Qwell | bhima: it's often a good idea to get hwec, if you can |
| 18:47.01 | Qwell | there are swec's in zaptel though |
| 18:48.10 | bhima | Qwell: Are they poorer quality, or more latency, or do they just suck a lot of CPU...? |
| 18:48.31 | Qwell | not sure about latency, but the first and third are pretty true |
| 18:49.05 | drmessano-LT | Its also about where youre applying it in the chain too |
| 18:49.18 | drmessano-LT | May as well nip it at the source then to try to DSP it out later |
| 18:49.50 | bhima | Ok, thanks. In the past I've seen hardware accelerators that got obsolete quickly as computer CPUs got faster. Hardware compression for Stacker was only worth using for a year or so... |
| 18:51.01 | *** join/#asterisk fiXXXerMet (n=kjohnson@dsl092-156-002.wdc2.dsl.speakeasy.net) |
| 18:52.09 | *** part/#asterisk maldous (n=user@f28115.upc-f.chello.nl) |
| 18:52.29 | drmessano-LT | Pop on, pop off echo cancellation on a card would rock |
| 18:52.41 | drmessano-LT | Which I am sure someone is gonna tell me already exists |
| 18:55.18 | bhima | Where does the echoing on a PRI come from? |
| 18:55.39 | *** join/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 18:55.56 | *** part/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 18:57.17 | *** join/#asterisk Winkie (n=urmom@general-kt-195.t-mobile.co.uk) |
| 19:02.54 | hmmhesays | from a pri plugged into a voip gateway terminated to an analog pstn line |
| 19:06.49 | bhima | I'm using Ethernet phones and going to a CISCO IAD device that is generating the PRI locally. |
| 19:07.07 | JaminCollins | that's enough to get echo |
| 19:08.10 | bhima | Where in that loop is the echo coming from, then...? |
| 19:09.14 | JaminCollins | probably happening at the conversion between the PSTN and VoIP networks |
| 19:09.27 | JaminCollins | but that's just speculation on my behalf... it is the likely point though |
| 19:11.35 | bhima | Right now AIUI the signal path is VoIP ethernet phone via SIP to Asterisk, via PRI to Cisco IAD2431, via SIP to something at NuVox HQ, via PRI to PSTN. |
| 19:11.47 | bhima | I need to find out if NuVox will just let me do SIP straight to them. |
| 19:12.31 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 19:12.36 | bhima | I poked around on the Cisco box but wasn't able to extract the SIP credentials from it. |
| 19:13.19 | *** part/#asterisk snuffie16 (n=chatzill@fw.receivia.com) |
| 19:13.26 | *** join/#asterisk drmessano-LT (n=nonya@207.230.140.240) |
| 19:17.39 | *** join/#asterisk ZX81 (n=ZX81@202.20.97.211) |
| 19:20.53 | drmessano-LT | hmm |
| 19:20.55 | _ShrikE | bhima: NO |
| 19:25.16 | *** join/#asterisk jeally-bean (i=user@63-76-119-176.directcom.com) |
| 19:27.33 | *** join/#asterisk UCFmethod (n=UCFmetho@office.eyestreet.com) |
| 19:27.39 | UCFmethod | howdy |
| 19:30.26 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 19:30.50 | *** join/#asterisk FlatFoot (n=chatzill@80.88.218.4) |
| 19:33.05 | *** join/#asterisk metfan2007 (n=metfan20@fw.grupositel.com.mx) |
| 19:33.24 | metfan2007 | Hi all, do you know if some h323 driver has transfer capabilities? |
| 19:33.58 | UCFmethod | I am getting the following error when trying to run 'make menuconfig' for zaptel-1.4.7.1 "Install ncurses to use the menu interface!" I have the following 2 rpm's installed ncurses-5.5-24.20060715 and ncurses-devel-5.5-24.20060715 on a CentOS 5.1 box |
| 19:34.12 | UCFmethod | what gives? |
| 19:35.24 | [TK]D-Fender | UCFmethod: You need libncurses |
| 19:35.34 | UCFmethod | thanks |
| 19:35.51 | *** join/#asterisk _jwd_ (n=jwd@pool-71-126-226-77.bstnma.east.verizon.net) |
| 19:36.04 | _jwd_ | hello there |
| 19:36.49 | UCFmethod | [TK]D-Fender: curious why asterisk doesnt seem to need the same package to run 'make menuconfig' |
| 19:36.49 | *** join/#asterisk G-nerd (n=AskMe@dhcp-077-249-041-129.chello.nl) |
| 19:36.49 | _jwd_ | hey UCF |
| 19:36.49 | G-nerd | Hello my best friends! |
| 19:36.57 | _jwd_ | I am a little confused on something |
| 19:36.58 | UCFmethod | _jwd_: hey |
| 19:37.22 | metfan2007 | I'm working right know in a Avaya H323 Asterisk integration, I can start calls from Avaya phone to H323 trunk, but Avaya sends messages about "codec mismatch", and Asterisk does not show any error in CLI, any idea? I just enable all codecs in ooh323.conf "allow=all" |
| 19:37.40 | _jwd_ | you need to have a T1 to create a voip phone setup |
| 19:37.43 | G-nerd | jwd, you are confused? why? you should be happy |
| 19:37.46 | _jwd_ | and if so you need to have the card |
| 19:37.54 | G-nerd | iz there a girl involved? |
| 19:37.57 | *** join/#asterisk K1W2U3 (n=K1W2U3@unaffiliated/K1W2U3) |
| 19:38.05 | _jwd_ | I am happy, I have been reading and reading and can't figure out my answer |
| 19:38.17 | G-nerd | what is the question? |
| 19:38.17 | _jwd_ | which in turn means I am asking the wrong question or a lame one |
| 19:38.29 | G-nerd | I just got in here |
| 19:38.44 | _jwd_ | in every case for asterisk setup you need a T1 card? |
| 19:39.04 | nDuff | _jwd_: no. |
| 19:39.12 | _jwd_ | okay so when would you |
| 19:39.15 | _jwd_ | and when wouldn't you |
| 19:39.22 | nDuff | _jwd_: if you're getting a PRI from the phone company, you need a phone card. |
| 19:39.23 | G-nerd | well I am not familiar with that card, but it depends with what you want |
| 19:39.40 | nDuff | s/phone card/t1 card/ |
| 19:39.44 | outtolunc | in every other case that doesn't require t1 connectivity to the PSTN or some other PBX/chanbank |
| 19:39.46 | nDuff | _jwd_: if you're connecting to most channel banks, you need a t1 card |
| 19:40.11 | nDuff | _jwd_: otherwise, if you're doing all VoIP or connecting to POTS through equipment that does SIP, you don't need one. |
| 19:40.24 | G-nerd | a PRI is like ISDN PRI? right? |
| 19:40.28 | _jwd_ | so to connect to a POTS channel bank you would need the card |
| 19:40.28 | *** join/#asterisk thinko (i=jdoe6alp@smaug.rackdragon.com) |
| 19:40.33 | _jwd_ | yes G-Nerd |
| 19:40.40 | UCFmethod | [TK]D-Fender: make menuselect solved it... go figure |
| 19:40.53 | nDuff | _jwd_: most channel banks, yes. There are some that are USB, but I don't think I'd trust them. |
| 19:41.19 | G-nerd | well if you have such a ISDN connection, than there is no other choise, unless you switch to another type of interface/connection |
| 19:41.35 | _jwd_ | so technically 1 FXS card and 1 FXO card |
| 19:41.50 | _jwd_ | would be suffucient using say Comcast Business |
| 19:41.58 | _jwd_ | no t1 |
| 19:42.14 | nDuff | _jwd_: I'd just buy a SPA-2100 or such in that case, personally. |
| 19:42.27 | nDuff | _jwd_: not as reliable faxing as using a PCI device, but much more convenient. |
| 19:43.01 | G-nerd | anyway, the point is, you need something to convert T1 to a connection which Asterisk can use on a pc |
| 19:43.29 | _jwd_ | you need the t1 card if you got a t1 from the phone company |
| 19:43.35 | _jwd_ | Comcast = Cable company |
| 19:43.39 | _jwd_ | not a t1 |
| 19:44.06 | _jwd_ | I guess my main question would be what is the reason for the t1 card in general. what does it do |
| 19:44.17 | G-nerd | Asterisk is just software (well it is more than just hahaha), so there are several adapters/converters to connect different type of telephoneconnections to the pc |
| 19:44.27 | FlatFoot | ~t1 |
| 19:44.28 | jbot | rumour has it, t1 is two pairs of copper wire that carry data at a rate of 1.544 Mbps. T1 lines are used to carry 24 DS-0 signals (i.e. 24 telephone conversations) or 1.536 Mbps of data. For more information see http://www.stromcarlson.com/docs/basics/t1svcfund.pdf |
| 19:44.50 | nDuff | _jwd_: over a PRI, you have 23/24 lines of channelized, time-division-multiplexed voice. It's not ethernet, so your ethernet card can't speak it. |
| 19:45.10 | nDuff | _jwd_: ...does that answer your question? |
| 19:45.22 | _jwd_ | getting there... |
| 19:45.41 | G-nerd | jwd, when I ask such questions, a lot of guys told me to read the book. They think I am lazy, but that is not true :( |
| 19:45.41 | *** join/#asterisk drmessano-LT (n=nonya@207.230.140.240) |
| 19:45.55 | drmessano-LT | EGAD |
| 19:46.21 | *** part/#asterisk jeally-bean (i=user@63-76-119-176.directcom.com) |
| 19:46.23 | _jwd_ | no I have been reading too, but I know what you are saying |
| 19:46.25 | G-nerd | nDuff is the Master telephone dude |
| 19:46.49 | G-nerd | nDuff, are those only TDM? |
| 19:47.00 | J4k3 | a PRI, under normal circumstances, is a T1. |
| 19:47.02 | G-nerd | no FDM too? |
| 19:47.11 | _jwd_ | thank you VERY much |
| 19:47.32 | drmessano-LT | hmm |
| 19:47.58 | G-nerd | like cablemodem, it modulates on a certain frequency and maybe in that particular frequency there is also TDM |
| 19:48.20 | G-nerd | because we get also TV from the same cable |
| 19:48.58 | J4k3 | and that simply doesn't happen with a PRI. |
| 19:49.37 | J4k3 | also, telephone twisted pair simply doesn't have that much headroom. all phone wire is good for is inter-pair crosstalk :P |
| 19:49.42 | G-nerd | really? maybe because it hasn't the properties for it |
| 19:50.05 | J4k3 | and because you're dealing with the phone network |
| 19:50.20 | J4k3 | pretty much every telco circuit out there was designed around the same time the transistor was invented. |
| 19:50.23 | drmessano-LT | You're getting a PRI from comcast cable? |
| 19:50.51 | G-nerd | yeah |
| 19:50.57 | G-nerd | no not me |
| 19:50.59 | drmessano-LT | Hmm |
| 19:51.13 | J4k3 | comcast can deliver a PRI. They're just another lamer-ass clec. |
| 19:51.16 | _jwd_ | so to use 4 lines I would need 1 FXS and 2 FXO cards |
| 19:51.19 | G-nerd | PRI comes from the classic telepfhone cable |
| 19:51.25 | drmessano-LT | Ok |
| 19:51.28 | drmessano-LT | I was gonna say |
| 19:51.31 | drmessano-LT | As a CLEC, yes |
| 19:51.38 | drmessano-LT | Over their cable, HAH |
| 19:51.45 | J4k3 | haha yeah |
| 19:52.07 | J4k3 | you can barely shove a couple decent voip calls over a docsis plant. I can't imagine trying to emulate a PRI over one. |
| 19:52.10 | J4k3 | that'd be one fucked up PRI. |
| 19:52.28 | drmessano-LT | Knology got as a BRI for an event we did.. They basically called AT&T up and sent us a bill for the cost + 10% |
| 19:52.31 | drmessano-LT | us* |
| 19:52.45 | drmessano-LT | A DOCPRI |
| 19:52.48 | drmessano-LT | That sounds HOT |
| 19:52.50 | J4k3 | haha |
| 19:52.57 | G-nerd | no, from the cable we have analog and digital tv, internet (LAN about 8 Mb/s) and maybe other stuff |
| 19:53.14 | *** join/#asterisk gardo (n=gardo@121.97.198.127) |
| 19:53.15 | J4k3 | pft |
| 19:53.17 | J4k3 | 'lan' my ass. |
| 19:53.21 | drmessano-LT | 23 lines of dropouts, and a somewhat working D channel |
| 19:53.21 | J4k3 | thats docsis, and it sucks balls |
| 19:53.39 | drmessano-LT | Comcast sucks balls |
| 19:53.45 | drmessano-LT | 8MB on their best day |
| 19:53.50 | drmessano-LT | 12 at night |
| 19:53.55 | drmessano-LT | Usually 3 during the day |
| 19:54.06 | drmessano-LT | I get the burst at 3am |
| 19:54.10 | G-nerd | anyway, but it is possible ;P |
| 19:54.13 | drmessano-LT | "YAY, Speedboost" |
| 19:54.16 | drmessano-LT | Uh no |
| 19:54.25 | drmessano-LT | They can order a PRI for you |
| 19:54.28 | drmessano-LT | But not ever DOCSIS |
| 19:54.38 | drmessano-LT | They can order you a hamburger too |
| 19:54.41 | drmessano-LT | Doesnt mean they make it |
| 19:55.03 | FlatFoot | lasagne for me please |
| 19:55.11 | G-nerd | anyway I am from holland, so I am not familiar with those companies |
| 19:55.24 | drmessano-LT | I hope to <insert name of preferred deity here> they dont install the PRI |
| 19:55.27 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 19:55.34 | drmessano-LT | They have whoever supplies it handle it |
| 19:55.53 | G-nerd | french fries? we got dutch fries |
| 19:55.55 | drmessano-LT | Coax connectors for RG-6 will NOT fit in an RJ-45 socket |
| 19:56.02 | Qwell | drmessano: push harder |
| 19:56.09 | G-nerd | with mayo, ketchup and unions |
| 19:56.11 | drmessano-LT | Exactly |
| 19:56.19 | *** part/#asterisk mintee (n=mintee@72-165-177-90.dia.static.qwest.net) |
| 19:56.22 | drmessano-LT | Wire Nut |
| 19:56.53 | drmessano-LT | God, I wouldnt want Comcast ordering ANY ACRONYM for me |
| 19:57.15 | J4k3 | I will say my T1s have more operating voltage (~300V) than my old Time Warner cable drop had (120V influence... you couldn't touch the damned coax without getting the shit whacked out of you) |
| 19:57.20 | J4k3 | ;) |
| 19:57.29 | drmessano-LT | LOL |
| 19:57.39 | J4k3 | that TW plant is now owned by comcrap |
| 19:57.41 | J4k3 | go figure. |
| 19:57.52 | drmessano-LT | The good news: |
| 19:58.09 | drmessano-LT | If youre using Comcast.. no more having to QoS BitTorrent traffic |
| 19:58.11 | drmessano-LT | THEY DO IT FOR YOU |
| 19:58.16 | J4k3 | HAHAHA |
| 19:58.16 | J4k3 | owned. |
| 19:58.29 | J4k3 | its comcastic! |
| 19:58.36 | J4k3 | comcastic craptastic |
| 19:58.39 | drmessano-LT | "Yet another service we provide free of charge. Thank you for using comcast" |
| 19:59.16 | drmessano-LT | Crap, I need to update my Comcast supplied McAfee Viruscan |
| 19:59.22 | drmessano-LT | :( |
| 19:59.23 | J4k3 | haha |
| 19:59.44 | J4k3 | mcafee... I think you'd be better off without a virus scanner at all |
| 19:59.54 | drmessano-LT | I'm glad they take network security seriously. "We don't support using a router, but here's some McAfee for ya" |
| 19:59.56 | FlatFoot | mccrappy |
| 20:00.12 | drmessano-LT | Talking about hitting a guy when he's down |
| 20:00.16 | J4k3 | mcafee = norton's boyfriend's name. |
| 20:00.43 | drmessano-LT | "Sir, do you have a virus on your PC?" "No" "Do you have 400,000 viruses on your PC?" "Yes, very yes :(" |
| 20:00.50 | FlatFoot | J4k3: what's the preffered antivirus then ? |
| 20:00.57 | drmessano-LT | Linux |
| 20:01.04 | FlatFoot | lol |
| 20:01.19 | J4k3 | carefulness. |
| 20:01.33 | outtolunc | a shack in the mountains <G> |
| 20:01.37 | drmessano-LT | Symantec is ok.. corporate, not Notin' Antivirus 2009 Premiere Pro Premium |
| 20:01.40 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 20:01.43 | J4k3 | no software can protect the PC of a dumbass user. |
| 20:01.45 | FlatFoot | anyone use AVG ? |
| 20:01.58 | J4k3 | I use AVG on PC's that people think they need an AV on |
| 20:02.01 | J4k3 | the price is right. |
| 20:02.09 | drmessano-LT | Norton AV "BUT I DIDNT INSTALL A DAMN FIREWALL!!!???!!!" |
| 20:02.10 | *** join/#asterisk nshm (n=shmyrev@ppp83-237-254-29.pppoe.mtu-net.ru) |
| 20:02.16 | drmessano-LT | "Oh" |
| 20:02.25 | J4k3 | everyone else... I'm pushing the 'if the user keeps getting the PC loaded with virii, its the user - not the OS" |
| 20:02.34 | FlatFoot | we as a firm moved to avg cos Norton messed up all install's of SAGE |
| 20:02.40 | nshm | hey all |
| 20:02.49 | drmessano-LT | AV helps when I got to warez...... SECURITY websites |
| 20:02.53 | nshm | I wonder if it's possible to install * in a custom prefix |
| 20:03.03 | drmessano-LT | go* |
| 20:03.05 | nshm | configure --prefix seems to be broken :( |
| 20:04.15 | J4k3 | drmessano-LT: warez sites suck, get some usenet. |
| 20:05.14 | *** join/#asterisk ZenBSDi (n=zenbsdi@unaffiliated/ZenBSDi) |
| 20:05.17 | J4k3 | on the windows platform, I've noticed that you basically need 512MB ram and a whole spare CPU core to not notice the AV slowing you down. |
| 20:05.25 | J4k3 | this laptop... loading an AV makes it pathetically slow. |
| 20:05.33 | J4k3 | without an AV, everything is quite perky. |
| 20:05.39 | drmessano-LT | I run Symantec corp and dsiable most of the BS |
| 20:05.45 | G-nerd | hi guys, do you know a good opensource softphone, I'm planning to get Asterisk running for the first time |
| 20:05.46 | drmessano-LT | Like the 99% CPU startup scan |
| 20:05.58 | ZenBSDi | G-nerd, for windows or linux? |
| 20:06.05 | FlatFoot | J4k3: AVG onb my laptop which is not much of a machine runs quite well |
| 20:06.20 | G-nerd | windose, because I have two laptops using windose |
| 20:06.30 | drmessano-LT | X-Lite |
| 20:06.36 | G-nerd | X-Lite? |
| 20:06.37 | J4k3 | FlatFoot: avg is pretty high overhead. I think part of it is related to HDD speed. |
| 20:06.41 | drmessano-LT | Yes |
| 20:06.49 | ZenBSDi | X-Lite is great g-nerd |
| 20:06.49 | J4k3 | if you've got a fast cpu and a 4200 rpm HD, you'll never notice the AV slowing you down. |
| 20:06.49 | G-nerd | allrighty |
| 20:06.55 | G-nerd | ok thnx guys |
| 20:07.04 | ZenBSDi | yup |
| 20:07.07 | FlatFoot | J4k3: on scan it slows but otherwise i don't notice it |
| 20:07.14 | J4k3 | I've got a brand new 7200 rpm hd connected to a slow-ish machine (P-M (not P4M) 1.7) |
| 20:07.18 | *** join/#asterisk rantsh (n=rantsh@190.36.185.139) |
| 20:07.27 | drmessano-LT | People can help you if you fuck up X-lIte, not many people know what HappyClownPhone 0.99Beta is |
| 20:07.31 | rantsh | Hello people |
| 20:07.34 | drmessano-LT | But they soon WILL... Muhahahaha |
| 20:07.43 | rantsh | ~agent |
| 20:07.44 | jbot | Newsflex has a free-agent like interface, but without requiring Gnome or KDE, you can find it at freshmeat |
| 20:08.05 | rantsh | ~agi |
| 20:08.05 | jbot | well, agi is the Asterisk Gateway Interface... similar to CGI for web applications AGI lets you script call control and access databases using your favorite language. AGI wrappers are available for Python (pyst), Perl (astperl?) and other languages |
| 20:08.24 | J4k3 | but, I also use radio mobile which chews through like 6GB of topo info every time I generate a map |
| 20:08.31 | J4k3 | and AVG insists on scanning that crap |
| 20:08.39 | J4k3 | oh well, I've been antivirus free for almost 6 months now |
| 20:08.54 | FlatFoot | J4k3: you do radio links then ? |
| 20:08.56 | J4k3 | I'll load AVG for a few hours, scan, take it back off... no problems. |
| 20:09.00 | *** part/#asterisk nshm (n=shmyrev@ppp83-237-254-29.pppoe.mtu-net.ru) |
| 20:09.01 | rantsh | anyone knows how I can get to do something like $agi->exec("show agents"); in an agi script? |
| 20:09.13 | J4k3 | FlatFoot: yeah, I do some rural WISPing for my real job. |
| 20:09.21 | drmessano-LT | Radio Mobile is fun with SAV too |
| 20:09.25 | FlatFoot | J4k3: what kit ? |
| 20:09.30 | *** join/#asterisk a1fa (n=a1fa@unaffiliated/a1fa) |
| 20:09.38 | a1fa | never do any business with valcom |
| 20:09.40 | a1fa | what a shady company |
| 20:09.44 | J4k3 | FlatFoot: 802.11 based exclusively. using ubiquiti gear at 900 mhz. |
| 20:09.57 | a1fa | they are advertising their products as sip capable... with some unknown firmware revision |
| 20:10.02 | G-nerd | you guys are using X-Lite too with asterisk? |
| 20:10.03 | a1fa | no documentation on the website... |
| 20:10.10 | a1fa | no firmware updates |
| 20:10.13 | drmessano-LT | "SIP Cabable" |
| 20:10.16 | a1fa | support is shady too |
| 20:10.19 | drmessano-LT | MY toaster is SIP CAPABLE |
| 20:10.20 | FlatFoot | J4k3: i use SkyPilot , Alvarion , OsBridge for most of my day job ( * for the rest ) |
| 20:10.27 | G-nerd | drmessano |
| 20:10.31 | a1fa | drmessano: noo. as in they do SIP |
| 20:10.31 | ZenBSDi | rantsh, what language are you scriptings with? |
| 20:10.33 | drmessano-LT | Yes |
| 20:10.40 | rantsh | ZenBSDi, perl |
| 20:10.58 | ZenBSDi | mmm.. I'm a asterisk-java user myself =p |
| 20:11.05 | drmessano-LT | Sorry, I dont like marketing terms |
| 20:11.16 | FlatFoot | J4k3: 5.4Ghz , 5.8Ghz only |
| 20:11.23 | a1fa | "SIP Wall Speakers" |
| 20:11.28 | drmessano-LT | Sip capable = It could definitely do SIP if we let it |
| 20:11.31 | a1fa | thats what they advertise their product as |
| 20:11.34 | rantsh | ZenBSDi, but agi should be about the same... have you been able to do something like that? |
| 20:11.40 | a1fa | yeah |
| 20:11.47 | a1fa | no.. they advertise it as "SIP" |
| 20:12.00 | drmessano-LT | Sip compatible = A SIP product can work with this.. not necessarily via SIP |
| 20:12.01 | a1fa | maybe it stands for Shitty IP Multicasting |
| 20:12.02 | ZenBSDi | show agents? haven't been worried about that yet.. I wanted to get credit card processing and database recording going |
| 20:12.04 | a1fa | no no.. |
| 20:12.12 | a1fa | drmessano: they advertise it as SIP |
| 20:12.17 | a1fa | SIP SPEAKER |
| 20:12.27 | drmessano-LT | SIP Aware: It has a SIP menu |
| 20:12.29 | ZenBSDi | but I'm hitting the asterisk-java docs now heh |
| 20:12.35 | a1fa | yeah |
| 20:12.37 | a1fa | stupid fucks |
| 20:12.39 | rantsh | hehe |
| 20:12.39 | a1fa | avoid them |
| 20:12.50 | a1fa | also avoid Cybergear |
| 20:12.52 | drmessano-LT | SIP compliant: Not SIP, but won't mess with it either |
| 20:13.09 | *** join/#asterisk Defraz (n=t0tal@fw.fuzecore.com) |
| 20:13.16 | drmessano-LT | SIP supported: We'll call you on our SIP phones, Mr Toaster owner |
| 20:13.18 | a1fa | try to avoid Cyberdata |
| 20:13.28 | a1fa | their sales team sucks |
| 20:13.47 | drmessano-LT | SIP Derived: We used SIP once, now its a closed protocol |
| 20:14.09 | rantsh | ZenBSDi, I thought of doing that too... but it concerns me that the line is not encrypted, so a man in the middle attack could steal dtmf and or conversation and eventually someone's identity |
| 20:14.17 | J4k3 | FlatFoot: ahh. too much foliage here to use 5ghz for anything except PTP |
| 20:14.17 | a1fa | drmessano: point noted |
| 20:14.30 | drmessano-LT | HA |
| 20:14.50 | drmessano-LT | IAX compatible: SIP, but if you connect it to an Asterisk box, very FTW |
| 20:14.57 | a1fa | CyberData also sucks |
| 20:14.57 | FlatFoot | J4k3: where are u then ? we mainly have hills and very flat bit's here in Kent UK |
| 20:15.07 | a1fa | drmessano: i dont know of any iax speakers |
| 20:15.18 | rantsh | me brb |
| 20:15.23 | drmessano-LT | Dude, ive moved on from your speaker issue |
| 20:15.44 | drmessano-LT | You need to ask people if it works with Asterisk.. if they dont know what Asterisk is, they dont know SIP |
| 20:15.50 | drmessano-LT | and if they dont know SIP, the box is useless |
| 20:16.07 | J4k3 | http://www.intrastar.net/~jsuter/stuff/3-31-05/ = pictures of the average terrain around here |
| 20:16.11 | J4k3 | from about 30m up. |
| 20:16.11 | a1fa | lol |
| 20:16.17 | J4k3 | (I'm in east texas) |
| 20:16.29 | drmessano-LT | They dont have to support it, or even care about Asterisk.. but if the dont know WUT N AKERISK R, run away |
| 20:16.31 | ZenBSDi | rantsh I don't follow... if the person is just using dial tones to input info and the network is secure where is the problem? |
| 20:16.38 | a1fa | drmessano: any IAX speakers? |
| 20:16.41 | J4k3 | http://www.intrastar.net/~jsuter/stuff/ (a few pics... I need to take more) |
| 20:16.49 | drmessano-LT | No, IAX is a silly protocol for clients |
| 20:16.57 | a1fa | so SIP |
| 20:17.07 | ZenBSDi | sip is the future :) |
| 20:17.16 | G-nerd | Is X_Lite really opensource?? |
| 20:17.19 | FlatFoot | J4k3: nice mast ! |
| 20:17.22 | drmessano-LT | IAX2 does exactly what its designed for, and well |
| 20:17.32 | drmessano-LT | X-Lite is free, not open source |
| 20:17.44 | drmessano-LT | If you want "OPEN SOURCE", good luck finding one that doesnt suck |
| 20:17.55 | *** join/#asterisk Maxous (n=stephen@74.7.13.242) |
| 20:18.12 | ZenBSDi | g-nerd you'll have to hit google and type in "softphone + opensource" if you want an OSS piece |
| 20:18.14 | G-nerd | maybe it is for me a challenge to develope one |
| 20:18.28 | drmessano-LT | Hmm |
| 20:18.29 | Maxous | Good afternoon all. |
| 20:18.38 | ZenBSDi | g-nerd, doubt it.. I've seen some opensource versions that were compiled for windows too |
| 20:18.38 | G-nerd | I have done that, most software are only free, but not opensource |
| 20:18.39 | drmessano-LT | Yes, the market really needs another SIP Softphone |
| 20:18.42 | drmessano-LT | Yep |
| 20:18.45 | adelas | does asterisk support multiple network cards? |
| 20:18.47 | J4k3 | I think the problem with opensource and softphone is the codecs |
| 20:18.53 | fiXXXerMet | So I installed ztdummy and the recompiled asterisk...... What next? Did it handle everything for me, or do I need to work on zapata.cnof or something? |
| 20:19.04 | J4k3 | nobody wants to listen to awful gsm, and g711 eats the bandwidth like mad |
| 20:19.04 | drmessano-LT | J4k3.. Shoosh.. we need another softphone |
| 20:19.04 | FlatFoot | J4k3: http://www.orbital.net/?l=wireless/wlcoverage thats us |
| 20:19.06 | drmessano-LT | Say it with me |
| 20:19.16 | drmessano-LT | "We need YET ANOTHER softphone" |
| 20:19.30 | G-nerd | ZenBSDi: in that case maybe I can help if there are some bugs |
| 20:19.33 | ZenBSDi | adelas, considering asterisk only cares about trunks from zapata compatible cards .. sip/iax is open for business on any amount of nics =p |
| 20:19.50 | G-nerd | but first comes first, to get Asterisk running |
| 20:20.00 | ZenBSDi | G-nerd, :) |
| 20:20.01 | drmessano-LT | Theres dozens of Free ones... and on Open Source one will be a useless project because the main developer will do all the work, and eventually tire of it. |
| 20:20.03 | adelas | well, i have 2 nics, and ony the first nic works with asterisk |
| 20:20.08 | J4k3 | FlatFoot: nice, thats actually a good chunk of turf. |
| 20:20.24 | drmessano-LT | At the end of they day you'll have a softphone |
| 20:20.27 | adelas | i can't seem to get asterisk to work with the 2nd nic |
| 20:20.27 | ZenBSDi | adelas, what protocol you using? sip or iax?' |
| 20:20.28 | drmessano-LT | Crap |
| 20:20.30 | drmessano-LT | the* |
| 20:20.30 | adelas | SIP |
| 20:20.33 | J4k3 | and thanks to my parents watching EastEnders on PBS for years, I know about where you're at (the opening theme music had a panning map of london, at least back in the 80s... haha) |
| 20:20.41 | G-nerd | drmessano, you are very optimistic |
| 20:20.44 | G-nerd | :p |
| 20:20.45 | ZenBSDi | you using softphones? |
| 20:20.51 | drmessano-LT | No, I am being sarcastic |
| 20:20.55 | rantsh | ZenBSDi, that's if the network is secure |
| 20:20.58 | FlatFoot | J4k3: what a terrible program that is |
| 20:21.00 | drmessano-LT | Thats like coding a new IRC client from scratch |
| 20:21.03 | drmessano-LT | Why oh why |
| 20:21.38 | J4k3 | FlatFoot: agreed. especially when it came on at the same time as benny hill |
| 20:21.49 | drmessano-LT | OMG |
| 20:21.50 | G-nerd | drmessano, because of the challenge, or to learn more on programming |
| 20:21.50 | rantsh | ZenBSDi, but i'm guessing your payer will connect to you through a public network right? you can still decode dtmf sounds to know what numbers someone presses |
| 20:21.53 | J4k3 | I'd wander off and watch some dirty old man vs watching some dirty ol' drama. |
| 20:22.01 | FlatFoot | J4k3: yep , now there was a funny man |
| 20:22.04 | adelas | ZenBSDi, its sip, and each nic is different subnet |
| 20:22.09 | ZenBSDi | rantsh, nope.. through a POTS =p |
| 20:22.12 | drmessano-LT | Well, good luck on that |
| 20:22.29 | G-nerd | wish me luck to get Asterisk running hahaha |
| 20:22.44 | ZenBSDi | I have a 4 line zapata card and 4 trunks configured |
| 20:22.51 | drmessano-LT | Forget asterisk, you have a softphone to code |
| 20:22.51 | G-nerd | that was meant sarcastic :p |
| 20:23.08 | drmessano-LT | Want to take over devel of HappyClownPhone 0.99? |
| 20:23.11 | ZenBSDi | so unless the caller is stupid and on a 700mhz wireless phone .. no problem :) |
| 20:23.12 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) |
| 20:23.55 | drmessano-LT | I'm pretty sure 2008 is the year of HappyClownPhone |
| 20:23.58 | drmessano-LT | 1.0 baby! |
| 20:24.25 | FlatFoot | i think i'll wait till 1.0.1.5 |
| 20:24.37 | drmessano-LT | THAT COULD BE YEARS |
| 20:24.39 | drmessano-LT | Um |
| 20:24.41 | FlatFoot | lol |
| 20:24.43 | drmessano-LT | I mean, "cool" |
| 20:25.01 | drmessano-LT | Gotta launch the PBX product too |
| 20:25.09 | drmessano-LT | HappyClownTricksPBX |
| 20:25.30 | FlatFoot | drmessano-LT: don't take on too much |
| 20:25.48 | jblack | So, I'm doing some googling about securing my voip communications. What I'm finding is not encouraging |
| 20:26.06 | ZenBSDi | adelas, in your sip.conf .. what are you using for host=? |
| 20:26.20 | drmessano-LT | We're very close to release on that too.. just have to get mod_honk out of beta |
| 20:26.54 | FlatFoot | drmessano-LT: and what have you been drinking tonight ? i think i want some :P |
| 20:27.00 | drmessano-LT | lol |
| 20:27.02 | jblack | drmessano-LT: Did they get the extra large shoes in this release? |
| 20:27.05 | drmessano-LT | No drinking, I am at work |
| 20:27.25 | drmessano-LT | mod_shoes was rewritten and is working |
| 20:27.26 | FlatFoot | drmessano-LT: soory , what time you got then ? |
| 20:27.35 | FlatFoot | sorry * |
| 20:27.35 | drmessano-LT | 3:27PM here |
| 20:27.40 | jblack | Good. It's important for this release to get caught flat footed. |
| 20:27.49 | drmessano-LT | NOT CLOSE ENOUGH TO 5:30 |
| 20:27.53 | drmessano-LT | LOL |
| 20:28.07 | FlatFoot | 8:30 and almost time for beer |
| 20:28.11 | Maxous | I just got the 3com Asterisk box. |
| 20:28.23 | a1fa | DrAk0:woot |
| 20:28.26 | a1fa | Maxous : woot |
| 20:28.29 | a1fa | how much $$$? |
| 20:28.31 | Maxous | lol |
| 20:28.40 | drmessano-LT | 3com isnt bankrupt? |
| 20:28.45 | Maxous | nah |
| 20:28.46 | Maxous | not yet |
| 20:28.48 | Maxous | :P |
| 20:28.59 | drmessano-LT | Thank god USR is still selling USB modems |
| 20:29.10 | Maxous | I think list it's about $1700 |
| 20:29.15 | a1fa | wow |
| 20:29.21 | a1fa | thats how much i payed for my new motorcycle ;P |
| 20:29.29 | jblack | man. the book's answer sucks even worse than google. |
| 20:29.34 | Maxous | nice |
| 20:29.34 | drmessano-LT | I remember when a 3C905c was a badass NIC card |
| 20:29.39 | drmessano-LT | Then I used them |
| 20:29.44 | ZenBSDi | hehe |
| 20:29.48 | drmessano-LT | and they started dying from static hits |
| 20:29.57 | drmessano-LT | Like, 3 a day during T-STorm season |
| 20:29.59 | ZenBSDi | I'm an all intel man myself |
| 20:30.02 | Maxous | the 3C number for it is 3CR10551A |
| 20:30.07 | FlatFoot | someone mention coax ? |
| 20:30.13 | jblack | "For example, a vpn between sites could be employed |
| 20:30.43 | *** join/#asterisk kamanashisroy (n=root@202.56.7.133) |
| 20:30.46 | Maxous | I have a question for yall. |
| 20:31.26 | Maxous | What is the best way to get into selling the Asterisk? |
| 20:31.28 | drmessano-LT | Linksys LNE-100, $20 each... 3C905Cs, $95 each.. the choice was easy |
| 20:31.32 | jblack | maxous: No, that dress makes you look just as skinny as when we met. |
| 20:32.10 | Maxous | jblack:Oh stop it, :-* |
| 20:32.15 | Maxous | hah |
| 20:32.16 | jblack | maxous: Perhaps an ad in the local paper, targetting small businesses. |
| 20:32.21 | drmessano-LT | "Selling the Asterisk"sounds like an 80s movie with Patrick Dempsey and Demi Moore |
| 20:32.32 | kamanashisroy | hi .. I have a question too. I want to send billing information in each second to the calling party .. for this reason I use dial with G() option .. this executes two dialplans in two channels .. But unfortunately one channel does not hangup if other channel hangs up :( .. any clue ? |
| 20:32.33 | J4k3 | drmessano-LT: yeah... I have them in my pc-based router. every time we take lightning around here I lose at least one. |
| 20:32.48 | jblack | drmessano: The sequel (Romancing the Akerisk) was even worse. |
| 20:32.53 | Maxous | jblack: I mean, getting educated on it. |
| 20:32.53 | drmessano-LT | lol |
| 20:32.59 | J4k3 | of course, last time we took lightning I ended up having things on the tower with melted ethernet transformers (the black boxes by the ethernet jack) |
| 20:33.00 | drmessano-LT | "License to Asterisk" |
| 20:33.08 | jblack | maxous: OH! That's easy. |
| 20:33.24 | jblack | You start off by learning how to install, deploy and maintain asterisk. Then... PROFIT! |
| 20:33.28 | drmessano-LT | Maxous: Ask [TK]D-Fender to teach you.. he loves it |
| 20:33.30 | J4k3 | 'Asterisk Overdrive' |
| 20:33.37 | J4k3 | a Stephen King horror story |
| 20:33.39 | drmessano-LT | 1. Install Ubuntu |
| 20:33.42 | jblack | [TK] is amazing. |
| 20:33.43 | drmessano-LT | 2. Install Asterisk |
| 20:33.45 | drmessano-LT | 3. ????? |
| 20:33.49 | jblack | PROFIT! |
| 20:33.50 | drmessano-LT | 4. Profit!! |
| 20:33.56 | J4k3 | prophets |
| 20:34.05 | Maxous | jblack: I mean from the get go. Like if you have no idea how to install linux. |
| 20:34.10 | drmessano-LT | wow |
| 20:34.14 | Maxous | jblack: Is there a beginers guide? |
| 20:34.15 | jblack | That reminds me, in this modern age, can prophets profit? |
| 20:34.16 | J4k3 | I should start a consulting company called the Profit Prophets. |
| 20:34.23 | J4k3 | oh snap, jblack |
| 20:34.24 | drmessano-LT | If you cant install linux, back away from the PBX |
| 20:34.26 | jblack | Maxous: There's (Dah Dah dah) THE BOOK |
| 20:34.47 | jblack | ~book |
| 20:34.48 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 20:35.11 | Maxous | jbot:TY TY! |
| 20:35.11 | jbot | ACTION shouts to the world in a very loud voice, "TY! is the most awesomest person ever! Extra! Extra! Read all about it." And of course, it's carried on every major network world-wide... |
| 20:35.21 | drmessano-LT | best way to learn Linux is do the opposite I did |
| 20:35.37 | jblack | Who's TY? TK's evil clone? |
| 20:35.39 | drmessano-LT | Dont install some packaged crap and keep fixing it, learning in the process |
| 20:35.42 | jblack | ... or is that KY... |
| 20:35.42 | Maxous | lol |
| 20:35.42 | ZenBSDi | best way to learn linux is to study a certification guide .. like LPI's .. |
| 20:35.51 | Maxous | TY = Thank you. |
| 20:35.56 | drmessano-LT | Install it, try to install crap, and break it |
| 20:35.59 | drmessano-LT | Break it more |
| 20:36.02 | a1fa | ZenBSDi : lol |
| 20:36.03 | Maxous | I am new to this IRC thing. |
| 20:36.04 | FlatFoot | Maxous: FreeBSD thats the way to go |
| 20:36.04 | drmessano-LT | Then fix it |
| 20:36.20 | ZenBSDi | or install vmware server .. install your linux distro.. use the nifty snapshot feature.. break it .. roll it back =p |
| 20:36.21 | kams | hi .. I have a question too. I want to send billing information in each second to the calling party .. for this reason I use dial with G() option .. this executes two dialplans in two channels .. But unfortunately one channel does not hangup if other channel hangs up :( .. any clue ? |
| 20:36.29 | Maxous | FreeBSD 'eh? Why not Ubuntu |
| 20:36.40 | ZenBSDi | /home/bsdi/bin/sys.sh -{c|d|k|m|v|a} |
| 20:36.41 | drmessano-LT | Had I known Linux wasn't nearly as intimidating, I would have started using it 10 years ago |
| 20:36.43 | ZenBSDi | DIS: Slackware 11.0.0 |
| 20:36.44 | jblack | Ohhh, you meant to thank me (as contrasted with your actual result of getting jbot to cheer a non-existant person on your behalf) |
| 20:36.44 | FlatFoot | Maxous: FreeBSD bit more secure |
| 20:36.49 | tzafrir_home | FreeBSD folks will tell you that Ubuntu is too easy |
| 20:36.54 | jblack | YW! YW! |
| 20:37.00 | kams | Maxous: or let me start a new company to help friends like you |
| 20:37.05 | drmessano-LT | Ubunturisk PBX FTW |
| 20:37.06 | kams | anyway .. |
| 20:37.12 | kams | hi .. I have a question too. I want to send billing information in each second to the calling party .. for this reason I use dial with G() option .. this executes two dialplans in two channels .. But unfortunately one channel does not hangup if other channel hangs up :( .. any clue ? |
| 20:37.15 | jblack | Maxous: Hey, I know someone you can hook up with. MrDigital. |
| 20:37.20 | drmessano-LT | ROFL |
| 20:37.21 | drmessano-LT | YES |
| 20:37.29 | jblack | He told me just a couple days ago that he's working on a 1/2 million dollar * project. |
| 20:37.39 | jblack | He could... use another eye... on the job |
| 20:37.48 | Maxous | lol |
| 20:37.58 | ZenBSDi | FlatFoot, fyi, ubuntu can be made to be just as secure.. just edit the hosts.allow and hosts.deny .. setup your iptables firewall and when you add users ... make sure their not in the same group that is already configured in sudoers .. |
| 20:38.03 | fiXXXerMet | Having trouble with conferencing. I've specified the 'i' option but I am not getting any messages when joining the room. Also, I am not being prompted for a password. |
| 20:38.05 | ZenBSDi | done |
| 20:38.05 | drmessano-LT | MrDigital is the great-great-step-uncledad of Steve Worziak.. Whose name rhymes with Steve Wozniak |
| 20:38.09 | drmessano-LT | He can help |
| 20:38.37 | jblack | Worziak is the original implementer of clownphone, right? |
| 20:38.49 | jblack | Did the prototype? |
| 20:38.54 | ZenBSDi | freebsd is only more secure out of the box than ubuntu .. once a little work is done ubuntu is just as secure as any stiff linux distro or BSD =p |
| 20:38.54 | drmessano-LT | He developed it until .03.. Then he had his accident :( |
| 20:39.08 | drmessano-LT | RIP Steve |
| 20:39.26 | drmessano-LT | That was a sad day for HappyClownPhone |
| 20:39.34 | jblack | ZenBSDi: check out the server release. Nothing sitting on an open port after install at all. Not even smtp |
| 20:40.01 | drmessano-LT | One day Ubuntu will be as secure as Windows Vista |
| 20:40.06 | a1fa | haha |
| 20:40.07 | jblack | Ironic that it's the server version that comes with no running servers at all. |
| 20:40.08 | Maxous | When does he sign on? |
| 20:40.11 | a1fa | drmessano: you are full of it |
| 20:40.22 | drmessano-LT | ROFL |
| 20:40.22 | kams | drmessano-LT: I hope that should not happend |
| 20:40.25 | jblack | I think he's right. |
| 20:40.32 | ZenBSDi | jblack, you mean ubuntu server? |
| 20:40.32 | jblack | I can't get vista to run long enough to be a target |
| 20:41.01 | jblack | (well, not really, but it sounded funny) |
| 20:41.09 | jblack | zenbsdi: Yes. There's a server release of ubuntu these days. |
| 20:41.17 | jblack | It's split into desktop and server. |
| 20:41.17 | drmessano-LT | Ubuntu Server is a joke... It's almost as much of a joke as when MS decided Windows NT was going to have a server edition |
| 20:41.25 | ZenBSDi | I know.. I'm using ubuntu 7.10 server for my asterisk server :) |
| 20:41.28 | drmessano-LT | It was a server because it was locked in the closet |
| 20:41.34 | kams | can anyone help me with dial parameters ? |
| 20:41.43 | Maxous | what is a trixbox? |
| 20:41.47 | drmessano-LT | LOL |
| 20:41.58 | drmessano-LT | Qwell: Stop me |
| 20:42.02 | jblack | Bah. everyone has a server in their closet. Who else, other than me, can say they have a computer permanantly stationed on their bed? |
| 20:42.08 | ZenBSDi | I just love when people put down ubuntu .. they have either never tried it or can't wrap their heads around something as simple as apt-get =p |
| 20:42.23 | drmessano-LT | Ubuntu is a fine desktop OS |
| 20:42.28 | drmessano-LT | Not a server |
| 20:42.41 | ZenBSDi | especially the gentoo folks .. they're the worst =p |
| 20:42.47 | kams | let us drop it .. |
| 20:43.06 | kams | ubuntu .. fedora .. centos .. freebsd all are fine |
| 20:43.14 | kams | now let us come to the point .. |
| 20:43.16 | jblack | ZenBSDi: There's just some people that think there are magical differences between kernels and stripped down installs. |
| 20:43.24 | kams | do you know the dial parameters well .. |
| 20:43.26 | drmessano-LT | Maxous: Trixbox is the center of all evil in the VoIPiverse |
| 20:43.28 | jblack | kams: Yeah? What about Lindows? |
| 20:43.29 | ZenBSDi | drmessano, I've been a nixer for 11 years now and I've been a working admin for 7 .. there is nothing "joke" about ubuntu server =p |
| 20:43.34 | kams | I am talking about the G() option |
| 20:43.45 | kams | lol |
| 20:43.51 | kams | let us come to the point .. |
| 20:44.01 | jblack | How do you say "G()"? g-spot? |
| 20:44.15 | kams | :)) |
| 20:44.18 | Maxous | drmessano-LT: hah, I am on trixbox.com now. |
| 20:44.19 | ZenBSDi | mmm I like to like the g-spot on sexy women :D |
| 20:44.32 | ZenBSDi | errr s/like to like/like to lick/g =p |
| 20:44.33 | drmessano-LT | Trixbox is the best way to learn nothing about Asterisk expect what someone wants you to see, and when it breaks you're more than screwed. |
| 20:44.38 | drmessano-LT | Crap |
| 20:44.40 | drmessano-LT | Except |
| 20:44.58 | ZenBSDi | amen to that.. edit the asterisk configs by hand.. be a man! |
| 20:45.03 | kams | when the dialplan execution stops on the separate channels when we use dial with G() option ?? |
| 20:45.07 | drmessano-LT | Trixbox is bike with training wheels that you suspiciously never feel like you own |
| 20:45.09 | Maxous | drmessano-LT: I see. So basically, its a good amount of fluff that smells like roses till it breaks. |
| 20:45.21 | J4k3 | updating my ancient trixbox install made call performance here suck |
| 20:45.23 | ZenBSDi | exten => 1300,1,Agi(agi://localhost/hello.agi) |
| 20:45.26 | *** join/#asterisk lackli (n=andyk@24-197-132-105.dhcp.spbg.sc.charter.com) |
| 20:45.28 | *** join/#asterisk tecnico (n=tecnico@user-24-214-56-217.knology.net) |
| 20:45.32 | J4k3 | calls get completed before my phones get the calls routed in, etc. |
| 20:45.34 | J4k3 | its a mess |
| 20:45.49 | J4k3 | I called someone a few minutes ago... they weren't sitting by the phone or anything, I never heard a ring... I heard... "hello!" |
| 20:45.50 | drmessano-LT | I will be the n00b bastard and say I like FreePBX for some things.. and it has its place.. But what they do with a Trixbox is a cancer compared to just installing FreePBX |
| 20:45.53 | J4k3 | I'm like... wtf |
| 20:45.56 | kams | why are you showing extensions ? |
| 20:46.09 | drmessano-LT | Its bloated and is more like a Windows 1000 tasks in one box than a PBX |
| 20:46.09 | J4k3 | and since its trixbox, its too hard to bother fix it |
| 20:46.19 | ZenBSDi | kams, why not? |
| 20:46.34 | *** join/#asterisk neoalex (n=chatzill@user-0ccengj.cable.mindspring.com) |
| 20:46.54 | lackli | i'm helping a network admin configure asterisk for his business. he may not be there long, and wants the next guy to have an easy time with it, so he's looking for GUIs...can anyone recommend some asterisk accessories? |
| 20:47.00 | kams | I think people have gone crazy .. I think the weather is bad .. |
| 20:47.08 | neoalex | hi guys, I'm having a weird problem... I have call pickup set up so that when I dial **Exten while the extension is ringing it picks up the call |
| 20:47.26 | kams | asterisk accessories !! what does that mean ? |
| 20:47.35 | Maxous | drmessano-LT: Thanks for letting me know that. |
| 20:47.37 | neoalex | that works fine, however when I set a function key on my Snom as a destination to ** something it doesn't work |
| 20:47.43 | Maxous | drmessano-LT: See my company is a 3Com Dealer. |
| 20:47.50 | drmessano-LT | you shouldnt need a 6GHZ Pentcore RAMblaster box to run Asterisk.. but throw in a CRM, spyware, GUIs for useless info you can get from the CLI, etc |
| 20:47.57 | lackli | some GUIs, kams, some GUIs.... |
| 20:47.58 | Maxous | drmessano-LT: But we know the days of 3Com might be comming to an end. |
| 20:48.14 | Maxous | drmessano-LT: So, we are in search of a new phone system to sell. |
| 20:48.24 | Qwell | Maxous: 3com sells an Asterisk box now |
| 20:48.37 | jblack | maxous: Here's how you can get rich. |
| 20:48.40 | cpm | does 3com even exist anymore? |
| 20:48.41 | Maxous | Qwell: Yep. I have it on my desk now. |
| 20:48.42 | neoalex | any ideas on what might be causing this |
| 20:48.46 | drmessano-LT | 1. Install Ubuntu |
| 20:48.47 | drmessano-LT | No |
| 20:48.52 | drmessano-LT | Go on, jblack |
| 20:49.02 | jblack | Put up a big * box that takes calls from people on a 976 line. Then, call them back later, and wake 'em up! |
| 20:49.04 | ZenBSDi | man.. I just don't understand some of the things people try with asterisk... I think of my setup like a business and go from there. whats this with picking up a line while it's ringing business? why not just setup an inbox and in the dialplan IfBusy goto voicemail? |
| 20:49.08 | a1fa | trixbox ftw |
| 20:49.08 | a1fa | :p |
| 20:49.14 | Maxous | cpm: Oh yes. 3Com is still around. With tons of great products. |
| 20:49.28 | Maxous | <PROTECTED> |
| 20:49.30 | G-nerd | hello guys, is MSN also based on SIP (I mean the speech part) |
| 20:49.33 | a1fa | Maxous : you are crazy for paying $1700 for asterisk appliance |
| 20:49.41 | a1fa | G-nerd : MSN has SIP |
| 20:49.47 | Maxous | a1fa: I didn't i'm a dealer. |
| 20:49.51 | a1fa | G-nerd : their integrated phone has it |
| 20:49.52 | cpm | tons eh? |
| 20:49.55 | drmessano-LT | Besides, Trixbox phones home more than Vista does.. Wearing your underwear inside out, Fonality already knows |
| 20:49.57 | G-nerd | so you couls also communicate with a MSN |
| 20:49.59 | a1fa | Maxous : drug dealer? |
| 20:50.04 | Maxous | a1fa: People will pay for it. 3Com Dealer. |
| 20:50.06 | G-nerd | oooh ok |
| 20:50.19 | a1fa | Maxous : liar |
| 20:50.20 | drmessano-LT | Drug dealers need PBXs too |
| 20:50.22 | drmessano-LT | and IVRs |
| 20:50.25 | a1fa | yes |
| 20:50.29 | Alan_Hicks | 3com ain't what it used to be though. |
| 20:50.34 | drmessano-LT | "For smack, smack the 1" |
| 20:50.40 | a1fa | Press #1 for Coke, Press #2 for Marry-Jane, Press #3 for E |
| 20:50.40 | Maxous | a1fa: :-/ ? |
| 20:50.45 | G-nerd | but what about the other which you can talk with other msn users, without that built in telephone |
| 20:50.48 | Alan_Hicks | They seem to have lost a lot of their visibility when they stopped making consumer NICs. |
| 20:50.51 | Maxous | You are right. they arn't as strong. |
| 20:50.58 | cpm | 15 years ago, heck, even 10 years ago, I loved 3com, then , , well, , they seemed to have lost focus |
| 20:50.58 | jblack | I thought all a drug dealer needed was a ho to carry around his cell phone? |
| 20:51.02 | a1fa | G-nerd : negative |
| 20:51.06 | G-nerd | anyway, I got the point |
| 20:51.08 | Maxous | <PROTECTED> |
| 20:51.13 | kams | asterisk should write chan_solid to deliver drugs .. |
| 20:51.13 | drmessano-LT | 3com is a company without an identity |
| 20:51.20 | a1fa | 3Croocks |
| 20:51.20 | G-nerd | a1fa I know why |
| 20:51.30 | Alan_Hicks | Maxous: Yeah I know. They just don't seem to know what that market is, IMHO. :^) |
| 20:51.44 | Alan_Hicks | rob0: Be honest! You never had focus! |
| 20:51.44 | jblack | I'll tell you what they do. |
| 20:51.44 | drmessano-LT | I bought a 3com toaster last week.. WTF |
| 20:51.48 | G-nerd | but it must use voip, with their own protocol I guess |
| 20:52.03 | jblack | "Convergence applications". "Open services networking" "Secure converged networks"... |
| 20:52.07 | jblack | Oh, and samba |
| 20:52.15 | Maxous | <PROTECTED> |
| 20:52.16 | ZenBSDi | NetBSD and AI scripting .. cause you know you want to see the toaster bang the dog =p |
| 20:52.19 | drmessano-LT | "Unified Communications" |
| 20:52.21 | Maxous | <PROTECTED> |
| 20:52.37 | drmessano-LT | Glue an Asterisk Box to a Toaster.. Unified Breakfast |
| 20:52.41 | Maxous | drmessano-LT: It's more like Secure Converged Networks. |
| 20:52.48 | Alan_Hicks | Maxous: s/Moron of a // |
| 20:52.58 | Alan_Hicks | There, fixed your redundancy. |
| 20:53.04 | Maxous | lol |
| 20:53.09 | jblack | drmessano: No. You have to glue the * box _inside_ the toaster. Makes the fork-work more interesting |
| 20:53.14 | drmessano-LT | lol |
| 20:53.15 | Maxous | drmessano-LT: aww, we don't need to be mean about 3Com. |
| 20:53.26 | G-nerd | Why Digium developed IAX? just to communicate between asterisk machines? |
| 20:53.40 | jblack | g-nerd: In simple terms, because sip sucks. |
| 20:53.40 | rob0 | Does 3com still own the digits 5 and 9? |
| 20:53.42 | Alan_Hicks | Seriously though, if I could buy 3co NICs and switches that were high quality without being ridiculously expensive, I would do so in a heart beat. They would get all my business. |
| 20:53.53 | jblack | Try it. make a sip sound with your mouth. Notice that you're sucking |
| 20:53.58 | Alan_Hicks | rob0: No, Dolly Parton copyrighted those. |
| 20:54.00 | a1fa | <jblack> Try it. make a sip sound with your mouth. Notice that you're sucking |
| 20:54.05 | drmessano-LT | maxous: No biggie, 3com is too far behind to develop technology to hear us |
| 20:54.12 | G-nerd | jblack, are you kidding? |
| 20:54.23 | a1fa | definatley sucking |
| 20:54.24 | a1fa | [TK]D-Fender |
| 20:54.26 | a1fa | yo |
| 20:54.30 | kams | jblack: lol |
| 20:54.33 | jblack | g-nerd: Not about the sucks part. It's a nightmare with firewalls and nat. |
| 20:54.38 | drmessano-LT | A 3com switch |
| 20:54.41 | Maxous | Alan_Hicks: I hear ya. They do have great switches. |
| 20:54.43 | drmessano-LT | Wow, next to my HP switch? |
| 20:54.44 | a1fa | 3com switch < gay |
| 20:54.51 | [TK]D-Fender | ? |
| 20:54.53 | Maxous | drmessano-LT: lol |
| 20:54.54 | a1fa | :) |
| 20:54.54 | G-nerd | and IAX will make change of that? |
| 20:55.00 | drmessano-LT | ProCurve FTW |
| 20:55.00 | jblack | as to whether or not your sipping sucks, that would depend upon whether or not you have cheeks.... |
| 20:55.05 | a1fa | [TK]D-Fender : just checking if you were alife |
| 20:55.17 | Alan_Hicks | ProCurve switches are just completely out of the picture for all my clients. |
| 20:55.27 | Maxous | a1fa: Hah. They have great switches. |
| 20:55.29 | Alan_Hicks | Far far far too expensive. |
| 20:55.33 | a1fa | ProCurve sucks |
| 20:55.36 | a1fa | boat anchors |
| 20:55.42 | Maxous | Alan_Hicks That depends on your needs. |
| 20:55.50 | drmessano-LT | 100VG FTFW!!!! |
| 20:55.52 | Maxous | Alan_Hicks: they have a huge range of switches. |
| 20:55.54 | Alan_Hicks | Maxous: I work with small and medium businesses. |
| 20:56.11 | kams | you know I am looking for an equestion ... (income - cost*0) = profit .. |
| 20:56.13 | Alan_Hicks | Most are served fine with a non-managed 24-port switch. |
| 20:56.29 | Maxous | Alan_Hicks: the Baseline switches and office connect switches are a good fit there. |
| 20:56.32 | jblack | kams: Sell your toe lint. |
| 20:56.34 | a1fa | <kams> you know I am looking for an equestion ... (income - cost*0) = profit .. |
| 20:56.41 | a1fa | kmas: you must spend money to make money |
| 20:56.45 | a1fa | next |
| 20:56.49 | drmessano-LT | I didnt know HP made switches until someone sent me a case of Ciscos in a shipping box for HPs |
| 20:56.51 | drmessano-LT | j/k |
| 20:56.52 | Alan_Hicks | Maxous: Still to expensive when you consider I can buy two or three lesser switches for the same ammount. |
| 20:56.57 | [TK]D-Fender | a1fa: An Equestrian? |
| 20:57.07 | *** join/#asterisk nhuisman_work (n=nhuisman@aeko.IfA.Hawaii.Edu) |
| 20:57.09 | [TK]D-Fender | a1fa: Go camel-jockey! |
| 20:57.11 | a1fa | [TK]D-Fender : whats that? |
| 20:57.12 | G-nerd | jblack when does SIP screws, if you make connections outside your locan network? |
| 20:57.15 | drmessano-LT | kmas |
| 20:57.20 | drmessano-LT | The equation is easy |
| 20:57.23 | drmessano-LT | 1. Install |
| 20:57.24 | Maxous | Alan_Hicks: Hum, really? Are they running VoIP as well? |
| 20:57.25 | drmessano-LT | 2. Configure |
| 20:57.29 | drmessano-LT | 3. ????? |
| 20:57.32 | drmessano-LT | 4. Profit!!! |
| 20:57.33 | kams | lol |
| 20:57.40 | jblack | g-nerd: Yes. especially then. |
| 20:57.43 | Maxous | Alan_Hicks: then you can use PoE. |
| 20:57.45 | nhuisman_work | does anyone know of a stand alone little bit of scripts and maybe gui for handling phone firmware provisioning and sip/skinny.conf |
| 20:57.55 | kams | 3 = customize .. |
| 20:58.02 | Alan_Hicks | Maxous: No, most of my clients aren't. We're not strictly VoiP. |
| 20:58.04 | kams | I do the 3rd part .. |
| 20:58.11 | nhuisman_work | and no, i don't want to go with one of the asterisk integrated packes, I have asterisk business edition and need to extend that. |
| 20:58.12 | kams | profit goes to my clients .. |
| 20:58.19 | G-nerd | hmmm, but how is it possible voip keeps groing, while using SIP. And how about H.323? |
| 20:58.25 | Maxous | Alan_Hicks: gotcha.\ |
| 20:58.53 | a1fa | i smell fear |
| 20:59.01 | Alan_Hicks | Mostly small businesses, smb and ISVs. |
| 20:59.25 | Maxous | Alan_Hicks: for a baseline, unmanaged, 24 port switch with 2 gigabit uplinks, the MSRP is .... |
| 20:59.42 | nhuisman_work | when I installed asterisk I got a whole bunch of sample configs put in place, is there some bare minimum number of config files i can have in the /etc/asterisk dir, i don't want the sample files anymore. |
| 20:59.45 | G-nerd | jblack, Firewall is just doing his job, to keep security. Maybe you need to learn from Skype, check which ports they are using |
| 20:59.52 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 21:00.00 | jblack | g-nerd: Well, as much as sip sucks, getting *ganked* by telcos sucks much worse. |
| 21:00.03 | *** part/#asterisk kams (n=root@202.56.7.133) |
| 21:00.19 | Maxous | 179 from CDW |
| 21:00.28 | jblack | g-nerd: uhm, skype is patented, and doesn't exactly give out their codebase. |
| 21:00.37 | Maxous | <PROTECTED> |
| 21:00.38 | Alan_Hicks | Yeah that's not too bad, but still pricey for many of my clients. |
| 21:00.47 | G-nerd | it is not that, but they use also SIP or H.323 |
| 21:00.48 | Maxous | Really? do they need the gig uplinks? |
| 21:00.50 | Alan_Hicks | Of course, many of my clients only have half a dozen computers too. |
| 21:00.59 | nhuisman_work | what is that an ata device? |
| 21:01.08 | nhuisman_work | oh, nm a switch |
| 21:01.12 | Maxous | lol |
| 21:01.26 | Alan_Hicks | Maxous: Like I said, small business. :-) |
| 21:01.28 | nhuisman_work | how is $180 for a 24 port switch too much money |
| 21:01.30 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 21:01.38 | jblack | g-nerd: Nope. just their own secret protocol. |
| 21:01.43 | Alan_Hicks | nhuisman_work: When I can get three for that cost. |
| 21:01.54 | G-nerd | jblack you don ' t convinced me about the SIP |
| 21:01.55 | *** join/#asterisk supjigator (n=sysgod@152.53.16.10) |
| 21:02.05 | jblack | that's one of the places sip doesn't suck. sip is an open protocol |
| 21:02.12 | *** part/#asterisk supjigator (n=sysgod@152.53.16.10) |
| 21:02.16 | nhuisman_work | Alan_Hicks, oh i guess you probably don't need the gigabit uplink ports, that probably makes it much cheaper without those. |
| 21:02.17 | G-nerd | if SIP is useless, than it was allready dead |
| 21:02.27 | Maxous | <PROTECTED> |
| 21:02.36 | *** part/#asterisk lackli (n=andyk@24-197-132-105.dhcp.spbg.sc.charter.com) |
| 21:02.43 | jblack | I said it sucked, not useless. |
| 21:02.47 | Alan_Hicks | Maxous: I only have a handful of clients with two dozen or more nodes. |
| 21:02.50 | Maxous | <PROTECTED> |
| 21:02.53 | [TK]D-Fender | G-nerd: SIP is useless? Tell taht to all these ITSPs.... |
| 21:03.09 | G-nerd | D-Fender, jblack told me it sucks |
| 21:03.10 | Alan_Hicks | Maxous: That's not bad at all. |
| 21:03.10 | nhuisman_work | why are you saying sip is useless? |
| 21:03.15 | a1fa | there is no alternative to sip |
| 21:03.34 | jblack | perhaps iax2 some day. |
| 21:03.42 | Maxous | <PROTECTED> |
| 21:03.48 | G-nerd | I thought IAX is to cummunicate between Asterisk machines |
| 21:03.52 | Alan_Hicks | s/no alternative/no superior alternative with native support in the phones/ |
| 21:03.58 | drmessano-LT | IAX is good for Trunks |
| 21:03.59 | [TK]D-Fender | G-nerd: It is |
| 21:04.06 | drmessano-LT | No |
| 21:04.07 | *** join/#asterisk nortex (n=chatzill@64.136.92.71) |
| 21:04.11 | drmessano-LT | IAX2 IS KICK ASS for trunks |
| 21:04.12 | Maxous | <PROTECTED> |
| 21:04.16 | G-nerd | well than jblack gives me totally wrong questions |
| 21:04.26 | G-nerd | I mean explenations |
| 21:04.48 | jblack | i don't see where I told you anything misleading or inaccurate. |
| 21:04.51 | Maxous | <PROTECTED> |
| 21:05.05 | drmessano-LT | G-nerd: You cant spell out iNTER aSTERISK ExCHANGE any clearer |
| 21:05.15 | drmessano-LT | Thats what it does |
| 21:05.28 | G-nerd | drmessano, tell it to jblack |
| 21:05.35 | drmessano-LT | IAX can do double duty as a client protocol |
| 21:05.37 | Alan_Hicks | Would be really nice if IP phones spoke IAX. |
| 21:05.41 | drmessano-LT | and it does the job well |
| 21:05.51 | a1fa | your mom speaks iax |
| 21:05.54 | Maxous | lol |
| 21:05.54 | a1fa | =) |
| 21:05.56 | jblack | I wish |
| 21:05.57 | G-nerd | hahahaha |
| 21:06.04 | a1fa | we communicate on the same protocol |
| 21:06.09 | Maxous | Well guys, it's been fun. |
| 21:06.14 | Maxous | i g2g install a firewall now. |
| 21:06.16 | a1fa | j/k |
| 21:06.16 | Maxous | fun. |
| 21:06.23 | G-nerd | maybe his mom could understand what the boy wants |
| 21:06.23 | drmessano-LT | Your mom is in my FreePBX phonebook |
| 21:06.23 | Alan_Hicks | Maxous: Later. |
| 21:06.40 | rantsh | anyone knows how I can get agent status from agi? |
| 21:06.49 | jblack | Unfortunatley, she passed many years ago. I guess she'd have to use the Ether-net |
| 21:06.51 | Alan_Hicks | Say what ya want about me, but leave Mama out of it. |
| 21:07.02 | drmessano-LT | "I have a custom extension for your girlfriends cellphone" |
| 21:07.11 | G-nerd | I saw a spider coming out of a man' s mouth |
| 21:07.32 | Alan_Hicks | G-nerd: Must have been on the IAX-Files. |
| 21:07.37 | G-nerd | I don' t talk about anyone's momma |
| 21:07.56 | drmessano-LT | HA |
| 21:08.02 | G-nerd | I respect everybody, also their relatives |
| 21:08.09 | drmessano-LT | "I SIP'ed your girlfriend" |
| 21:08.15 | nhuisman_work | it's easy to respect someone when you are on top of them. |
| 21:08.34 | drmessano-LT | I respect you, jblack |
| 21:08.58 | drmessano-LT | Im having a small problem with an SPA-3102 |
| 21:09.06 | G-nerd | I don' t know how jblack' s telephonebook looks like if the only thing he can say is suck |
| 21:09.12 | a1fa | lol |
| 21:09.21 | drmessano-LT | The power supply has the distinct smell of baking bread, and it wont come on |
| 21:09.23 | G-nerd | :p |
| 21:09.32 | ZenBSDi | exten => 1200,1,MeetMe(yer_mom) exten => 1200,2,DoMe(yer_mom) exten => 1200,3,LetMeSleep_now_thatwasgood(yer_mom) |
| 21:09.33 | ZenBSDi | =p |
| 21:09.45 | drmessano-LT | rofl |
| 21:09.54 | Alan_Hicks | drmessano-LT: Stick a screw-driver in it to jump it off. |
| 21:10.03 | drmessano-LT | Good idea |
| 21:10.28 | drmessano-LT | Maybe i'll stick YOUR MOM.. oh nevermind.. brb |
| 21:10.34 | G-nerd | Zen, I personally thnk you go to far with your low IQ joke, sorry to say that, but that is my opnion |
| 21:10.43 | *** join/#asterisk comprookie2000 (n=david@adsl-065-012-210-216.sip.bct.bellsouth.net) |
| 21:11.04 | drmessano-LT | Low IQ? |
| 21:11.09 | ZenBSDi | yer just jealous you can't write a pretty dialplan like I can =p |
| 21:11.20 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 21:11.22 | nhuisman_work | yeah i'd say that was a lot better then <insert sentence here> + your mom. |
| 21:11.31 | drmessano-LT | Yes |
| 21:12.15 | ZenBSDi | hey it's the asterisk room .. it's <insert dial plan here> + your mom =p |
| 21:12.46 | drmessano-LT | "Who is your ITSP?" "Your mom" |
| 21:13.10 | drmessano-LT | "Where can I paste this debug?" "your mom" |
| 21:13.10 | G-nerd | but anyway, he was just exploring the border |
| 21:13.12 | ZenBSDi | My trunk runs to "your mom" =p |
| 21:13.15 | drmessano-LT | It all works |
| 21:13.44 | G-nerd | come on guys, can we get back on asterisk, with no mamma |
| 21:13.47 | ZenBSDi | Welp, thanks for showing what a prick you are G-nerd.. last time I answer any of your questions =p |
| 21:14.31 | G-nerd | Zen, this doesn't mean I can' t give you my opinion |
| 21:14.33 | drmessano-LT | What do you expect.. Google Nerd is still in beta |
| 21:14.55 | G-nerd | But thank you very much for your help Zen, I still appreciate it |
| 21:14.58 | a1fa | [TK]D-Fender : quick question |
| 21:15.06 | a1fa | [TK]D-Fender : do you use allowguest in your installs? |
| 21:15.17 | a1fa | i am debating about it |
| 21:15.26 | a1fa | i have some users that would like to call me like that |
| 21:15.38 | jblack | Is it someone else's bandwidth? |
| 21:15.40 | [TK]D-Fender | a1fa: depends whose |
| 21:15.56 | ZenBSDi | you need to lighten up G-nerd .. this may be the asterisk room but obviously some people were having fun with momma jokes and I wanted in .. no need to call me "low iq" when the fact is I AM a linux admin and last time I checked you need atleast a Mid IQ for that =p |
| 21:16.12 | G-nerd | hmmm I need to concetrate on the asterisk book to get asterisk running |
| 21:16.20 | jblack | zenbsdi: Hey, there's a rumor going around that you have a low iq. Any truth to that/ |
| 21:16.22 | ZenBSDi | no.. you just need to learn to google |
| 21:16.31 | G-nerd | Zen hahahahahah |
| 21:16.41 | G-nerd | you are right Zen, sorry, I appologize |
| 21:16.56 | jblack | g-nerd: Did you have a specific question in mind? |
| 21:17.00 | G-nerd | Would you forgive me momma Zen? |
| 21:17.15 | ZenBSDi | jblack .. not as frupid as I am you think triend =p |
| 21:17.33 | ZenBSDi | no wait.. thats drunk =p |
| 21:17.36 | jblack | O RLY? K THX |
| 21:17.37 | *** join/#asterisk Lucky7 (i=Lucky7@207.200.28.175) |
| 21:17.55 | drmessano-LT | IAX2KTHXBYE |
| 21:17.57 | G-nerd | jblack, if it is not about SIP I' ll give ya a holla |
| 21:18.17 | ZenBSDi | lol@iax2kthxbye =p |
| 21:18.25 | Lucky7 | is there anyway to have the server beep when it starts recording a call? |
| 21:18.32 | G-nerd | ok, guys , I have to fix Asterisk man, I' ll catch ya up later |
| 21:18.40 | jblack | g-nerd: basically, you want iax, but most everything out there is sip. So, you need it. |
| 21:18.55 | ZenBSDi | lucky7 .. fastagi script .. script it to play a sound .. otherwise I don't think there is a built in way |
| 21:18.58 | [TK]D-Fender | jblack: And why would he want IAX? |
| 21:19.06 | G-nerd | ok jblack., I understand and I believe you, thank you |
| 21:19.16 | Lucky7 | ZenBSDi: figured it have to be something like that. |
| 21:19.23 | jblack | Easier to get through and around firewalls, calls are trunked. It's a neater, smaller package all around. |
| 21:19.28 | Lucky7 | ZenBSDi : Thanks. |
| 21:19.32 | jblack | I think it also gets your whites whiter without dulling your colors. |
| 21:19.36 | ZaVoid | burp |
| 21:19.37 | a1fa | trunk ftw |
| 21:19.43 | [TK]D-Fender | jblack: unless you need the bandwidth saving, I wouldn't touch it |
| 21:19.47 | drmessano-LT | IAX for clients isnt practical... Learn to fix NAT problems, and use IAX for trunks |
| 21:19.49 | G-nerd | so jblack, for communicationg in a local network like my house, I better need to use IAX? |
| 21:19.55 | ez` | i would liek to use my previous asterisk ; 1.4.17 seem have dtmf problem ; i can compile 1.4.15 and make insatll over my current asterisk ( 1.4.17 ) |
| 21:20.05 | ZenBSDi | yeah lucky7, all the really cool tricks require scripting .. and unless you're learning java I can't help .. my agi scripts are all java =p |
| 21:20.10 | [TK]D-Fender | G-nerd:no |
| 21:20.12 | jblack | g-nerd: People more experienced than I are saying no |
| 21:20.15 | drmessano-LT | Good god |
| 21:20.26 | drmessano-LT | IAX 4 CLIENTS = VERY NO |
| 21:20.27 | DrAk0 | why a2billing is so confusing... |
| 21:20.49 | drmessano-LT | Its a BETTER protocol, but its waste pondering it |
| 21:20.55 | G-nerd | more experienced than you, hmmm, than I have to think about D-Fender |
| 21:21.00 | ZaVoid | what makes it better? |
| 21:21.11 | ZaVoid | i've never quite understood this |
| 21:21.22 | drmessano-LT | The obvious NAT issues and less overheard with multiple channels |
| 21:21.30 | *** join/#asterisk qdk (n=qdk@195.242.194.41) |
| 21:21.31 | nhuisman_work | does anyone know what the bare minimum number of configs in /etc/asterisk is? |
| 21:21.32 | drmessano-LT | Its good for what its designed for |
| 21:21.34 | jblack | certainly there's room for agreement that opening up 10k ports is not comforting. |
| 21:21.36 | ZaVoid | aside from nat |
| 21:21.43 | nhuisman_work | i have a shit load of sample files in there but I don't want them messing up my install. |
| 21:21.51 | drmessano-LT | 10,000 ports to ONE listener |
| 21:21.57 | ZenBSDi | [TK]D-Fender is a very knowledgable person with asterisk. when I had some questions he had the answer .. so don't ignore [TK]D-Fender 's advice on things or if he just points you to the doc |
| 21:22.08 | drmessano-LT | 1 port on one bad listener is no worse than 10000 to a secure one |
| 21:22.31 | ZaVoid | SIP just seems a lot more flexible to me as a protocol |
| 21:22.31 | jblack | nhuisman_work: make a sample dir, move them into there. Then, as you realize you really do need them, move them back. Hopefully you'll learn sooner than I to move nearly all of them back. |
| 21:22.33 | drmessano-LT | I hate that argument about "the number of ports" immensly |
| 21:22.34 | G-nerd | D-Fender, the Asterisk Master |
| 21:22.40 | ZaVoid | but i came from the H.323 world.. so i love sip |
| 21:22.48 | drmessano-LT | SIP is very flexbile |
| 21:22.54 | drmessano-LT | and you can fix NAT if you know what youre doing |
| 21:22.56 | nhuisman_work | hmm |
| 21:23.02 | ZaVoid | nothing worse then LRQ's and ARQ's and h323 v2 vs v1 incompatabilities |
| 21:23.02 | nhuisman_work | so most of them are required eh |
| 21:23.05 | [TK]D-Fender | G-nerd: Not so much, but I've got a good grasp of the basics and a bit extra on top. |
| 21:23.05 | drmessano-LT | Everyone is SOO down on SIP/NAT/SIP/NAT |
| 21:23.08 | ZaVoid | hell ever read a h.323 debug? its not fun |
| 21:23.16 | ZaVoid | iax looks the same.. crappy debugs |
| 21:23.22 | G-nerd | ??? Zavoid, H323 is not the same as SIP though they are the same VOIP protocols |
| 21:23.37 | ZaVoid | they're both signaling protocols G-nerd |
| 21:23.50 | [TK]D-Fender | IIRC H.323 closely resembles PRI QSIG. |
| 21:23.51 | G-nerd | D-Fender, to me you are the MASTER, sensei |
| 21:24.01 | ZaVoid | it does fender |
| 21:24.26 | jblack | I'm sure I'm overly biased as I spent a lot of time trying to wedge siproxd and * on the same box. |
| 21:24.27 | ZaVoid | doesn't mean its elegant |
| 21:24.33 | [TK]D-Fender | ZaVoid: I recall some mention of it as such whil would explain why it is often preferred by carriers. |
| 21:24.36 | ZaVoid | nothing was worse then multi gatekeeper hopping |
| 21:24.49 | ZaVoid | well carrier world is moving away form it now |
| 21:24.54 | [TK]D-Fender | ZaVoid: Would mean more easily transportable signalling |
| 21:25.02 | ZaVoid | all the major SBC's support both and all new turnups unless requested seem to be SIP |
| 21:25.22 | ZaVoid | at least thats been my experience recently |
| 21:25.39 | ZaVoid | most major carriers are pushing SIP now. and slowly removing h.323 endpoints |
| 21:25.41 | [TK]D-Fender | Ok, checkout time at the office... I'm fried and heading home, BBIAB |
| 21:25.46 | *** join/#asterisk jdiego (n=root@190.144.32.10) |
| 21:25.47 | ZaVoid | see ya fender |
| 21:27.41 | drmessano-LT | Time for me to go fight with router install |
| 21:27.45 | drmessano-LT | BBL |
| 21:28.59 | *** join/#asterisk skirmisha (n=viki@87-126-34-63.btc-net.bg) |
| 21:29.04 | skirmisha | guys |
| 21:29.09 | *** join/#asterisk c4t3l (n=c4t3l@74.95.210.124) |
| 21:29.20 | c4t3l | howdy gang |
| 21:29.55 | skirmisha | when u have moh in queue and when queue anounce msg then moh song starts from begining, is this a bug or how can i fix it. I use native MOH |
| 21:31.01 | UnixDog | ok this day sucks |
| 21:31.07 | skirmisha | any ideas |
| 21:32.45 | nhuisman_work | Hey I setup my dhcp for phones but it seems like the phones aren't accepting the addresses, it just keeps looking with DHCPDISCOVER, DHCPOFFER, DHCPREQUEST, DHCPACK |
| 21:32.54 | nhuisman_work | any idea why the phone would jsut take the ip? |
| 21:33.12 | nhuisman_work | keeps looping those in the log |
| 21:33.33 | *** part/#asterisk jdiego (n=root@190.144.32.10) |
| 21:33.34 | c4t3l | skirmisha: if you use madplay to stream MOH you can specify a random order |
| 21:34.08 | skirmisha | i use native MOH |
| 21:34.12 | G-nerd | jblack was saying that SIP sucks because of bas soundquality, now I' ve read that SIP is like H.323 (which Zavoid corrected me) signal protocols. And the sound "the media" is taken care by RTP, so the soundquality is determined by the RTP or other alike protocols |
| 21:34.26 | skirmisha | but problem is that if queue anounce msg then moh starts over |
| 21:34.35 | ZaVoid | correct G-nerd |
| 21:34.51 | c4t3l | skirmisha: not sure if i can help with this one |
| 21:34.54 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 21:34.55 | nhuisman_work | anyone? |
| 21:35.25 | G-nerd | Thnx Zavoid, but could it be that the bad sound also means to many delays, instead of litches and pops |
| 21:35.27 | nhuisman_work | do the phones have to be configured in asterisk before they accept an ip? |
| 21:35.47 | puppet | anyone in sweden that could try faxing me or anyone anywhere just try to send a fax_ >( |
| 21:35.50 | ZaVoid | sure cutomers "bad quality" can be many reasons |
| 21:35.59 | ZaVoid | latency, jitter packet loss and much more |
| 21:36.11 | outtolunc | UnixDog: that special eh |
| 21:36.28 | G-nerd | hmmm, actually I have never experienced voip, I have no one to talk with in Skype :( |
| 21:36.44 | puppet | skype isnt real voip <P |
| 21:36.47 | puppet | ;P |
| 21:36.50 | UnixDog | its one of those days |
| 21:37.39 | *** part/#asterisk G-nerd (n=AskMe@dhcp-077-249-041-129.chello.nl) |
| 21:38.17 | *** join/#asterisk G-nerd (n=AskMe@dhcp-077-249-041-129.chello.nl) |
| 21:38.19 | *** join/#asterisk bsdwarrior (n=mfahey@fahey.enter.net) |
| 21:38.37 | G-nerd | my laptop sucks |
| 21:38.40 | G-nerd | pfff |
| 21:38.50 | bsdwarrior | I setup func_odbc.conf do I need to do a "reload" or will reload extensions work? |
| 21:39.18 | ZaVoid | lol how isnt skype real voice over internet protocol? |
| 21:39.18 | bsdwarrior | I am already using unixodbc for cdr, but the function I created in func_odbc.conf doesnt work |
| 21:39.39 | ZaVoid | restart now bsdwarrior |
| 21:40.05 | bsdwarrior | zarviod do I have to do a "reload" everytime I add a function to that file? |
| 21:40.28 | *** join/#asterisk CanWood (n=chatzill@24.108.64.80) |
| 21:40.48 | *** join/#asterisk CunningPike_ (n=arodgers@204.239.12.183) |
| 21:44.21 | CanWood | Hey Folks. I want to have an Asterisk box on a separate VLAN in our network. If I populate the Vlan Tag and Priority value in the Layer2 Qos settings on a GXP2000, does anyone know if the DHCP broadcasts it sends out will be tagged? That way I can run the DHCP server on the Asterisk box to service only that VLAN |
| 21:44.39 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 21:46.38 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:47.45 | *** part/#asterisk JaminCollins (n=jcollins@151.101.5.95) |
| 21:49.37 | *** join/#asterisk FunnyManVA (n=funnyman@12.171.153.133) |
| 21:49.56 | *** join/#asterisk grandpapadot (n=null@mail.heavylogic.com) |
| 21:51.55 | *** join/#asterisk trippss (n=ss@72.20.150.196) |
| 21:52.29 | bsdwarrior | im calling a function defined in func_odbc.conf but it doesnt put anything in the db. any suggestions |
| 21:53.30 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 21:56.36 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:00.16 | bsdwarrior | anyone around ? |
| 22:00.29 | a1fa | mom |
| 22:01.59 | G-nerd | Damn! sip.conf is a very long configuration file, pffff |
| 22:02.14 | G-nerd | are you still talking about mom? |
| 22:02.27 | *** part/#asterisk ddunavant (n=David@70-4-149-49.area3.spcsdns.net) |
| 22:02.49 | [TK]D-Fender | G-nerd, You have our permission to shorten it |
| 22:02.56 | UnixDog | what are you using odbc for |
| 22:03.05 | syzygyBSD | hmm, I am trying to get a "Please deposit 25c if you want to continue" type message going... but I can't find the right prompts in default sounds, anyone have some? |
| 22:03.17 | UnixDog | use the mysql and or pgsql connectors |
| 22:03.32 | G-nerd | no, there must be a reason for it |
| 22:03.33 | syzygyBSD | gonna have that be my zapateller message |
| 22:03.42 | UnixDog | syzygyBSD, you doing a payphone pbx |
| 22:03.52 | syzygyBSD | no payphone... |
| 22:03.56 | [TK]D-Fender | syzygyBSD, http://www.theivrvoice.com/ |
| 22:04.00 | G-nerd | D-Fender, is that meant sarcastic? |
| 22:04.13 | UnixDog | no |
| 22:04.14 | syzygyBSD | [TK]D-Fender: :) oh I have used allison a lot |
| 22:04.25 | [TK]D-Fender | G-nerd, Slightly. Its your config, do what you want with it. |
| 22:04.28 | UnixDog | I think its ment ask her to record the sound |
| 22:04.35 | syzygyBSD | even have her saying "No spitting swearing farting or picking your ass" |
| 22:04.45 | G-nerd | yeah, but not to mess my config up |
| 22:04.46 | syzygyBSD | just not the one i want right now... |
| 22:05.00 | G-nerd | btw it is almost all outcommented |
| 22:05.01 | UnixDog | that would not work sin [TK]D-Fender does all those |
| 22:05.15 | bsdwarrior | unixdog - im using postgres, I want to do an insert and nothing gets added to the db |
| 22:05.27 | *** join/#asterisk nybble (n=nybble@about/apple/performa/nybble) |
| 22:05.38 | G-nerd | D-Fender, will I get it run within a few hours? |
| 22:05.39 | bsdwarrior | <PROTECTED> |
| 22:05.41 | UnixDog | did you conpile asteris with pgsql support |
| 22:05.50 | [TK]D-Fender | G-nerd, thats just SAMPLE junk. That isn't your config, thats just junk documentation |
| 22:05.50 | UnixDog | or with unixodbc support |
| 22:05.57 | bsdwarrior | res_odbc.conf is correct. |
| 22:06.05 | bsdwarrior | unixdog - im already using unixodbc for cdr and thats working |
| 22:06.19 | UnixDog | you dont need odbc to connect to postgress now days |
| 22:06.29 | ZX81 | how do you pass a sip username in a dialstring? |
| 22:06.35 | UnixDog | there is a postgres connector |
| 22:06.44 | G-nerd | hmmm, you mean I need to start from scratch?. Anyway, I just follow the asterisk pdf book |
| 22:06.59 | ZX81 | is it possible? |
| 22:07.06 | bsdwarrior | unixdog, I can't redo the entire config right now for a live system, |
| 22:07.23 | bsdwarrior | unixdog, how can I enable debugging for func_odbc ? |
| 22:08.33 | UnixDog | dont use odbc dont know |
| 22:08.41 | bsdwarrior | is this syntax even correct anyone? exten => 7,n,Set(foobar=ADDCALLBACK_MAIN(${CALLBACKNUM} ) ) |
| 22:09.29 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 22:09.29 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 22:09.42 | ZX81 | I'm trying to set callerid on an outbound sip trunk but the from: field never gets populated |
| 22:09.44 | ZX81 | asterisk 1.4 |
| 22:09.55 | ZX81 | using set(CALLERID(all)=1234 <1234> |
| 22:10.02 | ZX81 | where 1234 is the cid |
| 22:10.17 | ZX81 | if I set fromuser it works |
| 22:10.24 | ZX81 | but want to just pass the cid |
| 22:10.33 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 22:10.46 | ZX81 | exten => 125,1,Set(CALLERID(all)=2122827000 <2122827000>) |
| 22:11.17 | ZX81 | exten => 125,2,Dial(SIP/366015162339718@nextcarrier) |
| 22:11.43 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
| 22:11.43 | mvanbaak | ZX81: did you try with 2 calls, one for CALLERID(name) and one for CALLERID(num) ? |
| 22:11.47 | mvanbaak | that's how I do it |
| 22:11.54 | ZX81 | yeah |
| 22:11.58 | ZX81 | I did that first |
| 22:12.15 | fiXXXerMet | When I call voicemail, conference - whatever - command line shows the 'Playing 'file-name' output, but I don't hear anything on the phone. |
| 22:12.38 | fiXXXerMet | All of the sound files are in /var/lib/asterisk/sounds |
| 22:12.47 | mvanbaak | fiXXXerMet: turn up the volume ;) |
| 22:12.51 | fiXXXerMet | lol |
| 22:12.54 | ZX81 | From: "Unknown" <sip:Unknown@207.251.81.10>;tag=as4a198239 |
| 22:13.10 | fiXXXerMet | I wish it were that easy mvanbaak |
| 22:14.14 | fiXXXerMet | It doesn't say that it can't find the file, it says that the file is playing - I just don't hear it. |
| 22:14.19 | bsdwarrior | I want to be able to use sql statements (postgres) in the dialplan. can someone point me in the right direction ? |
| 22:15.54 | fiXXXerMet | bsdwarrior: I don't think that is how it works. |
| 22:17.15 | bsdwarrior | fixxxermet I see people doing this exten => s,n,MYSQL(Connect connid 127.0.0.1 acd acdpass acd) |
| 22:17.25 | bsdwarrior | how can I do the same thing with postgres ? |
| 22:18.10 | fiXXXerMet | Yikes, no idea. |
| 22:18.19 | nhuisman_work | any of you folks use cisco 79XX phones? What version of SIP firmware do you suggest? |
| 22:18.28 | nhuisman_work | 7940/7960.. |
| 22:19.21 | bsdwarrior | P003-08-6-00 works fine for us |
| 22:20.05 | bsdwarrior | any gurus around |
| 22:20.09 | nhuisman_work | do you have any experience with upgrading SKINNY to sip and then back? |
| 22:20.16 | nhuisman_work | i want to make sure I can back out of the sip firmware. |
| 22:20.41 | bsdwarrior | nhuis no sorry |
| 22:21.09 | nhuisman_work | did you have to upgrade in stages? |
| 22:21.19 | nhuisman_work | somewhere I read you can't go all the way to latest in one upgrade |
| 22:22.16 | bsdwarrior | thats possible. its a pita |
| 22:22.24 | puck | nhuisman_work: I still have 7905's, but can't get the latest firmware for them anymore. :( |
| 22:22.27 | c4t3l | what is the general consensus here regarding asterisk and GUI interfaces? |
| 22:22.36 | nhuisman_work | puck do you have cisco support? |
| 22:22.44 | syzygyBSD | c4t3l: don't |
| 22:22.54 | syzygyBSD | !gui |
| 22:23.02 | puck | nhuisman_work: no, but a friend who does checked and couldn't find it |
| 22:23.05 | syzygyBSD | hmm, maybe another command |
| 22:23.07 | syzygyBSD | ~gui |
| 22:23.07 | jbot | somebody said gui was (Graphical User Interface) Any system that uses graphics to represent the functions of a program. All Windows operating systems are GUI. or point-and-drool, double-click, drag-and-drop, ... or at http://pla-netx.com/linebackn/guis/index.html. Of course Real Programmers use the command line interface. See cli |
| 22:23.11 | nhuisman_work | puck, checking |
| 22:23.26 | nhuisman_work | puck, sip? |
| 22:23.35 | puck | c4t3l: I've starting playing around with some desktop integration goodness |
| 22:23.37 | syzygyBSD | hmm, I liked fender's quote about the gui... |
| 22:23.37 | puck | nhuisman_work: yeah |
| 22:23.58 | bsdwarrior | lol |
| 22:24.36 | nhuisman_work | <PROTECTED> |
| 22:24.44 | [TK]D-Fender | syzygyBSD, Which? |
| 22:25.01 | nhuisman_work | firmware i mean |
| 22:25.11 | bsdwarrior | hey unixdog |
| 22:25.15 | syzygyBSD | the one about learning asterisk with a gui being like .... |
| 22:25.25 | c4t3l | syzgyBSD: I've been working with * for a couple of years now and the company I currently work with has made the descision to sell a GUI based * :( |
| 22:25.25 | puck | nhuisman_work: just checking |
| 22:25.27 | [TK]D-Fender | syzygyBSD, thats not my quote.. |
| 22:25.30 | [TK]D-Fender | ~zeeek |
| 22:25.31 | jbot | methinks zeeek is someone who once said "learning asterisk using a GUI is like learning sex through masturbation. You'll never get to the good stuff." |
| 22:25.46 | syzygyBSD | ahh, but you were the one that knew the command |
| 22:25.48 | puck | nhuisman_work: 1.02.00 |
| 22:26.12 | [TK]D-Fender | syzygyBSD, But I almost never call it.... that's usually ManxPower |
| 22:26.53 | syzygyBSD | I use a gui for my configuration... made it myself, it is a textbox for each extension if it is "custom", or you can have it automatically do 10 or so different types of extensions |
| 22:26.55 | c4t3l | I have installed 3 systems in the last 2 weeks that should have only taken 1 hour tops, but because the GUI is there things have to re-programmed-very unhappy |
| 22:27.06 | nhuisman_work | puck, kolea.ifa.hawaii.edu/~nhuisman/CP7905080001SIP060412A.ZIP |
| 22:27.09 | nhuisman_work | 8.0 |
| 22:27.41 | puck | nhuisman_work: awesome, thanks! |
| 22:27.47 | nhuisman_work | err 8.0(1) |
| 22:27.54 | puck | good for a 7905G? |
| 22:27.56 | c4t3l | i personally hate the damn gui |
| 22:28.05 | nhuisman_work | it's listed for 7905s |
| 22:28.25 | puck | great, thank you! |
| 22:28.27 | nhuisman_work | SIP software for Cisco 7905 IP Phone - build 060412A - for Non-CallManager applications |
| 22:28.30 | nhuisman_work | np |
| 22:28.38 | nhuisman_work | anyone else want firmware ? hehe |
| 22:28.46 | c4t3l | you really cant do anything with it. Maybe I should look for another job eh?? hint hint... |
| 22:28.57 | puck | heh, I've always been disappointed that these phones don't do the XML menus... :( |
| 22:29.17 | puck | oh, my baby daughters awake, off to get her up |
| 22:29.51 | UnixDog | hey |
| 22:30.01 | UnixDog | wow more bsd people here |
| 22:30.18 | Daviey | Hi, for some reason Hangup isn't working in this example - any ideas? http://pastebin.com/d2ba0112a |
| 22:30.32 | UnixDog | all the bsd people to the right side for the room and all the linux people leave the channel |
| 22:30.34 | UnixDog | lol |
| 22:30.51 | Daviey | ./wc |
| 22:31.26 | nhuisman_work | that'd be one small circle jerk left :P |
| 22:32.25 | UnixDog | lol |
| 22:32.27 | G-nerd | what is a dialplan exactly, I just started to configure sip.conf |
| 22:32.44 | UnixDog | look at extensions.conf and learn |
| 22:32.47 | syzygyBSD | lol UnixDog |
| 22:32.58 | G-nerd | UnixDogg, am not talking to you |
| 22:33.25 | CanWood | Hey UnixDog, Free, Net, or OpenBSD? |
| 22:33.32 | syzygyBSD | A dialplan tells how all calls will be handled |
| 22:33.59 | G-nerd | well that is clear, I' ve read that in asterisk book too, but still don' t have a clue |
| 22:34.04 | UnixDog | G-nerd, how are to ever learn if you dont jump in and doggy paddle |
| 22:34.15 | CanWood | the consultant setting us up said "linux or you'll regret it" and I'm looking for horror/success stories with OpenBSD |
| 22:34.34 | UnixDog | a dial plain is how you write what functions your system can do and your users have access to |
| 22:34.47 | UnixDog | like callwaiting and callforwarding |
| 22:34.59 | UnixDog | look at extensions.conf and read it many times |
| 22:35.04 | UnixDog | it will come to you |
| 22:35.21 | syzygyBSD | well, I used freebsd for a while, it was very stable and had great disk access speeds, eventually I moved off just to keep management down to one OS |
| 22:35.27 | [TK]D-Fender | G-nerd, You've been answered. dialp = extensions.conf <--- |
| 22:35.49 | G-nerd | Accoording to Asterisk book 2nd edition I have to start with SIP.conf, not with extension.conf ok |
| 22:36.06 | syzygyBSD | I hope you don't believe everything you read... |
| 22:36.09 | UnixDog | also at a asterisk cli do core show applications |
| 22:36.19 | UnixDog | read it and compaire it to the dial plan |
| 22:36.25 | UnixDog | it will help you to learn |
| 22:36.27 | G-nerd | aaah ok D-Fender, so I have to read extension to know what' s going on |
| 22:36.40 | UnixDog | now go padawond and learn the ways of asterisk |
| 22:36.41 | G-nerd | ok thnx UnixDog |
| 22:36.54 | UnixDog | become one with your pbx |
| 22:36.59 | UnixDog | commune with it |
| 22:37.27 | UnixDog | its how we all learned and now its your turn |
| 22:37.40 | G-nerd | yeah, so is my wife, work, rent......:( |
| 22:38.13 | G-nerd | anyway I want to get Asterisk running before I sleep |
| 22:38.16 | syzygyBSD | actually, I learned by osmosis. I slept on the book |
| 22:38.23 | *** join/#asterisk dacs (n=haiger@unaffiliated/dacs) |
| 22:38.45 | UnixDog | then go get centpbx and install it |
| 22:38.54 | UnixDog | it comes with a functional dial plan |
| 22:39.13 | UnixDog | and learn to use it and come back here when your really ready to learn |
| 22:39.25 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 22:39.26 | Daviey | make samples ; comes with a "functional dial plan" :) |
| 22:39.36 | UnixDog | not really |
| 22:39.43 | UnixDog | semi functional |
| 22:40.26 | UnixDog | it wont have all the NANPA Vertical dial plan in it |
| 22:40.31 | Daviey | anyway.. why isn't my hangups working! |
| 22:40.43 | G-nerd | UnixDog, come back here when you are really know how to communicate |
| 22:41.42 | G-nerd | just say nothing and read here how D-Fender does, you can learn a lot from him |
| 22:41.59 | Daviey | seriously? |
| 22:42.30 | Daviey | D-Fender hasn't won any prizes for his people skills AIUI |
| 22:42.54 | UnixDog | G-nerd, get real and listen we are trying to help you to learn |
| 22:43.06 | puppet | $["${PHASEESTATUS}" = "0"]11? isnt that right syntax? |
| 22:43.10 | file | UnixDog: not all those across the land require a "fully functional" dialplan |
| 22:43.35 | UnixDog | if you need a pbx now and are not willing to take the time to learn the get centpbx install |
| 22:43.51 | UnixDog | 90% do from what I have learned over the years |
| 22:44.05 | UnixDog | they are to lazy to learn |
| 22:45.16 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 22:45.51 | [TK]D-Fender | "fully functional" means what exactly? What is this miraculous benchmark that all others must compare to? |
| 22:46.05 | syzygyBSD | every feature ever! |
| 22:46.33 | [TK]D-Fender | syzygyBSD, And since you could call anything an exten does a "feature", then your dialplan would be infinitley sized. |
| 22:46.36 | UnixDog | that it has full nanpa vertical dial plan |
| 22:46.47 | *** join/#asterisk Grash (n=olaf_mar@cdbil1-a1-2-23.ipcom.comunitel.net) |
| 22:46.47 | G-nerd | UnixDog, let me explain you this. If I have a whole free day I will know 70% of the whole Asterisk thing, ok. But as a father, husband and employee (as a programmer for embedded systems and server applications for windose) I don' t have all the time. So if you just don' t compare my lifestyle to yours, than you can at least show some understanding. And that is the last thing I say. |
| 22:46.57 | UnixDog | matches what most standard pbx systems have |
| 22:47.41 | UnixDog | I have 2 kids a x wife 2 dogs and a house with 2 roomates |
| 22:47.47 | jameswf-home | can it poor me a beer |
| 22:47.59 | [TK]D-Fender | UnixDog, O RLY? For me NANPA means being able to dial 7-10-11 digit #'s on the PSTN. But wait, that assumes the system even has anything to DO with the PSTN. If I make a PBX for use solely within the scope of 1 building, do I need to give a hoot about NANPA? No. |
| 22:48.46 | UnixDog | most pbx systems have full nanpa callforwarding callwaiting call return enable/disable |
| 22:48.52 | UnixDog | and more |
| 22:49.03 | [TK]D-Fender | G-nerd, The "samples" that can't with * are just that, "samples". You should trash sip.conf & extensions.conf right after you install and start from scratch and use the samples as nothing more than inspiration. |
| 22:49.08 | UnixDog | and thats what alot of companies and users look for when they say pbx |
| 22:49.09 | puppet | i love recieving faxes now ;P |
| 22:49.13 | puppet | -- Executing SipT38SwitchOver("SIP/83.140.41.46-080db660", "") <3 |
| 22:49.17 | jameswf-home | my PBX can burn CDs |
| 22:49.25 | [hC] | puppet: what are you using for t38? |
| 22:49.32 | [hC] | puppet: at the gateway/client ? |
| 22:49.39 | puppet | [hC]: not asterisk im sorry to say |
| 22:49.46 | [TK]D-Fender | jameswf-home, My PBX plays me movies on its 120" HT screen :p |
| 22:49.58 | [TK]D-Fender | jameswf-home, And used to make me coffee. |
| 22:49.59 | UnixDog | it also has to do full status checking to see whats enabled and disabled on the line |
| 22:50.22 | syzygyBSD | well, last time I will be here in this country, g'bye everybody! |
| 22:50.22 | [TK]D-Fender | UnixDog, Does your PBX make you coffee? I think you might be missing some important "features" there :) |
| 22:50.29 | G-nerd | D-Fender thnx for the advice. I have commented all out the conf-file, so I can follow the instructions step by step accoording to the Asterisk book. Later I'll create new conf files from scratch. But thnx |
| 22:50.45 | jameswf-home | We were going to buy an old coke machine and make it asterisk ready |
| 22:50.49 | UnixDog | I dont drink coffie but it is linked to mr house and controls my house with dial plan |
| 22:51.09 | [TK]D-Fender | UnixDog, Ok, close enough... |
| 22:51.19 | [hC] | puppet: what was that output from? |
| 22:51.29 | puppet | a fork of asterisk |
| 22:51.38 | jameswf-home | I should have it read dirty stories to me |
| 22:51.44 | puppet | a fork of asterisk |
| 22:51.50 | puppet | wrong window the last :) |
| 22:52.56 | UnixDog | the sad part is this. Microshaft has done a good job at training people not to touch config files let the gui do it. |
| 22:53.01 | [hC] | puppet: your own, or? |
| 22:53.11 | [TK]D-Fender | [hC], Callweaver |
| 22:56.24 | jameswf-home | dirty forker |
| 22:56.27 | *** join/#asterisk catharina (n=ask@78-21-204-113.access.telenet.be) |
| 22:56.28 | jameswf-home | :) |
| 22:56.43 | syzygyBSD | gotta love them forks |
| 22:57.04 | catharina | hello all |
| 22:57.14 | syzygyBSD | catharina!!! how are you? |
| 22:57.21 | puppet | jameswf-home: well nothing bad about asterisks, but after i tried CW, no laggy MoH on sip channel, better faxrecieving, i havent seen anything bad so far :) and no more ztdummy. But then it comes from asterisks so :) |
| 22:57.21 | catharina | very fine, thank you |
| 22:57.33 | catharina | except for my asterisk issue :-) |
| 22:57.35 | catharina | sorry |
| 22:57.42 | nhuisman_work | is it possible to setup two dhcp servers on the same layer 2 but configure one to only answer to certain mac addresses? |
| 22:58.01 | syzygyBSD | whats the issue? |
| 22:58.32 | syzygyBSD | nhuisman_work: yes.. but you have to configure the other one not to answer for it either |
| 22:59.10 | catharina | i've registered to a sip provider in my country. i have a voip-in number. dialing in and out works perfectly (most of the time) |
| 22:59.15 | *** join/#asterisk eldon (n=eldon@216.207.245.1) |
| 22:59.19 | syzygyBSD | I was running two on my network without realizing it... huge problem when you are trying to change settings |
| 22:59.26 | catharina | after a while, the dialing in functionality stops working |
| 22:59.38 | catharina | although it shows as registered |
| 22:59.49 | syzygyBSD | registered at your end... |
| 22:59.57 | UnixDog | that could be many issues catharina |
| 22:59.57 | catharina | I have to make an outbound call (or do a sip reload) and then the dial in works again |
| 23:00.19 | catharina | seems as though it times out or something or looses the registration at their end |
| 23:00.25 | syzygyBSD | catharina: your sip registration is longer than your providers. make your's refresh sooner |
| 23:00.30 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 23:01.07 | nhuisman_work | syzygyBSD, hmm yeah. I wanted to test my asterisk server in the same network as the current voip |
| 23:01.12 | nhuisman_work | maybe i'll just figure out another network configuration to test. |
| 23:01.22 | catharina | is that the defaultexpiry option in sip.conf ? |
| 23:01.59 | syzygyBSD | I use zap as my provider, or IAX. Let me check |
| 23:02.07 | lzhang | dhadskjfdskjfakslnkndkfjbdskjfbskjbakbvjkqkdwedfwksdjcvkcfgrwkjbvsfkgarwkufgeriferwjhgwrg |
| 23:02.18 | syzygyBSD | yes the keyboard works |
| 23:02.24 | catharina | lol |
| 23:03.18 | syzygyBSD | catharina: registertimeout=3600 will reregister every hour |
| 23:03.27 | syzygyBSD | wait, nm |
| 23:04.17 | syzygyBSD | defaultexpirey |
| 23:04.23 | syzygyBSD | http://www.voip-info.org/wiki-Asterisk+config+sip.conf |
| 23:04.27 | syzygyBSD | the better place to look |
| 23:04.39 | syzygyBSD | anyway, make it reregister sooner |
| 23:04.48 | syzygyBSD | boss just brought beer, I am signing off now |
| 23:04.51 | syzygyBSD | good luck |
| 23:04.58 | G-nerd | Hello, what is trunk mean in Asterisk context? |
| 23:05.01 | catharina | bye |
| 23:05.02 | catharina | thanks |
| 23:05.22 | CanWood | from SIP to beer. A natural transition |
| 23:05.48 | mvanbaak | CanWood: to understand and like SIP you need shitloads of beer |
| 23:05.55 | mvanbaak | so yeah, it's pretty natural |
| 23:06.12 | G-nerd | Hello, what is trunk mean in Asterisk context? |
| 23:07.23 | *** join/#asterisk maldous (n=user@f28115.upc-f.chello.nl) |
| 23:07.25 | maldous | hi. |
| 23:08.04 | maldous | does anyone know if asterisk will run on a ADM5120 or if there would be codec limitations? |
| 23:08.15 | mvanbaak | G-nerd: can mean several things |
| 23:08.23 | mvanbaak | G-nerd: are you referring to the version |
| 23:08.38 | mvanbaak | G-nerd: or to something like 'SIP-trunk, IAX-trunk' |
| 23:08.49 | G-nerd | like what? I guess the latest ones |
| 23:09.52 | drmessano | G-nerd, are you reading the book? |
| 23:09.54 | catharina | does anyone here have some experience with a cisco 7912 ? |
| 23:10.13 | G-nerd | yes |
| 23:10.37 | G-nerd | the electronic one, I' m almost blind |
| 23:10.45 | catharina | lol |
| 23:10.49 | catharina | sorry |
| 23:12.33 | G-nerd | no reason for sorry cat |
| 23:12.44 | G-nerd | I'm a joke my whole life |
| 23:12.58 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 23:13.24 | *** part/#asterisk `paul (n=aldee@125.252.68.68) |
| 23:14.34 | *** join/#asterisk CanWood (n=chatzill@24.108.64.80) |
| 23:15.10 | puppet | hmm something strange here i fax number 75, and the channel says it takes 70:s settings |
| 23:15.18 | nhuisman_work | trying to load a 7940 via dhcp and tftp and it's saying load id incorrect, any ideas? |
| 23:18.10 | puppet | how come it always picks gsm when i dont even haev it in the list of the once i want to use |
| 23:18.14 | catharina | just a guess: check if the image version in SIPDefault.cnf matches the image on your tftp server |
| 23:22.12 | puppet | hmm cant i change codec on just one incoming number? do i really have to change it on all numbers? |
| 23:24.05 | *** join/#asterisk vrtk (n=bruno@201.9.57.7) |
| 23:26.35 | [TK]D-Fender | catharina, You running your server behind NAT? |
| 23:26.57 | catharina | yes |
| 23:27.06 | catharina | or |
| 23:27.10 | catharina | actually no |
| 23:27.12 | catharina | sorry |
| 23:27.16 | [TK]D-Fender | catharina, sounds like the UDP window is closing behind you |
| 23:27.21 | catharina | it runs on an openwrt box |
| 23:27.27 | catharina | which has a public adress on the outside |
| 23:27.35 | [TK]D-Fender | catharina, * on WRT w/ a WAN ip direct? |
| 23:27.36 | catharina | what do you mean [TK]D-Fender ? |
| 23:27.44 | catharina | yes |
| 23:27.50 | [TK]D-Fender | catharina, check your firewall on it. |
| 23:28.01 | [TK]D-Fender | cath make sure 5060,10000-20000 are open |
| 23:28.13 | catharina | incoming ? |
| 23:28.14 | [TK]D-Fender | catharina, (all UPD) |
| 23:28.18 | [TK]D-Fender | catharina, both ways |
| 23:28.28 | catharina | well it's open outgoing |
| 23:28.52 | catharina | and it's a statefull firewall |
| 23:29.13 | [TK]D-Fender | catharina, fat lod of good that'll do *... UDP is stateless. |
| 23:29.22 | catharina | ah ok |
| 23:29.27 | catharina | good point |
| 23:29.30 | catharina | not an expert here |
| 23:29.33 | catharina | :-) |
| 23:29.40 | [TK]D-Fender | catharina, np :) |
| 23:29.53 | catharina | you said the UDP window is closing, is there anything I can do on my side to keep it open ? |
| 23:30.00 | catharina | a sort of keepalive or something ? |
| 23:30.46 | [TK]D-Fender | catharina, that was under the assumption of NAT which isn't supposed to be the case. |
| 23:31.18 | [TK]D-Fender | catharina, I'd still look at your firewall mind you, and it would probably be a decent idea to enable Qualify on your SIP peer to your ITSP |
| 23:31.33 | catharina | what is qualify ? |
| 23:32.30 | *** join/#asterisk Paladine (n=paladine@ns2.scs-live.com) |
| 23:33.01 | catharina | ah ok |
| 23:33.02 | catharina | found it |
| 23:33.13 | Paladine | wow hope I am in the right channel given how many are in the topic lol |
| 23:33.19 | *** join/#asterisk etfonhomey (n=chatzill@74-143-197-2.static.insightbb.com) |
| 23:33.25 | Paladine | I am basically just looking for a little bit of advice |
| 23:34.49 | [TK]D-Fender | Paladine, just ask it. |
| 23:34.49 | Paladine | I would like to install asterisk for 2 pourposes, firstly because of the saving we can make by switching to VOIP and secondly because I need to collect evidence for a criminal case involving harassing phonecalls |
| 23:34.50 | Paladine | this would be at my home so we only have 1 line |
| 23:35.14 | Paladine | I have a spare server (real server not little PC) and I need to know what other equipment I need to plug it into the UK telephone system |
| 23:35.16 | nhuisman_work | can't you just use the phone bill for harassment logs? |
| 23:35.38 | Paladine | phonebill only gives me the number sI call not the numbers people call me from |
| 23:35.39 | nhuisman_work | you'll need an interface card |
| 23:35.49 | nhuisman_work | Paladine, i'm sure the phone company could give you that information |
| 23:35.51 | Paladine | and I need to actually record the calls for evidence (as in the conversation) |
| 23:36.07 | Paladine | no they can't, the phone company don't log incoming calls, only outgoing |
| 23:36.13 | Paladine | it is a common problem in the UK |
| 23:36.14 | [TK]D-Fender | Paladine, Digium TDM01B for your POTS line. |
| 23:36.15 | nhuisman_work | wow that's pretty stupid |
| 23:36.30 | Paladine | TK is that a PCI card? |
| 23:36.30 | [TK]D-Fender | Paladine, * can record calls and you get the callerID, etc... |
| 23:36.31 | nhuisman_work | of them. |
| 23:36.38 | [TK]D-Fender | Paladine, Yes |
| 23:37.29 | drmessano | You also need to check with local law enforcement and find out what they need from you specifically |
| 23:37.52 | [TK]D-Fender | Paladine, And verify the legality of your recording |
| 23:37.58 | drmessano | Unfortunately, a report that makes sense to you may look stupid to them |
| 23:38.21 | drmessano | So specifics are good.. |
| 23:38.36 | Paladine | already verified, it is perfectly legal in the UK for an individual to record calls |
| 23:38.45 | Paladine | only companies have to inform the other party |
| 23:38.52 | catharina | [TK]D-Fender: i've been running a tcpdump on my box the entire evening. |
| 23:39.00 | nhuisman_work | yeah but will law enforcement be able to use your recording against the offending party |
| 23:39.02 | catharina | i can see sip registrations every 10 minutes |
| 23:39.03 | nhuisman_work | is what you want to make sure of. |
| 23:39.07 | catharina | so registration is ok |
| 23:39.25 | catharina | a few minutes ago, incoming phones were not working |
| 23:39.48 | adelas | can anyone here help me with setting asterisk up to take 2 network cards? |
| 23:39.49 | [TK]D-Fender | catharina, Still sounds like a networking issue to me... |
| 23:39.49 | Paladine | drmessano, also done, I need to provde a sample of voice calls recorded (not all of the calls just a few) plus a log of dates/times durations of calls and number called from |
| 23:39.55 | catharina | so in the trace I can see a sip cancel message |
| 23:40.01 | adelas | i have my sip.conf to be listening at 0.0.0.0 |
| 23:40.19 | [TK]D-Fender | adelas, thats good so far |
| 23:40.21 | nhuisman_work | Paladine, if you are just setting up asterisk for that I might suggest just getting a tape recorder and some caller id box. |
| 23:40.26 | Paladine | nhuisman_work, yes they can, it is police who have asked me to take this action |
| 23:40.27 | nhuisman_work | but otherwise go for it. |
| 23:40.33 | drmessano | Paladine: and they will accept a CDR reports from an Asterisk phone system |
| 23:40.43 | Paladine | tape recorder won't work we have dect phones |
| 23:40.50 | [TK]D-Fender | drmessano, CDR is nice, recording, more so |
| 23:40.54 | Paladine | so there is no socket to plug the recorder into |
| 23:41.01 | nhuisman_work | i see |
| 23:41.02 | drmessano | They'll want logs |
| 23:41.06 | *** join/#asterisk cesar_CR (n=cesar@201.202.156.2) |
| 23:41.09 | adelas | [TK]D-Fender, but asterisk won't take incoming traffic from the 2nd nic. I know the 2nd nick works b/c of the http server and ping services |
| 23:41.21 | [TK]D-Fender | Paladine, well i told you the kind of card you'll want |
| 23:41.22 | drmessano | and logs in a format they are familiar with and that are admissible in court |
| 23:41.24 | adelas | is there any other problem? |
| 23:41.25 | Paladine | I presume asterisk can log calls effectively, failing that I suspect just the file creation time and date would be sufficient |
| 23:41.42 | Paladine | TK yeah I am trying to find one now, thanks for that |
| 23:41.54 | [TK]D-Fender | adelas, Well you haven't described whats on each of these NIC's, showed us your configs, etc... |
| 23:42.08 | adelas | each nic is a different subnet |
| 23:42.29 | [TK]D-Fender | adelas, elaborate..... |
| 23:42.36 | adelas | nic1 is like 10.11.12.x, while nic2 is 10.11.14.x |
| 23:43.01 | adelas | i was hoping for asterisk to be able to listen and take traffic on both nics |
| 23:43.04 | Paladine | TK, wow that is more expensive than I expected |
| 23:43.47 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 23:44.08 | [TK]D-Fender | adelas, set "localnet" for each of them |
| 23:44.37 | [TK]D-Fender | Paladine, You might get away with an X100P clone, but they are often flakey.... |
| 23:45.10 | [TK]D-Fender | Paladine, And I'm thinking that you'll want decent call quality and disconnect supervision. |
| 23:45.18 | [TK]D-Fender | Paladine, this is something tricky in the UK |
| 23:45.25 | adelas | [TK]D-Fender, what do you mean by "localnet" |
| 23:45.30 | catharina | i'm going to bed |
| 23:45.33 | catharina | thanks all for the help |
| 23:45.50 | catharina | see you later |
| 23:46.38 | Paladine | TK, yeah but $160 is waaay expensive when you are a student with an adult dependent, a 2 year old and a mortgage hehe |
| 23:46.42 | adelas | [TK]D-Fender, there are 2 physically differnet networks |
| 23:47.23 | [TK]D-Fender | Paladine, go find an X100P then. |
| 23:48.11 | [TK]D-Fender | adelas, under general add : localnet=10.11.14.0/24 |
| 23:48.14 | [TK]D-Fender | adelas, under general add : localnet=10.11.12.0/24 |
| 23:48.59 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177583847.dsl.bell.ca) |
| 23:51.25 | adelas | [TK]D-Fender, sorry i'm a noob, but you mean general under the sip.conf? |
| 23:52.52 | mvanbaak | yes |
| 23:52.54 | Paladine | [TK]D-Fender, would this be suitable? http://tinyurl.com/3cnzqh |
| 23:54.06 | *** join/#asterisk AndyGraybeal (n=andy@node216.36.251.72.1dial.com) |
| 23:54.46 | *** join/#asterisk SwK (n=SwK@user-24-214-55-149.knology.net) |
| 23:55.08 | drmessano | I think this is a better x100p: http://www.x100p.com/products/FXO.php |
| 23:58.30 | mvanbaak | drmessano: no |
| 23:58.45 | mvanbaak | all them clones are as bad as horsedoodles |
| 23:58.48 | mvanbaak | get a TDM |
| 23:59.21 | mvanbaak | the X100P is awefull |