| 00:05.36 | *** join/#asterisk EnigmaCurry (n=user@c-24-10-239-16.hsd1.ut.comcast.net) |
| 00:07.48 | *** join/#asterisk coppice (n=chatzill@39.192.17.210.dyn.pacific.net.hk) |
| 00:09.40 | obnauticus | De_Mon it says `wholesellers only' does that mean I can't sign up? |
| 00:09.44 | obnauticus | Because they have pretty freaking good deals. |
| 00:09.48 | obnauticus | and i want to be able to use them for my own stuff. |
| 00:13.25 | *** join/#asterisk felix_da_catz (n=felix@c-76-31-59-80.hsd1.tx.comcast.net) |
| 00:16.49 | *** join/#asterisk codeshah (n=codeshah@S01060011092d0063.ed.shawcable.net) |
| 00:20.07 | luke-jr | obnauticus: it means not to expect consume-level support, and you will need to pay a bit more |
| 00:20.36 | luke-jr | there's a fee on deposits under $500, for example |
| 00:20.42 | *** join/#asterisk Siya (n=djerk@217-195-248-251.dsl.easynet.nl) |
| 00:21.02 | *** join/#asterisk JayTee52 (n=jforde05@c-69-137-243-25.hsd1.in.comcast.net) |
| 00:21.25 | *** join/#asterisk bkw_ (n=brian@adsl-64-149-41-151.dsl.tul2ok.sbcglobal.net) |
| 00:23.09 | obnauticus | luke-jr, well their deals make up for that right? |
| 00:23.25 | luke-jr | define "deals" |
| 00:23.28 | luke-jr | I've seen cheaper |
| 00:24.03 | codeshah | hmm hey guys I just installed asterisk on ubuntu with apt-get, but when I try to go to /etc/asterisk I get permission denied... I am trying to go as su ... anyone else use the ubuntu package/ |
| 00:25.23 | obnauticus | like-jr where can you suggest then :/ |
| 00:25.25 | obnauticus | for an enduser... |
| 00:25.54 | obnauticus | well i don't need support but i just need good service, and an inbound as well as termination would be good too, but I can use ipkall because i'm in washington anyway. |
| 00:26.06 | fujin | codeshah: yes, you'll need to be root (sudo -i) |
| 00:26.48 | obnauticus | Also, I'm currently using IpKall and it's not receiving my DTMF tones for some reason |
| 00:26.51 | obnauticus | http://pastebin.ca/771496 <-- debug output |
| 00:27.09 | *** join/#asterisk phix (i=threat@123-243-44-131.tpgi.com.au) |
| 00:27.12 | phix | hey |
| 00:27.12 | codeshah | fujin, thanks . |
| 00:27.39 | phix | Does this look correct for dial plan 2 on PSTN line of a SPA3102? --> (S0<: 100@10.0.0.1 :5060>) |
| 00:27.50 | *** part/#asterisk bkw_ (n=brian@adsl-64-149-41-151.dsl.tul2ok.sbcglobal.net) |
| 00:28.02 | phix | dial plan 1 is used to incoming calls, I set dial plan 2 for outgoing calls on PSTN line |
| 00:28.08 | phix | (FXO port) |
| 00:33.05 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-dcf7d3428e15469e) |
| 00:35.42 | twisted | hi file |
| 00:36.04 | file | how goes? |
| 00:36.23 | twisted | routing hell |
| 00:36.25 | twisted | you? |
| 00:36.42 | file | meh, packing up my laptop to send it in for repair |
| 00:36.57 | twisted | fun |
| 00:41.05 | obnauticus | how do i get extreme debugging output in asterisk |
| 00:41.06 | obnauticus | i forgot... |
| 00:41.11 | codeshah | Im on the asterisk book, trying to setup a basic voip connection ... but I'm a bit confused about the whole DHCP server thingie and how to set that up |
| 00:41.19 | obnauticus | oh debug channel all |
| 00:41.20 | obnauticus | lol |
| 00:41.21 | codeshah | are there other resources I should look at |
| 00:42.06 | phix | obnauticus: ummm 65535 ? |
| 00:42.30 | obnauticus | .phix, i just want to see all the stuff going on |
| 00:42.42 | obnauticus | because something isn't working |
| 00:42.43 | phix | -d 65535 |
| 00:42.47 | obnauticus | http://pastebin.ca/771496 <-- |
| 00:43.25 | obnauticus | phix, it's still not saying anything |
| 00:43.46 | phix | it should be in /var/log/syslog |
| 00:44.19 | phix | lol oops |
| 00:44.27 | phix | use -vvvvvvvvvvvvvvvvvvv |
| 00:44.29 | obnauticus | ya |
| 00:44.32 | obnauticus | that's what im doing |
| 00:44.34 | obnauticus | it's to the max |
| 00:44.36 | obnauticus | i need more than that. |
| 00:44.37 | phix | instead of -d :) hehe |
| 00:44.39 | phix | oh |
| 00:44.46 | phix | *shrugs* |
| 00:44.50 | obnauticus | I'm using ipkall, and it's not seeing my dtmf tones |
| 00:44.51 | obnauticus | for some reason |
| 00:44.55 | obnauticus | I can hear |
| 00:44.59 | obnauticus | but i have a menu up |
| 00:44.59 | phix | strace asterisk :) |
| 00:45.04 | obnauticus | strace? |
| 00:45.08 | phix | yes |
| 00:45.18 | phix | prints out all system functions called by asterisk |
| 00:45.18 | TJNII | codeshah: You mean setting up a DHCP server to provision your phones? |
| 00:45.47 | phix | obnauticus: That is very verbose :) |
| 00:45.51 | obnauticus | lol |
| 00:45.53 | obnauticus | phix i saw one |
| 00:45.54 | obnauticus | once* |
| 00:46.01 | obnauticus | some dude told me to do something and it spat out A LOT of stuff |
| 00:46.06 | obnauticus | like too much to comprehend |
| 00:46.18 | obnauticus | it told me frames that it was sending |
| 00:46.19 | obnauticus | and stuff |
| 00:46.22 | phix | yeah, like 1/2 the source code worth of stuff |
| 00:47.21 | TJNII | ~vonage |
| 00:47.22 | jbot | it has been said that vonage is a bunch of monkeys |
| 00:47.30 | *** join/#asterisk ai-a (n=jake2@cpc5-oxfd1-0-0-cust523.oxfd.cable.ntl.com) |
| 00:48.07 | codeshah | TJNIT, is that what it is? I need to read up a bit more ... I essentially have this x-lite softphone ... and going through the SIP config part of the asterisk book . I am new to telephony, so may have missed something? |
| 00:48.29 | phix | TJNII: yay |
| 00:48.29 | codeshah | and that part of the book mentions setting up a DHCP server under essential server components |
| 00:49.03 | TJNII | codeshah: Which section of the book? I'd like to take a look at it |
| 00:49.17 | TJNII | Because I didn't set up a DHCP server specifically for asterisk |
| 00:49.23 | codeshah | TJNIT, page 85 ... this is the 2nd edition rls . |
| 00:49.31 | codeshah | 2007 . from Oreilly . |
| 00:49.40 | codeshah | dunno if the 1st free one has it lemme check |
| 00:50.27 | TJNII | No |
| 00:50.37 | TJNII | Page 85 of edition 1 doesn't mention DHCP |
| 00:51.20 | TJNII | Hmmmm... asterisk.org says "You have an error in your SQL syntax" |
| 00:51.43 | codeshah | TJNIT, hmm you are right ... the two books are very different on that part . |
| 00:51.53 | codeshah | TJNIT, maybe I should just go through the 1st edition .. |
| 00:52.29 | TJNII | Well, you need a DHCP server (and probably already have one) to give your devices IPs behind the NAT. |
| 00:52.57 | TJNII | It is possible to make your phones auto-provision through DHCP, but that is a feature, not a necessity |
| 00:53.27 | codeshah | hmm k |
| 00:54.19 | TJNII | And if you're using a softphone you really don't have to worry about it |
| 00:54.45 | TJNII | But don't try to run asterisk and the softphone on the same computer as they will both try to bind to port 5060 and it will not work |
| 00:55.12 | TJNII | There are ways around that, but for a newbie its easier to jut use another machine |
| 00:55.28 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 00:57.43 | codeshah | hmm /etc/asterisk/ has no config files ... weird . |
| 00:58.09 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 00:58.44 | TJNII | codeshah: You install from scratch? |
| 00:58.59 | codeshah | TJNIT, actually no, I installed the ubuntu package . |
| 00:59.04 | *** part/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-dcf7d3428e15469e) |
| 00:59.13 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 00:59.13 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 00:59.51 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-4a358ff69005d88b) |
| 00:59.53 | codeshah | let me purge and reinstall |
| 01:00.24 | TJNII | codeshah: There is a default config file build option in the source package, I don't know about ubuntu |
| 01:00.32 | TJNII | And my nick is TJNII |
| 01:00.33 | *** join/#asterisk PaulAviles (n=Miranda@dsl-7-36.cofs.net) |
| 01:00.50 | codeshah | sorry about that :) . sure, let me go to the source package |
| 01:01.16 | obnauticus | would anyone here know why I am not able to receive (on asterisk end) from a SIP trunk? |
| 01:01.20 | obnauticus | it's on ipkall |
| 01:01.36 | obnauticus | http://pastebin.ca/771496 <-- that's what it looks like :/ |
| 01:02.43 | PaulAviles | hey guys, I have a tricky question |
| 01:03.31 | tzafrir | hmmm, /etc/asterisk in debs may be is asterisk-config . But he has already left... |
| 01:04.10 | PaulAviles | has anyone compared the quality of asterisk vs like Skype? I was actually very disapointed with asterisk |
| 01:04.46 | obnauticus | PaulAviles, with skypes quality and commerical codec, you loose felxability. |
| 01:04.59 | obnauticus | Where with asterisk you make a compromise for quality to get a lot of flexibility. |
| 01:05.19 | PaulAviles | felxibility will never win over quality.... |
| 01:05.42 | obnauticus | ,,, |
| 01:05.42 | PaulAviles | makes no difference if you have 1000 buttons to press if you cannot hear the other side.... |
| 01:05.47 | obnauticus | I'm not going to argue. |
| 01:06.01 | tzafrir | PaulAviles, g722 and other codecs of higher sample rate are making their way into Asterisk |
| 01:06.15 | tzafrir | not sure about speex/wb, though |
| 01:06.15 | obnauticus | tzafrfir how much bandwidth does g722 take? |
| 01:06.19 | PaulAviles | is not an argument, i am not fighting.. is just an observation.... |
| 01:06.47 | obnauticus | Flexability (ie. using existing phones as opposed to having a computer attached to a device which is logged into Skype) |
| 01:07.24 | obnauticus | In the longrun it costs the enduser more to run skype as their primary voip solution (if they want PSTN hookups) than an asterisk box. |
| 01:08.01 | PaulAviles | well yes and no.. in our case we were using on my end xlite on a pc and we tried to do the same with another remote location and had to switch to skype because of quality |
| 01:08.33 | obnauticus | ?skype |
| 01:08.38 | obnauticus | ~skype |
| 01:08.38 | jbot | Skype is the bastard child of telephony. It uses a proprietary protocol for which only commercial channel drivers exist, all of which are DISGUSTING hacks at best. Forget about using Skype with Asterisk... |
| 01:08.38 | obnauticus | :/ |
| 01:08.56 | *** part/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 01:09.40 | *** join/#asterisk BBHoss (n=hoss@146.229.181.183) |
| 01:09.55 | PaulAviles | no, i am not trying to use it with asterisk, all I am saying and jbot you make an even worst point, if they are so bad how in the bloody hell is their quality better? |
| 01:10.39 | PaulAviles | also, I am not pushing for skype either.. |
| 01:10.40 | obnauticus | ya jbot |
| 01:10.40 | obnauticus | Your point makes no sense. |
| 01:10.40 | obnauticus | Why did you even say that? |
| 01:10.40 | obnauticus | ~skype |
| 01:10.40 | jbot | Skype is the bastard child of telephony. It uses a proprietary protocol for which only commercial channel drivers exist, all of which are DISGUSTING hacks at best. Forget about using Skype with Asterisk... |
| 01:10.44 | obnauticus | Ahh |
| 01:10.46 | obnauticus | but that makes sense |
| 01:11.34 | PaulAviles | so, only commercial channel drivers are better than anything currently supported in asterisk? |
| 01:11.43 | PaulAviles | yeah.. that makes sense..... |
| 01:12.05 | obnauticus | you tell em PaulAviles |
| 01:12.09 | obnauticus | I don't think he's gonna argue |
| 01:12.13 | obnauticus | he's clearly incorrect. |
| 01:12.40 | PaulAviles | again, I am not arguing......maybe you want to and that is another story... |
| 01:12.48 | obnauticus | ya you're right |
| 01:12.51 | obnauticus | I'm not gonna argue. |
| 01:12.56 | PaulAviles | whatever.. |
| 01:12.58 | JT | PaulAviles: jbot doesn't make points |
| 01:13.02 | obnauticus | rofl |
| 01:13.33 | obnauticus | JT you spoiled it... you really did. |
| 01:14.30 | PaulAviles | so, lets ask the questions in a different way.. how can you achieve better quality using asterisks that "resembles" commercial channel existing drivers that are DISGUSTING hacks without using the DISGUSTING hacks? |
| 01:14.38 | PaulAviles | happy now? |
| 01:14.47 | obnauticus | ask jbot. |
| 01:14.51 | obnauticus | was that directed towards me? |
| 01:14.58 | JT | g.722 is higher quality than g.711 |
| 01:14.59 | PaulAviles | anyone... |
| 01:15.02 | JT | is that what you meant |
| 01:15.17 | PaulAviles | g722 is what skype? |
| 01:15.20 | obnauticus | no |
| 01:15.24 | obnauticus | skype isn't using an open codec. |
| 01:15.41 | coppice | no |
| 01:15.56 | obnauticus | They took voip |
| 01:15.59 | obnauticus | and wiped poop allover it |
| 01:16.02 | obnauticus | and made the poop popular |
| 01:16.04 | obnauticus | it's disgusting. |
| 01:16.20 | PaulAviles | ok, pardon my ignorance in all the codecs, is there anything similar in quality then? I think xlite uses ulaw |
| 01:16.22 | *** join/#asterisk codeshah (n=codeshah@S01060011092d0063.ed.shawcable.net) |
| 01:16.39 | coppice | similar to what? |
| 01:16.50 | PaulAviles | the quality of the DISGUSTING hacks... |
| 01:17.10 | coppice | why do you want something like skype? |
| 01:17.16 | PaulAviles | no |
| 01:17.17 | PaulAviles | point is |
| 01:17.23 | obnauticus | was jbot wrong PaulAviles? |
| 01:17.28 | obnauticus | Was what he said incorrect? |
| 01:17.38 | PaulAviles | let me elaborate.. |
| 01:18.10 | *** join/#asterisk Hadi- (n=Hadi@CPE001310492769-CM001225e00576.cpe.net.cable.rogers.com) |
| 01:18.11 | *** part/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-4a358ff69005d88b) |
| 01:18.20 | Hadi- | Hello... anyone here using A2Billing? |
| 01:18.27 | obnauticus | I'm not. |
| 01:18.31 | Hadi- | just had a quick question |
| 01:18.43 | tzafrir | hmmm, /etc/asterisk in debs may be is asterisk-config |
| 01:18.53 | tzafrir | codeshah, --^ |
| 01:18.57 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-a3c4e1aebbf1d426) |
| 01:19.09 | PaulAviles | take an asterisk sys.. setup an internal phone cisco 7960. setup a remote extension overseas. Use xlite on pc. try talking.. bad quality. install DISGUTING hacks on both end points and woila... better bloody quality.. |
| 01:19.13 | codeshah | tzafrir, hey . whats up . :) |
| 01:19.30 | obnauticus | PaulAviles, you have a cisco 7960? |
| 01:19.33 | PaulAviles | yes |
| 01:19.37 | obnauticus | Because I just got one off'a ebay and it's in the mail |
| 01:19.38 | obnauticus | how is it? |
| 01:20.02 | PaulAviles | well, is greyish, have buttons and a handset... |
| 01:20.09 | obnauticus | I know that much. |
| 01:20.13 | obnauticus | how does it work? |
| 01:20.13 | Hadi- | any other asterisk billing system you guys can recommand |
| 01:20.17 | JT | PaulAviles: being a sarcastic idiot towards the whole channel, not a good way to get help |
| 01:20.22 | Hadi- | for calling card and wholesale termination |
| 01:20.32 | obnauticus | JT, he's dissin our bot. |
| 01:20.41 | PaulAviles | diff on being sarcasting and having humor JT... |
| 01:20.47 | obnauticus | How rude! |
| 01:20.53 | JT | sarcasting, interesting word |
| 01:21.09 | tripps | ~sarcasting |
| 01:21.15 | tripps | : |
| 01:21.17 | tripps | :) |
| 01:21.19 | obnauticus | ~help |
| 01:21.26 | PaulAviles | fat finger.. I think the phone is nice, but.. there are some features I don't like |
| 01:21.44 | PaulAviles | the boot process is a pain as you need every time a tftp |
| 01:21.46 | obnauticus | PaulAviles I think you changed jbot's opinion on the topic. |
| 01:21.48 | obnauticus | let's see eh? |
| 01:21.50 | obnauticus | ~skype |
| 01:21.50 | jbot | Skype is the bastard child of telephony. It uses a proprietary protocol for which only commercial channel drivers exist, all of which are DISGUSTING hacks at best. Forget about using Skype with Asterisk... |
| 01:21.58 | obnauticus | Nope...you wern't convincing enough. |
| 01:22.21 | tripps | PaulAviles: call john chambers and see if they'll rewrite the sip firmware |
| 01:23.25 | PaulAviles | the firmware on the phones does have a direct impact with quality. take the same phone on the same asterisk and switch the firmware to sccp and the sound is a lot better than sip |
| 01:24.04 | tripps | PaulAviles: sip is just a protocol. better codec if you want better sound |
| 01:24.55 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-636ce60305150aee) |
| 01:24.58 | PaulAviles | so what has the best quality and lower bandwidth requirements? |
| 01:24.58 | JT | PaulAviles: because cisco's implementation of sip is shit |
| 01:25.04 | JT | PaulAviles: avoid cisco :) |
| 01:25.26 | PaulAviles | maybe you are correct.. |
| 01:25.44 | PaulAviles | what do you recommend? |
| 01:25.51 | tripps | however the 7970's are great phones as well, though you need some earthmoving equipment to get them working with * |
| 01:26.01 | JT | polycom |
| 01:26.04 | tripps | ~polycom |
| 01:26.04 | jbot | polycom is, like, the manufacturer of one of the best IP phones in the market. http://polycom.com - Note: Here is where you can get some downloads: http://www.polycom.com/resource_center/0,,pw-6812-12612,00.html |
| 01:26.21 | PaulAviles | does it also need a tftp? |
| 01:26.37 | JT | polycom can use tftp, ftp, http, https |
| 01:26.47 | Hadi- | ~cisco |
| 01:26.48 | jbot | cisco makes the routers that move the pr0n across the internet at a very high rate of exchange. a company full of patent protected punks! |
| 01:26.50 | *** join/#asterisk TJNII (n=TJNII@209.234.89.226) |
| 01:27.06 | PaulAviles | well, that is a way to put it... |
| 01:27.16 | PaulAviles | you forgot rich too... |
| 01:27.30 | PaulAviles | or at least some of them are... |
| 01:27.55 | BBHoss | anyone here have a AA50 firmware (Asterisk Appliance), or access to the support site? |
| 01:31.56 | *** join/#asterisk Netgeeks (n=chris@gw0.office1.talkplus.com) |
| 01:34.08 | PaulAviles | jt so the polycoms need to boot on power on from a remote source like the ciscos? |
| 01:34.30 | obnauticus | hey JT |
| 01:34.34 | obnauticus | No application 'SetCallerId' for extension <-- what does that mean? |
| 01:34.57 | Qwell | obnauticus: means you're using something that's been deprecated for over 2 years |
| 01:35.10 | obnauticus | uhh |
| 01:35.12 | obnauticus | nuts? |
| 01:35.38 | Qwell | Set(CALLERID(num)=1234) |
| 01:36.36 | obnauticus | Right now i have: exten => _394.,1,SetCallerId,Name <Number> |
| 01:36.43 | obnauticus | but obviously replace name with the name and # with the # |
| 01:38.01 | *** join/#asterisk asdx (n=diego@adsl-146-228.click.com.py) |
| 01:38.48 | TJNII | Did any IAX port options change from 1.2 to 1.4 |
| 01:39.00 | TJNII | My IAX port is closed and I con't understand why |
| 01:39.20 | Qwell | TJNII: how are you checking if it's open? |
| 01:39.34 | PaulAviles | are you loading it? |
| 01:40.21 | TJNII | Error loading module 'iax2': /usr/lib/asterisk/modules/iax2.so: cannot open shared object file: No such file or directory |
| 01:40.26 | TJNII | Well, that would do it.... |
| 01:40.41 | PaulAviles | th4e file does not exist you will not be able to use it |
| 01:40.49 | PaulAviles | did you manually compile it? |
| 01:40.59 | JT | obnauticus: well and truly deprecated syntax |
| 01:41.34 | Qwell | TJNII: are you trying to load a module called "iax2"? |
| 01:41.52 | TJNII | Yea, that was wrong |
| 01:41.55 | TJNII | I got it now |
| 01:42.04 | PaulAviles | if the files does not exist inside modules it will not load it ... |
| 01:42.14 | TJNII | And to answer your earlier question I'm using nmap 127.0.0.1 -sU |
| 01:42.59 | TJNII | I see "Binding IAX2 to default address 0.0.0.0:4569." |
| 01:44.31 | TJNII | Hmmm. Well, a nmap 127.0.0.1 -P0 -sU show it filtered |
| 01:45.05 | TJNII | But other asterisk servers cannot connect in. They say "Cause 3 - No route to destination" |
| 01:45.23 | BBHoss | qwell, do you know where i can get firmware for the aa50? |
| 01:45.29 | fujin | Have you established a route to that destination? |
| 01:45.31 | Qwell | BBHoss: digium.com |
| 01:45.34 | fujin | (i.e.; configured an iax friend) |
| 01:45.36 | BBHoss | where? |
| 01:45.44 | Qwell | ABE portal I think |
| 01:45.51 | Qwell | I don't know - I don't have access :p |
| 01:45.55 | BBHoss | damn |
| 01:45.58 | Qwell | support can show you where |
| 01:46.11 | BBHoss | im not registered with support though |
| 01:46.16 | Qwell | well, register |
| 01:46.20 | BBHoss | how |
| 01:46.22 | TJNII | fujin: I'm trying to transition servers from one box to another. I copied the configs verbatim, and the other server has had nothing changed. This all worked before. |
| 01:46.33 | Qwell | BBHoss: I'm sure it's step 1 in the documentation |
| 01:46.39 | BBHoss | heh |
| 01:46.41 | fujin | TJNII: Have the IP addresses changed? |
| 01:46.52 | BBHoss | well i obtained this in a sort of unorthodox way |
| 01:47.01 | fujin | an Illegal way? |
| 01:47.07 | Qwell | so, you pirated hardware? O.o |
| 01:47.11 | fujin | That's fail. |
| 01:47.25 | BBHoss | no, mark gave it to me |
| 01:47.43 | Qwell | ahh, then you don't have an aa50, you have an aadk |
| 01:47.52 | BBHoss | no its an aa50, i promise |
| 01:48.03 | TJNII | fujin: No. Currently port forwarded in the router. I switched over the forwards, but lemme check them again |
| 01:48.04 | Qwell | no it's not |
| 01:48.10 | BBHoss | hmm |
| 01:48.13 | Qwell | if it was an aa50, you would have support |
| 01:48.17 | BBHoss | s844i? |
| 01:48.22 | Qwell | and a license for EBE |
| 01:48.39 | TJNII | No, the 4569 forward is there and UDP. |
| 01:48.43 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 01:48.44 | BBHoss | Asterisk Business Edition autotag_for_sx00i-1.0.1 |
| 01:48.54 | Qwell | BBHoss: yeah, you'll need to get it from Mark |
| 01:49.05 | Qwell | otherwise, you're stuck with building from aadk sources |
| 01:49.21 | BBHoss | hmm, im not going to attempt that |
| 01:50.28 | BBHoss | ok |
| 01:50.41 | BBHoss | answer this for me, where do i adjust txgain and rxgain |
| 01:50.54 | BBHoss | for zaptel of course |
| 01:50.57 | Qwell | in the gui |
| 01:51.24 | BBHoss | i haven't seen an option... |
| 01:51.30 | Qwell | upgrade |
| 01:51.41 | BBHoss | i wish i could |
| 01:53.35 | fujin | lear2notaa50 |
| 01:54.00 | BBHoss | ? |
| 01:54.04 | nestAr | password? |
| 01:54.06 | nestAr | woops! |
| 01:54.44 | *** join/#asterisk btorrenga (n=btorreng@adsl-68-75-160-56.dsl.emhril.ameritech.net) |
| 01:54.48 | tzafrir | BBHoss, in zapata.conf ? |
| 01:55.28 | phix | oh nice |
| 01:55.29 | obnauticus | Qwell can you help me with something really fast. |
| 01:55.47 | obnauticus | It |
| 01:55.51 | obnauticus | it's probably not that hard :/ |
| 01:55.52 | phix | I found the issue with my SPA3102, the default PSTN-VoIP ring delay is 16 seconds, changing it to 0 works great :D |
| 01:55.59 | phix | should add that to the forums :P |
| 01:56.25 | obnauticus | well anyone then: |
| 01:56.32 | obnauticus | http://pastebin.ca/771589 <-- I can hear on the phone I'm calling to my pbx |
| 01:56.38 | obnauticus | but i cannot send to my pbx from my phone |
| 01:57.02 | BBHoss | one way audio? |
| 01:57.02 | phix | also, ending dialed numbers with a # makes the ATA dial the numbe straight away (oh yeah and putting in a correct dialplan for your country also helps :)). |
| 01:57.43 | obnauticus | ya |
| 01:57.49 | BBHoss | nat? |
| 01:57.56 | obnauticus | word. |
| 01:57.58 | phix | BBHoss: insecure=invite? |
| 01:58.49 | *** join/#asterisk axscode (n=axscode@132.240.208.218.klj02-home.tm.net.my) |
| 01:58.55 | btorrenga | anyone seen a situation with one-way audio (outbound works, inbound audio doesn't), and then get disconnected after a few seconds with " no reply to our critical packet"? |
| 01:59.10 | obnauticus | ... |
| 01:59.12 | obnauticus | I'm uhh |
| 01:59.13 | btorrenga | maybe 20 seconds then disconnected |
| 01:59.16 | phix | Now to fix my last issue, callerid from PSTN -> asterisk, shows up as the PSTN SIP uername instead of the number calling from the PSTN line |
| 01:59.18 | obnauticus | having that exact same problemn right now dude. |
| 01:59.20 | obnauticus | http://pastebin.ca/771589 <-- |
| 01:59.25 | fujin | btorrenga: lag, nat? |
| 01:59.29 | btorrenga | nat |
| 01:59.31 | obnauticus | I'm behind an NAT too |
| 01:59.31 | btorrenga | lag is ok |
| 01:59.35 | fujin | NAT = the problem |
| 01:59.37 | obnauticus | We have the same problem fujin |
| 01:59.38 | BBHoss | hmm pastebin is not working on my end |
| 01:59.42 | obnauticus | What should we do ? |
| 01:59.45 | phix | btorrenga: ummmm yeah :) port map |
| 01:59.54 | obnauticus | I have all the ports configured correctly for SIP and etc. |
| 01:59.56 | phix | port forward even |
| 02:00.01 | fujin | depends on the setup. Have you got phones -> asterisk -> NAT -> internet? |
| 02:00.06 | obnauticus | i have |
| 02:00.10 | phix | obnauticus: are you forwarding them to your asterisk box though from your routeR? |
| 02:00.11 | fujin | or phones -> NAT -> SIP provider |
| 02:00.17 | obnauticus | internet -> NAT/Router/Firewall -> Asterisk/other clients |
| 02:00.19 | btorrenga | ports are good |
| 02:00.28 | obnauticus | ya |
| 02:00.31 | obnauticus | my ports are fine too. |
| 02:00.34 | btorrenga | works ok with TelIAX, I get the error with IPKall |
| 02:00.41 | obnauticus | btorrenga |
| 02:00.42 | obnauticus | me too |
| 02:00.43 | BBHoss | you don't have a telephony enabled modem do you? |
| 02:00.45 | phix | btorrenga: they use a different set of ports? |
| 02:00.46 | obnauticus | i never got it about a month ago. |
| 02:00.47 | btorrenga | started after I moved from a static to dynamic IP |
| 02:00.59 | phix | heh |
| 02:01.03 | phix | you need to register |
| 02:01.03 | fujin | canreinvite=no, nat=yes? |
| 02:01.12 | phix | register => |
| 02:01.23 | btorrenga | fujin, yup |
| 02:01.28 | fujin | o_0 |
| 02:01.34 | BBHoss | also make sure you define your local network, externip, etc |
| 02:01.39 | BBHoss | ~nat |
| 02:01.40 | jbot | from memory, nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 02:01.40 | fujin | ^^. |
| 02:01.44 | btorrenga | Ive never seen the error befoe |
| 02:02.01 | btorrenga | I have localnet set, but externIP doesnt accept a hostname, only IP |
| 02:02.08 | BBHoss | externhost does |
| 02:02.16 | btorrenga | the CLI complains |
| 02:02.16 | fujin | specify your IP, then? |
| 02:02.22 | btorrenga | dynamic IP... |
| 02:02.25 | fujin | get a static? |
| 02:02.26 | BBHoss | you'll need dyndns |
| 02:02.32 | obnauticus | Ya, I use dyndns for mine. |
| 02:02.41 | btorrenga | yeah, I have it setup with afraid dns |
| 02:03.05 | btorrenga | I had a static for years, but am going back to school... poor now. |
| 02:03.47 | fujin | I have no issues here specify nat=yes on my upstream SIP provider, canreinvite=no,. |
| 02:04.01 | fujin | and it goes (LAN) phones -> asterisk -> NAT -> tubes -> SIP provider |
| 02:04.25 | fujin | I don't even need port forwards (I only use it for outgoing traffic) |
| 02:04.54 | btorrenga | here is the CLi when externip=hostname: |
| 02:05.04 | btorrenga | [Nov 12 20:05:22] WARNING[32528]: chan_sip.c:16757 reload_config: Invalid address for externip keyword: brent.hostname.whatever |
| 02:05.04 | fujin | upgrade |
| 02:05.13 | btorrenga | running 1.4.132 |
| 02:05.18 | btorrenga | er, 1.4.13 |
| 02:05.18 | fujin | o_0 |
| 02:05.20 | fujin | That's odd. |
| 02:05.40 | obnauticus | btorrenga, can you tell me when you fix it |
| 02:05.45 | Hadi- | is there any other billing solutions for asterisk (similar to A2Billing) that you guys can recommand? |
| 02:05.46 | btorrenga | haha |
| 02:05.47 | obnauticus | because we are having the same exact issue. |
| 02:05.50 | obnauticus | I'm not kidding. |
| 02:05.51 | btorrenga | you assume I will fix it? |
| 02:05.53 | *** join/#asterisk ManxPower (n=manxpowe@71-8-61-95.dhcp.leds.al.charter.com) |
| 02:05.57 | btorrenga | thanks for the confidence. |
| 02:05.58 | fujin | btorrenga: use 'externhost' instead of 'externip'. |
| 02:05.59 | obnauticus | well they are helping you :/ |
| 02:06.05 | btorrenga | hmm |
| 02:06.06 | btorrenga | ok |
| 02:06.08 | btorrenga | brb |
| 02:06.16 | TJNII | Bah. Everythings working but IAX |
| 02:06.27 | PaulAviles | is it loaded? |
| 02:06.39 | TJNII | With no errors |
| 02:06.46 | TJNII | And outgoing calls work |
| 02:06.55 | TJNII | But not incoming or registrations |
| 02:07.00 | obnauticus | I GOT IT |
| 02:07.02 | obnauticus | btorrenga |
| 02:07.04 | obnauticus | i figured it out |
| 02:07.04 | btorrenga | no luck |
| 02:07.06 | btorrenga | ??? |
| 02:07.09 | obnauticus | hold on |
| 02:07.15 | btorrenga | same error with externhost |
| 02:07.16 | btorrenga | btw |
| 02:07.24 | obnauticus | update your externip = |
| 02:07.33 | fujin | btorrenga: give me some PASTIES |
| 02:07.33 | obnauticus | under sip.conf |
| 02:07.36 | fujin | I need to see sip.conf |
| 02:07.41 | fujin | of phones, and provider |
| 02:07.42 | obnauticus | btorrenga |
| 02:07.44 | btorrenga | ayeaye |
| 02:07.47 | obnauticus | if your ip changed you need to change that too |
| 02:07.49 | obnauticus | that's what i just did |
| 02:07.51 | obnauticus | and it worked |
| 02:07.52 | obnauticus | 100% |
| 02:07.59 | fujin | obnauticus: externhost will solve that problem, if you use a dynamic dns |
| 02:08.06 | PaulAviles | TJ does it work for sip incoming? |
| 02:08.09 | fujin | (not suggested for production environments, get a static, etc) |
| 02:08.12 | obnauticus | can you specify a dnymaic one? |
| 02:08.17 | obnauticus | well mine rarely changes. |
| 02:08.26 | btorrenga | heres the general section |
| 02:08.27 | fujin | asterisk will do dns lookups occasionally |
| 02:08.27 | btorrenga | [general] |
| 02:08.27 | btorrenga | tos_sip=cs3 |
| 02:08.27 | btorrenga | tos_audio=ef |
| 02:08.28 | btorrenga | tos_video=af41 |
| 02:08.28 | btorrenga | srvlookup=yes |
| 02:08.28 | btorrenga | ;externip=brent.torrenga.com |
| 02:08.30 | fujin | ~pb |
| 02:08.30 | jbot | A Pastebin is a place to paste your stuff 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 |
| 02:08.30 | btorrenga | externhost=brent.torrenga.com |
| 02:08.32 | btorrenga | ;extenip=68.75.160.56 |
| 02:08.32 | fujin | Don't paste to the channel |
| 02:08.34 | btorrenga | localnet=10.0.0.0/255.0.0.0 |
| 02:08.34 | obnauticus | holy shit! |
| 02:08.36 | btorrenga | context=inbound-guest-sip |
| 02:08.38 | btorrenga | disallow=all |
| 02:08.40 | btorrenga | allow=g729 |
| 02:08.42 | btorrenga | allow=g726 |
| 02:08.44 | btorrenga | allow=g723 |
| 02:08.46 | btorrenga | allow=gsm |
| 02:08.48 | btorrenga | allow=ulaw |
| 02:08.48 | obnauticus | btorrenga change your externip= |
| 02:08.50 | btorrenga | allow=alaw |
| 02:08.52 | btorrenga | allow=ilbc |
| 02:08.52 | obnauticus | Epic fail! |
| 02:08.54 | btorrenga | fromdomain=brent.torrenga.com |
| 02:08.56 | btorrenga | nat=yes |
| 02:08.58 | btorrenga | canreinvite=no |
| 02:09.00 | btorrenga | ignoreregexpire=no |
| 02:09.02 | btorrenga | oohh... |
| 02:09.04 | btorrenga | sorry about that. |
| 02:09.06 | btorrenga | whats the proper way to paste? |
| 02:09.12 | PaulAviles | pastebin.ca |
| 02:09.12 | fujin | ~pb |
| 02:09.13 | jbot | A Pastebin is a place to paste your stuff 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 |
| 02:09.14 | fujin | like I said :) |
| 02:09.16 | btorrenga | I havent used irc in about 10 years or so until tonight... |
| 02:09.18 | fujin | I prefer rafb.net/paste/ |
| 02:09.43 | *** join/#asterisk bkw_ (n=brian@adsl-64-149-41-151.dsl.tul2ok.sbcglobal.net) |
| 02:10.17 | btorrenga | ok, I "converted" it, if I paste that text, then it will not blow anything up? |
| 02:10.28 | *** part/#asterisk JayTee52 (n=jforde05@c-69-137-243-25.hsd1.in.comcast.net) |
| 02:10.35 | JT | btorrenga: ? |
| 02:11.05 | fujin | btorrenga: if you pasted it to the pastebin, paste us the link to it |
| 02:11.06 | btorrenga | what does rafb.net/paste do exactly? |
| 02:11.08 | fujin | "copy link" or whatever. |
| 02:11.10 | TJNII | When I forward the port to the old 1.2 server (which shows the iax port as being open, not filtered) it works. |
| 02:11.11 | btorrenga | ahhh |
| 02:11.13 | btorrenga | I see |
| 02:11.18 | fujin | btorrenga: it lets you paste large amounts of text to it, and then provides you with a linkback to it |
| 02:11.22 | fujin | to save our poor eyes |
| 02:11.22 | btorrenga | http://rafb.net/p/4dogRJ81.html |
| 02:11.24 | btorrenga | haha |
| 02:11.31 | btorrenga | ok, I get it now. that is my general section |
| 02:11.34 | *** join/#asterisk groogs[h] (n=gregm@cbl-66-102-80-229.wtccommunications.ca) |
| 02:11.36 | btorrenga | I'll get the ipkal stanza. |
| 02:11.42 | fujin | cool |
| 02:11.46 | fujin | paste em both in one pasty, makes it a bit easier |
| 02:12.49 | btorrenga | gotcha |
| 02:12.52 | btorrenga | heres both |
| 02:12.53 | btorrenga | http://rafb.net/p/YJF5Jh99.html |
| 02:13.29 | fujin | ok, now we're getting somewhere. |
| 02:14.23 | Hadi- | Hello... anyone here using A2Billing? |
| 02:14.59 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 02:15.04 | btorrenga | obnauticus: you fixed your with externip? |
| 02:15.42 | PaulAviles | sorry hadi no.. |
| 02:15.47 | Hadi- | :( |
| 02:16.46 | obnauticus | ya |
| 02:16.52 | fujin | btorrenga: if there's anything I can suggest, it's strip your conf down to the bare minimal |
| 02:16.53 | obnauticus | i just updated externip= in my sip.conf |
| 02:16.56 | obnauticus | to my current one. |
| 02:16.58 | obnauticus | and it worked. |
| 02:17.02 | btorrenga | grr |
| 02:17.15 | obnauticus | i changed nat=yes and the content handling hting to no |
| 02:17.20 | obnauticus | handling* |
| 02:17.32 | btorrenga | I cant go changing sip.conf/sip reload every time my IP changes |
| 02:17.40 | obnauticus | i know neither me :/ |
| 02:17.42 | obnauticus | but it works! |
| 02:17.47 | obnauticus | you can have a uhh |
| 02:17.51 | phix | What is the point of using voicemailbox= under a sip user in sip.conf? isn't that what voicemail.conf does? |
| 02:17.51 | obnauticus | crontab do it |
| 02:17.53 | fujin | btorrenga: take a look at http://rafb.net/p/YcgDsj17.html |
| 02:17.53 | obnauticus | and reload it every time |
| 02:17.54 | obnauticus | lol |
| 02:18.36 | btorrenga | ok, fujin, what am I seeing? |
| 02:18.50 | fujin | That's a working phones -> asterisk -> NAT <- SIP |
| 02:19.00 | btorrenga | hmm, ok |
| 02:19.20 | fujin | see, I don't even specify localnet |
| 02:19.23 | fujin | externhost or anything. |
| 02:19.26 | btorrenga | I appreciate that. I'll take it as a starting point, and build up |
| 02:19.38 | fujin | Specifying externhost would be necessary for you, as I guess you're taking incoming calls over it |
| 02:19.45 | fujin | where I use wxc for strictly one-way outbound calls. |
| 02:19.53 | fujin | but that's the *only* change I believe you'll need. |
| 02:20.16 | btorrenga | uhuh. Its funny that this worked for two or three years with a static IP. |
| 02:20.26 | fujin | Static IP makes lots of stuff happy. |
| 02:20.31 | btorrenga | I wish externhost= would accept a host instead of IP.... |
| 02:20.41 | fujin | externhost does, externip does not. |
| 02:20.45 | fujin | they are different configuration values |
| 02:20.55 | btorrenga | I'll show you... |
| 02:20.58 | fujin | externhost causes asterisk to poll the hostname specifiied, and pretty much sets externip=$HOST |
| 02:21.08 | fujin | You pasted before, externip=blabla.com |
| 02:21.17 | btorrenga | [Nov 12 20:07:58] WARNING[32528]: chan_sip.c:16764 reload_config: Invalid address for externhost keyword: brent.torrenga.com |
| 02:21.28 | fujin | does brent.torrenga.com resolve? |
| 02:21.33 | btorrenga | it should |
| 02:21.36 | btorrenga | can you try it? |
| 02:21.38 | fujin | host brent.torrenga.com |
| 02:21.51 | fujin | yes, it resolves here. |
| 02:21.55 | fujin | does it resolve on your asterisk box |
| 02:21.59 | btorrenga | AHHHH! |
| 02:22.02 | fujin | bingo |
| 02:22.03 | btorrenga | thats a good question |
| 02:22.05 | btorrenga | NO it wont |
| 02:22.10 | fujin | vi /etc/resolv.conf |
| 02:22.11 | btorrenga | (dont ask why, its complicated) |
| 02:22.16 | fujin | It needs to. |
| 02:22.18 | TJNII | The IAX pokes arn't getting through.... |
| 02:22.48 | btorrenga | huh! thanks for your help. To fix this will take a bit of work, actually. |
| 02:23.02 | fujin | I don't see why? point it at a recursive nameserver, done? |
| 02:23.06 | btorrenga | it has to do with VPN's and other crap... |
| 02:23.24 | btorrenga | and local DNS servers that fake certain IP's as being on the localnet. |
| 02:23.32 | fujin | oh god, split brain for the lose. |
| 02:23.52 | fujin | btorrenga: if you specify the nameserver in you rresolv.conf, it'll query that directly |
| 02:23.55 | fujin | and ignore your localnet stuff. |
| 02:23.57 | fujin | =winwin situation |
| 02:24.01 | fujin | no more split brain, correct dns! |
| 02:24.04 | btorrenga | yeah, good idea. |
| 02:24.14 | btorrenga | Im trying to think if it will break anything else... |
| 02:24.14 | TJNII | IAX is also UDP, correct? |
| 02:24.29 | btorrenga | I dont think it will |
| 02:24.38 | fujin | TJNII: can be both, 4569/tcp,udp |
| 02:25.05 | JT | no |
| 02:25.09 | JT | it does not use tcp |
| 02:25.12 | JT | that would be illogical |
| 02:25.31 | fujin | can not, or does not? |
| 02:25.34 | J4k3 | theres always fun stuff like icmp |
| 02:25.35 | J4k3 | ;) |
| 02:25.36 | JT | can not |
| 02:25.40 | fujin | kk |
| 02:25.44 | JT | it uses UDP 4569 |
| 02:25.46 | fujin | someone should tell whoever maintains /etc/services this |
| 02:25.54 | JT | iax is combined signalling and media |
| 02:26.01 | fujin | IANA believes that IAX can run on TCP |
| 02:26.02 | fujin | o_0 |
| 02:26.05 | JT | media cannot travel over tcp with reasonable performance |
| 02:26.25 | JT | heh, damn iana |
| 02:26.38 | fujin | the RFC for IAX doesn't mention TCP, at all? |
| 02:27.02 | fujin | actually, I lie, debian may have added IAX to /etc/services |
| 02:27.05 | fujin | just read the rest of the header |
| 02:28.35 | BBHoss | is there any way i can see what dtmf asterisk is seeing on zaptel? |
| 02:29.50 | *** join/#asterisk mordaunt (n=mordaunt@unaffiliated/mordaunt) |
| 02:30.04 | TJNII | The one machine is sending pokes, the other is not recieving. The port is forwarded. If I forward the port to the old asterisk install (which has the same configs) it works. |
| 02:30.07 | TJNII | I'm stumped |
| 02:30.23 | fujin | TJNII: nmap |
| 02:30.47 | BBHoss | what kind of device are you using? |
| 02:31.21 | TJNII | fujin: The IAX port shows open|filtered on all machines I've run it against |
| 02:31.38 | TJNII | BBHoss: Two asterisk 1.4 servers on Debian |
| 02:32.06 | BBHoss | router-wise |
| 02:32.16 | BBHoss | what is doing the routing? |
| 02:32.42 | TJNII | an Actiontek router |
| 02:32.44 | BBHoss | you have two separate boxes each running asterisk 1.4, behind a firewall/router. correct? |
| 02:32.52 | fujin | TJNII: is the new asterisk binding IAX to 0.0.0.0? |
| 02:32.58 | fujin | i.e.; not just a specific IP address, or 127.0.0.1? |
| 02:33.05 | fujin | netstat -l|grep -v unix|grep iax |
| 02:33.10 | BBHoss | 0.0.0.0 is all |
| 02:33.18 | TJNII | fujin: Binding IAX2 to default address 0.0.0.0:4569. |
| 02:33.34 | fujin | I'm aware of that, just wanted to check it was binding to all addresses. |
| 02:33.52 | TJNII | Yea, that was one of the first things I checked. :) |
| 02:34.54 | TJNII | fujin: netstat shows "udp 0 0 *:iax *:*" |
| 02:35.33 | TJNII | I'm trying to get the actiontek nat out of the picture. When I get this server running it will sit on the public internet instead of port forwarding. |
| 02:35.40 | fujin | public internet, for the win. |
| 02:35.43 | fujin | iptables + public internet = fine. |
| 02:35.50 | J4k3 | I'd bet a large quantity of pocket lint that your router sucks. use one of the asterisk boxes as your router. |
| 02:35.55 | TJNII | But it doesn't seem to want to work |
| 02:36.03 | BBHoss | actiontek sounds like one of those cheapo modems that qwest and others give out |
| 02:36.12 | J4k3 | BBHoss: *ding ding* |
| 02:36.16 | BBHoss | j4k3: i second that |
| 02:36.19 | J4k3 | the best garbage china can deliver |
| 02:36.24 | J4k3 | especially when it comes to the software, generally. |
| 02:36.34 | TJNII | J4k3: That is a bet I wouldn't take. It is crap. When I get this up it will just be a bridge. |
| 02:38.12 | *** join/#asterisk Mackes (n=root@cpe-76-180-236-119.buffalo.res.rr.com) |
| 02:38.38 | *** part/#asterisk Mackes (n=root@cpe-76-180-236-119.buffalo.res.rr.com) |
| 02:38.51 | TJNII | I don't know what else to check here/ |
| 02:42.48 | TJNII | Let me try pointing the old server at the new server..... |
| 02:43.41 | *** join/#asterisk icewater1an (n=immagine@i53874644.versanet.de) |
| 02:44.03 | fujin | I'm not too familiar with IAX, otherwise I'd try help you out a little more |
| 02:44.12 | fujin | s/not // |
| 02:44.15 | fujin | gah |
| 02:44.20 | fujin | yeah, |
| 02:44.31 | btorrenga | haha |
| 02:44.45 | btorrenga | it has a long ways to go, doesnt it? |
| 02:45.04 | btorrenga | IAX I mean |
| 02:45.38 | fujin | SIP has had more interest from the genral publiz |
| 02:46.08 | obnauticus | They really should come up with a better standard |
| 02:46.21 | obnauticus | What's the highest quality codec? |
| 02:46.23 | *** join/#asterisk techie (n=techie@adsl-76-214-20-56.dsl.lsan03.sbcglobal.net) |
| 02:46.23 | TJNII | Well, the kicker is I haven't changed the configs! |
| 02:46.34 | JT | of the 8kHz ones, G.711 |
| 02:46.46 | JT | btorrenga: yes, SIP is > IAX2 |
| 02:46.47 | TJNII | I just put another * install on a second machine and cp -rvp |
| 02:46.48 | brookshire | g722 is nive |
| 02:46.54 | btorrenga | I've tried setting up IAX twice between a couple offices - each time it worked great in testing. It would blow up in production. |
| 02:47.00 | brookshire | nice also |
| 02:47.18 | JT | G.722 is wideband |
| 02:47.21 | JT | 16kHz |
| 02:48.00 | obnauticus | Is g729 good? |
| 02:48.14 | TJNII | Well, the 2 asterisk boxes behind NAT can talk to each other, so I wonder if it isn't something screwy with the router |
| 02:48.24 | BBHoss | im sure it is |
| 02:48.33 | TJNII | Odd, though |
| 02:48.43 | TJNII | That it would work for one machine and not the other |
| 02:48.56 | JT | obnauticus: not really |
| 02:49.00 | JT | obnauticus: it's low bandwidth |
| 02:49.09 | obnauticus | in compairison to ulaw :/ |
| 02:49.15 | *** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com) |
| 02:49.37 | obnauticus | It's better than ulaw though right |
| 02:49.49 | *** join/#asterisk asteriskguy (n=learnast@cpe-66-75-92-47.socal.res.rr.com) |
| 02:49.51 | JT | no way |
| 02:49.53 | JT | are you on drugs? |
| 02:49.56 | obnauticus | lol |
| 02:49.58 | asteriskguy | drugs? |
| 02:50.03 | obnauticus | im messin with you |
| 02:50.03 | asteriskguy | some would be nice |
| 02:50.07 | obnauticus | ... |
| 02:50.08 | asteriskguy | how's it going JT? |
| 02:50.15 | obnauticus | Anyway |
| 02:50.22 | obnauticus | how is Cisco CM>? |
| 02:50.22 | PaulAviles | 729 is supposed to be very good... |
| 02:50.33 | PaulAviles | a real pain the the ... |
| 02:50.45 | JT | it's not that good |
| 02:50.54 | asteriskguy | asterisk & virtual server |
| 02:51.06 | asteriskguy | how's that for an idea? VMWARE ESX |
| 02:51.31 | fujin | It's a fine idea, works great here. |
| 02:51.32 | BBHoss | TJNII: do a hard reset of the modem/router, then add the rules again for the NEW server, the actiontek may be having brainfarts |
| 02:51.37 | PaulAviles | can have problems with zaptel for time sync |
| 02:51.48 | asteriskguy | fujin, how are you running it |
| 02:51.56 | asteriskguy | what if you don't use zaptel |
| 02:51.58 | fujin | With an init script. |
| 02:52.03 | fujin | I don't use zaptel, it's rubbish. |
| 02:52.05 | asteriskguy | sip provider |
| 02:52.10 | PaulAviles | what about confenrecing like meetme? |
| 02:52.16 | fujin | ztdummy |
| 02:52.18 | asteriskguy | fujin, care to elaborate on the init script? |
| 02:52.25 | fujin | lol~ |
| 02:52.25 | PaulAviles | should be ok then |
| 02:52.27 | fujin | It comes with Ubuntu. |
| 02:52.40 | fujin | I grabbed the inits cript from it, nuked the rest and built from source. |
| 02:52.59 | *** join/#asterisk grimsy (n=chatzill@203-206-220-214.perm.iinet.net.au) |
| 02:53.02 | asteriskguy | yes, I have a test box running * 1.4.13 along with ztdummy on vmware esx |
| 02:53.13 | asteriskguy | what about multiple * running on esx? |
| 02:53.20 | fujin | yes, what's the problem with that? |
| 02:53.27 | fujin | I have two running, HA-style. |
| 02:53.29 | fujin | hot/cold |
| 02:53.30 | asteriskguy | maybe 20-30 * running on esx |
| 02:53.37 | fujin | 20-30 would be a dumb idea |
| 02:53.47 | asteriskguy | reason? |
| 02:53.50 | fujin | You'd never need that many asterisk boxes. |
| 02:53.53 | TJNII | BBHoss: I'm not going to bother. My 2 asterisk boxes can talk and everything else works, so I'm going to say "good enough for now" and move on. When I get my block of IPs and my network reconfigured I'll do more testing and a more gracefull transition. |
| 02:54.00 | asteriskguy | yes we do |
| 02:54.13 | asteriskguy | 1/per branch office |
| 02:54.18 | asteriskguy | 200+ offices |
| 02:54.25 | asteriskguy | 16 phones / office |
| 02:54.31 | fujin | hrm |
| 02:54.36 | fujin | yes, I guess in that situation, you could |
| 02:54.43 | btorrenga | asteriskguy: SIP or IAX connecting those offices? |
| 02:54.48 | asteriskguy | iax for now |
| 02:54.55 | asteriskguy | but we'll need to put in a SIP proxy |
| 02:54.55 | fujin | although It might be smarter to have a centralised farm of asterisk servers, and run OpenSER or something in your offices |
| 02:54.58 | fujin | let it handle the routing. |
| 02:55.02 | PaulAviles | probably will be better and cheaper to buy separate desktop computers than the total cost of esx and the server |
| 02:55.05 | fujin | and asterisk handle the backend, inter-office routing. |
| 02:55.15 | btorrenga | jitter is ok with IAX in your setup??? |
| 02:55.17 | fujin | better and cheaper? cheaper maybe, better no |
| 02:55.42 | PaulAviles | no ha? what is your super hyper dupper esx server crashes? |
| 02:55.54 | asteriskguy | IAX is fine, we have 2 big offices running on IAX |
| 02:55.58 | *** join/#asterisk red9012 (n=marc3234@76-10-149-62.dsl.teksavvy.com) |
| 02:55.58 | PaulAviles | with separate boxes you can have some failover |
| 02:56.01 | asteriskguy | over a DS3 though |
| 02:56.11 | asteriskguy | ESX has HA |
| 02:56.15 | btorrenga | ah. |
| 02:56.19 | asteriskguy | as an option, probably cost more though |
| 02:56.30 | fujin | I don't do my HA on ESX. |
| 02:56.37 | fujin | I'm making use of the heartbeat packages |
| 02:56.40 | asteriskguy | but our budget can handle that if need to |
| 02:56.42 | *** join/#asterisk PepOSX (n=pepOSX@190.72.153.45) |
| 02:56.49 | fujin | as it can do in-depth checks, and sharing a virtual IP |
| 02:56.50 | PaulAviles | I think so. pretty familiar with vmwareb, but I still think is too risky |
| 02:56.54 | red9012 | how does asterisk handle quad core processor? are all cores used? |
| 02:57.09 | PaulAviles | can the kernel use them? |
| 02:57.15 | asteriskguy | i know it handles dual core fine |
| 02:57.33 | red9012 | I mean is the load distributed across the cores? |
| 02:57.36 | fujin | red9012: Linux has no issue sharing processes among them. |
| 02:57.38 | asteriskguy | fujin we're interested in setting something like that up |
| 02:57.40 | fujin | on a SMP kernel, yes |
| 02:57.41 | PaulAviles | it shoul... |
| 02:57.46 | PaulAviles | d |
| 02:57.52 | asteriskguy | do you have information you can share? |
| 02:58.06 | fujin | uhm |
| 02:58.21 | red9012 | as usual, not a clear answer is given -- and we will now proceed to harsh comments that I am about to get. |
| 02:58.45 | fujin | You'll have to be a little more specific. I built the callcentre here, 50~ phones, two asterisk servers, HA, everythings happy |
| 02:58.49 | fujin | what do you want to know |
| 02:59.00 | fujin | red9012: yes. It does. all cores are used. |
| 02:59.18 | asteriskguy | ok, 200 locations, 16 phones/loc, on esx |
| 02:59.21 | fujin | apt-get install htop && htop, do some testing. You'll see asterisk spawning children on each core/proc |
| 02:59.23 | asteriskguy | plus a sip proxy |
| 02:59.45 | fujin | ESX at each location? that's a little silly, imo |
| 02:59.46 | asteriskguy | HA is a must |
| 02:59.54 | asteriskguy | no, not esx at each location |
| 03:00.01 | fujin | yes, well, I'm not going to design your system for you, not for free :) |
| 03:00.08 | asteriskguy | esx is running at a centralized location |
| 03:00.21 | asteriskguy | then each instance of * will handle each location |
| 03:00.43 | asteriskguy | would having 20-30 instance of * have any issue with ztdummy? |
| 03:00.56 | fujin | uh |
| 03:01.08 | fujin | 20-30 instances on one server? |
| 03:01.11 | fujin | or 20-30 virtual servers |
| 03:01.29 | BBHoss | ztdummy doesn't work right with xen sometimes |
| 03:01.50 | red9012 | fukin -- only one asterisk is running - wondering if all cores will be used. |
| 03:01.50 | fujin | yes, I've heard issues with ztdummy on soem virtualisation software. |
| 03:01.54 | asteriskguy | 20-30 virtual server |
| 03:02.02 | fujin | red9012: as I said before, yes |
| 03:02.06 | asteriskguy | running on 1 physical machine |
| 03:02.08 | fujin | it'll spawn children as load increases |
| 03:02.14 | *** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com) |
| 03:02.20 | fujin | which will head to each processor/core as necessary, on a SMP kernel |
| 03:02.27 | BBHoss | red9012: asterisk doesn't scale well with multiple processors, too many mutex locks |
| 03:02.34 | asteriskguy | the server has (4) dual core CPUs with 32GB of RAM |
| 03:02.48 | red9012 | fukin -- asterisk is monothread. I woul be surprised to see it use more than core |
| 03:03.13 | asteriskguy | red9012, we have * running on (2) dual core CPUs |
| 03:03.15 | asteriskguy | works fine |
| 03:03.16 | fujin | I've seen it making use of both cores, here. |
| 03:03.33 | asteriskguy | on an SMP kernel ofcourse |
| 03:03.39 | red9012 | works fine is not the issue. |
| 03:03.53 | red9012 | I want to see all cores in use. |
| 03:03.54 | asteriskguy | fujin, how's that spec for 20-30 virtual server running asterisk? |
| 03:04.12 | BBHoss | i know with asterisk, two separate boxes are nearly always better than 1 box with 2 cpus, twice the ram, etc |
| 03:04.15 | fujin | should be fine, as I said though I think you're doing it wrong |
| 03:04.18 | red9012 | fukin -- running 20-30 asterisk server on one host is not recommended. |
| 03:04.46 | fujin | I'd do it differently |
| 03:04.55 | asteriskguy | ok, how should i do this then? did I miss something? |
| 03:04.58 | obnauticus | LOL |
| 03:05.02 | obnauticus | 20-30 asterisk servers? |
| 03:05.04 | obnauticus | daemons* |
| 03:05.04 | obnauticus | wow. |
| 03:05.25 | red9012 | you must be out of your mind to run 20-30 asterisk servers on one host. |
| 03:05.27 | BBHoss | i think he's running them in virtual instances |
| 03:05.35 | obnauticus | we know. |
| 03:05.35 | obnauticus | lol |
| 03:05.37 | red9012 | and if you do queues, then it will crash./ |
| 03:05.37 | BBHoss | well with 8 cores and 32gb of ram |
| 03:05.42 | obnauticus | Oh |
| 03:05.48 | obnauticus | :/ |
| 03:05.49 | obnauticus | i dunno then |
| 03:05.49 | obnauticus | it |
| 03:05.52 | obnauticus | it's worth a try |
| 03:05.52 | asteriskguy | well, no queues, no voicemail |
| 03:05.54 | obnauticus | i say go for it. |
| 03:05.58 | asteriskguy | just direct calls and parking |
| 03:05.59 | BBHoss | still i recommend a rack of asterisk servers |
| 03:06.17 | BBHoss | but im sure you already have the hard ware :( |
| 03:06.18 | red9012 | anyone here knows how many concurrent queues I can run on a single 3ghz server? |
| 03:06.24 | asteriskguy | yeah |
| 03:06.38 | asteriskguy | it was bought for a MS project that of yanked |
| 03:06.46 | BBHoss | heh |
| 03:06.49 | asteriskguy | so we're using it for * virtualization |
| 03:07.03 | asteriskguy | fujin, how would you do it then? |
| 03:07.16 | fujin | I'd have a central farm of asterisk servers to handle the backend |
| 03:07.19 | BBHoss | just dont expect to be able to do anything that requires ztdummy |
| 03:07.20 | fujin | and put openser proxies at each office |
| 03:07.24 | *** join/#asterisk [hC] (n=hardcore@70.68.142.245) |
| 03:07.27 | red9012 | fukin -- what soft do you use for virtualization? |
| 03:07.28 | BBHoss | fujin: i agree |
| 03:07.33 | fujin | ESX |
| 03:08.34 | fujin | red9012: depends on codecs, network, a number of things |
| 03:08.37 | fujin | not just the speed of the processor |
| 03:08.42 | asteriskguy | hmm....kinda backward from what we envisioned. we envisioned SIP provider -> OpenSER -> *s |
| 03:08.53 | fujin | you're doing it wrong |
| 03:09.07 | fujin | you can connect your sip provider to asterisk, define all your openser offices |
| 03:09.12 | red9012 | openser is needed as far as I can tell, only when you need loadbalancing |
| 03:09.18 | fujin | then openser will bridge (reinvite) with your sip provider |
| 03:09.24 | obnauticus | Can someone tell me what is wrong with this |
| 03:09.25 | obnauticus | exten => #,2,MeetMe(1,i,2) |
| 03:09.33 | obnauticus | It's saying it's not a valid conference number |
| 03:09.41 | obnauticus | while in meetme.conf it's configured |
| 03:09.43 | fujin | show application meetme |
| 03:10.07 | [hC] | obnauticus: you must not have ztdummy loaded as a kernel module |
| 03:10.15 | obnauticus | No active conferences. |
| 03:10.16 | [hC] | obnauticus: or, a full fledged zaptel timing source. |
| 03:10.17 | obnauticus | oh |
| 03:10.26 | asteriskguy | doesn't MOH requires ztdummy? |
| 03:10.26 | obnauticus | it wokred before:" |
| 03:10.32 | obnauticus | What should i do then |
| 03:10.36 | obnauticus | kldload ztdummy ? |
| 03:10.36 | red9012 | moh does not require ztdummy |
| 03:10.37 | [hC] | modprobe ztdummy |
| 03:10.39 | fujin | I don't think MOH requires ztdummy. |
| 03:10.54 | red9012 | moh does have bugs though. |
| 03:11.03 | obnauticus | k it woked |
| 03:11.05 | J4k3 | hmm, lack of ztdummy will screw up conf? my confs go to shit at call #3 |
| 03:11.07 | fujin | native MOH works fine. |
| 03:11.15 | J4k3 | but I've also got a toy of an asterisk box (P3-700) |
| 03:11.36 | obnauticus | Is there any reason why my PBX isn't accepting DTMF tones from ANYTHING? |
| 03:11.41 | obnauticus | I have no idea why it's not. |
| 03:11.58 | fujin | You're sending them incorrectly? |
| 03:12.07 | obnauticus | by...pressing the button :/ |
| 03:12.17 | asteriskguy | MP3 MOH plays garbled music sometimes, but after we switch to native everything was fine |
| 03:12.25 | fujin | non-native MOH is dumb. |
| 03:12.31 | *** join/#asterisk TJNII_ (n=TJNII@209.234.89.226) |
| 03:12.33 | [hC] | make sure you set dtmfmode on both sides to rfc2833 |
| 03:12.36 | fujin | transcode all your MOH music to <native_codec_here> |
| 03:12.46 | fujin | obnauticus: what format DTMF is the PBX expecting? |
| 03:12.46 | asteriskguy | fujin, do you do consulting? |
| 03:12.53 | fujin | asteriskguy: I've not done it before, no. |
| 03:13.06 | TJNII_ | GAAH! Now IAX is working and SIP is failing! |
| 03:13.07 | obnauticus | [hC] it is. |
| 03:13.31 | fujin | are both devices able to send and receive rfc2833 DTMF? |
| 03:13.37 | fujin | I always had issues with it, and went with INBAND EVERYWHERE |
| 03:13.40 | fujin | Made lots of stuff easy. |
| 03:13.43 | btorrenga | wow |
| 03:13.50 | btorrenga | you only use g711? |
| 03:13.52 | *** join/#asterisk chode_ (n=chode@pD9E896CD.dip0.t-ipconnect.de) |
| 03:13.52 | obnauticus | fujin, not sure, how do I tell? |
| 03:13.59 | fujin | btorrenga: correct. |
| 03:14.07 | J4k3 | inband + lossy codec = sketchy dtmf |
| 03:14.17 | fujin | obnauticus: well, what are the two devices, asterisk / sip phone? |
| 03:14.25 | obnauticus | sip device.. |
| 03:14.26 | J4k3 | g729 usually gets dtmf ok, but others are sketchy |
| 03:14.28 | obnauticus | to meetme conference :/ |
| 03:14.37 | BBHoss | inband + codec!=*law = no dtmf :) |
| 03:14.43 | [hC] | you're trying to send dtmf into a meetme? |
| 03:14.54 | btorrenga | BBH: nice |
| 03:15.31 | obnauticus | k it works |
| 03:15.32 | obnauticus | hold on |
| 03:15.32 | BBHoss | actually it would be this: inband + (codec!=*law) = no dtmf |
| 03:15.34 | fujin | obnauticus: check dtmfmode in sip.conf, in the device configuration |
| 03:15.41 | obnauticus | i think i got it to work |
| 03:16.07 | obnauticus | nevermind |
| 03:16.13 | obnauticus | It's working on my outbound trunk... |
| 03:16.37 | obnauticus | k the one it's working on has dytmfmode as rfc2833 |
| 03:16.40 | obnauticus | and so does the one it's not working on. |
| 03:17.20 | *** join/#asterisk craigk (n=ckowald@58.174.122.198) |
| 03:18.12 | BBHoss | what is the device/trunk its not working on? |
| 03:18.21 | obnauticus | uhh |
| 03:18.22 | obnauticus | it |
| 03:18.26 | obnauticus | it's a sip device :/ |
| 03:18.27 | obnauticus | hold on |
| 03:18.29 | obnauticus | i'll pastebin it |
| 03:19.09 | obnauticus | http://pastebin.ca/771667 |
| 03:19.48 | BBHoss | use this: http://rafb.net/paste/ |
| 03:20.02 | BBHoss | for some odd reason i can't get to pastebin.ca |
| 03:20.08 | obnauticus | http://rafb.net/p/rSa3R458.html |
| 03:20.57 | *** join/#asterisk angom (n=Angel@201.170.35.218) |
| 03:24.20 | obnauticus | BBHoss, when i call into my PBX from my cell phone, it accepts the DTMF tones i send from my cell phone. But when I call to the conference directly from a softphone of mine using the same dtmfmode, it's not seeing it. |
| 03:24.53 | BBHoss | thats odd, what softphone is it |
| 03:25.04 | obnauticus | it was called idefisk |
| 03:25.06 | obnauticus | but they changed names |
| 03:25.09 | BBHoss | zoiper |
| 03:25.10 | obnauticus | ZoIPer |
| 03:25.11 | obnauticus | ya |
| 03:25.50 | BBHoss | is is an iax trunk or sip trunk |
| 03:25.55 | obnauticus | sip |
| 03:26.59 | BBHoss | it doesnt show support for RFC2833 |
| 03:27.03 | BBHoss | odd |
| 03:27.08 | obnauticus | ? |
| 03:27.09 | obnauticus | Huh? |
| 03:27.11 | obnauticus | the uhh |
| 03:27.19 | obnauticus | ZoIPer doesn't support RFC2833? |
| 03:27.35 | BBHoss | it doesn't show support on the website faq |
| 03:27.42 | BBHoss | SIP: |
| 03:27.42 | BBHoss | RFC 3261, RFC 2045, RFC 2046, RFC 2181, RFC 2617, RFC 2782, RFC 2915, RFC 3263, RFC 3265, RFC 3515, RFC 4028, RFC 4566 |
| 03:27.49 | obnauticus | What should I use? |
| 03:28.01 | BBHoss | try inband for the softphone |
| 03:28.08 | obnauticus | dtmfmode=inband ? |
| 03:28.16 | fujin | use inband, or notify |
| 03:28.18 | BBHoss | yes, for your softphone only |
| 03:28.34 | obnauticus | k |
| 03:28.47 | obnauticus | no workie. |
| 03:29.31 | BBHoss | try calling your cell with your softphone, then hit numbers and see if you hear sounds |
| 03:29.40 | obnauticus | alreqady did |
| 03:29.41 | obnauticus | it didn't. |
| 03:29.44 | BBHoss | hmm |
| 03:29.54 | BBHoss | try a different softphone then i guess |
| 03:30.22 | fujin | obnauticus: try notify |
| 03:30.30 | fujin | don't forget, you must configure the DEVICE to use that dtmf signaling |
| 03:30.34 | fujin | AND the sip.conf section |
| 03:30.37 | fujin | they *must* match |
| 03:30.44 | obnauticus | [2007-11-12 11:10:09] WARNING[2364]: chan_sip.c:16386 handle_common_options: Unknown dtmf mode 'notify' on line 686, using rfc2833 |
| 03:30.49 | BBHoss | yeah there may be a setting in zoiper |
| 03:30.53 | obnauticus | k |
| 03:30.56 | obnauticus | i'll look |
| 03:31.04 | fujin | notify? |
| 03:31.05 | fujin | hrm |
| 03:31.09 | BBHoss | i think fujjin means dtmfmode=info> |
| 03:31.09 | fujin | is it 'sip'? i forget |
| 03:31.11 | fujin | info. |
| 03:31.12 | fujin | that's it. |
| 03:31.24 | obnauticus | k |
| 03:31.28 | BBHoss | leave off that > its supposed to be a ? :) |
| 03:31.34 | obnauticus | ZoIPer has no options for DTMF |
| 03:31.34 | obnauticus | lol |
| 03:31.53 | BBHoss | i don't think it supports info either |
| 03:32.01 | BBHoss | that is rfc2976 |
| 03:32.08 | obnauticus | nope |
| 03:32.09 | BBHoss | which is not shwon |
| 03:32.10 | obnauticus | info didn't work |
| 03:32.10 | obnauticus | :/ |
| 03:32.12 | obnauticus | Wtf... |
| 03:32.16 | obnauticus | what kind of dtmf does DOES it support. |
| 03:32.22 | BBHoss | probably nothing |
| 03:32.25 | obnauticus | the only options it has for dtmf is `disable dtmf' |
| 03:32.25 | obnauticus | lol. |
| 03:32.32 | BBHoss | try iax |
| 03:32.39 | obnauticus | k :| |
| 03:32.41 | BBHoss | thats the only reason i would use zoiper |
| 03:32.56 | fujin | use um, the free eyebeam |
| 03:32.57 | fujin | x-lite |
| 03:33.01 | fujin | you can set the dtmf mode in that. |
| 03:33.08 | obnauticus | i'll get x-lite |
| 03:33.11 | obnauticus | never liked ZoIPer anyway |
| 03:33.53 | asteriskguy | is there any training for OpenSER in the US |
| 03:34.09 | asteriskguy | other then the one that just passed this last Nov 1st at VON |
| 03:35.52 | *** join/#asterisk TJNII_ (n=TJNII@209.234.89.226) |
| 03:36.11 | TJNII_ | Yea, it's the asctiontek |
| 03:36.29 | BBHoss | hehehe |
| 03:37.02 | obnauticus | kj it works |
| 03:38.06 | *** join/#asterisk speekac (n=alwin@60.51.217.61) |
| 03:38.11 | obnauticus | wtf is eyebeam 1.5 |
| 03:38.30 | BBHoss | xlite |
| 03:38.30 | obnauticus | I need the `transfer' feature |
| 03:38.30 | obnauticus | lol. |
| 03:38.45 | BBHoss | use zoiper with IAX2, it SHOULD work |
| 03:38.47 | TJNII_ | Now my SIP doesn't work..... |
| 03:39.03 | obnauticus | well for some reason BBHoss, my IAX2 isn't working :/ |
| 03:39.53 | BBHoss | TJNII_: trash that router |
| 03:40.03 | BBHoss | literally |
| 03:40.20 | BBHoss | don't stuff it in a closet, or donate it to chairity |
| 03:40.35 | *** join/#asterisk LakeSolon (n=blake@12-202-201-70.client.mchsi.com) |
| 03:41.43 | TJNII_ | BBHoss: I got this one because their next better router had to be factory reset every week.... |
| 03:42.00 | BBHoss | heh |
| 03:42.06 | BBHoss | are you on dsl or cable |
| 03:42.11 | *** join/#asterisk speshak (n=speshak@209.234.88.44) |
| 03:42.12 | TJNII_ | DSL |
| 03:42.17 | fujin | get a Linksys! |
| 03:42.28 | BBHoss | screw linksys |
| 03:42.35 | BBHoss | get a pfSense :) |
| 03:42.42 | fujin | what the crap is that? |
| 03:42.49 | BBHoss | pfsense.org |
| 03:42.52 | fujin | must be crap |
| 03:42.53 | fujin | xD |
| 03:42.55 | BBHoss | freebsd based router |
| 03:42.56 | obnauticus | okay BBHoss my ZoIPer isn't even trying to register with my server. |
| 03:42.58 | obnauticus | it's not doing ANYTHING |
| 03:42.58 | fujin | ah! |
| 03:42.59 | obnauticus | lol. |
| 03:43.01 | fujin | freebsd! crap! |
| 03:43.04 | obnauticus | it says registering |
| 03:43.10 | obnauticus | but i don't see any requests on asterisk |
| 03:43.11 | BBHoss | lets not start that shit |
| 03:43.18 | obnauticus | i <3 freebsd |
| 03:43.19 | fujin | lol@bsd. |
| 03:43.19 | obnauticus | anyweay |
| 03:43.26 | fujin | obnauticus: use x-lite |
| 03:43.28 | obnauticus | i say lol @ ubuntu. |
| 03:43.34 | obnauticus | I am using xlite, but I want to get iax working too. |
| 03:43.57 | obnauticus | aww F*ck it |
| 03:43.57 | fujin | oh, right. |
| 03:43.59 | obnauticus | i hate IAX2 anyway. |
| 03:44.01 | BBHoss | it uses pf from openbsd, then ALTQ from freebsd i believe |
| 03:44.05 | BBHoss | i LOVE iax2! |
| 03:44.11 | obnauticus | BBHoss pfsense? |
| 03:44.16 | BBHoss | yes |
| 03:44.19 | BBHoss | pfsense.org |
| 03:44.20 | obnauticus | I run pfsense. |
| 03:44.23 | obnauticus | remtard. |
| 03:44.27 | obnauticus | PfSense is pretty good |
| 03:44.30 | obnauticus | it has multi-wan routing support. |
| 03:44.32 | obnauticus | which is nice. |
| 03:44.42 | fujin | oh, really? |
| 03:44.42 | obnauticus | with load balancing and etc. |
| 03:44.46 | obnauticus | good traffic shaping |
| 03:44.46 | obnauticus | ya. |
| 03:44.55 | obnauticus | you can install 3rd party packages on it too |
| 03:45.01 | obnauticus | (eg. snort) |
| 03:45.02 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 03:45.06 | fujin | so, I could route multiple /32's to it? |
| 03:45.08 | bintut | hello all.. |
| 03:45.08 | obnauticus | I have a cisco catalyst |
| 03:45.13 | obnauticus | fujin, yes. |
| 03:45.25 | obnauticus | I have a cisco catalyst, and an Extreme Networks Alping 3808 |
| 03:45.30 | obnauticus | Alpine* |
| 03:45.33 | BBHoss | multiwan is a bit like dundi if you know what i mean |
| 03:45.39 | bintut | anyone here ever implemented a fax over ip on asterisk? what do you suggest? :) |
| 03:45.48 | obnauticus | Multiwan is nice...if you know how to use it. |
| 03:45.53 | asteriskguy | yes obnauticus |
| 03:45.56 | fujin | heh |
| 03:45.56 | asteriskguy | hylaFax |
| 03:45.57 | obnauticus | most people think it's like if oyu have two 30mbit lines it's 30+30 |
| 03:46.00 | fujin | multiwan could bypass many billing systems. |
| 03:46.05 | obnauticus | when it's actually 30&30 |
| 03:46.15 | fujin | can pfSense do ADSL? |
| 03:46.19 | asteriskguy | works ok, though I heard it's not reliable |
| 03:46.20 | obnauticus | Yes. |
| 03:46.31 | obnauticus | My Pfsense's box has been up for months |
| 03:46.31 | obnauticus | lol. |
| 03:46.35 | asteriskguy | but hylafax works ok in a test enviroment |
| 03:46.37 | BBHoss | yes |
| 03:46.44 | fujin | pppoatm? |
| 03:46.44 | obnauticus | PfSense is good. |
| 03:46.45 | obnauticus | imo. |
| 03:46.51 | obnauticus | ppp? |
| 03:46.51 | obnauticus | ya |
| 03:46.54 | obnauticus | it has VPN support |
| 03:46.58 | obnauticus | 802.11q vlan support |
| 03:47.01 | btorrenga | openvpn? |
| 03:47.03 | obnauticus | everything you could want, im not kidding. |
| 03:47.03 | obnauticus | yes |
| 03:47.06 | obnauticus | you can install openvpn |
| 03:47.12 | obnauticus | with a click of a button seriously. |
| 03:47.15 | BBHoss | you could put a supported card in there and it would probably work |
| 03:47.15 | fujin | pppoatm = adsl |
| 03:47.19 | obnauticus | i dunno. |
| 03:47.20 | fujin | ah, needs a card. |
| 03:47.21 | fujin | hrm. |
| 03:47.31 | obnauticus | It supports pretty much anything, or so im told |
| 03:47.43 | btorrenga | sangoma 518? |
| 03:47.48 | TJNII_ | I hear that guy speshak is a guru on routing. Perhaps we should ask him. |
| 03:47.51 | obnauticus | lol i right click in xlite, open diagnostics folder and it opens my temp folder |
| 03:47.53 | obnauticus | what a fail. |
| 03:48.31 | obnauticus | Whats an other good SIP phone |
| 03:48.34 | obnauticus | that supports tranfering |
| 03:48.35 | obnauticus | and etc. |
| 03:48.43 | btorrenga | windows? |
| 03:48.49 | *** join/#asterisk ai-a (n=jake2@cpc5-oxfd1-0-0-cust523.oxfd.cable.ntl.com) |
| 03:48.49 | fujin | xlite |
| 03:48.50 | obnauticus | ya |
| 03:48.51 | fujin | is the best free one |
| 03:48.54 | obnauticus | other than xlite |
| 03:48.55 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 03:48.58 | obnauticus | it won't let me transfer |
| 03:48.59 | fujin | what's wrong with xlite? |
| 03:49.00 | obnauticus | it's being a dickx. |
| 03:49.05 | fujin | you fail |
| 03:49.10 | obnauticus | wtf |
| 03:49.11 | obnauticus | dude |
| 03:49.16 | obnauticus | it's telling me to buy shit to `xfer' |
| 03:49.21 | fujin | oh, pwnt |
| 03:49.25 | fujin | warez it |
| 03:49.30 | obnauticus | altready tried |
| 03:50.00 | btorrenga | i recall a VOIP outfit in the mideast had the pay versionon their site |
| 03:50.19 | JT | obnauticus: eyebeam |
| 03:50.24 | obnauticus | free@ |
| 03:50.38 | fujin | hehe |
| 03:50.46 | fujin | ~cheap |
| 03:50.56 | jbot | from memory, cheap is a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 03:50.56 | obnauticus | I'm 15. |
| 03:50.59 | obnauticus | I don't pay for Sh*t |
| 03:51.05 | fujin | You're doing it wrong. |
| 03:51.10 | bintut | anyone? |
| 03:51.11 | fujin | Get a job at a petrol station, buy eyebeam |
| 03:51.12 | fujin | game over |
| 03:51.18 | obnauticus | ... |
| 03:51.20 | Nivex | or, as I like to say "Free shit is still shit" |
| 03:51.22 | btorrenga | jbot: true true true... |
| 03:51.23 | obnauticus | I have a cisco SIP Phone in the mail |
| 03:51.24 | obnauticus | remtard |
| 03:51.26 | obnauticus | that's coming soon |
| 03:51.29 | obnauticus | so i'll kick your ass |
| 03:51.30 | obnauticus | rofl. |
| 03:51.47 | fujin | O WOW A CISCO? |
| 03:51.48 | fujin | THATS L##T |
| 03:51.51 | obnauticus | not really |
| 03:51.52 | obnauticus | it are fail. |
| 03:51.53 | btorrenga | haha |
| 03:51.55 | obnauticus | Or so im told. |
| 03:52.01 | obnauticus | I wanna try CCM too though. |
| 03:52.06 | JT | especially if it has SCCP firmware |
| 03:52.12 | fujin | You'll need a better job than petrol station. |
| 03:52.20 | obnauticus | JT I can get SCCP and SIP firmware for free. |
| 03:52.21 | obnauticus | or |
| 03:52.23 | obnauticus | SIP firmware :/ |
| 03:52.26 | btorrenga | fujn, where are you at? |
| 03:52.26 | obnauticus | And CCM supports SIP. |
| 03:52.44 | btorrenga | "petrol" |
| 03:52.55 | *** join/#asterisk bmg505 (n=leon@196.209.183.44) |
| 03:53.15 | fujin | I'm in New Zealand. |
| 03:53.33 | fujin | Petrol, evidently, is short for petrolium |
| 03:53.45 | btorrenga | yes |
| 03:54.02 | btorrenga | is it 9am or so by you right now? |
| 03:54.19 | fujin | 4:54pm, nzdt |
| 03:55.02 | btorrenga | btw, thanks for that resolv.conf idea. I'm still waiting for IPkall to take my new settings, though. |
| 03:56.08 | fujin | :D |
| 04:00.39 | bintut | anybody knows where can i find a good documentation on the best practices of implementing fax over ip on asterisk? |
| 04:01.52 | BBHoss | what do you mean by over ip |
| 04:02.13 | BBHoss | sip/iax trunks, or zaptel-->sip/iax2-->fax machine |
| 04:02.42 | JT | btorrenga: what's wrong with the word petrol? |
| 04:02.50 | btorrenga | haha, nothing |
| 04:02.59 | btorrenga | it just caught my eye |
| 04:03.15 | bintut | BBHoss: fax machine (sender) => asterisk => iax2 peering over internet => asterisk => fax machine (receiver) |
| 04:03.20 | btorrenga | in Chicago we say "gas" |
| 04:04.04 | J4k3 | I have the cheapest hardware sip phones on the market (that I know of) and I'm perfectly content with them (ok not really, but they're 100% functional as what they are...) |
| 04:04.07 | BBHoss | ahh |
| 04:04.55 | fujin | btorrenga: don't you also say 'cawfee'? |
| 04:04.57 | fujin | or is that NY |
| 04:04.59 | TJNII | And now, magically, sip is working again. |
| 04:04.59 | J4k3 | I'd rather, say, go on a long-weekend vacation with my money, than say, dump it on something that I couldn't really give a rats ass less about |
| 04:05.07 | BBHoss | you really want t.38 support |
| 04:05.27 | btorrenga | depends if you're from the city. "Awwfice", too. |
| 04:05.48 | fujin | Heh. |
| 04:05.54 | bintut | BBHoss: and i am talking about the usual way of sending fax from a typical fax machine not knowing the sender (person) is faxing it over ip.. that means, the sender (person) fax it the usual way of faxing a document |
| 04:06.07 | BBHoss | and i think thats only on SIP right now, but i may be wrong |
| 04:06.10 | fujin | ughrh |
| 04:06.16 | fujin | J4k3: what about 'aint no'? |
| 04:06.19 | fujin | double negative! |
| 04:06.25 | J4k3 | and the most non-racist white person there still drops the n-bomb in every other sentence... good lord they're racist. |
| 04:06.28 | *** join/#asterisk Law (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 04:07.27 | J4k3 | I've been exposed to too many cultures to be racist, myself. I find everyone sucks equally, in their own special way |
| 04:08.06 | J4k3 | fujin: I didn't catch any of that. just 'yall' |
| 04:08.18 | J4k3 | which is just a southern version of 'youse guyz' |
| 04:08.27 | btorrenga | don't worry, itll wear off. |
| 04:09.08 | btorrenga | youse guyezes |
| 04:09.08 | J4k3 | people from new joy-zee say 'youse guys', and it makes me cringe. |
| 04:09.08 | btorrenga | sandwich = sangwich |
| 04:09.16 | J4k3 | I'm ready to fight another civil war over the yall/youse guys situation. |
| 04:09.25 | BBHoss | heh |
| 04:09.25 | BBHoss | bintut, all you need is * 1.4, some ATAs that support t.38, some fax machines that work well with t.38, and some elbow grease |
| 04:09.31 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 04:09.32 | BBHoss | bintut: yes i understand that |
| 04:09.36 | BBHoss | two atas with t.38 support |
| 04:09.38 | BBHoss | then * 1.4 with t.38 passthrough enabled |
| 04:10.02 | btorrenga | BBH: can you recommend an ATA for T38? |
| 04:10.33 | bintut | BBHoss: you mean, t.38 is not in the current asterisk 1.4? |
| 04:10.34 | fujin | uh |
| 04:10.35 | fujin | fax=bad idea |
| 04:10.48 | J4k3 | fax = dead |
| 04:10.57 | btorrenga | fax = callweaver? |
| 04:11.35 | BBHoss | fax is not dead, believe me... |
| 04:11.54 | fujin | it's dead |
| 04:11.59 | fujin | by a copier than can do fax->emailo |
| 04:12.01 | bintut | fax will definitely cannot be dead at all |
| 04:12.04 | fujin | provision an analogue line |
| 04:12.07 | fujin | be done with the bloody thing |
| 04:12.37 | Kobaz | at my last job i had to work on printing out forms from the accounting system so they could be faxed to a system that did ocr and then processed the forms |
| 04:12.42 | BBHoss | some people cant afford copiers that do fax |
| 04:12.55 | Kobaz | instead of uhh, submitting something to a simple web form |
| 04:12.58 | bintut | BBHoss: precisely |
| 04:13.06 | *** join/#asterisk axscode (n=axscode@132.240.208.218.klj02-home.tm.net.my) |
| 04:13.25 | BBHoss | 1.4 has t.38 passthrough, but nothing else |
| 04:13.36 | bintut | BBHoss: you mean, t.38 is in the current asterisk 1.4? |
| 04:13.40 | BBHoss | yes |
| 04:14.00 | bintut | BBHoss: so, what shall i need to make my faxing requirement work? |
| 04:14.01 | BBHoss | http://www.voip-info.org/wiki/view/Asterisk+T.38 somewhat good info |
| 04:14.05 | TJNII | Hmmmm... It seems my system clock is off by about 5 hours..... despite the use of ntp... |
| 04:14.06 | bintut | ok |
| 04:14.30 | BBHoss | the sipuras usually work good |
| 04:14.46 | btorrenga | now Lynksys, right? |
| 04:14.58 | BBHoss | yeah |
| 04:16.01 | obnauticus | Ew Linksys |
| 04:16.14 | fujin | Linksys is fine. |
| 04:16.17 | fujin | They're awesome, in fact |
| 04:16.17 | bintut | BBHoss: is it possible to make fax over ip work without the use of an ATA? |
| 04:16.21 | obnauticus | ARe you serious? |
| 04:16.25 | obnauticus | other than being total crap you are right. |
| 04:16.27 | fujin | better than GS/Cisco/Mitel |
| 04:16.34 | BBHoss | maybe |
| 04:16.34 | obnauticus | Linksys > Cisco |
| 04:16.38 | obnauticus | that is ... |
| 04:16.40 | obnauticus | disturbing. |
| 04:16.42 | fujin | That's what I said. |
| 04:16.50 | fujin | ~phones |
| 04:17.01 | jbot | i guess phones is http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. places ... |
| 04:17.01 | obnauticus | OHH |
| 04:17.02 | fujin | spa9x2 series. |
| 04:17.02 | obnauticus | I thought you were tlaking about routersd. |
| 04:17.08 | fujin | No. not routers. heh. |
| 04:17.15 | fujin | Comparing cisco to linksys for routers |
| 04:17.16 | fujin | is like |
| 04:17.18 | fujin | yeah, no. |
| 04:17.23 | fujin | DOES NOT COMPUTE |
| 04:17.25 | fujin | INVALID EXPRESSION |
| 04:17.48 | btorrenga | Aastra > Cisco? |
| 04:18.05 | fujin | absolutely |
| 04:18.07 | J4k3 | jbot, a service brought to you by Polycom (tm) |
| 04:18.11 | *** join/#asterisk axisys_ (n=axisys@ip70-174-179-120.dc.dc.cox.net) |
| 04:18.12 | btorrenga | cisco is low on that list due to features, not quality, right? |
| 04:18.18 | fujin | no, quality |
| 04:18.23 | btorrenga | really |
| 04:18.24 | fujin | and manageability |
| 04:18.31 | btorrenga | I can see manageability |
| 04:18.33 | fujin | put a polycom and a cisco side by side |
| 04:18.42 | btorrenga | but Cisco's are pretty solid, so I thought. |
| 04:18.53 | fujin | eh, in a CCM environment, maybe.. |
| 04:19.00 | BBHoss | yeah |
| 04:19.00 | btorrenga | ah. |
| 04:19.05 | BBHoss | they are designed to work with CCM |
| 04:19.10 | J4k3 | btorrenga: if you believe the people in here, your whole system will fall over dead on its face unless you buy polycom |
| 04:19.18 | J4k3 | YOU MUST BUY POLYCOM, OMG, YOU MUST. |
| 04:19.19 | btorrenga | haha |
| 04:19.21 | fujin | that's incorrect, and an exageration |
| 04:19.25 | fujin | although, we do suggest buying Polycom |
| 04:19.27 | J4k3 | fujin: then prove otherwise. |
| 04:19.29 | fujin | as they're awesome(tm) |
| 04:19.37 | fujin | J4k3: I run linksys phones, my system hasn't fallen over |
| 04:19.39 | fujin | <proven> |
| 04:19.39 | fujin | eof. |
| 04:19.41 | fujin | next question |
| 04:19.52 | J4k3 | I run grandstream budgetone 101's, same result. |
| 04:19.55 | fujin | sweet |
| 04:20.07 | fujin | everyone knows grandstream is for cheap-asses, featureset wise |
| 04:20.16 | fujin | wasn't there just an exploit published for them, too? |
| 04:20.18 | J4k3 | I pick up the phone, people talk, they hang up. |
| 04:20.19 | fujin | re: sip NOTIFY |
| 04:20.26 | J4k3 | fujin: who gives a rats ass? |
| 04:20.39 | fujin | I do, as an engineer concerned about security. |
| 04:20.58 | J4k3 | if you're worried about security you run much better hardware than this crappy SIP shit |
| 04:20.59 | fujin | Considering you *chose* grandstream, I understand why you don't. |
| 04:21.21 | J4k3 | and polycom has never had an exploit? |
| 04:21.23 | J4k3 | lets see... |
| 04:21.42 | fujin | no idea, I wasn't making a point that they haven't |
| 04:21.48 | fujin | I was making a point that grandstream had one just recently |
| 04:22.15 | J4k3 | but your point lacked sharpness |
| 04:22.21 | J4k3 | try again. |
| 04:22.47 | J4k3 | see, this channel basically pushes the 'you must spend a lot on phones to expect anything to work at all' which is quite simply incorrect. |
| 04:23.00 | fujin | spend a lot on PHONES AND HARDWARE |
| 04:23.17 | J4k3 | most people need neither |
| 04:23.24 | fujin | most people are happy with trixbox |
| 04:23.40 | J4k3 | exactly. |
| 04:23.41 | btorrenga | uggg |
| 04:24.23 | J4k3 | makes calls, takes calls, doesn't screw up. |
| 04:26.19 | J4k3 | so exactly what is the problem? |
| 04:26.19 | btorrenga | to get "under the hood" requires jumping through some hoops, doesnt it? |
| 04:26.19 | J4k3 | who cares |
| 04:26.19 | J4k3 | if I wanted more I wouldn't use trixbox :P |
| 04:26.19 | btorrenga | perfect |
| 04:26.19 | J4k3 | if I wanted more from a phone, I wouldn't buy grandstream |
| 04:26.19 | J4k3 | but quite simply the way this channel puts it, neither product ever works |
| 04:26.20 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 04:26.20 | J4k3 | and thats quite simply wrong, and shows a lot of ignorance on those who take part in it. |
| 04:27.29 | fujin | I'm heading home, seeyas. |
| 04:27.45 | btorrenga | see ya |
| 04:27.48 | btorrenga | thanks again |
| 04:28.11 | [TK]D-Fender | btorrenga, So looking to get into * and wondering what to get? |
| 04:28.12 | TJNII | It's OK J4k3. I run Grandstreams too. |
| 04:28.34 | btorrenga | as far as phones or * hardware? |
| 04:28.38 | TJNII | It's not that bad, I mean I can't her the echo on the speakerphone, after all |
| 04:28.52 | [TK]D-Fender | btorrenga, Yes, both. |
| 04:29.17 | btorrenga | I use Cisco 7940's and 7960's with the SIP firmware |
| 04:29.27 | btorrenga | and just a robust P4 with a gig or 2 of RAM |
| 04:29.31 | [TK]D-Fender | btorrenga, currently? |
| 04:29.33 | btorrenga | yes |
| 04:29.39 | *** join/#asterisk vnn (n=nostalge@modemcable083.222-80-70.mc.videotron.ca) |
| 04:29.44 | [TK]D-Fender | btorrenga, And how are they? |
| 04:29.54 | btorrenga | and sangoma PSTN cards for analogue lines |
| 04:30.00 | J4k3 | TJNII: I've never done anything with the speakerphone except check voicemail *Shrug* |
| 04:30.14 | btorrenga | I like them, though the Cisco's are outdated nowadays |
| 04:30.21 | J4k3 | my cellphone has an excellent speakerphone, I've used it a couple times. |
| 04:30.25 | btorrenga | people talk a lot about Polycom phones |
| 04:30.38 | [TK]D-Fender | btorrenga, I'm not sure I'd say outdated so much as not focused on SIP... |
| 04:30.43 | J4k3 | you're going to spend a lot on phones and cards |
| 04:30.46 | J4k3 | and plug fucking POTS into it? |
| 04:30.51 | TJNII | J4k3: Echoy speakerphone and numeric only caller ID. Otherwise I have no complaints about bt100s. |
| 04:31.12 | J4k3 | TJNII: yep, agreed. |
| 04:31.21 | btorrenga | only 3 or four PSTN lines at the offices |
| 04:31.39 | btorrenga | we used TDM400P's for a few years |
| 04:31.43 | JT | btorrenga: why use POTS :) |
| 04:31.47 | btorrenga | switched to Sangoma |
| 04:31.55 | btorrenga | reliability |
| 04:32.01 | *** part/#asterisk vnn (n=nostalge@modemcable083.222-80-70.mc.videotron.ca) |
| 04:32.01 | btorrenga | cant afford T1 |
| 04:32.08 | btorrenga | ADSL |
| 04:32.14 | J4k3 | ISDN BRI is pretty solid around here |
| 04:32.25 | [TK]D-Fender | btorrenga, Sounds like a decent setup. Typically Polycoms are a fair bit cheaper than Cisco, and offer better call handling, SIP support, at a noticably lower price. |
| 04:32.26 | J4k3 | and cheap, at least compared to buying a lot of business POTS. |
| 04:32.33 | J4k3 | a BRI costs about 85% of what two B1s runs. |
| 04:32.44 | [TK]D-Fender | btorrenga, But you could do far worse. |
| 04:32.46 | btorrenga | not by me |
| 04:32.56 | De_Mon | arg _find-XXX anyone care to guess why this doesnt match Goto(mycontext,find-800,1) |
| 04:33.01 | btorrenga | BRI's were way expensive when I priced it a while back |
| 04:33.15 | J4k3 | of course, I also live in a weird place where my T1+ITSP is more reliable than my friggin ILEC-delivered POTS line. |
| 04:33.19 | [TK]D-Fender | btorrenga, Where are you located? |
| 04:33.32 | btorrenga | Chicagoland, northwest indiana actually. |
| 04:33.43 | btorrenga | one office is AT&T the other is Verizon |
| 04:33.51 | [TK]D-Fender | btorrenga, Yup... BRI = huh what?! ;) |
| 04:33.58 | btorrenga | haha, ya |
| 04:33.59 | J4k3 | I was going to terminate it into my * box, then realized the crappy modem-turned-pots-adapter I had a ton of overhead |
| 04:34.04 | [TK]D-Fender | btorrenga, So its either partial PRI or analog... |
| 04:34.07 | De_Mon | does anyone really use lowercase wildcards for pattern matching? nxxnxxx |
| 04:34.32 | J4k3 | of course, terminating BRI is a whore in * |
| 04:34.41 | J4k3 | there is *no* reasonably priced gear |
| 04:34.42 | btorrenga | as far as I was concerned, a partial PRI made sense at about 7 analogue lines as far as price goes |
| 04:35.28 | J4k3 | PRI starts making sense if you're running a lot of traffic |
| 04:35.45 | J4k3 | makes sense to get a better internet connection with a decent SLA, and an ITSP |
| 04:35.49 | J4k3 | usually |
| 04:36.24 | [TK]D-Fender | btorrenga, cost effectiveness comes at one factor, and functionality if needed has its own judgement. |
| 04:37.08 | btorrenga | functionality is fine, we have hunt groups on the analogue lines, and then forward-on-busy to a VOIP toll-free number |
| 04:37.12 | De_Mon | how do you excape wildcards in the dialplan I've tried both _fi\nd-XXX and _fi\\nd-XXX and it still cant find my extension |
| 04:37.41 | btorrenga | De_>, I dont think you can. I think you need to use a regex function |
| 04:37.54 | btorrenga | (someone correct me if I am wrong) |
| 04:38.03 | De_Mon | huh? regex in an extension? |
| 04:38.19 | *** join/#asterisk serpent-fly (n=serpent@194.79.34.10) |
| 04:38.31 | btorrenga | yeah, like match to _. or _X., and then use a regex function to evaluate ${EXTEN} |
| 04:38.50 | btorrenga | (I think) |
| 04:38.57 | obnauticus | Hey., what's that asterisk driver called where you can use a cell phone via USB |
| 04:39.01 | obnauticus | as a trunk. |
| 04:39.05 | J4k3 | icagoland, northwest indiana actually. |
| 04:39.06 | J4k3 | 22:33 < btorrenga> one icagoland, northwest indiana actually. |
| 04:39.07 | btorrenga | chan_mobile |
| 04:39.14 | J4k3 | ack |
| 04:39.18 | J4k3 | sorry about that, mouse insanity |
| 04:39.31 | De_Mon | oh, good grief that wouldn't work |
| 04:39.50 | De_Mon | it would _work_ but its not the design I had in mind |
| 04:40.03 | De_Mon | having to avoid n is annoying |
| 04:40.04 | J4k3 | err |
| 04:40.13 | btorrenga | I dont think you can escape matching |
| 04:40.15 | obnauticus | btorrenga does it work via USB yet? |
| 04:40.15 | J4k3 | I don't think, at least CDMA phones, you can do voice-over-usb |
| 04:40.17 | *** join/#asterisk axscode (n=axscode@132.240.208.218.klj02-home.tm.net.my) |
| 04:40.34 | obnauticus | Damn it. |
| 04:40.41 | btorrenga | someone I spoke with had it working |
| 04:40.41 | J4k3 | obnauticus: bluetooth is the only way to go, I *know* qualcomm cdma phones don't/won't push voice over the data cable. |
| 04:40.51 | De_Mon | I'm pretty sure you could in 1.2, I just switched methods before figuring out the right way |
| 04:40.53 | btorrenga | over bluetooth |
| 04:40.58 | De_Mon | and never implimented it after I did find it |
| 04:40.58 | J4k3 | you *might* be able to do it via a sound card and the earphone plug, dialing/answering via usb and AT codes. |
| 04:41.00 | obnauticus | i can solder the mic to a twisted pair |
| 04:41.04 | obnauticus | and plug it into an FXO |
| 04:41.07 | obnauticus | and for the speaker |
| 04:41.10 | obnauticus | plug it into an FXS |
| 04:41.11 | J4k3 | obnauticus: go to a sound card. |
| 04:41.15 | obnauticus | no |
| 04:41.21 | obnauticus | like into FXO/FXS lol. |
| 04:41.32 | J4k3 | that'd be expensive/worthless. |
| 04:41.36 | obnauticus | I know. |
| 04:41.44 | J4k3 | just do it via bt, bt-equipped phones are cheap |
| 04:41.54 | J4k3 | of course, you start running into a lot of shitty codecs chained together |
| 04:41.59 | obnauticus | Where can i get a bluetooth adpter :/ |
| 04:42.01 | obnauticus | adapter* |
| 04:42.01 | btorrenga | it made my box explode on incoming calls |
| 04:42.03 | J4k3 | I have no idea how bad the end result sounds, I can't imagine dtmf'ing over it :) |
| 04:42.22 | J4k3 | obnauticus: pretty much anybody that sells flash memory sells usb bt adapters. |
| 04:42.22 | btorrenga | outbound calling seemed to work fine. |
| 04:42.46 | J4k3 | at least around here... office depot, staples, best buy, circuit city, radio shack, etc. |
| 04:42.55 | J4k3 | or spend about 1/3rd as much and buy online |
| 04:44.02 | obnauticus | rofl |
| 04:44.12 | obnauticus | moving x-lite around the screen on windows really fast reminds of me kidpix. |
| 04:44.17 | obnauticus | anyone here remember kidpix? |
| 04:45.22 | obnauticus | http://comsewogue.k12.ny.us/~ssilverman/whaletales/hirner/hirner.htm <-- our future. |
| 04:45.53 | obnauticus | Problem: How does blubber help whales? |
| 04:45.59 | btorrenga | I thought telephony was our future? |
| 04:46.03 | obnauticus | Hypohtisis: I think blubber helps them stay warm in cold water! |
| 04:46.21 | obnauticus | btorrenga that's a common misconception braught upon the american citizens by the United States government. |
| 04:46.27 | obnauticus | Primarily Barack Obama. |
| 04:46.39 | obnauticus | That page has the key to our future. |
| 04:46.50 | obnauticus | Look, the kids even drew the sperm whale! |
| 04:47.34 | *** join/#asterisk IgorG (n=FeedomPa@195.162.32.126) |
| 04:49.54 | TJNII | J4k3: Echoy speakerphone and numeric only caller ID. Otherwise I have no complaints about bt100s. |
| 04:50.14 | J4k3 | TJNII: yep |
| 04:50.41 | TJNII | Sorry, that up enter was ment for another window |
| 04:51.04 | btorrenga | bed time. |
| 04:51.08 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 04:51.14 | *** part/#asterisk btorrenga (n=btorreng@adsl-68-75-160-56.dsl.emhril.ameritech.net) |
| 04:53.25 | *** join/#asterisk s0lid (n=_freq@60.51.125.159) |
| 05:04.12 | JT | J4k3: bri is cheap to terminate a single BRI in asterisk if it has ETSI signalling |
| 05:06.04 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 05:09.45 | *** join/#asterisk metabsd (n=metabsd@modemcable103.201-131-66.mc.videotron.ca) |
| 05:17.46 | phix | hmmmmm |
| 05:17.58 | phix | What is the advantage of call groups? |
| 05:18.28 | *** join/#asterisk IgI (n=FeedomPa@195.162.32.126) |
| 05:19.17 | *** join/#asterisk xtr-II (n=94752345@216.19.191.191.novuscom.net) |
| 05:20.57 | [pyro] | hmm does asterisk support BLA yet? |
| 05:23.04 | *** join/#asterisk Hadi- (n=Hadi@CPE001310492769-CM001225e00576.cpe.net.cable.rogers.com) |
| 05:23.10 | *** join/#asterisk BobbieG (n=bob@209.146.182.130) |
| 05:23.15 | [TK]D-Fender | [pyro], No. |
| 05:23.32 | [TK]D-Fender | [pyro], And from what I hear, not in 1.6 either |
| 05:25.37 | FremWork | you sure? |
| 05:25.47 | FremWork | BLA is the busy lamp thing yeh? |
| 05:26.27 | TJNII | http://www.doretel.com/cisco-armored-products.php |
| 05:26.35 | grimsy | bridged line appearance i thought |
| 05:26.46 | ManxPower | Doesn't asterisk call that SLA? |
| 05:26.58 | BobbieG | has anyone had an issue with the queues on 1.4.13? |
| 05:27.05 | [TK]D-Fender | ManxPower, Same thing, equaly fictional. |
| 05:27.14 | De_Mon | BobbieG no.. |
| 05:28.13 | *** join/#asterisk serpent-fly (n=serpent@194.79.34.10) |
| 05:28.21 | FremWork | ah, what I was thinking of was BLF |
| 05:28.32 | BobbieG | i know it is one setting i was using in an earlier version fine just not sure which one and using so many |
| 05:28.42 | ManxPower | FremWork: And that is why it is important to get the words right. |
| 05:28.58 | ManxPower | If you just want BLF...well that is in 1.2+ |
| 05:29.29 | ManxPower | Because of one letter, we have lost 5 mins of our lives, which we will never get back. |
| 05:30.13 | FremWork | SLA BLF BLA SIP IAX CID DID... I need a break from the acronyms (and that's just some of the asterisk one's I deal with daily... don't get me started on ARS, TMS, etc |
| 05:30.35 | grimsy | good old TLA's |
| 05:30.50 | FremWork | I concur |
| 05:32.43 | hellop | I should have concurred! |
| 05:32.44 | *** join/#asterisk izaak (n=izaak@modemcable132.248-130-66.mc.videotron.ca) |
| 05:33.12 | *** join/#asterisk axscode (i=axscode@58.26.60.120) |
| 05:33.26 | [pyro] | haha |
| 05:33.42 | [pyro] | [TK]D-Fender: i was just reading about SLA's here |
| 05:33.42 | [pyro] | http://www.voip-info.org/wiki/view/Asterisk+SLA |
| 05:34.17 | *** join/#asterisk moprilo (n=jjohn@sv-cpe-dynamic-190-53-14-251.amnetsal.com) |
| 05:34.31 | moprilo | i did "modprobe ztdummy", but i need to undo it.. how do i do that? |
| 05:35.11 | [TK]D-Fender | moprilo, "rmmod ztdummy" |
| 05:35.40 | phix | Service Level Agreements? |
| 05:35.56 | moprilo | modprobe -r did it .. thanks |
| 05:39.28 | *** join/#asterisk atomicd (n=atomicd@adsl-69-109-58-155.dsl.irvnca.pacbell.net) |
| 05:44.46 | [pyro] | it looks like SLA is supported and should work (tm) |
| 05:44.57 | [pyro] | turn asterisk into a keying system :) |
| 05:45.08 | [pyro] | http://www.asterisk.org/node/48342 |
| 05:45.39 | [TK]D-Fender | [pyro], that is a fugly 2-bit hack pretending to be SLA |
| 05:46.13 | [pyro] | [TK]D-Fender: but does it work? |
| 05:47.42 | [TK]D-Fender | [pyro], in some limited capacity I suppose, but with vulnerabilities, and a requirement for a large number of BLF capable speed-dials |
| 05:47.48 | [TK]D-Fender | [pyro], And only for LINES. |
| 05:48.03 | [TK]D-Fender | [pyro], No comment about the unnaturalness of it all. |
| 05:49.15 | *** join/#asterisk Steven_elvisda (n=Steven_E@202.47.107.60) |
| 05:49.20 | [pyro] | [TK]D-Fender: yeah i dont think ill set it up here at work as i dont need it. But some people just wont look at asterisk as a solution unless it can do SLA |
| 05:49.44 | [pyro] | id rather just pick up the phone and dial, who cares what line it uses |
| 05:49.45 | [TK]D-Fender | [pyro], pathetic. no need really. Thats what parking is for. |
| 05:50.37 | ManxPower | [pyro]: Really Asterisk is not the solution if they need the features of a real Key System |
| 05:50.55 | [pyro] | yeah |
| 05:51.35 | [TK]D-Fender | ManxPower, And there is virtually no need for key system style management. |
| 05:51.59 | [pyro] | hmm my aastra phones when paged dont beep |
| 05:52.28 | [pyro] | the page connects to the phone and you can hear whats going on at said phone. No beep warning for the paged party :) |
| 05:53.49 | ManxPower | [TK]D-Fender: My customers like BLF quite a but. |
| 05:53.55 | ManxPower | but they don't need shared lines, etc |
| 05:54.14 | [TK]D-Fender | ManxPower, BLF hell yeah.. key system channel grabbing? No. |
| 05:57.28 | *** join/#asterisk IgorG (n=FeedomPa@195.162.32.126) |
| 05:57.45 | ManxPower | *nod* |
| 05:57.51 | [pyro] | doesnt asterisk send a beep after *80 to page an extension? |
| 05:58.12 | [TK]D-Fender | [pyro], *80? no such thing... what did YOU do there? |
| 05:58.14 | ManxPower | My clients want BLF just so the operator can tell the caller that Ms. Realestate Agent Asshole is on the phone. |
| 05:58.30 | [pyro] | [TK]D-Fender: hehe |
| 05:58.42 | [pyro] | [TK]D-Fender: yes i know my soul is forefit |
| 05:58.47 | ManxPower | [pyro]: it is up to the phone to auto answer the call and play a beep |
| 05:58.50 | [pyro] | i shall ask in the correct channel |
| 05:58.51 | [pyro] | :D |
| 05:58.56 | [TK]D-Fender | [pyro], The first step is admitting you have a problem. |
| 05:59.04 | [pyro] | ManxPower: yeah i have the option setup in the phone to play the beep |
| 05:59.12 | [pyro] | ManxPower: .. and autoanswer |
| 05:59.19 | ManxPower | [pyro]: you realize we have a herd of aligators that have been specially trained to crave meat tainted with a GUI, right? |
| 05:59.28 | [pyro] | lol |
| 05:59.37 | [TK]D-Fender | Crikey! |
| 05:59.52 | [TK]D-Fender | Oi she's a beaut! |
| 05:59.54 | [pyro] | oh comon guys, whats wrong with a GUI? :) its nice and fast!! |
| 06:00.01 | ManxPower | It's the only thing that works to keep out the GUI people |
| 06:00.07 | [pyro] | lol' |
| 06:00.11 | ManxPower | [pyro]: it is also IMPOSSIBLE for us to troubleshoot. |
| 06:00.21 | ManxPower | ~zeeek |
| 06:00.54 | jbot | hmm... 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." |
| 06:00.54 | [pyro] | ManxPower: yeah i can see how that would be a problem |
| 06:00.59 | ManxPower | [pyro]: it's like if you bought a totally customized ford car, custom engine, custom power windows, custom power train, etc. Then you bring that car to a ford dealer and expect them to fix it. |
| 06:01.02 | [TK]D-Fender | [pyro], Basically you can't fine tune SHIT in there and noone wants to much around in their tunnel-visioned world. |
| 06:01.42 | ManxPower | well, Asterisk GUIs totally customize Asterisk with really bizarre stuff and designs...then people expect us to fix it. |
| 06:02.16 | [pyro] | yeah granted |
| 06:02.51 | [TK]D-Fender | *80? Ask me how much I care about design flaws you can't even explain because you didn't make it. Think we're going to track it down only to watch any effort get tossed the next time you commit a change? No way in hell. |
| 06:02.58 | ManxPower | We say "ok, show us the CLI output of a failed call", expecting to see 5 lines or so of output. If they are using a GUI it is a hundred lines (at least!) of custom AGI scripts, custom macros, and custom dialplan design. |
| 06:02.59 | lowlevel | you need crippling social disorders to learn asterisk. :) |
| 06:03.00 | [pyro] | you guys are the only ones that answer questions too, thats whats frustrating. Not much convo happens in #freepbx |
| 06:03.36 | ManxPower | [pyro]: go ask about Redhat on a Debian channel. |
| 06:03.40 | lowlevel | what do you expect for free? |
| 06:03.46 | tzanger | ManxPower: I need you and [TK]D-Fender to help me beat up the guy I'm currently contracting for |
| 06:03.49 | tzanger | he doesn't get that |
| 06:03.54 | [pyro] | [TK]D-Fender: yeah *80 comes from a freepbx module called Paging and Intercom |
| 06:04.00 | [TK]D-Fender | [pyro], They are equally clueless. If you;re using a GUI you are typically not messing around with it |
| 06:04.02 | lowlevel | you should just get northern telecom stuff from some telecom provider |
| 06:04.03 | ManxPower | tzanger: Those are what I call "former clients" |
| 06:04.19 | tzanger | I just keep telling him "I make magic happen with the drivers for your hardware. stop asking me to make the gui work, that's not what I'm good at" |
| 06:04.19 | [TK]D-Fender | tzanger, No news on my new blade :( |
| 06:04.20 | lowlevel | or cheaper... panasonic or somthing |
| 06:04.28 | tzanger | [TK]D-Fender: how long did they say it'd take? |
| 06:04.29 | [TK]D-Fender | tzanger, But my old one is still nice & sharp :D |
| 06:04.34 | s0lid | hi anyone tried asterisk on macos? |
| 06:04.37 | ManxPower | tzanger: One of my customers occasionally whines bout lack of a GUI. I say "OK, give me a list of design requirements for a GUI for you" and then he is silent for a few more months |
| 06:04.45 | lowlevel | s0lid; I want to... but I need IP phones first :/ |
| 06:04.46 | s0lid | im using mezzo packages currently and need some info about it? |
| 06:04.46 | tzanger | [TK]D-Fender: hahaha I needed a nice sharp blade to cut my roast beef tonight |
| 06:04.47 | [TK]D-Fender | tzanger, about 4 weeks, but I never heard owrd sine making payment.... |
| 06:05.02 | [TK]D-Fender | tzanger, Then you're doing something very wrong :) |
| 06:05.06 | s0lid | lowlevel: you need it as in you need to buy one? what country you from? |
| 06:05.07 | *** join/#asterisk axscode (i=axscode@58.26.60.120) |
| 06:05.18 | [TK]D-Fender | tzanger, my fillet mignon has been cutting like butter all week :) |
| 06:05.22 | tzanger | ManxPower: this guy has a decent list of requirements, but I keep telling him if he wants that he's gonna have to write it. $2500 is too much for switchvox for 1500 channels and freepbx just isn't filling it |
| 06:05.28 | [pyro] | [TK]D-Fender: ManxPower: no stress im not asking you to fix anyting or blaming you guys for anything. its good to be in here because at least you guys talk :) |
| 06:05.37 | tzanger | [TK]D-Fender: next time I'm gonna cook a roast at 300... this was 350 and it was good but not perfect |
| 06:05.42 | lowlevel | solid: I mostly use macs at home, but I have to keep a linux box to run the phones cause I went with just using old analog phones on a digium card in linux. |
| 06:05.47 | tzanger | seared the outside with the hottest pan I could make |
| 06:05.57 | tzanger | lowlevel: use an ATA |
| 06:05.58 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 06:06.05 | [TK]D-Fender | tzanger, mine is about 6min a side pre-heat to 500 and then set to broil. |
| 06:06.06 | tzanger | [TK]D-Fender: that's always worrying |
| 06:06.09 | [pyro] | k gtg talk later :) |
| 06:06.14 | lowlevel | solid: so.. not only would I need asterisk to work.. I'de need a pci bus... or yeah... about 4 ata's |
| 06:06.14 | *** part/#asterisk [pyro] (n=Pyro@tor/regular/bracketed-pyro) |
| 06:06.18 | ManxPower | lowlevel: Even if you could get Asterisk to work on a Mac, the userbase would be so small you would get virtually no help from the community |
| 06:06.21 | [TK]D-Fender | tzanger, comes out blue/rare. |
| 06:06.23 | [TK]D-Fender | yummmmmmmmmmmmm |
| 06:06.31 | tzanger | [TK]D-Fender: I love it like that... wife likes it well done however :-( |
| 06:06.33 | s0lid | lowlevel: just get ATA it would be cheaper and faster to imlement |
| 06:06.40 | ManxPower | My first Asterisk install was on analog phones. NEVER EVER AGAIN. |
| 06:06.42 | s0lid | lowlevel: what do you want to do anyway |
| 06:06.51 | lowlevel | manx; uhm... when did I ask for support? |
| 06:06.52 | lowlevel | *boggle* |
| 06:06.53 | lowlevel | ;) |
| 06:06.57 | tzanger | anyway it's late... I gotta get ot bed :-( |
| 06:07.19 | ManxPower | lowlevel: Just making sure you are not in the middle of a fit of insanity. |
| 06:07.32 | lowlevel | no. i've been using asterisk for over a year |
| 06:07.38 | lowlevel | and.. I don't use any steenking GUI's |
| 06:07.44 | obnauticus | same here man |
| 06:07.51 | obnauticus | I'm 16 and i prefer CLI only with manual text files. |
| 06:07.58 | obnauticus | (abnormal for my age apperentally( |
| 06:08.05 | lowlevel | just.. I made a choice to use this PCI card from digium with old phones... so I'm stuck with a linux box I'de rather not have around |
| 06:08.36 | lowlevel | I dont want to spend money on ATA's , I'm just going to buy some nice ip phones.. i only really need 3 or 4 |
| 06:08.39 | lowlevel | just putting it off |
| 06:09.11 | [TK]D-Fender | ATA's are fine for most uses I find... |
| 06:09.26 | lowlevel | well... I have a small place here.. and I find them to be clunky and annoying |
| 06:09.52 | lowlevel | plus.. my analog phones are getting old... I'de rather rid the phones now and skip the atas |
| 06:10.09 | ManxPower | Polycoms are where it's at. |
| 06:10.14 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 06:10.15 | lowlevel | yeah, they just look gay |
| 06:10.19 | lowlevel | ;) |
| 06:10.26 | [TK]D-Fender | lowlevel, as a matter of choice, hey why not, but there's little functionality you need that demands a hard-phone. |
| 06:10.32 | ManxPower | not gay, "metrosexual" |
| 06:10.44 | [TK]D-Fender | lowlevel, You get used to them fast actually... |
| 06:10.45 | lowlevel | no, gay. metrosexual.. is more like the cisco ones |
| 06:10.53 | lowlevel | which I dont mind |
| 06:11.48 | lowlevel | but yeah.. I'm just using this to run my home phone system really , nothing commercial.... |
| 06:11.49 | obnauticus | Seriously |
| 06:11.56 | obnauticus | what is so fucking bad about a Cisco Hardphone? |
| 06:12.00 | lowlevel | nothing. |
| 06:12.02 | obnauticus | I've never used one so I don't understand |
| 06:12.30 | ManxPower | obnauticus: almost all the problems with Cisco phones are not techical or design, they are LICENSING |
| 06:12.34 | obnauticus | Everyone bitches about them, they look cool like they LOOK cooler than the polycom ones...which are better but they still look cooler. |
| 06:12.47 | obnauticus | ManxPower, why? |
| 06:12.55 | lowlevel | yeah licensing cisco products has been difficult for me as well |
| 06:12.58 | [TK]D-Fender | obnauticus, inferior call handling. No presence support. Licenced firmware. Higher cost. More like why would you ever consider them over Polycom? |
| 06:13.06 | nestAr | i gotta a polycom ip550 and ip300 here in my basement, they both make calls quite well. |
| 06:13.37 | obnauticus | [TK]D-Fender, the cisco phones do look cooler I'm not gonna lie. |
| 06:13.43 | lowlevel | I'm purely looking at 'style' , and not really internals/design/function |
| 06:13.44 | obnauticus | But the SIP firmware is free for the Cisco phones. |
| 06:13.44 | ManxPower | To LEGALLY run SIP on the phones you have to pay an extra $100 or so, on top of the cost of the phone. If you do not want to run power over ethernet, you need to buy a power supply (also not included in the base price of the phone), an extra $45 |
| 06:13.51 | [TK]D-Fender | IP550's only point if the guy who's too cheap to get the 650 and is desperate for a backlight |
| 06:14.04 | obnauticus | ManxPower I have both of those, what else :/ |
| 06:14.06 | ManxPower | so really, the cost is sometimes as much as twice what a polycom would cost. |
| 06:14.10 | lowlevel | maybe polycom will out some new modles |
| 06:14.15 | [TK]D-Fender | obnauticus, I'm not taking away from your aestetics comment, but I buy a phone for how it WORKS. |
| 06:14.29 | obnauticus | I know, i was just being optimistic :P |
| 06:14.33 | lowlevel | *nod* |
| 06:14.50 | obnauticus | Anyway, I got a polycom ip500 and it's fine... |
| 06:14.52 | ManxPower | If it was not for Cisco's licensing and power supply issues, we would be using Cisco for the 200 or so IP phones we have -- instead we are all Polycom |
| 06:14.58 | [TK]D-Fender | frankly they should have woken up and backlit ALL of their line. |
| 06:15.00 | obnauticus | but I got a Cisco 7960 comming from ebay. |
| 06:15.06 | obnauticus | Ya i know dude. |
| 06:15.15 | obnauticus | there's a mod to put one in though, I'm gonna do that. |
| 06:15.15 | obnauticus | lol |
| 06:15.23 | ManxPower | obnauticus: so when it arrives you'll be here whining about having to buy the firmware. |
| 06:15.30 | [TK]D-Fender | obnauticus, if it were a 501 you'd have a microbrowser... |
| 06:15.35 | ManxPower | You realize the the firmware that comes with the phone is NOT transferrable, right? |
| 06:15.38 | obnauticus | wtf. |
| 06:15.45 | lowlevel | manx: yeah, but its on there ;) |
| 06:15.54 | ManxPower | now, you can pirate it pretty easy, but it's not LEGAL |
| 06:16.28 | ManxPower | i.e. you might not care for your own use, but if your client is a $600mil/year company, they will care. |
| 06:16.32 | lowlevel | (I guess you guys know exactly why I'm still using the digium with old phones ;)) |
| 06:16.32 | obnauticus | Can't you just change it via TFTP? |
| 06:16.45 | ManxPower | obnauticus: you can't download the firmware |
| 06:17.38 | ManxPower | If you have a support contract you are ABLE to download the firmware, but you are not legally licensed to use it just because you have a support contract. |
| 06:18.08 | lowlevel | yeah... so, if its for business, don't buy any used ones. |
| 06:18.17 | obnauticus | That's kinda gay. |
| 06:18.20 | obnauticus | I'll get it working though. |
| 06:18.23 | lowlevel | ob: take it up with cisco |
| 06:18.33 | obnauticus | Well they have their methods of getting money too :P |
| 06:18.44 | obnauticus | http://www.cisco.com/pcgi-bin/tablebuild.pl/sip-ip-phone7960 <-- |
| 06:18.52 | obnauticus | SIP Flash Image for 7940/7960 IP Phone v8.2(0)- Non-CallManager <-- |
| 06:18.59 | lowlevel | I've only got one cisco 'licence' to deal with thankfully |
| 06:18.59 | ManxPower | some of the cool features of using cisco phones with Cisco Call manager are not available on the SIP firmware |
| 06:19.00 | obnauticus | I'm not gonna be using Cisco CM primarily. |
| 06:19.02 | lowlevel | for now... |
| 06:19.03 | lowlevel | :/ |
| 06:19.12 | hellop | Yes, but can you put a picture of boobies on your polycom 501's LCD? |
| 06:19.15 | lowlevel | and... a perpetual support agreement (ass rape) |
| 06:19.17 | obnauticus | LOL |
| 06:19.23 | obnauticus | hellop you can on the Cisco one! |
| 06:19.35 | ManxPower | So you can see why Cisco is not a popular choice around here. |
| 06:19.41 | obnauticus | I'm not gonna lie though, Cisco releases some good stuff. |
| 06:19.47 | [TK]D-Fender | hellop, yup |
| 06:19.50 | obnauticus | Like their networking hardware works the way it's supposed to. |
| 06:20.07 | obnauticus | but much like microsoft they take advantage of their popularity and are dicks about it. |
| 06:20.07 | nestAr | [TK]D-Fender: Maybe, but they were a good price.. I didn't even look at the 650. I don't know what the difference is. |
| 06:20.18 | lowlevel | ob: hmm, yeah.. pretty much |
| 06:20.20 | ManxPower | obnauticus: um, try to ACTUALLY download that firmware |
| 06:20.32 | obnauticus | I have it |
| 06:20.33 | obnauticus | on my desktop. |
| 06:20.41 | ManxPower | you must have a CCO account |
| 06:20.42 | obnauticus | you can login anonomyously |
| 06:20.43 | nestAr | I bought the 550's so certain people could feel like they were more important. ;) |
| 06:20.46 | obnauticus | brb. |
| 06:20.56 | lowlevel | ob; honestly.. I like that I dont' get called about 'the internet is down' ... so I always recommend the ass rape. |
| 06:21.01 | [TK]D-Fender | nestAr, 2 more line keys, support the expansion modules & USB, and I believe come with PS as well. |
| 06:21.17 | nestAr | the 550's came with a PS |
| 06:21.25 | nestAr | didn't use it, have POE switch |
| 06:21.37 | [TK]D-Fender | nestAr, ok, nix that then, the rest stands. 650 has a future, 550 is a dead end. |
| 06:21.48 | ManxPower | lowlevel: my clients mostly use all Cisco routers (2621/2621XM) and all Cisco switches (Catalyst 550x) |
| 06:21.55 | [TK]D-Fender | and a pricey one at that. |
| 06:22.03 | nestAr | not sure that i need the 2 extra line keys or a expansion module. |
| 06:22.10 | ManxPower | [TK]D-Fender: is the USB useful for anythin yet? |
| 06:22.15 | [TK]D-Fender | nestAr, its about protecting your investment. |
| 06:22.20 | lowlevel | manx; thats much bigger than what I deal with ;) (ASA 55xx's and such lately) |
| 06:22.27 | [TK]D-Fender | ManxPower, not much that I've seen, but it opens doors. |
| 06:23.09 | lowlevel | but.. previously the odd PIX |
| 06:23.09 | ManxPower | lowlevel: the catalysts are so cheap on ebay, we can buy hot spares and still come out cheaper |
| 06:23.09 | lowlevel | or whatever |
| 06:23.09 | nestAr | :shrug: i have so little invested in the phones it doesn't really matter. they make and take calls, that's what's important to me. |
| 06:23.09 | lowlevel | manx: yeah, they're solid |
| 06:23.12 | [TK]D-Fender | nestAr, Suppose there's that... |
| 06:23.19 | ManxPower | [TK]D-Fender: I suggest IP 501 for the operator, they can always redeploy that to "an important person" if they need BLF, etc |
| 06:23.29 | obnauticus | ManxPower you don't need a CCO accoun.t |
| 06:23.30 | obnauticus | account* |
| 06:23.38 | [TK]D-Fender | ManxPower, 501 for presence? ew. |
| 06:23.38 | nestAr | i had 501's at my previous company, i figured the 550 was a s |
| 06:23.40 | obnauticus | Read the Note: section: http://www.cisco.com/pcgi-bin/Software/Tablebuild/doftp.pl |
| 06:23.44 | [TK]D-Fender | ManxPower, no lit indicators |
| 06:23.48 | ManxPower | obnauticus: so what did you provide as the userid, password? |
| 06:23.48 | nestAr | set up from the 501 |
| 06:23.56 | obnauticus | ManxPower, read the Note section. |
| 06:24.12 | [TK]D-Fender | nestAr, it is, but it costs the same as a 601 without the expansion capabilities. |
| 06:24.14 | lowlevel | hmm, wonder if I got my 30amp outlets while I was off |
| 06:24.17 | obnauticus | It's Anonymous:your@email.com |
| 06:24.18 | nestAr | ;v r4 |
| 06:24.19 | nestAr | ' |
| 06:24.22 | ManxPower | obnauticus: my session is no longer valid |
| 06:24.24 | obnauticus | The SIP Firmware is a free release. |
| 06:24.30 | nestAr | sorry, weinerdog at the keyboard |
| 06:24.59 | lowlevel | ah well, work in the morning :/ night guys |
| 06:25.05 | obnauticus | Now... Asterisk supports SCCP, correct? |
| 06:25.07 | obnauticus | skinny.conf? |
| 06:25.12 | nestAr | what would i use the expansion moduel for? i honestly don't know. |
| 06:25.16 | [TK]D-Fender | I'm outta here too... |
| 06:25.21 | [TK]D-Fender | g'night all |
| 06:25.24 | nestAr | later |
| 06:25.32 | *** part/#asterisk atomicd (n=atomicd@adsl-69-109-58-155.dsl.irvnca.pacbell.net) |
| 06:25.46 | [TK]D-Fender | nestAr, BLF / system functions like page, parking, etc |
| 06:25.56 | nestAr | true |
| 06:25.57 | obnauticus | ManxPower: there's this too refer pm |
| 06:26.09 | *** join/#asterisk h3x (i=Justino@64.192.116.17) |
| 06:30.12 | hellop | Polycom Question: You know the Soundpoint L phones? They look nearly identical to the 501s except RJ11 not Ethernet. Can someone point me on a search to figure out how to make the SoundPoint L buttons work? Inet searches yeild little. |
| 06:30.55 | hellop | Just that the SoundPoint L is an Integrated analog feature telephone designed for systems like TeleVantage CTM and the Intel is touting the Intel Converged Communications Platform (ICCP) . |
| 06:31.11 | hellop | IOW best use = doorstop? |
| 06:31.23 | linagee | how do i make an iax uri call to another friend of mine? |
| 06:31.54 | linagee | i add a custom extension with this in the dial line: iax://1004@ip.ip.17.98/1004 |
| 06:32.11 | linagee | it says all lines are busy/congested in the CLI debugging. :( |
| 06:32.16 | *** join/#asterisk IgI (n=FeedomPa@195.162.32.126) |
| 06:32.41 | ManxPower | linagee: Asterisk does not support IAX IRLs |
| 06:32.45 | ManxPower | or URLs |
| 06:32.48 | linagee | ManxPower: hrm?? |
| 06:32.54 | linagee | ManxPower: only SIP URIs? |
| 06:33.08 | ManxPower | It doesn't support SIP URIs either. |
| 06:33.25 | linagee | ManxPower: Dial("sipuri") something like that? |
| 06:33.38 | ManxPower | "SIP/1235@thehost" and "IAX/thehost/2345" are not URIs |
| 06:33.53 | linagee | ManxPower: hrm! so that's the correct format! :) |
| 06:33.56 | linagee | ManxPower: thanks. :) |
| 06:33.59 | *** join/#asterisk di||itante (n=michael@pool-70-105-171-72.nwrknj.fios.verizon.net) |
| 06:34.10 | Shaun2222 | with the iaxy s101 units how do you transfer calls |
| 06:34.19 | Shaun2222 | if there a specific set of keys i need to press. |
| 06:34.23 | ManxPower | linagee: you would have known that if you had read even one page of documentation |
| 06:34.29 | ManxPower | Shaun2222: FLASH |
| 06:34.48 | linagee | ManxPower: lol. ;) |
| 06:34.56 | ManxPower | I don't recall if it's FLASH NUMBER HANGUP or FLASH NUMBER FLASH HANGUP |
| 06:35.24 | hellop | I guess analog feature phones where just dropped but the industry. |
| 06:35.34 | hellop | but=by |
| 06:35.56 | Shaun2222 | ManxPower: weird but works thanks |
| 06:36.18 | Shaun2222 | i did flash number flash |
| 06:36.21 | Shaun2222 | hangup |
| 06:37.19 | *** join/#asterisk lemanal (n=lemanal@214.sub-75-209-130.myvzw.com) |
| 06:37.25 | linagee | ManxPower: i dialed SIP and his side rang. i dialed IAX and i got the busy/congested message. :-/ |
| 06:37.53 | ManxPower | Shaun2222: not weird at all. That has been the standard way for analog centrex for 20 years. |
| 06:38.03 | ManxPower | linagee: I'm sorry, but you just don't know enough. |
| 06:38.08 | linagee | ManxPower: aw |
| 06:38.17 | linagee | for the lose |
| 06:43.06 | *** join/#asterisk LoF^[Lawbringer] (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 06:46.38 | linagee | ManxPower: is there a way to tell asterisk to always use an external IP instead of the IP on the interface? (i'd say externip if it was a trunk) |
| 06:46.45 | linagee | hmm |
| 06:47.37 | J4k3 | I believe you can attach the asterisk daemon itself to an IP |
| 06:47.39 | linagee | (i looked at the log and i am getting his internal IP. he's got a 1:1 NAT set up) |
| 06:48.15 | linagee | his asterisk box is sending out what it thinks is it's IP, but it's really an internal IP. hrm |
| 06:50.44 | *** join/#asterisk UnFred (n=UnFred@S010600095b44774f.vs.shawcable.net) |
| 06:50.48 | hellop | linagee, sounds like an easy .conf fix |
| 06:51.52 | linagee | hellop: true. reading about it now |
| 06:55.47 | *** join/#asterisk jeebusmobile (n=jeebusmo@cpe-72-132-155-43.dc.res.rr.com) |
| 07:04.47 | *** join/#asterisk marc7 (n=marc@S010600131024913b.vc.shawcable.net) |
| 07:04.50 | *** join/#asterisk IgorG (n=FeedomPa@195.162.32.126) |
| 07:05.28 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 07:06.02 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 07:07.05 | hellop | linagee, tell him to plug directly into the DSL modem and assign his PC the external IP. |
| 07:07.55 | linagee | hellop: lol. i think he's got a cable modem and he wants to route it through a m0n0wall box or something dumb like that |
| 07:09.02 | BBHoss | m0n0wall is certainly not dumb |
| 07:09.13 | obnauticus | M0n0wall is ok |
| 07:09.17 | obnauticus | i like pfsense better. |
| 07:09.24 | BBHoss | this is true |
| 07:09.28 | BBHoss | what is he trying to do |
| 07:09.34 | ManxPower | A properly managed system doesn't need a firewall. |
| 07:09.40 | BBHoss | bullshit |
| 07:09.43 | linagee | BBHoss: right but trying to do a 1:1 NAT and such. ugh. :( |
| 07:10.04 | BBHoss | you can get by with forwarding ports |
| 07:10.13 | [hC] | a properly managed system can still need a firewall if you need services exposed to some people and not others, publically |
| 07:10.28 | ManxPower | BBHoss: OK then. How does a firewall protect ports that have nothing running on them and how does a firewall protect applications that ARE running on a port? |
| 07:10.31 | linagee | [hC]: do the firewall using iptables in the asterisk box itself. ;) |
| 07:10.40 | ManxPower | [hC]: that I can agree with. |
| 07:10.42 | [hC] | linagee: that is fine. |
| 07:10.48 | [hC] | i have nothing wrong with local iptables rules |
| 07:11.16 | ManxPower | However, I do thing that is better handled at the application, rather than the network |
| 07:11.21 | [hC] | but if you are depending on a firewall to allow access to services, but protect you from exploits, say goodnight. |
| 07:11.28 | [hC] | that too. |
| 07:11.28 | BBHoss | yes |
| 07:12.04 | [hC] | for example i run ssh on my servers but only allow connections from certain hosts, likewise with ftp and sometimes http, incase a new vulnerability is discovered and i am hit unknowingly |
| 07:12.05 | ManxPower | now, on a system that is not properly managed (like my Windows laptop, which I am NOT qualified to properly manage) a firewall can be pretty handy. |
| 07:12.11 | BBHoss | firewalls are needed because of the human aspect |
| 07:12.18 | BBHoss | human error |
| 07:12.23 | *** join/#asterisk harpal (n=Harpal@124.125.255.223) |
| 07:12.39 | [hC] | firewalls act as a buffer between humans and keeping systems patched, configured and up to date |
| 07:12.52 | BBHoss | they are required |
| 07:12.52 | [hC] | also, when a machine does get rooted somehow, a firewall prevents it from becoming much worse. |
| 07:12.55 | ManxPower | BBHoss: like accidentally running an FTP server? |
| 07:12.57 | [hC] | or.. can. |
| 07:13.01 | BBHoss | no |
| 07:13.25 | BBHoss | like forgetting to specify eth1 instead of the default which is all interfaces |
| 07:13.25 | ManxPower | BBHoss: Oh, so like protecting your SMTP server from being exploited? |
| 07:13.36 | *** join/#asterisk J4zen (n=Jvan4zen@a213-84-139-244.adsl.xs4all.nl) |
| 07:13.45 | ManxPower | Sorry, I sometimes forget people run more than one interface. |
| 07:13.57 | BBHoss | thats a limited example |
| 07:14.08 | [hC] | thats also a configuration problem |
| 07:14.09 | ManxPower | and that they use the interfaces as a sort of security method |
| 07:14.17 | [hC] | all manx said was that you dont NEED a firewall if the system has been configured and maintained 'properly' |
| 07:14.30 | BBHoss | im not saying thats false |
| 07:14.43 | BBHoss | but properly means perfectly in my book |
| 07:14.53 | BBHoss | humans!=perfect |
| 07:15.00 | ManxPower | BBHoss: bullshit <-- I assume that was for linagee |
| 07:15.06 | [hC] | which is why a lot of people have firewalls :) |
| 07:16.01 | ManxPower | I have a firewall on my windows box, but not on the linux boxes I "manage" |
| 07:16.01 | BBHoss | i am sorry, i didnt think that through |
| 07:16.33 | [hC] | in my opinion a firewall acts as a safety net for configuration error, and can help lock down services from trusted hosts so you dont run into accidental breach by means of "0 day exploits" of services, and such. |
| 07:16.42 | BBHoss | firewalls also help protect you if you get hacked from a zero-day exploit |
| 07:16.43 | hellop | [hC], well some firewalls offer nifty stuff like application monitoring("Notepad is trying to access the internet"), and detection/logging/notification of DoS, probing, and other attacks. Without one, you won't know what the problem is during a DoS. |
| 07:16.53 | ManxPower | [hC]: *nod* |
| 07:17.28 | ManxPower | BBHoss: How exactly does a firewall help protect you if you get hacked? |
| 07:17.48 | BBHoss | depending on configuration, it can help protect information going out |
| 07:17.53 | ManxPower | One might assume "hacked" means "root level access" and anyone with root level access can turn off the firewall |
| 07:18.18 | BBHoss | i am speaking in reference to one that does nat |
| 07:18.32 | [hC] | well, not if the firewall is a separate device of course, so yes if you get hacked a standalone firewall can help prevent the opening of a new port that you didnt want open, or data leaving that you didnt want going out |
| 07:18.38 | [hC] | but you're pretty much already screwed anyways |
| 07:18.40 | BBHoss | and external device thats not installed on the computer |
| 07:18.41 | ManxPower | BBHoss: firewall and nat are not the same thing. |
| 07:18.43 | *** join/#asterisk nibbler_de (n=nibbler@as250.net) |
| 07:18.46 | BBHoss | no shit |
| 07:18.50 | nibbler_de | re |
| 07:18.54 | nibbler_de | i have configured a callerid in sip.conf - is there any variable which i get the "real" callerid of the user from? |
| 07:18.57 | linagee | ManxPower: we just got it working. he had to do some localnet option |
| 07:18.58 | BBHoss | they are commonly packged |
| 07:19.11 | [hC] | BBHoss: more than you'd realize, they are not commonly packaged. |
| 07:19.13 | ManxPower | [hC]: oh, I think there are many, many, many reasons to have a standalone firewall. I was only referring to a host firewall |
| 07:19.22 | [hC] | BBHoss: unless you're strictly focusing on soho router firewalls. |
| 07:20.06 | ManxPower | I also doubt you are going to be running NAT on a host that is not a router. |
| 07:20.08 | BBHoss | if you were referring to host firewalls, then thats fine, i agree, but standalone firewalls are necessary |
| 07:20.28 | [hC] | i think you're wrong by saying necessary, again. |
| 07:20.45 | ManxPower | When I said "system" I was referring to "host", sorry I was not more clear. |
| 07:20.52 | BBHoss | fine by me |
| 07:20.55 | ManxPower | not router. |
| 07:22.13 | ManxPower | and not a network |
| 07:22.16 | *** join/#asterisk saftsack (n=saftsack@pD9E05A8E.dip.t-dialin.net) |
| 07:23.01 | linagee | a netnotwork |
| 07:23.13 | BBHoss | heh, sure |
| 07:24.53 | *** join/#asterisk IgI (n=FeedomPa@195.162.32.126) |
| 07:25.05 | *** join/#asterisk mosty (n=mostyn@ppp191-34.static.internode.on.net) |
| 07:25.17 | hellop | nibbler_de, you mean like exten => 4,2,SayDigits(${CALLERID(num)}) |
| 07:25.21 | nibbler_de | yes |
| 07:25.26 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 07:25.40 | nibbler_de | but calerid(num) gives me the configured callerid - not the real one |
| 07:26.01 | ManxPower | no, like the variables listed in /path/to/src/asterisk/doc/README.variables (or channelvariables.txt if you are on 1.4) |
| 07:26.17 | ManxPower | nibbler_de: if you want the real one, then don't configure callerid. |
| 07:26.32 | ManxPower | "the real one" is whatever the device sends as it's callerid |
| 07:26.39 | nibbler_de | ManxPower: i have already looked into the variables documentation - without success |
| 07:26.44 | nibbler_de | i have this config: |
| 07:26.53 | nibbler_de | [2342] |
| 07:26.53 | nibbler_de | callerid=9112753355 |
| 07:26.57 | hellop | nibbler_de, Manx seems to have explained it for you. |
| 07:26.58 | nibbler_de | username=2342 |
| 07:27.39 | nibbler_de | in my sip. conf - so you are trying to tell me that i can not get the information "2342" during the call? and all i'll ever get will be the callerid 9112753355? |
| 07:28.02 | nibbler_de | that would make things rather complicated for me *sigh* |
| 07:28.10 | ManxPower | why not setvar=CIDOVERRIDE=9112753355 then CALLERID(num) has the callerid it is sending, and you have the callerid yo would set. |
| 07:28.38 | ManxPower | nibbler_de: Um, 2342 is not the "real callerid", it is the SIP userid |
| 07:29.03 | ManxPower | and you should be able to get lots of info about that. |
| 07:29.03 | nibbler_de | ManxPower: hmm, i can define per-user setvar= entries in sip.conf? |
| 07:29.14 | ManxPower | nibbler_de: yes. |
| 07:29.36 | nibbler_de | that would be a workaround i can live with - but - if i can get the sip userid i'd be much happier |
| 07:30.22 | ManxPower | when you change your queestion from "how do I get the real callerid" to "how can I get the sip userid" both the question AND the answer change |
| 07:31.01 | nibbler_de | well - i regard the sip userid as the "real callerid" but - yeah - what i meant was in deed the sip userid |
| 07:31.31 | ManxPower | SIPURI might have the info, you would have to parse it out. CHANNEL might also have that info and you would have to parse that out. |
| 07:31.36 | *** join/#asterisk Putzz (n=me@CPE001a707d4d4e-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 07:31.54 | *** join/#asterisk BeeBuu (n=chatzill@125.95.248.142) |
| 07:32.04 | BeeBuu | help all |
| 07:32.14 | hellop | BeeBuu, yes please do. |
| 07:32.34 | J4zen | Hi there |
| 07:32.40 | BeeBuu | i men i need help please. |
| 07:32.51 | J4zen | i'm having some odd issues with my PBX, allow me to explain the situation |
| 07:32.58 | ManxPower | ~ask |
| 07:32.58 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 07:33.13 | BeeBuu | is there any idea to check which channel is avaliable? |
| 07:33.46 | ManxPower | BeeBuu: "show application chanisavail" "show applications" is also your friend. love it, hold it, buy it chocolates |
| 07:34.02 | J4zen | Recently we moved our (test)-PBX to our Datacenter, the PBX worked just fine in our local network. After moving it to the datacenter i am unable to register my SNOM320's at the PBX, i AM however able to register my X-lite softphones on it. |
| 07:34.10 | J4zen | I have generated two log files |
| 07:34.18 | J4zen | one from the SIP debug on the office IP |
| 07:34.21 | J4zen | http://www.pastebin.ca/771795 |
| 07:34.25 | J4zen | and one from the SNOM320 on level 9 |
| 07:34.30 | ManxPower | J4zen: make sure you do NOT have a bindaddr setting in Asterisk |
| 07:34.33 | J4zen | http://www.pastebin.ca/771797 |
| 07:34.48 | nibbler_de | ManxPower: great, thanks - works like a charm - i just have to cut the sip: and the @ip away - finally it made sense that i use strictly four-digit numbers everywhere ;-) |
| 07:34.51 | J4zen | There shouldn't be, the PBX is not behind a NAT |
| 07:35.00 | J4zen | the SNOM's are however behind a NAT |
| 07:35.04 | BeeBuu | ManxPower: which application for me? |
| 07:35.06 | J4zen | and have the qualify=yes and nat=yes switch |
| 07:35.09 | J4zen | in their sip.conf |
| 07:35.09 | BBHoss | has the IP address changed? |
| 07:35.30 | ManxPower | J4zen: make SURE there are NO nat settings enabled on the SNOM |
| 07:35.56 | J4zen | Well not on the SNOM itself |
| 07:36.02 | ManxPower | asterisk nat=yes precludes using the phone's nat support. Also if you are using STUN, turn it off. |
| 07:36.06 | J4zen | but in the sip.conf, their extension |
| 07:36.18 | J4zen | Am not using STUN |
| 07:36.28 | *** join/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 07:36.36 | BeeBuu | ManxPower: which application for check channel? |
| 07:37.03 | J4zen | So the sip.conf extensions for the SNOM320's shouldn't have the nat=yes and qualify=yes switches enabled? |
| 07:37.05 | ManxPower | ManxPower: BeeBuu: "show application chanisavail" "show applications" is also your friend. love it, hold it, buy it chocolates |
| 07:37.18 | J4zen | lol @ ManxPower |
| 07:37.28 | BeeBuu | ManxPower: thanks |
| 07:37.34 | ManxPower | J4zen: yes, they should have nat=yes, qualify=yes, but the SNOMs them selves have NAT options as well. |
| 07:37.47 | J4zen | i see, i must have overlooked that |
| 07:38.12 | J4zen | The NAT Identity Settings ? |
| 07:38.23 | ManxPower | J4zen: I've not used SNOMs so I can't say for sure, what I can say is that every brand of IP phone I've used has options for NAT stuff |
| 07:38.44 | BBHoss | just make sure offer ICE is off and the rest of the fields are blank |
| 07:38.49 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 07:39.13 | BeeBuu | ManxPower: can i use this: chanlsavail(zap/g0)? |
| 07:39.28 | mosty | BeeBuu, try it |
| 07:40.10 | *** join/#asterisk lemanal_ (n=lemanal@71.9.108.98) |
| 07:40.16 | J4zen | The settings were as you described BBHoss, Offer ICE is off and all fields are blank(which they were by default)( |
| 07:40.23 | obnauticus | hey my asterisk crashed and now it's being all weird. |
| 07:40.46 | BBHoss | j4zen: have the phones always been on a separate nat-enabled network from the server |
| 07:40.56 | *** join/#asterisk XTR-III (n=94752345@216.19.191.191.novuscom.net) |
| 07:40.57 | obnauticus | like when i type reload it only reprases some configuration files. |
| 07:41.03 | *** join/#asterisk callguy_ (n=callguy@pool-71-255-162-167.bstnma.east.verizon.net) |
| 07:41.29 | ManxPower | What do you expect it to do? Sit up and recite a poem? |
| 07:41.38 | obnauticus | work? |
| 07:41.53 | J4zen | BBHoss; No |
| 07:41.57 | ManxPower | reload tells Asterisk to reparse it's config files. |
| 07:42.01 | phix | hey, is there any way to use pam or a database or LDAP etc.. for authentication of sip, mailboxes, etc.? |
| 07:42.03 | mosty | i'm trying to auto provision a polycom ip 550, do they support http for settings files, or does it have to be tftp or ftp? |
| 07:42.08 | J4zen | BBHoss; They used to be in our office LAN, all within the same NAT-enabled network |
| 07:42.09 | phix | I don't like putting plaintext passwords in files |
| 07:42.09 | obnauticus | ManxPower it's not reparsing all of them |
| 07:42.12 | obnauticus | or the ones it was before. |
| 07:42.20 | J4zen | And they worked just fine |
| 07:42.21 | BBHoss | then you moved outside NAT? |
| 07:42.23 | mosty | phix, lookup asterisk realtime, but there are downsides. see the wiki |
| 07:42.24 | J4zen | Yes |
| 07:42.27 | BBHoss | heh |
| 07:42.30 | J4zen | well the PBX moved outside the NAT |
| 07:42.31 | phix | mosty: thank you :) |
| 07:42.32 | BBHoss | welcome to hell |
| 07:42.33 | J4zen | the SNOM's stayed |
| 07:42.36 | J4zen | lol |
| 07:42.42 | *** join/#asterisk lemanal (n=lemanal@214.sub-75-209-130.myvzw.com) |
| 07:42.43 | J4zen | I have the vague impression |
| 07:42.47 | BeeBuu | :-O |
| 07:42.49 | J4zen | that SNOM's have some sort of cache i cannot seem to clear |
| 07:42.52 | *** join/#asterisk xheliox (n=jeff@193.251.121.70.cfl.res.rr.com) |
| 07:43.03 | ManxPower | mosty: I beleive that ALL polycoms except for the original 300, 500, and 600 support http and https provisioning |
| 07:43.41 | kiscokid | obnauticus: what else is not working now? |
| 07:43.48 | obnauticus | well i can't register on it |
| 07:43.55 | obnauticus | it doesn't have the commands im used to |
| 07:43.59 | obnauticus | liek iax and sip... |
| 07:44.08 | obnauticus | Asterisk SVN-branch-1.4-r71230 built by root @ asterisk on a i686 running Linux on 2007-06-23 00:39:02 UTC |
| 07:44.12 | mosty | ManxPower, i set the server-name option to something like "http://my-host-here", but i do a packet log and the phone dhcp's etc, but never tries to connect to the web server |
| 07:44.21 | BBHoss | ~nat |
| 07:44.22 | jbot | hmm... nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 07:44.37 | kiscokid | sounds like those modules are not getting loaded |
| 07:44.38 | BBHoss | try that j4zen |
| 07:44.50 | obnauticus | now it keeps repeating |
| 07:44.53 | J4zen | Well, the PBX itself is not behind a nat |
| 07:44.54 | obnauticus | Remote Unix connection |
| 07:44.56 | J4zen | the setup is as follows |
| 07:44.58 | obnauticus | remote UNIX connection disconnected |
| 07:45.01 | BBHoss | oh yeah |
| 07:45.10 | BBHoss | this is gonna be great |
| 07:45.13 | ManxPower | obnauticus: um, we don't really support SVN o this channel. |
| 07:45.18 | obnauticus | :/ |
| 07:45.21 | J4zen | SNOM > ROUTER/GW > INTERNET > PBX ( directly on an outside IP , at datacenter ) |
| 07:45.22 | mosty | snom auto provisioning is so much easier than polycom :( |
| 07:45.25 | BBHoss | you'll need a true proxy |
| 07:45.26 | obnauticus | well I don't know what to do it was working before :/ |
| 07:45.32 | J4zen | BBHoss; Excuse me? |
| 07:45.39 | ManxPower | obnauticus: "stop now" service asterisk start |
| 07:45.40 | obnauticus | before it crashed |
| 07:45.42 | BBHoss | like OpenSER |
| 07:45.44 | linagee | J4k3: define PBX |
| 07:45.46 | kiscokid | obnauticus: remote unix connection means someone did asterisk -r |
| 07:45.47 | BBHoss | or some SBC |
| 07:45.52 | BBHoss | session border controller |
| 07:45.53 | J4zen | linagee; Asterisk. |
| 07:46.01 | phix | mosty: I mostly want a secure storage of passwords, like a hash with salt for example, instead of plaintext |
| 07:46.12 | J4zen | Hmm, you |
| 07:46.15 | J4zen | are creeping me out |
| 07:46.16 | mosty | phix: you can do that in a sip.conf file |
| 07:46.21 | ManxPower | phix: you mean like md5 passwords? |
| 07:46.32 | BBHoss | welcome to the world of sip!! |
| 07:46.46 | J4zen | BBHoss; Why would i need that in the first place? Asterisk cannot connect thru my NAT ? |
| 07:46.48 | phix | ManxPower: like keeping the md5 hash in the config file instead of the plaintext version would also be good |
| 07:47.01 | phix | a hash + salt would be better |
| 07:47.06 | ManxPower | J4zen: you don't need any proxies or other crap like that. |
| 07:47.07 | BBHoss | obnauticus: kill asterisk, then start up with asterisk -cvvvvvvvvvvvvvvvvvvv |
| 07:47.14 | ManxPower | phix: Asterisk can do that. |
| 07:47.15 | linagee | phix: hash + salt + ketchup is EVEN better. :) |
| 07:47.17 | ManxPower | ~book |
| 07:47.58 | 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 |
| 07:47.59 | phix | linagee: yay |
| 07:47.59 | linagee | ketchup goes great with has |
| 07:47.59 | linagee | h |
| 07:47.59 | phix | ManxPower: where? |
| 07:47.59 | phix | ManxPower: oh in the book, what chapter? I have that book :) |
| 07:47.59 | ManxPower | phix: sip.conf.sample should provide several examples |
| 07:47.59 | kiscokid | ~thebook |
| 07:48.09 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 07:48.09 | obnauticus | what should I be looking for BBHoss? |
| 07:48.10 | ManxPower | that file is included in the asterisk source code, btw |
| 07:48.10 | BBHoss | pastebin it |
| 07:48.10 | obnauticus | oh ok |
| 07:48.21 | J4zen | ManxPower: What do i need , if not a proxy? |
| 07:48.27 | phix | ManxPower: oh |
| 07:48.36 | *** join/#asterisk MicW (n=michael@dslb-088-074-130-220.pools.arcor-ip.net) |
| 07:48.38 | MicW | hi |
| 07:48.45 | ManxPower | J4zen: to figure out what config option where is wrong. |
| 07:49.04 | phix | hmmmm, I should use the [authentication] sip.conf feature? |
| 07:49.05 | ManxPower | make sure your hosting prover is not providing a helpful firewall. |
| 07:49.05 | J4zen | I was afraid you'd say that |
| 07:49.17 | J4zen | No all ports are opened, no firewall |
| 07:49.38 | phix | username#md5hash@hostname interesting, does that work with register too? |
| 07:49.52 | phix | using # instead of : to seperate username and password / hash ? |
| 07:50.17 | MicW | i'm running asterisk with external sip providers. that works fine until my ip changes. then the sip connnections get lost and I see a lot of "Registration for '...' timed out" |
| 07:50.39 | MicW | when i reload asterisk, they register again ind it works until the next ip change |
| 07:51.05 | obnauticus | BBHoss: http://pastebin.ca/771811 |
| 07:51.10 | phix | ? |
| 07:51.15 | obnauticus | Sorry for the colors. |
| 07:51.18 | mosty | micw: reload sip when your ip changes |
| 07:51.22 | obnauticus | you might want to cat that into a shell that supports colors |
| 07:51.24 | obnauticus | or soemthing |
| 07:51.31 | mosty | micw: or reload periodically, or just get a static ip |
| 07:52.14 | MicW | is it possible to change the configurtaion so that sip re-registers if the registration gets lost? |
| 07:52.26 | BBHoss | apparently, non of your modules have descriptions, so they can't be loaded |
| 07:52.32 | obnauticus | wtf? |
| 07:52.34 | obnauticus | It was JUSt working lol. |
| 07:53.05 | mosty | MicW, the sip client doesn't know that the registration was lost, that's the problem |
| 07:53.30 | obnauticus | BBHoss what should I do? |
| 07:53.54 | BBHoss | not sure, could be a simple fix, or it might have corrupted everything |
| 07:53.58 | BBHoss | i would reinstall |
| 07:54.06 | phix | ManxPower: I can't find my salt! |
| 07:54.10 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 07:54.16 | obnauticus | fuuuuuuck. |
| 07:55.10 | J4zen | ManxPower: I dont see any misconfiguration in the SIP.conf file: http://www.pastebin.ca/771815 |
| 07:55.19 | J4zen | Could you take a look please? |
| 07:57.45 | mosty | can anyone tell me what setting in need in dhcpd.conf i need to tell a polycom phone to get it's settings from a web server? nothing i've found on the web seems to work |
| 07:58.14 | BBHoss | option 66 |
| 07:58.39 | BBHoss | oh web server |
| 07:58.50 | obnauticus | l |
| 07:58.52 | obnauticus | k BBHoss |
| 07:58.59 | obnauticus | I just reinstsalled real fast |
| 07:59.31 | mosty | bbhoss: do you know the name for option 66 in dhcpd v3? |
| 08:00.05 | *** join/#asterisk Bilbolodz (n=bilbo@pc-bilbo.man.lodz.pl) |
| 08:00.11 | *** part/#asterisk Bilbolodz (n=bilbo@pc-bilbo.man.lodz.pl) |
| 08:02.04 | kiscokid | mosty: use: option boot-server code 66 = string; |
| 08:02.52 | mosty | kiscokid, /etc/dhcp3/dhcpd.conf line 11: unknown option dhcp.boot-server |
| 08:04.33 | obnauticus | Internal RTCP NTP clock skew detected |
| 08:04.38 | obnauticus | how do i fi9x that? |
| 08:05.44 | kiscokid | mosty: is line 11 option boot-server code 66 = string; ? |
| 08:06.04 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 08:06.18 | mosty | kiscokid, tftp-server-name seems to work now, strangely enough even if i put http://something |
| 08:07.07 | bintut | hello all.. anyone here have personally deployed fax over ip service on the asterisk? |
| 08:07.36 | mosty | bintut, no, give up now is my advice |
| 08:07.38 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 08:07.58 | obnauticus | Does anyone here know what the error `Device does not match ACL' means while doing a handle_request_register |
| 08:08.46 | bintut | mosty: why? is it because of a buggy software or something? i need to have a foip functionality on my box.. |
| 08:09.27 | mosty | bintut, asterisk only has pass-thru t.38 support |
| 08:10.39 | bintut | mosty: yeah, i read that from one of the pages i found on the internet.. but i am looking for a practical/actual deployment if there is.. |
| 08:11.18 | mosty | bintut, well you probably need something that supports t.38 well, and asterisk doesn't (yet) |
| 08:11.24 | BBHoss | based on the situation you described before, * can do it |
| 08:11.26 | bintut | actually, i found from the articles that asterisk+iaxmodem+hylafax will do the trick but i'm looking for your advice here in this community |
| 08:11.42 | BBHoss | the only problem i see is that you want to run it over an IAX trunk |
| 08:11.54 | bintut | BBHoss: what combinations you have in there without using a hardware ata? |
| 08:12.02 | BBHoss | t.38 is sip<-sip-> only |
| 08:12.04 | mosty | bintut, fax over voip will never work well |
| 08:12.11 | BBHoss | t.38 can be great |
| 08:12.36 | bintut | BBHoss: regardless if iax or sip.. i want to have a reliable foip using t.38 |
| 08:12.47 | *** join/#asterisk J4zen|Ghost (n=Jvan4zen@a213-84-139-244.adsl.xs4all.nl) |
| 08:12.49 | BBHoss | you have to have a sip ata that supports |
| 08:12.52 | BBHoss | t.38 |
| 08:13.05 | BBHoss | and pass it over sip, to another sip ata |
| 08:13.28 | *** part/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 08:13.30 | BBHoss | now if you want to dial someone using POTS and use t.38, then you'll need a provider that supports t.38 |
| 08:13.30 | bintut | BBHoss: other than using an ata hardware, any other solution? |
| 08:13.39 | BBHoss | you can TRY callweaver |
| 08:13.51 | BBHoss | they seem to have origination/termination |
| 08:14.05 | BBHoss | or you could use hylafax, but ive never tried it |
| 08:14.26 | bintut | BBHoss: yeah, i read from another article that callweaver has a full support for pass through and termination but their site is down |
| 08:15.34 | bintut | BBHoss: i actually have a digium card dev kit here and i believe i can fax to pots.. my problem is faxing through ip.. |
| 08:16.10 | *** join/#asterisk _ys (n=yuri@236-069.nat.mns.ru) |
| 08:16.16 | mosty | bintut, are you trying to get asterisk to send outgoing fax via sip or iax? |
| 08:16.19 | BBHoss | that is everyones problem |
| 08:16.51 | bintut | mosty: regardless of protocol, doesn't matter to me.. the important thing is i can send/receive fax over ip |
| 08:17.10 | mosty | bintut, you cannot do it reliably with asterisk as an endpoint |
| 08:17.22 | mosty | unless you only use PSTN channels |
| 08:18.08 | BBHoss | * cant act as an endpoint, it can only pass t.38 through to an ATA |
| 08:18.09 | bintut | mosty: actually, i am peered with an asterisk box also but to cut off long distance fax bills, i'm looking for ways that both sides can send/receives fax over ip |
| 08:18.36 | mosty | bintut, you can use a web interface to hylafax |
| 08:20.33 | *** join/#asterisk cypherdelic (n=cypher@p5B27C8C4.dip.t-dialin.net) |
| 08:23.08 | *** join/#asterisk modu (n=modu@rue92-6-82-237-172-115.fbx.proxad.net) |
| 08:23.12 | modu | hello |
| 08:24.56 | *** join/#asterisk nexilus (n=nexilus@gate.compodium.se) |
| 08:25.00 | obnauticus | BBHoss |
| 08:25.03 | obnauticus | how do i fix this: Internal RTCP NTP clock skew detected |
| 08:25.30 | nexilus | how would i go about if i want any dialed extension to have the same ",1," functionality? to be frank, i want all calls to go through an AGI script first, and then use the apropriate dialplans |
| 08:25.58 | BBHoss | thats probably just debug info |
| 08:26.05 | BBHoss | try installing from a tarball |
| 08:26.18 | obnauticus | Your name is traball. |
| 08:26.19 | obnauticus | lol |
| 08:26.48 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 08:27.44 | modu | When I pickup a call I want my phone to show the real caller (not *9+exten) is there a way to do this ? |
| 08:28.16 | obnauticus | ~cisco |
| 08:28.17 | jbot | cisco makes the routers that move the pr0n across the internet at a very high rate of exchange. a company full of patent protected punks! |
| 08:28.44 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-f6ae8512ed792744) |
| 08:32.43 | *** join/#asterisk maagic (i=maagic@fsck.fi) |
| 08:33.37 | *** join/#asterisk parag00n (n=parag0n@87-194-9-117.bethere.co.uk) |
| 08:36.31 | modu | Someone use pickup here ? |
| 08:41.32 | bintut | mosty: sorry, i left for a while.. |
| 08:41.41 | bintut | BBHoss: sorry, i left for a while.. |
| 08:42.11 | mosty | bintut, you can use a web interface to hylafax |
| 08:42.17 | mosty | is the last thing i said |
| 08:43.55 | nexilus | if i use exten s,1,...... does that affect all calls ? |
| 08:45.44 | *** join/#asterisk speekac (n=alwin@60.51.217.61) |
| 08:46.03 | mosty | only calls sent to extension s, priority 1 in that context |
| 08:46.03 | *** join/#asterisk J4zen|Ghost (n=Jvan4zen@a213-84-139-244.adsl.xs4all.nl) |
| 08:50.50 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:53.32 | *** join/#asterisk Mongos (i=Mongos@81.90.9.124) |
| 08:58.33 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 09:00.09 | *** join/#asterisk Mongos_ (i=Mongos@81.90.9.124) |
| 09:00.30 | nexilus | mosty... but isnt s the start of any call...? |
| 09:00.37 | mosty | no |
| 09:01.14 | nexilus | then ... either someone should update the wikipedia or im misreading terribly :S |
| 09:01.50 | mosty | you must be misreading |
| 09:02.04 | nexilus | http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf |
| 09:02.12 | nexilus | under "predefined extensions" |
| 09:02.16 | nexilus | what do they mean by S then..? |
| 09:02.23 | JT | bintut: maybe you're not understanding |
| 09:02.41 | JT | bintut: you CANNOT fax over IP over WANs over VOICE codecs. |
| 09:03.24 | bintut | mosty: yeah.. thanks.. :) |
| 09:04.03 | bintut | JT: ok. thanks. |
| 09:05.59 | nexilus | ...anyone? |
| 09:07.25 | JT | nexilus: |
| 09:07.26 | mosty | calls that come in on an analogue line are sent to extension s, priority 1, in whatever context you set in sip.conf |
| 09:07.27 | JT | ~thebook |
| 09:07.27 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 09:07.53 | JT | mosty: analogue line... sip.conf? |
| 09:08.07 | mosty | er, zapata.conf sorry. |
| 09:08.49 | *** join/#asterisk Dovid (n=Dovid@bzq-79-182-14-102.red.bezeqint.net) |
| 09:08.50 | nexilus | im having trouble accepting mosty's answers since they contradict everything ive read so far... |
| 09:09.15 | JT | nexilus: then read something that isn't wrong |
| 09:09.19 | JT | nexilus: like the book |
| 09:09.26 | JT | and asterisk's internal documentation |
| 09:09.59 | Dovid | JT: If i want asteirsk to call two people and then bridge the call what is the easiest way to do it ? |
| 09:10.19 | Dovid | .call file to the first party and then send the call to a context that calls the second party ? |
| 09:10.37 | mosty | dovid: that would work |
| 09:10.43 | nexilus | i have read two books, and the online wikipedia.. both state as far as i remember that the predefined extension s is used for all calls within a context (usually tho when there is no predefined extensions in the context) but shouldnt that in theory work even if there is extensions in a context but theres a general "step 1" for each extension? |
| 09:10.45 | mosty | or use the originate cli command |
| 09:11.10 | mosty | nexilus, you either misread (most likely), or you're reading something that's wrong |
| 09:11.18 | *** join/#asterisk harpal (n=Harpal@124.125.255.223) |
| 09:11.19 | *** join/#asterisk viperdudeuk (n=chatzill@84-45-129-190.no-dns-yet.enta.net) |
| 09:11.52 | Dovid | mosty: Any simpler way to accomplish that ? |
| 09:12.05 | mosty | dovid: that is simple |
| 09:12.14 | Dovid | .call or originate ? |
| 09:12.22 | mosty | they are effectively the same thing |
| 09:12.29 | mosty | pick whichever you prefer |
| 09:12.59 | JT | nexilus: that's not right |
| 09:13.36 | JT | Dovid: AMI originate should be better for high call volumes |
| 09:13.42 | JT | no stuffing around with the FS |
| 09:14.00 | J4k3 | linagee: the closest thing I'll ever get to owning my own switch? ;) |
| 09:14.00 | Dovid | been a while since I played with the AMI |
| 09:14.27 | bintut | brb.. |
| 09:14.42 | bintut | thanks to all who tried to help me.. |
| 09:15.03 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 09:15.30 | Siya | anyone who can point me to a simple way to telling why cdr_mysql won't build? |
| 09:15.46 | Siya | menuselect shows XXX so I've been looking at dependencies for ages |
| 09:15.49 | mosty | you'll have to look at the error from the build |
| 09:15.59 | mosty | do you have mysql dev libs installed? |
| 09:17.36 | Siya | I used to have it workin |
| 09:17.55 | Siya | mosty: yes |
| 09:18.56 | Siya | heh, I lost some txt here |
| 09:19.08 | modu | Someone have configured pickup ? |
| 09:19.37 | *** join/#asterisk Schumie (i=SteveWri@cpc2-rdng2-0-0-cust382.winn.cable.ntl.com) |
| 09:19.40 | Siya | I had to rebuild my machine after a HD crash had it working before the crash. now it won't build and I see no errors |
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| 09:25.18 | BeeBuu | any one know what kinds format can be record by mixmonitor()? |
| 09:27.46 | *** join/#asterisk defswork (n=andy@77.44.54.34) |
| 09:28.05 | defswork | \o/ We've got a new phone system here |
| 09:28.26 | defswork | I asked if we could install asterisk - no chance - we have some shitty LG system |
| 09:28.33 | defswork | and it is /so/ bad |
| 09:29.21 | Siya | mosty: shouldn't there be some error file? |
| 09:29.41 | Siya | no errors being spat at me on the cli |
| 09:33.18 | *** join/#asterisk michael-i (n=michael-@Lc18f.l.pppool.de) |
| 09:33.32 | mosty | siya: you're trying to build asterisk-addons right? and not asterisk? |
| 09:37.13 | obnauticus | How does the music on hold module choose in which order to load music? |
| 09:37.20 | obnauticus | Is it alphabetically in the specified directory? |
| 09:37.25 | Dovid | anyone know the command to quit the AMI ? |
| 09:37.32 | Dovid | i dont see it any where on the wiki |
| 09:38.11 | Dovid | Action: Quit did not work |
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| 10:00.14 | tzafrir | Dovid? logoff? logout? |
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| 10:01.12 | Siya | possibly it was only run the first time I tried installing and not after as I was trying to resolve the dependencies |
| 10:03.08 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
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| 11:22.34 | last1 | anyone alive here ? |
| 11:22.50 | viperdudeuk | hi |
| 11:22.59 | ftugrul | hi |
| 11:23.24 | last1 | how do I know ur not bots ? :) |
| 11:23.36 | J4zen | hi |
| 11:23.41 | MicW | hi |
| 11:23.49 | viperdudeuk | you give me your phone number and i call you |
| 11:23.54 | J4zen | <FATAL ERROR: #381 PLEASE CONTACT ADMINISTRATOR> |
| 11:24.14 | last1 | lol |
| 11:24.36 | last1 | I have two accounts with a voip provider. each account gives me a separate phone #. I connect to them through SIP |
| 11:24.55 | J4zen | Yeah. |
| 11:25.00 | last1 | each account is defined in it's own context [in1] and [in2] in sip.conf ( each has different user, password, etc ) |
| 11:25.09 | last1 | each also calls a different context in extensions.conf |
| 11:25.32 | last1 | the problem is that when a call comes in for the number defined in [in1] it gets picked up by [in2] |
| 11:25.40 | last1 | not sure why |
| 11:26.00 | ftugrul | where can I find the dependency list for asterisk 1.4.13 please? |
| 11:26.08 | ftugrul | I'm trying to compile it. |
| 11:26.09 | viperdudeuk | last1: are you registering the SIP accounts? |
| 11:26.13 | MicW | have you set "context=in1" and "context=in2" in sip.conf (different for both)? |
| 11:26.29 | last1 | [in1] and [in2] are defined exactly alike except for differences in user/password and context they go in extensions.conf |
| 11:26.37 | last1 | they do get registered yes |
| 11:26.58 | last1 | both show up fine if I use: sip show registry |
| 11:27.05 | viperdudeuk | set the extension to ring at the end of the register line |
| 11:27.12 | viperdudeuk | ie |
| 11:27.36 | viperdudeuk | register => user:pass@myvoiprovider/myextentocall |
| 11:28.11 | *** join/#asterisk f0rqu3 (n=f0rqu3@unaffiliated/f0rqu3) |
| 11:28.17 | J4zen | <PROTECTED> |
| 11:28.18 | *** part/#asterisk f0rqu3 (n=f0rqu3@unaffiliated/f0rqu3) |
| 11:28.28 | last1 | le me try that |
| 11:28.45 | viperdudeuk | J4zen: SCP? |
| 11:28.57 | J4zen | "SCP" ? |
| 11:29.04 | viperdudeuk | secure copy |
| 11:29.07 | viperdudeuk | works over SSH |
| 11:29.13 | J4zen | oh it does? i had no idea |
| 11:29.21 | viperdudeuk | if you can SSH in you should be able to SCP |
| 11:29.31 | J4zen | Yeah i can ssh in |
| 11:30.07 | viperdudeuk | get a scp client and use that then |
| 11:30.17 | rob0 | pipe a tar zc through nc, nc -l piped to tar zx on the receiving side |
| 11:30.27 | last1 | I made that change and now it shows: can not find [in1] in context [in2] |
| 11:30.40 | last1 | I should have mentioned that when the number rings I want it to go to a context, not ring directly |
| 11:31.20 | rob0 | "FTP account on Asterisk"? |
| 11:31.31 | J4zen | Sorry, it wasn't on asterisk as it turned out |
| 11:31.35 | J4zen | im using a GUI to asterisk |
| 11:31.38 | J4zen | which came with an FTP |
| 11:31.41 | last1 | not sure where it gets that [in2] is the default context, except for maybe that in2 is defined after in1 ? |
| 11:32.25 | J4zen | that shouldnt mather right? as far as i know the context aren't linear scripted |
| 11:33.11 | last1 | I'm even more confused as to why I need to specify user/password in the contexts if I already have them defined in the register lines |
| 11:33.21 | last1 | can't I have a generic context without user/pass ? |
| 11:33.51 | viperdudeuk | last1: they are used if you use that SIP context to make outoging calls |
| 11:34.18 | last1 | true, just realized that |
| 11:35.08 | last1 | so now the message is as follows: Looking for c1 in c2 |
| 11:35.15 | last1 | after specify /c1 on the register line |
| 11:35.16 | ftugrul | where can I find software dependency list for asterisk 1.4.13 please? |
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| 11:38.12 | last1 | ftugrul: did u try the website ? |
| 11:38.26 | ftugrul | last1, yes, I've checked but I can't find. |
| 11:38.29 | last1 | I'm sure even in the README files that come with the package it will say |
| 11:38.33 | ftugrul | my compilation gave error |
| 11:39.30 | ftugrul | last1, README doesn't contain such information :/ |
| 11:40.12 | last1 | more update: sip.conf does appear to be order specific |
| 11:40.22 | last1 | puttin [in1] context after [in2] fixes the problem |
| 11:40.24 | Uatec | Hi, |
| 11:40.35 | last1 | but makes incoming calls to in2 number impossible now |
| 11:41.33 | Uatec | I have many phones int he office, some of which might be unattended. I can press *8 to pickup these calls. However if I press *8 i don't know who the call was coming from since I don't have the CLID on my screen. |
| 11:42.01 | Uatec | Does anybody know if there is anyway I can present the caller ID to myself when picking up someone elses call? |
| 11:46.15 | rob0 | ftugrul: pastebin the last few lines of the error. |
| 11:46.42 | last1 | for all those that couldn't find an answer to my problem... here it is |
| 11:46.44 | last1 | The only benefit of type=user is when you _want_ to match on username |
| 11:46.44 | last1 | regardless of IP the calls originate from. If the peer is registering to |
| 11:46.44 | last1 | you, you don't need it. If they are on a fixed IP, you don't need it. |
| 11:46.44 | last1 | 'type=peer' is _never_ matched on username for incoming calls, only |
| 11:46.44 | last1 | matched on IP address/port number (unless you use insecure=port or higher). |
| 11:47.38 | last1 | both calls for both numbers were originating from the same IP so it was using the last defined context |
| 11:48.12 | ftugrul | rob0, http://rafb.net/p/PxZg6w44.html |
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| 11:50.26 | rob0 | "Last few lines" = 1000, lol. |
| 11:50.52 | ftugrul | rob0, ^_^ |
| 11:51.01 | rob0 | I'd guess you don't have glibc maybe. |
| 11:51.10 | rob0 | what OS is this? |
| 11:51.39 | last1 | and what language is that |
| 11:51.51 | ftugrul | rob0, glibc 2.3.6 |
| 11:51.59 | ftugrul | rob0, Pardus 2007.2 |
| 11:52.04 | ftugrul | Turkish |
| 11:52.08 | last1 | Pardus ? wtf |
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| 11:52.19 | ftugrul | http://www.pardus.org.tr |
| 11:52.50 | rob0 | Can you compile other things? |
| 11:52.56 | ftugrul | sure |
| 11:53.07 | ftugrul | I've packaged ekiga for Pardus. |
| 11:53.14 | ftugrul | same system, same installation |
| 11:54.25 | rob0 | probably some include file is missing, related to libedit |
| 11:54.38 | ftugrul | libedit? |
| 11:54.40 | rob0 | but I'm only guessing |
| 11:54.45 | rob0 | line 1008 |
| 11:54.55 | ftugrul | is it a dependency or part of asterisk please? |
| 11:55.52 | rob0 | asterisk-1.4.13/main/editline/libedit.a |
| 11:56.52 | ftugrul | rob0, there's no such file there |
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| 11:58.19 | last1 | rob: that's a library that asterisk has to build, no ? |
| 11:58.27 | ftugrul | is it a dependency? |
| 11:58.30 | last1 | if his compile fails, that file won't exist |
| 11:58.38 | ftugrul | yes |
| 11:58.51 | ftugrul | file is not in .tar.gz source |
| 11:59.32 | last1 | even better |
| 11:59.32 | last1 | Registering multiple SIP accounts with one SIP provider has been a nightmare in Asterisk. Or, rather, still is. The match-on-IP scheme for peers is a hack to handle registrations, but not a very good hack. If you register for multiple accounts, the incoming calls will all match the same peer. A poor solution. |
| 12:00.32 | rob0 | ftugrul: Look in the directory. There's an INSTALL file. |
| 12:00.51 | rob0 | Yes, libedit.a is the compiled object. |
| 12:00.59 | ftugrul | found it |
| 12:04.23 | ftugrul | rob0, have you got any idea about how to fix it please? |
| 12:05.09 | last1 | ftugrul: I don't have a solution, sorry. but I am interested to find out why you use this distribution |
| 12:05.25 | *** join/#asterisk myiagy (n=myiagy@189.34.11.211) |
| 12:05.36 | ftugrul | last1, I'm a developer of Pardus, I'm packaging software for Pardus. |
| 12:08.25 | rob0 | If it was me, I might try --disable-readline as per the INSTALL file. |
| 12:10.04 | ftugrul | rob0, already tried :) |
| 12:10.06 | ftugrul | same |
| 12:10.33 | last1 | do a make clean in between |
| 12:10.51 | last1 | might also help to remove config.cache / config.log etc |
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| 12:13.30 | ftugrul | ok, let me try. |
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| 12:20.03 | ftugrul | hmm, it seems like there are dependencies: http://www.debianhelp.co.uk/asterisk.htm#KonaLink8 |
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| 12:26.06 | ftugrul | rob0, I'm still unable to find official/project page of libedit2 |
| 12:26.06 | ftugrul | :) |
| 12:26.17 | ftugrul | searching... |
| 12:27.53 | ftugrul | finally, http://www.thrysoee.dk/editline/ |
| 12:27.55 | ftugrul | :) |
| 12:28.03 | Zuchmir | i'm trying to settup my account with freeworlddialup.com and i can't get it working (i installed a clean switchvox) :-( |
| 12:28.32 | rob0 | I don't have libeditline or libedit, and * compiled for me. |
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| 12:29.13 | ftugrul | hmm. same version rob0? |
| 12:29.34 | rob0 | Asterisk 1.4.13? |
| 12:29.40 | ftugrul | yes |
| 12:31.26 | Zuchmir | any ideas why following simple instructions don't work |
| 12:34.06 | rob0 | Zuchmir, it has been at least a year since I looked at FWD, but I do remember discovering that their SIP service was broken. I think I got it working with IAX2. Or maybe the other way around. :) |
| 12:35.52 | Uatec | I have many phones int he office, some of which might be unattended. I can press *8 to pickup these calls. However if I press *8 i don't know who the call was coming from since I don't have the CLID on my screen. |
| 12:35.55 | Uatec | Does anybody know if there is anyway I can present the caller ID to myself when picking up someone elses call? |
| 12:36.00 | Zuchmir | rob0: do i not use iax.conf for that? (I followed instructions from: http://www.freeworlddialup.com/help/?p=knowledgebase&c=18&a=76 ) |
| 12:36.52 | Zuchmir | rob0: is there another easy to test (free) voip service that is easier to setup |
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| 12:42.15 | ftugrul | rob0, it's the same. asterisk includes source code of editline. it seems like. |
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| 12:46.53 | Zuchmir | I am getting: Host dnsmgr Username Perceived Refresh State |
| 12:46.53 | Zuchmir | 192.246.69.186:4569 N xxxxxx <Unregistered> 60 Rejected |
| 12:47.10 | Zuchmir | when i type iax2 show peers |
| 12:48.38 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 12:52.54 | *** join/#asterisk sawik (n=sawik@82-70-174-190.dsl.in-addr.zen.co.uk) |
| 12:52.56 | *** join/#asterisk UserReg_CL (n=dede@200.113.128.22) |
| 12:53.10 | UserReg_CL | hi.. helpme please.. need update mysql |
| 12:55.30 | modu | Someone have experience with pickup ? |
| 12:56.52 | tzafrir | jbot, tell modu about ask |
| 12:57.17 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 12:57.32 | puzzled | hi |
| 12:58.31 | *** join/#asterisk shtoom (n=shtoom@59.93.120.128) |
| 12:59.06 | modu | When I pickup a call I want my phone to show the real caller (not *9+exten) is there a way to do this ? |
| 12:59.51 | modu | The Contact: header is not updated |
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| 13:02.22 | modu | no one use pickup ? |
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| 13:04.43 | modu | tzafrir: with specifics questions there is no answers |
| 13:05.41 | tzanger | modu: perhaps nobody's run in to your particular problem |
| 13:05.59 | tzanger | I don't use directed pickup myself, but I am pretty sure *8 shows me incoming callerid |
| 13:06.10 | modu | How can someone use asterisk without Pickup ? |
| 13:06.24 | tzanger | modu: phone rings, pick itup |
| 13:06.47 | tzanger | I ran systems with thousands of incoming calls a day without ever using *9 |
| 13:07.03 | Zuchmir | any ideas why i get chan_iax2.c: Registration of 'xxxxxx' rejected: 'Registration Refused' from: '192.246.69.186' |
| 13:07.17 | modu | how do you intercept call ? |
| 13:07.45 | tzanger | I don't often have to do that, and whenI do it's *8 because my phones are in the same pickupgroup |
| 13:08.38 | modu | yes but *8 or *9 is the same things, I'm talking about the pikcup functionality |
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| 13:09.10 | rob0 | -- Registered IAX2 to '192.246.69.186', who sees us as my.ip.add.ress:4569 with no messages waiting |
| 13:09.30 | rob0 | Zuchmir: that's my fwd |
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| 13:13.55 | rob0 | Zuchmir: did you just sign up? Maybe it takes awhile to get your credentials to their server. |
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| 13:20.09 | Zuchmir | rob0: i setup the account yesterday, and did tests with FWD.communicator and it worked |
| 13:22.15 | Zuchmir | rob0: can you msg me the contents of your iax.conf (or can i send msg mine) to compare |
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| 13:23.46 | Davee3 | Hey, would i need (open)SER for a setup with 250 extensions, infrequent calls? |
| 13:23.52 | Zuchmir | rob0: does your iax.conf match http://pastebin.ca/45128 ? |
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| 13:24.51 | viperdudeuk | Davee3: not neccesarily, if you dont have the phones registering too often then asterisk wll cope with 250 extrensions |
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| 13:25.09 | NirS | hello everybody |
| 13:25.11 | NirS | anybody home ? |
| 13:25.22 | viperdudeuk | I have asterisk boxes with 400+ extensions on them |
| 13:25.41 | viperdudeuk | just make sure plenty of RAM and fast CPU |
| 13:26.18 | rob0 | Zuchmir, I have a context in my FWD peer section |
| 13:26.22 | Davee3 | viperdudeuk, I was thinking 4Gb ram - that should be enough right? |
| 13:26.49 | viperdudeuk | yep more than enough I have 400+ with 1Gb ram |
| 13:27.05 | Davee3 | groovy |
| 13:27.27 | NirS | anyone ever used regular expressions inside an Asterisk GotoIF statement ? |
| 13:27.32 | viperdudeuk | it is on a hosted pbx service I run, although now I put openser in front as I have 1000's of subscribers |
| 13:28.05 | viperdudeuk | plus openser allows white labelling |
| 13:28.32 | Davee3 | viperdudeuk, well this will be on a local installation |
| 13:28.34 | *** join/#asterisk Sunmoon__ (n=Sunmoon@ip72-206-113-190.om.om.cox.net) |
| 13:28.38 | rob0 | Zuchmir, and I don't have a type=user block for inbound ... but then ... I don't know for sure that inbound works. :) |
| 13:28.47 | Zuchmir | rob0: i'm behind NAT would that be causing trouble |
| 13:28.52 | rob0 | shouldn't |
| 13:29.02 | *** part/#asterisk Sunmoon__ (n=Sunmoon@ip72-206-113-190.om.om.cox.net) |
| 13:29.42 | *** join/#asterisk jozu (n=torrent@84.120.184.91.dyn.user.ono.com) |
| 13:31.56 | Zuchmir | rob0: i get "fwd-peer/xxxxxx 192.246.69.186 (S) 255.255.255.255 4569 OK (307 ms)", but when i dial the number i get busy signal |
| 13:33.10 | [TK]D-Fender | Zuchmir: You should pastebin the complete call attempt with channel debug enabled. You aren't showing anything useful. |
| 13:33.20 | [TK]D-Fender | Zuchmir: Verbose 10 |
| 13:34.52 | *** join/#asterisk bkw_ (n=brian@adsl-64-149-41-151.dsl.tul2ok.sbcglobal.net) |
| 13:35.47 | *** join/#asterisk stimpie (n=stimpie@84-104-5-115.cable.quicknet.nl) |
| 13:35.56 | Zuchmir | tkd-fender i'm calling from fwd.communicator |
| 13:36.04 | Zuchmir | TO asterisk |
| 13:36.07 | asteriskguy | we have about a 1000 handsets registered to our * server |
| 13:36.19 | asteriskguy | 2 dual core cpus w/8GB of RAM |
| 13:36.22 | brainspiral | Hey. Has anyone used Cisco 79xx phones successfully with any version of asterisk? |
| 13:37.40 | Zuchmir | "iax2 show peers" shows status: ok, but the log shows: chan_iax2.c: Registration of 'xxxxxx' rejected: 'Registration Refused' from: '192.246.69.186' |
| 13:37.54 | [TK]D-Fender | Zuchmir: you have that on your same lan and are calling yourself through IAX? |
| 13:38.24 | *** part/#asterisk bkw_ (n=brian@adsl-64-149-41-151.dsl.tul2ok.sbcglobal.net) |
| 13:38.27 | [TK]D-Fender | brainspiral: plenty of people |
| 13:38.51 | *** join/#asterisk ReDNeQ (i=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 13:38.58 | *** join/#asterisk syneus (n=syneus@host11-100-dynamic.15-87-r.retail.telecomitalia.it) |
| 13:39.01 | *** join/#asterisk Zefk (n=Zefk@wsc-fo.b.astral.ro) |
| 13:39.17 | brainspiral | Thanks. I've deployed, just can't seem to get the same kind of quality out of them (and the Linksys 94x) compared to a Polycom or Aastra. |
| 13:39.30 | *** join/#asterisk ReDNeQ- (i=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 13:39.46 | Zuchmir | tkd-fender: i'm trying to setup asterisk to take calls from my account @ freeworlddialup.com |
| 13:40.43 | modu | brainspiral : do you use pickup features ? |
| 13:41.10 | [TK]D-Fender | Zuchmir: And are trying to register the same account from both FWD-comm & * at the same time? |
| 13:41.28 | rob0 | aha! |
| 13:41.32 | [TK]D-Fender | brainspiral: what kind of "quality"? |
| 13:42.18 | brainspiral | Pickup features are available outside of CallManager? |
| 13:42.36 | rob0 | When using Polycom or Aastra, he hears John F. Kennedy. When using the Cisco 79xx he hears GW Bush. |
| 13:42.45 | Zuchmir | tkd-fender: i setup 2 accounts i am using uno w/FWD-comm and the other in * |
| 13:42.53 | modu | brainspiral: yes but I'm looking for someone that have successfully use it |
| 13:43.30 | brainspiral | Snap, Crackle, Pop & occasional echo. |
| 13:43.45 | rob0 | Kellogg's will sue! |
| 13:43.50 | [TK]D-Fender | Zuchmir: well looks like you got your auth wrong on the register so you are indeed dead in the water. |
| 13:43.51 | Zefk | I developed a small network of asterisk servers across Europe. I need some documentation about how to build a good dial plan. I use on each server contexts like from.trunk1, from.trunk2 ,,, from.trunkn, to.trunk1, to.trunk2 ,,, to.trunkn and a lot of includes like: context from.trunk2 =< { includes => { to.trunk1; to.trunk3;} }. Is this a good solution ? Thx |
| 13:44.04 | brainspiral | I was also a fan of RK. With lots of sugar. |
| 13:44.53 | Zuchmir | tkd-fender: so why is the status: OK? |
| 13:45.16 | [TK]D-Fender | Zuchmir: tahts only the result of a QUALIFY test, not an AUTH test. |
| 13:45.34 | *** join/#asterisk BlackH8t (n=bl@dslb-088-064-156-228.pools.arcor-ip.net) |
| 13:45.34 | Zuchmir | 1 iax2 peers [1 online, 0 offline, 0 unmonitored] |
| 13:45.35 | [TK]D-Fender | Zuchmir: as in "Yes, the host I am pointing to is up". |
| 13:45.56 | [TK]D-Fender | Zuchmir: Its just saying the other side is THERE. It doesn't mean to want to listen to YOU |
| 13:45.57 | *** join/#asterisk stimpie_ (n=stimpie@84-104-5-115.cable.quicknet.nl) |
| 13:46.18 | Zuchmir | oh, ok |
| 13:46.21 | *** join/#asterisk shido6_ (n=shido6@204.126.120.132) |
| 13:46.35 | rob0 | Zuchmir: try swapping the accounts between * and FWD-comm ? |
| 13:47.52 | [TK]D-Fender | rob0: no need |
| 13:48.02 | [TK]D-Fender | rob0: let him configure the one he has right. |
| 13:48.19 | rob0 | I was just thinking maybe the password was wrong |
| 13:50.00 | *** join/#asterisk axisys__ (n=axisys@ip70-177-183-25.dc.dc.cox.net) |
| 13:50.35 | Zuchmir | ...actually when i swap accounts it "hangs up" immediately (the other way gave a busy signal) |
| 13:50.53 | Zuchmir | i can log into either account with FWD-comm |
| 13:51.48 | Zuchmir | Defaddr->IP : 0.0.0.0 Port 0 |
| 13:52.56 | *** join/#asterisk BlackH8t (n=atorvita@dslb-088-064-156-228.pools.arcor-ip.net) |
| 13:54.43 | *** join/#asterisk mao (n=mao@190.84.254.27) |
| 13:55.16 | Zuchmir | how do i change verbose level |
| 13:55.17 | *** join/#asterisk bantu (n=Miranda@p54A3294E.dip0.t-ipconnect.de) |
| 13:55.19 | *** join/#asterisk axscode (i=axscode@187.223.48.60.klj04-home.tm.net.my) |
| 13:55.48 | [TK]D-Fender | Zuchmir: "Thats nice", now go fix your Register statement. |
| 13:55.58 | *** join/#asterisk Bladerunner05 (n=feelme@81-174-56-54.static.ngi.it) |
| 13:56.53 | Zuchmir | tkd-fender: register => xxxxxx:password@iax2.fwdnet.net |
| 13:58.12 | Bladerunner05 | Hi all |
| 13:59.01 | Zuchmir | ... where xxxxxx=myFWDnumber, password=mySecret |
| 13:59.27 | Bladerunner05 | I compile asterisk 1.4.13 with spandsp (latest version), please see http://www.pastebin.ca/772013 for notice. I get 3 errors for underined symbol and if I comment out that lines in app_rxfax.c I can receive a fax. But I notice that the fax is compressed (too small and too large) |
| 14:01.01 | coppice | if you used the latest version of spandsp you wouldn't need to comment out those lines. |
| 14:01.03 | coppice | I guess you don't use FAX much. the too small/too large thing is due to broken viewers |
| 14:01.56 | *** join/#asterisk southtel (n=southtel@68-114-23-151.dhcp.gwnt.ga.charter.com) |
| 14:03.46 | *** join/#asterisk chode (n=chode@pD9E8BE9D.dip0.t-ipconnect.de) |
| 14:04.07 | *** join/#asterisk [intra]lanman (n=lanman@va-76-6-212-80.dhcp.embarqhsd.net) |
| 14:04.41 | Bladerunner05 | •coppice• I used spandsp.0.0.4 |
| 14:04.48 | *** join/#asterisk duckz (n=duckz@85.204.47.228) |
| 14:04.56 | coppice | then those functions exists |
| 14:05.15 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:05.16 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 14:05.38 | Bladerunner05 | •coppice• I understand but If I don't comment out that lines asterisk crash receving fax |
| 14:06.11 | coppice | perhaps you have an older version of spandsp somewhere on your machine, and that is being picked up at runtime |
| 14:06.23 | codefreeze | Siya: what I do, is look thru the config.log file and try to see why the configure didn't catch your mysql stuff. You can correlate that with the configure script. |
| 14:06.45 | Siya | codefreeze: thanks |
| 14:07.25 | Siya | I googled loads but nowhere did I find the suggestion to run ./configure (ignorance was not bliss this time...) |
| 14:07.40 | Siya | cdr_mysql works fine now though :) |
| 14:09.20 | *** join/#asterisk robl^ (n=robl@pdpc/supporter/monthlybyte/robl) |
| 14:11.17 | *** join/#asterisk axscode (i=axscode@187.223.48.60.klj04-home.tm.net.my) |
| 14:12.12 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 14:12.29 | *** join/#asterisk fskrotzki (n=fskrot@host198.textwise.com) |
| 14:13.35 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:14.52 | _x86_ | hey guys.... |
| 14:15.12 | _x86_ | I have a bunch of analog FXS stations going to a channel bank, then going over a T1 to Asterisk |
| 14:15.33 | _x86_ | zapata.conf has usecallerid=yes, but no initial caller ID is set on the channels in zapata.conf |
| 14:16.04 | _x86_ | when a call is dialed, I have an AGI that sets the caller ID, and it puts it into CDR properly |
| 14:16.30 | _x86_ | but when one of the analog stations calls, say, a SIP phone, the only caller ID that comes up is the IP of the asterisk server |
| 14:16.43 | _x86_ | how can i make it so that it sends the caller ID that the AGI sets? |
| 14:18.05 | [TK]D-Fender | _x86_: pastebin is your friend.... |
| 14:18.20 | [TK]D-Fender | _x86_: include plenty of backup. |
| 14:19.11 | Bladerunner05 | •coppice• U mean libspandsp ? |
| 14:19.56 | _x86_ | [TK]D-Fender: what do you want me to pastebin? |
| 14:21.00 | coppice | yes |
| 14:21.14 | *** join/#asterisk raydogg`` (n=truz_`24@74-129-166-232.dhcp.insightbb.com) |
| 14:21.52 | [TK]D-Fender | _x86_: Think on it.... |
| 14:22.31 | *** join/#asterisk nfi|ermes (n=ErMeS@217.220.121.62) |
| 14:23.16 | _x86_ | [TK]D-Fender: my sykick powars tell me you want zapata.conf? |
| 14:23.20 | asanchez_ | is recomendable asterisk 1.4 for production ? or better using 1.2 ? |
| 14:24.39 | *** join/#asterisk kv0s (n=kv0s@p4FD256DC.dip.t-dialin.net) |
| 14:24.41 | kv0s | Hi! |
| 14:24.55 | [TK]D-Fender | _x86_: Zapata, your dialplan, your scripts, EVERYTHING. Trust = 0 :) |
| 14:24.55 | _x86_ | alejandro: 1.2 is no longer supported |
| 14:25.04 | _x86_ | [TK]D-Fender: trust++ buddy! ;) |
| 14:25.23 | *** join/#asterisk af_ (n=getsmart@88-149-240-124.dynamic.ngi.it) |
| 14:25.31 | badcfe | is there a way of running asterisk as asterisk user and not doing chmod a+w /var/run |
| 14:26.09 | badcfe | it seems to be needed cause of creationg of the /var/run/asterisk/ctl |
| 14:26.09 | *** join/#asterisk saftsack (n=saftsack@pD9E05A8E.dip.t-dialin.net) |
| 14:26.09 | badcfe | but i dont like doing /var/run world writeable |
| 14:27.13 | *** join/#asterisk dioedu (n=dioedu@201.7.117.114) |
| 14:27.24 | alejandro | _x86_: I know, but for example, debian still uses 1.2 because they consider more stable. |
| 14:27.25 | *** join/#asterisk Dabba (n=d@ipv6.mfnx.ip6net.net) |
| 14:27.33 | alejandro | I wanted to know other opinions. |
| 14:27.46 | _x86_ | alejandro: debian will use 1.2 for the next 10 years because debian is so slow it's pathetic ;) |
| 14:27.49 | *** join/#asterisk bl4q (i=me@1.1.1.vg) |
| 14:27.51 | *** join/#asterisk IPetrov2 (i=IPetrov2@ppp85-140-235-143.pppoe.mtu-net.ru) |
| 14:28.01 | _x86_ | [TK]D-Fender: http://pastebin.ca/772032 <-- relevant section of zapata.conf |
| 14:28.30 | Dabba | hi all, fresh 1.4.13 asterisk RH9 installed ok, seg faults as soon as calls in/out any ideas :-) |
| 14:28.56 | tzafrir | Dabba, RH9??? |
| 14:29.08 | tzafrir | why would you inflict such a pain upon yourself? |
| 14:29.19 | Dabba | legacy box was running 1.2 from ages ago, its on site and im not |
| 14:29.40 | _x86_ | [TK]D-Fender: http://pastebin.ca/772034 <-- muh script |
| 14:30.25 | [TK]D-Fender | _x86_: 1.4? |
| 14:30.25 | tzafrir | alejandro, Debian stable has 1.2 . Debian Sid has 1.4 . Sadly Lenny still has 1.2 due to sad technical reasons |
| 14:30.35 | dioedu | hi, i get a thing in my asterisk when i was in a call, and i had any dtmf, asterisk manager always sent me a unlink and a link... this is a normal feature ? |
| 14:30.58 | dioedu | sip <=> sip |
| 14:31.36 | Dabba | is a core dump any use as that is what asterisk says segmentation fault (core dumped) |
| 14:31.58 | _x86_ | [TK]D-Fender: hmm... wait a sec.... |
| 14:32.25 | _x86_ | [TK]D-Fender: my dialplan is jacked.... it only sets caller ID after the SIP extension matches |
| 14:34.17 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 14:35.29 | nexilus | is there any common issue that results in some phones causing it not to ring on the "caller" side, but that the phone rings as it should on the callee side? |
| 14:35.29 | _x86_ | [TK]D-Fender: ok, but when those same analog FXS stations call to a remote asterisk extension, the caller ID shows up as the IP of the original asterisk server |
| 14:36.02 | [TK]D-Fender | _x86_: why am I not seeing your dialplan and call execution? |
| 14:36.26 | *** join/#asterisk debiano777 (n=nana@213-140-19-123.fastres.net) |
| 14:36.36 | debiano777 | hi all |
| 14:37.01 | debiano777 | anyone can help me to setup asterisk CDR in mysqul? |
| 14:37.06 | debiano777 | mysql |
| 14:37.20 | badcfe | so is it _really_ the only way of running as asterisk user to make /var/run/ worls writeable? |
| 14:38.08 | nexilus | ah.. found my error.. since i use "Answer" and pull the call through an agi first it gets answered directly, thus no "ringing" on the callee side |
| 14:38.14 | nexilus | is there a way to remedy that? |
| 14:38.53 | JT | don't answer? |
| 14:39.17 | *** join/#asterisk zydrunas (n=Miranda@216.49.228-ip-184.ckt.net) |
| 14:39.30 | nexilus | JT: if i remove Answer will the agi still run tho? suppose it does |
| 14:39.36 | rob0 | badcfe: ?? |
| 14:39.46 | _x86_ | [TK]D-Fender: because it's jacked! i'm fixin it ;) |
| 14:40.00 | *** join/#asterisk blq- (i=me@1.1.1.vg) |
| 14:40.05 | rob0 | badcfe, set appropriate paths in asterisk.conf (paths that your user can write.) |
| 14:40.44 | *** join/#asterisk david_totalcom (n=david@i14-98.shosting.atw.hu) |
| 14:40.55 | david_totalcom | Hellello! |
| 14:41.12 | *** join/#asterisk bantu (n=Miranda@p54A3294E.dip0.t-ipconnect.de) |
| 14:41.45 | _x86_ | [TK]D-Fender: yeah it was my own dumbness... fixed the dialplan and it works perfectly now :) |
| 14:43.00 | david_totalcom | Is there any way to reorganize the sounds of the asterisk voicemail system? I try to change the language, but in hungarian the grammar differences don't make my life easier. I have to change the order of the number and other sounds. |
| 14:43.24 | debiano777 | anyone can help me to setup cdr.conf for mysql database? |
| 14:45.34 | *** join/#asterisk felix_da_catz (n=felix@PixNationalGroup.phonoscope.com) |
| 14:48.08 | *** join/#asterisk McDouglas (i=lala@wkgpwy3nv7.adsl.datanet.hu) |
| 14:48.50 | *** join/#asterisk bl4q (n=Bl@dslb-088-064-156-228.pools.arcor-ip.net) |
| 14:48.50 | *** join/#asterisk blq (i=me@1.1.1.vg) |
| 14:48.52 | McDouglas | anyone knows how can i check if asterisk makes a blind transfer or an attended transfer upon a transfer request? |
| 14:49.59 | *** join/#asterisk saftsack (n=saftsack@pD9E05A8E.dip.t-dialin.net) |
| 14:50.01 | tzafrir | jbot, tell debiano777 about ask |
| 14:50.02 | McDouglas | i added atxfer => # to features.conf, but it loks like asterisk makes a blindtransfer when someone uses the # button to initiate a transfer |
| 14:50.11 | [TK]D-Fender | McDouglas: Lookup the doc on variables and read it over a few times. |
| 14:51.39 | *** join/#asterisk callguy (n=callguy@pool-71-255-162-167.bstnma.east.verizon.net) |
| 14:52.10 | *** join/#asterisk ming_zym (n=ming_zym@124.14.233.71) |
| 14:53.16 | debiano777 | i intall asterisk-addon1.4, i create a mysql database, and configure cdr_mysql.conf with my database how can i do to put the asterisk cdr in mysql database without csv ? |
| 14:53.47 | debiano777 | i must add some row in cdr.conf? |
| 14:58.26 | codefreeze | debiano777: did you install the .so (s) from the addons into your asterisk dist? Does it load? Can you see any console messages when asterisk starts about mysql? |
| 15:04.05 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 15:04.08 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 15:04.15 | dandre | hello, |
| 15:05.34 | dandre | I have see on xorcom site some nice astribanks. Is there some alternative with less port? For instance I need a bank with 2 BRI, 4 FXO and 1 or 2 FXS. |
| 15:06.43 | tzafrir | dandre, hi (/me from Xorcom) I'm afraid that this combination is not really possible. You can have a 6+2 combination, though |
| 15:08.52 | *** join/#asterisk coppice_ (n=chatzill@102.204.17.210.dyn.pacific.net.hk) |
| 15:09.08 | dandre | Yes I have seen it but I target some small companies who may have needs for few FXS, FXO and BRI. All other stuff is VOIP SIP Phones. If I add to use these I would be 3 times the expected channel price |
| 15:09.58 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 15:13.36 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 15:14.14 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 15:14.34 | _x86_ | tzafrir: i was looking for you the other day... we already had a channel bank, but we were needing to purchase a T1 card to integrate it... was going to buy a 24 port astribank if it was cheaper than the T1 card (Sangoma A102D-x) |
| 15:15.11 | *** join/#asterisk jo_edu (n=chatzill@201-95-153-132.dsl.telesp.net.br) |
| 15:15.21 | jo_edu | hi |
| 15:15.31 | tzafrir | _x86_, hmm... I don't think that it is cheaper than a single-span T1 card |
| 15:16.25 | [TK]D-Fender | _x86_: Add the CB cost in there too..... be fair |
| 15:17.10 | jo_edu | is the first time I use IRC. I study asterisk. |
| 15:17.43 | jo_edu | I like change knowloge |
| 15:19.56 | _x86_ | tzafrir: we only buy dual and quad-port T1 cards, with HWEC |
| 15:20.01 | *** join/#asterisk gerhard7 (n=gerhard@82-170-9-68.dsl.ip.tiscali.nl) |
| 15:20.24 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 15:20.24 | _x86_ | tzafrir: the sangoma A102D-x was about $1400, and your 24 port FXS astribank was a little higher |
| 15:20.27 | [TK]D-Fender | _x86_: You get CB's for free? Can I have a few? :) |
| 15:20.41 | _x86_ | [TK]D-Fender: we had one spare, in this case |
| 15:20.41 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:21.09 | *** part/#asterisk kv0s (n=kv0s@p4FD256DC.dip.t-dialin.net) |
| 15:22.00 | ManxPower | _x86_: we use all 2-port cards |
| 15:22.50 | _x86_ | ManxPower: we only have 3 PCIe slots available per server (2U HP box), so we sometimes have to use 4 port cards to get the density we need |
| 15:24.02 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 15:24.13 | ManxPower | The last Adtran TA 750 I bought cost me $300 and that was with 8xFXO and the rest FXS. |
| 15:24.20 | ManxPower | eBay, of course. |
| 15:26.25 | _x86_ | that TA750 looks nice |
| 15:26.35 | *** part/#asterisk Dabba (n=d@ipv6.mfnx.ip6net.net) |
| 15:26.54 | ManxPower | it also included the battery pack. |
| 15:27.20 | ManxPower | Granted, I did get an unusually good deal on it. Most of the time the cost is closer to $500 - $600 |
| 15:28.58 | _x86_ | you can get one of those with 24 FXS ports for $500? |
| 15:29.00 | _x86_ | new? |
| 15:29.15 | ManxPower | I didn't say it was new. |
| 15:29.54 | ManxPower | For stuff from eBay my customers usually keep one spare device for each region they do business in. |
| 15:33.14 | *** join/#asterisk mistik1 (n=mistik1@ool-4352c7d3.dyn.optonline.net) |
| 15:33.44 | *** join/#asterisk mrw (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 15:33.46 | mistik1 | hello |
| 15:34.16 | mistik1 | Is there any video conference software for asterisk out there? |
| 15:34.58 | ManxPower | mistik1: no, but most SIP software should work with Asterisk. Asterisk's video support is not very complete, but people have used it. |
| 15:35.46 | mistik1 | ManxPower: Yes, I've tried it with SIP and IAX2 protocols and it works great |
| 15:36.02 | mistik1 | I heard about video support and was wondering how that is going |
| 15:38.06 | *** join/#asterisk mao (n=mao@190.84.254.27) |
| 15:39.51 | *** part/#asterisk david_totalcom (n=david@i14-98.shosting.atw.hu) |
| 15:43.00 | *** join/#asterisk Belgarath (i=belgarat@banda.pl) |
| 15:43.53 | *** join/#asterisk mihinomenest (i=1oXd@66.255.220.17) |
| 15:43.54 | agx | anyone is using a big FXS gateway (ports>20)? what do you suggest? Patton?? |
| 15:44.17 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-724b8ba603112084) |
| 15:45.28 | *** join/#asterisk sigmounte (n=sigmount@88.172.80.96) |
| 15:48.39 | [TK]D-Fender | agx: Heard little about them, but that much has been somewhat positive |
| 15:48.56 | [TK]D-Fender | agx: Mediatrix is pretty good, AudioCodes confusing, but also pretty decent |
| 15:50.59 | coppice_ | mediatrix sucks. they have the buggiest protocols I have ever seen |
| 15:52.37 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 15:52.51 | dioedu | Hello, I'm using asterisk 1.2 and asterisk manager to receive the events. Today I see that when I am in a call (SIP <=> SIP) and press some key (DTMF) in the conversation, I receive a unlink event follow by link event. Is this normal ? |
| 15:52.52 | *** join/#asterisk JackEStorm (n=no@ip68-225-77-136.no.no.cox.net) |
| 15:52.53 | [TK]D-Fender | coppice_: And specific items on your "offenders" list? |
| 15:53.55 | coppice_ | I things only work with mediatrix boxes because people implement workarounds for them. Their T.38 some bizarre stuff. Their SIP is weird too |
| 15:56.59 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:02.32 | *** join/#asterisk axscode (n=axscode@187.223.48.60.klj04-home.tm.net.my) |
| 16:05.16 | MicW | is there a way to create an "initial hint" for phones which are offline after asterisk is startet (i see this snom-phones as "online" until they conencted at least once with asterisk) |
| 16:06.43 | modu | Is there new people here that have successfully used pickup functions ? |
| 16:07.35 | *** join/#asterisk russell (n=russell@75.153.47.179) |
| 16:08.19 | modu | I desperately looking for info on pickup |
| 16:09.26 | ManxPower | modu: We usually use parking, instead of Pickup. What specific issue are you having? |
| 16:10.07 | modu | ManxPower: when i pickup a call I want to see the initial caller on my phone : no *9+ext |
| 16:11.01 | ManxPower | modu: you cannot do that with Asterisk |
| 16:11.03 | modu | my phone (aastra) support "sip udate callerid" based on the Contact header, but asterisk does not update it |
| 16:11.15 | ManxPower | asterisk does not support that feature. |
| 16:11.28 | Qwell | there is a patch on the bug tracker |
| 16:11.35 | Qwell | I think |
| 16:11.54 | Qwell | maybe that's something else.. called party id |
| 16:12.18 | modu | thanks very much I'm google it ... |
| 16:14.35 | *** join/#asterisk netstatic (n=aboroda@38.113.5.165) |
| 16:15.06 | modu | Qwell: do you know if that was usable for pickup or only manually updating callerid ? |
| 16:15.34 | *** join/#asterisk Nugget (i=nugget@carrera.macnugget.org) |
| 16:15.35 | [TK]D-Fender | last I heard it was only manual |
| 16:17.29 | *** join/#asterisk Nugget (i=nugget@carrera.macnugget.org) |
| 16:21.00 | modu | Ok so the last chance is to get some light from #asterisk-dev :-) |
| 16:24.53 | *** join/#asterisk gpowers (n=glenn@208.66.168.244) |
| 16:25.00 | *** join/#asterisk techie (n=techie@adsl-76-214-20-56.dsl.lsan03.sbcglobal.net) |
| 16:25.54 | *** join/#asterisk coderAst (n=root@200.93.195.132) |
| 16:26.43 | coderAst | hi everybody. I need some info about making an outgoing call with Asterisk::Outgoing perl module |
| 16:28.41 | gpowers | usually, you can just write a "call" file, info at: http://www.voip-info.org/wiki/index.php?page=Asterisk+auto-dial+out |
| 16:28.47 | gpowers | which perl mod are you refering to? |
| 16:28.48 | *** join/#asterisk EnigmaCurry (n=user@c-24-10-239-16.hsd1.ut.comcast.net) |
| 16:30.06 | coderAst | i'm using Asterisk::Outgoing perl module. it generates the call file and place it inside /var/spool/asterisk/outgoing |
| 16:30.23 | gpowers | what happens next? |
| 16:30.24 | coderAst | but i doesn't call |
| 16:30.37 | coderAst | i want to call to a local extension |
| 16:30.39 | coderAst | sip |
| 16:31.05 | coderAst | for example i want to call to ext 112 sip |
| 16:31.17 | coderAst | so i set on Channel: sip/112 |
| 16:31.23 | coderAst | is this ok? |
| 16:31.27 | gpowers | what are the console messages? |
| 16:31.39 | coderAst | no messages about sip 112 |
| 16:31.48 | nestAr | lmao |
| 16:32.07 | nestAr | first attempt at a microbrowser page and i crash the phone |
| 16:32.11 | nestAr | damn, i'm good. |
| 16:32.54 | nestAr | well, to be totally fair, i just threw a url at it |
| 16:33.22 | nestAr | it did not like it.. it made me laugh though.. so i guess that can go on the pro side of the list |
| 16:33.49 | coderAst | gpowers what is supposed to appear inthe console wiht outbound console |
| 16:34.10 | *** join/#asterisk e` (n=e@38.102.196.202) |
| 16:34.37 | MicW | seems that the initial hints are correct. but the snom's leds are off for hint "unavaiable" |
| 16:34.43 | MicW | can i change this? |
| 16:35.43 | *** join/#asterisk zydrunas (n=Miranda@216.49.228-ip-184.ckt.net) |
| 16:35.59 | [TK]D-Fender | MicW: Go read your admin guide. |
| 16:36.22 | MicW | [TK]D-Fender: is this an asterisk or a snom setting? |
| 16:36.33 | [TK]D-Fender | MicW: Snom clearly. |
| 16:36.45 | [TK]D-Fender | MicW: On a Polycom they'd be flashing. |
| 16:36.56 | MicW | ok, sorry for non-topic |
| 16:38.35 | *** part/#asterisk coderAst (n=root@200.93.195.132) |
| 16:38.53 | *** join/#asterisk coderAst (n=root@200.93.195.132) |
| 16:41.15 | coderAst | has anyone worked with perl module Asterisk::Outgoing? |
| 16:42.04 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 16:42.04 | *** join/#asterisk muiro (n=muiro@cpe-76-189-84-108.neo.res.rr.com) |
| 16:42.10 | *** join/#asterisk maxie9 (n=maxie9@125.212.91.153) |
| 16:42.47 | *** part/#asterisk Modu (n=modu@rue92-6-82-237-172-115.fbx.proxad.net) |
| 16:45.55 | *** join/#asterisk Tomasu (n=moose@S0106001bfcce6c34.ed.shawcable.net) |
| 16:45.57 | *** join/#asterisk kaylinx (n=kaylinx@24.66.32.135) |
| 16:46.02 | *** join/#asterisk maxie9 (n=maxie9@125.212.91.153) |
| 16:46.07 | kaylinx | hello |
| 16:46.19 | kaylinx | having some issues with setting up trix |
| 16:46.31 | MicW | [TK]D-Fender: seems to be an saterisk issue: http://lists.digium.com/pipermail/asterisk-users/2007-March/183310.html |
| 16:46.32 | [TK]D-Fender | kaylinx: You're in the wrong channel. |
| 16:46.35 | [TK]D-Fender | ~trixbox |
| 16:46.36 | jbot | rumour has it, trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox seriously painful to support, and thus you will find little help here for it. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
| 16:46.41 | kaylinx | ok |
| 16:46.43 | kaylinx | sorry |
| 16:46.54 | Tomasu | I'm wondering how I can make my asterisk setup allow calling into a certian extension that asks for a number, and then calls you back just before it starts rining the number you put in... |
| 16:47.03 | *** part/#asterisk kaylinx (n=kaylinx@24.66.32.135) |
| 16:47.34 | tzafrir | Tomasu, show application Read |
| 16:47.49 | coderAst | hey do you have info about a channel for developers with perl for astersik or something like that? |
| 16:47.54 | *** join/#asterisk e` (n=e@38.102.196.202) |
| 16:48.04 | Tomasu | ok, but how about the part where it lets you hang up and calls two numbers (yours and the one you entered) and connects them? |
| 16:48.26 | debiano777 | i try to compiling asterisk addons and i have this error cdr_addon_mysql.so': No such file or directory |
| 16:48.28 | maxie9 | can anyone help me setting up asterisk flash operator panel |
| 16:48.45 | debiano777 | someone cat day me why? |
| 16:49.33 | tzafrir | debiano777, please pastebin a more complete trace |
| 16:49.48 | [TK]D-Fender | Tomasu: Why would it hang up and call you back, or are you planning on providing *2* different numbers? |
| 16:49.58 | MicW | how can i see if my running asterisk is bristuff+patched? |
| 16:50.06 | *** join/#asterisk phearless (n=phear@host217-34-75-65.in-addr.btopenworld.com) |
| 16:50.23 | tzafrir | MicW, normally: 'show version' |
| 16:50.32 | tzafrir | also: do you have the command 'bri'? |
| 16:50.54 | Tomasu | [TK]D-Fender: I'd like to beable to call my asterisk number from my cell phone, enter a phone number, hang up, and have it call me back and the other number, then connect the two.. so I can use some of the free incomming minutes on my cell ;) |
| 16:50.59 | MicW | No such command 'bri' |
| 16:51.05 | MicW | (on the console) |
| 16:51.16 | [TK]D-Fender | Tomasu: Ah. Go lookup "call files" on the WIKI |
| 16:51.19 | [TK]D-Fender | ~wikis |
| 16:51.20 | jbot | [wikis] http://www.voip-info.org |
| 16:51.35 | tzafrir | MicW, you have 'zap' ? |
| 16:51.58 | Tomasu | [TK]D-Fender: hmm, willd o thanks |
| 16:52.04 | *** part/#asterisk coderAst (n=root@200.93.195.132) |
| 16:52.20 | MicW | if i have bristuff, i installed only the patch, not the whole bristuff. don't know if this makes the difference |
| 16:52.35 | maxie9 | help me please setup asterisk flash operator panel... |
| 16:52.40 | tzafrir | MicW, what patch do you refer to, exactly? |
| 16:52.50 | tzafrir | asterisk.patch? |
| 16:52.52 | debiano777 | for x in app_addon_sql_mysql.so app_saycountpl.so cdr_addon_mysql.so res_config_mysql.so; do /usr/bin/install -c -m 755 $x /usr/lib/asterisk/modules ; done |
| 16:52.54 | MicW | the asterisk-patch from the bristuff download |
| 16:52.55 | [TK]D-Fender | maxie9: This is not a FOP support channel. |
| 16:52.59 | MicW | yes |
| 16:53.17 | tzafrir | Why do you want to use bristuff? What for, specifically? |
| 16:53.36 | MicW | make my snom leds light when the phones are not registered |
| 16:53.54 | tzafrir | debiano777, so that has failed to build for some reason, I guess |
| 16:54.53 | MicW | i have recompiled (and i'm sure that i have the bristuff patch) but show version shows nothing of it |
| 16:55.08 | tzafrir | MicW, So you don't really need patched zaptel, libpri and certainly not libgsmat |
| 16:55.26 | tzafrir | what you did should be fine |
| 16:55.28 | MicW | no. i simply installed asterisk.patch |
| 16:55.33 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 16:55.33 | *** mode/#asterisk [+o anthm] by ChanServ |
| 16:55.35 | MicW | but it is not working |
| 16:55.42 | *** join/#asterisk axscode (n=axscode@187.223.48.60.klj04-home.tm.net.my) |
| 16:55.53 | debiano777 | for x in app_addon_sql_mysql.so app_saycountpl.so cdr_addon_mysql.so res_config_mysql.so; do /usr/bin/install -c -m 755 $x /usr/lib/asterisk/modules ; done |
| 16:55.57 | MicW | the leds blink when someone is called and lights when someone is calling |
| 16:56.00 | debiano777 | - /usr/bin/install: impossibile fare stat di `app_addon_sql_mysql.so': No such file or directory |
| 16:56.07 | MicW | but are off when someone's phone is off |
| 16:56.19 | debiano777 | this is the problem wen i do make install |
| 16:56.22 | tzafrir | jbot, tell debiano777 about pb |
| 16:56.51 | [TK]D-Fender | MicW: This seems to be rather exclusive behaviour with SNOM, and may be normal.... |
| 16:57.02 | [TK]D-Fender | ~pb |
| 16:57.03 | jbot | A Pastebin is a place to paste your stuff 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 |
| 16:57.09 | MicW | [TK]D-Fender: it was working with asterisk 1.2 |
| 16:57.25 | [TK]D-Fender | MicW: And what version are you running exactly now? |
| 16:57.34 | tzafrir | [TK]D-Fender, actually that "tell" form works well |
| 16:57.39 | MicW | 1.4.11 |
| 16:57.43 | MicW | +bristuff patch |
| 16:57.53 | lirakis | how can i kill a channel on cli |
| 16:57.55 | lirakis | i always forget |
| 16:58.11 | tzafrir | soft hangup <tab><tab> |
| 16:58.24 | [TK]D-Fender | tzafrir : but ~pb is gauranteed and we KNOW thet he reads it instead of missing some probably buried notive he'lll never see in his client. |
| 16:59.15 | [TK]D-Fender | MicW: And What does "core show hint" tell you? |
| 16:59.23 | [TK]D-Fender | MicW: And What does "core show hints" tell you? |
| 16:59.50 | debiano777 | i posted my problem in http://paste.debian.net/42313 |
| 16:59.56 | Tomasu | [TK]D-Fender: heh, thanks, it seems it'll be easier than I expected. found a blog via the wiki that has a config that'll work (after some tweaks). :) |
| 17:00.23 | MicW | "unavailable" |
| 17:00.28 | MicW | for the phones which are off |
| 17:00.53 | MicW | i changed asterisk like mentioned in the mailing list and now it works |
| 17:02.34 | nestAr | guess i'll need to read polycom's guidelines, it doesn't seem to love 100% legit XML |
| 17:02.43 | nestAr | err XHTML |
| 17:04.40 | MicW | what parameter is used to set the time after which a phone which is switched off gets "unavailable" in the hints? |
| 17:05.52 | *** join/#asterisk axscode (i=axscode@134.84.48.60.klj04-home.tm.net.my) |
| 17:07.28 | *** join/#asterisk zerohalo (n=zeroHalo@pool-71-255-162-167.bstnma.east.verizon.net) |
| 17:09.49 | reber | what is the cheapest sip provider for french people ? |
| 17:11.32 | docelmo | reber check out google or some of the better known providers out there |
| 17:11.36 | ManxPower | MicW: I don't know for sure, but I suspect that if qualify=X is used, then anytime the phone does not respond in X milliseconds then it will be unavailable. If qualify is not used, then when the registration times out it will become unavailable. |
| 17:11.46 | *** join/#asterisk dexteruk (n=charper@78.90.15.216) |
| 17:12.10 | reber | docelmo, i have a list, but comparing each other is a work ... |
| 17:12.20 | dexteruk | im compiling zaptel and i get this error |
| 17:12.22 | MicW | the phones gets "unreachable" shortly after i switched them off |
| 17:12.29 | MicW | but the hints remain "idle" |
| 17:12.35 | dexteruk | '/usr/src/zaptel/tor2.c:603: warning: asm operand 0 probably doesn’t match constraints' |
| 17:12.37 | ManxPower | MicW: then I was wrong |
| 17:14.00 | MicW | ok, i'll try again tomorow. bye and thanks a lot |
| 17:14.57 | *** part/#asterisk dexteruk (n=charper@78.90.15.216) |
| 17:15.40 | *** join/#asterisk dexteruk (n=charper@78.90.15.216) |
| 17:17.11 | dexteruk | im having problems compiling asterisk and zaptel this is my error can anyone help? '/usr/src/zaptel/tor2.c:603: warning: asm operand 0 probably doesn’t match constraints' |
| 17:17.28 | *** join/#asterisk viperdudeuk (n=chatzill@84-45-129-190.no-dns-yet.enta.net) |
| 17:17.32 | dexteruk | im runing fedora core 7 |
| 17:17.47 | viperdudeuk | hi |
| 17:18.22 | dexteruk | hi |
| 17:18.38 | viperdudeuk | i am having trouble transferring calls with linksys 942's * reports "both legs must reside on Asterisk box to transfer at this time", anyone know what is going on? |
| 17:19.35 | viperdudeuk | i presume it is a linksys setting but my customer is non techie so i need to find out what |
| 17:20.35 | *** join/#asterisk axscode (i=axscode@134.84.48.60.klj04-home.tm.net.my) |
| 17:22.24 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 17:23.40 | steve | what dial rule do I need to match everything beginning with 0? |
| 17:23.49 | Qwell | _0. |
| 17:23.57 | steve | thansk |
| 17:23.58 | steve | thanks* |
| 17:26.08 | dexteruk | im having problems compiling asterisk and zaptel this is my error can anyone help? '/usr/src/zaptel/tor2.c:603: warning: asm operand 0 probably doesn’t match constraints' |
| 17:26.52 | Qwell | dexteruk: If you don't use tor2, disable it in menuselect |
| 17:27.29 | steve | hmm... now my american friend is telling me "all circuits are busy now" .. but they aren't, nobody except me is using this system? |
| 17:27.50 | steve | the system is meant to route outbound calls to my zaptel card |
| 17:27.51 | Qwell | steve: using something silly like trixbox/freepbx? |
| 17:27.53 | tzafrir | tor2 has inline assmebly? |
| 17:28.10 | Qwell | tzafrir: no idea.. |
| 17:28.34 | *** join/#asterisk Marc29MQC (n=iluvjdmc@125.60.241.175) |
| 17:28.40 | De_Mon | this is odd.. I have an extension that parks calls, so I dial 7 and get parked. I open another line on the phone do the same thing, now I get a warning. |
| 17:28.44 | De_Mon | [Nov 13 12:29:02] WARNING[6585]: channel.c:2336 __ast_read: Exception flag set o |
| 17:28.47 | De_Mon | n 'SIP/jon-0824b300', but no exception handler |
| 17:29.02 | De_Mon | put that call on hold, dial 7 from a 3rd line on the same phone, now I get the same warning twice... |
| 17:29.16 | tzafrir | dexteruk, what version of Zaptel? |
| 17:29.17 | steve | Qwell: yes :( |
| 17:29.21 | Qwell | ~freepbx |
| 17:29.22 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 17:30.16 | *** join/#asterisk gardo (n=gardo@121.97.177.128) |
| 17:31.02 | *** join/#asterisk markit (n=marco@88-149-177-66.static.ngi.it) |
| 17:31.23 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
| 17:31.59 | dexteruk | tzafrir: SVN-trunk-r3093M |
| 17:32.08 | markit | hi, just noticed that monitoring calls, now I have an error: sox soxio: Failed reading `/var/spool/asterisk/monitor/20071113-171303-mycall.gsm': unknown file type `auto'. Any clue? maybe sox has changed and is not compatible with * 1.4.x? |
| 17:33.41 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:33.58 | tzafrir | dexteruk, svn switch http://svn.digium.com/svn/zaptel/trunk http://svn.digium.com/svn/zaptel/branches/1.4 #please don't use zaptel trunk |
| 17:34.25 | Qwell | tzafrir: the trunk url isn't needed there |
| 17:34.53 | nestAr | tip, polycom microbrowser hate <pre> |
| 17:34.54 | tzafrir | right. I didn't really test that line... |
| 17:35.06 | nestAr | probably why it's in the unsupported list. |
| 17:35.25 | [TK]D-Fender | nestAr: U R SMRT |
| 17:35.45 | Qwell | why the heck is <pre/> not supported? |
| 17:36.01 | moemoe | anyone here who has asterisk with hfc running on netbsd? i can only find ftp://ftp.netbsd.org/pub/NetBSD/packages/pkgsrc/comms/zaptel-netbsd/README.html, but this is really outdated. and the pkgsrc asterisk comes neither with zaptel nor with chan_capi |
| 17:36.05 | Qwell | and what's up with the non-conforming syntax it requires? |
| 17:36.16 | *** join/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 17:36.37 | *** join/#asterisk lemanal (n=lemanal@wifi-233-27.sc07.org) |
| 17:38.20 | nestAr | well, <br /> is also in the unsupported list, but it works.. so who knows. |
| 17:38.51 | Qwell | probably can't count on the behavior of it then |
| 17:38.59 | Qwell | like in certain elements maybe |
| 17:39.46 | nestAr | dunno, the page was simple enough that pre was perfect, i don't have a whole lot of elements. |
| 17:39.54 | *** join/#asterisk axscode (n=axscode@134.84.48.60.klj04-home.tm.net.my) |
| 17:39.54 | nestAr | just wanted to spit out some text, more or less. |
| 17:40.41 | *** join/#asterisk crichardson (n=crichard@38.113.5.185) |
| 17:41.56 | crichardson | hi guys i have a question i dont know much about t1s but i was woundering what kind of hardware do i need for a mixed t offering both data and voice? and how do i break out the data back to the network ? |
| 17:44.21 | nestAr | my phone company does it for, not really sure how they do it.. |
| 17:44.32 | nestAr | i think the newer digium cards support what you want |
| 17:44.45 | Bladerunner05 | I compile asterisk 1.4.13 with spandsp (latest version), when I receive a fax after: Executing [fax@default:2] RxFAX("Zap/1-1","xxxx") in new stack appears: Segmentation fault (core dumped) and asterisk crash |
| 17:44.47 | nestAr | throws the internet channels to the kernel |
| 17:44.51 | ManxPower | crichardson: any one of a number of devices. We use an Adtran TSU 120 for that sort of thing, but you can do it all in zaptel or use some form of IAD like a VINA or a channel bank |
| 17:45.25 | ManxPower | as nestAr said, the telco can do it for you too |
| 17:46.13 | crichardson | if the telco did it would they just supply 2 ether drops from the dmarc that were broken out to data and voice? |
| 17:46.47 | *** join/#asterisk zerohalo (n=zeroHalo@pool-71-255-162-167.bstnma.east.verizon.net) |
| 17:46.52 | nestAr | that's how mine did it.. i plug my network and the digium card into the same adtran box |
| 17:47.14 | ManxPower | crichardson: no, they would supply one network connection (ethernet or V.35) for data and a DXS-1 (T-1) port for voice. |
| 17:47.31 | nestAr | ManxPower is correct |
| 17:47.35 | ManxPower | nestAr: your telco handed you voice over ethernet?? 8-) |
| 17:47.57 | nestAr | i think he was saying ether, speaking of the cable, not the transport |
| 17:48.02 | ManxPower | actually the network connection could be DSX-1 or ethernet or V.35 |
| 17:48.22 | ManxPower | nestAr: if he's calling twisted pair "ethernet" he is far, far beyond our help. |
| 17:48.36 | crichardson | hehe |
| 17:48.49 | *** join/#asterisk s1d (n=s1d_@62.244.178.194) |
| 17:49.09 | s1d | he damn it, is there is agi error still showing up on http://www.dongs.dk rather then content |
| 17:49.11 | crichardson | sorry i have only dealt with fiber i didnt think people still used ts anymore :o |
| 17:49.19 | *** join/#asterisk theo_ (n=theo@c-67-166-100-135.hsd1.ut.comcast.net) |
| 17:49.43 | ManxPower | Personally I like the voice to be DXS-1 (channelizecd T-1) and the data to be either DSX-1 or V.35. If they hand you data as ethernet then your router has no idea when the line goes down,. as the ethernet link will still be present regardless of the status of the T-1. |
| 17:49.44 | nestAr | bored. |
| 17:50.09 | *** join/#asterisk muiro (n=muiro@cpe-76-189-84-108.neo.res.rr.com) |
| 17:50.24 | nestAr | ManxPower: true. actually, my telco is handing me the voice portion as a PRI, data as ethernet |
| 17:50.26 | *** part/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 17:50.40 | *** join/#asterisk klauwhamer (n=felixdhc@ipd50af070.speed.planet.nl) |
| 17:51.07 | s1d | anyone test the dongs.dk i'm stuck at a D.C. |
| 17:51.12 | ManxPower | nestAr: *nod* so you have no idea when the data goes down except for the fact your packets are not getting to/from where they are supposed to be. |
| 17:51.35 | De_Mon | hmm |
| 17:51.53 | nestAr | indeed. |
| 17:51.57 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:53.42 | Bladerunner05 | I compile asterisk 1.4.13 with spandsp (latest version), when I receive a fax after: Executing [fax@default:2] RxFAX("Zap/1-1","xxxx") in new stack appears: Segmentation fault (core dumped) and asterisk crash |
| 17:53.51 | Bladerunner05 | how can I check what cause that error ' |
| 17:53.52 | Bladerunner05 | ? |
| 17:53.57 | nestAr | i considered going the zaptel route, but i didn't want to bite off more than i could chew. |
| 17:54.01 | ManxPower | nestAr: I'm a big fan of Adtran TSU 120s, as they are modular and so you can hand off the different pieces in any format you want to. |
| 17:54.15 | s1d | <PROTECTED> |
| 17:54.18 | ManxPower | nestAr: I'm a rebel -- PCs are not routers. routers are routers. |
| 17:54.35 | ManxPower | s1d: exactly how does this apply to Asterisk? |
| 17:54.44 | nestAr | i think s1d is a spammer |
| 17:54.47 | s1d | no.. |
| 17:54.52 | s1d | it's my asterisk gateway |
| 17:54.56 | ManxPower | are you needing asterisk help or are you just cluttering up the channel with off topic crap? |
| 17:55.05 | nestAr | tastes like spam. yum. |
| 17:55.20 | *** part/#asterisk s1d (n=s1d_@62.244.178.194) |
| 17:55.27 | nestAr | heh |
| 17:55.48 | nestAr | yay, penis. man, it was hard to guess what that was going to be. |
| 17:59.05 | tzanger | so where's that kb when you need one |
| 18:00.24 | lirakis | i have an agent who is "stuck" on a call... and cant log in or out. They are not actually on a call, but asterisk shows them as busy... which is why i think i cant log them off in any way |
| 18:00.48 | ManxPower | lirakis: did they recently TRANSFER a call? |
| 18:00.51 | lirakis | any one know how to resolve this... i have tried hanging up the channels that are associated with the agents exten. |
| 18:01.01 | lirakis | ManxPower: i dont know... |
| 18:01.11 | De_Mon | I just parked someone on a parking extension already in use, and * just hung up on them. |
| 18:01.28 | lirakis | ManxPower: ive been trying to get them logged out for like 10 min though |
| 18:01.50 | ManxPower | at least at one point there was an issue where queue would consider the agent "on a call" if they accepted a call, then transfered the call. the agent would become abvailable when the original call/channel ended |
| 18:02.17 | ManxPower | De_Mon: huh? parking does not allow you to select where the call gets parked at |
| 18:02.34 | *** join/#asterisk s1d (n=s1d_@62.244.178.194) |
| 18:02.42 | De_Mon | ManxPower Set(PARKINGEXTEN=anumber) |
| 18:03.08 | ManxPower | De_Mon: Ah. an you also noticed that you have to handle parking failures in your dialplan. |
| 18:03.21 | lirakis | .. so no other suggestions |
| 18:03.38 | De_Mon | ManxPower I tried adding h,1 that redirects the call, but its too late they are already hungup with h is processed |
| 18:03.43 | lirakis | on how to "kill" this call and or get the agent logged of... i think it is a "phantom" call really |
| 18:03.53 | De_Mon | and adding another extension after Park wasn't processed either! |
| 18:04.24 | ManxPower | De_Mon: no, you handle it as the next priority in your dialplan, Did you read "core show application park"? |
| 18:04.31 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 18:04.51 | De_Mon | see the line right above yours, I'll try it again though... |
| 18:05.24 | De_Mon | I do see that last line of the description this time, maybe I found a bug ;) |
| 18:05.42 | ManxPower | (1.4 only) If you set the PARKINGEXTEN variable to an extension in your parking context, park() will park the call on that extension, unless it already exists. In that case, execution will continue at next priority. |
| 18:05.54 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.172) |
| 18:06.00 | ManxPower | in 1.2 you can't fo PARKINGEXTEN |
| 18:06.01 | *** mode/#asterisk [+b *!*n=s1d_@62.244.178.*] by Qwell |
| 18:06.02 | *** kick/#asterisk [LOLdongs!i=north@pdpc/sponsor/digium/Qwell] by Qwell (Qwell) |
| 18:06.53 | De_Mon | ManxPower this is 1.4.13, and park hangs up instead of continuing if the extension is in use |
| 18:07.07 | De_Mon | lol dongs |
| 18:07.07 | ManxPower | De_Mon: sounds like it is time to report a bug. |
| 18:07.22 | ManxPower | thank you, Qwell |
| 18:07.36 | *** join/#asterisk Blackthorn (i=blacktho@76.77.160.10) |
| 18:07.38 | muiro | oh man, I love asterisk. I've just started learning and I love it. |
| 18:07.55 | [T]ank | having issues installing fglrx. I have all of the packages installed except for one dependency which is giving me grief. Could anyone assist? Here are a few details: http://pastebin.ca/772270 |
| 18:07.57 | Blackthorn | How do you set a cron job in ubuntu to reobot the asterisk server at 3 A.M. every day? |
| 18:08.07 | [T]ank | oops. wrong channel |
| 18:08.08 | [T]ank | ;-) |
| 18:08.11 | lirakis | Blackthorn: dude wrong chan. |
| 18:08.21 | ManxPower | Blackthorn: you are on the wrong channel |
| 18:08.37 | [T]ank | so is [T]ank |
| 18:08.46 | [T]ank | habit :-D |
| 18:08.50 | ManxPower | whois s1d |
| 18:09.39 | Qwell | oh, right |
| 18:09.40 | *** mode/#asterisk [+r] by Qwell |
| 18:09.42 | Qwell | duh |
| 18:10.53 | *** join/#asterisk dswillia (n=me2@199.3.247.34) |
| 18:12.48 | dswillia | hey all I have a sangoma 2 port pri card installed in my asterisk box, I have it configured and the pri plugged in with no alarms. When I place a incomming call to the box, I get a regular busy (no fast) is there a way I can watch and see if the call is comming into the asterisk server via the cli? |
| 18:13.00 | De_Mon | muiro welcome to the club |
| 18:13.14 | [TK]D-Fender | dswillia: "pri debug span 1" |
| 18:13.24 | Strom_M | ~reorder |
| 18:13.24 | jbot | rumour has it, reorder is what's commonly (and mistakenly) called "fast busy." Reorder tone indicates a problem with call completion; it doesn't indicate that the called party is busy. |
| 18:14.28 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 18:16.12 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 18:16.23 | nestAr | is there a key combo to reboot the ip550? the one for the 500 doesn't work. |
| 18:16.36 | Strom_M | menu 3 2 456 enter 3 yes |
| 18:17.13 | Strom_M | i've done that way too many times :) |
| 18:17.41 | dswillia | I cannot issue that from the CLI says its unknown |
| 18:19.12 | *** join/#asterisk hmmhesays (n=Neg@24-119-176-74.cpe.cableone.net) |
| 18:19.18 | nestAr | lol, i yeah, i know that |
| 18:19.40 | nestAr | i was just looking for the key combo so i didn't have to explain all that to the users. ;) |
| 18:20.42 | *** join/#asterisk davidcsi (n=davidcsi@180.Red-213-97-249.staticIP.rima-tde.net) |
| 18:20.57 | davidcsi | hello all, i'm trying to compile asterisk-oh323 |
| 18:21.15 | davidcsi | and get lots of error, although openh323 and pwlib compiled fine |
| 18:21.34 | davidcsi | seems like the compiler can't find any file |
| 18:22.37 | ManxPower | or hold down 468* wait for prompt and beep (2 - 3 seconds), enter password 456, select enter |
| 18:23.05 | *** join/#asterisk jsmith (n=jsmith@68.246.137.171) |
| 18:23.05 | *** mode/#asterisk [+o jsmith] by ChanServ |
| 18:25.02 | davidcsi | openh323flags.mak:2: /openh323u.mak: No such file or directory |
| 18:25.28 | [TK]D-Fender | nestAr: Pull the plug :) |
| 18:26.21 | *** join/#asterisk Aughey (n=jha@64.219.54.125) |
| 18:30.19 | Aughey | I'm having issues with incoming DTMF on a sangoma FXO card. Most of the time it work, but several times a day we either get double DTMF digits recognized (press 2 receives 2 2) or missed DTMF digits |
| 18:30.32 | Aughey | I've tried the relaxdtmf=ye value, and that doesn't seem to help |
| 18:30.58 | *** part/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 18:31.21 | ManxPower | Aughey: remove relaxdtmf and use rxgain for those channels. some value between 4 and -4 should work. try 4, 2, -2, and -4 and see which one improves the issue |
| 18:31.47 | ManxPower | relaxdtmf can CAUSE the issue you are seeing. other things can also cause it. |
| 18:32.08 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 18:32.20 | coppice | if you are using crappy phones, a recent fix the the DTMF code may be important for you |
| 18:32.40 | ManxPower | coppice: the calls are coming from the MSTN |
| 18:32.43 | ManxPower | andPSTN too |
| 18:32.46 | Aughey | I'm getting reports from many people using may different phones |
| 18:32.52 | Aughey | external dial-in |
| 18:32.59 | ManxPower | or I assume they are, as he has the issue on an analog FXO port. |
| 18:33.39 | ManxPower | Aughey: I know of at least one other person that experienced the same issue. He solved it by using rxgain=-2 for the problem channels |
| 18:33.53 | coppice | well, if the calls are coming from a crappy source, a recent fix to DTMF might be important. I don't know how far the propagated that fix |
| 18:34.09 | *** join/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 18:34.22 | Aughey | Ok, I'll give rxgain=-2 a try and see |
| 18:34.33 | Aughey | where can I get the DTMF fix? |
| 18:34.39 | *** part/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:34.51 | coppice | if the gain makes much difference something is horribly broken |
| 18:43.39 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 18:45.21 | *** join/#asterisk axscode (i=axscode@134.84.48.60.klj04-home.tm.net.my) |
| 18:46.04 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 18:49.59 | *** join/#asterisk bmg505 (n=leon@196.209.183.44) |
| 18:53.36 | *** join/#asterisk Tourinho (n=tourinho@cm-virtua-poa-C8B0F83B.dynamic.brdterra.com.br) |
| 18:54.09 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 18:54.15 | flujan | hi guys... hi [TK]D-Fender |
| 18:54.30 | flujan | yesterday I pasted a problem with segmentation faults and asterisk. |
| 18:54.41 | Tourinho | good morning guys, me again :) |
| 18:54.43 | flujan | today I started asterisk from a shell and here it is |
| 18:54.47 | flujan | glibc detected *** corrupted double-linked list |
| 18:54.49 | [TK]D-Fender | flujan: try to get * to crash in one piece, ok? |
| 18:55.08 | flujan | [TK]D-Fender: ehehe I have the core dumped... but todays it only shows this |
| 18:55.26 | *** join/#asterisk Kandinsky (n=Kandinsk@perla2.tm.ew.ro) |
| 18:57.45 | flujan | [TK]D-Fender: any ideas? |
| 18:57.55 | flujan | http://pastebin.com/m6db7b4f |
| 18:58.27 | [TK]D-Fender | flujan: no clue. I know nothing about * coding. |
| 19:01.45 | flujan | ok [TK]D-Fender thanks |
| 19:04.11 | ManxPower | coppice: I've found that if the audio level is too low, asterisk can miss DTMF digits or duplicate DTMF digits, same if it's too high. |
| 19:04.58 | ManxPower | (too high causing problems is uncommon in my experience, but it happened to me on at least 2 analog PSTN lines connected into Asterisk |
| 19:05.02 | coppice | the DTMF receiver should have a wide dynamic range. it should take a very quiet or overloaded signal to be outside the range. |
| 19:05.17 | ManxPower | fujin: that pastebin is NOT a backtrace |
| 19:05.33 | ManxPower | coppice: yes, but this is Asterisk we are talking about. |
| 19:05.33 | coppice | a lot of people run their asterisk systems in alsmost continuous clipped, with the gain at maximum |
| 19:06.21 | ManxPower | coppice: I think what happened was that the gain was so loud Asterisk took the dtmf echo (analog on both ends) to be actual dtmf |
| 19:06.57 | ManxPower | but that is a guess, all I know was playing with the gains fixed it (this was in pre-1.0 days) |
| 19:09.54 | ManxPower | it was really bizarre |
| 19:10.46 | *** join/#asterisk ghento (n=ghento@75.155.241.7) |
| 19:11.56 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 19:12.17 | coppice | yeah. the crappy V.23/Bell 202 modem in * doesn't have a problem with decoding echo, but my one is spandsp does. it requires some extra logic in things like land line SMS handling to avoid problems :-) |
| 19:18.55 | jameswf | anyone have anything good or bad to say about a dell pe 2950 or the intel x5000 chipset ? |
| 19:19.04 | *** join/#asterisk _ys (n=yuri@80.70.236.69) |
| 19:19.11 | jsmith | jameswf: The Dell PE2950 is a great box |
| 19:23.02 | *** join/#asterisk angom (n=angom@200.38.31.239.dsl.dyn.telnor.net) |
| 19:23.12 | Bladerunner05 | I compile asterisk 1.4.13 with spandsp (latest version), when I receive a fax after: Executing [fax@default:2] RxFAX("Zap/1-1","xxxx") in new stack appears: Segmentation fault (core dumped) and asterisk crash |
| 19:23.16 | *** join/#asterisk doodoodoo (n=piet@c1-67-9.rrba.isadsl.co.za) |
| 19:23.26 | doodoodoo | help please |
| 19:24.40 | doodoodoo | i have installed a new asterisk box to replace old one, using the exact configs as on old system, everything works but i have no sound when dialing voicemail |
| 19:25.17 | ManxPower | doodoodoo: common problem with ztdummy. don't load ztdummy |
| 19:25.29 | *** join/#asterisk TJNII (n=TJNII@209.234.89.237) |
| 19:25.30 | ManxPower | doodoodoo: if you were on the asterisk-users mailing list you would know this is a current topic. |
| 19:25.44 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.172) |
| 19:26.05 | De_Mon | Why on earth would I use gosubif instead of gotoif? |
| 19:26.32 | jsmith | doodoodoo: That would also happen if you had a T1 card in the system, but didn't have it configured. |
| 19:26.37 | ManxPower | De_Mon: goto does not save it's pplace in the dialplan and return to it |
| 19:26.38 | doodoodoo | ManxPower what do you mean by mailing list? |
| 19:26.44 | ManxPower | ~mailinglist |
| 19:26.45 | jbot | Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.htmm |
| 19:27.03 | De_Mon | gosubif goes to a label, not context,priority,extension |
| 19:27.08 | [T]ank | i am noticing something strange on my system. maybe i do not have a flag set correctly or something. In the dialing of a telephone number from my cell phone, i hear "you have reached a non working number". However from asterisk I do not hear anything at all. It just shows ringing. All other calls come through just fine. Any ideas? |
| 19:27.30 | *** join/#asterisk agx (n=badpengu@81-174-46-174.dynamic.ngi.it) |
| 19:27.40 | De_Mon | so, im stuck in the same extension might as well hardcode the entire subroutine |
| 19:27.43 | De_Mon | and use gotoif! |
| 19:28.37 | [TK]D-Fender | Wrong. |
| 19:28.38 | ManxPower | De_Mon: um, it can use either |
| 19:28.45 | ManxPower | a label is just a textual priority |
| 19:29.55 | *** join/#asterisk angom_h (n=angom@200.38.31.239.dsl.dyn.telnor.net) |
| 19:30.12 | *** join/#asterisk Assid (n=assid@unaffiliated/assid) |
| 19:30.13 | Assid | heya |
| 19:30.27 | Assid | anyone here have much experience with a linksys 3102? |
| 19:30.46 | De_Mon | eeh? a 'label' isn't something you define for a priority? |
| 19:30.48 | ManxPower | De_Mon: you might be right, but if you are, you can easily combine ExecIf with a Gosub |
| 19:30.52 | doodoodoo | @jsmith : using tdm 24xxp |
| 19:31.11 | Assid | im thinking of upgrading my firmware to start with. however site says to check version, but the damn device doesnt have a verison there |
| 19:31.16 | ManxPower | De_Mon: I SUSPECT the show application GosubIf is wrong. |
| 19:31.17 | De_Mon | ManxPower it it is jsut a label, I can gosubif to a label that does a real gosub, I'll try it I suppose |
| 19:31.46 | De_Mon | Does gotoif support jumping contexts and extensions? |
| 19:31.55 | doodoodoo | i can see in the cosole its playing the sounds but nothing comes trough on the hanset |
| 19:32.05 | De_Mon | I've always used it to jump to labels in the same extension |
| 19:32.16 | De_Mon | enuf talking time to get some answers |
| 19:32.51 | Kandinsky | anybody knows how to assign an MSN to a specific extension? |
| 19:33.00 | Kandinsky | an ISDN MSN |
| 19:34.03 | Tourinho | guys, is there a way to debug and discover why my agi script does not continue running after I execute (DIAL)? Sometime it does continue, sometimes doest |
| 19:34.21 | *** join/#asterisk angom (n=angom@200.38.31.239.dsl.dyn.telnor.net) |
| 19:35.10 | ManxPower | Kandinsky: you really can't do that, and there really is no need to do that. |
| 19:35.25 | ManxPower | Tourinho: execing Dial from an AGI is generally not a good idea. |
| 19:35.47 | Kandinsky | i have an asterisk server with 2 isdn BRI pci cards using hfc-s chipsets |
| 19:36.07 | Kandinsky | and i want to assign a MSN number to a specific sip account |
| 19:36.39 | tzafrir | Kandinsky, incoming calls from ISDN go to the extension number of the MSN |
| 19:36.57 | ManxPower | Kandinsky: Asterisk can't do that directly. When a sip phone makes a call to the PSTN via the BRI, there are actually TWO calls, SIP Phone -> Asterisk and another call Asterisk -> PSTN via a Zaptel channel |
| 19:37.08 | Kandinsky | yes |
| 19:37.09 | tzafrir | e.g: if the MSN is 1234567, then you can have extension 1234567 |
| 19:37.10 | Kandinsky | precisely |
| 19:37.14 | tzafrir | do there whatever you want |
| 19:37.16 | ManxPower | But you CAN make a specific phone dial out on a specific zap channel |
| 19:37.23 | Kandinsky | yes |
| 19:37.25 | *** part/#asterisk agx (n=badpengu@81-174-46-174.dynamic.ngi.it) |
| 19:37.36 | Tourinho | ManxPower: is there another way to originate a call from AGI? Im trying to write an application since last week.. but seens that AGI is not powerfull enougth :S |
| 19:37.36 | Kandinsky | but how do i specify a msn in the zaptel channel |
| 19:38.04 | [TK]D-Fender | Tourinho: What would make the decisions about placing this call? And why do you feel its too complex? |
| 19:38.08 | ManxPower | Kandinsky: that would be specified in wherever you put the MSN in a config file. |
| 19:38.10 | tzafrir | in zapata.conf you just specifiy to where calls will go (to which dialplan context) |
| 19:38.21 | Kandinsky | which config file? |
| 19:38.32 | ManxPower | tzafrir: I think he wants the opposite of what you are saying. |
| 19:38.36 | tzafrir | in extensions.conf (or whatever way you use to write your dialplan) you set that |
| 19:38.39 | ManxPower | Kandinsky: what config file do you specify the MSNs in? |
| 19:38.45 | tzafrir | set MSNs? |
| 19:38.54 | Kandinsky | none..right now..that's what i want top find out |
| 19:38.56 | ManxPower | misdn.conf,. visdn.conf, zapata.conf? |
| 19:38.58 | Kandinsky | yes |
| 19:39.04 | Kandinsky | tzafrir |
| 19:39.12 | Tourinho | [TK]D-Fender: Im trying to write a simple callingcard application, so I need to know the number that the caller wants to call. |
| 19:39.31 | Kandinsky | exten => _97.,1,Dial(Zap/g2/${EXTEN:1}) would always dial the first msn allocated to that zap channel |
| 19:39.32 | ManxPower | Tourinho: your question implies that you have incoming and outgoing calls working on your BRI. |
| 19:39.35 | Kandinsky | if u understand |
| 19:39.42 | [TK]D-Fender | Tourinho: "show application read" <- this is DIALPLAN stuff... |
| 19:39.47 | ManxPower | sorrry |
| 19:39.55 | ManxPower | Kandinsky: your question implies that you have incoming and outgoing calls working on your BRI. |
| 19:39.57 | Kandinsky | i want something like exten => _97.,1,Dial(Zap/g2-MSNnumberX /${EXTEN:1}) |
| 19:40.04 | doodoodoo | sorted ztcfg -vv not loaded on start up |
| 19:40.09 | ManxPower | Kandinsky: and I said you cannot do that. |
| 19:40.09 | Kandinsky | indeed i do |
| 19:40.16 | *** join/#asterisk Sweeper (i=sweeper@softcheese.net) |
| 19:40.18 | ManxPower | Kandinsky: ok, where did you configure your PBI? |
| 19:40.21 | doodoodoo | ManxPower how does one join the mailing list? |
| 19:40.21 | ManxPower | ..BRI? |
| 19:40.27 | Kandinsky | yes |
| 19:40.29 | Kandinsky | bri |
| 19:40.34 | *** join/#asterisk Assid (n=assid@unaffiliated/assid) |
| 19:40.35 | Kandinsky | 2 pci cards in the asterisk server |
| 19:40.38 | ManxPower | doodoodoo: the instructions are on lists.digium.conf |
| 19:40.46 | ManxPower | lists.digium.com |
| 19:40.57 | ManxPower | Kandinsky: what actual file did you set the MSN in to make your BRI work????????????????????????????????????????????????? |
| 19:41.07 | Kandinsky | asterisk is my pbi ... and i have 2 ISDN NT |
| 19:41.09 | doodoodoo | thx ManxPower |
| 19:41.12 | De_Mon | ManxPower you were right, goto, gotoif and gosubif all use label to mean [[context,]extension,]priority |
| 19:41.21 | Sweeper | hey, what's voip provider that's really good at LNP? |
| 19:41.26 | ManxPower | De_Mon: make a bug report |
| 19:41.45 | *** join/#asterisk saftsack (n=saftsack@pD9E05A8E.dip.t-dialin.net) |
| 19:41.45 | Kandinsky | zapata.conf |
| 19:41.54 | Tourinho | [TK]D-Fender: Im able to read DTMF input from user.. and dial to destination. The problem is that sometimes, after my agi calls DIAL, it doesnt returns to AGI. |
| 19:41.58 | ManxPower | ok, now what channel in zapata.conf is associated with which MSN? |
| 19:42.01 | Kandinsky | zapata.conf for the channels |
| 19:42.23 | ManxPower | channel 1 or channel 2? |
| 19:42.26 | Kandinsky | that's what I want to do!!! ..I don't have any msn associated to anything |
| 19:42.39 | Tourinho | [TK]D-Fender: Im just trying to findout a way to debug this situation and learn when it could happens |
| 19:42.44 | Kandinsky | i just use g1 or g2 in dial |
| 19:43.01 | ManxPower | Kandinsky: you could also use 1 or 2 in Dial |
| 19:43.02 | Kandinsky | and it will use my first ISDN NT or the other |
| 19:43.16 | *** join/#asterisk bmg505 (n=leon@196.209.183.44) |
| 19:43.17 | Kandinsky | but an NT has an s0 bus |
| 19:43.20 | Kandinsky | with 7 MSN |
| 19:43.21 | ManxPower | Kandinsky: what are your two MSNs?> |
| 19:43.29 | Kandinsky | understand? |
| 19:43.49 | ManxPower | Kandinsky: not really, but you will have to talk to an Asterisk + BRI expert and there are none of those in the USA. |
| 19:44.07 | Kandinsky | on each ISDN NT conected to a pci card through a s0 bus i can use 7 msns |
| 19:44.28 | Kandinsky | but how do i specify which msn within the bus to use? |
| 19:44.30 | ManxPower | Kandinsky: I suspect you cannot do what you want to do. |
| 19:44.42 | ManxPower | Kandinsky: you do that by taking to an expert in BRI |
| 19:45.04 | tzafrir | isn't setting the MSN is done by setting the caller ID or something? or am I totally confusing things? |
| 19:45.07 | ManxPower | Kandinsky: your extensive search of the asterisk mailinglists was not helpful? |
| 19:45.28 | Kandinsky | wait a minute |
| 19:45.53 | *** join/#asterisk Falle (n=falle@diana.falle.se) |
| 19:46.05 | ManxPower | tzafrir: what I don't understand is WHY he wants to do that. It doesn't accomplish anything useful as far as I can tell. |
| 19:46.11 | Kandinsky | i know i can override and give a specific ID number (the msn i would like) but will that work good? |
| 19:46.32 | ManxPower | Kandinsky: do you want to force the MSN or do you want to force the Caller*ID? |
| 19:46.32 | Kandinsky | when i make an outgoing call |
| 19:46.42 | Kandinsky | i would like to force the msn |
| 19:47.00 | ManxPower | Kandinsky: then you will have to talk to someone that is an expert in ISDN BRI. |
| 19:47.06 | Kandinsky | but i think the caller id works for the purpose |
| 19:47.08 | Kandinsky | right? |
| 19:47.10 | tzafrir | for proper billing with the telco and such? |
| 19:47.14 | ManxPower | As that is a BRI specific thing. |
| 19:47.14 | Kandinsky | yeah |
| 19:47.18 | ManxPower | It' |
| 19:47.19 | Kandinsky | that was the plan |
| 19:47.27 | Assid | hrm... anyone here played witha linksys 3102 ? |
| 19:47.31 | ManxPower | It's EASY to set the callerid, if your carrier permits that. |
| 19:47.35 | Assid | i need some helpgetting this pstn section |
| 19:47.55 | Kandinsky | ok...i'll make some tests to see if it works |
| 19:48.06 | Kandinsky | i found an example using capi.conf |
| 19:48.07 | ManxPower | just remember that +, 0 and 00 are NOT part of the callerid |
| 19:48.07 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 19:48.18 | Kandinsky | [ISDN1] |
| 19:48.18 | Kandinsky | isdenmode=msn |
| 19:48.18 | Kandinsky | msn = 435253 |
| 19:48.31 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 19:48.32 | Kandinsky | and i thought i could do that in zaptel.conf or something |
| 19:48.38 | Kandinsky | but i don't use capi |
| 19:48.42 | Kandinsky | just the ideea |
| 19:48.51 | Kandinsky | to specify the msn directly |
| 19:48.55 | ManxPower | Kandinsky: CAPI and Zaptel are utterly and totally different. |
| 19:48.58 | Kandinsky | in a user context |
| 19:49.06 | Kandinsky | didn't know that :) |
| 19:49.11 | ManxPower | now, since you refuse to follow my advice, I will have to put you on /ignore. |
| 19:49.29 | Kandinsky | lol....i don't refuse your advice |
| 19:49.40 | Kandinsky | just trying toexplain more |
| 19:49.48 | Kandinsky | i am a novice |
| 19:49.56 | ManxPower | Yes, you did. You have not tried to find a BRI expert, you have not even tried searching the mailing list archives. |
| 19:50.03 | ManxPower | ~mailinglist |
| 19:50.04 | jbot | Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.htmm |
| 19:50.14 | Tourinho | where can I found docs about agi? besides the book? |
| 19:50.17 | ManxPower | much better |
| 19:50.20 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 19:50.27 | ManxPower | Tourinho: what version of asterisk? |
| 19:51.42 | Tourinho | Asterisk 1.4.13 |
| 19:51.47 | ManxPower | Tourinho: Dial will jump out of your AGI and go to the "h" extension when one of the legs of the call hangs up. (I don't recall if it's the calling leg or the callee leg) |
| 19:52.02 | ManxPower | Which is why I said it is not a good idea to exec Dial from an AGI |
| 19:52.57 | Tourinho | ManxPower: thanks.. but it is weird that sometime AGI continues |
| 19:53.16 | ManxPower | Tourinho: it would depend on which side hangs up firsty. |
| 19:54.23 | Aces1up | i know this may be off topic, but i'm in the need of a 1-800 number, preferrably with virtual pbx functionality, the only thing i am concerned about is getting a number from just anyone as i have had problems in the past with not all calling parties being able to get a hold of me. |
| 19:54.40 | Aces1up | just wondering what i should look for in a provider to make sure my calls are reliable. |
| 19:55.00 | Aces1up | as well as what a reasonable rate i should be paying per-minute for a toll-free number. |
| 19:55.22 | Tourinho | ManxPower: what about asterisk manager? |
| 19:55.22 | Bladerunner05 | I compile asterisk 1.4.13 with spandsp (latest version), when I receive a fax after: Executing [fax@default:2] RxFAX("Zap/1-1","xxxx") in new stack appears: Segmentation fault (core dumped) and asterisk crash. any ideas? |
| 20:07.07 | *** join/#asterisk macros73_ (n=cs@c-24-3-246-27.hsd1.mn.comcast.net) |
| 20:07.11 | ManxPower | Bladerunner05: asking over and over won't get you an answer. What it will get you is 303 pissed off people, |
| 20:07.47 | Bladerunner05 | •ManxPower• sure |
| 20:08.48 | *** join/#asterisk Assid (n=assid@unaffiliated/assid) |
| 20:08.55 | Assid | stupid 3102 |
| 20:09.03 | Assid | i cant do squat with the fxo port |
| 20:09.04 | ManxPower | Google: Results 1 - 10 of about 45 from lists.digium.com for rxfax segfault spandsp AND "too lazy to use google" |
| 20:10.25 | Bladerunner05 | MaxPower: I do that and look around it without finding a way |
| 20:10.29 | muiro | likelihood of getting an outboard 33.3k baud us robotics sportser to play nicely with asterisk to do a one time demo IVR for a customer? |
| 20:10.51 | duki | sudo reboot |
| 20:14.38 | ManxPower | muiro: the chances are NONE AT ALL |
| 20:14.39 | *** join/#asterisk mace (n=mace@debian/developer/mace) |
| 20:14.43 | Tourinho | ManxPower: another weired thing just happened now.. * run AGI a little bit longer after DIAL ends :S |
| 20:15.19 | mace | i'd like to monitor calls that are held in a queue via a BLF; is this possible? |
| 20:15.35 | Tourinho | I was able to get variables values from asterisk, and issue an acc request, but it stops right after |
| 20:15.40 | ManxPower | muiro: Asterisk does not support modems as a voice interface. |
| 20:16.00 | muiro | ManxPower: thanks |
| 20:16.02 | ManxPower | Tourinho: Asterisk kills everything having to do with the call when the call ends. What do I have to say to make you listen |
| 20:17.33 | *** join/#asterisk Potato663 (n=[hpcw]ni@200.172.5.10) |
| 20:17.36 | Potato663 | WRAARRRR!!! I'm the Tomato Monstahhhhh! WRAARRRR!!! |
| 20:17.41 | Potato663 | WRAARRRR!!! I has the Cookies Tooo! WRAARRRR!!! |
| 20:17.49 | Potato663 | WRAARRRR!!! I'm the Tomato Monstahhhhh! WRAARRRR!!! |
| 20:18.09 | *** part/#asterisk Potato663 (n=[hpcw]ni@200.172.5.10) |
| 20:18.43 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 20:19.13 | outtolunc | he stole my tomato! stop him |
| 20:19.31 | outtolunc | is it friday yet? |
| 20:19.38 | *** join/#asterisk jetlagmk2 (i=jetlag@pool-70-106-89-254.hag.east.verizon.net) |
| 20:19.52 | muiro | it's still basically tuesday |
| 20:19.54 | rob0 | Every day is Friday for the True Believers. |
| 20:21.44 | *** join/#asterisk craigk (n=ckowald@58.174.122.198) |
| 20:22.32 | nestAr | how can i change where a parked call goes when it times out? |
| 20:23.21 | *** join/#asterisk DaneM (n=DaneM@ppp-209-77-228-246.dsl.chi2ca.pacbell.net) |
| 20:24.38 | *** join/#asterisk gardo (n=gardo@121.97.176.170) |
| 20:25.55 | DaneM | Hello, all. Does "Set(TIMEOUT(response)=15)" set it so that it will wait 15 seconds AFTER a subsequent Background command, or do you have to set it so that it's the length of the Background message PLUS 15 seconds? |
| 20:26.30 | *** join/#asterisk polerin (n=erin@c-71-228-222-87.hsd1.tn.comcast.net) |
| 20:27.11 | Tourinho | ManxPower: sorry, Im a bit desperated :) |
| 20:27.24 | nestAr | DaneM: just the length you want the timeout to be. |
| 20:27.28 | De_Mon | nestAr features.conf |
| 20:27.42 | DaneM | nestAr: after the message completes? |
| 20:28.11 | nestAr | DaneM: I think you're using it incorrectly. |
| 20:28.17 | nestAr | when the message ends, it ends.. |
| 20:28.31 | nestAr | if you want to exten, play some silence |
| 20:28.36 | nestAr | extend* |
| 20:28.41 | ManxPower | DaneM: it has nothing to do with Background, it has to do with WaitExen |
| 20:29.15 | DaneM | hmmmm...basically what I'm trying to do is give the caller 15 seconds after the Background message completes to enter an extension. |
| 20:29.23 | DaneM | which command should I use for that? |
| 20:29.28 | mace | .- |
| 20:29.32 | ManxPower | DaneM: you would put a WaitExten after the Background |
| 20:29.47 | DaneM | OK. thanks...I'll try that. |
| 20:30.25 | DaneM | btw, what is the Set(TIMEOUT(response)=x) used for then? I found the wiki page a bit confusing. |
| 20:30.44 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 20:31.11 | [TK]D-Fender | DaneM: that is how I would do it... |
| 20:31.32 | DaneM | ok. I was just wondering. |
| 20:33.35 | Bladerunner05 | how can I do to download asterisk via svn |
| 20:34.19 | *** join/#asterisk Rhinoo_ah (n=ahonea@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 20:34.31 | nestAr | De_Mon: i don't see anything in features.conf that gives me a option to change that.. can you give me a hint? |
| 20:34.35 | Assid | [Nov 13 15:33:44] WARNING[18517]: chan_sip.c:6460 determine_firstline_parts: Bad request protocol 22042420@ip.address.of.the.server:5060 SIP/2.0 |
| 20:34.40 | Assid | can someone help me with this? |
| 20:34.51 | Assid | im trying to use my spa 3102 |
| 20:36.32 | lirakis | Bladerunner05: google.. it will take < 30 to find |
| 20:36.37 | lirakis | *30sec |
| 20:36.46 | *** join/#asterisk dijungal (n=kdaniel@63.175.159.171) |
| 20:36.47 | *** join/#asterisk |dennis| (n=dennis@200.32.236.10) |
| 20:37.47 | *** join/#asterisk BBHoss (n=hoss@146.229.191.76) |
| 20:39.13 | fetcher | Is there a way to have Asterisk log SIP registration & expiration events (only) to a file? |
| 20:39.50 | fetcher | trying to troubleshoot some connectivity problems, dropped calls etc. that aren't reported until after the fact |
| 20:42.37 | Assid | anyone here using a spa3102 for pstn-voip gateway ? |
| 20:47.10 | DaneM | Thanks for your help with WaitExten(), all. That seems to have fixed my problem. :-) |
| 20:47.17 | muiro | fetcher: can't you just grep the registration events out of the normal log? |
| 20:47.48 | *** part/#asterisk DaneM (n=DaneM@ppp-209-77-228-246.dsl.chi2ca.pacbell.net) |
| 20:54.30 | Assid | okay seriously? anyone here got a spa3000 or spa3102 |
| 20:54.37 | Assid | the one with an fxo and fxs port |
| 20:55.09 | *** join/#asterisk moprilo (n=jjohn@sv-cpe-dynamic-190-53-14-251.amnetsal.com) |
| 20:56.13 | *** join/#asterisk syneus (n=syneus@host71-92-dynamic.17-79-r.retail.telecomitalia.it) |
| 20:57.54 | *** join/#asterisk dijungal (n=kdaniel@63.175.159.171) |
| 21:08.00 | *** join/#asterisk lemanal (n=lemanal@wifi-233-27.sc07.org) |
| 21:14.11 | *** join/#asterisk PC_Clone (n=pc_clone@c-76-16-79-154.hsd1.il.comcast.net) |
| 21:24.02 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 21:24.12 | PC_Clone | hi everyone....I was wondering if anyone has experienced the "Bridge stops bridging channels" and why * does that |
| 21:28.01 | fujin | to confuse you |
| 21:28.33 | PC_Clone | fujin: |
| 21:28.37 | PC_Clone | fujin: it's workign |
| 21:28.54 | PC_Clone | but the users have audio dropouts one way when that happens |
| 21:28.55 | fujin | I haven't experienced it personally |
| 21:29.03 | fujin | have you got canreinvite=yes? |
| 21:29.32 | [TK]D-Fender | heading home, BBIAB |
| 21:29.38 | PC_Clone | no |
| 21:29.42 | PC_Clone | canreinvite=no |
| 21:29.50 | fujin | then there's no reason for it to try bridging |
| 21:30.14 | *** join/#asterisk techie (n=techie@adsl-76-214-20-56.dsl.lsan03.sbcglobal.net) |
| 21:30.26 | *** join/#asterisk nullogic (n=nullogic@208.52.147.166) |
| 21:30.32 | PC_Clone | this is from Zap to SIP |
| 21:31.10 | PC_Clone | and it only happens on the WAN (Point to point) |
| 21:31.35 | PC_Clone | while the routers are telling me that bandwith isn't the problem |
| 21:34.58 | *** part/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 21:36.24 | J4k3 | PC_Clone: I had a similar problem, the ethernet interface on the router was not nway-handshaking with my cheapish switch |
| 21:36.48 | J4k3 | so the router was at 100/hdx, the switch was at 100/fdx, or vice versa, I forget |
| 21:36.58 | J4k3 | packets were lost and only in one direction |
| 21:37.04 | PC_Clone | hmmm |
| 21:38.12 | J4k3 | does it 100% drop out/stop/quit or is this a 'sound quality' issue? |
| 21:38.21 | J4k3 | mine was SQ, not a 100% failure. |
| 21:38.23 | PC_Clone | wouldn't I see errors on the routers ethernet interface? |
| 21:38.30 | J4k3 | I didn't, thankyoucisco. |
| 21:38.44 | PC_Clone | they say they hear "dead silence" |
| 21:38.54 | J4k3 | hrm, thats not going to be random packet loss then |
| 21:38.55 | J4k3 | ick |
| 21:38.57 | PC_Clone | for a few seconds |
| 21:39.15 | J4k3 | ahh, dead silence but it returns? |
| 21:39.21 | PC_Clone | ya |
| 21:39.36 | PC_Clone | and it's only audio TO them |
| 21:39.42 | PC_Clone | audio FROM them is still heard |
| 21:39.59 | J4k3 | hrm... that could be so much... I'd vote network issues but I also had the ghettoest of internal networks before I fixed (replaced switches, rewired bad/old cables, etc) it |
| 21:40.15 | J4k3 | yeah SIP is UDP, traffic in/out is pretty well seperate |
| 21:40.33 | J4k3 | its not like tcp where loss in one way will cause a 'stop' of the other direction's traffic. |
| 21:40.37 | PC_Clone | well, this is a remote office |
| 21:40.43 | PC_Clone | and they have a pretty bad network |
| 21:40.46 | PC_Clone | but the router |
| 21:40.54 | PC_Clone | and switch are new |
| 21:41.00 | nestAr | J4k3: that HDX/FDX thing happens a lot, esp with Sun systems and Cisco switches |
| 21:41.12 | nestAr | used to have that problem all the time at my old job |
| 21:41.15 | PC_Clone | and this happens on phones plugged directly into the switch |
| 21:41.31 | J4k3 | nestAr: yep, and its darned annoying when it happens :) |
| 21:41.38 | nestAr | indeed. |
| 21:41.52 | J4k3 | PC_Clone: hrm... what kind of phones are these, and what kind of interfaces do they have? |
| 21:42.00 | PC_Clone | Polycom 550 |
| 21:42.03 | *** join/#asterisk funxion (n=x@63.214.236.169) |
| 21:42.09 | PC_Clone | 100 fdx |
| 21:42.46 | funxion | does anyone have any experience with xtradius and quintum? |
| 21:43.20 | J4k3 | powered by 803.3af? |
| 21:43.32 | PC_Clone | most, not all |
| 21:43.40 | PC_Clone | actually, most aren't |
| 21:43.44 | J4k3 | hrm... |
| 21:43.49 | PC_Clone | 2 yes, 3 no |
| 21:44.16 | J4k3 | well if its consistant across both ways, that isn't it |
| 21:44.19 | *** part/#asterisk zerohalo (n=zeroHalo@pool-71-255-162-167.bstnma.east.verizon.net) |
| 21:44.33 | J4k3 | I just wonder if their negotiating properly with your switch |
| 21:44.46 | J4k3 | are there any PCs on the same switch? |
| 21:45.01 | J4k3 | I'd try setting up some pings to see if anything goes goofy |
| 21:45.07 | J4k3 | have the PC ping the phone and the router |
| 21:45.27 | PC_Clone | I've been pinging from the other side of the wan with no issues |
| 21:45.48 | PC_Clone | except when I saturate the link and QoS drops me to hell |
| 21:45.58 | J4k3 | yeah, data circuits usually have decent error notification/sensing |
| 21:46.16 | J4k3 | ethernet stuff got too cheap, and therefore it got kinda cheezy |
| 21:47.00 | J4k3 | now a cheap switch is 5% of the price of a decent switch. |
| 21:47.09 | PC_Clone | well, I can ping from that router |
| 21:47.29 | J4k3 | ahh, set up a pretty rapid ping, if possible |
| 21:47.34 | J4k3 | at least 10 requests/second |
| 21:47.47 | J4k3 | thats how I finally got my problem to show its head |
| 21:47.58 | J4k3 | lots of pinging while talking on the phones |
| 21:48.09 | PC_Clone | one sec |
| 21:49.07 | PC_Clone | rapid ping on cisco? |
| 21:49.23 | nestAr | increasing packet size will often show a network problem as well |
| 21:49.35 | J4k3 | yep |
| 21:49.50 | PC_Clone | how do i rapid ping |
| 21:49.50 | J4k3 | PC_Clone: hrm... cisco routers piss me off these days :) |
| 21:50.14 | PC_Clone | nestAr: When I increase the size, the pings actually become more uniform in timing |
| 21:50.24 | PC_Clone | .1 ms dev |
| 21:51.40 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:52.03 | J4k3 | thats pretty stable then. |
| 21:52.17 | J4k3 | maybe [TK]D-Fender knows, he's the self-decreed official mouthpiece of polycom |
| 21:53.00 | PC_Clone | haha |
| 21:53.01 | [TK]D-Fender | J4k3, Now who's sounding bitter? :) |
| 21:53.29 | [TK]D-Fender | J4k3, and I've made no such claim either. Any more words you'd care to put in my mouth? |
| 21:54.13 | fujin | [TK]D-Fender: he can't afford Polycom, and insults everyone in here for recommending them |
| 21:54.17 | fujin | fail is fail, i guess. |
| 21:54.28 | J4k3 | I dunno, my phones work *shrug* |
| 21:54.31 | PC_Clone | interesting....just pinged a phone 200 times |
| 21:54.37 | J4k3 | PC_Clone's don't. |
| 21:54.48 | PC_Clone | ? |
| 21:54.54 | J4k3 | maybe I should tell him what you tell people to do... throw his phones away and buy something that works? |
| 21:55.23 | [TK]D-Fender | PC_Clone, what model, what issue? |
| 21:55.41 | fujin | ip550's |
| 21:55.46 | PC_Clone | 550....random dropped audio but only on the point to point |
| 21:55.53 | PC_Clone | and only to the phone |
| 21:56.03 | fujin | does phone<->phone across the LAN work fine? |
| 21:56.09 | PC_Clone | no |
| 21:56.10 | fujin | and you've got canreinvite=no applied globally, yes? |
| 21:56.15 | PC_Clone | yes |
| 21:56.21 | fujin | firewalls between phones and asterisk? |
| 21:56.24 | PC_Clone | no |
| 21:56.28 | [TK]D-Fender | PC_Clone, clarify "point to poit", codec used, and whats on the other end of the call, as well as the networking in between. |
| 21:56.30 | fujin | o_0 |
| 21:56.38 | fujin | PC_Clone: tried it with reinvite=yes? |
| 21:56.58 | fujin | s/reinvite/canreinvite/ |
| 21:57.03 | PC_Clone | [TK]D-Fender: codec is ulaw, zap (PRI) on the other end |
| 21:57.29 | [TK]D-Fender | PC_Clone, Ok, netowk path between the phone and * please. |
| 21:57.30 | J4k3 | PC_Clone: does the problem occur when talking to voicemail and other local * services? |
| 21:57.34 | PC_Clone | or even another polycom 550 but at the other end of the point to point (private t1) |
| 21:58.31 | PC_Clone | J4k3: afaik just extentsion calls |
| 21:59.36 | PC_Clone | [TK]D-Fender: *=192.168.1.5 gw 192.168.1.250 -> 192.168.2.250 -> 192.168.2.175-180=polycom phones |
| 22:00.27 | [TK]D-Fender | PC_Clone, got a route added to you * server for that? Also pastebin your route table, iptables, and sip.conf masking only passwords. |
| 22:01.07 | fujin | why have the phones and asterisk on different networks? |
| 22:01.08 | *** join/#asterisk __freedom__lover (n=eduardo@clipper.provale.com.br) |
| 22:01.09 | fujin | does not compute |
| 22:01.34 | fujin | seperate vlans or? |
| 22:01.42 | PC_Clone | fujin: the phones are on a private t1 to another office |
| 22:01.50 | *** join/#asterisk Darthclue (n=e054502@fw149.northside.isd.tenet.edu) |
| 22:01.56 | fujin | that doesn't make sense |
| 22:02.01 | J4k3 | woah, tenet.edu still exists?!?! |
| 22:02.08 | fujin | so how do they communicate with asterisk? |
| 22:02.09 | J4k3 | tenet was my first internet experience evar. |
| 22:02.12 | fujin | through thte t1 router? |
| 22:02.15 | PC_Clone | ya |
| 22:02.31 | fujin | any ACL's stopping traffic? |
| 22:02.40 | fujin | is it a one-to-one nat, or just routed? |
| 22:03.22 | PC_Clone | [TK]D-Fender: route http://pastebin.com/m3909071 |
| 22:03.26 | PC_Clone | routed |
| 22:03.57 | fujin | i dunno, try canreinvite=yes |
| 22:03.58 | *** join/#asterisk _pepo_ (n=c9eea608@190.10.187.20) |
| 22:03.58 | Darthclue | is transcoding from gsm to ulaw bad or just overhead? |
| 22:04.00 | fujin | see if happy stuff happens. |
| 22:04.06 | fujin | Darthclue: overhead, uneccesarry |
| 22:04.08 | _pepo_ | hi friends |
| 22:04.15 | fujin | if everything is ulaw, pre-transcode all your recorded stuff to ulaw |
| 22:04.22 | fujin | (all my moh, voicemail = alaw) |
| 22:04.25 | [TK]D-Fender | PC_Clone, what traffic do you pump over this link besides *? |
| 22:04.43 | PC_Clone | inet |
| 22:04.54 | Darthclue | is there a command to force ulaw encoding in cepstral? |
| 22:04.55 | _pepo_ | Do I can use the authentification of my extensions with LDAP? |
| 22:05.02 | fujin | what the crap is cepstral? |
| 22:05.11 | fujin | _pepo_: no |
| 22:05.12 | De_Mon | nestAr remind me what you're looking for |
| 22:05.12 | jsmith | fujin: Text to speech |
| 22:05.13 | [TK]D-Fender | PC_Clone, well its entirely probable that some random browsing cuts into your VoIP traffic... |
| 22:05.27 | PC_Clone | [TK]D-Fender: I'm running QoS on the cisco's |
| 22:05.31 | PC_Clone | dscp |
| 22:05.43 | fujin | PC_Clone: I generally don't route voip traffic, unless entirely necessary. shortest path to and from asterisk to the phones |
| 22:05.47 | fujin | always works best |
| 22:05.54 | fujin | and you can always do cos/tos on that switch fabric |
| 22:06.08 | [TK]D-Fender | PC_Clone, Ok, is your call dropped completely or does it just "wink" during a conversation? |
| 22:06.12 | Darthclue | and if not in cepstral, then in sox? |
| 22:06.15 | PC_Clone | fujin: these offices are 45 miles apart....I didn't have enough ethernet cable |
| 22:06.24 | fujin | wow, that's a dumb idea. |
| 22:06.26 | fujin | learn2plan. |
| 22:06.50 | PC_Clone | [TK]D-Fender: I guess wink.....drops audio for a sec or two or 5 |
| 22:06.59 | PC_Clone | but comes back |
| 22:07.09 | fujin | and this 45 miles, it's carried by the intertrons? |
| 22:07.12 | fujin | or like, ipsec |
| 22:07.18 | [TK]D-Fender | PC_Clone, Ok, then I'll leave it as a "neworking issue" with regards to QoS / your uplink. |
| 22:07.27 | fujin | Darthclue: yes, you can transcode from anything to ulaw with sox |
| 22:07.40 | [TK]D-Fender | PC_Clone, hate to say... if you want that solved I'll be you'll need a Cisco tech in... |
| 22:07.58 | PC_Clone | [TK]D-Fender: Weird, b/c the ciscos look fine (no drops) |
| 22:07.58 | [TK]D-Fender | PC_Clone, that kind of dropout can be attributed to packet loss or jitter compensation. |
| 22:08.35 | PC_Clone | fujin: it's not ipsec or anything.....it's a private t1 line |
| 22:08.42 | PC_Clone | think frame relay without the cloud |
| 22:08.59 | [TK]D-Fender | fujin, P2P T1. Just bridged through the telco |
| 22:09.06 | fujin | ugh |
| 22:09.07 | [TK]D-Fender | fujin, direct synch |
| 22:09.07 | fujin | do not want. |
| 22:09.22 | [TK]D-Fender | fujin, Yeah, you WOULD (if E10 wasn't available) |
| 22:09.49 | [TK]D-Fender | PC_Clone, what do they charge you for it? |
| 22:10.19 | PC_Clone | about 700/mo |
| 22:10.27 | PC_Clone | USD |
| 22:10.29 | [TK]D-Fender | PC_Clone, for BOTH sides I hope.... |
| 22:10.32 | PC_Clone | ya |
| 22:10.45 | [TK]D-Fender | PC_Clone, not "terrible"..... but meh... |
| 22:10.46 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 22:10.46 | PC_Clone | not 1400 total |
| 22:11.15 | PC_Clone | well...it also provides inet to the remote |
| 22:11.24 | PC_Clone | so that's about 400 a month oof |
| 22:11.52 | PC_Clone | so for an extra 300 you get a vpn without all the hassles and a full t1 for internal traffic |
| 22:12.21 | *** join/#asterisk stybba (n=stybba@190.10.0.136) |
| 22:17.57 | *** join/#asterisk grandpapadot (n=null@mail.heavylogic.com) |
| 22:18.35 | grandpapadot | Hi all. Are dialplan "hints" context sensitive? i.e., if I have a hint for 810 in one context, and another context have a hint for 810, will the phone watching 810 in the first context see hints from the second? |
| 22:22.06 | ManxPower | grandpapadot: the hint should either be in the same context as context=whatever is specified in sip.conf for that device, you can also have subscribecontext=whatever to override the contrext where the hint is. |
| 22:22.53 | ManxPower | so if your phone is in the bob context and the hint is in the tom context, then you would need sibscribecontext=tom, without subscribecontext, the hint would have to be in the bob context. |
| 22:27.59 | [TK]D-Fender | ManxPower, little too late for his impatience... |
| 22:29.04 | *** join/#asterisk callguy (n=callguy@pool-71-255-162-167.bstnma.east.verizon.net) |
| 22:29.21 | *** join/#asterisk macros73_ (n=cs@c-24-3-246-27.hsd1.pa.comcast.net) |
| 22:31.14 | __freedom__lover | \quit |
| 22:31.32 | *** part/#asterisk fskrotzki (n=fskrot@host198.textwise.com) |
| 22:34.41 | Assid | [TK]D-Fender: you ever played with a spa3102 ? |
| 22:36.20 | *** join/#asterisk gabiru (n=gabiru@213.37.159.28.dyn.user.ono.com) |
| 22:49.17 | *** join/#asterisk gabiru (n=gabiru@213.37.159.28.dyn.user.ono.com) |
| 22:57.39 | *** join/#asterisk AJaymn (i=AJaymn@71-82-218-158.dhcp.mdsn.wi.charter.com) |
| 23:00.18 | *** join/#asterisk gabiru (n=gabiru@213.37.159.28.dyn.user.ono.com) |
| 23:00.51 | *** join/#asterisk CrashHD (n=crashhd@67-107-9-130.starstream.net) |
| 23:02.30 | *** join/#asterisk CVirus (n=GoD@196.205.192.246) |
| 23:02.36 | CrashHD | Hello everyoen |
| 23:09.17 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 23:11.24 | *** join/#asterisk irule (n=irule@200.53.61.4) |
| 23:12.24 | *** join/#asterisk zapp-branigan (n=zapp-bra@140.23.220.87.dynamic.jazztel.es) |
| 23:20.53 | *** join/#asterisk corpcomp (n=corpcomp@125-236-174-245.broadband-telecom.global-gateway.net.nz) |
| 23:21.26 | corpcomp | I have a 1.2 server setup. I have setup a SIP trunk to my provider but itseems to be having a problem. In the CLI I get the following message "-- parse_srv: SRVmapped to host fep1.2talk.co.nz, port 5060" and I am not entirely sure why or what it means. Verbose is set to 6 |
| 23:25.04 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 23:26.17 | ManxPower | ~siptrunk |
| 23:26.18 | jbot | [siptrunk] Asterisk does not support SIP Trunks. Set trunk=no in sip.conf and then set up the device normally in sip.conf. |
| 23:26.33 | fujin | huh, what's trunk=? do |
| 23:26.53 | fujin | /usr/src/asterisk-1.4.11/configs# grep trunk sip.conf.sample |
| 23:26.56 | fujin | that's not even an option?? |
| 23:27.12 | ManxPower | fujin: don't worry about it. |
| 23:27.19 | fujin | 1.6? |
| 23:28.40 | CrashHD | what is a sip trunk? |
| 23:28.48 | fujin | lol. |
| 23:28.49 | Qwell | ~siptrunk |
| 23:28.49 | jbot | siptrunk is probably Asterisk does not support SIP Trunks. Set trunk=no in sip.conf and then set up the device normally in sip.conf. |
| 23:28.57 | CrashHD | :) |
| 23:29.03 | ManxPower | I set that to keep people from whining about "sip trunks" |
| 23:29.05 | CrashHD | it is something asterisk does not support |
| 23:29.06 | ManxPower | ~trunk |
| 23:29.07 | jbot | methinks trunk is is a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. |
| 23:29.11 | ManxPower | that is the real one |
| 23:29.30 | fujin | It's a marketing buzzword. |
| 23:29.38 | CrashHD | hah |
| 23:29.41 | fujin | I've heard lots of "sip trunk" from salesmen. |
| 23:29.41 | ManxPower | since there is no such thing as a sip trunk, asterisk can't support sip trunks |
| 23:29.41 | CrashHD | too funny |
| 23:29.54 | ManxPower | invalid config options are silently ignored. |
| 23:30.01 | CrashHD | got my laugh for the day |
| 23:30.13 | corpcomp | TY for your help. |
| 23:30.15 | *** part/#asterisk corpcomp (n=corpcomp@125-236-174-245.broadband-telecom.global-gateway.net.nz) |
| 23:30.18 | ManxPower | so you could put mymotherisacrackwhore=yes in sip.conf and asterisk would not complain about it |
| 23:30.49 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 23:31.07 | pigpen | anyone know if anyone has integrated OnQ with Asterisk? |
| 23:31.10 | nestAr | De_Mon: how to control where parked calls go when they time out. The documentation says that it'll call back the exten that it came from, but it tries to dial the Zap channel the parked call is on.. |
| 23:31.18 | pigpen | I have a hotel that is needing this as a requirement. |
| 23:31.47 | JT | ManxPower: probably due to asterisk not parsing config files properly |
| 23:31.56 | ManxPower | nestAr: the docs are totally wrong about parking timeouts |
| 23:32.14 | nestAr | ManxPower: i am jack's utter suprise |
| 23:32.15 | nestAr | lol |
| 23:32.34 | ManxPower | JT: It's a good thing actually. I have several options in voicemail.conf that an AGI script parses for voicemail notification phone numbes |
| 23:32.37 | nestAr | sorry fight club is on |
| 23:32.55 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 23:33.19 | ManxPower | notify=5551512 for example on the user's voicemail.conf entry |
| 23:33.55 | De_Mon | how do I use Set: in a callfile? Set: var=FOO works for 1 variable, what about more than 1 variable? |
| 23:34.17 | De_Mon | multiple Set:'s ? |
| 23:34.28 | Qwell | De_Mon: what happened when you tried that? |
| 23:34.39 | mosty | is it possible to reload g729 from the manager interface? if so then what privilege do i need? |
| 23:34.40 | De_Mon | I haven't tried anything yet I'm looking at wiki |
| 23:34.58 | De_Mon | it doesn't really give any examples |
| 23:36.22 | De_Mon | man, using all these cool featuers is a great way to find improvements in the documentation :) |
| 23:36.33 | De_Mon | *ways to improve* |
| 23:36.55 | De_Mon | Qwell whats the story with Park? Is it supposed to continue to the next priority or not? |
| 23:37.06 | Qwell | if you use PARKINGEXTEN, yes |
| 23:37.33 | De_Mon | whew! |
| 23:38.08 | fujin | De_Mon: the wiki is a REALLy bad source of information |
| 23:38.27 | ManxPower | mosty: there is no reason to "reload" g729 because g729 has no config options |
| 23:38.52 | fujin | mosty: reload codec_g729.so |
| 23:38.55 | mosty | manxpower: it's not loaded, i just want to load it without restarting asterisk |
| 23:38.57 | ManxPower | De_Mon: there was a bug fix with regards to parking made today |
| 23:39.07 | ManxPower | mosty: load codec_g729a.so |
| 23:39.25 | ManxPower | you want to LOAD it not RELOAD it. |
| 23:39.26 | JT | ManxPower: although the "surprise screwup" side of things isn't such a good thing ;) |
| 23:39.32 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 23:39.36 | mosty | it says "Unable to load module codec_g729a.so" |
| 23:39.47 | ManxPower | mosty: then I guess you have not purchased it from Digium. |
| 23:40.24 | mosty | ManxPower, we have several licences, that have been in use for a couple of years |
| 23:40.31 | Qwell | on that machine? |
| 23:40.31 | mosty | all on this box |
| 23:40.41 | Qwell | mosty: call support |
| 23:40.41 | fujin | so why's it unloaded? |
| 23:40.44 | ManxPower | If you DID download the codec and bought the license then it is either not installed or the filename is slightly different (maybe codec_g729.so) |
| 23:41.00 | mosty | it's unloaded because asterisk crashed, and we tracked it back to a problem with g729 |
| 23:41.14 | fujin | sounds terrible |
| 23:41.15 | ManxPower | it should be pretty obvious what the filename is by looking in /usr/lib/asterisk/modules |
| 23:41.56 | De_Mon | fujin do you have a BETTER source of information? show application doesn't always tell me everything I need to know (for example on call files?) |
| 23:42.21 | fujin | De_Mon: the documentation that comes with the source packages? (configs/*.sample, docs/*) |
| 23:42.34 | fujin | generally provides pretty reasonable examples |
| 23:42.49 | fujin | like the AEL hotdesking dynamic queue member system, I adapted mine from that one |
| 23:42.52 | mosty | ManxPower, the file is there, /usr/lib/asterisk/modules/codec_g729a.so |
| 23:43.11 | fujin | mosty: got it configured in modules.conf? |
| 23:43.22 | fujin | nm |
| 23:43.24 | fujin | shouldn't need it in there |
| 23:43.29 | mosty | fujin, yes. is there a way to force a reload of modules.conf just in case? |
| 23:43.50 | fujin | don't think so, no |
| 23:43.55 | fujin | why not just punt asterisk? |
| 23:43.57 | Qwell | dude.. call support |
| 23:43.59 | De_Mon | asterisk-1.4/doc$ vi callfiles.txt |
| 23:43.59 | fujin | stand up your other HA box, kill that one |
| 23:44.01 | De_Mon | doh |
| 23:44.13 | fujin | ^^ De_Mon :P |
| 23:44.15 | De_Mon | asterisk-1.4/doc/callfiles.txt says to use setvar |
| 23:44.21 | mosty | fujin, because we have lots of active calls, who are already pissed enough, heh |
| 23:44.40 | fujin | blame it on sola flares |
| 23:45.26 | De_Mon | Happily enough that document did mention that multiple setvar's can be used. So is it Setvar: or Set: for 1.4+ |
| 23:45.42 | ManxPower | mosty: FIRST, find the name of the actual file. look in /usr/lib/asterisk/modules. What is the file name? |
| 23:45.44 | Qwell | De_Mon: both work |
| 23:45.54 | fujin | setvar is just deprecated, is it not, Qwell? |
| 23:45.57 | De_Mon | Qwell neither is being depriciated? |
| 23:46.00 | Qwell | don't think so |
| 23:46.03 | Qwell | not sure |
| 23:46.17 | fujin | ah. I've used setvar happily throughout dialplan, but changed them all to Set cause it's prettier. |
| 23:46.22 | mosty | ManxPower, it's /usr/lib/asterisk/modules/codec_g729a.so |
| 23:46.28 | Qwell | Set is a lot prettier, heh |
| 23:46.31 | De_Mon | considering the SetVar() Application is depriciated I thought it was maybe setvar isn't depriciated either |
| 23:46.43 | Qwell | mosty: just call support, and be done with it |
| 23:46.55 | ManxPower | mosty: then you can do a load codec_g729.so in the asterisk CLI |
| 23:46.59 | JT | De_Mon: there is a sample.call included too |
| 23:47.10 | ManxPower | if that does not work, put down the keyboard and call digium support. |
| 23:47.29 | mosty | i don't have access to the CLI, just AMI |
| 23:47.34 | mosty | hmm ok |
| 23:47.45 | De_Mon | omg, you have made me feel like a noob! I HOPE YOUR HAPPY!!! |
| 23:47.48 | ManxPower | mosty: you have to have access to the CLI if you want to load or unload a module. |
| 23:47.58 | De_Mon | s/YOUR/YOU'RE/ |
| 23:48.11 | ManxPower | only a moron would come here asking for help and not have access to the CLI or a shell on the asterisk box |
| 23:48.12 | mosty | ManxPower, hmm ok, i'll try to get into the cli then |
| 23:48.31 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 23:48.42 | ManxPower | that's like bringing your car to a mechanic and taking the keys with you when you leave. |
| 23:48.48 | mosty | i have a shell on the asterisk box, but asterisk -r does not work, the asterisk.ctl file does not exist |
| 23:49.10 | ManxPower | mosty: if asterisk.ctl does not exist then asterisk is not running |
| 23:49.18 | ManxPower | and you have a seriously fucked up box |
| 23:49.30 | fujin | or permissions are wrong on /var/run/asterisk |
| 23:49.36 | fujin | (generally the latter.. :P) |
| 23:49.47 | ManxPower | fujin: or asterisk.conf points it somewhere else |
| 23:49.52 | fujin | indeed |
| 23:49.53 | mosty | ManxPower, it is definitely running. another admin ran asterisk as root (i have a root) shell, but i don't think they used the init script, so i'm not sure where it created the asterisk.ctl file |
| 23:49.56 | Qwell | which means it's running as asterisk, which means the perms on the module may be wrong also |
| 23:50.11 | ManxPower | fujin: it seems like he's set up the system to fail, doesn't it? |
| 23:50.15 | fujin | soudns like it |
| 23:50.48 | mosty | believe me, i did not put the box in this state, i'm just trying to recover from it |
| 23:51.03 | ManxPower | mosty: stop worrying about g279, you have way more serious issues. |
| 23:51.15 | De_Mon | Thats odd, I didn't get an email when the case was closed, just when it was assigned |
| 23:51.23 | mosty | asterisk is running as root, according to ps |
| 23:51.29 | mosty | i am also root |
| 23:51.42 | mosty | the g729 module is world readable and executable |
| 23:51.52 | ManxPower | mosty: if you can't access the CLI, your asterisk box is screwed up, regardless of if it's running or not. |
| 23:52.11 | ManxPower | mosty: the only way to load the codec is from the CLI or on startup. |
| 23:52.19 | mosty | ManxPower, i have AMI access, can't i use the AMI "Command" command? |
| 23:52.22 | ManxPower | if you can't get to the CLI then you can't load the codec. |
| 23:52.28 | Qwell | mosty: ...to do what? |
| 23:52.37 | Qwell | asterisk -rx "load module codec_g729a.so"? |
| 23:52.40 | ManxPower | mosty: no you cannot. |
| 23:52.56 | De_Mon | Qwell When I blindxfer a call to parking with PARKINGEXTEN, the parked person gets to hear where they were parked. Is a flag the right way to park them silently, or checking if PARKINGEXTEN exists be just as good? |
| 23:53.02 | mosty | Qwell, do load or reload codec_g729a.so |
| 23:53.19 | De_Mon | s!if!is! |
| 23:54.23 | De_Mon | whoops that wasnt the right substitution... ah well english not first words |
| 23:54.47 | fujin | GAH |
| 23:54.52 | fujin | DO NOT USE ! FOR A DELIMITERRRRR |
| 23:55.23 | ManxPower | fujin: when 1.6 comes out I believe | will be officially considered deprecated |
| 23:55.34 | Qwell | ! != | |
| 23:55.50 | fujin | o_0 |
| 23:55.56 | jameswf | | != ! |
| 23:56.01 | fujin | ! just breaks my brain |
| 23:56.03 | jameswf | eieio |
| 23:56.09 | ManxPower | != != = |
| 23:56.25 | ManxPower | bang equal is not equal to equal |
| 23:56.29 | jameswf | :(|) |
| 23:56.43 | Qwell | != == = |
| 23:58.11 | jameswf | || != !! ! .=! == !! |
| 23:58.26 | De_Mon | fujin how about s@this@that? |
| 23:58.46 | fujin | dnoooo |
| 23:58.50 | fujin | use a slash |
| 23:58.56 | De_Mon | s^this^that? |
| 23:59.02 | fujin | s/omg/die/ |
| 23:59.17 | De_Mon | if I uses slashes jbot will correct me! |
| 23:59.29 | ManxPower | I use ^ because that's what things like the M() option to dial uses to separate options. |
| 23:59.30 | fujin | jbot: don't do sed autocorrection aymore! dirty bot |
| 23:59.36 | jameswf | preg_replace .... |