| 00:00.06 | jameswf | s/poo/poop/ |
| 00:00.08 | De_Mon | have you tried using sithisithati? I thik it'd work too |
| 00:00.17 | fujin | ohmygod |
| 00:00.18 | fujin | donotwant |
| 00:00.23 | Qwell | ! ! ! |
| 00:00.28 | fujin | no, you'd have to escape the \i in 'this'. |
| 00:00.43 | Qwell | sbthisbthatb |
| 00:00.44 | Qwell | duh |
| 00:00.55 | jameswf | aeiou and sometimes Y |
| 00:01.14 | fujin | aj@gatekeeper:~$ echo poo | sed 'sipooipeei' |
| 00:01.15 | fujin | pee |
| 00:01.16 | fujin | oh my :\ |
| 00:02.15 | jameswf | linux is dirty |
| 00:04.23 | jameswf | unzip ; strip ; touch ; grep ; finger ; mount ; fsck ; more ; yes ; umount ; sleep; |
| 00:05.37 | jameswf | james@james-desktop:~$ man sex |
| 00:05.37 | jameswf | No manual entry for s |
| 00:06.30 | TJNII | My asterisk server is no longer behind nat. (Whooo!) |
| 00:08.24 | Corydon76-dig | "No manual entry for virgin in section sex" |
| 00:09.23 | saftsack | hi is someone here who lives in germany? |
| 00:10.22 | jameswf | saftsack, where do you live |
| 00:10.34 | saftsack | germany ... |
| 00:10.47 | jameswf | saftsack, then the answer is yes... you |
| 00:11.28 | saftsack | do you feel luck if youre acting like a nitpicker? |
| 00:12.39 | saftsack | nevertheless is someone else here who is out of germany? |
| 00:12.55 | TJNII | "When designing finger puppets, it's critical to only portray them from the waist up, because as soon as you put legs on them it looks like you're giving your diminutive new friend one hell of a ride." |
| 00:13.13 | nestAr | vitelity's support staff is only a little better than brain dead. |
| 00:14.54 | jameswf | saftsack, http://voip-info.org/wiki/view/Asterisk+consultants |
| 00:16.41 | saftsack | ok the germans are in theirs beds now ... gn8 |
| 00:16.42 | *** part/#asterisk saftsack (n=saftsack@pD9E05A8E.dip.t-dialin.net) |
| 00:17.47 | *** join/#asterisk mindCrime (i=chatzill@nat/redhat/x-7783a2b538b3d96f) |
| 00:22.49 | *** join/#asterisk coppice (n=chatzill@102.204.17.210.dyn.pacific.net.hk) |
| 00:23.14 | *** join/#asterisk Bawbatos (n=Bawbatos@67.109.14.227.ptr.us.xo.net) |
| 00:24.42 | *** join/#asterisk macros73_ (n=cs@c-24-3-246-27.hsd1.mn.comcast.net) |
| 00:25.31 | Bawbatos | dumb question, but i set this up 6 months ago and have not looked at it since. how do i set the outbound id on a analog channel. someone was trying to send a fax internal to test and while the lines have exts the id that shows up it asterisk |
| 00:26.13 | tzanger | sigh |
| 00:26.22 | tzanger | I see a thousand pics of openmoko but nothing showing how thick it is |
| 00:26.54 | tzanger | unfortnately some of the shots I see show it fairly thick |
| 00:27.04 | coppice | i think if you care about that, its not for you :-) |
| 00:27.12 | tzanger | howso? |
| 00:27.15 | *** part/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 00:27.43 | tzanger | I want a totally open phone, from bootloader to dialer app, but I also don't want a damn brick |
| 00:27.43 | coppice | it has "early adopter" wriitten all over it. they even say you shouldn't try to make phone calls with it :-) |
| 00:27.50 | tzanger | I'm not buying it *now* |
| 00:27.57 | tzanger | I'm looking at it for the first commercial release |
| 00:28.10 | coppice | then you probably want the 2009 model from google |
| 00:28.22 | tzanger | no, I have a feeling the google offering will ultimately fail |
| 00:28.35 | Bawbatos | google will not be offering any hardware! |
| 00:28.38 | coppice | i'm damned certain openmoko will |
| 00:29.02 | tzanger | coppice: I agree, I just want the damn open phone |
| 00:29.29 | coppice | I suspect one or two key people, like HTC, are desperate to make google work |
| 00:29.45 | tzanger | why does HTC care? |
| 00:29.55 | tzanger | they sell WM6 already and do very well |
| 00:29.58 | tzanger | what will google offer them? |
| 00:30.11 | fujin | google maps runs perfect on my wm6 phone |
| 00:30.15 | tzanger | N800 would be right if it had a damn radio in it |
| 00:30.15 | fujin | that's about as much google as I need. |
| 00:30.22 | coppice | because everyone says "lovely hardware, but the software sucks". at some point they need a way out |
| 00:30.34 | tzanger | coppice: nah, everyone "loves" WM6 |
| 00:30.36 | tzanger | or is it 5 |
| 00:30.37 | tzanger | I can't remember |
| 00:30.38 | ai-a[afk] | Bawbatos: http://www.voip-info.org/wiki/view/Setting+Callerid |
| 00:30.46 | fujin | 5 is bad, 6 is nice. |
| 00:30.54 | tzanger | and if they aren't a WM fanboy then they're a fucking crackberry head |
| 00:30.54 | fujin | my tytn runs 6 quite happily, bosses tytn II is pretty smooth. |
| 00:31.10 | coppice | yeah, they absolutely adore buying a box so bare they have to spend a fortune filling in all the holes |
| 00:32.19 | coppice | they don't even pu a worthwhile calculator in WM6. most of the buyers are people like salesmen who really want a decent calculator. it pisses them off to spend $29.95 just to get a simple function |
| 00:32.32 | tzanger | I dunno they seem to have it all bottled up |
| 00:32.41 | tzanger | I liked palm before they totally fucked themselves |
| 00:32.57 | tzanger | I now run a TX and an L6 |
| 00:33.23 | coppice | i know quite a few HTC users who have dumped them in the last few months because of WinCE |
| 00:33.32 | tzanger | hmm I dunno |
| 00:33.43 | tzanger | if they're willing to put android on, why not oe? |
| 00:33.55 | tzanger | I don't see the carriers accepting android |
| 00:34.10 | tzanger | they've spent a fortune locking you in, android is a direct opposite in that regard |
| 00:34.12 | fujin | coppice: there are many free scientific calcs for wm6 |
| 00:34.18 | fujin | nearly all custom firmware comes with it |
| 00:34.26 | fujin | woudln't catch me dead running the stock TyTN firmware :\ |
| 00:34.34 | tzanger | fujin: again, it's hacking the damned firmware |
| 00:34.37 | tzanger | I'm sick of that |
| 00:34.39 | fujin | that's half the fun |
| 00:34.54 | coppice | for you, but that's a niche thing |
| 00:34.58 | fujin | indeed |
| 00:35.03 | fujin | but, it does what I want :) |
| 00:35.09 | coppice | HTC need something as acceptable as the iPhone |
| 00:35.17 | fujin | all of the other employees of this company run my custom firmware, and are happy |
| 00:35.17 | fujin | xD |
| 00:35.18 | tzanger | fujin: I think there is more fun in actually making it better than screwing about trying ot get to the fun part |
| 00:35.25 | fujin | oh, of course |
| 00:35.34 | fujin | I'd much rather take a product which has similar power/functionality and customisability |
| 00:35.35 | fujin | but, there isn't one |
| 00:35.38 | fujin | don't even suggest symbian |
| 00:35.45 | tzanger | no I wouldn't |
| 00:35.48 | Bawbatos | ai-a[afk] maybe i am being brain dead but i do not see that i set that up on my internal ip phone context and the ext comes up as the correct thing. is the analog bit different |
| 00:38.09 | coppice | the mobile platforms are all rather sad. although Linux is the second biggest behind symbian, they linux phones are all locked down so its irrelevant what they run |
| 00:39.34 | ai-a[afk] | iPhone sucks. |
| 00:39.53 | Bawbatos | why? |
| 00:39.57 | ai-a[afk] | useless camera, no 3g, expensive.. |
| 00:40.12 | tzanger | I've heard good things about the razr2 v8 |
| 00:40.15 | ai-a[afk] | oh wait,, you can browse the internet, and detects you moving it around 90degrees,, so useful. |
| 00:40.21 | Bawbatos | hum, best browser, just works, ssh, vnc, apache, mt-daap |
| 00:40.23 | tzanger | and have a contact with one of their linux devs, but I have yet to see wha thte phone'll do |
| 00:40.33 | ai-a[afk] | who needs multi-touch display that small ? |
| 00:40.49 | ai-a[afk] | Bawbatos: for that price you can buy a laptop you fool. |
| 00:40.50 | Bawbatos | 3g would be nice however having owned a ton of smart phones it is the best one ever made. |
| 00:40.57 | Bawbatos | i have a laptop |
| 00:41.05 | ai-a[afk] | Bawbatos: then you ssh from your phone ? |
| 00:41.09 | ai-a[afk] | your using it now ? |
| 00:41.11 | Bawbatos | sometimes |
| 00:41.13 | Bawbatos | no but i could. |
| 00:41.15 | ai-a[afk] | "sometimes" haha. |
| 00:41.19 | Bawbatos | i have colloquy on it |
| 00:41.20 | ai-a[afk] | "OH, I could"... but i dont.. |
| 00:41.25 | Bawbatos | i have and do |
| 00:41.29 | ai-a[afk] | why dont i ?? because the keyboard is pointless. |
| 00:41.31 | Bawbatos | when i have network issue |
| 00:41.32 | ai-a[afk] | Hah, a joke. |
| 00:41.38 | Bawbatos | i type just fine on it. |
| 00:41.43 | ai-a[afk] | "But i could,, if i wanted to " |
| 00:41.47 | ai-a[afk] | If so,, use it. |
| 00:41.51 | ai-a[afk] | leave your laptop at home. |
| 00:41.54 | ai-a[afk] | ... hmmm, cant can you.. |
| 00:42.01 | ai-a[afk] | so.. its worthles. |
| 00:42.09 | zapp-branigan | hi, i have a problem Unable to create channel of type 'Local' (cause 0 - Unknown) |
| 00:42.10 | Bawbatos | it is not a laptop replacement. and on my last 3 trips i did leave it at home. |
| 00:42.18 | zapp-branigan | what is the problem ? |
| 00:42.23 | ai-a[afk] | why have a device that duplicates what your other device does at less quality.,. and pay so much for it ? |
| 00:42.52 | ai-a[afk] | Bawbatos: do you have a ipod ? |
| 00:42.53 | Bawbatos | did not cost alot as far as i can see. it was bout the same as a new treo when it came out. |
| 00:42.56 | Bawbatos | no. |
| 00:42.59 | ai-a[afk] | why not? |
| 00:43.04 | Bawbatos | i dumped it when i got the phone |
| 00:43.16 | ai-a[afk] | your laptop plays music,,, your iphone plays music,, why not get a 3rd device,, just incase. |
| 00:43.38 | ai-a[afk] | hey, why not carry 5 devices that have wifi... just to show off... "Because I Could.. If i wanted" |
| 00:43.41 | Bawbatos | i like less stuff. i need a pda. i need a light email client so i do not have to open my laptop all the time. |
| 00:43.42 | ai-a[afk] | .. but i dont need to.. |
| 00:43.49 | ai-a[afk] | Bawbatos: less stuff = 1 laptop. |
| 00:44.07 | Bawbatos | i cannot open my laptop, pull up a phone number in 2 secs. |
| 00:44.11 | ai-a[afk] | Bawbatos: how long does your iphone battery last ? |
| 00:44.16 | Bawbatos | all day |
| 00:44.23 | ai-a[afk] | all day when its active ? |
| 00:44.31 | De_Mon | oh brother |
| 00:44.31 | ai-a[afk] | or all day on standby. |
| 00:44.35 | Bawbatos | never had it run out on me yet. |
| 00:44.37 | ai-a[afk] | its a phone. |
| 00:44.39 | Bawbatos | it is a tool |
| 00:44.40 | De_Mon | "all day with normal use' |
| 00:44.57 | De_Mon | ai-a[afk] how long does a bloody laptop battery last when its active? |
| 00:45.04 | Bawbatos | it does what i need it to do. heck i fixed a major network issue from dinner a few weeks ago from my phone |
| 00:45.15 | Bawbatos | and i would have had to drive home, get my computer, etc.. |
| 00:45.23 | ai-a[afk] | De_Mon: but he doesnt use the SSH, wifi, browser.... |
| 00:45.23 | Bawbatos | so works for me. |
| 00:45.24 | ai-a[afk] | why not ? |
| 00:45.48 | ai-a[afk] | Bawbatos: sorry, i dont believe you. |
| 00:45.56 | De_Mon | bwahaha |
| 00:45.57 | Bawbatos | hum, do not really care. |
| 00:45.58 | ai-a[afk] | i know guys who have it,, and now.. its just a phone. |
| 00:46.26 | ai-a[afk] | its a gimmick gadget that you just want.. |
| 00:46.31 | De_Mon | I ssh and remotedesktop from my blackberry all the time |
| 00:46.45 | ai-a[afk] | then 5 weeks after showing your friends and family.. its sat on the side ringing like any other phone. |
| 00:46.48 | De_Mon | okay, once a month. but still! it beats turning on a laptop |
| 00:46.53 | Bawbatos | when i reinstalled my work laptop the other day i did not even both to setup my person email account cuz they are on the phone. |
| 00:46.55 | ai-a[afk] | Once a month Hehe |
| 00:47.07 | ai-a[afk] | De_Mon: if its for SSH / rdp, then use it.. |
| 00:47.11 | ai-a[afk] | why use the laptop ? |
| 00:47.17 | ai-a[afk] | they say its used for browsing,, use it. |
| 00:47.21 | ai-a[afk] | you dont... |
| 00:47.27 | ai-a[afk] | so you gotta ask youself why you dont. |
| 00:47.31 | ai-a[afk] | ..because.. its a PHONE. |
| 00:47.44 | De_Mon | ai-a[afk] you keep using the word 'you' and you don't even know me! |
| 00:47.53 | Bawbatos | it is not met to be a laptop replacement. it is a tool for when you do not have that. and its is a PDA which i need very much. i have had a PDA since the 1st palm was USR |
| 00:47.54 | ai-a[afk] | your the one saying you use it. |
| 00:48.17 | De_Mon | your the one saying i don't |
| 00:48.20 | dijungal | hey guys... here's an interesting question: Server A passes a call to Server B then to agent 1, Agent 1 does a transfer (hit the transfer button on the phone), how do I get Server B to drop off the call path and the call be Transferred from Server A? |
| 00:49.00 | ai-a[afk] | i have a phone. and a 12mpix camera, and a mp3 player... i am glad that myp3 player runs out of battery and i can still make calls when im kaking down a river. |
| 00:49.05 | ManxPower | dijungal: what protocol? |
| 00:49.13 | De_Mon | dijungal create a forward extension on server A that you can transfer the call to that then does the transfer. |
| 00:49.20 | dijungal | IAX between the servers, SIP to the client |
| 00:49.23 | dijungal | to the agent |
| 00:49.46 | ManxPower | dijungal: you basically can't do that because you are mixing protocols |
| 00:49.55 | dijungal | oooh... use the Transfer command on Server B? |
| 00:49.55 | De_Mon | actually that prolly wolnt work |
| 00:50.26 | De_Mon | it'd just go A->B->A-> wherever |
| 00:50.29 | dijungal | ooo |
| 00:50.39 | dijungal | thought so |
| 00:50.46 | dijungal | server B will still be in the call path |
| 00:50.50 | ManxPower | if you were not mixing protocols and using only SIP for all call legs, AND all legs of the call are using the same codec, AND there are no options on the Dial(... line that would cause asterisk to stay in the media path. |
| 00:51.05 | Bawbatos | Darwin 192.168.119.121 9.0.0d1 Darwin Kernel Version 9.0.0d1: Wed Sep 19 00:08:43 PDT 2007; root:xnu-933.0.0.203.obj~21/RELEASE_ARM_S5L8900XRB iPhone1,1 unknown |
| 00:51.08 | ManxPower | then server A will drop out of the audio (but NOT signalling) path |
| 00:51.19 | dijungal | hmmm |
| 00:51.42 | ManxPower | you can disable this by specifying canreinvite=no. |
| 00:51.58 | ManxPower | Oh! also, I'm assuming there is no NAT involved anywhere in any call leg. |
| 00:52.07 | *** join/#asterisk |dennis| (n=dennis@200.32.236.10) |
| 00:52.09 | dijungal | what is i use the Transfer() function? |
| 00:52.28 | dijungal | NAT will be involved |
| 00:52.39 | dijungal | hence the reason for IAX trunking the server |
| 00:52.45 | ManxPower | dijungal: it's mainly used for IVRs |
| 00:53.03 | ManxPower | dijungal: even if you were using IAX for all the legs you would have a problem if NAT was involved. |
| 00:53.25 | dijungal | hmm... |
| 00:53.37 | ManxPower | the problem with NAT is that there is no STATE in the NAT box for server B to talk to the origination call leg |
| 00:53.42 | *** join/#asterisk ghento (n=ghento@75.155.241.7) |
| 00:54.01 | dijungal | that sucks |
| 00:54.23 | ManxPower | *shrug* Redesign things so NAT is not involved. |
| 00:55.33 | dijungal | so this does not look very possible |
| 00:55.47 | dijungal | and there's a mix of technologies |
| 00:56.12 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-5b7506af75db9afc) |
| 00:56.16 | ManxPower | dijungal: perhaps you can diagram it like: SIP ITSP <-> Internet <-> NAT Router <-> Asterisk A <-> Internet <-> Server B <-> IAX Phone. |
| 00:56.30 | dijungal | PSTN -> ServerA -> (IAX) -> ServerB -> (SIP) -> Agent1 |
| 00:56.39 | ManxPower | dijungal: there is no nat in there |
| 00:56.58 | dijungal | there might be a NAT between serverA and B |
| 00:56.58 | ManxPower | also you did not specify what technology the PSTN is using to get into Asterisk. |
| 00:57.10 | ManxPower | A diagram is not something you just spew out, THINK about it, then diagram it |
| 00:57.38 | ManxPower | I also see NO internet connection in that diagram |
| 00:57.59 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 00:58.11 | dijungal | ServerA <-> (IAX) <-> NAT ROUTER<-> INTERNET <-> NAT ROUTER <->(IAX) <-> SERVERB |
| 00:58.36 | dijungal | PSTN<->ServerA <-> (IAX) <-> NAT ROUTER<-> INTERNET <-> NAT ROUTER <->(IAX) <-> ServerB |
| 00:58.53 | ManxPower | PSTN = PRI, analog, BRI, SIP ITSP or IAX IOTSP |
| 00:59.07 | ManxPower | NOW you are STARTING to get a decent diagram |
| 00:59.16 | dijungal | i have the diagram in dia, but it can change.. because it's not implemented yet.... i'm trying to figure out the best way to achieve the transfers so it does not tie-up 2 servers |
| 00:59.23 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 00:59.32 | De_Mon | RAWR I hate lowercase 'n' pattern matching too many words contain the letter n! |
| 00:59.46 | dijungal | PSTN - PRI - 4 T1s - TE410P |
| 00:59.48 | ManxPower | dijungal: I suggest looking into VPN between the two networks so there is no nat involved. |
| 01:00.00 | dijungal | hmmm |
| 01:00.09 | ManxPower | then change everything to SIP and you might be good to go |
| 01:00.29 | dijungal | hmmmm |
| 01:00.38 | ManxPower | we use all SIP for phones and for inter-asterisk connections |
| 01:00.51 | dijungal | sip trunk between the servers |
| 01:01.05 | ManxPower | there is no such thing as a sip trunk. |
| 01:01.07 | ManxPower | ~trunk |
| 01:01.08 | jbot | trunk is, like, 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. |
| 01:01.21 | ManxPower | perhaps "peer" or "sip.conf entry" is what you are looking for. |
| 01:01.36 | ManxPower | unfortunatly using all sip will increase the bandwidth required. |
| 01:01.59 | dijungal | yes it will |
| 01:02.20 | ManxPower | Also, how the hell do you expect Server A to get out of the media path when that is the server that handles the interface to the PSTN via PRI? |
| 01:02.41 | dijungal | i need ServerB out of the media path |
| 01:02.50 | dijungal | ServerA should handle the transfer of the call |
| 01:03.07 | ManxPower | no, the server the phone is connected to will handle the transfer of the call. |
| 01:03.14 | dijungal | but the request comes from the Agent on ServerB |
| 01:03.53 | ManxPower | so server B will handle the transfer |
| 01:04.02 | *** join/#asterisk RyanW (n=cableguy@ge0-0-15-lns0.207alg.qx21.net) |
| 01:04.11 | ManxPower | so basically you are expecting the Agent (SIP phone) to manage to talk directly to server A without being registered or in any way setup to use server a |
| 01:04.16 | RyanW | I have a site using SNOM360's and i need a visual indicator for the receptionist to indicate the night switch is active. any ideas? |
| 01:05.32 | *** join/#asterisk axscode (i=axscode@134.84.48.60.klj04-home.tm.net.my) |
| 01:05.42 | ManxPower | Even if that worked (it prolly won't) you still lose the bandwidth savings you get by using IAX |
| 01:06.27 | dijungal | ManxPower: Agent (Sip Phone) does a transfer (this phone is connected to ServerB), ServerB tells Server A, can you transfer this ungoing call, then serverA does the transfer, dropping serverB |
| 01:06.31 | ManxPower | BTW, don't expect to use IAX phones, softphones suck and all the IAX hardphones REALLY suck. |
| 01:06.32 | dijungal | and the agent from the call |
| 01:06.50 | ManxPower | dijungal: that might work. |
| 01:07.05 | dijungal | ManxPower: why do softphones suck.... that's what we mostly use |
| 01:07.05 | ManxPower | there are several options in iax.conf that would control that |
| 01:07.12 | *** join/#asterisk axscode (i=axscode@134.84.48.60.klj04-home.tm.net.my) |
| 01:07.16 | Maxxed | i have a asterisk pbx and a two cisco 7960's, i hear there is a way to have the phone open things on a pc, ie. if a customer calls, some fancy agi gets the caller id, queryes a sql db, and opens up a webbrowser |
| 01:07.28 | dijungal | ManxPower: it seems logical enough, just can't figure out in my mind how it can be done on asterisk |
| 01:07.31 | Maxxed | is there a proof of concept out there any where? |
| 01:07.32 | ManxPower | dijungal: many reasons. The OS, the cheap sound hardware, cheap headsets, crappy programmers. |
| 01:07.46 | ManxPower | dijungal: it happens AUTOMATICALLY if ASterisk can do it unless you disable it. |
| 01:08.38 | ManxPower | Maxxed: once you get to "fancy agi" you are no longer dealing with Asterisk. So you really are looking for some magical softare that can do that outside of ASterisk |
| 01:08.45 | Maxxed | ah |
| 01:08.46 | Maxxed | figured |
| 01:09.21 | Maxxed | i guess one could..hum.. have an ajax like routine poll a sql table for things to do |
| 01:09.49 | Maxxed | agi writes caller id info to the table, the browser picks up the row, gets the customer info, displays it |
| 01:09.52 | Maxxed | all in a sec |
| 01:09.54 | Maxxed | heh |
| 01:09.57 | Maxxed | sounds doable |
| 01:09.58 | ManxPower | Maxxed: or a program running on the windows box that listens on a socket and is controled by a TCP connection from a shell (aka AGI) script. |
| 01:10.14 | Maxxed | ah, another good one |
| 01:10.21 | *** join/#asterisk mosty (n=mostyn@60-241-198-194.static.tpgi.com.au) |
| 01:10.29 | Maxxed | im no windows developer though so that would meen a lil work on my part |
| 01:10.39 | Maxxed | but its cool, i know there are ways to do it :) |
| 01:10.42 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 01:10.58 | ManxPower | Most windows developers burst into flames when they enter the channel. |
| 01:11.07 | Maxxed | aha |
| 01:11.07 | ManxPower | I've not heard many complaints about it. 8-) |
| 01:11.08 | Maxxed | :p |
| 01:11.22 | Maxxed | thanks for the idea ManxPower |
| 01:11.29 | Maxxed | man i sware you live in here |
| 01:11.35 | Maxxed | wtf do u do for a living? |
| 01:12.06 | ManxPower | Maxxed: I do asterisk, WAN design and management, LAN, and Linux consulting |
| 01:12.13 | Maxxed | ah cool :) |
| 01:12.22 | Maxxed | you solo or you working for a consulting shop? |
| 01:12.49 | ManxPower | I don't normally do one off consulting gigs, I generally only do consulting stuff that would be a long term relationship with the customer. I'm an indie consultant. |
| 01:13.01 | Maxxed | ah cool cool |
| 01:13.07 | Maxxed | you have a website by chance? |
| 01:13.17 | ManxPower | I don't have any customers that have been with me LESS than 5 years. The longest has been with me for 12 years. |
| 01:13.31 | Maxxed | damn, must have that customer service shit down :) |
| 01:13.37 | ManxPower | Maxxed: Customers apply to ME to be customer, not the other way around. No need for a web site. |
| 01:13.53 | ManxPower | Maxxed: I'm *quite* good at what I do. |
| 01:13.58 | Maxxed | sounds like it! |
| 01:14.22 | Maxxed | you have an hourly rate? say if i fuck something up and need a hand? |
| 01:14.37 | ManxPower | It also helps that the New Orleans area (where my customers are located) is full of tech wannabes with no real tech knowledge. |
| 01:14.45 | Maxxed | hehe |
| 01:15.11 | ManxPower | Maxxed: I have my rate for "interesting things" and a rate for "uninteresting things" |
| 01:15.18 | Maxxed | ah |
| 01:16.07 | ManxPower | For the most part unless you are using PRI, Polycom, Asterisk, Digium or Sangoma, Cisco router I'm not all that interested. |
| 01:16.39 | ManxPower | Let me put it this way. If you have a dynamic IP address, don't expect me to be interested in your needs. |
| 01:16.50 | Maxxed | lol |
| 01:16.54 | Maxxed | no i get you :p |
| 01:17.02 | ManxPower | Maxxed: my issues with Cisco are licensing |
| 01:17.11 | Maxxed | ManxPower: same here |
| 01:17.22 | Maxxed | but i have had great luck with their phones |
| 01:17.24 | mosty | when i register g729 licences, is there a cache kept of the key? |
| 01:17.31 | Maxxed | havent messed with anybody else really |
| 01:17.34 | ManxPower | mosty: on disk, yes. |
| 01:17.54 | JT | cisco phones have rabies ;) |
| 01:18.00 | mosty | ahh, /var/lib/asterisk/licenses |
| 01:18.09 | ManxPower | Maxxed: polycoms are basically just as good as Cisco phones, but without the issues with having to buy a power supply and buy firmware. The polycoms don't look as butch as Cisco. |
| 01:18.38 | Maxxed | mm, whats a good polycom modle to compete with the 7960? |
| 01:19.06 | puzzled | ManxPower, isn't the polycom more configurable than the cisco too? |
| 01:19.09 | ManxPower | It's been so long since I've used a Cisco phone, I don't recall the features of the 7960 |
| 01:19.33 | ManxPower | puzzled: yes. Some of Cisco's cool features are only available when running SCCP and Call Manager. |
| 01:19.54 | Maxxed | ManxPower: what polycom model u perfer? |
| 01:20.14 | puzzled | I still use a couple of 7961s and they work fine with chan_sccp |
| 01:20.46 | puzzled | Maxxed, I prefer the IP650 but it all depends on budget. the IP430 looks pretty solid too |
| 01:21.42 | Maxxed | mmm, il have to check some of these out |
| 01:21.54 | ManxPower | I think the 430 is what my customers mostly use, I would have to check to be sure. |
| 01:22.02 | Maxxed | il be doing a 40 some odd station deployment 1st quarter of next year |
| 01:22.10 | ManxPower | most of the time I don't even see the actual phone. |
| 01:22.24 | ManxPower | I manage around 200 polycoms |
| 01:22.40 | Maxxed | i have a good 170 79xx ciscos |
| 01:22.44 | ManxPower | where "manage" is mostly they just work. |
| 01:22.49 | Maxxed | nothing fancy at all, they just work |
| 01:22.53 | Maxxed | heh |
| 01:22.53 | Maxxed | yeah |
| 01:23.31 | Maxxed | i would like to stick with the 7960's but if the polycoms are cheaper and provide the same feature set, hell |
| 01:23.36 | Maxxed | bottem line wins |
| 01:25.10 | ManxPower | I standardized on Polycom |
| 01:25.31 | Maxxed | i standr'd on cisco a long while back |
| 01:25.35 | Maxxed | but im ot afraid of change |
| 01:25.41 | Maxxed | or mixed enviorments :) |
| 01:25.48 | Maxxed | or spelling |
| 01:25.48 | Maxxed | heh |
| 01:25.51 | ManxPower | mixed enviroments are evil. |
| 01:25.56 | ManxPower | if you have Cisco, stick with Cisco |
| 01:26.00 | Maxxed | yep |
| 01:26.31 | ManxPower | windows is managed by a different person at my clients |
| 01:26.34 | Maxxed | i have a few grandstream junk phones at a one site along with the 7960's no problem |
| 01:26.39 | Maxxed | ah |
| 01:27.05 | Maxxed | i wear to many hats |
| 01:27.12 | Maxxed | its a good thing and a bad thing |
| 01:27.20 | Maxxed | staying up to date with everything is rough |
| 01:27.33 | Maxxed | i rather be an expert in one feild than be half assed in 10 |
| 01:27.48 | Maxxed | but thats just the way the chips have fallen i guess |
| 01:30.13 | Maxxed | ell im off for now, im gona think about how im gona do this weird agi, browser thing |
| 01:30.18 | Maxxed | s/ell/well |
| 01:30.50 | Maxxed | i think the agi writing to a db that a ajax based web app polls will work aigh |
| 01:30.55 | Maxxed | i dont care for the polling crap much |
| 01:31.07 | Maxxed | but, it means i aint gota do no damn windows code'n |
| 01:31.09 | Maxxed | :D |
| 01:32.03 | Maxxed | hum.. the windows route seems alot prettier.. idk, il figure it out |
| 01:32.08 | Maxxed | thx again ManxPower :) |
| 01:32.17 | obnauticus | Okay, why when i have more than one call on 1 line comming inbound from a trunk that's through IPKall... |
| 01:32.19 | obnauticus | i cannot hear |
| 01:32.25 | obnauticus | but only when it's >1 line |
| 01:32.31 | obnauticus | with just 1 person calling i can hear fine :/ |
| 01:34.29 | De_Mon | using Dial's G option, how do you pass variables to the called party? |
| 01:35.03 | De_Mon | Right now I can only pass 1 variable and thats using _dial-. |
| 01:35.10 | *** join/#asterisk fedealves (i=johndoe@97-243.187-72.tampabay.res.rr.com) |
| 01:35.49 | fedealves | question for the wise and the smart: id I know the uniqueid of a call, how do I close it? |
| 01:36.13 | fedealves | is there a way to close a call from the command line ?? |
| 01:37.49 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 01:38.28 | *** join/#asterisk mitcheloc (n=mitchel@adsl-68-120-227-239.dsl.irvnca.pacbell.net) |
| 01:45.59 | Corydon76-dig | fedealves: there is no direct way with that information |
| 01:46.12 | Corydon76-dig | fedealves: you need the channel NAME not the unique id |
| 01:46.25 | *** join/#asterisk groogs[h] (n=gregm@cbl-66-102-80-229.wtccommunications.ca) |
| 01:47.27 | obnauticus | Corydon76-dig, if i have that info how do I do that? |
| 01:48.05 | Corydon76-dig | "soft hangup <name>" |
| 01:48.20 | obnauticus | I see. |
| 01:48.28 | obnauticus | can you force a call? |
| 01:48.39 | Corydon76-dig | not from the CLI |
| 01:48.52 | obnauticus | ... :/ |
| 01:48.55 | Corydon76-dig | You can do an Originate command from the manager |
| 01:49.18 | obnauticus | How do I run the manager again? |
| 01:49.35 | Corydon76-dig | telnet localhost 5038 |
| 01:49.35 | Nugget | telnet is eeeeeeevil! |
| 01:50.14 | Corydon76-dig | Actually, there's an originate command in trunk, dunno about 1.4 |
| 01:50.25 | Corydon76-dig | for the CLI, I mean |
| 01:51.25 | obnauticus | Huh./ |
| 01:51.31 | obnauticus | Is there any way to do it from the CLI in 1.4 :/ |
| 01:51.46 | Corydon76-dig | Maybe |
| 01:52.05 | ManxPower | PBX-1 *> !telnet localhost 5038 |
| 01:52.15 | obnauticus | ya well i know that :/ |
| 01:52.37 | Corydon76-dig | Yes, it appears to be in 1.4, if you've loaded res_clioriginate.so |
| 01:52.38 | ManxPower | that would be a "way to do it from the CLI in 1.4 :/" |
| 01:53.25 | obnauticus | oh I see. |
| 01:55.48 | obnauticus | Is there any site with syntax examples for Origionate? |
| 01:55.58 | obnauticus | Because I don't understand the example given in the cli :/ |
| 01:56.32 | ManxPower | obnauticus: there is a secret place for docs for Asterisk. Look in /path/to/src/asterisk-1.4/doc Don't tell anyone! |
| 01:56.38 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 01:56.39 | obnauticus | OGM!!! |
| 01:58.07 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 01:58.07 | *** mode/#asterisk [+o russellb] by ChanServ |
| 02:01.09 | obnauticus | ManxPower, you have any idea that when i have more than one call going on the same inbound line |
| 02:01.22 | obnauticus | i can only receive from one of the channels. |
| 02:01.48 | obnauticus | like say i have 2 phones that are on a landline, i call in on 1, i can hear it, then at the same time i call in on the second, i cannot hear it, but the 2nd phone can hear me. |
| 02:01.49 | ManxPower | obnauticus: what you said makes no sense |
| 02:02.15 | obnauticus | by `hear me' i mean speaking on my PBX |
| 02:02.18 | ManxPower | You have "2 phones on a landline" That would be outside of Asterisk connected to your local telco? |
| 02:02.24 | obnauticus | ya. |
| 02:02.30 | obnauticus | i call in on them.. |
| 02:02.36 | ManxPower | and the call is getting into Asterisk via a PRI from the telco? |
| 02:02.39 | obnauticus | asterisk can hear the first one to call. |
| 02:02.44 | obnauticus | no. |
| 02:02.50 | obnauticus | it's an ipkall line :/ |
| 02:02.53 | obnauticus | trunk |
| 02:02.55 | ManxPower | perhaps you should stop making me guess. |
| 02:03.01 | ManxPower | ~siptrunk |
| 02:03.02 | jbot | i heard siptrunk is Asterisk does not support SIP Trunks. Set trunk=no in sip.conf and then set up the device normally in sip.conf. |
| 02:03.17 | obnauticus | wait I may have been wrong |
| 02:03.20 | obnauticus | what IS a trunk again? |
| 02:03.33 | ManxPower | Ok, so you are using an analog PSTN phone to call into Asterisk via ipkall. Then where does the call go? |
| 02:03.35 | ManxPower | ~trunk |
| 02:03.36 | jbot | hmm... 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. |
| 02:03.55 | obnauticus | Oh okay. |
| 02:04.29 | ManxPower | Ok, so you are using an analog PSTN phone to call into Asterisk via ipkall, then the call gets sent to a Polycom SIP phone? |
| 02:04.49 | obnauticus | no no no.. it's not that complicated. |
| 02:04.52 | ManxPower | This is a fun game! It's called "guess at all the important information that obnauticus doesn't give out!" |
| 02:05.03 | obnauticus | I don't know how to describe it dude |
| 02:05.06 | ManxPower | It is a game I tired of quickly. |
| 02:05.07 | obnauticus | i haven't been using it a long time |
| 02:05.21 | obnauticus | I would give you a full description if i had it, it's not like I don't apperciate your help. |
| 02:05.24 | ManxPower | obnauticus: what is the device that the other person is holding in their hand? |
| 02:05.30 | obnauticus | a cell phone |
| 02:05.34 | ManxPower | all calls have two ends. The caller and the callee. |
| 02:05.46 | obnauticus | k |
| 02:05.57 | obnauticus | k lemme show you |
| 02:06.21 | *** join/#asterisk axscode (n=axscode@73.192.48.60.klj02-home.tm.net.my) |
| 02:06.22 | obnauticus | Caller 1 -[can hear]-> PBX; 2nd caller -[/]-> PBX |
| 02:06.27 | obnauticus | in chronological order. |
| 02:06.28 | ManxPower | So we have Analog Phone -> PSTN -> IPKall (SIP) -> (SIP) Asterisk-> ????????????? -> Cell Phone |
| 02:07.03 | ManxPower | now what is ??? |
| 02:07.06 | obnauticus | 2 people just calling my asterisk box through the PSTN. |
| 02:07.18 | obnauticus | It's less complicated than you are thinking... |
| 02:07.22 | obnauticus | Just start over. |
| 02:07.24 | obnauticus | sorry :/ |
| 02:07.42 | ManxPower | OK. The calls are connected to Asterisk. I don't think you are holding the asterisk server to your ear to talk to the callers. |
| 02:08.02 | obnauticus | caller 1 <-> PSTN <-> SIP <-> Asterisk |
| 02:08.23 | obnauticus | caller 2 <- PSTN <- SIP <- Asterisk |
| 02:08.28 | obnauticus | <-> shows 2 way communication |
| 02:08.41 | obnauticus | That's what is happening. |
| 02:08.46 | ManxPower | now do you know you do not have 2-way communication on caller 2? |
| 02:08.56 | obnauticus | yes. |
| 02:08.57 | obnauticus | that is correct |
| 02:08.59 | ManxPower | A call has a phone on both ends. |
| 02:09.00 | obnauticus | but with caller 1 it is fine. |
| 02:09.06 | obnauticus | ya |
| 02:09.12 | obnauticus | but my * box can't hear me. |
| 02:09.13 | obnauticus | apperentally. |
| 02:09.16 | ManxPower | you are showing me ONE end of each call, you are not showing me the other end of the call. |
| 02:09.18 | obnauticus | if i'm the 2nd one calling on that SIP |
| 02:09.28 | ManxPower | HOW DO YOU KNOW ASTERISK BOX CAN'T HEAR YOU?????????????? |
| 02:09.38 | obnauticus | Because when i call in and try to talk to myself |
| 02:09.46 | obnauticus | i cannot hear myseelf |
| 02:09.58 | ManxPower | obnauticus: I'm sorry, you don't know enough for me to continue helping you. |
| 02:10.06 | obnauticus | ,,, |
| 02:10.18 | obnauticus | i'll draw a picture then you will get it.. |
| 02:10.27 | obnauticus | i think you are just assuming when im dumb, my question is extremely simple |
| 02:10.30 | obnauticus | and im just getting tied up |
| 02:10.52 | ManxPower | When I call into an asterisk box, I don't hear myself either. |
| 02:11.03 | ManxPower | unless I'm running the Echo app, of course. |
| 02:11.11 | fujin | ^^ |
| 02:11.22 | ManxPower | Would SOMEONE please try to help this guy? |
| 02:11.24 | obnauticus | im listening on both ends. |
| 02:11.29 | fujin | I was trying to help him the other day |
| 02:11.36 | obnauticus | and it worked :/ |
| 02:11.40 | ManxPower | fujin: did it end up going like this? |
| 02:11.46 | fujin | kind of yeah, it was an uphill battle |
| 02:11.47 | fujin | ;P |
| 02:11.51 | fujin | obnauticus: what's the problem? |
| 02:11.54 | fujin | I cbf scrollback. |
| 02:12.19 | Mw3 | general user error |
| 02:12.20 | ManxPower | perhaps you mean something like Analog Phone <> PSTN <> IPKALL <> SIP <> Asterisk <> IPKALL <> PSTN <> Analog Phone? |
| 02:12.33 | obnauticus | Yes. |
| 02:12.35 | fujin | solar flares? |
| 02:12.40 | fujin | you gonna explain it again, or not |
| 02:12.49 | ManxPower | then why did you not diagram it that way in the first place? |
| 02:12.56 | obnauticus | im drawing a picture.. |
| 02:13.09 | obnauticus | ManxPower I assumed that <-> could make up for the other side of the diagram :/ |
| 02:13.15 | obnauticus | and I didn't know that I even need to write the other part :/ |
| 02:13.19 | ManxPower | don't assume that. |
| 02:13.40 | ManxPower | It just as easily could have been Analog Phone <> PSTN <> IPKALL <> SIP <> Asterisk <> SIP <-> IP Phone |
| 02:13.57 | ManxPower | or Analog Phone <> PSTN <> IPKALL <> SIP <> Asterisk <> SIP <> Softphone |
| 02:14.09 | ManxPower | or Analog Phone <> PSTN <> IPKALL <> SIP <> Asterisk <> H323 <> H323 carrier |
| 02:14.10 | fujin | I wish there was an easier way to draw OSI-style communication layers, in IRC~! |
| 02:14.15 | obnauticus | It doesn't matter... |
| 02:14.18 | obnauticus | can you just... hold on |
| 02:14.22 | obnauticus | this diagram SHOULD explain it all :/ |
| 02:14.26 | ManxPower | now, take my diagram and put in where there is NAT happening |
| 02:14.28 | obnauticus | I don't want to piss you off more ... |
| 02:14.30 | fujin | diagram-man-dem |
| 02:15.23 | ManxPower | Or am I going to have to guess where you have NAT? |
| 02:15.33 | obnauticus | I am behind an NAT yes. |
| 02:15.35 | fujin | guess, it's more fun, like that |
| 02:15.47 | ManxPower | so take the diagram I gave you and insert where NAT is happening |
| 02:16.07 | obnauticus | between IPKALL and my * box. |
| 02:16.26 | ManxPower | ok, now you know that you have an issue with one-way audio with NAT. |
| 02:16.47 | ManxPower | the answer to that is usually "put canreinvite=no in the sip.conf entry for the device/peer/host you are having issues with" |
| 02:17.24 | obnauticus | I already have that. |
| 02:17.45 | obnauticus | lemme try this though hold on |
| 02:18.19 | ManxPower | uh, "this" = what? |
| 02:18.20 | *** join/#asterisk gerphimum (n=trekkie@cpe-70-125-148-108.satx.res.rr.com) |
| 02:18.49 | ManxPower | ignore it is |
| 02:19.01 | ManxPower | much better. Someone else can help you |
| 02:19.01 | obnauticus | I didn't say you couldn't |
| 02:19.05 | obnauticus | I don't RELY on your help |
| 02:19.08 | obnauticus | stop being fucking arrogent |
| 02:19.09 | obnauticus | god damn. |
| 02:19.25 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 02:22.15 | *** join/#asterisk Own4g33| (n=Sexy@ip68-227-185-62.ri.ri.cox.net) |
| 02:22.32 | Own4g33| | Can I use Asterisk with Vonage? |
| 02:22.42 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 02:23.17 | Strom_M | no |
| 02:23.18 | Strom_M | not in any way that isn't a complete kludge anyway |
| 02:23.30 | obnauticus | fujin: http://img256.imageshack.us/img256/3920/nutskd5.png |
| 02:23.32 | Own4g33| | I know this isn't the place to ask, but what VOIP provider can i use with asterisk? |
| 02:23.40 | fujin | obnauticus: safe for work? |
| 02:23.48 | Own4g33| | preferably a large company like vonage... |
| 02:23.48 | obnauticus | Yes. |
| 02:23.56 | Strom_M | Own4g33|: look at teliax |
| 02:24.02 | rob0 | Dozens of providers. Many are listed at the wiki. |
| 02:24.09 | Strom_M | also, vonage is dying a slow and painful death. |
| 02:24.09 | fujin | obnauticus: canreinvite=yes/no? |
| 02:24.13 | obnauticus | no |
| 02:24.19 | obnauticus | =no |
| 02:24.25 | fujin | tried with =yes? |
| 02:24.29 | Own4g33| | well i would like to be able to spoof caller id with asterisk and have a large provider |
| 02:24.31 | obnauticus | Nope. |
| 02:24.39 | Own4g33| | any suggestions? also, where is the wiki? |
| 02:24.59 | Strom_M | Own4g33|: I already gave you a suggestion |
| 02:25.10 | [TK]D-Fender | ~wikis |
| 02:25.11 | jbot | wikis is probably http://www.voip-info.org |
| 02:25.12 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^ |
| 02:25.26 | Own4g33| | hmm, alright. i think i looked at that site. |
| 02:25.33 | Own4g33| | none of them were really large companies though |
| 02:25.41 | *** join/#asterisk axscode (n=axscode@73.192.48.60.klj02-home.tm.net.my) |
| 02:25.44 | Strom_M | Own4g33|: what difference does it make? |
| 02:25.48 | Strom_M | vonage is junk |
| 02:25.59 | Own4g33| | well i would rather have a large reputable company for my home phone, ya know? |
| 02:26.13 | Own4g33| | caller id spoofing is a must |
| 02:26.21 | Own4g33| | as well as private caller id unblocking |
| 02:26.31 | Strom_M | both of which are questionably legal at best |
| 02:26.32 | Own4g33| | and asterisk compatibility |
| 02:26.47 | Own4g33| | whats questionable about unblocking private caller id? |
| 02:27.01 | obnauticus | Okay fujin, when i set trunk=no it gives me 1-way audio. |
| 02:27.03 | obnauticus | I figured that much out. |
| 02:27.10 | Strom_M | it's in direct violation of federal and state telecom regulations |
| 02:27.12 | fujin | trunk=no isn't even a setting. |
| 02:27.16 | fujin | dd |
| 02:27.19 | obnauticus | lol. |
| 02:27.47 | fujin | grep trunk /usr/src/asterisk-*/sip.conf.sample |
| 02:27.50 | fujin | && echo $? |
| 02:27.53 | fujin | =0 |
| 02:28.02 | ManxPower | Own4g33|: a "large reputable company" would not let you spoof callerid and would not let you override calls that come in as callerid blocked. |
| 02:28.21 | fujin | I'm with a SIP provider that lets me specify outbound callerid |
| 02:28.24 | fujin | to anything, too |
| 02:28.27 | fujin | kinda handy. |
| 02:28.36 | Own4g33| | what provider is that fujin? |
| 02:28.45 | fujin | wxc, in New Zealand. |
| 02:28.52 | Own4g33| | ahh nvm |
| 02:29.09 | Own4g33| | there is nothing illegal about spoofing caller id |
| 02:29.17 | Strom_M | congratulations |
| 02:29.19 | Own4g33| | the law is clear on what is illegal, and simply spoofing is not |
| 02:29.23 | Strom_M | now go be 31337 somewhere else |
| 02:29.30 | fujin | pwnt by Strom_M |
| 02:29.32 | Own4g33| | it has nothing to with that |
| 02:29.35 | ManxPower | Own4g33|: only if you use it for fraud, but unblocking blocked callerid IS illegal |
| 02:29.51 | Own4g33| | really manx? |
| 02:30.00 | Own4g33| | i had no idea... so only 800 numbers can legally do it? |
| 02:30.09 | Own4g33| | wait, but dominos pizza does |
| 02:30.09 | fujin | eh? unblocking blocked callerid?? |
| 02:30.12 | Own4g33| | and it has a local number |
| 02:30.15 | ManxPower | Own4g33|: 800 numbers do not unblock callerid |
| 02:30.26 | Own4g33| | well ANI |
| 02:30.26 | fujin | how do you get something from nothing? |
| 02:30.28 | Own4g33| | my bad |
| 02:30.35 | ManxPower | 800 users receive ANI which cannot be blocked, spoofed, or otherwise faked. |
| 02:30.46 | Own4g33| | i heard ANI can be spoofed |
| 02:30.56 | obnauticus | fujin noreinvite=yes makes it 1 - way audio |
| 02:30.57 | Own4g33| | because people spoof ani to 911 all the time |
| 02:30.58 | ManxPower | The FCC considers it the right of the person paying for the call to know who made the call. |
| 02:31.05 | obnauticus | im figuring this out hopefully :/ |
| 02:31.05 | fujin | obnauticus: canreinvite |
| 02:31.11 | ManxPower | *sigh* |
| 02:31.24 | Own4g33| | SPOOF ANI to 911 FOR A QUICK LAUGH |
| 02:31.25 | [TK]D-Fender | fujin, first your you charge up the phase induction coils and re-route the plasma flow through the Heisenberg compensators.... |
| 02:31.44 | obnauticus | Own4g33 |
| 02:31.47 | obnauticus | | |
| 02:31.51 | Own4g33| | YESSIR? |
| 02:31.53 | obnauticus | that is epic lulx |
| 02:31.59 | Own4g33| | EPIC LULZ |
| 02:32.00 | Own4g33| | :-D |
| 02:32.16 | Strom_M | fucking hell...the channel is being overrun with twelve year old idiots. |
| 02:32.29 | Own4g33| | somebody ddos Strom_M |
| 02:32.29 | [TK]D-Fender | Strom_M, And there's only one of them! :p |
| 02:32.29 | fujin | It was bound to happen eventually. |
| 02:32.40 | ManxPower | Strom_M: BTW, do you know how common it is for PSAPs to do E911 location based Caller*ID .vs. ANI .vs. BTN? |
| 02:32.55 | Strom_M | ManxPower: not offhand, no |
| 02:32.55 | ManxPower | Strom_M: I TOLD you that. 8-) |
| 02:33.00 | Strom_M | heh |
| 02:33.02 | Own4g33| | somebody ddos strom off irc |
| 02:33.06 | [TK]D-Fender | Own4g33|, GTFO... no really.... GTFO :) |
| 02:33.38 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 02:33.38 | *** mode/#asterisk [+o russellb] by ChanServ |
| 02:33.48 | fujin | hiho russellb! |
| 02:33.49 | *** join/#asterisk poor_man (n=chatzill@213.63.2.234) |
| 02:33.50 | ManxPower | russellb! Our hero! |
| 02:33.51 | [TK]D-Fender | Own4g33|, what looks like your first day in and you're hunting down every kind of trouble you can find. |
| 02:33.57 | obnauticus | okay ya |
| 02:34.03 | obnauticus | fujin this problem are weird. |
| 02:34.09 | fujin | MY PROBLEM ARE WEIRD ALSO |
| 02:34.34 | ManxPower | Strom_M: One would hope most of them would use BTN |
| 02:34.47 | obnauticus | When I create a new channel on my inbound line i get 1-way communication on the new channel... |
| 02:34.52 | obnauticus | on the existing channel i still have 2-way. |
| 02:34.54 | *** mode/#asterisk [+b Own4g33|!*@*] by russellb |
| 02:34.54 | *** kick/#asterisk [Own4g33|!n=russell@asterisk/developer-and-stable-maintainer/drumkilla] by russellb (russellb) |
| 02:35.09 | Strom_M | thanks :) |
| 02:35.10 | fujin | zwned |
| 02:35.14 | fujin | ^5 russellb |
| 02:35.46 | ManxPower | It seems to be the sort to change it's nick to avoid the ban |
| 02:36.10 | russellb | yeah, but that's annoying to do, because you have to register to get in here |
| 02:36.16 | russellb | so, at least there is a little annoyance involved |
| 02:36.16 | russellb | heh |
| 02:36.35 | ManxPower | Ah! I had forgotten about that. |
| 02:36.39 | [TK]D-Fender | russellb, counter-annoyance initiative... I like it :) |
| 02:40.23 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 02:40.38 | *** mode/#asterisk [-b Own4g33|!*@*] by russellb |
| 02:41.56 | TJNII | Gosh darn it, broadvoice registrations are failing again. |
| 02:42.28 | obnauticus | fujin okay the problem is even more weird now :/ |
| 02:42.37 | fujin | awesome |
| 02:42.44 | fujin | with canreinvite=yes? |
| 02:42.48 | obnauticus | =no |
| 02:42.53 | obnauticus | with yes i get 1 way audio no matter what |
| 02:42.59 | fujin | no, you had it set to =no before |
| 02:43.09 | fujin | so you get the same behaviour with canreinvite=yes |
| 02:43.10 | fujin | ? |
| 02:43.14 | obnauticus | okay listen to this... |
| 02:43.14 | fujin | insecure=very, nat=? |
| 02:43.23 | obnauticus | insecure=very ya |
| 02:43.25 | obnauticus | nat=yes |
| 02:43.31 | *** join/#asterisk icewaterman (n=immagine@i538742FE.versanet.de) |
| 02:43.51 | fujin | well, uh |
| 02:44.06 | obnauticus | okay just hear me out for a sec.. |
| 02:44.19 | obnauticus | i call in the first time with an analog phone (it doesn't matter who's answering) I get 2-way audio |
| 02:44.25 | obnauticus | and EVERY call after that i can only get 1 way audio. |
| 02:44.33 | obnauticus | even if i hang up all my lines and call in again. |
| 02:44.40 | obnauticus | it will still be 1-way audio. |
| 02:45.34 | *** join/#asterisk mihinomenest (i=gWmQ@66.255.220.17) |
| 02:45.40 | fujin | what router |
| 02:45.45 | fujin | sounds like a portforwarding issue |
| 02:45.46 | obnauticus | pfsense freebsd box. |
| 02:45.51 | fujin | ah, that's the problem |
| 02:45.55 | fujin | toss that piece of junk out |
| 02:46.08 | fujin | what rtp range have you defined on your asterisk server? does it match up with IPKALL? |
| 02:46.26 | obnauticus | yes, it does. |
| 02:47.27 | fujin | you probably want canreinvite=no, for your diagram |
| 02:47.33 | fujin | will keep the path sane |
| 02:47.35 | obnauticus | Ya... |
| 02:47.53 | obnauticus | if it's yes then i get 1-way audio no matter what... |
| 02:48.16 | fujin | and what port forwarding have you got setup on pfsense? |
| 02:48.18 | obnauticus | 5060 - 5082 <-- that's my RTP rang |
| 02:48.18 | obnauticus | e |
| 02:48.29 | fujin | that's not a very big rtp range |
| 02:48.33 | obnauticus | well im not sure if that's the RTP.. |
| 02:48.38 | obnauticus | 10000 - 20000 <-- that's an other one i set |
| 02:48.48 | fujin | where have you set that? in rtp.conf? |
| 02:49.03 | fujin | rtpstart=10000 |
| 02:49.06 | fujin | rtpend=20000 |
| 02:49.21 | obnauticus | yup |
| 02:49.38 | fujin | and you've got 10k-20k portforwarded to the asterisk box? |
| 02:49.42 | fujin | no iptables on asterisk? |
| 02:49.48 | obnauticus | nope |
| 02:49.52 | obnauticus | just 10,000-20,000 |
| 02:50.04 | obnauticus | It's pf so :/ |
| 02:50.05 | fujin | nope to which one |
| 02:50.10 | *** join/#asterisk Steven_elvisda (n=Steven_E@202.47.107.60) |
| 02:50.18 | obnauticus | No iptables on my * server. |
| 02:50.21 | fujin | k |
| 02:50.53 | fujin | and you're sure the portforwarding is all correct? |
| 02:50.57 | obnauticus | Yes. |
| 02:51.11 | obnauticus | Those are the 2 ranges i forwarded to my * box. |
| 02:51.12 | fujin | have you tried tsharking a sip conversation? you should be able to step through it and see the rtp |
| 02:51.18 | fujin | well why do you have two rangeS? |
| 02:51.23 | fujin | you should only have one range setup |
| 02:51.31 | obnauticus | i really..don't know |
| 02:51.33 | fujin | and sip, specifically sdp, will choose which ports to use |
| 02:51.50 | *** join/#asterisk Steven_elvisda_ (n=Steven_E@202.47.107.60) |
| 02:51.58 | fujin | 5060udp needs to be forwarded, aslong as 10k-20k |
| 02:52.10 | obnauticus | 5060 - 5082 |
| 02:52.17 | obnauticus | 10000 - 20000 |
| 02:52.50 | fujin | all udp? |
| 02:52.55 | obnauticus | TCP/UDP |
| 02:53.02 | obnauticus | both. |
| 02:54.11 | fujin | heh |
| 02:54.15 | fujin | I dunno what to say, all sounds fine |
| 02:54.21 | fujin | try tshark the sip convo and analyze it with wireshark |
| 02:54.51 | obnauticus | affermative. |
| 02:55.09 | fujin | your configs sound sane, the only thing I'm not certain about is your pfsense box. |
| 02:55.25 | fujin | The only other thing I could suggest is use your asterisk box as the network termination unit |
| 02:55.28 | fujin | run iptables |
| 02:55.33 | obnauticus | well if it were being flaky, then it wouldn't be flaky at the exact same time every time. |
| 02:55.49 | fujin | sure it would, if it was an issue with the way things were handled by the firewall |
| 02:56.13 | fujin | (i.e.; put asterisk on a public ip, firewall it down, take nat out of the equation) |
| 02:56.32 | obnauticus | or just put asterisk before my NAT |
| 02:56.33 | fujin | although one-way transmitting through nat generally works fine, I'm not sure how works for phone->nat->asterisk->nat->phone |
| 02:56.57 | fujin | you could probably use openser aswell, unessl you've got some reason to run asterisk |
| 02:57.07 | fujin | looks like all you're doing is proxying stuff out a SIP provider |
| 02:57.45 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177582303.dsl.bell.ca) |
| 02:58.32 | *** join/#asterisk Mackes (n=root@cpe-76-180-236-119.buffalo.res.rr.com) |
| 02:58.55 | obnauticus | <PROTECTED> |
| 02:58.57 | obnauticus | That's one frame. |
| 02:59.10 | obnauticus | with 2 way communication :/ |
| 02:59.29 | obnauticus | wait wtf. |
| 02:59.35 | obnauticus | now i get 2-way >: |} |
| 02:59.58 | obnauticus | LOL |
| 03:00.04 | obnauticus | when i hang up from the client that's calling. |
| 03:00.24 | obnauticus | it gives me 1-way communication and f*cks up the whole thing. |
| 03:00.26 | obnauticus | and i have to reload. |
| 03:00.38 | obnauticus | but if i reload from my softphone, then it's all good >: / |
| 03:01.43 | ManxPower | For anyone curious about 911 service, take a look at nena.org |
| 03:02.36 | obnauticus | fujin, could it be something wrong with jitterbuffer? |
| 03:02.41 | JT | obnauticus: 5060udp only + rtp range |
| 03:02.45 | fujin | no |
| 03:03.12 | obnauticus | I think I found somethign weird though fujin |
| 03:03.27 | fujin | oh? |
| 03:03.34 | obnauticus | when i hang the phone up from the caller it makes everything 1-way. but from the callee it makes calls in the future 2-way. |
| 03:03.48 | obnauticus | but the first person calling me to hang up before me... |
| 03:03.55 | obnauticus | makes everything in the future for me 1-way until i reload. |
| 03:06.16 | fujin | what version of asterisk? |
| 03:06.19 | fujin | built from source? |
| 03:06.26 | obnauticus | asterisk*CLI> core show version |
| 03:06.27 | obnauticus | Asterisk SVN-branch-1.4-r71230 built by root @ asterisk on a i686 running Linux on 2007-06-23 00:39:02 UTC |
| 03:06.43 | fujin | why are you running svn? |
| 03:06.47 | obnauticus | I dunno |
| 03:06.52 | obnauticus | I...don't know :/ |
| 03:07.10 | fujin | I'm not sure how smart of an idea that is |
| 03:07.26 | Strom_M | fujin: svn 1.4 branch is fine |
| 03:07.30 | Strom_M | it's not svn head |
| 03:07.37 | Strom_M | er, svn trunk |
| 03:08.58 | *** join/#asterisk PepOSX (n=pepOSX@190.72.153.45) |
| 03:12.58 | *** join/#asterisk cryptnix (n=andrew@fw.levelsync.com) |
| 03:13.10 | cryptnix | hmm, looking for some type of reminder system anyone know of any |
| 03:13.44 | obnauticus | fujin |
| 03:13.56 | obnauticus | I'm not exactly comfertable with my configuration :| |
| 03:13.57 | Qwell | fujin seems like a pretty poor reminder system |
| 03:14.10 | obnauticus | lol\ |
| 03:14.31 | [TK]D-Fender | cryptnix, Post-it-notes |
| 03:15.02 | cryptnix | damn |
| 03:15.04 | cryptnix | that works |
| 03:15.13 | cryptnix | only i need it to be where i can call in to it |
| 03:15.19 | fujin | huh!? |
| 03:15.23 | fujin | I'm not a reminder system |
| 03:15.24 | cryptnix | ;) know of any options that are already pre-built |
| 03:15.30 | JT | cryptnix: cron + .call files |
| 03:15.33 | fujin | obnauticus: then nuke it, and make one you're comfortable with? |
| 03:15.34 | [TK]D-Fender | cryptnix, no such thing as "built in" |
| 03:15.43 | fujin | cron + callfiles is a good idea, I hadn't though to fthat. |
| 03:15.53 | [TK]D-Fender | cryptnix, get the book, sit down and READ. |
| 03:15.55 | [TK]D-Fender | ~book |
| 03:15.55 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 03:15.56 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^ |
| 03:16.10 | [TK]D-Fender | cryptnix, And you'll find out how to do thing like that. |
| 03:16.17 | cryptnix | http://www.asteriskvoipnews.com/asterisk_help/telephone_reminders_30_the_free_asterisk_telephone_reminder_system.html |
| 03:16.20 | [TK]D-Fender | cryptnix, for the parts where * can come in |
| 03:16.40 | cryptnix | thanks fender |
| 03:16.49 | *** join/#asterisk rasix (n=rasix@ip70-110.cbn.net.id) |
| 03:18.06 | obnauticus | BBLLLARGH |
| 03:18.20 | obnauticus | [2007-11-13 10:57:05] WARNING[22479]: chan_sip.c:12432 handle_response: Remote host can't match request BYE to call '4cfb5fe07fae61ac72bbe46912b50a09@10.0.0.109'. Giving up. |
| 03:18.23 | obnauticus | what does that mean? |
| 03:19.17 | [TK]D-Fender | obnauticus, you dealing with NAT in there somewhere? |
| 03:19.22 | obnauticus | ya |
| 03:19.28 | JT | probably that the remote host never knew a call existed |
| 03:19.29 | obnauticus | this is really frustrating i seriously have no idea what to do >:| |
| 03:19.31 | [TK]D-Fender | obnauticus, what make & model? |
| 03:19.42 | obnauticus | FreeBSD pfsense. |
| 03:19.56 | [TK]D-Fender | obnauticus, make sure it isn't doing ANY SIP transform, etc. |
| 03:19.59 | JT | obnauticus: have you pastebinned sip.conf? |
| 03:20.07 | obnauticus | do you want me to ? |
| 03:20.11 | JT | obnauticus: is asterisk acting as sip server? |
| 03:20.13 | JT | obnauticus: yes. |
| 03:20.16 | obnauticus | i think so |
| 03:20.34 | JT | obnauticus: are you connecting to a sip server or a phone or ata? |
| 03:20.48 | obnauticus | im not connecting to an ata |
| 03:20.49 | obnauticus | so sip server. |
| 03:21.05 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 03:21.05 | *** mode/#asterisk [+o russellb] by ChanServ |
| 03:21.43 | JT | obnauticus: is your server and ata on the same lan? |
| 03:21.51 | JT | wait |
| 03:21.52 | obnauticus | I don't have an ATA |
| 03:21.54 | JT | sip server |
| 03:21.59 | JT | so... |
| 03:22.04 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 03:22.08 | JT | Asterisk > NAT > Server on Inet |
| 03:22.16 | obnauticus | yes |
| 03:22.25 | JT | obnauticus: WHY are you port forwarding? |
| 03:22.45 | obnauticus | i have WAN -> NAT/Firewall(PFSENSE) -> Asterisk |
| 03:22.48 | obnauticus | that is my setup |
| 03:22.57 | JT | only port forward if someone on the Internet needs to access your asterisk as a server |
| 03:23.12 | JT | you do not need to port forward to connect to a sip provider |
| 03:23.47 | obnauticus | Clients are connecting to my server also. |
| 03:23.48 | obnauticus | over sip. |
| 03:23.53 | obnauticus | http://pastebin.ca/772881 <-- my sip.conf. |
| 03:24.44 | fujin | gah |
| 03:24.48 | fujin | strip the comments out next time |
| 03:25.00 | obnauticus | sorry |
| 03:25.25 | JT | obnauticus: is the externip= correct? |
| 03:25.29 | obnauticus | Yes |
| 03:25.38 | obnauticus | I made sure yesterday |
| 03:25.41 | obnauticus | and that fixed my last problem. |
| 03:25.41 | JT | obnauticus: why is localnet=10.0.0.109/255.255.0.0 ? |
| 03:25.48 | JT | that's NOT how to specify a network |
| 03:25.51 | obnauticus | LOL I put that? |
| 03:25.56 | obnauticus | 10.0.0.0/16 |
| 03:26.05 | fujin | meh, I've never used localnet. |
| 03:26.12 | JT | 10.0.0.0/255.0.0.0 |
| 03:26.18 | JT | as if use a class A |
| 03:26.18 | fujin | and do phones -> asterisk -> NAT -> sip provider -> pstn |
| 03:26.23 | JT | class C is fine for most |
| 03:26.41 | JT | fujin: yes, localnet is for asterisk > NAT > phones |
| 03:26.55 | fujin | which is not what he's doing, I believe? |
| 03:27.12 | obnauticus | no |
| 03:27.18 | fujin | he's doing PSTN <> sip provider <- NAT <- asterisk |
| 03:27.28 | fujin | in and out of the pstn |
| 03:28.03 | *** join/#asterisk cesar_cR (n=cesar@celord.ice.co.cr) |
| 03:28.08 | JT | he said he also has phones in the Internet |
| 03:28.14 | fujin | oh, I'd not heard that |
| 03:30.02 | CrazyTux | Anyone here ever played with SPA9000? |
| 03:30.34 | *** join/#asterisk jmacz (n=jmacz@190.25.35.244) |
| 03:32.06 | *** join/#asterisk mordaunt (n=mordaunt@unaffiliated/mordaunt) |
| 03:32.10 | Mackes | Is anyone using IPCOP with Asterisk? Or, what would be a enterpise |
| 03:32.33 | Mackes | level firewall for VOIP |
| 03:33.33 | rob0 | enterprise |
| 03:34.07 | fujin | enterprise level firewall for asterisk.. hmm |
| 03:34.26 | fujin | I'm not sure you're tackling the problem correctly. |
| 03:35.08 | Mackes | How is that? |
| 03:35.40 | fujin | What is the problem you're faced with? |
| 03:35.54 | rob0 | Usually the "enterprise" solution is to throw away large sums of money on a proprietary firewall which doesn't give as much control as Linux/iptables. |
| 03:36.30 | rob0 | But, that's because they wouldn't have anyone on staff who would know enough to manage iptables. |
| 03:36.33 | Mackes | Well, Not really a problem. My Company runs on Asterisk, and we are switching from several PRI's to SIP accounts |
| 03:36.46 | fujin | I'm sure. |
| 03:36.52 | fujin | For outbound, or two-way calling? |
| 03:36.52 | Mackes | Which means I have to expose my Asterisk system to the outside world |
| 03:36.58 | fujin | Not entirely. |
| 03:37.20 | Mackes | Iptables is great. |
| 03:37.30 | fujin | I have two systems here, on a private VLAN (192.168.108.x), and use a SIP provider for outgoing calls |
| 03:37.51 | fujin | but, if you need incoming calls to go through the NAT, now, there's a tricky one |
| 03:37.55 | Mackes | But I guess I would rather go with a PIX or something like that, only because its alittle clearer to me as to what is happening |
| 03:37.58 | fujin | generally involves portforwarding |
| 03:38.06 | fujin | oh god, don't even think of using a pix. |
| 03:38.17 | fujin | you'd be better off to build a nix server as a nat mangler |
| 03:38.18 | fujin | iptables |
| 03:38.26 | *** part/#asterisk rasix (n=rasix@ip70-110.cbn.net.id) |
| 03:38.27 | Mackes | Pix, no good? |
| 03:38.30 | fujin | n+1 nics |
| 03:38.32 | fujin | no, they're terrible |
| 03:38.36 | Mackes | Hmmm |
| 03:38.36 | Mackes | Ok |
| 03:38.42 | obnauticus | N+1? |
| 03:38.46 | Mackes | We have several for all other data |
| 03:38.46 | obnauticus | N+1 are good |
| 03:38.53 | JT | PIX is shite |
| 03:38.56 | JT | especially for SIP |
| 03:38.58 | Mackes | N+1? |
| 03:38.59 | fujin | aye |
| 03:39.00 | rob0 | n+2 if you want dual ISPs for failover. |
| 03:39.05 | Mackes | ok, wow, good to know |
| 03:39.05 | obnauticus | ya. |
| 03:39.07 | *** join/#asterisk emist_ (n=emist@unaffiliated/emist) |
| 03:39.09 | Mackes | ok no PIX |
| 03:39.11 | obnauticus | I have dual wan's right now anyway |
| 03:39.13 | obnauticus | i just use both of them |
| 03:39.13 | obnauticus | lol. |
| 03:39.24 | fujin | loadbalanced, or individually routed? |
| 03:39.33 | obnauticus | load balanced. |
| 03:39.38 | obnauticus | and individually routed. |
| 03:39.43 | fujin | how do you control the return paths? |
| 03:39.46 | obnauticus | dynamically individually routed. |
| 03:39.46 | obnauticus | lol. |
| 03:39.54 | obnauticus | They return on the same ip. |
| 03:39.55 | Mackes | How are you load balancing your systems |
| 03:40.07 | fujin | I use linux-ha to achieve HA asterisk |
| 03:40.14 | obnauticus | it doesn't change the speed limit, it just gives the highway more lanes, if you know what i mean |
| 03:40.18 | fujin | I don't believe loadbalancing with asterisk isn't the righ tway. |
| 03:40.38 | fujin | loadbalancing with SER, or similar, sure |
| 03:40.51 | Mackes | Do you have two Asterisk systems? |
| 03:41.06 | fujin | yes, here, I have two |
| 03:41.07 | fujin | hot/cold |
| 03:41.16 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 03:41.16 | fujin | with drbd for the voicemail |
| 03:41.26 | Mackes | Does the Hot and Cold has the same configs? |
| 03:41.38 | Mackes | drbd for the voicemail? |
| 03:41.43 | Mackes | drdb? |
| 03:41.47 | JT | google |
| 03:41.50 | Mackes | ok |
| 03:41.55 | fujin | distributed replicating block device |
| 03:42.06 | fujin | "shares" a mountpoint between servers, but isn't gay like NFS |
| 03:42.29 | Mackes | ahhh.. found it |
| 03:42.36 | Mackes | Is it fiberchannel |
| 03:42.51 | fujin | no |
| 03:42.53 | fujin | it's software |
| 03:43.01 | Mackes | SAN? |
| 03:43.06 | fujin | no, it's software |
| 03:43.09 | Mackes | Ok, I see |
| 03:43.10 | fujin | these are two standaloen servers |
| 03:43.12 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 03:43.23 | fujin | although in an esx/san environment, a voicemail LUN might work well. |
| 03:43.27 | fujin | for hot,cold anyway |
| 03:43.32 | bintut | anyone here using an aztech ip150 ip phone? |
| 03:43.37 | Mackes | ok |
| 03:43.46 | Mackes | So, one hot, one cold...... |
| 03:43.55 | Mackes | And shared sotrage for voicemail |
| 03:44.11 | bintut | i'm looking for a way to make auto-provisioning work for this ip phone.. also, make blf work.. |
| 03:44.12 | Mackes | Do you program the Cold as a second proxy in each phone? |
| 03:44.15 | *** join/#asterisk rasix (n=jeruk@ip70-110.cbn.net.id) |
| 03:44.26 | *** join/#asterisk grimsy (n=chatzill@203-206-220-214.perm.iinet.net.au) |
| 03:45.37 | bintut | you can find information of my phone here => http://www.aztech.com/prod_iptelephony_ip150.html |
| 03:46.44 | bintut | i'm wondering if anyone on this channel uses the same ip phone and made it acquired the configuration from a dhcpd server (auto-provisioning) and also the blf functionality |
| 03:46.55 | bintut | by the way, on asterisk-1.2.x |
| 03:48.00 | Mackes | Cool |
| 03:48.33 | Mackes | I have seen these phones bintut |
| 03:48.46 | Mackes | They will do IAX with a firmware change |
| 03:49.17 | bintut | really? currently, it's using sip only.. that would be nice if they will also support iax |
| 03:49.28 | bintut | s/iax/iax2 |
| 03:49.39 | [TK]D-Fender | Mackes, So have I..... by their ORIGINAL BRAND : http://www.snom.com/en/snom300_voip_phone0.html |
| 03:50.33 | [TK]D-Fender | gay-ass rebranded BS |
| 03:50.50 | bintut | [TK]D-Fender: i see.. it's a snom 300 also.. |
| 03:50.58 | [TK]D-Fender | really?!?! |
| 03:51.28 | riddlebox | lol |
| 03:51.33 | fujin | pwnt |
| 03:51.40 | fujin | are they rebadging snoms? |
| 03:52.01 | fujin | what jews |
| 03:53.06 | russellb | no jokes about race / religion / sexual orientation / etc. please :-p |
| 03:53.44 | fujin | oh, sorry |
| 03:53.57 | fujin | what rebadgers! |
| 03:54.14 | [TK]D-Fender | fujin, no anti-beastiality either! :p |
| 03:54.21 | fujin | oh my |
| 03:54.25 | fujin | what is this, #ubuntu? |
| 03:54.27 | fujin | ;[ |
| 03:54.36 | Mackes | Neato |
| 03:54.49 | Mackes | I just got a Snom 370, and I love it |
| 03:55.06 | dijungal | how do i get asterisk to store the queue_log in a database? |
| 03:55.09 | [TK]D-Fender | fujin, http://www.badgerbadgerbadger.com/ |
| 03:55.30 | fujin | limb by limb we are gon cut dem down, send fi the hacksaw, take out the tongue |
| 03:56.17 | dijungal | lol |
| 03:56.17 | Mackes | fujin, How big is your Asterisk Rollout? |
| 03:56.28 | russellb | [TK]D-Fender: the internet is weird. |
| 03:56.30 | fujin | not very big |
| 03:56.38 | fujin | 50 phones, 20~ hot-deskable helpdesk staff |
| 03:56.44 | fujin | two servers |
| 03:56.46 | Mackes | But you have Hot/Cold servers? |
| 03:57.01 | fujin | Yes. Losing all telephony capabilities for an ISP would be bad. |
| 03:57.04 | Mackes | How many concurrent phones? |
| 03:57.07 | [TK]D-Fender | russellb, ain't it grand? :) |
| 03:57.11 | fujin | Not many. |
| 03:57.14 | [TK]D-Fender | ~whee |
| 03:57.14 | jbot | [~whee] Weeeeeeeee! http://www.albinoblacksheep.com/flash/weeee |
| 03:57.20 | [TK]D-Fender | russellb, ;) |
| 03:57.32 | fujin | 12 channels, 7 active calls (right now) |
| 03:57.39 | fujin | I've tested upto a few hundred with real calls |
| 03:57.47 | fujin | few thousand with SIPp and a captured RTP stream. |
| 03:57.48 | Mackes | Very Cool |
| 03:57.50 | fujin | listening to MoH |
| 03:58.01 | Mackes | What brand servers, how beefy? |
| 03:59.52 | fujin | not very beefy |
| 03:59.54 | fujin | I didn't buy them |
| 04:00.00 | fujin | just inherited them, and built it |
| 04:00.03 | fujin | they're uh, hp's |
| 04:00.11 | Mackes | Cool yo |
| 04:00.13 | fujin | 1gb ram each, 3ghz xeon |
| 04:00.21 | Mackes | Single Core? |
| 04:00.21 | fujin | mirror raid |
| 04:00.25 | fujin | dual |
| 04:00.38 | Mackes | Do you do much trans coding? |
| 04:00.47 | Mackes | All Ulaw, for example? |
| 04:02.17 | *** join/#asterisk techie (n=techie@adsl-76-214-20-56.dsl.lsan03.sbcglobal.net) |
| 04:02.33 | fujin | Everything is alaw |
| 04:02.38 | PC_Clone | [TK]D-Fender: hey, thought I'd update you on our earlier discussion.....about the dropping audio......I think I found the culprit. A bad switch hanging off the main switch |
| 04:02.40 | fujin | voicemail is transcoded, all music-on-hold is transcoded |
| 04:02.42 | fujin | everything is alaw. |
| 04:03.24 | [TK]D-Fender | PC_Clone, Swapped and things seem stable now? |
| 04:03.40 | PC_Clone | tomorrow morning I will be |
| 04:04.09 | PC_Clone | I finally saw some errors on the main switch from that port that coincided with audio dropouts |
| 04:04.34 | PC_Clone | and the switch (Linksys EFSP42) has been known to cause some packet issues under load |
| 04:05.42 | [TK]D-Fender | PC_Clone, A reasonable suspect and far from the worst that could happen. Guess you'll have some confirmation shortly. |
| 04:06.18 | PC_Clone | yes sir |
| 04:06.41 | PC_Clone | fujin: you get a Point to point installed yet ? :) |
| 04:06.54 | fujin | no, I've no reason to want one. |
| 04:07.21 | PC_Clone | where are you from? |
| 04:07.32 | PC_Clone | seeing that you use alaw |
| 04:07.55 | Mackes | Yeah alaw? |
| 04:08.01 | fujin | I'm in New Zealand. |
| 04:08.04 | PC_Clone | ah |
| 04:08.08 | Mackes | Ahh |
| 04:08.57 | [TK]D-Fender | fujin, I'm collecting reasons to head that way. Got 1 client there already... |
| 04:09.42 | Mackes | Why is Ulaw and Alaw regional? |
| 04:10.46 | PC_Clone | same reason ami and b8zf are on T1's in the states |
| 04:11.00 | PC_Clone | they started using it, and had to keep using it |
| 04:11.14 | fujin | Mackes: because America is difficult? |
| 04:11.23 | Mackes | Well... |
| 04:11.24 | Mackes | ok |
| 04:11.37 | *** join/#asterisk Splat (n=splat@home.heehawhills.com) |
| 04:11.37 | Mackes | We could use Alaw. |
| 04:11.43 | PC_Clone | fujin: just our governement and their granted duopolies |
| 04:12.08 | Mackes | However, and please excuse my ignorace. Why have two standards? Is it for more then SIP? |
| 04:12.15 | Mackes | I know its a codec |
| 04:12.21 | fujin | yes, the analogue phone system runs xLaw aswell. |
| 04:12.23 | Mackes | but how else is it used |
| 04:12.35 | Mackes | so digital systems use it |
| 04:12.49 | Mackes | On PRI tie lines? |
| 04:13.07 | fujin | everything uses it |
| 04:13.13 | fujin | pri, bra, bri |
| 04:13.29 | asteriskguy | hey fujin |
| 04:13.33 | fujin | hello :) |
| 04:13.41 | Mackes | Ahhh.. Ok, so its the toll Quality Codec for all Digital voice, I got it |
| 04:13.43 | obnauticus | is externhost= so that asterisk can determen your external ip by doing an A lookup on the domain you sepcify? |
| 04:13.53 | asteriskguy | so I brought the vm esx server idea to my boss |
| 04:14.00 | Mackes | We run ESX |
| 04:14.06 | [TK]D-Fender | obnauticus, Yes |
| 04:14.11 | asteriskguy | the cost was going to be 90K including servers |
| 04:14.16 | fujin | Mackes: it's the codec for *everything*. Unless, your upstream provider converts to sip somewhere across the country. |
| 04:14.19 | asteriskguy | lol, I don't think he liked it |
| 04:14.21 | obnauticus | If I have externhos specified, do I have to specify externip? |
| 04:14.22 | fujin | or transcode it forwhatever reason |
| 04:14.26 | obnauticus | That owuldn't make sense. |
| 04:14.30 | asteriskguy | Mackes you run * on ESX? |
| 04:14.36 | Mackes | Yep |
| 04:14.45 | asteriskguy | how many clients? |
| 04:14.46 | Mackes | and on Vmware Free Server as well |
| 04:14.53 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 04:15.00 | asteriskguy | on linux I hope |
| 04:15.22 | Mackes | 400 regesters clients, however, we only have a few calls active at one time |
| 04:15.29 | asteriskguy | oh.... |
| 04:15.37 | Mackes | However, I have 15 other VM's on the same steal |
| 04:15.43 | asteriskguy | we're pretty busy |
| 04:15.46 | asteriskguy | lots of sales |
| 04:16.01 | asteriskguy | 16 phones/location |
| 04:16.14 | Mackes | I think Asterisk does a great job on Vmware |
| 04:16.18 | asteriskguy | mackes, what kind of codec? |
| 04:16.31 | asteriskguy | are all clients local or remote? |
| 04:16.41 | Mackes | Well |
| 04:16.49 | Mackes | That is a hard question |
| 04:17.02 | Mackes | We have 30 retail locations, and 3 regonial offices |
| 04:17.08 | [TK]D-Fender | obnauticus, You don't |
| 04:17.09 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 04:17.15 | Mackes | however the Asterisk servers are all at our HQ |
| 04:17.25 | asteriskguy | how many * servers? |
| 04:17.25 | Mackes | and we only put phones (aastra) at our locations |
| 04:17.58 | asteriskguy | currently we have the corporate office (~250 handsets) + 40 locations (16 handsets/loc) |
| 04:18.27 | asteriskguy | actually, scratch that, more like ~350 for corporate |
| 04:18.28 | Mackes | Its Complicated, but for normal "phone call" use, one IBM 3650, 4 GB RAM, 2 Dual Cores (4 Cores) |
| 04:18.33 | asteriskguy | all running on 1 server |
| 04:18.50 | asteriskguy | 2 dual cores, 8 GB |
| 04:19.07 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:19.25 | asteriskguy | no problems so far except limitation like single parking lot for the entire system |
| 04:19.26 | Mackes | and we have a virtual server (ESX) that supports 400 softclients and Zultys WiFi phones that we use to monitor our employees |
| 04:19.36 | Mackes | right, same with us |
| 04:19.43 | Mackes | We like the single parking lot |
| 04:19.46 | asteriskguy | hmm....what do you mean monitor? |
| 04:20.02 | fujin | chanspy I assume. |
| 04:20.07 | asteriskguy | no our case, we had guys from california picking sale calls in new york |
| 04:20.18 | asteriskguy | oh...chanspy |
| 04:20.19 | asteriskguy | :) |
| 04:20.21 | Mackes | I have set up softclients on our cash regesters for auto answer with USB mics so that we can listen to customer interactions |
| 04:20.31 | asteriskguy | oh wow |
| 04:20.42 | asteriskguy | 007 stuff here |
| 04:20.51 | Mackes | And we have WiFi phones that we clip ot our outdoor Cashiers to monitor them |
| 04:20.54 | obnauticus | YAA |
| 04:20.59 | asteriskguy | what kind of chain are you running? |
| 04:21.12 | obnauticus | I have a laser mic hooked up to a twisted pair plugged into my asterisk box with a channel |
| 04:21.16 | obnauticus | pointed at my neighboor's house |
| 04:21.17 | asteriskguy | do they know it's a wifi phone? |
| 04:21.19 | obnauticus | and i can call in and listen to them! |
| 04:21.33 | Mackes | Check it out.... www.deltasoniccarwash.com |
| 04:21.51 | Mackes | Largest Carwash chain (by Volume) in the US |
| 04:22.42 | asteriskguy | is there such a thing as a wifi phone, only the base is wifi and the handset is attached to the base? |
| 04:22.43 | Mackes | Yep... We give them bonuses based on how friendly our employees are |
| 04:23.03 | asteriskguy | damn....I need to go work for you guys |
| 04:23.06 | Mackes | yep.. Ethernet WiFi, SIP based... |
| 04:23.09 | fujin | meh, dect phones with analogue<>sip adapters are much better than WiFi. |
| 04:23.14 | asteriskguy | do they give bonuses to (nice) IT guys? |
| 04:23.35 | Mackes | I have alot of toys. |
| 04:23.44 | Mackes | and the pay is fair |
| 04:23.51 | asteriskguy | Mackes, got a link to that wifi phone? |
| 04:24.02 | asteriskguy | fair? depends on the market though right? |
| 04:24.07 | JT | Mackes: no |
| 04:24.12 | JT | Nugget: not ethernet wifi |
| 04:24.19 | JT | DECT to ethernet |
| 04:24.24 | Mackes | Sure.... I have quite a few "test" WiFi phones .. which one? |
| 04:24.44 | asteriskguy | i'm looking for one that has a wireless base |
| 04:24.49 | Mackes | DECT to ethernet? |
| 04:25.12 | JT | yes |
| 04:25.12 | asteriskguy | but the handset itself is attached to the base |
| 04:25.12 | JT | ie without the shit component |
| 04:25.12 | JT | wifi |
| 04:25.24 | Mackes | Ohh.. So you would like a phone without needing 802.11X APs? |
| 04:25.24 | [TK]D-Fender | ~wifisip |
| 04:25.25 | jbot | Wi-Fi SIP phones suck. All of them. HARD. Some only slightly less than others... |
| 04:25.33 | fujin | asterisk <-> analogue/sip ethernet adaptor <-> DECT base station/handunit |
| 04:25.37 | JT | Mackes: most people don't want wifi |
| 04:25.39 | Mackes | Na, they can be VERY good. |
| 04:25.40 | JT | for voip |
| 04:25.40 | Mackes | Really |
| 04:25.42 | JT | haha |
| 04:25.44 | JT | crackpot |
| 04:25.59 | fujin | wm6 softphone over wifi performs ok, I've used it when I absolutely need to |
| 04:26.04 | fujin | but that's not your average wifi phone. |
| 04:26.09 | obnauticus | my wireless sip phone is sexy |
| 04:26.15 | asteriskguy | well, JT...we tested a phone out via wireless |
| 04:26.35 | russellb | fujin: what wm6 softphone do you use |
| 04:26.38 | fujin | I wouldn't roll wifi SIP devices into production, man, there'd be so many complaints |
| 04:26.40 | Mackes | http://www.zultys.com/index.jsp?tab=productdetail&product=wip2&detail=datasheet-wip2&type=phones |
| 04:26.40 | [TK]D-Fender | obnauticus, Yes, and we're now quite sure you leave it on vibrate and in interesting places... |
| 04:26.51 | asteriskguy | sound quality was actually better then when calling our 800 number |
| 04:26.58 | fujin | russellb: I use the built-in one, mostly, but have tried that ting or thing or zing or whatever it's called |
| 04:27.01 | asteriskguy | internally that is, via a t1 connx |
| 04:27.10 | JT | Mackes: the problem is the technology |
| 04:27.12 | Mackes | The Hitachi is also a great WiFi Sip phone |
| 04:27.24 | russellb | fujin: didn't know there was one built in, heh |
| 04:27.34 | fujin | aye, just need a configuration tool to tweak the settings |
| 04:28.09 | Mackes | We use Alcatel WiFi AP's with an Enterrpise controller, and we can roam between all of our locations without dropping while at the location |
| 04:28.23 | fujin | I'm off, hometime. seeya guys. |
| 04:28.35 | JT | Mackes: sure, spend a billion dollars on APs and it might work ok |
| 04:28.41 | Mackes | Our HQ has 4 floors and we can roam between 16 AP's without an issue |
| 04:29.02 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 04:29.10 | asteriskguy | hmm...what about cisco? |
| 04:29.15 | Mackes | Linksys 54G with DDWRT works very well... very inexpensive solution |
| 04:29.23 | Mackes | We had Cisco 1200's |
| 04:29.40 | Mackes | Very Nice AP's but its hard to admin them all when you have alot |
| 04:29.58 | asteriskguy | don't they have somekind of controller? |
| 04:30.03 | Mackes | And, they dont allow roaming as well as they should |
| 04:30.08 | Mackes | Cisco sells software |
| 04:30.14 | Mackes | but its expensive |
| 04:30.30 | asteriskguy | yeah |
| 04:30.35 | asteriskguy | expensive |
| 04:30.50 | Mackes | Check out DDWRT. |
| 04:30.57 | asteriskguy | to convert our corporate office to voip using call manager was going to cost us 750K |
| 04:31.12 | Mackes | US dollars? |
| 04:31.17 | asteriskguy | yeah |
| 04:31.28 | Mackes | WAAAAYYYY TO MUCH |
| 04:31.45 | asteriskguy | so we converted the corporate office + 40 branch offices all on Polycom & PoE switches |
| 04:31.54 | asteriskguy | actually, two corporate offices |
| 04:32.01 | asteriskguy | total? <250K |
| 04:32.18 | Mackes | 65,000 will get you hunderds of Voip phones (Cisco 7960's) Asterisk on an IBM Server, an great WiFI Infastucture and WiFi Phones |
| 04:32.31 | Mackes | Polycom is great |
| 04:32.43 | asteriskguy | yeah, i like them |
| 04:33.07 | asteriskguy | support is good too, we have our own sales engineer with their company |
| 04:33.11 | JT | why would you get cisco phones + asterisk |
| 04:33.13 | JT | ? |
| 04:33.14 | JT | crazy |
| 04:33.35 | Mackes | Cisco phones are the best I thing |
| 04:33.38 | Mackes | think |
| 04:33.40 | JT | nope |
| 04:33.42 | JT | polycom are |
| 04:33.43 | Mackes | ok |
| 04:33.52 | JT | cisco sip offerings are second rate |
| 04:33.52 | Mackes | Polycom are great |
| 04:34.11 | Mackes | Yeah, Kinda of.. It depends. |
| 04:34.23 | Mackes | I think Polycom and Cisco are the same |
| 04:34.34 | JT | audio quality in the polycoms are better |
| 04:34.44 | JT | the company policies of polycom are much nicer than cisco |
| 04:34.50 | JT | cisco are arseholes |
| 04:34.52 | Mackes | Snom is a cool hobby phone, and I like Aastra for mass deployment |
| 04:35.03 | Mackes | grandstream are only good for overheard paging |
| 04:35.03 | JT | snoms are a waste of money ;) |
| 04:35.06 | asteriskguy | I would agree, polycom |
| 04:35.28 | asteriskguy | price & features combination = Polycom |
| 04:35.38 | Mackes | Dont you think Polycom are hard to Mass Deploy? TFtp? |
| 04:35.40 | JT | i'd rather a linksys than a snom |
| 04:35.43 | Mackes | The XML is a bitch |
| 04:35.43 | JT | snoms are so ugly |
| 04:35.49 | asteriskguy | really? tell me a little about grandstream and overhead paging |
| 04:35.58 | JT | polycoms do tftp, ftp, http, https |
| 04:36.00 | asteriskguy | we use FTP |
| 04:36.12 | asteriskguy | works fine |
| 04:36.19 | Mackes | Yeah, but the XML configs are, well difficult |
| 04:36.20 | JT | xml is easy |
| 04:36.27 | asteriskguy | linksys? |
| 04:36.29 | JT | it's easy to autogenerate too |
| 04:36.46 | Mackes | Cisco's and Aastra and clear text, and make sense to read |
| 04:36.54 | asteriskguy | yeah, we got several scripts to generate config files for polycom |
| 04:37.04 | JT | sure, and make no sense to parse |
| 04:37.11 | Mackes | yeah |
| 04:37.21 | Mackes | Polycom are a well made phone |
| 04:37.32 | Mackes | our conference room phones are SIP Polycom |
| 04:37.41 | asteriskguy | Mackes, grandstream & overhead paging |
| 04:37.46 | Mackes | oh yeah |
| 04:37.47 | asteriskguy | do you use overhead paging? |
| 04:37.50 | Mackes | ywp |
| 04:37.52 | Mackes | yep |
| 04:37.59 | Mackes | every location |
| 04:38.11 | asteriskguy | so phones at remote location connected to * and corporate and page overhead? |
| 04:38.30 | asteriskguy | we got it working like that through using a viking and linksys spa device |
| 04:38.38 | Mackes | set the grandstram to autoanswer, and tap your PA amp into the speaker |
| 04:38.52 | Mackes | works great |
| 04:38.53 | Mackes | no issues |
| 04:38.59 | asteriskguy | not very pleasant though cuz after the page you get this busy tone over the overhead speaker |
| 04:39.07 | asteriskguy | so you have to rig the phone? |
| 04:39.24 | asteriskguy | what model of grandstream are you using? |
| 04:39.31 | Mackes | gxp 2000 |
| 04:39.45 | nestAr | you have my condolences |
| 04:39.47 | nestAr | ;) |
| 04:39.50 | Mackes | We use them for all sorts of wierd uses, because they are ceap |
| 04:40.05 | Mackes | never had the busy tone |
| 04:40.10 | asteriskguy | hmm....when you said speaker you mean speaker phone right? |
| 04:40.27 | Mackes | We have used Zone paging units and Linksys FXO ATA's as well |
| 04:40.38 | Mackes | right, but they have a jack as well |
| 04:41.37 | asteriskguy | speaker jack? or headset jack? |
| 04:41.58 | Mackes | yes |
| 04:42.19 | asteriskguy | ? |
| 04:42.39 | asteriskguy | which one? |
| 04:42.56 | Mackes | http://www.voip-info.org/wiki/index.php?page=Asterisk+Paging+and+Intercom |
| 04:43.00 | TJNII | Will a goto work with a pattern match as a recipiant? I mean will goto(outgoing,18002858585,1) go to exten => 1800XXXXXXX,1,blah? |
| 04:43.12 | Mackes | ? |
| 04:43.22 | TJNII | s/exten => 1800XXXXXXX,1,blah/exten => _1800XXXXXXX,1,blah/ |
| 04:43.42 | Mackes | I think so |
| 04:43.56 | TJNII | Let me try it. |
| 04:44.12 | *** join/#asterisk Juggie (n=Juggie@CPE001601df17fb-CM001a6682d7b6.cpe.net.cable.rogers.com) |
| 04:44.21 | Mackes | You might need to use the Local statement |
| 04:44.48 | asteriskguy | hmm...we got the polycom phones to autoanswer in speaker mode |
| 04:44.54 | asteriskguy | only problem is the mic is enabled |
| 04:45.58 | Mackes | Yeah, well if you are going to use it for overhead paging, its worth disconnecting the mic |
| 04:46.19 | Mackes | But Im sure you could put something in the XML config to disable to mic? |
| 04:46.42 | asteriskguy | polycom? |
| 04:46.59 | asteriskguy | hmm....never thought of that, maybe I should talk to the sales engineer? |
| 04:47.18 | Mackes | I know that Aastra, Snom and Grandstream have methods for "paging" that disable the mic |
| 04:47.38 | Mackes | We use it in the office for paging single rooms out the phone only |
| 04:48.03 | Mackes | I wish there was a way to "Page" several SIP phones and have them all connect at once.. |
| 04:48.12 | Mackes | So that I would not need overhead paging |
| 04:48.34 | asteriskguy | we have that working |
| 04:48.43 | Mackes | HOW?> |
| 04:48.47 | Mackes | Please, |
| 04:48.51 | asteriskguy | we can page 16 polycom phones on their speaker phones at once |
| 04:48.54 | asteriskguy | one sec |
| 04:49.01 | Mackes | Really!!! |
| 04:49.13 | Mackes | I have tried everything |
| 04:49.19 | asteriskguy | yeah, the only problem is like I said, the mic is enabled |
| 04:49.32 | Mackes | once I connect to one, Asterisk stops attempting to connect to the rest |
| 04:49.44 | Mackes | I have tried to put them in a conference room |
| 04:49.55 | asteriskguy | use the page command |
| 04:49.56 | Mackes | but, well, its not practical |
| 04:50.07 | Mackes | in asterisk? |
| 04:50.15 | asteriskguy | it's basically a meetme conference |
| 04:50.17 | asteriskguy | yeah |
| 04:50.19 | Mackes | Thats an extention of the Meetme? |
| 04:50.21 | Mackes | right |
| 04:50.26 | Mackes | hmmmm |
| 04:50.34 | Mackes | I will have to look at that again |
| 04:50.41 | asteriskguy | hangon |
| 04:50.44 | asteriskguy | I'll send you the info |
| 04:50.46 | Mackes | I know what you are refering to |
| 04:50.48 | Mackes | cool |
| 04:50.50 | Mackes | thanks |
| 04:51.31 | Mackes | Have any of you gone to Digium Boot camp? |
| 04:51.42 | asteriskguy | I did |
| 04:51.52 | TJNII | Yea, that does work OK. |
| 04:52.28 | Mackes | Did you think it was worth the money? |
| 05:00.51 | grimsy | i'm going next week, so ask me then :P |
| 05:02.14 | Mackes | cool |
| 05:03.54 | *** join/#asterisk codeshah (n=codeshah@S01060011092d0063.ed.shawcable.net) |
| 05:04.57 | Corydon76-dig | Mackes: I went for the last day |
| 05:06.04 | Corydon76-dig | It's definitely a good refresher course, plus I learned a few new things |
| 05:06.36 | Corydon76-dig | Mackes: just remember, it's NOT for newbies... |
| 05:06.56 | Corydon76-dig | You have to know Linux and you should have gotten your feet wet already |
| 05:07.16 | Corydon76-dig | but the instruction is good |
| 05:12.06 | asteriskguy | depens on the instructor though |
| 05:12.20 | asteriskguy | we had a total of 3 guys went through the course |
| 05:12.39 | asteriskguy | 1 of us was lucky, he had the author of THE book as the instructor |
| 05:13.07 | Corydon76-dig | Which author? |
| 05:13.14 | Qwell | probably Jared |
| 05:13.19 | Corydon76-dig | Both Jared and Leif teach |
| 05:13.36 | Qwell | seeing as how asteriskguy is in the US :) |
| 05:13.54 | Corydon76-dig | Is he? |
| 05:14.02 | Qwell | socal.res.rr.com |
| 05:14.27 | Corydon76-dig | Yeah, but I don't trust cable providers to get hostnames correct |
| 05:14.42 | Qwell | rr isn't in Canada |
| 05:14.49 | Qwell | at least, I don't think they are |
| 05:14.50 | Corydon76-dig | I know people in Tennessee who have New Jersey hostnames |
| 05:15.11 | BBHoss | heh, they probably steal cable |
| 05:15.22 | BBHoss | internet that is |
| 05:15.31 | BBHoss | by using a mac address from a different node |
| 05:15.32 | Corydon76-dig | I know RR is in Minnesota, which might as well be Canada |
| 05:15.34 | Qwell | yeah, would only need 1000 miles of coax |
| 05:15.44 | BBHoss | check it out |
| 05:15.48 | BBHoss | tcniso.net |
| 05:17.35 | *** join/#asterisk emist (n=emist@unaffiliated/emist) |
| 05:22.29 | *** join/#asterisk TrentCreek (n=TrentCre@cpe-70-117-207-168.rgv.res.rr.com) |
| 05:23.30 | *** join/#asterisk ManxPower (n=manxpowe@71-8-61-95.dhcp.leds.al.charter.com) |
| 05:24.09 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.141.173) |
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| 05:38.53 | *** join/#asterisk yxa (n=lonari@58.185.90.101) |
| 05:39.56 | yxa | does anyone has experience creating call files thru php, controlled by arguments of POST data? |
| 05:41.18 | grimsy | yxa: tried this? http://www.voipjots.com/2006/02/click-to-call-with-your-asteriskhome.html |
| 05:43.40 | yxa | grimsy looking at it now. i'm just not familiar with php |
| 05:43.59 | grimsy | ah, probably wouldn't help then ;) |
| 05:44.19 | grimsy | i've come across lots of projects on sourceforge etc that have click to call functionality |
| 05:44.38 | grimsy | might be worth taking a look at some of those, pulling them apart and learning from there |
| 05:49.08 | BBHoss | yxa |
| 05:49.27 | BBHoss | i have an example script that you could easily enough adapt |
| 05:49.55 | *** part/#asterisk techie (n=techie@adsl-76-214-20-56.dsl.lsan03.sbcglobal.net) |
| 05:50.01 | BBHoss | it reads in numbers from a file, then sets it to a variable |
| 05:50.17 | BBHoss | you could just make the POST the variable and it would work |
| 05:51.02 | BBHoss | you want it? |
| 05:51.21 | endre | do not want |
| 05:51.25 | *** part/#asterisk Mackes (n=root@cpe-76-180-236-119.buffalo.res.rr.com) |
| 05:51.29 | BBHoss | this aint woot! |
| 05:52.19 | BBHoss | http://rafb.net/p/FCNOmG18.html if youre interested |
| 05:55.26 | ManxPower | yxa: why not do it in a language you are familiar with? |
| 05:58.49 | yxa | ManxPower the requirement was POST. i guess php is the easiet |
| 05:59.26 | BBHoss | did you look at my paste? |
| 06:00.26 | yxa | BBHoss yeah i want it. can you paste it to me private? |
| 06:00.37 | BBHoss | what? |
| 06:00.42 | BBHoss | i did paste it |
| 06:00.51 | yxa | BBHoss sorry got it |
| 06:01.13 | obnauticus | play |
| 06:01.16 | obnauticus | BBHoss can you help me? |
| 06:01.16 | yxa | BBHoss thanks |
| 06:01.24 | obnauticus | I've been having a really weird problem lately. |
| 06:01.31 | BBHoss | just feed the number into the $num variable |
| 06:03.51 | obnauticus | BBHoss, sometimes I'm able to have 2-way audio, and sometimes i only have 1-way audio |
| 06:04.01 | obnauticus | depending at what intervals i call, and how many calls im handling |
| 06:04.06 | obnauticus | on my * box |
| 06:04.17 | BBHoss | whats the specs of the box |
| 06:04.25 | obnauticus | Pos. |
| 06:04.35 | obnauticus | it works uhh |
| 06:04.43 | obnauticus | It's really hard to explain. |
| 06:04.45 | yxa | BBHoss not familiar with php and POST. need a little more handholding if you don't mind |
| 06:04.46 | obnauticus | because it's really flaky. |
| 06:04.58 | BBHoss | heh |
| 06:05.03 | obnauticus | like |
| 06:05.03 | obnauticus | okay |
| 06:05.06 | obnauticus | Here's one situation |
| 06:05.10 | obnauticus | I call in, i get 2-way audio |
| 06:05.14 | BBHoss | you ought to go to #php then because im no expert either |
| 06:05.15 | obnauticus | i hang up, immidetally call back |
| 06:05.23 | obnauticus | and i only get 1-way audio |
| 06:05.26 | obnauticus | i wait for like |
| 06:05.28 | obnauticus | a min or something |
| 06:05.32 | obnauticus | call back i have 2-way audio |
| 06:05.38 | obnauticus | that's one situation. |
| 06:05.51 | obnauticus | an other is, I receive more than one call at the same time... |
| 06:06.05 | obnauticus | the first caller is able to have 2-way audio, but the second is only 1-way |
| 06:06.10 | obnauticus | this is all external through IPKall |
| 06:06.15 | BBHoss | that is odd |
| 06:06.35 | BBHoss | did you ever compile from the tarball, or are you still using the SVN code |
| 06:06.40 | obnauticus | SVN |
| 06:06.48 | BBHoss | try a ball of tar |
| 06:06.49 | obnauticus | I asked here earlyer they said that the SVN release i had was fine. |
| 06:06.51 | obnauticus | ... |
| 06:06.52 | obnauticus | HEY . |
| 06:06.54 | obnauticus | Ball of tar. |
| 06:06.54 | yxa | does IPkall still restrict calls to callerid? |
| 06:06.55 | obnauticus | lol. |
| 06:07.03 | obnauticus | k |
| 06:07.17 | BBHoss | or do a reinstall, unless you did that already |
| 06:07.23 | obnauticus | I did that already. |
| 06:07.37 | BBHoss | then its either your hardware, or your provider |
| 06:08.14 | JT | BBHoss: have you tried disabling the firewall? |
| 06:08.41 | obnauticus | JT I'm just looking for multiple answers :/ |
| 06:09.23 | BBHoss | but it is reproducable (as in it does it every time) |
| 06:09.34 | obnauticus | ya |
| 06:09.53 | obnauticus | I think if I hang up on the line im calling IN on (ie. my cell phone) then it gives me 1-way audio for a while too |
| 06:09.55 | obnauticus | but im not sure. |
| 06:09.55 | BBHoss | try another provider and see if you have the same problem |
| 06:10.10 | BBHoss | what about outbound? |
| 06:10.20 | obnauticus | I can't really test that |
| 06:10.24 | obnauticus | im not paying for a trunk...yet |
| 06:10.25 | obnauticus | soon though |
| 06:10.34 | obnauticus | ya.. thios is weird |
| 06:10.37 | BBHoss | voipjet gives 25 cents free |
| 06:10.44 | obnauticus | i don't know I don't have that though |
| 06:10.49 | obnauticus | it still says i have 0 credit. |
| 06:18.23 | *** join/#asterisk BeeBuu (n=chatzill@125.95.248.142) |
| 06:18.57 | *** join/#asterisk saftsack (n=saftsack@pD9E07653.dip.t-dialin.net) |
| 06:19.13 | obnauticus | well BBHoss you know what's weird is |
| 06:19.31 | obnauticus | if it really couldn't hear my phone wouldn't it say it could't reply to the critical packet |
| 06:19.34 | obnauticus | or something like that |
| 06:21.06 | BBHoss | thats why im thinking it might be a provider issue |
| 06:22.43 | obnauticus | I don't know if this could be anything of concern but I get this |
| 06:22.44 | obnauticus | Internal RTCP NTP clock skew detected: lsr=2637328271, now=2637394649, dlsr=66387 (1:012ms), diff=9 |
| 06:22.48 | obnauticus | Dunno how to fix it |
| 06:23.40 | obnauticus | That's weird |
| 06:25.39 | BBHoss | probably just useless debug info |
| 06:27.48 | *** join/#asterisk Steven_elvisda (n=Steven_E@202.47.107.60) |
| 06:28.34 | *** join/#asterisk l0verb0y (n=l0verb0y@210.1.137.41) |
| 06:28.37 | l0verb0y | join #freepbx |
| 06:28.48 | obnauticus | ... |
| 06:28.52 | l0verb0y | oops! |
| 06:28.56 | obnauticus | lol |
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| 06:48.29 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
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| 07:05.03 | obnauticus | BBHoss, I think it's a provider issue now. |
| 07:05.20 | obnauticus | This is because when i reload asterisk, and if it was an asterisk problem, i would get 2-way audio again after i restart it. |
| 07:05.24 | obnauticus | I mean restart*** |
| 07:05.32 | obnauticus | But when i restart it, i don't. |
| 07:05.40 | obnauticus | I still get 1-way |
| 07:05.49 | obnauticus | until i wait `a little bit' (i don't know how long) then I can get 2-way. |
| 07:07.46 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 07:09.44 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-200d8b0c2ff626bb) |
| 07:12.38 | BeeBuu | hello,all |
| 07:13.35 | BeeBuu | when i retrydial(),there is "Music class default requested but no musiconhold loaded." in CLI> |
| 07:13.53 | BeeBuu | anyone tell me what's that mean? thanks. |
| 07:18.34 | obnauticus | how do i define multiple extension numbers within one line? |
| 07:18.36 | obnauticus | like |
| 07:18.45 | obnauticus | exten => {*4,*3} |
| 07:19.42 | awk | anyone here dev a billing software |
| 07:19.57 | awk | I have a query as to how to bill a certain type of call |
| 07:23.52 | orkid | give it to 'em for free :) |
| 07:29.09 | awk | :P |
| 07:30.31 | Sweeper | charge them as much as you can gt away with! |
| 07:31.30 | awk | well if somebody does have any insight it would be great, because I don't know a company that bills these calls correctly |
| 07:31.30 | orkid | NOOOO |
| 07:31.31 | awk | or at all |
| 07:31.31 | orkid | lol |
| 07:31.44 | awk | so anyone done and dev work on asterisk billing please let me know |
| 07:32.04 | orkid | sorry to be such a pest. |
| 07:32.06 | orkid | gl |
| 07:32.44 | *** join/#asterisk xnam (n=xnam@S0106006097940f68.vw.shawcable.net) |
| 07:34.33 | Sweeper | awk: do you mean, you want to know how much to call, or you want to know how to properly flag it? |
| 07:34.48 | Sweeper | s/call/charge/ |
| 07:35.50 | awk | Sweeper: how to bill transfer calls, as transfer calls are split over to cdr fields |
| 07:36.11 | awk | I need to find a way to bill for that entire call, as even the unique ids are different due to initiating dial twice |
| 07:36.38 | awk | If I could set a channel variable throughout a call, and then just update the cdr(userfield) that way, I can link a call on the userfield column. I just need to make sure I can control the entire callflow, including the transfering, maybe __TRANSFER_CONTEXT=mtransfer |
| 07:37.13 | Sweeper | ahh |
| 07:37.21 | Sweeper | yea, chan var sounds good |
| 07:37.47 | *** join/#asterisk axscode (n=axscode@73.192.48.60.klj02-home.tm.net.my) |
| 07:38.16 | awk | I still want to know how other people get this right |
| 07:38.29 | awk | I guess I could write an agi script but that would take quite some time.. |
| 07:38.33 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
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| 07:54.49 | obnauticus | <PROTECTED> |
| 07:55.04 | obnauticus | kind of like using VoipJet with IpKall but something that has imbound included in the service too. |
| 07:55.50 | Strom_M | teliax.com |
| 07:56.25 | obnauticus | Is there one that is `pay as you go' sort of speak. |
| 07:56.28 | JT | awk: how would a chan var help? |
| 07:56.30 | obnauticus | one that doesn't require a subscription |
| 07:56.39 | Strom_M | teliax is pay as you go |
| 07:56.46 | Strom_M | read the page before jumping to conclusions |
| 07:56.48 | obnauticus | oh |
| 07:56.49 | obnauticus | i didn't see |
| 07:56.50 | obnauticus | lol. |
| 07:56.54 | obnauticus | i just looked at the main page, sorry |
| 07:57.16 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 07:59.02 | obnauticus | Storm_M I can change my ANI with this service, so I can just change it to my IPKall # and not have to pay for it.. |
| 07:59.10 | obnauticus | but then i minus well be getting voipjet :/ |
| 08:00.08 | obnauticus | Oh I see |
| 08:00.14 | obnauticus | this is pretty good Storm_M, thanks. |
| 08:01.31 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkgroup.com.au) |
| 08:01.36 | Strom_M | who the hell is storm? |
| 08:01.48 | Strom_M | and why are you so averse to paying for phone service? |
| 08:01.50 | obnauticus | My tab complete isn't working too well through this BNC with network name. |
| 08:01.53 | Strom_M | do you steal electricity too? |
| 08:01.59 | obnauticus | Because I'm 16. |
| 08:02.23 | Strom_M | good. perfect opportunity to get a part-time job |
| 08:02.29 | obnauticus | I have a part time job. |
| 08:02.35 | Strom_M | one day a month will pay for your telephone service |
| 08:02.45 | obnauticus | I like to be thrifty, i don't necessarily need the best of things if I just want to learn how to use this stuff and be able to use it. |
| 08:02.55 | obnauticus | through the PSTN. |
| 08:03.02 | Strom_M | theft in the name of learning is still theft |
| 08:03.10 | endre | lmao |
| 08:03.17 | Strom_M | it's two god damned cents per minute. |
| 08:03.20 | obnauticus | ARe you saying that being cheap is equivilant to stealing? |
| 08:03.36 | Strom_M | no, i'm saying that theft of service is equivalent to stealing |
| 08:03.44 | obnauticus | Who said i was steling service? |
| 08:03.49 | Strom_M | you did |
| 08:03.52 | obnauticus | Wtf where? |
| 08:04.04 | Strom_M | <obnauticus> Storm_M I can change my ANI with this service, so I can just change it to my IPKall # and not have to pay for it.. |
| 08:04.23 | obnauticus | I didn't know that they didn't offer a plan WITHOUT an incoming line. |
| 08:04.29 | obnauticus | they did* |
| 08:04.42 | obnauticus | Therefore, changing my ANI to my other line would NOT be stealing in that situation, would it. |
| 08:04.59 | Strom_M | do you know the difference between ANI and calling party number? |
| 08:05.05 | obnauticus | Yes. |
| 08:05.20 | obnauticus | It helps people just look at the number and then call it manually. |
| 08:05.22 | *** join/#asterisk harpal (n=Harpal@124.125.255.223) |
| 08:05.26 | Strom_M | uh |
| 08:05.31 | obnauticus | i said that incorrectly. |
| 08:05.33 | obnauticus | i mean like |
| 08:05.35 | obnauticus | if i were to change my ANI |
| 08:05.41 | obnauticus | I know the difference ok. |
| 08:05.41 | Strom_M | ~101 |
| 08:05.42 | jbot | i guess 101 is Telephony 101, which is a good read if you're unfamiliar with traditional TDM telephony. You can download it at http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf |
| 08:05.42 | obnauticus | lol. |
| 08:05.51 | Strom_M | go read that |
| 08:05.51 | obnauticus | I know what ANI is dude. |
| 08:06.00 | obnauticus | And I have already read it. |
| 08:06.19 | Strom_M | ok, so the problem is that you're just a horrid communicator |
| 08:06.29 | obnauticus | I'm quadrouple tasking right now. |
| 08:06.29 | BeeBuu | when i retrydial(),there is "Music class default requested but no musiconhold loaded." in CLI> |
| 08:06.32 | BeeBuu | anyone tell me what's that mean? thanks. |
| 08:06.32 | Strom_M | because clearly i'm misunderstanding you left and right |
| 08:06.44 | obnauticus | Ya, I don't blame you, I don't even understand what I am saying. |
| 08:06.51 | Strom_M | *blink* |
| 08:06.56 | Strom_M | go away and come back when you can be coherent |
| 08:06.59 | Strom_M | thanks! |
| 08:07.02 | Strom_M | -- the management |
| 08:07.03 | obnauticus | in the morning :/ |
| 08:07.03 | obnauticus | k |
| 08:09.52 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 08:12.15 | BeeBuu | what is "Music class default requested but no musiconhold loaded." mean? |
| 08:16.09 | *** join/#asterisk SuurMyy (n=SuurMyy_@195.238.211.98) |
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| 08:42.21 | J4zen | Any asterisk gurus building voip systems in holland? |
| 08:45.26 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:47.53 | *** join/#asterisk jhiver (i=jhiver@164-242.206-83.static-ip.oleane.fr) |
| 08:48.33 | jhiver | Hi, I've just installed Asterisk 1.4.13... 'sip show peers' doesn't work anymore! |
| 08:48.46 | jhiver | synapseuk*CLI> sip show peers |
| 08:48.47 | jhiver | No such command 'sip show' (type 'help' for help) |
| 08:49.05 | jhiver | and of course doing asterisk -rx 'help' |grep sip yields NOTHING =( |
| 08:49.08 | jhiver | any ideas? |
| 08:52.11 | jhiver | found it =) config issue, no modules loaded |
| 08:53.26 | *** join/#asterisk slowshutt (n=d@196.211.34.2) |
| 08:55.41 | slowshutt | Hi there i created my own wer server where all my voicemail files are stored, for easy management, i have the problem where i can not delete the messages from the voicemailbox be cause asterisk writes with read only permissions, where and what must one edit to get asterisk to write it with diferent permissions? any help welcom |
| 08:57.05 | slowshutt | Hi there i created my own web server where all my voicemail files are stored, for easy management, i have the problem where i can not delete the messages from the voicemailbox because asterisk writes it with read only permissions, where and what must one edit to get asterisk to write it with diferent permissions? any help welcome please |
| 09:06.02 | *** join/#asterisk _ys (n=yuri@80.70.236.69) |
| 09:11.04 | *** join/#asterisk qdk (n=qdk@85.235.253.139) |
| 09:13.25 | *** join/#asterisk Greek-Boy (n=email@41.221.58.2) |
| 09:13.27 | Greek-Boy | Good morning |
| 09:13.46 | BBHoss | sup |
| 09:17.03 | Greek-Boy | nm |
| 09:17.07 | Greek-Boy | u? |
| 09:17.20 | BBHoss | bout to go to sleep :) |
| 09:17.22 | Greek-Boy | i'm just struggling with my linksys SPA942 mass deployment |
| 09:17.44 | BBHoss | do they even do provisioning? |
| 09:17.59 | BBHoss | oh |
| 09:18.05 | Greek-Boy | yip they do |
| 09:18.06 | BBHoss | ip phones |
| 09:18.18 | BBHoss | for some reason i was thinking ATAs |
| 09:18.20 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 09:18.41 | Greek-Boy | nah, phones |
| 09:19.06 | Greek-Boy | some of the firmware supported different config file formats. some required xml |
| 09:19.24 | BBHoss | its probably best to have ALL the same firms |
| 09:20.23 | Greek-Boy | yi |
| 09:21.02 | BBHoss | this tool apparently works well |
| 09:21.02 | BBHoss | http://spc.pifiu.com/ |
| 09:21.56 | Greek-Boy | yeah I've got it |
| 09:22.05 | Greek-Boy | but I cant figure out what to do with it |
| 09:22.18 | Greek-Boy | allI did is generate a profile example out of it |
| 09:22.30 | BBHoss | ive personally never played with any linksys |
| 09:23.38 | Greek-Boy | oh |
| 09:23.50 | Greek-Boy | thanks for your help anyway :) |
| 09:25.45 | *** join/#asterisk ghento (n=ghento@75.155.241.7) |
| 09:26.52 | BBHoss | <PROTECTED> |
| 09:26.52 | BBHoss | parameter-value pairs into a properly formatted and encrypted .cfg file. |
| 09:27.12 | *** join/#asterisk Narkov- (n=Narkov@c220-237-77-174.kelvn1.qld.optusnet.com.au) |
| 09:27.45 | Narkov- | is 200 active calls "magic" in Asterisk? I can't seem to get over it even though seemingly having plentiful resources |
| 09:28.12 | BBHoss | what do you mean by plentiful |
| 09:28.28 | Narkov- | low cpu...close to nill disk...plenty of bandwidth |
| 09:28.51 | Narkov- | plenty of resources on terminating end |
| 09:28.54 | Narkov- | call quality excellent |
| 09:29.20 | Narkov- | manager events seem to be ignored after ~200 calls |
| 09:31.19 | *** join/#asterisk saftsack (n=saftsack@s0338.vpn.hrz.tu-darmstadt.de) |
| 09:32.06 | Narkov- | any ideas? |
| 09:34.05 | J4zen | Has anyone ever used the services of a company called gnTel? |
| 09:35.26 | *** join/#asterisk sergey (n=sergey@91.189.233.71) |
| 09:41.26 | slowshutt | Hi there i created my own wer server where all my voicemail files are stored, for easy management, i have the problem where i can not delete the messages from the voicemailbox be cause asterisk writes with read only permissions, where and what must one edit to get asterisk to write it with diferent permissions? any help welcom |
| 09:45.05 | slowshutt | yes J4zen |
| 09:45.17 | slowshutt | what do you want to know about them? |
| 09:46.29 | *** join/#asterisk ectospasm (n=ectospas@c-68-62-219-11.hsd1.al.comcast.net) |
| 09:46.33 | slowshutt | if anyone can read this please say so |
| 09:46.41 | Narkov- | hi |
| 09:46.47 | Narkov- | slowshutt: copy :) |
| 09:46.56 | slowshutt | thx |
| 09:47.14 | slowshutt | i thought my mirc is acting up |
| 09:47.19 | slowshutt | quite today |
| 09:47.38 | *** join/#asterisk davevg-btwtech (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net) |
| 09:47.42 | slowshutt | whats up Narkov-? |
| 09:48.39 | *** join/#asterisk agallo (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 09:48.42 | Narkov- | nothing much mate...you? |
| 09:52.53 | slowshutt | are you a asterisk guru? Narkov-? |
| 09:54.12 | Narkov- | Hrmm...define guru? I derive an income from Asterisk....I'm not a code monkey but I am an system engineer |
| 09:54.41 | Narkov- | yourself slowshutt? |
| 09:56.02 | slowshutt | kinda know my way around, but stuck on something simple |
| 09:56.10 | slowshutt | you mind trying to help? |
| 09:56.16 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 09:56.31 | Narkov- | slowshutt: I can try...whats the issue? |
| 09:57.28 | Narkov- | got it |
| 09:57.38 | Narkov- | have you tried changing the directory permissions? |
| 09:57.51 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 09:58.16 | slowshutt | app_voicemail writes the files in var/spool/asterisk.voicemail/default folder i need to change the permisions on the file it writes in order for me to display it in my webpage i need it to change the permission to write to enable me to delete and edit the messages |
| 09:58.24 | slowshutt | yes |
| 09:58.44 | slowshutt | does not work |
| 09:59.04 | slowshutt | asterisk writes the permision to every wav file recorded to read only |
| 09:59.35 | slowshutt | meens i have to manualy go set the permisions for new file |
| 09:59.48 | Narkov- | tried changing the umask for asterisk? |
| 10:00.00 | Narkov- | it should be system based and not an Asterisk stting |
| 10:00.04 | Narkov- | setting |
| 10:01.47 | slowshutt | now you lost me what does umask mean |
| 10:02.35 | Narkov- | it sets the default permissions for file creation |
| 10:02.50 | slowshutt | is this a linux feature? |
| 10:02.59 | slowshutt | where do i do this? |
| 10:03.02 | Narkov- | i believe you should be able to set it for a particular user (ie. asterisk) |
| 10:03.10 | Narkov- | what distro slowshutt? |
| 10:05.15 | slowshutt | suse 10.2 |
| 10:08.15 | *** join/#asterisk nfi|ermes (n=ErMeS@217.220.121.62) |
| 10:10.58 | *** join/#asterisk salzh (n=salzh@124.77.5.180) |
| 10:11.43 | slowshutt | did i scare you off Narkov- |
| 10:11.46 | slowshutt | ? |
| 10:12.08 | JT | sounds like normal fear ;) |
| 10:13.16 | Narkov- | slowshutt: sorry mate...i get destracted easily |
| 10:13.19 | Narkov- | slowshutt: http://lists.suse.com/archives/suse-sles-e/2006-Jul/0026.html |
| 10:13.54 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 10:28.08 | Narkov- | does anone know about Asterisk ignoring managr events when it gets busy? |
| 10:30.50 | *** join/#asterisk d00gster (n=doughant@bas1-toronto12-1128685020.dsl.bell.ca) |
| 10:31.49 | J4zen | What would it mean if sip debug my_voip_trunk would result in : |
| 10:31.49 | J4zen | <PROTECTED> |
| 10:31.49 | J4zen | <PROTECTED> |
| 10:31.49 | J4zen | <PROTECTED> |
| 10:31.50 | J4zen | SIP/2.0 407 Proxy Authentication Required |
| 10:31.56 | J4zen | After recieving a call |
| 10:32.04 | J4zen | or ehm, While recieving a call |
| 10:32.08 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 10:32.15 | *** join/#asterisk thieums (n=Mathieu@rny93-4-82-231-54-139.fbx.proxad.net) |
| 10:32.19 | J4zen | I call the number connected the the VOIP trunk, but immediatly after it is answered..it is hung up |
| 10:32.23 | J4zen | my SIP debug shows the above |
| 10:33.11 | *** join/#asterisk Steven_elvisda_ (n=Steven_E@202.47.107.60) |
| 10:33.19 | slowshutt | what codex are you running J4zen? |
| 10:33.31 | J4zen | oh |
| 10:33.32 | J4zen | actually |
| 10:33.35 | J4zen | i just noticed |
| 10:33.42 | J4zen | after the 407, it actually authenticates for the proxy |
| 10:33.50 | J4zen | and then returns a SIP/2.0 403 Forbidden for the actual call |
| 10:34.16 | thieums | hello, do you know how to map a specific source ip address on an outbound iax call ? |
| 10:35.10 | slowshutt | you are not connected are you using register => ???:???@ashdashd.co.zxa:5060\xxxx ?? j4shen? |
| 10:35.25 | J4zen | not the portnumber and not the DID number |
| 10:35.40 | J4zen | but, if im not registered i shouldnt be able to make any outbound calls either |
| 10:35.45 | J4zen | and that.. i can |
| 10:36.27 | slowshutt | past your sip.conf and extension.conf in pastbin and i will ry and help you |
| 10:36.40 | J4zen | Thanks :) |
| 10:36.53 | slowshutt | are you using iax or sip to connect to remote site? |
| 10:36.57 | J4zen | sip |
| 10:37.57 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 10:39.09 | slowshutt | k past lets look |
| 10:39.17 | J4zen | Working on it :) |
| 10:39.24 | slowshutt | k |
| 10:39.29 | Greek-Boy | im using firmware 5.1.15a for linksys spa942. I want to do mass provisioning but can't figure out which configuration file format to use with that firmware version. I'm using the latest SPC tool. |
| 10:39.46 | J4zen | its rather large so i'm sorting what stuff might be interesting to you and what not lol |
| 10:40.46 | Greek-Boy | and I cant find older firmware out there |
| 10:43.11 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 10:43.50 | *** join/#asterisk d00gster (n=doughant@bas1-toronto12-1128685020.dsl.bell.ca) |
| 10:47.50 | J4zen | slowshutt: I've sent you a PM |
| 10:48.41 | slowshutt | k |
| 10:52.45 | *** join/#asterisk michael-i (n=michael-@Lf5d7.l.pppool.de) |
| 10:55.28 | *** join/#asterisk zapp-branigan (n=zapp-bra@140.23.220.87.dynamic.jazztel.es) |
| 10:56.50 | zapp-branigan | hi i have a problem i can use the (!) in the iax.conf in a asterisk free version for the users ? |
| 10:59.56 | Strom_M | huh? |
| 11:04.39 | *** part/#asterisk Narkov- (n=Narkov@c220-237-77-174.kelvn1.qld.optusnet.com.au) |
| 11:06.13 | *** join/#asterisk _ys (i=ys@91.151.196.254) |
| 11:08.15 | FlatFoot | mornign all |
| 11:12.40 | thieums | somebody could please help me with my iax problem ? i've got a multi ip box and i want to specify the ip source address on outbound iax calls, do you know how to proceed ? |
| 11:13.09 | *** join/#asterisk cisplatine (n=chatzill@83.145.94.46) |
| 11:14.21 | *** join/#asterisk viperdudeuk (n=chatzill@84-45-129-190.no-dns-yet.enta.net) |
| 11:14.45 | cisplatine | Hi there, I have a question about the GXP2000 provisioning, if anyone is interested, thx |
| 11:14.54 | FlatFoot | just installed zaptel 1.4.6 and modprobe does not seem to be there . Has it changed aswell as a lot of other stuff since i last built a * ??? |
| 11:15.16 | Strom_M | FlatFoot: modprobe is part of linux, not zaptel... |
| 11:15.39 | FlatFoot | Strom_M: see that'll be me being a plank again |
| 11:16.29 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 11:16.39 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 11:17.23 | Strom_M | a plank? |
| 11:18.14 | FlatFoot | Strom_M: sorry south eastern England expression for an idiot |
| 11:18.25 | Strom_M | ah, heh |
| 11:18.34 | Strom_M | didn't hear that one whilst I was in London :) |
| 11:18.38 | FlatFoot | jbot might like to know that one |
| 11:18.47 | FlatFoot | down in glorious KENT |
| 11:18.57 | viperdudeuk | hi |
| 11:18.59 | cisplatine | does anyone know how to enable SIP compact-header with gxp2000 via a prov file ? I found some old release notes, but looks outdated |
| 11:19.03 | viperdudeuk | from sunny shropshire |
| 11:19.17 | Strom_M | <---- Los Angeles |
| 11:19.18 | FlatFoot | is it nice and chilly up there ? |
| 11:19.29 | FlatFoot | Strom_M: bet thats warm |
| 11:19.36 | viperdudeuk | dunno in a warm office :-) |
| 11:19.56 | FlatFoot | viperdudeuk: warm office that would be nice |
| 11:19.59 | Strom_M | today could have been t-shirt and shorts weather |
| 11:20.22 | FlatFoot | Strom_M: see now thats just making me annoyed ;) |
| 11:20.42 | Strom_M | right now, at 3:20 AM, it's 18 centigrade |
| 11:20.44 | viperdudeuk | has anyone had problems with linksys 942 transferring calls? |
| 11:21.04 | FlatFoot | Strom_M: 3:20am ( thats bed time ) |
| 11:21.07 | Strom_M | just chilly enough to warrant wearing a light jacket |
| 11:21.10 | DrAk0 | is there any way of doing n-way calls ? |
| 11:21.23 | Strom_M | yeah, i decided to pop out just now and get a burrito at one of the local taco trucks |
| 11:21.29 | Strom_M | so delicious :D |
| 11:21.42 | viperdudeuk | asterisk says both call legs must reside on asterisk box |
| 11:21.50 | *** join/#asterisk saftsack (n=saftsack@pD9E07653.dip.t-dialin.net) |
| 11:22.11 | saftsack | hi, is someone else here who is out of germany? |
| 11:23.05 | DrAk0 | here 7:22am, 26C outside, my room 16C |
| 11:23.24 | FlatFoot | DrAk0: where are you then ? |
| 11:23.33 | DrAk0 | Margarita Island, Venezuela. |
| 11:25.28 | saftsack | is there a national telephony flat available for business customers? because most flats here in germany arent available for business customers .. |
| 11:26.32 | Strom_M | saftsack: that's because business customers generally make a metric crapload more calls than residential customers :) |
| 11:27.15 | saftsack | yes i know that ;) and the telcos know that too ;) but isnt there any telco which sells a flatrate for more money/month? |
| 11:27.24 | FlatFoot | Strom_M; i found i needed kldstat instead |
| 11:28.12 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 11:33.25 | *** join/#asterisk coppice (n=chatzill@102.204.17.210.dyn.pacific.net.hk) |
| 11:33.54 | slowshutt | anayone know how to change the voicemail app to write wave file with permision 777? |
| 11:34.55 | coppice | that's a very strange permission for a wave file |
| 11:35.34 | *** join/#asterisk MicW (n=michael@dslb-088-074-149-043.pools.arcor-ip.net) |
| 11:35.35 | MicW | hi |
| 11:36.18 | MicW | i'm running asterisk 1.4.11 with snom phones and i use hints. interestingly currently 2 phone-leds are flashing. the hint (show hints) shows "Hold". |
| 11:36.25 | MicW | but there's no hold on the phones |
| 11:43.02 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 11:43.12 | *** part/#asterisk robl^ (n=robl@pdpc/supporter/monthlybyte/robl) |
| 11:43.13 | slowshutt | can you maybe point me in right direction coppice |
| 11:44.04 | coppice | trying to use 777 is definitely a bad direction |
| 11:44.21 | slowshutt | what do you recommend? |
| 11:45.24 | slowshutt | but where can you ammend it? |
| 11:45.27 | coppice | well, 777 would make the file executable. sounds kinda inappropriate |
| 11:45.30 | MicW | is there a way to force a refresh of the "hints" list? |
| 11:46.12 | slowshutt | where would change the permissions for the voicemail application? coppice? |
| 11:53.30 | *** join/#asterisk Steven_elvisda (n=Steven_E@202.47.107.60) |
| 11:59.32 | *** join/#asterisk bantu (n=Miranda@p54A32B11.dip0.t-ipconnect.de) |
| 12:07.01 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 12:07.04 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 12:17.55 | *** join/#asterisk gerhard7 (n=gerhard@82-170-9-68.dsl.ip.tiscali.nl) |
| 12:18.10 | *** join/#asterisk duckz (n=duckz@85.204.47.228) |
| 12:22.20 | *** join/#asterisk salog (n=Salog@h081217000091.dyn.cm.kabsi.at) |
| 12:26.45 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 12:27.24 | salog | Hi! My sipprovider overwrites SetCallerID, so I need to dial #31# in front of my number, to hide my ID. When I try this out from a sip client, it works, but using "exten => _XXX.,1,Dial(SIP/#31#${EXTEN}@provider)" results in a circuit-busy. Is there anything I need to take care of? Do I need to escape #, or anything else? |
| 12:40.06 | *** join/#asterisk ussrback (n=MAX@87.253.33.50) |
| 12:40.15 | ussrback | Hi all |
| 12:40.29 | ussrback | How can i install H323 on my asterisk 1.4.13? |
| 12:41.10 | tzafrir | ussrback, asterisk has chan_h323 that depends on opeh323 . There's also an independen chan_ooh323c in asterisk-addons |
| 12:41.18 | tzafrir | what distro do you use? |
| 12:41.53 | ussrback | i use 1.4.13 |
| 12:42.10 | ussrback | so should i install some additional software to use H323? |
| 12:43.06 | *** join/#asterisk oej (n=olle@gw-sthlm01.rebtel.com) |
| 12:43.10 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.106.70) |
| 12:43.31 | ussrback | should i install open323 and then use chan_323 ? |
| 12:44.24 | JT | ussrback: you should use chan_woomera |
| 12:45.15 | tzafrir | ussrback, what linux distribution (or what OS) do you use? |
| 12:45.31 | ussrback | i use EL-4 |
| 12:46.10 | tzafrir | I figure that chan_ooh323c is the path of least resistance. Not really sure about the quality of both |
| 12:46.21 | tzafrir | or of woomera/opal |
| 12:46.29 | JT | both chan_h323 and chan_oh323c are complete disasters |
| 12:46.38 | JT | woomera is the least of a disaster |
| 12:46.50 | ussrback | so what is a better module for h323? |
| 12:47.15 | tzafrir | well, I'm not sure how much people are familiar with woomera around here |
| 12:47.30 | JT | tzafrir: about as much as they are familiar with the others |
| 12:47.53 | coppice | if you know about woomera, does that make you a rocket scientist? |
| 12:49.02 | ussrback | so may be its better to use GNUGK for H323 connection? |
| 12:49.29 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 12:49.31 | tzafrir | use SIP... |
| 12:49.42 | tzafrir | sorry |
| 12:50.02 | ussrback | no i need H323 connection with some operator |
| 12:51.50 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 12:51.55 | *** join/#asterisk Bladerunner05 (n=feelme@81-174-56-54.static.ngi.it) |
| 12:54.20 | *** join/#asterisk shay|work (n=shay@unaffiliated/shay) |
| 12:54.25 | shay|work | Hello folks |
| 12:54.28 | JT | ussrback: yes, GnuGK would be much better |
| 12:54.36 | JT | tzafrir: such an asteriskism ;) |
| 12:55.28 | Bladerunner05 | to download asterisk via svn I have to use trunk ? |
| 12:55.46 | JT | Bladerunner05: no |
| 12:56.23 | Bladerunner05 | JS: and wich one ? |
| 12:57.27 | tzafrir | Bladerunner05, branches/1.4 is the branch from which the next 1.4 release (1.4.14) will be released |
| 12:57.51 | Bladerunner05 | •tzafrir• thanks |
| 12:57.58 | tzafrir | (the svn also has tags for each released tarball, but getting them is a bit pointless) |
| 13:02.05 | Bladerunner05 | •tzafrir• is this correct svn checkout http://svn.digium.com/svn/asterisk/branches/1.4/ asterisk |
| 13:02.20 | tzafrir | yes |
| 13:02.45 | Bladerunner05 | I do the same for zaptel.... etc etc... ? |
| 13:02.49 | tzafrir | Bladerunner05, you can also try simply browsing tp://svn.digium.com/svn/asterisk/branches/1.4/ |
| 13:02.53 | tzafrir | Bladerunner05, yes |
| 13:03.05 | tzafrir | note that with Zaptel you *should not* use trunk |
| 13:03.26 | *** join/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 13:04.13 | tzafrir | browsing http://svn.digium.com/svn/asterisk/branches/1.4/ , that is. The URL you intend to checkout |
| 13:04.24 | Bladerunner05 | •tzafrir• doing svn checkout http://svn.digium.com/svn/asterisk/branches/1.4/ asterisk it will download all the folders ? |
| 13:05.03 | tzafrir | yes |
| 13:05.03 | Bladerunner05 | ok |
| 13:05.05 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 13:05.21 | *** join/#asterisk agx (n=AGX@88.34.216.63) |
| 13:13.24 | *** join/#asterisk tim|macbook (n=tim@office.sit.kumina.nl) |
| 13:14.46 | *** join/#asterisk eurodrew555 (n=eurodrew@83-244-164-130.cust-83.exponential-e.net) |
| 13:16.29 | *** join/#asterisk Tourinho (n=tourinho@cm-virtua-poa-C8B0F83B.dynamic.brdterra.com.br) |
| 13:16.39 | Tourinho | good day |
| 13:17.20 | *** join/#asterisk ussrback (i=MAX@87.253.33.236) |
| 13:18.04 | codefreeze | awk: about your billing/cdr/xfer problem; if you figure out something neat, and it works, let me know, ok? |
| 13:20.00 | slowshutt | hi there can anyone help me please i am connecting to a paid service provider, can not connect i think it is because of the g729 licence they insist me on having how does this work and how do you install it? |
| 13:20.43 | viperdudeuk | slowshutt: goto to Digiums site, buy the licences and follow install instructions |
| 13:23.29 | *** join/#asterisk robl^ (n=robl@pdpc/supporter/monthlybyte/robl) |
| 13:23.56 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:23.56 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 13:26.40 | slowshutt | if i have one sip account haw many licences must i buy, they relate it to channels? |
| 13:26.56 | *** join/#asterisk zerohalo (n=zeroHalo@pool-71-255-162-167.bstnma.east.verizon.net) |
| 13:27.39 | viperdudeuk | 1 per channel |
| 13:28.54 | eurodrew555 | hi |
| 13:29.06 | eurodrew555 | we are looking at implementing multiple domains within Asterisk |
| 13:29.32 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 13:29.35 | eurodrew555 | so we can handle multiple separate company's using different domains |
| 13:29.41 | slowshutt | meaning if i hava i iax connection to another server and i have 2 calls between the two i would requir 4 licences? |
| 13:29.43 | eurodrew555 | found a bit of a prob though |
| 13:29.56 | eurodrew555 | while asterisk easily handles multiple domains when making outbound calls |
| 13:30.14 | viperdudeuk | eurodrew555: i am not sure asterisk can do this... I use openser for multiple domains in front of asterisk |
| 13:30.21 | eurodrew555 | there doesn't seem to be a concept of multiple domains for inbound calls |
| 13:30.23 | awk | codefreeze well my issue is how to bill transfered calls |
| 13:30.28 | eurodrew555 | we have got OpenSER as our front end |
| 13:30.33 | eurodrew555 | and it handles the multiple domains OK |
| 13:30.39 | viperdudeuk | ok |
| 13:30.42 | eurodrew555 | but we forward everything to asterisk for call processing and voicemail |
| 13:30.48 | viperdudeuk | ok |
| 13:30.57 | viperdudeuk | this is also what I do |
| 13:31.06 | eurodrew555 | but we have found for example... asterisk realtime is always just doing a select for just the username |
| 13:31.19 | viperdudeuk | but i use a agi script to strip out the domain part |
| 13:31.23 | eurodrew555 | so our problem is if we have user@companya and user@companyb |
| 13:31.28 | viperdudeuk | to identify the exten on that domain |
| 13:31.33 | viperdudeuk | i dont use realtime |
| 13:31.39 | eurodrew555 | can asterisk even handle that? |
| 13:31.51 | viperdudeuk | i custom wrote all my stuff,, much more control that way |
| 13:32.10 | eurodrew555 | sip show peers for example doesn't show up domains.. so you would have multiple users with the same username, which I don't think is going to work |
| 13:32.23 | viperdudeuk | eurodrew555: it can handle the multi domains if you use agi |
| 13:32.28 | slowshutt | can there licences be linked to iax channels? |
| 13:32.51 | viperdudeuk | g729 license works across all channels |
| 13:33.15 | eurodrew555 | ok that might work, but what about registrations to the asterisk boxes |
| 13:33.25 | viperdudeuk | its the realtime app that *CANT* handle multi domains AFAIK |
| 13:33.33 | viperdudeuk | registrations are to openser |
| 13:33.44 | eurodrew555 | we forward on all our registrations using sipsak |
| 13:33.46 | eurodrew555 | it works very well |
| 13:34.03 | eurodrew555 | we have multiple asterisk boxes behind openser for redundancy+capacity |
| 13:34.24 | viperdudeuk | so do i |
| 13:34.51 | eurodrew555 | ok I'm starting to think we might have to do most of this within OpenSER |
| 13:35.06 | viperdudeuk | i use a shell script that decides what box to send to using exec_dset |
| 13:35.19 | eurodrew555 | so if we get user@companyb we will have to send the registration back as user_companyb |
| 13:35.32 | viperdudeuk | we have virtual domains working very well here across multiple boxes |
| 13:35.33 | eurodrew555 | so all usernames on asterisk will be unique, I suppose that would get around this |
| 13:35.48 | viperdudeuk | i dont have *ANY* usenames on asterisk |
| 13:36.27 | eurodrew555 | sounds quite different to our setup |
| 13:36.27 | viperdudeuk | when i want to call a extension its like 501@cust1.mydomain.net |
| 13:36.36 | viperdudeuk | then asterisk forwards it to openser |
| 13:36.59 | viperdudeuk | becauase their is a peer for cust1.mydomain.net in the sip.conf |
| 13:37.05 | eurodrew555 | we send all SIP through openser, and all registrations get replicated to all asterisk servers |
| 13:37.13 | viperdudeuk | ouch |
| 13:37.23 | viperdudeuk | asterisk is bad for lots of registrations |
| 13:37.31 | eurodrew555 | we don't see any problems |
| 13:37.42 | viperdudeuk | how many extens? |
| 13:37.46 | eurodrew555 | but then whats a lot of registrations? we've had at most say 100 sip users online |
| 13:37.53 | viperdudeuk | i ma talking 10000's |
| 13:37.56 | eurodrew555 | aha |
| 13:38.04 | eurodrew555 | big difference there! |
| 13:38.14 | viperdudeuk | no need to register on asterisk if using openser |
| 13:39.08 | eurodrew555 | how many asterisk boxes have you got behind openser? how many E1s on each box? |
| 13:39.30 | eurodrew555 | we have 3x asterisk (dual xeon 3.4) with quad E1 (sangoma) |
| 13:39.38 | viperdudeuk | i have around 10 asterisk boxs behind ser... we dont use E1's |
| 13:39.50 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:39.53 | viperdudeuk | i send out to PSTN via SIP to another upstream provider |
| 13:40.06 | eurodrew555 | in the process of moving all our users & call centre from Avaya Definity over to OpenSER/Asterisk |
| 13:40.21 | eurodrew555 | hmm so pure SIP no ISDN anywhere? |
| 13:40.28 | viperdudeuk | thats right |
| 13:41.10 | eurodrew555 | we are now looking at handling other companies comms through our setup, thats why we are looking at handling multiple domains |
| 13:41.37 | *** join/#asterisk oej (n=olle@gw-sthlm01.rebtel.com) |
| 13:42.15 | eurodrew555 | everything works well for our current setup and don't see any issues with registrations, but again we don't have that many users on at present |
| 13:42.33 | eurodrew555 | probably 40 max now, but we have about 300 to move over from our Avaya |
| 13:43.56 | eurodrew555 | thanks for the info viperdudeuk, off for lunch now |
| 13:44.02 | eurodrew555 | cheers |
| 13:46.16 | *** part/#asterisk tim|macbook (n=tim@office.sit.kumina.nl) |
| 13:47.19 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 13:48.33 | *** join/#asterisk twans (n=arcraig@2001:470:1f05:d3:91f3:bca8:c890:a756) |
| 13:50.50 | *** join/#asterisk Rusty1 (n=Rusty1@cpe-72-226-109-62.nycap.res.rr.com) |
| 13:52.37 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 13:53.41 | *** part/#asterisk Rusty1 (n=Rusty1@cpe-72-226-109-62.nycap.res.rr.com) |
| 13:55.04 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 13:56.35 | puzzled | hi |
| 13:59.26 | *** join/#asterisk penguinFunk (n=penguin@unaffiliated/penguinfunk) |
| 14:03.30 | TJNII | Morning |
| 14:04.37 | J4zen | I was just doing some logging , checking how the communication worked between Asterisk and a VOIP-Trunk provider |
| 14:04.47 | J4zen | and i noticed that it repeats two data-packets over and over again |
| 14:04.51 | J4zen | it doesn't seem right |
| 14:04.52 | J4zen | http://pastebin.com/m1b6a8037 |
| 14:04.55 | J4zen | Thats the two packets |
| 14:05.01 | J4zen | it sends a request, and recieves a 404 |
| 14:05.13 | J4zen | no active lines on the pbx |
| 14:05.16 | J4zen | no ongoing calls |
| 14:05.17 | J4zen | no nothing |
| 14:05.20 | J4zen | why does it do that? |
| 14:05.26 | J4zen | and should it be doing that? |
| 14:05.45 | file | you have qualify=yes set |
| 14:06.03 | file | Asterisk sends an OPTIONS packet to see whether the other side is there or not, and how long it takes it to process/respond back |
| 14:06.21 | J4zen | But what is this "unknown" and why does the server respond with a 404? |
| 14:06.36 | J4zen | # |
| 14:06.37 | J4zen | Reliably Transmitting (no NAT) to 194.140.246.34:5060: |
| 14:06.37 | J4zen | # |
| 14:06.37 | J4zen | OPTIONS sip:voip.zb.gntel.nl SIP/2.0 |
| 14:06.37 | J4zen | # |
| 14:06.37 | J4zen | Via: SIP/2.0/UDP 81.18.164.73:5060;branch=z9hG4bK484e8538;rport |
| 14:06.39 | J4zen | # |
| 14:06.39 | file | Unknown is the callerid value if none is specified, which in this case is perfectly normal |
| 14:06.41 | J4zen | From: "Unknown" <sip:Unknown@81.18.164.73>;tag=as3a7cee2e |
| 14:06.46 | J4zen | I see |
| 14:06.53 | file | as for the remote server that is how it is configured to respond, or it's implementation does that automatically |
| 14:06.53 | J4zen | so it supplies unknown if its a broadcast of some sort |
| 14:07.09 | J4zen | Great, gave me some more insight :) |
| 14:07.12 | J4zen | Thanks @file |
| 14:08.37 | J4zen | Qualify = yes , is only required if the PBX is behind a NAT right? |
| 14:08.43 | *** join/#asterisk oej (n=olle@gw-sthlm01.rebtel.com) |
| 14:09.04 | *** join/#asterisk zydrunas (n=Miranda@216.49.228-ip-184.ckt.net) |
| 14:09.28 | Bladerunner05 | using s |
| 14:10.05 | Bladerunner05 | using svn asterisk and spandsp0.0.4 when I receive a fax after 10 seconds return *** glibc detected *** ./asterisk-bin/sbin/asterisk: corrupted double-linked list: 0x081b8d60 *** |
| 14:10.11 | Bladerunner05 | and a lot of other lines |
| 14:10.51 | [TK]D-Fender | J4zen: Yes, primarily for NAT but also good to inform * about hosts that may drop out without de-registering. |
| 14:11.07 | [TK]D-Fender | J4zen: But I typically advise against it for ITSP's |
| 14:14.33 | J4zen | [TK]D-Fender; Thanks |
| 14:16.33 | J4zen | What is generally misconfigured if my PBX cannot recieve incoming calls, sip debug shows this error first: |
| 14:16.34 | J4zen | SIP/2.0 407 Proxy Authentication Required |
| 14:16.53 | J4zen | Followed by a SIP/2.0 403 Forbidden |
| 14:20.02 | [TK]D-Fender | J4zen: pastebin your full CLI output w/ SIP debug & configs masking only passwords |
| 14:20.26 | J4zen | Will do. Give me one moment |
| 14:21.42 | J4zen | Heres my logfile: http://pastebin.com/m2dd46010 |
| 14:22.19 | *** join/#asterisk asteriskmonkey (n=philip@69.77.169.14) |
| 14:22.59 | asteriskmonkey | is there anyway or a patch to set individual mailbox max messeages in asterisk voicemail? |
| 14:23.53 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:25.13 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 14:27.37 | J4zen | SIP.conf and such: http://pastebin.com/m2df9018f |
| 14:29.20 | [TK]D-Fender | J4zen: [VOIP Trunk(gNtel)] <- this really hold something else? |
| 14:29.34 | J4zen | it holds |
| 14:29.34 | J4zen | [from-trunk-VOIP Trunk(gNtel)] |
| 14:29.34 | J4zen | include => from-trunk-VOIP Trunk(gNtel)-custom |
| 14:29.34 | J4zen | exten => _.,1,Set(GROUP()=OUT_2) |
| 14:29.53 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 14:30.48 | [TK]D-Fender | J4zen: That isn't a valid looking context name or method of setting this up |
| 14:31.21 | [TK]D-Fender | J4zen: And you don't INCLUDE "devices" from sip.conf as if they were CONTEXTS in extensions.conf |
| 14:32.20 | [TK]D-Fender | J4zen: This setup is definately not "sane" |
| 14:32.45 | J4zen | [TK]D-Fender: I am led to believe you're right. |
| 14:32.57 | J4zen | What i just described is FreePBX's method of setting it up |
| 14:33.00 | J4zen | i am done with it |
| 14:33.05 | J4zen | I'll go to vanilla asterisk |
| 14:33.18 | [TK]D-Fender | J4zen: And being FreePBX I'm not sure how to rename this properly and work with the rest. |
| 14:33.32 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 14:33.37 | J4zen | I was more curious if it was a more common issue |
| 14:34.17 | [TK]D-Fender | J4zen: it isn't matching the user properly and section naming comes into pley as well as "fromuser" |
| 14:34.35 | J4zen | section naming? |
| 14:34.38 | J4zen | fromuser i am aware of |
| 14:34.38 | [TK]D-Fender | J4zen: Also not sure if you need "invite" as well as "port" for your "insecure=" setting |
| 14:34.40 | J4zen | i have tried that setup |
| 14:34.54 | J4zen | port was a recommended setting from the voip provider |
| 14:35.16 | [TK]D-Fender | J4zen: well I'm sure it won't hit [default] without that.. |
| 14:35.28 | [TK]D-Fender | J4zen: it isn't a catch-all like it might sound. |
| 14:35.35 | J4zen | how do you recommend i set it up? |
| 14:35.45 | [TK]D-Fender | J4zen: but as always I recommend ditching the GUI and doing it yourself. |
| 14:36.07 | J4zen | Yes well im in a situation it'd be best if i had a GUI, not for my sake. but for my co-workers |
| 14:36.28 | J4zen | I'm leaving this company by next summer |
| 14:36.39 | Qwell | I was about to say - get new coworkers... |
| 14:36.42 | J4zen | meaning i have to give several courses to get other staff familiar enough with VOIP/Asterisk |
| 14:36.45 | Qwell | but it looks like you've got that covered |
| 14:36.55 | J4zen | hehe |
| 14:36.58 | jameswf | Setting up asterisk with a front end is like building a website with frontpage, it may work in the end but if you have to edit the back end you will likely have no idea what the heck is going on..... insanity is the result of flexibility in automation |
| 14:37.17 | J4zen | haha i love that |
| 14:37.23 | J4zen | consider yourself quoted |
| 14:37.27 | Qwell | ~zeek |
| 14:37.31 | slowshutt | did you come right J4zen?? |
| 14:37.32 | Qwell | ~zeeek |
| 14:37.33 | jbot | well, 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." |
| 14:37.44 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 14:37.46 | J4zen | slowshutt: Nope |
| 14:38.00 | J4zen | lol |
| 14:38.01 | J4zen | alright |
| 14:38.08 | jameswf | maybe thats why my wife says I suck in bed :) |
| 14:38.15 | J4zen | i'm going to ditch FreePBX/Trix whatever crap GUI they come up with |
| 14:38.34 | slowshutt | there you go |
| 14:38.56 | J4zen | i suppose asterisk is to diverse to hook a GUI to |
| 14:39.07 | J4zen | and im a notepad fan anyway. |
| 14:39.26 | jameswf | I have a burning tower controlled by asterisk, you insert a disk and enter an extension for the iso you want to burn, asterisk rocks |
| 14:40.01 | slowshutt | are you sore that account details is for a sip and not a iax account? |
| 14:40.30 | rob0 | notepad?? |
| 14:40.42 | jameswf | should set it up at a school "Dial-a-Distro |
| 14:40.50 | jameswf | notepad BAD |
| 14:40.50 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 14:40.58 | J4zen | notepad rocks |
| 14:41.06 | J4zen | notepad or vim |
| 14:41.08 | J4zen | your choice |
| 14:41.16 | Qwell | notepad > vim |
| 14:41.16 | slowshutt | are you sore that account details is for a sip and not a iax account? J4zen? |
| 14:41.19 | jameswf | notepad does not properly format for *nix |
| 14:41.35 | rob0 | Notepad is on Windows. It does not understand Unix text. |
| 14:41.41 | J4zen | slowshutt; yeah 100% |
| 14:41.50 | J4zen | rob0; Make it VIM ;) |
| 14:42.12 | jameswf | ^notepad^kate || ^notepad^nano only vi when I am sober or drunk I forget...... |
| 14:43.13 | [TK]D-Fender | J4zen: http://pastebin.com/m246736ff |
| 14:43.19 | jameswf | I think there is a windows app called notepad++ which can handle unix formatting... |
| 14:43.37 | slowshutt | where you from J4zen? |
| 14:43.40 | J4zen | uh oh |
| 14:43.43 | J4zen | Netherlands |
| 14:43.50 | J4zen | jameswf: yes there is |
| 14:43.53 | rob0 | There are indeed various 3rd-party Windows-based editors which do Unix. But I don't do Windows, myself. :) |
| 14:44.03 | J4zen | slowshutt: You? |
| 14:44.08 | jameswf | at some point you just say screw it and format windows right off your system |
| 14:44.14 | jameswf | YAY DEBIAN |
| 14:45.41 | *** join/#asterisk PepOSX (n=pepOSX@190.72.153.45) |
| 14:46.03 | hmmhesays | he's going the distance |
| 14:46.05 | hmmhesays | he's going for speed |
| 14:46.23 | jameswf | the last thing I thought I might need windows for was photoshop, then I got a book read how to use Gimp, turns out I havent needed windows in 2 years and counting... |
| 14:46.48 | tzanger | hmmhesays: awesome osng |
| 14:47.20 | jameswf | hmmhesays, dont tease the fat man with cake |
| 14:47.26 | slowshutt | south africa |
| 14:48.43 | J4zen | Cool, you speak semi-dutch then ;) ? |
| 14:48.48 | tzanger | hmm, is the pci-express digium offering driven by the same te2xx driver as the pci version? |
| 14:49.23 | hmmhesays | lol, hard to play and sing |
| 14:49.55 | hmmhesays | extremely |
| 14:51.11 | hmmhesays | the gimp is decent for photo editing |
| 14:51.29 | J4zen | [TK]D-Fender: adding the DID didn't do the trick, unfortunatly. |
| 14:52.37 | [TK]D-Fender | J4zen: TAHTS ONLY PART OF THE PICTURE |
| 14:52.52 | tzanger | OMGWTFBBQ!!!11!11eleventy |
| 14:52.56 | J4zen | rofl |
| 14:52.58 | J4zen | caps-police |
| 14:53.18 | [TK]D-Fender | tzanger: plz2bnotmokingtehcapsY0kthxbibi |
| 14:53.37 | [TK]D-Fender | mmmmmm BBQ |
| 14:53.44 | tzanger | what's sad is that people send me emails that look like that |
| 14:53.49 | tzanger | I tend to ignore them |
| 14:54.02 | J4zen | i'm not even going to bother with freepbx anymore lol |
| 14:54.14 | J4zen | even if you give me the perfect answer which will solve the entire problem |
| 14:54.25 | J4zen | ill down right ignore u lol |
| 14:54.42 | J4zen | mmm.. fillet mignon |
| 14:54.46 | J4zen | D'oh! |
| 14:54.49 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:54.49 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:55.14 | tzanger | mmmmm |
| 14:55.21 | asteriskmonkey | is 1.4 as stable as 1.2 for large enterprise installments yet? |
| 14:55.57 | tzanger | asteriskmonkey: of course not |
| 14:55.57 | *** join/#asterisk fskrotzki (n=fskrot@host198.textwise.com) |
| 14:56.02 | FlatFoot | does meat travel down fibre ? |
| 14:56.18 | [TK]D-Fender | I bought a two 4-packs of 6oz marinated top sirloin (pepper, and teriyaki) and am a little disappointed... back to my FM tonight :) |
| 14:56.30 | thieums | asteriskmonkey: i don't think so |
| 14:56.35 | [TK]D-Fender | FlatFoot: Sure it does if you tenderize it right :p |
| 14:56.59 | FlatFoot | [TK]D-Fender: superb ! send it down then |
| 14:57.04 | slowshutt | i am using asterisk 1.4 380 sip users on site, 48 iax services, 480 remote sip users |
| 14:57.05 | thieums | 1.4 is so easy to coredumped |
| 14:57.26 | [TK]D-Fender | FlatFoot: Sorry, glass fibre isn't soluable enough ;) |
| 14:57.46 | FlatFoot | [TK]D-Fender: bugger i was looking forward to the FM |
| 14:58.16 | [TK]D-Fender | LOL!!!! "The government of New South Wales is fed up with street racers terrorizing pedestrians, so it's imposed the "YouTube Law." Confiscated cars will be videoed undergoing "crash test experiments," then shown on YouTube so the former owner (and the world) will see them being demolished. That'll learn 'em!" |
| 14:58.24 | [TK]D-Fender | Australia FTW! |
| 14:58.46 | FlatFoot | [TK]D-Fender: good thinking Aus |
| 14:58.59 | nestAr | lol. |
| 14:59.45 | nestAr | glad i don't live in australia |
| 14:59.46 | hmmhesays | I think I'm going to need to go see american gangster |
| 15:00.49 | J4zen | i heard about that movie |
| 15:00.57 | J4zen | theres this big hype going on isnt there? |
| 15:01.11 | nestAr | yea |
| 15:01.14 | J4zen | i only rarely come out of my cave, so i wouldnt know really |
| 15:01.23 | JT | [TK]D-Fender: i live in new south wales |
| 15:01.24 | nestAr | i wanted to go see it.. my wife wanted to see lions for lambs |
| 15:01.27 | nestAr | i lost |
| 15:01.33 | J4zen | owned. |
| 15:01.41 | nestAr | her pick was so so at best. |
| 15:01.56 | J4zen | Like you expected otherwise ;) |
| 15:02.05 | muiro | if I set a global variable, will it contain that value for all current connections or is it store on a per-connection basis? |
| 15:02.15 | [TK]D-Fender | nestAr: Marriage tip : You can be right.... or you can be HAPPY. Key words "Yes, dear" |
| 15:02.23 | JT | nestAr: much rather live in australia than the US |
| 15:02.25 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 15:02.57 | [TK]D-Fender | JT: Thats a temporary thing. AU + UK are seriously going "1984"..... |
| 15:03.10 | JT | [TK]D-Fender: america is much worse |
| 15:03.36 | [TK]D-Fender | JT: Not sure I'd say that so much as "getting worse progressively faster". |
| 15:03.36 | JT | [TK]D-Fender: you really need more comprehensive evidence to back up your claim |
| 15:04.38 | JT | like a lot of places, cars can already be forfeighted after a number of a certain type of driving offences |
| 15:04.47 | FlatFoot | ok i'm really having a 'BLONDE' day can't work out how to split the location of vm and monitor HELP !!!!! |
| 15:04.58 | JT | this car crashing thing is just a slight extension to that |
| 15:05.08 | JT | and wouldn't affect more than 8 vehicles a year |
| 15:05.43 | [TK]D-Fender | JT: 8 street race arrests? You don't make a law if you think you'll catch so few... |
| 15:05.49 | tzafrir | FlatFoot, you're really having a 'blonde' day if you think 'HELP!!!!' is a way to ask for help here |
| 15:05.58 | rob0 | In the USA, police can legally steal and sell a car or other property if "illegal drugs" are found. Without due process for the owner. |
| 15:06.02 | JT | [TK]D-Fender: 8 that are bad enough to warrant vehicle destruction |
| 15:06.16 | FlatFoot | just typing outloud ( brain is slowly dying ) need BEER |
| 15:06.32 | [TK]D-Fender | JT : You start a law like that and the goal is to implement it. Either that or gov't has nothing better to do with their time. |
| 15:06.36 | tzafrir | Besides, right now Asterisk talk seems to be off-topic ;-) |
| 15:06.46 | rob0 | what's asterisk? |
| 15:06.52 | FlatFoot | :-) |
| 15:06.57 | JT | [TK]D-Fender: it's a kneejerk because our media has a bee in their bonnet about road safety |
| 15:07.05 | JT | they've had this bee for the last few yeas |
| 15:07.07 | JT | years |
| 15:07.13 | tzafrir | rob0, the character Asterix the Gaul is based on |
| 15:07.24 | JT | the media is over-representing the level of "public outrage" |
| 15:07.33 | _ys | tzafrir http://bugs.digium.com/view.php?id=11244 |
| 15:08.06 | _ys | tzafrir sorry, this not for you |
| 15:08.59 | tzafrir | _ys, ask on #asterisk-dev (and why is strcpy used there?) |
| 15:09.02 | JT | [TK]D-Fender: a number of high profile road deaths and what not |
| 15:09.26 | *** join/#asterisk chendy (n=chendy@218.82.196.62) |
| 15:09.53 | _ys | tzafrir strcpy used correctly :) but offset - no |
| 15:09.55 | lirakis | JT: you dont hapen to be a street racer do you? lol |
| 15:10.04 | chendy | hey, guys, does anyone knows in which function handle the channel uniqueid? |
| 15:10.27 | JT | lirakis: no, i don't see how you could possibly have come to that conclusion? |
| 15:11.07 | [TK]D-Fender | chendy: channel |
| 15:11.21 | chendy | thanks |
| 15:12.19 | chendy | in which case does the channel itself changes the uniqueid? |
| 15:14.09 | lirakis | i have had an agent "suck" for over 24 hours now. "show queues" reports them as "agent/404 (Busy) has taken no calls yet". They cant log out or log in, I cant seem to soft hangup whatever channel they are on either. |
| 15:14.24 | JT | lirakis: ? |
| 15:14.35 | lirakis | does anyone know how to free up this agent without restarting asterisk? |
| 15:14.39 | Sweeper | lirakis: restart asterisk :P |
| 15:14.48 | *** join/#asterisk twoshadetod (n=twoshade@c-76-123-96-239.hsd1.fl.comcast.net) |
| 15:14.53 | Sweeper | use freswitch! |
| 15:15.27 | lirakis | Sweeper: cant.. calls are up with media. "use freeswitch", why? |
| 15:15.42 | twoshadetod | can someone recommend a good sip (outgoing and incoming) |
| 15:15.45 | twoshadetod | provider |
| 15:15.46 | Sweeper | *facetious |
| 15:15.49 | ManxPower | lirakis: you can't softhangup unless you know the channel name |
| 15:15.50 | jameswf | not freeswitch freswitch |
| 15:16.01 | lirakis | ManxPower: did "show channels" and got the channel name |
| 15:16.24 | ManxPower | lirakis: what version of Asterisk. What is the channel name you saw? |
| 15:16.39 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:16.48 | Sweeper | lirakis: use 'restart gracefully'? |
| 15:16.55 | lirakis | ManxPower: 1.2 SIP/1047-0a2569f0 |
| 15:17.25 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 15:17.27 | reber | can asterisk use a stun proxy ? |
| 15:17.37 | jameswf | I like restart now .... a few dropped calls never hurt anyone, if they did cingular wouldnt be so big |
| 15:17.38 | ManxPower | lirakis: and what does "sip show channel SIP/1047-0a2569f0" give you? |
| 15:17.39 | muiro | quostion: are global variables in the dial plan set per call or are they for all current calls? |
| 15:18.03 | lirakis | ManxPower: No such SIP Call ID starting with 'SIP/1047-0a2569f0' |
| 15:18.03 | [TK]D-Fender | reber: No. |
| 15:18.05 | ManxPower | muiro: they are global, all calls, all everything |
| 15:18.09 | muiro | thanks |
| 15:18.32 | *** join/#asterisk `Sean (i=Sean@CPE002211569301-CM0011e6be76d9.cpe.net.cable.rogers.com) |
| 15:18.35 | ManxPower | [TK]D-Fender: I thought 1.4 added primitive support for STUN? Not that STUN is really useful these days. |
| 15:18.40 | muiro | in that case, best method to pass data between contexts? |
| 15:18.52 | [TK]D-Fender | ManxPower: Not in release... maybe trunk |
| 15:18.52 | muiro | nah, don't answer that |
| 15:18.53 | coppice | STUN is extremely useful |
| 15:18.53 | Sweeper | muiro: channel variables :P |
| 15:18.59 | muiro | yeah, I just remember, heh |
| 15:19.01 | lirakis | ManxPower: i also see a channel for "Agent/404" which does show info.. i will pb |
| 15:19.03 | J4zen | How do i allow anonymous SIP connections? I'm going to try that as a last resort. |
| 15:19.09 | [TK]D-Fender | coppice: I'm still trying to set my laser printers on KILL here... |
| 15:19.22 | J4zen | what was the entry needed , i forgot =\ |
| 15:19.34 | lirakis | ManxPower: http://pastebin.ca/773437 |
| 15:19.38 | muiro | Sweeper,ManxPower, thanks |
| 15:19.43 | coppice | you need bigger lasers. find some sharks, and steal theirs |
| 15:20.34 | J4zen | sharks with lasers, pewpew |
| 15:20.42 | [TK]D-Fender | coppice: There are very few problems that can't be solved with ... really ... REALLY ...big lasers :D |
| 15:20.45 | *** join/#asterisk eurodrew555 (n=eurodrew@83-244-164-130.cust-83.exponential-e.net) |
| 15:20.59 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:21.07 | ManxPower | lirakis: no, that is not what it is looking for. Do a "sip show channels" The Call-ID collumn is what you are looking for. |
| 15:21.13 | coppice | we have a chain of sports shops called GigaSports. I often wonder if its intended to be pronounced jiga- :-) |
| 15:21.44 | lirakis | ManxPower: so i guess i need the IP of the phone |
| 15:21.48 | eurodrew555 | hi |
| 15:21.55 | coppice | [TK]D-Fender: tell that to the guys that can never get the 747 laser gun to do anything effective |
| 15:22.10 | eurodrew555 | can I ask if anyone is using any CTI integration with Asterisk |
| 15:22.13 | *** join/#asterisk svenna (n=svenna@p548D0704.dip0.t-ipconnect.de) |
| 15:22.36 | ManxPower | [TK]D-Fender: do a "stun debug" |
| 15:22.39 | lirakis | ManxPower: asteriskcc*CLI> sip show channel 66c0f44c4c8 |
| 15:22.40 | lirakis | No such SIP Call ID starting with '66c0f44c4c8' |
| 15:23.00 | *** join/#asterisk twans (n=arcraig@c-71-195-114-214.hsd1.ca.comcast.net) |
| 15:23.51 | ManxPower | lirakis: put the output of sip show channels on a pastebin. what is the ip of the problem phone? |
| 15:24.58 | [TK]D-Fender | eurodrew555: Be specific please. That is a dangerously vague term. |
| 15:25.11 | lirakis | ManxPower: hang on checking |
| 15:25.35 | ManxPower | [TK]D-Fender: rtp.c sure has a lot of stuff about stun in 1.4 |
| 15:26.26 | [TK]D-Fender | ManxPower: News to me... never saw anything in the configs regarding it... |
| 15:26.58 | ManxPower | [TK]D-Fender: I can't find anything in the sample configs about it, but the source has it. |
| 15:27.01 | lirakis | ManxPower: okay .. having the call center mgr get that for me... what should i be looking to do? I mean, if i have the "right" channel will "soft hangup xxxxxxxxx" work, where xxxxxxx is the channel? |
| 15:27.26 | lirakis | ManxPower: or rather the "call Id" from sip show channels |
| 15:27.35 | [TK]D-Fender | ManxPower: Hey I know... lets start a rumor! ;) |
| 15:28.13 | ManxPower | [TK]D-Fender: yet another "bug". An undocumented feature |
| 15:28.54 | [TK]D-Fender | ManxPower: Ones causing core dumps shall henceforth be named "Special Features" |
| 15:28.56 | ManxPower | lirakis: you can't know the right callid without knowing the ip address (at least as far as I can tell) |
| 15:29.24 | lirakis | ManxPower: http://pastebin.ca/773447 I think the ip is 10.10.2.65 the agent was last logged into ext 1047. |
| 15:29.38 | *** part/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:29.44 | ManxPower | [TK]D-Fender: Where "special" is closer to "special Olympics" than "special diamond" |
| 15:29.55 | ManxPower | lirakis: well come back when you KNOW |
| 15:30.42 | *** join/#asterisk ming_zym (n=ming_zym@124.14.233.88) |
| 15:30.51 | [TK]D-Fender | lirakis: pastebin "show channels concise" and "show queues" |
| 15:31.09 | [TK]D-Fender | lirakis: "show channels" truncates channel names... |
| 15:31.21 | ManxPower | lirakis: [TK]D-Fender is much better at queues than I am. |
| 15:31.41 | [TK]D-Fender | ManxPower: A scary prospect... |
| 15:31.48 | ManxPower | I think Asterisk's queue system should be taken out behind the barn and shot and put it out of our misery. |
| 15:32.21 | jameswf | ahhh fond memories of being behind the barn |
| 15:32.29 | coppice | Ask a brit. queuing is a national passtime there |
| 15:32.55 | _x86_ | just got off the phone with AT&T, and i requested E&M Wink signalling... |
| 15:33.01 | lirakis | [TK]D-Fender: http://pastebin.ca/773452 |
| 15:33.02 | _x86_ | this is for a CAS T1 order |
| 15:33.32 | _x86_ | they tell me theyre going to set me up with NI2 and DTMF Wink signalling |
| 15:33.34 | _x86_ | is that the same |
| 15:33.36 | _x86_ | ? |
| 15:33.52 | _x86_ | i thought NI2, 4ESS, 5E, etc was all PRI-specific |
| 15:33.57 | *** join/#asterisk Greek-Boy (n=email@41.221.58.2) |
| 15:34.13 | ManxPower | _x86_: if they tell you that you should get a different carrier, as your current carrier has morons working for them |
| 15:34.42 | FlatFoot | coppice: queuing , thats a thing of the past . Barging is the new pasttime |
| 15:34.50 | FlatFoot | * Bargeing |
| 15:34.56 | [TK]D-Fender | lirakis: "soft hangup SIP/69.24.128.135-b6d87930" |
| 15:34.58 | Greek-Boy | Anyone here got experience with mass provisioning of linksys spa942's? I'm trying to figure out which configuration files to use with firmware version 5.1.15a |
| 15:34.59 | FlatFoot | * or summit like that |
| 15:35.17 | coppice | ho, I used to love playing on canal barges as a kid |
| 15:35.33 | [TK]D-Fender | lirakis: Agent/104:internal:s:1:Up:Bridged Call:SIP/69.24.128.135-b6d87930:2037463780::3::SIP/69.24.128.135-b6d87930 |
| 15:35.45 | [TK]D-Fender | lirakis: Agent/404:internal:s:1:Up:Bridged Call:SIP/1021-b7a9fc28:1021::3::SIP/1021-b7a9fc28 |
| 15:35.50 | [TK]D-Fender | lirakis: one of these two... |
| 15:36.16 | ManxPower | [TK]D-Fender: it looks like softhangup wants a CallID, nor a Channel Name |
| 15:36.24 | lirakis | <PROTECTED> |
| 15:36.29 | [TK]D-Fender | ManxPower: No, its the channel... |
| 15:36.40 | [TK]D-Fender | ManxPower: call-id is a SIP specific thing |
| 15:36.53 | [TK]D-Fender | IIRC |
| 15:37.43 | lirakis | [TK]D-Fender: same for the other .... "SIP/1021-b7a9fc28 is not a known channel" |
| 15:38.31 | *** join/#asterisk harpal (n=Harpal@124.125.255.223) |
| 15:38.44 | eurodrew555 | hey Greek-Boy I'm currently working on this very thing |
| 15:38.55 | eurodrew555 | trying to sort out mass provisioning of all our SPA942s |
| 15:39.01 | ManxPower | lirakis: what happens if you power off the device 1021? |
| 15:39.30 | lirakis | ManxPower: good question... ill have to try that and device 1047 |
| 15:39.50 | ManxPower | lirakis: 1047? |
| 15:39.59 | ManxPower | why would you want to power off 1047? |
| 15:40.34 | *** join/#asterisk truz_`24 (n=truz_`24@74-129-166-232.dhcp.insightbb.com) |
| 15:40.58 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:41.00 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:41.06 | lirakis | ManxPower: it was the last device used .. the one that i think, agent/404 was using. |
| 15:41.15 | *** join/#asterisk truz_`24 (n=truz_`24@74-129-166-232.dhcp.insightbb.com) |
| 15:41.20 | lirakis | ManxPower: the other side of the bridged call with 1021 i think |
| 15:41.45 | lirakis | ManxPower: does that seem reasonable? |
| 15:41.49 | ManxPower | no. |
| 15:41.55 | lirakis | ManxPower: why ? |
| 15:42.12 | ManxPower | because agent 404 looks to be using device 1021 based on... |
| 15:42.19 | ManxPower | Agent/404:internal:s:1:Up:Bridged Call:SIP/1021-b7a9fc28:1021::3::SIP/1021-b7a9fc28 |
| 15:42.53 | lirakis | ManxPower: I thoght that meant 404 was calling 1021 |
| 15:43.18 | reber | [TK]D-Fender, ok |
| 15:43.20 | ManxPower | lirakis: An agent is a "virtual" think. |
| 15:43.26 | ManxPower | thing |
| 15:44.03 | ManxPower | lirakis: what version of 1.2.x are you using? |
| 15:44.15 | reber | ManxPower, " Not that STUN is really useful these days." Do you mean that stun IS or IS NOT usefull ? |
| 15:44.55 | lirakis | ManxPower: right... i saw this line right below "SIP/1047-0a2569f0:internal:s:1:Up:(None)::1047::3::SIP/1021-b7a9fc28" which i thought was the "real" version of the mapped agent .. |
| 15:45.02 | ManxPower | reber: STUN is a way for SIP devices that don't understand NAT to work behind NAT. Virtually all SIP devices support NAT so STUN is not useful for most people |
| 15:45.16 | lirakis | ManxPower: im having 1021 power cycled |
| 15:45.37 | ManxPower | lirakis: it might be. go ahead and power both of them off. NO! DO NOT POWER CYCLE! Leave the device off for 5 mins |
| 15:45.40 | lirakis | [TK]D-Fender: any idea why softhangup "couldnt find" those channels |
| 15:45.51 | lirakis | ManxPower: oh so it will loose registration? |
| 15:45.57 | coppice | ManxPower: wrong. STUN lets two boxes sitting behind different NATs talk to each other without hassle |
| 15:46.20 | ManxPower | lirakis: so it will lose registrationl rtp, etc |
| 15:46.34 | ManxPower | coppice: um, two devices behind nat can talk to each other without stun too. |
| 15:46.56 | [TK]D-Fender | lirakis: no idea |
| 15:46.58 | coppice | how? |
| 15:47.03 | lirakis | ManxPower: okay.. doing it for both 1021 and 1047 |
| 15:47.11 | ManxPower | coppice: using Asterisk, of course. |
| 15:47.29 | *** join/#asterisk techie (n=techie@adsl-76-214-20-56.dsl.lsan03.sbcglobal.net) |
| 15:47.30 | coppice | I said talk to each other. not chat through a third party |
| 15:47.37 | lirakis | [TK]D-Fender: okay .. if the power off doesnt work, i will have to wait till off hours and restart asterisk i guess |
| 15:48.05 | ManxPower | coppice: but for the moment lets ignore that. How does the NAT router set up the NAT translations between the two hosts when there is no outgoing connection between the two hosts. |
| 15:48.12 | ManxPower | (when using stun) |
| 15:48.20 | [TK]D-Fender | lirakis: that's ManxPower's suggestion, but I totally agree with it. |
| 15:48.26 | coppice | well, that is why STUN is so very useful |
| 15:48.58 | lirakis | [TK]D-Fender: yeah i think it makes sense too.. so i hope it works.. glad there is some consensus on it though |
| 15:49.19 | ManxPower | coppice: you are way too smart to ignore your claim because I think it's wrong. So help me out here. You have a very good understanding of NAT, right? |
| 15:49.46 | coppice | well, there isn't a whole lot to understand |
| 15:50.12 | [TK]D-Fender | ManxPower: Only reason it works with * is because of the port forwarding you do to it. hence going through * isn't really "works behind NAT" so much... |
| 15:50.27 | [TK]D-Fender | ManxPower: basically you can't reinvite at will... |
| 15:51.22 | ManxPower | Host A sends a packet to Host B. Host A is behind a NAT router. When the NAT router sees the packet it sets up a translation to translate the internal IP address and port to an external IP and port (the internal port and external port might be differnet). Right? |
| 15:51.31 | coppice | STUN lets you find what you look like to the outside world. from then on you can behave just as if you are on the outisde world using that port. no restrictions at all |
| 15:52.03 | ManxPower | this translation would also handle response packets to the outgoing packets. |
| 15:52.21 | asteriskmonkey | is there a patch for imap voicemail support for 1.2? |
| 15:52.39 | ManxPower | The nat router's translation table would be basically internal IP, internal port, external ip, external port. Right? |
| 15:52.49 | ManxPower | asteriskmonkey: No! |
| 15:52.55 | *** join/#asterisk dswillia (n=me2@199.3.247.34) |
| 15:53.02 | coppice | right |
| 15:53.48 | ManxPower | OK. Now, lets say that there is no initial connecton going outbound thru the nat, so there is no entry in the translation table. Without portforwarding the connection will fail, right? |
| 15:54.06 | [TK]D-Fender | ManxPower: yup. |
| 15:54.12 | ManxPower | sorry, without portforwarding and INCOMING connection would fail. |
| 15:54.13 | coppice | it would |
| 15:54.22 | dswillia | hey all, asterisk 1.4.13 with Sangoma A102 dual pri card. I have ran the wanrouter setup and the card seems to be good. (No alarms) However how can i check to see if asterisk actually see's it in an up state? |
| 15:54.23 | *** join/#asterisk Sonderblade (n=fsldfj@53dbd052.umea.cust.skycom.se) |
| 15:54.23 | [TK]D-Fender | ManxPower: typically NAT routers keep paths open for UDP only a very short time |
| 15:54.46 | [TK]D-Fender | dswillia: "pri show span [span number]" |
| 15:54.48 | ManxPower | OK, so when STUN is used, the translation table would have internal IP, internal port, external STUN IP, external port, right? |
| 15:55.01 | coppice | not that short. they need to ride over things like VAD |
| 15:55.11 | asteriskmonkey | If you use nfs as a backing for your voicemail store, and connection is lost dose asterisk crash? |
| 15:55.37 | coppice | well, external port. there is nothing STUNish about it |
| 15:55.49 | dswillia | ok I am somewhat new, but when I issue the command pri show span 1 I get a "No such command 'pri show' (type 'help' for help)" |
| 15:55.55 | Sonderblade | anyone know how to configure the dialtone for gxp 2000 phones? |
| 15:56.01 | ManxPower | coppice: I didn't say "external stun port" |
| 15:56.06 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 15:56.23 | coppice | sorry. well, external IP. there is nothing STUNish about it |
| 15:56.31 | ManxPower | OK, so now the SIP client gets a response back from the STUN server telling the SIP device what IP and port it sees the conneciton coming from, right? |
| 15:57.16 | [TK]D-Fender | Sonderblade: ... huh?! |
| 15:57.19 | coppice | SIP generally plays no part in this. STUN is most useful to get access RTP port to RTP port |
| 15:57.27 | ManxPower | the problem with that is that that external ip + port is only valid for the connection to the stun server. If another device tried to send packets to that external ip and port, the nat router would reject it. |
| 15:57.49 | coppice | nope. you are thinking like TCP. UDP doesn't work that way |
| 15:57.55 | ManxPower | because those packets would not be coming from a host that is in the nat routers translation tables. |
| 15:58.05 | [TK]D-Fender | dswillia: Doesn't look like you compiled * with libpri support, and possibly Zaptel.... |
| 15:58.14 | coppice | I can fire off packets in all directions from a single UDP port |
| 15:58.31 | dswillia | so just recompile then |
| 15:58.33 | ManxPower | coppice: I understand TCP .vs. UDP, but in this case that's not really an issue, the nat router "fakes" state for UDP stuff |
| 15:58.50 | Sonderblade | [TK]D-Fender: you can configure the dialtone for GXP-2000, im looking for a recipe for swedish dialtones |
| 15:58.52 | [TK]D-Fender | dswillia: Yes. Libpri, then Zaptel, then Wanpipe, then Asterisk |
| 15:59.10 | coppice | for UDP, what is in the nat table is the external port and IP on the NAT box. not the ones for the remote end |
| 15:59.21 | *** join/#asterisk Tiffon (n=name@unaffiliated/tiff0n) |
| 15:59.27 | [TK]D-Fender | Sonderblade: Go check your admin guide or the forums on www.voxilla.com |
| 15:59.33 | _x86_ | ManxPower: that's what I said! I was like "Isn't NI2, 5ESS, 4ESS, etc all PRI-specific?" and the "provisioning coordinator" said "absolutely not" |
| 15:59.48 | coppice | the NAT box cannot restrict the other end, or UDP breaks badly |
| 15:59.53 | ManxPower | _x86_: the correct response is "you know best, telecom god" |
| 16:00.00 | ManxPower | coppice: why not? |
| 16:00.48 | coppice | as I said earlier, UDP is not liek TCP. TCP is point to point. UDP is point to a variety of points. every incoming UDP packet carries with it the IP and port is came from |
| 16:00.58 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 16:01.10 | coppice | every ougoing UDP packet carries its destination |
| 16:01.16 | lirakis | ManxPower [TK]D-Fender: still seeing "Agent/404:internal:s:1:Up:Bridged Call:SIP/1021-b7a9fc28:1021::3::SIP/1021-b7a9fc28" after at least 5 minutes here :\ |
| 16:01.42 | _x86_ | ManxPower: i just hope it gets provisioned as E&M Wink heh |
| 16:01.51 | _x86_ | ManxPower: no clue what DTMF Wink is |
| 16:02.34 | coppice | once you fit a STUN server, you have an external UDP port that can send to anywhere or receive from anywhere. the first thing you get is a STUN packet telling you what that extermal IP/port pair it |
| 16:02.54 | ManxPower | Here's an example translation (internal) 192.168.1.1:1234 is translated to (external) 209.16.72.130:32764 and (dest host) 65.34.1.6:5060 |
| 16:03.14 | coppice | _x86_ DTMF wink is the same things as E&M wink these days |
| 16:03.40 | ManxPower | now if 65.34.1.6 sends a packet from 5060 to 209.16.72.130:32764 the NAT router will change the dest port and IP to be 192.168.1.1:1234, right? |
| 16:03.48 | *** join/#asterisk jarg (n=jarg@200.56.225.61) |
| 16:03.56 | coppice | what you are missing here is there is *no* specific destination host with UDP |
| 16:04.00 | *** join/#asterisk harpal (n=Harpal@124.125.255.223) |
| 16:04.12 | ManxPower | coppice: there is always a destination host and IP. |
| 16:04.29 | coppice | per packet, but not per local port |
| 16:04.50 | florz | coppice: but that's not the case for TCP, either |
| 16:05.06 | ManxPower | if what you say is correct then if there is an active UDP translation happening on a NAT router then any host on the internet can connect thru that port and hack the source cost with a buffer overflow or something |
| 16:05.21 | coppice | florz: TCP is a connection to a specific distant IP/port pair |
| 16:05.25 | ManxPower | since the NAT router is a slut and will accept anyone sending packets to it's external port. |
| 16:05.45 | florz | coppice: no, TCP is a connection between two specific IP/host pairs |
| 16:05.46 | coppice | florz: every packet in the connection had better go to the same IP/port pair for TCP |
| 16:06.08 | coppice | florz: sure. I mean it has a specifc far end. UDP doesn't |
| 16:06.35 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 16:06.53 | ManxPower | Apparently Cisco DOES keep the destination host/ip in it's table |
| 16:06.54 | florz | coppice: Well, per socket (exception: no far end), but not per local address/port pair |
| 16:06.55 | ManxPower | Pro Inside global Inside local Outside local Outside global |
| 16:06.55 | ManxPower | tcp 209.16.79.102:51473 172.16.7.13:51473 69.28.9.12:80 69.28.9.12:80 |
| 16:07.08 | flujan | hi guys, I have a simple question... I am trying to install a second version of asterisk on a different path... no int /usr but in /opt/local/ |
| 16:07.13 | flujan | how can I do this? |
| 16:07.23 | coppice | Manxpower: once an outgoing UDP packet gets an external side port allocated for your internal side port, all packets from anywhere , headed to that external IP/port pair will be forwarded to your box |
| 16:07.31 | ManxPower | sorry, bad paste |
| 16:07.35 | flujan | I tried the --prefix=/opt/local in the configure but this dont work. :( |
| 16:08.04 | florz | coppice: A web server is sending from its address/port to numerous different remote address/port pairs "at the same time" |
| 16:08.13 | ManxPower | Here's a better one |
| 16:08.14 | ManxPower | Pro Inside global Inside local Outside local Outside global |
| 16:08.15 | ManxPower | udp 209.16.79.102:41991 172.16.9.2:41991 63.209.3.132:53 63.209.3.132:53 |
| 16:08.15 | ManxPower | udp 209.16.79.102:1425 172.16.9.195:32768 208.172.65.40:53 208.172.65.40:53 |
| 16:08.42 | coppice | florz: right. a specific TCP socket is a 5-tuple. |
| 16:08.47 | ManxPower | now if the host 45.24.7.1 tries to send a packet to 209.16.79.102:1425 the router should reject the packet |
| 16:09.02 | florz | coppice: And it's merely a limitation in the implementation that you can't connect outbound to multiple destinations from a single local address/port |
| 16:09.16 | ManxPower | coppice: are you saying the NAT router would happily forward the packet to 172.16.9.195:32768? |
| 16:09.33 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 16:09.44 | coppice | florz: this is getting off the point. we are trying to discuss UDP here |
| 16:10.17 | ManxPower | That's AWESOME! /me DDOS the internet with UDP port 53 packets designed to crash Windows DNS resolution! |
| 16:10.19 | florz | coppice: Yeah, kindof, but you were claiming some fundamental difference between TCP and UDP that simply isn't there |
| 16:10.22 | salog | Hi! My sipprovider overwrites SetCallerID, so I need to dial #31# in front of my number, to hide my ID. When I try this out from a sip client, it works, but using "exten => _XXX.,1,Dial(SIP/#31#${EXTEN}@provider)" results in a circuit-busy. Is there anything I need to take care of? Do I need to escape #, or anything else? |
| 16:10.56 | FlatFoot | ManxPower: good thread what was the original question ? |
| 16:11.06 | coppice | florz: it is as far as the NAT box is concerned. it treats every specific TCP flow separately, but not every UDP flow |
| 16:11.09 | _x86_ | wtf is DTMF Wink? same as E&M Wink? |
| 16:11.18 | *** join/#asterisk Cazper (n=cazper@85.196.120.126) |
| 16:11.19 | ManxPower | FlatFoot: the original question was "I don't think STUN is very useful" |
| 16:12.08 | florz | coppice: well, but that's implementation and configuration specific, too |
| 16:12.18 | coppice | if x:y sends through a nat box to a stun box, and gets a:b on the outside, the stun box sends back to a:b, saying "you are a:b". that packet goes through to x:y. |
| 16:12.22 | FlatFoot | ah ok , so how did it get so deep into nat trans ? also don't forget that the nat router will store the destination specific to a MAC and should in theory only send packets to that mac so lon as it mathces th ip |
| 16:12.35 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 16:12.57 | ManxPower | coppice: I agree. |
| 16:13.03 | coppice | Now, x:y can send to anywhere else, and anywhere can send to a:b, where the packets will be forwarded to x:y |
| 16:13.13 | florz | coppice: And UDP applications usually do have a clear distinction between server and client, just as is the case with TCP ... |
| 16:13.16 | *** part/#asterisk Cazper (n=cazper@85.196.120.126) |
| 16:13.41 | ManxPower | But I do not think that if x:y sends through a nat box to a stun box, and gets a:b on the outside, the stun box sends back to a:b, saying "you are a:b". that packet goes through to x:y. and another random host sends a response to a:b that the NAT router should reject it. You say this not the case. |
| 16:13.45 | florz | coppice: And that's as little dictated by the protocol with TCP as it is with UDP |
| 16:14.10 | coppice | florz: the original and currne biggest use for UDP is media, where there is no sense of client or server at all. its pure peer-topeer |
| 16:14.58 | florz | coppice: Uh? I'd say in case of RTP there is a pretty clear distinction between client and server, for example. |
| 16:15.03 | ManxPower | There's the thing, if NAT routers operate the way coppies says the operate, then you STILL should not need STUN, as Asterisk can see the apparent source ip/port and tell the other SIP device what those numbers are |
| 16:15.07 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 16:15.13 | Greek-Boy | very odd. I have sip.conf with asterisk 1.4 but it didnt load it as a module. |
| 16:15.31 | florz | coppice: The server announces, say, via sdp, where it can be reached, and the client uses that to send its data there |
| 16:16.23 | florz | coppice: or you have symmetric RTP, where it's even clearer that one side waits for the other to "connect" in order to then reply to that "connection" |
| 16:16.25 | ManxPower | coppice: so are you saying that reinvites will work if you are using STUN? |
| 16:16.37 | *** join/#asterisk techie (n=techie@adsl-76-214-20-56.dsl.lsan03.sbcglobal.net) |
| 16:16.40 | coppice | only in crummy setups. in clean setups two clients find each other with help from a server, then the media goes pure peer to peer |
| 16:17.06 | coppice | why reinvite, when you can chat directly to begin with? |
| 16:17.18 | _x86_ | "E&M Wink" == "E&M DTMF Wink" == "DTMF Wink" ? |
| 16:17.44 | florz | plus, it's not uncommon for NAT to be paired with a stateful packet filter, which at least won't let pass any packets that belong to a UDP flow where it hasn't seen any outbound packets recently. |
| 16:17.49 | [TK]D-Fender | coppice: That'd be dial by IP though, no? |
| 16:17.56 | _x86_ | coppice: but then you lose features like CDR :) |
| 16:17.58 | ManxPower | coppice: I don't know, but my qurstion is would renvites work when using stun? |
| 16:18.19 | coppice | _x86_ E&M wink one has the option of pulse dialing, but you'll never see that any more. its always the same as DTMF wink these days |
| 16:18.32 | _x86_ | coppice: excellent |
| 16:19.04 | [TK]D-Fender | And reinvites are goo when the call is being bridged in a scenario where * doesn't need to be involved (RTP DTMF, recording, etc) |
| 16:19.05 | coppice | manxPower: of course it would work. once stun has helped you, you behave just like you are on the public internet |
| 16:19.30 | ManxPower | coppice: I disagree, but I see now why people would think they need STUN. |
| 16:19.42 | coppice | reinvites are dumb. direct initial connections are the sane approach. reinvites are for control freaks |
| 16:20.14 | ManxPower | Maybe I'll get off my lazy ass and do some actual testing. |
| 16:20.37 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 16:21.32 | coppice | well, reinvites are also for things like redirection and changing to FAX, but reinvites for what you are talking about it just for people who want control over you. they don't assist you at all |
| 16:22.49 | ManxPower | I used reinvites as an example because that's something that doesn't work thru nat |
| 16:22.57 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 16:23.32 | *** join/#asterisk ussrback (n=MAX@80.92.183.84) |
| 16:24.04 | ManxPower | coppice: so how do you get the two phones to talk directly to each other without a reinvite when using Asterisk? |
| 16:24.14 | ManxPower | (assuming no nat is involved) |
| 16:24.55 | coppice | STUN is very simple. It takes only a couple of pages of code to add it to your app. You send one packet and get one packet to figure out a port. that means the server load is so low people have set up small servers all over the place with free access. it just works |
| 16:25.33 | [TK]D-Fender | coppice: control over me? No I'm talking for my server.... I've got a God complex to support here! |
| 16:25.51 | ManxPower | If you use my server than I get to control you. |
| 16:26.03 | ManxPower | if you don't like that then you don't have to use my server 8-) |
| 16:27.05 | De_Mon | if I remove someone from a queue while they are on a call, is it supposed to disconnect them? |
| 16:28.44 | jameswf | cox communications sucks... |
| 16:29.02 | Corydon76-dig | De_Mon: shouldn't, no |
| 16:29.28 | ManxPower | jameswf: they are cox suckers |
| 16:29.29 | jameswf | De_Mon, if for whatewver reason you reload it will |
| 16:29.36 | ManxPower | sorry, those would be their customers |
| 16:29.40 | De_Mon | that can't be what happened they wern't a dynamic members... that was weird |
| 16:29.54 | jameswf | cox is having networkk issues and its f***in with my tivo schedule |
| 16:30.03 | De_Mon | cox suckers |
| 16:30.07 | Qwell | jameswf: simple solution - don't use tivo |
| 16:30.10 | jameswf | now i have to resort to piracy |
| 16:30.13 | De_Mon | don't use cox |
| 16:30.19 | ManxPower | Qwell: no need to get vulgar |
| 16:30.20 | ussrback | what is a best H323 channel for * ? |
| 16:30.28 | ManxPower | ussrback: they ALL suck. |
| 16:31.25 | ussrback | ManxPower: ohhh no dont killmy heart :), what should i try to use H323 ? |
| 16:31.30 | jameswf | thank god for torrents |
| 16:31.40 | ussrback | I have to connect with other operator throught h323 |
| 16:31.50 | ManxPower | ussrback: try the one in asterisk-addons |
| 16:32.26 | ussrback | what about gnugk +asterisk? |
| 16:32.57 | ManxPower | what about it? You still need an h323 channel driver for Asterisk |
| 16:33.59 | ussrback | is not it possible to use sip between gnugk and asterisk |
| 16:34.53 | *** join/#asterisk hfb (n=hfb@pool-71-106-219-180.lsanca.dsl-w.verizon.net) |
| 16:38.22 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 16:40.10 | De_Mon | jameswf if you have nntp access you can probably get whatever from there too, it it is new |
| 16:40.42 | ai-a | ussrback: dont put 'not' in that question. creates ambiguity in the answer. |
| 16:40.44 | *** join/#asterisk mrtelephone (n=MrTeleph@h697179-171.picriverisp.net) |
| 16:40.54 | mrtelephone | have you ever seen a sip device with multiple authorization headers? |
| 16:41.21 | ai-a | mrtelephone: you can auth as many as you like over sip. |
| 16:41.22 | mrtelephone | I have these sip devices that stop authenticating after a few days.. and I ngrep'd the output and this is what happens.. |
| 16:41.27 | mrtelephone | one sec |
| 16:41.32 | mrtelephone | ~pastebin |
| 16:41.32 | jbot | methinks pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or http://rafb.net/paste/, or http://pastebin.com is usually painfully too slow and unresponsive to use, use one of the other pastebin sites, or dpaste.com is a very nice pastebin as well |
| 16:41.57 | jameswf | yeah, I actualy like opening up nntp files then inspecting my ,wine folder |
| 16:42.08 | *** join/#asterisk irule (n=irule@200.53.61.4) |
| 16:42.25 | Qwell | jameswf: I once downloaded an exe packed set of...umm...files |
| 16:42.41 | Qwell | moved my .wine dir, ran the exe, let it virus me, and blew it away, heh |
| 16:43.08 | mrtelephone | http://pastebin.ca/773543 |
| 16:43.20 | Toerkeium | hello everyone. I think it's a off-topic question, but someone might help. |
| 16:43.25 | mrtelephone | take a look at that.. is that a mistake int he firmware where the variable isn't cleared? |
| 16:43.50 | mrtelephone | ai-a, multple authorizations with different nonces? |
| 16:44.03 | Toerkeium | I am running trixbox (and I am pretty sure it has nothing to do with trixbox). But when a operator speaks (using a softphone) with her headset, the other person (who called) hears every sound played in the computer. I have 10 operators with the same hardware and configuration. I tried changing the soundcard, the headset and assing another operator extension, but if you call, you hear even your own voice, because any sound played in the computer goes to as |
| 16:44.46 | *** join/#asterisk UserReg_CL (n=dede@200.113.138.20) |
| 16:44.52 | ai-a | Toerkeium: keyboards dont make sound,, apart from noise from the keys clicking,, so your mic is picking up this noise. |
| 16:45.11 | Toerkeium | ai-a.: no, I tried unplugging the headset |
| 16:45.14 | ManxPower | Toerkeium: sounds like you need to turn off the speakers. |
| 16:45.24 | UserReg_CL | hi, please helpme... need block call all number and only permit 2 number |
| 16:45.34 | ManxPower | Toerkeium: this is really an issue for the vendor of the softphone you purchased. |
| 16:45.43 | Toerkeium | ManxPower, there is no speakers. |
| 16:45.47 | mrtelephone | does asterisk have to generate a new nonce everytime?> |
| 16:46.02 | Toerkeium | ManxPower: it's the SJPhone, which I have working on 9 other PCs, with the same hardware. |
| 16:46.09 | Toerkeium | ManxPower: can be a configuration issue? |
| 16:46.50 | ManxPower | Toerkeium: it is not a configuration issue with Asterisk. It might be a configuration issue with the softphone or the operating system the softphone is running on. |
| 16:47.12 | Toerkeium | yeah, that's what i meant |
| 16:47.14 | rob0 | When the OS/GUI plays a sound, it goes over the softphone. |
| 16:47.23 | mrtelephone | Manxpower, did you take a look at the pastebin? |
| 16:47.34 | Toerkeium | I mean, I replaced all hardware already |
| 16:47.51 | Toerkeium | rob0, but not to the mic output, correct? |
| 16:47.55 | ManxPower | mrtelephone: what about it? |
| 16:48.38 | mrtelephone | does that look like a sip firmware problem? multiple authorizations? |
| 16:48.45 | ManxPower | mrtelephone: I have no idea. |
| 16:48.48 | Toerkeium | I am thinking of another software which could modified the normal behavior of sound cards |
| 16:48.55 | ManxPower | If I had an idea I would have said something. |
| 16:48.59 | mrtelephone | :( |
| 16:49.04 | mrtelephone | what do you know about nonces? |
| 16:49.17 | ManxPower | mrtelephone: nothing whatsoever' |
| 16:49.35 | ManxPower | I've never had to know anything about them. |
| 16:50.03 | [TK]D-Fender | Toerkeium: That is typical of picking the wrong source for input. Instead of "Mic" you seem to have picked what Creative labels "What U Hear" <- |
| 16:51.08 | ManxPower | [TK]D-Fender: so it's an OS config problem, just like everyone has been telling him. |
| 16:51.34 | Toerkeium | I am sure it has to do with OS |
| 16:51.43 | jameswf | ~pirates |
| 16:51.45 | ManxPower | But this is #asterisk, not #Windows |
| 16:51.48 | [TK]D-Fender | ManxPower: that is a resounding DUH. * can't magically sleat audio from a host PC! |
| 16:51.57 | UserReg_CL | :( |
| 16:51.57 | [TK]D-Fender | Steal* |
| 16:52.04 | ManxPower | and not #voip either |
| 16:53.04 | *** join/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 16:54.55 | nestAr | lol |
| 16:54.56 | nestAr | US to spend $1,600,000,000,000 on universal health care, free college education for all. Just kidding, it's for war, and it's already gone |
| 16:55.48 | Toerkeium | Did I ever blame to asterisk? |
| 16:56.00 | Qwell | Toerkeium: yes |
| 16:56.09 | Toerkeium | not I didn't |
| 16:56.15 | Qwell | You are in #asterisk |
| 16:56.33 | Toerkeium | just asked here because someone may already had this problem |
| 16:56.53 | [TK]D-Fender | Toerkeium: Stop. Now. We've told you what to look for. Let go, and move on. |
| 16:57.47 | Toerkeium | Stop. Now. ? take it easy, boy.. it's ok, won't ask anythinhg else |
| 16:57.47 | mrtelephone | hahah nesTAR |
| 16:58.33 | *** join/#asterisk linxroute (n=linx@203.190.164.86) |
| 16:58.48 | *** join/#asterisk cpina (n=carles@ip23498.bcn.altecom.net) |
| 16:58.59 | ussrback | when im trying to install ooh323c from addons i got error http://pastebin.ca/773554. how can i fix it? |
| 16:59.35 | tzafrir | cpina, BTW: current kernel in backports.org is 2.6.22 . IIRC it supports hires timers |
| 17:00.17 | tzafrir | right. It does |
| 17:00.33 | coppice | hires timers are nice. HPET might be nice too, in time |
| 17:00.58 | jameswf | woohoo wards newtoy is out |
| 17:01.39 | ai-a | can i use the manager port to track call transfers (attended) calls ? and who out of all Dial(SIP/,....) finally picks up the call ? |
| 17:04.48 | Toerkeium | it was the "What you hear" stuff |
| 17:04.52 | cpina | tzafrir: are you talking about ztdummy? .-) |
| 17:05.03 | tzafrir | yes |
| 17:05.17 | cpina | tzafrir: thanks, i will test it |
| 17:05.21 | cpina | one moment :-) please |
| 17:05.30 | cpina | (i wanted to ask other thing but doesn't matter now...) |
| 17:05.52 | Qwell | tzafrir: removing .EXPORT_ALL_VARIABLES doesn't fix 2.6.24 |
| 17:05.59 | tzafrir | ussrback, I figure it means that due to some earlier build error, that file was not built |
| 17:05.59 | Qwell | same CFLAGS error |
| 17:06.00 | Toerkeium | thanks |
| 17:06.17 | mrtelephone | can someone help me fix chan_sip.c |
| 17:06.59 | tzafrir | Qwell, can you check with env if you set it in your environment? |
| 17:07.05 | Qwell | huh? |
| 17:07.13 | Qwell | set what? |
| 17:07.19 | tzafrir | CFLAGS |
| 17:07.23 | Qwell | no, it's not set |
| 17:07.26 | tzafrir | Or will it be ignored then |
| 17:07.32 | UserReg_CL | hi, please helpme... need block call all number and only permit 2 number |
| 17:07.44 | Qwell | you mean my bash env, right? |
| 17:07.46 | *** join/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 17:08.05 | *** part/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 17:08.13 | tzafrir | And if you rem-out setting CFLAGS (and just build kerne module: 'make modules'), does it work? |
| 17:08.19 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 17:09.05 | Qwell | CFLAGS is set a bunch of places.. I tried that before, and if failed for some other reason |
| 17:09.10 | tzafrir | 'make modules'? in what way exactly? |
| 17:09.51 | Qwell | /usr/src/zaptel/branches/1.4/pciradio.c:59:27: error: zaptel/zaptel.h: No such file or directory |
| 17:09.57 | *** join/#asterisk trippss (n=ss@72.20.150.196) |
| 17:10.44 | trippss | hello people. can someone help me use sipsak to debug a NOTIFY message response? I don't quite get sipsak usage |
| 17:11.41 | ManxPower | trippss: um, sipsak sends custom sip packets, maybe you want "sip debug" in the Asterisk CLI instead? |
| 17:12.43 | Qwell | tried setting EXTRA_CFLAGS=-I., but it didn't like that either |
| 17:12.50 | *** part/#asterisk harpal (n=Harpal@124.125.255.223) |
| 17:12.58 | trippss | ManxPower: no i'm trying to simulate what betatester did in this post - http://bugs.digium.com/view.php?id=8575 |
| 17:13.39 | trippss | ManxPower: more specifically http://bugs.digium.com/view.php?id=8575#56410 |
| 17:14.53 | tzafrir | Qwell, EXTRA_CFLAGS should be set in Makefile.kernel26 |
| 17:15.20 | tzafrir | That error means -DSTANDALONE_ZAPATA is not set, right? |
| 17:15.36 | Qwell | yeah, must |
| 17:16.10 | De_Mon | http://pastebin.ca/773570 |
| 17:16.49 | De_Mon | Why does initiating a Wait() hangup the Local channel and start using the real SIP channel mid execution? |
| 17:16.51 | Qwell | tzafrir: yeah, it gets past that if I set EXTRA_CFLAGS += -DSTANDALONE_ZAPATA |
| 17:17.06 | tzafrir | set where? |
| 17:17.14 | Qwell | Makefile.kernel26 |
| 17:17.42 | Qwell | tzafrir: what's the kernel make command to prep a tree for building? |
| 17:17.51 | cpina | tzafrir: just today i prepared the conference in other computer (not dell) without any problem, and i took that dell with hpet computer to office to investigate or add hardware... |
| 17:17.52 | Qwell | I'm getting a genksyms not found error |
| 17:18.13 | Tourinho | is that possible to handle Spawn extension ? |
| 17:18.38 | tzafrir | Qwell, sorry, I don't remember |
| 17:18.46 | tzafrir | I don't think it is needed |
| 17:18.57 | De_Mon | Tourinho huh? |
| 17:23.00 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 17:23.08 | [TK]D-Fender | De_Mon: Thats what happens when you use Local withough doing "\n" at the end. It will hand over the channel the moment it hits a native bridge. |
| 17:23.16 | Qwell | /n |
| 17:27.04 | ManxPower | [TK]D-Fender: you mean /n, right? |
| 17:27.30 | Qwell | that was so 4 minutes ago |
| 17:27.41 | FlatFoot | a newline ? |
| 17:27.44 | [TK]D-Fender | Procrastination : the art of keeping up with yesterday. |
| 17:27.59 | trippss | ManxPower: do you have any ideas of how to use sipsak to issue a NOTIFY message such as that link to debug MWI? |
| 17:28.24 | ManxPower | trippss: If I did I would have said something. |
| 17:28.26 | FlatFoot | or are you leaning on the 'n' for some reason ( thats not nice mistreating random letters ) |
| 17:29.31 | trippss | anyone have any ideas of how sipsak works to issue custom messages? |
| 17:30.26 | De_Mon | Wait is a native bridge? |
| 17:31.32 | De_Mon | I will go learn more about this n you speak of |
| 17:32.00 | De_Mon | although, I really want it switched because I need those variables! |
| 17:32.41 | muiro | question: adding mailboxes. Does one need to reload the configuration to make changes? If so, what's the command? |
| 17:32.51 | De_Mon | Ahh I can do an answer.. sweet |
| 17:33.47 | [TK]D-Fender | muiro: "module reload app_voicemail.so" |
| 17:33.52 | *** join/#asterisk bmg505 (n=leon@196.209.183.81) |
| 17:33.59 | muiro | [TK]D-Fender: thank you |
| 17:36.00 | *** join/#asterisk Assid (n=assid@unaffiliated/assid) |
| 17:36.35 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 17:38.04 | De_Mon | oookay, Answer() didn't remove the local channel |
| 17:38.22 | De_Mon | stupid wiki, /me goes in search of real docs on Local/ channels |
| 17:38.39 | ManxPower | De_Mon: that is the wrong place to look. |
| 17:38.46 | ManxPower | De_Mon: can you guess the right place to look? |
| 17:40.00 | ManxPower | <PROTECTED> |
| 17:40.43 | ManxPower | Pop Quiz time! Everyone, what is the first place to look for Asterisk documentation? |
| 17:40.53 | Qwell | the book |
| 17:40.56 | ManxPower | no. |
| 17:40.58 | ManxPower | wrong! |
| 17:41.09 | ManxPower | The book is out of date before it goes to press. |
| 17:41.38 | ManxPower | There are two possible answers. 1) the CLI or 2) /path/to/src/asterisk-1.4/doc/ |
| 17:41.49 | ManxPower | What is the 2nd place to look for Asterisk documentation? |
| 17:41.55 | Qwell | the book |
| 17:41.56 | ManxPower | There are two possible answers. 1) the CLI or 2) /path/to/src/asterisk-1.4/doc/ |
| 17:42.00 | *** join/#asterisk angom (n=angom@200.38.31.239.dsl.dyn.telnor.net) |
| 17:42.10 | ManxPower | What is the 3rd place to look for Asterisk docs? |
| 17:42.12 | Qwell | the book |
| 17:42.16 | ManxPower | That's right! The Book! |
| 17:42.24 | ManxPower | What is the 4th place to look for Asterisk docs? |
| 17:42.26 | ManxPower | The Wiki. |
| 17:42.27 | Qwell | nowhere |
| 17:42.30 | Qwell | because the wiki sucks |
| 17:43.06 | De_Mon | The wiki is easier to get to alright! Yes I've learned my lesson and when the wiki gives me bad information I'll go to the docs. But only after! not first... |
| 17:43.17 | ManxPower | the mailing list would also be a good place to check before trying the Wiki, but the mailing list archives have almost as much wrong information as the Wiki does. |
| 17:43.27 | ManxPower | De_Mon: the wiki is frequently wrong! |
| 17:43.32 | De_Mon | liiiies |
| 17:44.16 | De_Mon | I need a web based docs repository I can search thru using google before I switch |
| 17:44.23 | ManxPower | De_Mon: how do you know the wiki gave you wrong information? For example the Wiki page about Asterisk dialplan variables, as of about a month ago only listed the variables in 1.0.x, but doesn't tell you that. 1.4 has at least 5 times as many dialplan variables, but you would not know that if you only looked the Wiki |
| 17:44.30 | anonymouz666 | why docs? "source code must be useful after all" |
| 17:44.49 | ManxPower | De_Mon: it's called ssh |
| 17:45.08 | ManxPower | BTW a similar problem exists with the list of applications on the Wiki |
| 17:45.08 | De_Mon | why must you rip out my heart and jump on it wearing cleats! |
| 17:45.28 | ManxPower | De_Mon: because you did something stupid. You checked the Wiki first. |
| 17:45.30 | De_Mon | the wiki just needs to be fixed |
| 17:45.40 | ManxPower | because you were too lazy to look in the asterisk source. |
| 17:45.42 | De_Mon | its a wiki after all, I blame YOU for not updating it |
| 17:45.49 | ManxPower | De_Mon: the wiki needs to be destroyed and restarted. |
| 17:46.00 | De_Mon | the suffffering |
| 17:46.01 | Greek-Boy | anyone here know how to upload a ringtone to a linksys SPA-942? The linksys ringtone generator appears to be for the SPA841 and does not upload to the phone. |
| 17:46.10 | ManxPower | De_Mon: If I spend 8 hours a day fixing the Wiki I'd never even come close to cleaning it up in a year. |
| 17:47.25 | De_Mon | you're so mean |
| 17:47.54 | *** join/#asterisk lemanal (n=lemanal@wifi-233-27.sc07.org) |
| 17:48.12 | ManxPower | De_Mon: If I was mean I'd let you happily use wrong information from the Wiki and not say anything. |
| 17:48.31 | De_Mon | strangely enough, the wiki and the doc/localchannel.txt match |
| 17:48.51 | De_Mon | I Answer()'d my local channel and it wasn't removed from the call path |
| 17:48.55 | ManxPower | De_Mon: the universe can be amazing sometimes, but most of the time it's just amazingly hateful. |
| 17:49.05 | De_Mon | a Wait(.1) did the trick though... |
| 17:49.30 | ManxPower | De_Mon: check the wiki dialplan variables page and compare it to the docs variables document |
| 17:49.38 | rob0 | "Never attribute to malice that which can be explained by stupidity." |
| 17:49.50 | De_Mon | Okay but before I do that, my alterego is going to update that page |
| 17:50.11 | [TK]D-Fender | rob0: If they do an especially good job of it, thats what the Darwin Awards are for :p |
| 17:50.13 | ManxPower | De_Mon: make sure you flag what variables are available in which Asterisk versions |
| 17:50.14 | De_Mon | That is one thing the wiki is really bad at, distinguishing 1.2 features from 1.4 |
| 17:50.18 | De_Mon | yah hehe |
| 17:50.32 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 17:50.38 | ManxPower | or at LEAST make a note on the page saying "this is correct for 1.4.x, and may not apply to earlier and later versions of asterisk" |
| 17:51.39 | De_Mon | but seriously, Local channel doesn't contain my variables so I need it out of the callpath is Wait(.1) 0 might work, the quickest way to drop it from the call path? |
| 17:51.53 | *** join/#asterisk Haq (n=liquid@6a.ab.5446.static.theplanet.com) |
| 17:52.03 | De_Mon | The doc says Just answering is sufficient but it's not |
| 17:52.28 | *** join/#asterisk irule (n=irule@200.53.61.4) |
| 17:52.28 | ManxPower | De_Mon: adding __ to the beginning of the variable name will make it be inherited by all child channels. |
| 17:52.48 | ManxPower | as you would have known if you had read doc/channelvariables.txt |
| 17:53.15 | De_Mon | Or I could just remove the local channel from the callpath and leave it at one _ |
| 17:54.21 | De_Mon | I notice you avoided the error I pointed out in the holy /doc |
| 17:55.03 | ManxPower | De_Mon: which error is that? |
| 17:55.37 | *** join/#asterisk agx (n=badpengu@81-174-9-212.dynamic.ngi.it) |
| 17:55.52 | ManxPower | I simply never have problems with Local/ and variables |
| 17:56.28 | De_Mon | If you Answer() the local channel its not removed from the call path, but a Wait(.1) does remove it. |
| 17:56.51 | De_Mon | <PROTECTED> |
| 17:57.15 | ManxPower | De_Mon: I would report that as a bug |
| 17:57.52 | De_Mon | No! no more bugs I don't want to find any more |
| 17:58.37 | jameswf | there is a movie called bug, rent it and watch it with a tweaker |
| 17:59.11 | ManxPower | De_Mon: you are using 1.4, there will be lots of bugs |
| 17:59.20 | De_Mon | 1.4.13 |
| 17:59.34 | De_Mon | its the no-bug release of 1.4 |
| 17:59.39 | ManxPower | most of them fairly minor or uncommon by now. |
| 17:59.39 | De_Mon | (supposed to be!) |
| 17:59.58 | De_Mon | I duno, within the last 2 days I've found 4 counting this one |
| 18:00.11 | De_Mon | they are minor thats true, and very obscure |
| 18:00.16 | ManxPower | De_Mon: And people wonder why I'm not upgrading from 1.2 |
| 18:00.40 | De_Mon | this dialplan is twisting asterisk in ways that its never even concieved of |
| 18:01.12 | ManxPower | De_Mon: My dialplan macros would make the Marquis de Sade blush. |
| 18:01.15 | De_Mon | Life without the featuers in 1.4 are unbarable |
| 18:01.37 | ManxPower | De_Mon: what specific 1.4 features? |
| 18:02.18 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 18:02.32 | De_Mon | I knew you were going to say that... |
| 18:03.15 | ManxPower | There are some 1.4 features that I like, but none that I can't live without. |
| 18:04.23 | ManxPower | And some features that I really want that did not make it into 1.4 and prolly won't make it into 1.6 either. One of those is the SIP qualify smoothing feature |
| 18:04.24 | De_Mon | meetme without review, func_odbc (backported I dont care its still not a feature) some new feature in voicemail, dictate (just ditched my dialplan version last week), GoSub, ChanSpy Whisper modes... |
| 18:04.45 | ManxPower | De_Mon: gosub is in 1.2 |
| 18:05.11 | ManxPower | we don't use MeetMe very often, we don't use databases, we don't use dictate, we don't use ChanSpy |
| 18:05.52 | De_Mon | I do, cuz I'm cool |
| 18:07.12 | robl^ | some people use MeetMe heavily -- for one of my install, it more heavily used than any other feature -- many people work at remote sites and its used daily for virtual conference rooms |
| 18:09.38 | cpm | yup |
| 18:09.39 | [TK]D-Fender | MeetMe is on a pseudo chopping block with 1.6 along with Chanspy. |
| 18:09.48 | cpm | meetme is about the only thing my * box gets used for. |
| 18:09.57 | ManxPower | [TK]D-Fender: what will replace their functionality. |
| 18:10.08 | De_Mon | 1.6 has some extra mediapath stuff that replaces chanspy |
| 18:10.24 | Qwell | meetme likely won't be outright replaced |
| 18:10.28 | ManxPower | [TK]D-Fender: removing ChanSpy isn't going to fix the deadlock problems |
| 18:10.32 | [TK]D-Fender | ManxPower: 1.6 will have a new bridging method that makes splicing & dicing channel extremely easy. |
| 18:10.34 | *** join/#asterisk _ys (n=yuri@236-069.nat.mns.ru) |
| 18:10.57 | [TK]D-Fender | ManxPower: MeetMe's current way is rather intensive. |
| 18:11.36 | De_Mon | I thought chanspy would still be there, just the internals would change |
| 18:11.51 | Greek-Boy | SetVar no longer used in 1.4? |
| 18:12.08 | mvanbaak | nope |
| 18:12.19 | mvanbaak | Set(${somevar}=blaat) |
| 18:12.20 | ManxPower | Greek-Boy: it is not, as you would have known if you had read UPGRADE.txt in the Asterisk source |
| 18:12.40 | Greek-Boy | ManxPower: yup I'm guilty |
| 18:13.05 | ManxPower | Greek-Boy: I hope you stop doing everything and go read it now. It has large amounts of information you need. |
| 18:13.16 | De_Mon | Why isn't Setvar depriciated from callfile docs? |
| 18:13.41 | Greek-Boy | ManxPower: will do that right away. |
| 18:13.53 | Greek-Boy | thanks |
| 18:14.06 | ManxPower | De_Mon: SetVar in the dialplan is an application, SetVar in a call file is a setting, not an application. |
| 18:14.18 | Qwell | [TK]D-Fender: all new polycoms are standard poe with no adapter needed, right? |
| 18:14.31 | Qwell | like the 330 and 430 |
| 18:15.10 | De_Mon | but it matches the application, it coulda been called ponies: and depriciating it wouldn't make sense, but it wasnt. |
| 18:15.12 | [TK]D-Fender | Qwell: IP 320,330,430,550,6XX all natively support PoE. IP 320/330 don't come with a wall-wart. |
| 18:15.23 | ManxPower | De_Mon: yes, that would make sense. |
| 18:15.40 | [TK]D-Fender | Qwell: And the IP 430 is all but pointless thanks to the 320/330 |
| 18:15.50 | Qwell | [TK]D-Fender: customer thinks the 330 is ugly |
| 18:15.51 | Qwell | heh |
| 18:15.52 | *** join/#asterisk _Sam-- (n=sam@fresco.kneedraggers.com) |
| 18:15.55 | Qwell | so I'm having him look at the 430 |
| 18:16.18 | [TK]D-Fender | Qwell: I wouldn't... and it costs more. |
| 18:16.36 | [TK]D-Fender | Qwell: IP430 = dead choice pretty much. |
| 18:18.23 | Qwell | ~phones |
| 18:18.24 | jbot | well, 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 like ... |
| 18:19.31 | mvanbaak | grandstream > * |
| 18:19.32 | mvanbaak | ;) |
| 18:20.06 | moemoe | snom 370! |
| 18:20.09 | [TK]D-Fender | Qwell: I'm going to redo that matrix.... |
| 18:20.19 | Qwell | the bani matrix? |
| 18:20.26 | *** join/#asterisk felix_da_catz (n=felix@66.195.225.1) |
| 18:20.27 | Qwell | where has he been, anyhow? He was a pretty cool guy.. |
| 18:20.28 | mvanbaak | yeah, move the cisco a bit up in the list wil ya |
| 18:21.29 | ManxPower | mvanbaak: tell Cisco to stop charging for SIP firmware and that might happen |
| 18:23.02 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:23.02 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 18:23.09 | De_Mon | Arnt the SPA's just rebranded ciscos? |
| 18:23.16 | ManxPower | De_Mon: no. |
| 18:23.27 | Qwell | Cisco is the one doing the rebranding |
| 18:23.41 | De_Mon | so cisco's are rebranded SPAs? |
| 18:23.48 | *** join/#asterisk ReD-MaN (i=daemon@172-220.static.golden.net) |
| 18:23.48 | Qwell | no |
| 18:23.53 | De_Mon | ok then |
| 18:23.56 | Qwell | the Cisco conference phones are rebranded polycoms |
| 18:24.17 | [TK]D-Fender | De_Mon: SPA's are a better build Sipura. |
| 18:24.22 | De_Mon | wonder where I got that spa rebranded idea |
| 18:24.26 | ManxPower | De_Mon: the SPAs are from SIPura, which Cisco bought. |
| 18:24.28 | Qwell | De_Mon: cisco owns them |
| 18:24.30 | Qwell | indirectly... |
| 18:24.34 | [TK]D-Fender | De_Mon: because Linksys bought SIPURA |
| 18:24.35 | Qwell | cisco > linksys > sipura |
| 18:24.36 | De_Mon | oooh that makes sense |
| 18:24.55 | De_Mon | thats funny |
| 18:25.00 | ManxPower | The SIPura/Linksys brand is Cisco's "consumer" brand. |
| 18:25.12 | [TK]D-Fender | De_Mon: So the SPA is jsut a physical build looking more like Cisco, but pretty much an SPA-841 inside. Firmware is entirely Sipura. |
| 18:25.14 | De_Mon | and their enterprise brand is polycom |
| 18:25.22 | ManxPower | they are totally different from the devices in Cisco's "corporate brand". |
| 18:25.31 | ManxPower | De_Mon: no it is not. |
| 18:25.47 | ManxPower | Cisco's enterprize brand is, oddly enough "Cisco" |
| 18:26.04 | De_Mon | which "brand" is polycom then |
| 18:26.09 | De_Mon | business? |
| 18:26.14 | *** join/#asterisk lemanal (n=lemanal@wifi-233-27.sc07.org) |
| 18:26.16 | De_Mon | corporate |
| 18:26.17 | ManxPower | De_Mon: The The Polycom brand is Polycom. |
| 18:27.21 | ManxPower | Polycom has nothing whatsoever to do with Cisco, except (maybe) Cisco licensed Polycom's speakerphone features and maybe rebrands the Polycom conference phones as Cisco conference phones, but Cisco's non-conference phones are made by Cisco, not Polycom. |
| 18:28.02 | ManxPower | Qwell: does Polycom's firmware work on the Cisco conference phones? |
| 18:28.13 | Qwell | ManxPower: don't know |
| 18:28.40 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 18:28.50 | [TK]D-Fender | ManxPower: I'd suspect they built the firmware for their stacks.... |
| 18:29.35 | *** part/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:29.36 | hmmhesays | vitelity international termination is pissing me off today |
| 18:29.43 | ManxPower | I suspect Cisco just licensed the speakerphone tech from Polycom and Cisco actually builds their own phones, but that is speculation. |
| 18:30.48 | nestAr | hmmhesays: vitelity termination in general has been flakey for a couple days. |
| 18:31.09 | nestAr | i couldn't register to their SIP server, they blamed my firewall, my isp, sun spots, etc.. |
| 18:31.19 | nestAr | changed to IAX, connects no problem |
| 18:31.38 | trippss | since i can't figure out sipsak, anyone have other testing tools that would allow me to send sip messages for testing? |
| 18:31.39 | nestAr | i am pretty sure my firewall didn't all of the sudden start blocking sip traffic.. ;) |
| 18:32.53 | [TK]D-Fender | trippss: have you considered... ASTERISK? ;) |
| 18:34.02 | ManxPower | [TK]D-Fender: he wants to lovingly hand craft the packets out of fair trade bits |
| 18:34.43 | [TK]D-Fender | ManxPower: With those little imperfections showing they were made by the honest hard working indigenous people of.... wherever... |
| 18:34.54 | ManxPower | [TK]D-Fender: exactly! |
| 18:35.13 | nestAr | lol |
| 18:35.42 | ManxPower | trippss: have you considered the sipsak mailing lists, documentation, wiki, or whatever other docs it has. |
| 18:36.04 | ManxPower | I doubt more than 1% of people on this channel have ever even heard of sipsak, let alone used it. |
| 18:36.12 | trippss | ManxPower: yeah I'm going to post that up there |
| 18:36.35 | ManxPower | I used it once, a few years ago, when we experimented with SIP IM to Polycom phones. |
| 18:36.37 | nestAr | man, dilemia.. hungry, do i a) cook (like, really cook, nothing to microwave) or b) get dressed and go get something |
| 18:36.51 | nestAr | tough call |
| 18:36.56 | ManxPower | The Polycom IM interface was so horrid, none of the users were interested in using it. |
| 18:37.23 | ManxPower | nestAr: going to get something without getting dressed might make the day more interesting |
| 18:37.38 | *** join/#asterisk CrashHD (n=crashhd@67-107-9-130.starstream.net) |
| 18:37.53 | trippss | taking a look at sipp too . . . |
| 18:38.06 | nestAr | well, i mean, i'm in sweats, i could go out, the girl at mc_whereever might look at me funny, but then again she's the one working at a fast food joint |
| 18:39.00 | rob0 | But then again, she's the one with the ... ahem, female equipment. |
| 18:39.18 | nestAr | rob0: probably 40 years old and 900 lbs |
| 18:39.19 | nestAr | PASS |
| 18:39.34 | nestAr | grill time! |
| 18:39.52 | rob0 | 40? What's wrong with 40? 900 pounds, yes, that could be a problem. |
| 18:40.39 | *** join/#asterisk murdmath (n=vircuser@mail.kimballequipment.com) |
| 18:40.46 | murdmath | howdy all. |
| 18:41.55 | murdmath | Is there a command at the asterisk console that will reset the state of a phone? When I do a show hints it shows an extension is busy when it isn't |
| 18:42.06 | De_Mon | where would I put a bug for local channels? |
| 18:42.23 | De_Mon | oh leme search maybe there are other bugs that'll give me a good idea |
| 18:42.37 | ManxPower | murdmath: nothing obvious in the output of "help" in the CLI? |
| 18:42.53 | murdmath | ManxPower: Nope, still searching. |
| 18:43.28 | Qwell | [TK]D-Fender: <Qwell> the cable that some of them come with (and that you buy for the 320/330) is basically just a poe injector |
| 18:43.29 | *** join/#asterisk VoicePulse (n=contact@unaffiliated/voicepulse) |
| 18:43.30 | Qwell | ^ true statement? |
| 18:44.16 | ManxPower | murdmath: what version of Asterisk? |
| 18:44.19 | [TK]D-Fender | Qwell: The IP 500/501 came with a non PoE Ethernet cable that their wall-wart plugged INTO |
| 18:44.47 | hmmhesays | can anyone point me at a decent international itsp? |
| 18:44.51 | ManxPower | Qwell: I believe the cable converts PoE into whatever the phone expects |
| 18:44.52 | [TK]D-Fender | Qwell: The IP 30X/50X both required ANOTHER special cable for PoE negotiation. |
| 18:44.58 | ManxPower | All ITSPs suck |
| 18:45.12 | De_Mon | oh there is a chan_local how did I not see it in the list... |
| 18:45.13 | hmmhesays | actually i've had good luck with vitelity, save a few destinations |
| 18:45.22 | [TK]D-Fender | Qwell: IP 320/330 do PoE natively and have a completely seperate wall-wart. |
| 18:46.00 | ManxPower | But the 300/301 have a separate wall wart, but don't do PoE native |
| 18:46.10 | [TK]D-Fender | ManxPower: Yes, as I said |
| 18:46.16 | murdmath | ManxPower: 1.2.x |
| 18:46.27 | murdmath | ManxPower: 1.2.22 |
| 18:47.41 | nestAr | rob0: i'm 27, i am trying to stay with in +/- 3 years of myself.. |
| 18:47.44 | *** join/#asterisk ghento (n=ghento@75.155.241.7) |
| 18:47.56 | nestAr | 40's not bad, if she's smokin' hot. |
| 18:48.16 | [TK]D-Fender | nestAr: Aston Kutcher & Demi Moore.... |
| 18:48.24 | *** join/#asterisk d-k-t (n=dt@125.120.138.74) |
| 18:48.51 | nestAr | true |
| 18:49.01 | nestAr | i retract the 40 year old statement |
| 18:49.03 | [TK]D-Fender | nestAr: But not a bad idea.I'm 32 now. At 29 I had a GF who was 35. A little out of place for a few morereasons too, but strangely the most in common with... |
| 18:49.12 | rob0 | I'm quite a few years past Hill 40, so I hear that and think "Hmmm ... young." :) |
| 18:49.21 | nestAr | s/40 years old/missing most of her teeth/ |
| 18:49.56 | [TK]D-Fender | I have always seemed to like mid 30's women.... |
| 18:50.13 | nestAr | my friend works loss prevention, he sent me a text the other day.. "dinner posponed, waiting for 5-0, this guy just stole more dvd's than he has teeth" |
| 18:50.13 | *** join/#asterisk Netyciukas (n=root@87-239-84-128.kis.user.lt) |
| 18:50.16 | nestAr | lol |
| 18:53.32 | *** join/#asterisk Netyciukas (n=bigman@87-239-84-128.kis.user.lt) |
| 18:55.18 | *** join/#asterisk Goldfisch (n=gturnqui@158-147-54-92.harris.com) |
| 18:57.25 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 19:03.37 | *** join/#asterisk gabiru (n=gabiru@213.37.159.28.dyn.user.ono.com) |
| 19:04.35 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 19:06.28 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 19:09.09 | *** join/#asterisk bantu (n=Miranda@p54A32B11.dip0.t-ipconnect.de) |
| 19:10.13 | *** join/#asterisk hohum (n=dcorbe@gate.globecommsystems.com) |
| 19:10.19 | hohum | guys |
| 19:10.29 | hohum | HOW do I get asterisk to STOP acting like an RTP proxy |
| 19:10.31 | hohum | I don't want it to |
| 19:10.34 | hohum | I have all my peers |
| 19:10.36 | hohum | canreinvite=yes |
| 19:10.36 | hohum | directrtpsetup=yes |
| 19:10.39 | hohum | and nat=never |
| 19:10.49 | hohum | but it still insists on being in the RTP stream |
| 19:11.39 | ManxPower | hohum: what options are on your Dial line? |
| 19:11.48 | ManxPower | hohum: how do you know it's still in the RTP stream? |
| 19:11.58 | ManxPower | hohum: what is the codec used for the channels? |
| 19:12.18 | hohum | exten => _X.,1,Dial(SIP/${EXTEN}@core) |
| 19:12.19 | hohum | that's it |
| 19:12.33 | hohum | and core is a peer in sip.conf with all 3 options |
| 19:12.33 | ManxPower | perhaps there's a codec mismatch |
| 19:12.47 | hohum | well one side shows codec G711 the other side shows Unknown |
| 19:13.00 | ManxPower | that might be the problem. |
| 19:13.22 | ManxPower | how do you know asterisk is still in the RTP stream? |
| 19:13.42 | hohum | because I turned rtp debugging on and I can see it sending packets |
| 19:14.15 | hohum | 206.71.169.114 1212937790 6daea89532f 00102/00000 unkn No Tx: ACK |
| 19:14.16 | hohum | 64.61.93.109 6319512967 151cbc9a3b3 00101/00102 ulaw No Rx: ACK |
| 19:14.16 | ManxPower | hohum: that would be a good way. |
| 19:14.31 | ManxPower | That is an ACTIVE call? |
| 19:14.39 | hohum | yes |
| 19:14.49 | ManxPower | I would not expect a format of unknown to work on an ACTIVE answered call. |
| 19:15.11 | ManxPower | hohum: what is your allow= and disallow= lines in sip.conf? |
| 19:15.52 | hohum | allow=all |
| 19:15.58 | ManxPower | don't do that |
| 19:16.09 | ManxPower | do a disallow=all and then allow= the ONE codec you want. |
| 19:16.58 | ManxPower | If the codec for the two legs of the call are different, then Asterisk must stay in the media path to transcode. |
| 19:18.40 | hohum | I think that did the trick |
| 19:19.20 | hohum | I don't see any more rtp packets at least when I do an rtp debug |
| 19:22.04 | hohum | so I think that worked, thanks |
| 19:28.16 | *** join/#asterisk wishes (n=wishes@60.234.20.178) |
| 19:28.16 | *** join/#asterisk grndslm (n=grndslm@24-116-87-97.cpe.cableone.net) |
| 19:29.22 | wishes | i have a problem :( i set up a bunch of sip DDIs to go to extensions, they all work except for the first one. all identical setup exactly, but th first one goes to the main line rather than the extension its supposed to |
| 19:29.28 | wishes | any ideas? |
| 19:29.56 | *** join/#asterisk UserReg_CL (n=dede@200.113.138.20) |
| 19:30.13 | kaldemar | wishes: are you using plain asterisk? |
| 19:30.30 | wishes | plain as in ? |
| 19:30.44 | kaldemar | plain as in without a gui for configuration. |
| 19:30.47 | wishes | oh yeah |
| 19:31.20 | wishes | they have guis? :D |
| 19:31.29 | kaldemar | pastebin your dialplan and tell what the DDI is so you'll be more likely to get help. |
| 19:31.38 | wishes | j/k, not sure i could handle a gui now - too used to console |
| 19:32.01 | kaldemar | you'll be better of without a gui. :P |
| 19:32.30 | UserReg_CL | helpme... need block all call in asterisk for one extension ? ... thank |
| 19:32.42 | hohum | heh |
| 19:32.44 | hohum | "DDI" |
| 19:32.47 | hohum | foreigners.... |
| 19:32.53 | hohum | they're called DIDs |
| 19:32.57 | hohum | ;) |
| 19:33.10 | Strom_M | holy god, you mean English isn't consistent across speakers?! WOW I NEVER REALIZED THAT !!!!!!!!!!!!!!!!!!!!!!!!!!!!! |
| 19:33.13 | Strom_M | moron |
| 19:33.37 | kaldemar | who are you calling a foreigner, foreigner? |
| 19:34.02 | UserReg_CL | (sorry by bad english, I am speak spanish) |
| 19:34.05 | hohum | kaldemar: you :P |
| 19:34.31 | irule | spanish == español |
| 19:34.32 | UserReg_CL | (alguien habla español) ? |
| 19:34.39 | irule | yo si |
| 19:34.43 | [TK]D-Fender | foreign films are definitely best left to foreigners... |
| 19:34.53 | wishes | kaldemar, http://pastebin.ca/773765 |
| 19:35.01 | *** join/#asterisk ZenBSDi (n=zenbsdi@unaffiliated/ZenBSDi) |
| 19:35.10 | wishes | its pretty cut down, but i pasted the relevant bits and had to edit out all the passwords etc :/ |
| 19:35.10 | *** part/#asterisk Sweeper (i=sweeper@softcheese.net) |
| 19:35.39 | ZenBSDi | how can I make asterisk accept an extension while a playback is happening? |
| 19:35.52 | ZenBSDi | I'm using WaitExten currently after PlayBack |
| 19:35.59 | hmmhesays | you should be using background |
| 19:36.00 | hmmhesays | or read |
| 19:36.04 | hohum | ZenBSDi: I think you want Background() not Playback() |
| 19:36.08 | muiro | perhaps I'm missing something. I want to get a date said in a natural spoken format. Whats the application I pass it to? And how do I format the string? I thought it was YYYYMMDDHHmm.ss-dow-doy. Is that not right? |
| 19:36.09 | ZenBSDi | ahhh |
| 19:36.14 | ZenBSDi | thanks guys! |
| 19:36.18 | irule | UserReg_CL to block calls you need to create trunks in their own [contects] and only include what you want |
| 19:37.08 | *** join/#asterisk Mercestes (n=Merceste@c-76-30-153-176.hsd1.tx.comcast.net) |
| 19:37.09 | wishes | kaldemar, was that any help? |
| 19:37.21 | irule | UserReg_CL so create [internal exts only], [local calls] and [long distance] and in your phone, include what you need |
| 19:37.22 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 19:38.13 | kaldemar | wishes: not really. to see where incoming calls land, you have to know the incoming context and take a look at the dialplan, i.e. extensions.conf. that was just parts of sip.conf. |
| 19:41.10 | *** join/#asterisk J4k3 (n=jsuter@openwrt.us) |
| 19:41.20 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 19:41.30 | wishes | kaldemar, im thinkjing it mnight be upstream |
| 19:43.12 | *** join/#asterisk Gamercjm (n=Gamercjm@pool-71-254-178-138.lsanca.fios.verizon.net) |
| 19:43.29 | UserReg_CL | (alguien habla español) ? |
| 19:43.35 | kaldemar | if it's upstream from your point of view and you don't have control over the next hop, then there's little that you can do. you have to bother someone else. |
| 19:44.19 | Gamercjm | is there a way in a context or in meetme to limit the conversation to like a minute or two? I know Dial() you can add a call length |
| 19:44.54 | [TK]D-Fender | Gamercjm: "show function TIMEOUT" |
| 19:45.27 | rob0 | I made 2 new ipkall numbers today, pointing to 2 different proxies. One, on Comcast in AL, works. The other, on AT&T/SBC DSL in AR, a busy signal. Nothing from ipkall hits the firewall in AR. |
| 19:46.25 | rob0 | Funny thing, a Stanaphone SIP number does ring in AR. So if AT&T is blocking SIP, they're not blocking *all* SIP. |
| 19:47.29 | rob0 | I realize that probably no one but ipkall could help me, but any ideas/WAG's would be appreciated. |
| 19:48.51 | muiro | question: Is there any asterisk application to convert a more human readable time value into unix time? |
| 19:49.43 | [TK]D-Fender | muiro: "show functions" <- go read, you'll find it.. |
| 19:49.48 | muiro | thanks |
| 19:51.13 | muiro | ah, yup, there it is |
| 19:52.58 | Gamercjm | [TK]D-Fender: Thanks |
| 19:54.35 | variable_office | what would be a good way to set asterisk up so that I can transfer callers to another individuals voicemail? |
| 19:55.27 | *** join/#asterisk Snake-eyes (n=hgjg@70.55.220.203.static.comindico.com.au) |
| 19:56.03 | hmmhesays | that should be a default config in voicemail.conf |
| 19:56.09 | *** join/#asterisk waverly360 (n=waverly@adsl-070-148-122-203.sip.bna.bellsouth.net) |
| 19:56.35 | variable_office | hmmhesays, what do you mean default config? |
| 19:56.37 | [TK]D-Fender | variable_office: Exten => 123,1,Voicemail(u456@wherever) ; Yippy-kai-yay |
| 19:57.12 | variable_office | how can i transfer someone to that extension; |
| 19:57.15 | waverly360 | Hey guys. Can anyone here tell me what the 'Status' field stands for in the 'QueueMemberStatus' event in the asterisk manager interface? I can't find it documented anywhere. |
| 19:57.26 | variable_office | when i transfer on the polycoms they require the other side to pick up first |
| 19:58.08 | [TK]D-Fender | variable_office: Thats because you're not choosing to do a BLIND TRANSFER |
| 19:58.27 | [TK]D-Fender | variable_office: It'd be a positively brilliant idea for you to have read the User's Guide.... |
| 19:59.10 | UserReg_CL | mmm |
| 19:59.46 | [TK]D-Fender | variable_office: And for reference, you don't transfer a call to "voicemail", you transfer it to an EXTEN taht does something you specifically want it to. |
| 20:00.11 | variable_office | yes, i understand that, i just didnt know how to do the blind transfrer thing |
| 20:01.05 | [TK]D-Fender | variable_office: if you'd read the Guide you'd know to do [Transfer] [Blind] destination [Send] |
| 20:03.20 | *** join/#asterisk chadkouse (n=chadkous@165.236.120.14) |
| 20:03.30 | variable_office | thanks for direction |
| 20:03.37 | chadkouse | hey guys -- what's the best way to allow a user to monitor another users call in real time ? |
| 20:03.49 | chadkouse | i'm guessing move it to a conference and mute the listener. |
| 20:06.56 | *** join/#asterisk techie (n=techie@adsl-76-214-20-56.dsl.lsan03.sbcglobal.net) |
| 20:08.00 | wishes | hmm ok ive traced my bug down to the following ... http://pastebin.ca/773804 - the Looking for 801 in default (domain ...) vs Looking for 099651031 in default (domain ...) which is the wrong one - why would it be looking for the full phone number ? |
| 20:08.36 | wishes | it should be " Looking for 801 in default .." |
| 20:09.08 | [TK]D-Fender | wishes: "fromuser" <- |
| 20:09.45 | wishes | ? |
| 20:10.57 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 20:11.05 | wishes | please explain? |
| 20:12.45 | wishes | the two lines are set up identically in every single way in the sip.conf - one works, one doesnt |
| 20:13.42 | *** join/#asterisk blackhole (i=Mishu@unaffiliated/blackhole) |
| 20:14.00 | *** join/#asterisk CrazyTux (n=CrazyTux@216-110-94-231.static.twtelecom.net) |
| 20:14.19 | blackhole | I am trying to develop a application which uses j2me to place SIP calls. Can someone tell me how can i place a sip call.. |
| 20:14.36 | CrazyTux | Hey guys, would you happen to know if in the default voicemail app if I can make it so like the traditional phone networks, if you call your own number, it goes to voicemail and you hit # or * and it'll ask you for your pin and you can check your voicemail remotely? |
| 20:14.55 | hmmhesays | anyone using broadvoice with asterisk 1.4? |
| 20:15.48 | wishes | blackhole, sounds like you need to confer with the RFCs |
| 20:16.12 | wishes | CrazyTux, yeah we have that set up here |
| 20:16.27 | wishes | actually no - its similar but different |
| 20:16.29 | blackhole | wishes, hmm Okay |
| 20:16.48 | CrazyTux | wishes, what exactly is involved? |
| 20:16.58 | *** join/#asterisk _ys (n=yuri@80.70.236.69) |
| 20:17.25 | *** join/#asterisk mihinomenest (i=9fmK@66.255.220.17) |
| 20:17.57 | *** join/#asterisk Gamercjm (n=Gamercjm@pool-71-254-178-138.lsanca.fios.verizon.net) |
| 20:18.24 | [TK]D-Fender | wishes: pastebin the full CLI output at verbose 10, SIP DEBUG enabled for a failed call attempt |
| 20:19.01 | wishes | [TK]D-Fender, eek, ok |
| 20:19.05 | wishes | ill give it a go |
| 20:19.16 | wishes | bit hard though now because everyone is in the office |
| 20:19.19 | *** join/#asterisk Voicemeup (n=VoiceMeU@modemcable132.108-83-70.mc.videotron.ca) |
| 20:19.22 | [TK]D-Fender | CrazyTux: this has nothing to do with Voicemail, but rather yrou DIALPLAN. |
| 20:19.24 | wishes | phones are in regular use from 20-30 people :/ |
| 20:19.34 | *** join/#asterisk gvasterisk (n=prueba@200.69.249.33) |
| 20:19.44 | Voicemeup | can i ask where you set the # of rings queues try to ring an agent for ? i have timeout.. but thats the whole queue timeout right ? |
| 20:20.06 | CrazyTux | [TK]D-Fender, how would I intercept the DTMF, after I've already launched the app? |
| 20:20.07 | gvasterisk | Hi ppl, I'm having a little problem with agi, shall I paste the code of the .php script here? |
| 20:20.15 | Voicemeup | no |
| 20:20.20 | Voicemeup | gvasterisk pastebin.ca |
| 20:20.28 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 20:20.32 | gvasterisk | what? |
| 20:20.39 | Voicemeup | paste code on pastebin.ca |
| 20:20.57 | Voicemeup | ~whatis pb |
| 20:21.06 | *** join/#asterisk Antarius (n=Antarius@pD9E0F824.dip0.t-ipconnect.de) |
| 20:21.15 | [TK]D-Fender | CrazyTux: You jsut said "go to vm if you dial your own number", what does that have to do with DTMF detection? |
| 20:21.29 | fujin | ~pb |
| 20:21.30 | 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 |
| 20:21.32 | blackhole | wishes, Actually i am looking for something similar to fring which can use my sip addr and its burned into my software so any call that uses my software would pass through my server. Do you know any sample of test app which i can try putting my sip add ect? |
| 20:21.39 | gvasterisk | ok 1 sec I'd paste it |
| 20:21.40 | CrazyTux | [TK]D-Fender, from an external line |
| 20:21.47 | Voicemeup | again sorry for pushing this to top |
| 20:21.48 | Voicemeup | can i ask where you set the # of rings queues try to ring an agent for ? i have timeout.. but thats the whole queue timeout right ? |
| 20:21.54 | CrazyTux | [TK]D-Fender, for example, from a pay phone on the street. |
| 20:22.00 | fujin | Voicemeup: there is no #rings timeout |
| 20:22.07 | [TK]D-Fender | CrazyTux: And as for how to enter you box on pressing "*" or "#", you should read up on Asterisk Standard Extensions. You seem to be short on at least 10% of them... |
| 20:22.35 | Antarius | Hello, I'm not sure if asterisk is the right software for me, so I ask here. It's possible to recieve and send a Fax with asterisk? |
| 20:22.36 | CrazyTux | [TK]D-Fender, I don't think your understanding the question. |
| 20:22.38 | [TK]D-Fender | Antarius: Is that all you want to do>? |
| 20:22.51 | [TK]D-Fender | CrazyTux: Yes, I do, and there are 2 parts to what you asked. |
| 20:23.17 | CrazyTux | [TK]D-Fender, What would that have anything to do with DialPlan, after the voicemail app takes over? |
| 20:23.36 | Antarius | [TK]D-Fender, Yes.....I want to build a fax server with remote entry for fax applications |
| 20:23.37 | [TK]D-Fender | CrazyTux: have you solved that first part? |
| 20:23.52 | CrazyTux | [TK]D-Fender, negative. |
| 20:23.59 | [TK]D-Fender | Antarius: Sounds like you should just get Hylafax |
| 20:24.01 | [TK]D-Fender | ~hylafax |
| 20:24.02 | jbot | A telecommunication system for UNIX systems. URL: http://www.hylafax.org |
| 20:24.28 | [TK]D-Fender | CrazyTux: First part was seeing if you called your own exten so as to act differently. You saying you haven't gotten that part fown? |
| 20:24.38 | wishes | [TK]D-Fender, mm its scrolling a tad to much |
| 20:24.40 | wishes | :/ |
| 20:24.49 | Antarius | jbot, Thank you |
| 20:24.49 | jbot | Antarius: bitte |
| 20:25.04 | CrazyTux | [TK]D-Fender, I'm not talking about calling my own extensions, or a pre-defined "authenticated" set of incoming numbers |
| 20:25.11 | gvasterisk | http://www.pastebin.ca/773840 <= there I've wrote what I've in my extensions.conf file and my file.php |
| 20:25.37 | CrazyTux | [TK]D-Fender, I'm talking about externally, calling my extension/DID. |
| 20:25.38 | gvasterisk | it should say those digits, ( the answer command is also in the dialplan ) I've forgotten to write it but I've tried |
| 20:25.48 | gvasterisk | and nothing happens |
| 20:25.55 | [TK]D-Fender | CrazyTux: Whatever. If you want to do something on pressing "*" or "#", thats what the "a", and "o" Asterisk Standard Extensions are for. |
| 20:25.55 | gvasterisk | I can't hear anything |
| 20:26.41 | Voicemeup | anyone ? |
| 20:26.58 | gvasterisk | the debug console says: AGI script file.php, completed returning 0 |
| 20:27.06 | gvasterisk | but nothing happens :( |
| 20:27.10 | [TK]D-Fender | Voicemeup: timeout <---- |
| 20:27.23 | Voicemeup | aint htat for wholel queue wait time ? |
| 20:27.35 | Voicemeup | how you limite whole queue then ? with the queue cmd ? |
| 20:29.15 | blackhole | wishes, Actually i am looking for something similar to fring which can use my sip addr and its burned into my software so any call that uses my software would pass through my server. Do you know any sample of test app which i can try putting my sip add ect? |
| 20:30.00 | [TK]D-Fender | Voicemeup: Correct |
| 20:30.52 | Voicemeup | k thanks |
| 20:31.51 | jameswf | ~rtfm |
| 20:31.52 | jbot | rtfm is probably Read The F*cking Manual (TM). It is a suggestion to do your homework before posting a question. Sometimes used as RTFM $SPECIFIC_MANUAL to refer to a specific source of information. See also http://en.wikipedia.org/wiki/RTFM |
| 20:32.05 | gvasterisk | can anyone help me? with php agi |
| 20:32.11 | Voicemeup | gvasterisk never PM anyone without asking first..thanks |
| 20:32.18 | Voicemeup | and btw no dont pm me |
| 20:32.31 | gvasterisk | oh ok..... |
| 20:32.42 | Voicemeup | its like some one entering your house for loose hcange |
| 20:32.47 | lirakis | can i specify e164 or noa when sending PRI ? |
| 20:32.53 | Voicemeup | you just want to kick them out |
| 20:33.05 | gvasterisk | ok man! |
| 20:33.05 | Voicemeup | gvasterisk.. you need to read on it. .or hire people to do it |
| 20:33.29 | gvasterisk | ok thank u |
| 20:34.02 | Mercestes | This coming from the man asking how to set the # of rings on a call queue. >.> |
| 20:34.16 | Voicemeup | yes |
| 20:34.20 | Voicemeup | tought it changed in 1.4 |
| 20:34.22 | muiro | hoo boy. Application to "say" currency values? |
| 20:34.28 | Voicemeup | since now its not behaving same way |
| 20:34.45 | Voicemeup | nm anyhow we dropping asterisk.. just doing vm now |
| 20:34.47 | Voicemeup | thanks again |
| 20:34.48 | *** part/#asterisk Voicemeup (n=VoiceMeU@modemcable132.108-83-70.mc.videotron.ca) |
| 20:34.59 | Mercestes | Awww, I hurt his feelings. |
| 20:36.38 | fujin | pwnt |
| 20:36.45 | fujin | chalk one up for Mercestes |
| 20:38.02 | steve | has anyone successfully got an X100P to receive caller ID data on a UK BT line? |
| 20:38.15 | Mercestes | gvasterisk, the "returning 0" means your php script completed without errors. |
| 20:38.58 | Mercestes | gvasterisk, If you include #phpagi shouldn't you be able to exec the say_digits function (or whatever it's called) instead of writing to stdout? |
| 20:39.12 | Mercestes | Most of the dialplan commands are functions in phpagi as I recall. |
| 20:40.50 | Mercestes | I don't see anything glaringly wrong with your code, however, although I'm not familiar with the writing commands to stdout method. I tended to use phpagi when I was using AGIs. |
| 20:42.00 | [TK]D-Fender | Mercestes: 'Mornin sunshine! |
| 20:42.14 | Mercestes | [TK]D-Fender, hey, baby. How's it hangin? |
| 20:42.25 | [TK]D-Fender | Mercestes: Currently... to the left. |
| 20:42.35 | Mercestes | damn left wingers... |
| 20:42.56 | Mercestes | You should be more liberal withit. |
| 20:43.03 | [TK]D-Fender | Mercestes: Not quite... Libertarian more like... RP 2008! |
| 20:43.40 | [TK]D-Fender | Mercestes: Either him, Kucinich, or Gravel. Basically the most reviled of their respective parties ;) |
| 20:43.52 | Mercestes | gvasterisk, just as a random thought, if * was seeing \n as text then say digits would freak out and do nothing, btw. just a thought. |
| 20:44.18 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 20:44.25 | Mercestes | Yes, I am so very much looking forward to see how the US will do worse than GW in the next term. |
| 20:45.19 | Mercestes | our presidents have gotten so bad they aren't even worth shooting anymore. |
| 20:46.37 | [TK]D-Fender | Mercestes: 2 words : Rudy Giuliani = Most likely to start WW3. |
| 20:47.00 | [TK]D-Fender | Mercestes: Sure they are, as long as you line of the VP to cut back on munitions cost! |
| 20:47.11 | [TK]D-Fender | s/of/up/ |
| 20:47.32 | Mercestes | WW3 would be good for the economy tho. |
| 20:47.45 | Mercestes | We coudl use asterisk for encrypted communications. WE'd all be employed. |
| 20:48.41 | [TK]D-Fender | Mercestes: And when you're glowing in the dark night fishing will be a breeze! |
| 20:49.12 | Mercestes | And no one would loose me again! |
| 20:49.24 | Mercestes | ...the sterility thing is kind of a downer but..people like me shouldn't reproduce anyways. |
| 20:49.57 | [TK]D-Fender | Mercestes: You can be a (non-re)productive member of society once more! |
| 20:50.11 | Mercestes | Yes! |
| 20:50.46 | Mercestes | gvasterisk, Did I manage to utter anything useful to you? |
| 20:50.57 | wishes | oh well ive fired off an email to our provider for now |
| 20:51.01 | Mercestes | a nod, a grunt, the bird...some kind of response....>.> |
| 20:51.03 | wishes | see if its something on their end |
| 20:51.10 | slowshutt | ok i installed the g729 licences to try and sort out my delay on speech for iax connections, no luck not working like i exspected it to work, anyone know how to to tweek it? |
| 20:52.39 | Mercestes | slowshutt, What kind of delay on speech? Like, 1-2 seconds before a voice path is established? Or you say something and it's 3 seconds before they hear you? |
| 20:53.24 | slowshutt | echo on line |
| 20:53.39 | [TK]D-Fender | slowshutt: Whats on each end? |
| 20:54.56 | slowshutt | i am connecting to a paid service if i connect to them using just a snom phone there is no echo, they insisted i get a g729 licence to work with their equipment |
| 20:55.24 | slowshutt | is the snom phones licence for g729 codex? |
| 20:55.37 | [TK]D-Fender | slowshutt: Sure. |
| 20:55.38 | slowshutt | sorry licenced |
| 20:55.53 | [TK]D-Fender | slowshutt: that should have no impact on echo. |
| 20:56.04 | slowshutt | that explain why i can connect to them without any hastles |
| 20:56.06 | [TK]D-Fender | slowshutt: Its either your phone (unlikely), or their service. |
| 20:56.28 | slowshutt | the snom phone is working fine |
| 20:56.30 | Mercestes | Or the network between you and them. |
| 20:57.00 | slowshutt | its when i set up a sip connection to them using asterisk where everthing seems to go downhill |
| 20:57.14 | slowshutt | i have a 4m dsl line at home |
| 20:57.29 | Mercestes | slowshutt, and what's the upspeed? |
| 20:57.45 | slowshutt | 580kb |
| 20:58.12 | Mercestes | slowshutt, and, you said you have echo on IAX, now it "goes downhill" on SIP. |
| 20:58.14 | slowshutt | i am using this link purely for asterisk |
| 20:58.31 | Mercestes | Doesn't matter, residential broadband is very hit and miss for VoIP. |
| 20:58.34 | slowshutt | sorry its a sip connect |
| 20:58.43 | Mercestes | Ok, well sip connections do not echo... |
| 20:59.16 | Mercestes | service providers *DO* play your voice back to you to assure you that you are indeed talking...and if there is a substantial delay in their playback, or the network, then it could be mistaken for an echo. |
| 20:59.24 | slowshutt | i have a echo on inbound and outbound calls |
| 20:59.29 | Mercestes | It is, however, a recording of your voice being played back to you. |
| 20:59.51 | slowshutt | link please |
| 20:59.56 | Mercestes | what are you using for a phone when you setup your sipconnect? |
| 21:00.01 | Mercestes | Link? |
| 21:00.03 | Mercestes | ~sipecho |
| 21:00.06 | slowshutt | snom |
| 21:00.11 | Mercestes | damn you. |
| 21:00.32 | Mercestes | So the snom connected directly tot hem works but a sipconnect using the snom hooked to your asterisk does not? |
| 21:00.42 | slowshutt | yes |
| 21:00.56 | Mercestes | perfect. DO you have "keepalive" or something similiar set on yoru SNOM? |
| 21:01.14 | *** join/#asterisk remmo (n=junk@203.32.47.250) |
| 21:01.18 | slowshutt | loaded 4 channel licence, shows me its using the g729 channels |
| 21:02.15 | *** join/#asterisk galeras (n=Martin@201.244.247.149) |
| 21:02.24 | slowshutt | no all standaard out of the bos just set up identity ond of i go , rings the instand you dial the number, no echo no delay |
| 21:03.55 | slowshutt | i could not get the asterisk to answer the calls for i didn't have g729 lics, after install rings sip phone no problem dialout works fine, its just the delay or echo thats bugging me |
| 21:03.57 | Mercestes | Hrm. Try setting qualify= to a numerical value, something obscene like 200 or 300 and test it. |
| 21:04.12 | slowshutt | k |
| 21:04.18 | Mercestes | This should "force" an active path between you adn then, but don't leave it like that. |
| 21:04.25 | Mercestes | it generates a bit of traffic. |
| 21:05.13 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 21:08.18 | slowshutt | what does that prove if it decreased the echo |
| 21:08.44 | *** part/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 21:08.56 | [TK]D-Fender | slowshutt: Only proves there is nothing consistent about your problem whatsoever |
| 21:09.10 | BBHoss | heh, thats an all to common situation :) |
| 21:09.15 | Mercestes | It proves there is slow routing between you and your provider. |
| 21:09.23 | [TK]D-Fender | Mercestes: Poor test. |
| 21:09.42 | Mercestes | I disagree. |
| 21:09.45 | slowshutt | show me how [TK]D-Fender |
| 21:09.55 | BBHoss | you could also try an iperf test, to check for jitter and such |
| 21:10.36 | [TK]D-Fender | Jitter != echo. |
| 21:10.42 | Mercestes | slowshutt, Do an MTR and and bring up all the fields. Particularly jitter and st.dev |
| 21:11.03 | [TK]D-Fender | Echo = phone gains are whacko - or - ITSP sucks ass. |
| 21:11.10 | BBHoss | yes i know that, it was just a suggestion because jitter can cause all sorts of problems |
| 21:11.22 | [TK]D-Fender | BBHoss: Except the one he's asking about. |
| 21:11.31 | Mercestes | What we did was bombard your network with packets to try and force yourself into a "static route" with the arp tables. If no one has to lookup where you are you get a faster path. |
| 21:11.44 | BBHoss | how do you know EXACTLY what he's experiencing? |
| 21:12.30 | Mercestes | It's not uncommon on residential broadband to have slow lookups. |
| 21:12.42 | Mercestes | If you are using domain names I would suggest using IP addresses. |
| 21:12.56 | [TK]D-Fender | BBHoss: slowshutt>i have a echo on inbound and outbound calls <-- is this not clear to you somehow? Sounds pretty blatant to me... |
| 21:12.57 | BBHoss | basically with VoIP, the best way to make sure you have a good connection is to pay for one |
| 21:13.08 | Mercestes | Basically |
| 21:13.56 | *** join/#asterisk svenna (n=svenna@p548D0704.dip0.t-ipconnect.de) |
| 21:14.30 | *** join/#asterisk Cyon (n=cyon@216.179.31.170) |
| 21:15.21 | Mercestes | Welp, I gotta get going. Good luck. |
| 21:15.40 | BBHoss | slowshutt: what provider are you using? |
| 21:16.21 | slowshutt | for? |
| 21:16.31 | BBHoss | ITSP and ISP |
| 21:16.48 | slowshutt | telfree and internet sulutions |
| 21:16.58 | slowshutt | south africa |
| 21:17.02 | BBHoss | ahh |
| 21:17.22 | BBHoss | sounds to me like its probably your provider |
| 21:17.33 | slowshutt | why? |
| 21:17.51 | slowshutt | they are not free |
| 21:18.12 | BBHoss | well, if you have echo when you're on pure voip, then its most likely an issue wher your provider terminates the calls |
| 21:18.33 | BBHoss | just because you pay for them doesn't mean they don't suck |
| 21:18.48 | [TK]D-Fender | ^^^ very true |
| 21:19.16 | BBHoss | i don't know of any good SA providers, as i don't live there ) |
| 21:19.41 | slowshutt | the connection works fine on a snom phone(using my ISP with the same service provider) connecting with sip and utilising their service, when i try it into fy faultarisk box it just goes all wrong |
| 21:20.04 | slowshutt | true bbhoss |
| 21:20.12 | BBHoss | retype that last sentence |
| 21:20.28 | slowshutt | true bbhoss |
| 21:20.36 | fujin | when i try it into fy faultarisk box it just goes all wrong |
| 21:20.55 | BBHoss | what are you trying to do when it goes wrong? |
| 21:21.32 | slowshutt | reload sip on falarisk |
| 21:21.56 | BBHoss | when you get the echo |
| 21:22.26 | slowshutt | fix it |
| 21:22.29 | CrazyTux | How does asterisk handle mailbox cleanup ? of messages in /var/spool/asterisk/voicemail/MAILBOX when say a mailbox is deleted, will it automatically clean that up, or would I have to manualyl do that? |
| 21:23.01 | slowshutt | voicemail removes the wave and text file |
| 21:23.12 | slowshutt | when you delete it |
| 21:23.12 | BBHoss | ok dude, im not understanding you, restate your problem |
| 21:23.28 | CrazyTux | BBHoss, me? |
| 21:23.35 | BBHoss | no slowshutt |
| 21:23.47 | CrazyTux | slowshutt, do you know over what duration / time period it would delete the file? |
| 21:23.54 | CrazyTux | s/file/files/ |
| 21:24.56 | slowshutt | i live in south africa and it looks to me although i have decent speed line the people i am connecting to to use there not so free trunk trouting failities, i thing the problem is on their side |
| 21:25.41 | BBHoss | yes but are you experiencing the "echo" problem when using the snom phone or not? |
| 21:26.01 | [TK]D-Fender | slowshutt: Test between 2 phones only directly linked to your server. If thats fine, its your ITSP |
| 21:26.01 | slowshutt | i just connected two of my boxes one about 100km away from the othe, both using the same isp, sip connect between the two with no echo or delay |
| 21:26.19 | BBHoss | id say get a new itsp |
| 21:26.30 | [TK]D-Fender | heading out, BBL |
| 21:26.49 | slowshutt | that sounds about right BBHoss |
| 21:27.29 | BBHoss | does it only echo sometimes, or all of the time? |
| 21:28.00 | slowshutt | Cheers TK |
| 21:28.28 | slowshutt | no echo now if i connect my two asterisk boxes |
| 21:29.01 | slowshutt | permanent when i connect to tellfree(not free) |
| 21:29.19 | BBHoss | yeah, its most likely on there end |
| 21:31.07 | *** join/#asterisk Gamercjm (n=Gamercjm@pool-71-254-178-138.lsanca.fios.verizon.net) |
| 21:31.56 | Gamercjm | What is the command to view how many actives calls are going through? |
| 21:31.57 | BBHoss | you said your isp is internet solutions? |
| 21:32.14 | slowshutt | yes |
| 21:32.32 | BBHoss | http://www.is.co.za/vois/VoIS.htm |
| 21:32.52 | BBHoss | they might let you use their network |
| 21:33.46 | slowshutt | BBHoss looks like you are a local here in south africa |
| 21:33.57 | BBHoss | no, i just google well :) |
| 21:35.11 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 21:35.22 | slowshutt | they are not that good had a big fight with them the other day got a 4 m link with uncapped adsl at home, then limited me to speed of a 512k link |
| 21:35.40 | slowshutt | a322holes |
| 21:37.25 | TJNII | ~fax |
| 21:37.26 | jbot | Well, apperantly the fax was concieved of by Napoleon Bonaparte. He commissioned a system of devices that could transmit a traced image electrically over telegraph lines to a remote device that would redraw the image identically. |
| 21:37.46 | TJNII | Humorous, but worthless.... |
| 21:38.04 | *** join/#asterisk fholmes (n=felix@66.195.225.1) |
| 21:38.17 | BBHoss | Gamercjm: "core show channels" on asterisk 1.4 |
| 21:39.10 | JT | slowshutt: echo is from the provider |
| 21:39.46 | slowshutt | figured it out thanks all and belated JT |
| 21:41.34 | slowshutt | what type of server should i look at if i want to get 4 pri with 1800 did lines to create my own trunk service? |
| 21:42.15 | JT | slowshutt: what was the problem? |
| 21:42.55 | TJNII | I assume the difference between G711u and G711a is u-law and a-law? |
| 21:43.02 | BBHoss | so you're going from a cheapo ITSP to 4 pris? |
| 21:43.33 | slowshutt | become my own ITSP |
| 21:43.54 | BBHoss | heh |
| 21:44.22 | BBHoss | how many channels are PRIs there? |
| 21:44.33 | waverly360 | wow..my first entry into voip-info.org...such a happy day.. :P |
| 21:44.44 | slowshutt | won't call it telfree, something more like notforfree |
| 21:44.50 | slowshutt | 30 ch |
| 21:45.06 | *** join/#asterisk rpm (n=russell@75.153.47.179) |
| 21:45.09 | BBHoss | so 120 simultaneous channels |
| 21:45.23 | slowshutt | i've been in the pabx game for +-20 years |
| 21:45.28 | JT | BBHoss: generally 30 channels in most places out side of north america and japan |
| 21:45.35 | slowshutt | asterisk is sure a eye opener |
| 21:45.36 | BBHoss | yeah i wasnt sure though |
| 21:45.55 | JT | you probably want a sip proxy |
| 21:45.58 | JT | and a few backends |
| 21:46.18 | waverly360 | Alas..no one cares... |
| 21:46.29 | BBHoss | heh |
| 21:46.33 | waverly360 | :) |
| 21:46.40 | Wonka | go cry, emo kid ;) |
| 21:46.47 | hmmhesays | LOL |
| 21:46.56 | hmmhesays | thats cry and let everyone feel your pain |
| 21:47.17 | waverly360 | heh heh |
| 21:47.30 | waverly360 | goodness..bad storm here |
| 21:47.37 | waverly360 | it was bright and shiny outside a few minutes ago |
| 21:47.48 | BBHoss | oh youre up in tennesee eh? |
| 21:47.53 | waverly360 | indeed |
| 21:48.05 | waverly360 | it got dark enough to make all the parking lot lights come on early |
| 21:48.07 | waverly360 | sweeeet |
| 21:48.13 | BBHoss | yeah |
| 21:48.23 | BBHoss | too bad it didnt come down further south |
| 21:48.32 | BBHoss | i like storms :) |
| 21:48.34 | waverly360 | You like bad storms too eh? |
| 21:48.35 | waverly360 | :) |
| 21:48.48 | waverly360 | yeah..I really dig em. I used to love them a lot more when I didn't own a house |
| 21:49.04 | BBHoss | im sure you know, but it has been a dry season in north alabama these past 2 years |
| 21:49.08 | waverly360 | now there's that tinge of fear in the back of my mind....will my house still be there when I get home? |
| 21:49.17 | waverly360 | I can imagine |
| 21:49.28 | waverly360 | this has been the worst summer that I can remember |
| 21:49.47 | waverly360 | a lot of farmers that I know (<- is from the stix) had a horrible time this year |
| 21:49.54 | BBHoss | yeah me too |
| 21:50.11 | BBHoss | you can't live in the south and not know smalltime farmers |
| 21:50.25 | waverly360 | I suppose that's true enough |
| 21:50.32 | waverly360 | I'm still from the stix though :) |
| 21:50.37 | waverly360 | redneck geek |
| 21:50.55 | BBHoss | me too |
| 21:51.02 | waverly360 | sweet |
| 21:51.45 | BBHoss | main problem is though, every event that happens in the tech world is usually at least 500 miles away |
| 21:51.56 | BBHoss | like expos and such |
| 21:52.25 | fetcher | Are there any well-known problems with Grandstream Handytones to watch out for? I can't seem to get an HT-288 to work |
| 21:52.53 | BBHoss | heh |
| 21:52.56 | *** join/#asterisk nhuisman_work (n=nhuisman@hiloguest223.IfA.Hawaii.Edu) |
| 21:53.03 | waverly360 | BBHoss: that's why you work for people who send you out to said events |
| 21:53.08 | fetcher | it registers successfully into the SIP database, but never transmits anything else |
| 21:53.15 | BBHoss | that would be nice |
| 21:53.16 | nhuisman_work | does anyone know if there exists wireless headsets for VOIP that don't use wifi? |
| 21:53.20 | slowshutt | how you coping with the gumlaws over there |
| 21:53.24 | nhuisman_work | ie 900mhz, 2ghz, 5.8ghz? |
| 21:53.25 | slowshutt | gun |
| 21:53.30 | BBHoss | HAHA |
| 21:53.31 | *** join/#asterisk stybba (n=stybba@190.10.0.136) |
| 21:53.40 | nhuisman_work | our building has really bad wifi reception and I have users that want wireless handsets |
| 21:53.43 | vlt | Hello. Are there any known problems using an NFS mount for /var/spool/asterisk/monitor? |
| 21:53.50 | BBHoss | nhuisman_work: check out AASTRA SIP-DECT |
| 21:54.07 | slowshutt | try bluetooth headsets |
| 21:54.15 | BBHoss | he meant handsets i think |
| 21:54.22 | waverly360 | nhuisman_work: We have a couple of people that use some nice wireless plantronics headsets that attach to a polycom. |
| 21:54.33 | waverly360 | nhuisman_work: the range on them is pretty good too. |
| 21:54.35 | nhuisman_work | I have users that want to be able to take their phone out of their office a ways |
| 21:54.42 | nhuisman_work | or just walk around in their office with a phone I guess |
| 21:54.50 | BBHoss | yeah you want a handset |
| 21:54.54 | BBHoss | you want DECT |
| 21:54.59 | nhuisman_work | looking at that now |
| 21:55.04 | BBHoss | and the best american version is the aastra |
| 21:55.06 | slowshutt | plantronic blueteeth headsets work the best |
| 21:55.09 | BBHoss | afaik |
| 21:55.23 | nhuisman_work | those plantronik they just pair with a base station? |
| 21:55.45 | slowshutt | how do you tie in dect with asterisk |
| 21:55.48 | *** join/#asterisk olinux (i=olinux@ip68-107-11-147.sd.sd.cox.net) |
| 21:55.50 | BBHoss | SIP |
| 21:56.02 | waverly360 | nhuisman_work: yeah.. cs70n is the model number..only 300 ft range though |
| 21:56.06 | nhuisman_work | wait |
| 21:56.12 | nhuisman_work | those dect use their own antennas? |
| 21:56.13 | slowshutt | do you get cordless ip phones? BBHoss? |
| 21:56.18 | BBHoss | snom also has an offering, called the M3, though i have not tried it |
| 21:56.29 | BBHoss | i have used the SIP-DECT from aastra |
| 21:56.40 | waverly360 | Aastra has a neat system that I should be getting in this week. Their Aastra 142 dect phone |
| 21:56.41 | nhuisman_work | that doesn't really solve my problem |
| 21:56.52 | nhuisman_work | since the office are far apart |
| 21:56.54 | BBHoss | why not |
| 21:56.57 | BBHoss | heh |
| 21:57.00 | BBHoss | welcome to DECT |
| 21:57.01 | nhuisman_work | i'd still have to put those everywhere |
| 21:57.02 | waverly360 | the phones tie into a base station that uses sip over dect |
| 21:57.07 | slowshutt | i hate south africa we are the rugby world champions but we dont have cordless ip phones |
| 21:57.17 | slowshutt | what is the range on these ip phones? |
| 21:57.24 | nhuisman_work | maybe i'm not understanding how the dect systemw orks |
| 21:57.26 | BBHoss | what the sip-dect? |
| 21:57.31 | waverly360 | nhuisman_work: I've been fighting the wifi thing for a LONG time....you're not gonna have a satisfied customer using a wifi phone |
| 21:57.40 | nhuisman_work | but that seems like the same as having a wifi phone |
| 21:57.43 | BBHoss | DECT over wifi ANY day |
| 21:57.53 | nhuisman_work | what does dect give you over wifi? |
| 21:57.58 | slowshutt | why wav tech not up to scratch yet? |
| 21:57.58 | BBHoss | it is, but it is DESIGNED for voice, uses it own frequency |
| 21:58.12 | waverly360 | plus the technology requires WAY less power |
| 21:58.18 | waverly360 | wifi phone..you might get 4 hours of talk time |
| 21:58.29 | BBHoss | yeah youll get like 60% or more battery life |
| 21:58.32 | waverly360 | a dect phone will get up to 12 hours talk |
| 21:58.37 | waverly360 | and days of stdby |
| 21:58.48 | fetcher | wifi has huge per-packet overhead ... and VoIP tends to send large numbers of very small packets. Not a good combination |
| 21:59.05 | waverly360 | only problem is that you can't use existing wireless infrastructure..you have to put dect access points everywhere |
| 21:59.06 | slowshutt | do you tei in dect analog sets using fxs ports? |
| 21:59.08 | waverly360 | but..it's worth it |
| 21:59.09 | nhuisman_work | how come people don't just make a normal wireless handsets? |
| 21:59.09 | hmmhesays | thats why you turn your cap time up |
| 21:59.13 | BBHoss | if you want wireless+sip, you have to get DECT |
| 21:59.16 | fetcher | plus the handoff between different access points can be dodgy, in settings with more than one AP |
| 21:59.18 | hmmhesays | so you have lower amounts of small packets |
| 21:59.26 | waverly360 | slowshutt: you can..but there are systems out there that work better |
| 21:59.33 | nhuisman_work | i have about 8 users far apart which want wireless handsets |
| 21:59.38 | BBHoss | handoff is seamless with DECT |
| 21:59.40 | nhuisman_work | and our walls suck for wireless signals it seems |
| 21:59.47 | BBHoss | do they need to work everywhere? |
| 21:59.52 | nhuisman_work | just in their offices |
| 21:59.59 | BBHoss | what kind of walls |
| 22:00.01 | nhuisman_work | or a bit outside |
| 22:00.14 | waverly360 | Here..everyone.. http://abptech.com/products/Aastra/sip_dect.html |
| 22:00.21 | BBHoss | my dads building sucked with anything 2.4ghz |
| 22:00.25 | waverly360 | I haven't used them yet..but it looks promising |
| 22:00.25 | fetcher | hmmhesays: yeah, it's a tradeoff between efficiency and voice lag, though 30ms (iLBC default) is tolerable |
| 22:00.33 | nhuisman_work | i guess rebar and cement |
| 22:00.36 | BBHoss | then we moved to DECT and it works great everywhere |
| 22:00.46 | nhuisman_work | I just don't want to have to blanket the building with dect and wifi |
| 22:00.52 | waverly360 | those access points have a 300,000 square foot range in an open space |
| 22:00.53 | BBHoss | his bulding is cinderblock walls, with some sheet metal |
| 22:01.04 | waverly360 | and can support 8 phones talking through them at one time |
| 22:01.05 | BBHoss | dect and wifi don't interfere |
| 22:01.23 | waverly360 | nhuisman_work: you're just giving everyone cancer at an earlier age..no big deal :) |
| 22:01.29 | nhuisman_work | heh |
| 22:01.56 | BBHoss | nhuisman_work: you want me to send you some REALLY detailed info on the SIP-DECT? |
| 22:02.06 | nhuisman_work | wonder if they make phones with bluetooth handsets |
| 22:02.08 | slowshutt | those access points have a 300,000 square foot range in an open space( in the salt lakes or deserts) |
| 22:02.12 | waverly360 | nhuisman_work: get ready for the firehose.... |
| 22:02.18 | waverly360 | BBHoss: :) |
| 22:02.28 | BBHoss | ive got the training presentation for them |
| 22:02.43 | waverly360 | BBHoss: oh really? I need that...gonna have to play with mine soon |
| 22:02.55 | BBHoss | hang on ill put it on teh interwebz |
| 22:03.00 | waverly360 | wootz |
| 22:03.22 | nhuisman_work | basically I want self-contained wireless handsets, not a building level solution. |
| 22:03.54 | BBHoss | yeah they dont have to be able to talk to each other, as long as they are all plugged into the network |
| 22:03.57 | waverly360 | oh |
| 22:04.04 | BBHoss | well |
| 22:04.05 | waverly360 | maybe try something like the aastra 480i |
| 22:04.18 | BBHoss | 480iCT |
| 22:04.26 | waverly360 | yeah..that one |
| 22:04.29 | BBHoss | but thats 2.4ghz |
| 22:04.44 | nhuisman_work | is that an issue? |
| 22:04.55 | waverly360 | I think the frequency ranges clashe |
| 22:04.57 | BBHoss | great thing about DECT is, you can go to another base station and have your phone work |
| 22:05.12 | BBHoss | so another person could bring their phone into ther others office and talk still |
| 22:05.14 | waverly360 | but I'm not versed on them well enough to know how |
| 22:05.34 | BBHoss | 2.4 ghz sucks realy all-around |
| 22:05.34 | nhuisman_work | clash with wifi? |
| 22:05.59 | BBHoss | between the microwave ovens, industrial traffic, and wifi, it suckzorz |
| 22:06.01 | waverly360 | nhuisman_work: I can't say without possibly sounding like an idiot |
| 22:06.10 | nhuisman_work | what about 5.8ghz? |
| 22:06.11 | waverly360 | but I think that's what BB is saying |
| 22:06.24 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 22:06.34 | *** join/#asterisk ghento (n=ghento@75.155.241.7) |
| 22:06.43 | nhuisman_work | maybe this one? http://www.voipsupply.com/product_info.php?products_id=923 |
| 22:06.51 | BBHoss | yeah it might work, but at one of my locations it was no better than 2.4, not because of interference, but because the radio waves can't penetrate walls well enough |
| 22:06.57 | BBHoss | comes down to physics |
| 22:07.18 | BBHoss | 1.9ghz has a longer wavelength than 24 or 5.8, therfore it penetrates better |
| 22:07.23 | BBHoss | 2.4 |
| 22:07.30 | nhuisman_work | guess can always goto 900mhz again |
| 22:07.34 | BBHoss | yeah |
| 22:07.35 | nhuisman_work | that has lower range though? |
| 22:07.46 | nhuisman_work | why did we ever goto 2.4 and 5.8? |
| 22:08.01 | BBHoss | yes, because of power emmision standards by the fcc |
| 22:08.32 | nhuisman_work | basically I have some people using analog phones and I want to switch them to our voip |
| 22:08.42 | nhuisman_work | so that when I upgrade I don't need additional analog fxos |
| 22:08.46 | BBHoss | im telling you, you want DECT |
| 22:09.05 | nhuisman_work | now question: say I need many dect stations |
| 22:09.07 | nhuisman_work | to cover the building |
| 22:09.10 | BBHoss | saves in the long run |
| 22:09.20 | nhuisman_work | does it have the same really lame problem of channel overlap? |
| 22:09.21 | BBHoss | you need an engineer for that, or someone whos trained |
| 22:09.23 | nhuisman_work | as wifi? |
| 22:09.25 | BBHoss | like me :) |
| 22:09.29 | BBHoss | its automatic |
| 22:09.59 | nhuisman_work | so how many can be near each other and not overlap? |
| 22:10.03 | Nugget | http://www.flickr.com/photos/ehgradman/sets/72157603124164552/ <-- awesomeness |
| 22:10.13 | nhuisman_work | ie wifi has only 3 real distinct channels |
| 22:10.19 | BBHoss | hang on im upping the training right now |
| 22:10.55 | nhuisman_work | how much does an astra dect run per handset? |
| 22:10.58 | BBHoss | you can put them side by side |
| 22:11.15 | nhuisman_work | like 300 something? |
| 22:11.34 | BBHoss | but you can't have more than 8 calls at once per site |
| 22:11.34 | BBHoss | lemme check MY costs |
| 22:11.43 | nhuisman_work | oh wait I was looking at the higher model |
| 22:11.54 | BBHoss | 2gac.net/training.ppt |
| 22:11.57 | BBHoss | have a look |
| 22:12.47 | nhuisman_work | now each handset can be a different number right? |
| 22:12.55 | *** join/#asterisk stimpie (n=stimpie@84-104-5-115.cable.quicknet.nl) |
| 22:13.01 | nhuisman_work | it's not just a multiplication of one number? |
| 22:13.09 | tzanger | by default (i.e. bare bare bare zapata.conf) echocancel=yes and echocancelwhenbridged=no? |
| 22:13.10 | BBHoss | what do you mean a different numbers |
| 22:13.12 | BBHoss | just read the ppt |
| 22:13.25 | [hC] | BBHoss: is this aastra dect? |
| 22:13.29 | BBHoss | yes |
| 22:13.35 | [hC] | its not 8 calls per site, its 8 concurrent calls per antenna |
| 22:13.39 | [hC] | "node" |
| 22:13.40 | BBHoss | yes |
| 22:13.44 | BBHoss | now i remember |
| 22:14.00 | [hC] | you can just toss more antennas up and they will take the 9th, 10th, etc call respectively |
| 22:14.15 | BBHoss | the 142 is $237.59 cost |
| 22:14.43 | [hC] | each dect handset can be independent registrations, yes |
| 22:14.47 | BBHoss | then you have to decide what RFP you want, indoor or outdoor |
| 22:14.54 | nhuisman_work | those run about 1k each? |
| 22:14.57 | BBHoss | it is a GREAT design |
| 22:14.58 | nhuisman_work | indoor obviously |
| 22:14.58 | BBHoss | uhh |
| 22:15.09 | BBHoss | my indoors MSRP is $799.99 |
| 22:15.10 | nhuisman_work | i'm looking at a site now that was listing them at that |
| 22:15.14 | nhuisman_work | I think they are marking them up |
| 22:15.21 | BBHoss | and my cost is $575.99 |
| 22:15.43 | hmmhesays | does chan_sip send a 183 session progress message under any circumstances? |
| 22:16.08 | BBHoss | then the outdoor is $949.99 and 683.99, respectively |
| 22:16.27 | [hC] | hmmhesays: probably easiest to search chan_sip.c for "183" :) |
| 22:18.16 | nhuisman_work | so If I have 6 places to place these |
| 22:18.39 | BBHoss | the range on these it more than 2.4 ghz phones, much more |
| 22:18.54 | nhuisman_work | it looks like there is 5 dect-channels? |
| 22:18.58 | hmmhesays | yeah probably |
| 22:19.01 | *** part/#asterisk galeras (n=Martin@201.244.247.149) |
| 22:19.12 | BBHoss | lemme look |
| 22:19.21 | nhuisman_work | page 15 of your slides |
| 22:20.10 | BBHoss | there are 5 carriers, with 24 slots per carrier |
| 22:20.33 | BBHoss | so for each carrier, you can have 8 channels of voice |
| 22:20.46 | nhuisman_work | what's a slot used for then? |
| 22:20.58 | BBHoss | 8 slots-pairs for voice |
| 22:21.11 | BBHoss | 2 slot-pairs for hand-in from another base station |
| 22:21.18 | nhuisman_work | ah |
| 22:21.19 | BBHoss | then 2 slots for signalling/sync |
| 22:21.22 | *** join/#asterisk b1ch0 (n=ralabiso@static-200-105-209-46.acelerate.net) |
| 22:21.45 | BBHoss | open space these things have 300,000 sq ft coverage |
| 22:22.41 | BBHoss | look on slide 25 |
| 22:22.41 | *** join/#asterisk X-Rob (n=Rob@ppp214-210.static.internode.on.net) |
| 22:22.41 | BBHoss | that will give you how much loss |
| 22:22.47 | slowshutt | what do you guys think of SIPxchange? |
| 22:22.47 | _mm_ | i'm having an issue using chanspy to do multiple spygroups, the documentation says to use : as a delimiter, and i do ( ChanSpy(|g(93:114)) ), but all it does when I spy, is beep repeatedly (and if i do either of those groups alone, it works fine, and there are people in both groups during my testing) I've checked the source of app_chanspy, and it does look for :, asterisk 1.4.11, i'm rather out of ideas, any suggestions? |
| 22:22.50 | nhuisman_work | yeah my building is all separate offices |
| 22:22.56 | nhuisman_work | with all cement walls |
| 22:23.02 | BBHoss | solid cement? |
| 22:23.13 | BBHoss | or cinder block? |
| 22:23.15 | nhuisman_work | not sure if it's solid or cinder |
| 22:23.25 | nhuisman_work | i think solid |
| 22:23.28 | nhuisman_work | in some places |
| 22:23.57 | BBHoss | look at 27 |
| 22:24.01 | BBHoss | also 26 |
| 22:24.03 | nhuisman_work | hmm yeah I don't hink all rooms are cement |
| 22:24.39 | nhuisman_work | suppose I'll buy a couple phones and a transmitter and test it |
| 22:24.43 | JT | the problem isn't so much the concrete |
| 22:24.53 | JT | but more the steel rebar |
| 22:25.09 | JT | also, if it was just cement it would be pretty weak ;) |
| 22:25.13 | Strom_M | welcome to Faraday Cages 101 |
| 22:25.15 | nhuisman_work | obviously :P |
| 22:25.21 | BBHoss | heh |
| 22:25.49 | BBHoss | starting on slide 30, it gives you how much capacity you will need |
| 22:26.13 | nhuisman_work | i'm really looking at a few scattered users probably like 10 at most |
| 22:26.24 | nhuisman_work | who knows though if I tell them they can have wireless handsets |
| 22:26.33 | BBHoss | the guy told me that airbus uses this system in europe with thousands of phones and hundreds of RFPs |
| 22:26.47 | BBHoss | its not the exact system though |
| 22:27.10 | BBHoss | do you have wifi coverage everywhere? |
| 22:27.55 | *** join/#asterisk D-N-A (n=Alam@41.233.138.103) |
| 22:28.13 | BBHoss | that guide will tell you probably all you need to know about setting it up |
| 22:28.20 | BBHoss | installing them is a breeze too |
| 22:28.23 | *** join/#asterisk |dennis| (n=dennis@200.32.233.84) |
| 22:28.30 | D-N-A | hello, any one can guide me how can i configure TDM 01b ? i'm using ubuntu |
| 22:28.31 | JT | are you talking about wifi handsets or dect? |
| 22:28.37 | nhuisman_work | I have wifi but the coverage drops in some rooms |
| 22:28.52 | nhuisman_work | I think there are channel overlap problems |
| 22:28.52 | BBHoss | how many APs do you have |
| 22:28.58 | nhuisman_work | i have three on right now |
| 22:29.03 | nhuisman_work | I used to have 6 |
| 22:29.06 | BBHoss | and that covers everything? |
| 22:29.06 | nhuisman_work | 3 on each floor |
| 22:29.10 | BBHoss | ohh |
| 22:29.19 | nhuisman_work | right now I have 3 with 1watt amps |
| 22:29.22 | BBHoss | how many apprx sq feet are the floors |
| 22:29.25 | BBHoss | oh i see |
| 22:29.33 | D-N-A | any help ? |
| 22:29.34 | BBHoss | well amps can cause lots of problems too! |
| 22:29.51 | nhuisman_work | yeah I was trying to see if that would improve the signal through the walls |
| 22:30.05 | BBHoss | D-N-A: call digium support, they can help you set it up (for free) |
| 22:30.06 | JT | lol, amps on omnis, how irresponsible |
| 22:30.25 | nhuisman_work | it's not open... |
| 22:30.29 | JT | err |
| 22:30.34 | JT | that's completely irrelevant |
| 22:30.37 | BBHoss | i wouldnt be surprised if you could get away with one per floor, maybe two |
| 22:30.37 | nhuisman_work | but i'm sure it's not the greatest idea |
| 22:30.41 | tzanger | fucking weird, man |
| 22:30.42 | *** join/#asterisk PodMan99a (n=keith@82-34-164-205.cable.ubr02.maid.blueyonder.co.uk) |
| 22:30.43 | JT | it's irresonsible from an RF viewpoint |
| 22:30.44 | D-N-A | BBHoss, i already googled , but do you know any docs for that ? |
| 22:30.51 | JT | you are massively raising the noise floor |
| 22:31.07 | PodMan99a | hey all have a problem with asterisk on CE4 app_dial.c: Unable to create channel of type 'IAX2' (cause 3 - No route to destination) |
| 22:31.16 | PodMan99a | anyone able to help .... im such a jackass lol |
| 22:31.17 | tzanger | same tdmoe endpoint, same tdm400 for timing, system A = great audio. take tdm400 to system B (a better system in terms of HW/CPU) and it's got echo... tdm400 isn't actually used for audio, just timing |
| 22:31.19 | b1ch0 | hi everybody, does anybody can help with a codec translation question ? i want to use GSM codec on my phones on my box .... wich codec is used for each pstn call (using tdm card) from IP phone to PSTN ? |
| 22:31.48 | JT | tzanger: tdmoe... dodgy hack ;) |
| 22:31.50 | nhuisman_work | hmm |
| 22:31.58 | gvasterisk | ppl I've problems with php agi in debug says: Usage: php [-f] etc. |
| 22:32.00 | tzanger | JT: how else am I gonna get 1200 zap channels intoa system? :-) |
| 22:32.12 | De_Mon | gvasterisk this isn't #php |
| 22:32.14 | *** join/#asterisk cesar_CR (n=cr@201.195.35.62) |
| 22:32.18 | BBHoss | b1ch0: PSTN uses ulaw/alaw |
| 22:32.22 | JT | tzanger: don't? :) |
| 22:32.23 | PodMan99a | its a simple pure IP network no PTSN lines just VoIP number purchased online |
| 22:32.24 | gvasterisk | I know, but the agi is for asterisk |
| 22:32.30 | nhuisman_work | so even though they are on different channels it is detrimental to have them on amps? |
| 22:32.36 | De_Mon | gvasterisk but the problem is with php |
| 22:32.36 | b1ch0 | ok thanks |
| 22:32.38 | gvasterisk | in agi debut is the problem |
| 22:32.43 | JT | gvasterisk: G.711 |
| 22:32.43 | gvasterisk | debug |
| 22:32.45 | JT | err |
| 22:32.47 | JT | b1ch0: i mean |
| 22:32.48 | nhuisman_work | or is it just like noise pollution |
| 22:32.57 | gvasterisk | an error of the codec? how do I solve that? |
| 22:33.02 | BBHoss | nhuisman_work: yes, as they all have a bit of noise, amps amplify the noise too |
| 22:33.09 | JT | nhuisman_work: yes it creates heaps of noise |
| 22:33.24 | JT | BBHoss: you're missing that the signal IS a noise source |
| 22:33.40 | BBHoss | thats what i meant |
| 22:33.56 | BBHoss | alright im off to dinner, bbl! |
| 22:33.57 | gvasterisk | jt how do I solve that problem? |
| 22:33.58 | JT | b1ch0: if it's an analogue card, it probably uses SLIN internally |
| 22:34.00 | De_Mon | besides amps how else can you boost signal? |
| 22:34.11 | BBHoss | better antennas |
| 22:34.35 | JT | gvasterisk: umm, have you even explained what the problem is? |
| 22:34.41 | nhuisman_work | got any favorite high gain omnis? |
| 22:34.42 | JT | De_Mon: more base stations |
| 22:34.51 | nhuisman_work | yeah but more base stations just interfere with each other |
| 22:34.55 | De_Mon | hehe I had a feeling you were gonna say that |
| 22:34.57 | PodMan99a | any one any good at asterisk *.conf have an issue with iax.conf extensions.conf ... well think thats it ... cant work out this error and google is no use |
| 22:35.03 | nhuisman_work | i was thinking about trying to use wds |
| 22:35.05 | JT | no, not if you set up the base stations right |
| 22:35.11 | nhuisman_work | but that just makes it slow as fuck with more aps |
| 22:35.15 | JT | try taking a lesson from cell network designers |
| 22:35.17 | gvasterisk | jt the problem is the following in the dialplan I call a file.agi that only answer saydigits1234 but, in the debug of the agi I get usage php [-f] blablabla |
| 22:35.27 | JT | design the network like a cell network |
| 22:35.43 | b1ch0 | yes it a 400P, so even if i the best codec to preserve bandwith and CPu consumption ... the PCI card alway use other codec (maybe G711 or SLIN) to bridge the call from IP phones to the PSTN world |
| 22:35.44 | De_Mon | they don't have to be repeaters |
| 22:36.00 | JT | use a wired network to feed all the APs |
| 22:36.00 | *** join/#asterisk agx (n=badpengu@81.174.9.212) |
| 22:36.12 | nhuisman_work | JT it is |
| 22:36.13 | PodMan99a | any geniuses free? |
| 22:36.27 | De_Mon | PodMan99a all the ones in here charge a fortune |
| 22:36.28 | JT | b1ch0: are the ip phones on your lan? |
| 22:36.30 | nhuisman_work | we have an L shaped building |
| 22:36.34 | nhuisman_work | with two floors |
| 22:36.35 | gvasterisk | jt any idea? |
| 22:36.49 | nhuisman_work | I have APS at the ends and in the middle of the L |
| 22:36.52 | nhuisman_work | on each floor |
| 22:36.52 | JT | gvasterisk: well obviously the agi doesn't work |
| 22:37.17 | JT | you REDUCE the output power of each AP so it's just enough to cover its area |
| 22:37.20 | gvasterisk | I know but it has the lines OK but I get that error :S |
| 22:37.28 | JT | and the adjoining area is on a different channel |
| 22:37.30 | PodMan99a | suppose it depends on what they can help me with ... lol i know im missing a small bit of this ... but its my 1st and internet is not helping so trying the IRC's as you all nicer |
| 22:37.56 | nhuisman_work | and always have one channel of buffere between? |
| 22:38.13 | nhuisman_work | so overlap is on a different channel, and the similar channel is another AP away |
| 22:38.21 | JT | nhuisman_work: at least 1 channel, yes |
| 22:38.41 | JT | cell networks are designed around theoretical hexagons |
| 22:38.55 | nhuisman_work | well how narrow do you go with the channels? there are three pretty distinct but you can goto 4 or 6 |
| 22:39.49 | JT | the distinct ones i guess |
| 22:39.54 | PodMan99a | [Nov 14 22:04:59] WARNING[28482] app_dial.c: Unable to create channel of type 'IAX2' (cause 3 - No route to destination) |
| 22:39.54 | PodMan99a | [Nov 14 22:04:59] NOTICE[28482] cdr.c: CDR on channel 'SIP/keith-09078b80' not posted |
| 22:40.12 | JT | PodMan99a: as it says... no route to destination |
| 22:40.13 | PodMan99a | is what i have in error log but no sense to me ... i can change the error but not fix it |
| 22:40.24 | JT | the error is simple |
| 22:40.28 | JT | exactly what it says |
| 22:40.29 | PodMan99a | yes... but firewall is open and usernames and passwords are corretn |
| 22:40.40 | PodMan99a | i tried the basics and everything is set |
| 22:40.44 | JT | it can't access whatever remote host it is trying to access |
| 22:42.10 | PodMan99a | ummmmm..... ill call the F***rs ... they should be able to change something... it makes sense what you say but surely this is standard stuff for online voip providers |
| 22:42.31 | JT | can you ping the host |
| 22:42.35 | JT | from your asterisk box? |
| 22:42.56 | PodMan99a | ping no... firewall blocks icmp |
| 22:43.04 | PodMan99a | but yes can communicate with world |
| 22:43.04 | *** join/#asterisk KryoStoffer (n=kri@helium.kri.dk) [NETSPLIT VICTIM] |
| 22:43.04 | *** join/#asterisk CtRiX (n=CtRiX@aretha.navynet.it) [NETSPLIT VICTIM] |
| 22:43.15 | JT | what a useless firewall |
| 22:43.28 | PodMan99a | is this not a good thing? |
| 22:43.37 | JT | blocking icmp is stupid as hell, so no |
| 22:43.48 | PodMan99a | y is this just out of interest |
| 22:44.08 | JT | Internet CONTROL Message Protocol |
| 22:44.20 | JT | CONTROL messages need to get through |
| 22:44.59 | PodMan99a | arnt ICMP used for ddos attacks though? |
| 22:45.22 | jameswf | netgear + sip no workie.... |
| 22:45.27 | _x86_ | anything could be used for DoS attacks |
| 22:45.34 | JT | and there is almost NO reason to block OUTBOUND ICMP, EVER unless you really don't trust people in your own network |
| 22:45.42 | JT | all protocols are used for DDoS attacks |
| 22:45.58 | _x86_ | to be immune to DoS attacks, there is a very easy solution that you can do.... unplug your computer from the network |
| 22:45.59 | PodMan99a | true.... outbound blocking bad.... so inbound blocking is ok... |
| 22:46.07 | JT | not as stupid |
| 22:46.10 | JT | but still stupid |
| 22:46.20 | _x86_ | PodMan99a: ICMP ping inbound is generally acceptable to block |
| 22:46.23 | PodMan99a | its webhost so nothing complicated ... smtp www pop3 |
| 22:46.53 | _x86_ | PodMan99a: but you should not block ALL icmp inbound, or else things like ICMP redirects, source quench, destination unreachable, etc wont get through... which is Very Bad(tm) |
| 22:47.11 | PodMan99a | lemme see whats blocked |
| 22:47.31 | JT | _x86_: not a great idea though |
| 22:47.37 | JT | to block icmp |
| 22:47.41 | JT | rate limit perhaps |
| 22:48.01 | _x86_ | JT: icmp ping inbound is ok to block |
| 22:48.14 | _x86_ | JT: not a horrible idea at all |
| 22:48.15 | JT | ok as in still not a very good idea |
| 22:48.36 | _x86_ | it's not a bad thing at all |
| 22:48.45 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 22:48.46 | JT | justify yourself |
| 22:48.49 | *** join/#asterisk asdx (n=diego@adsl-130-56.click.com.py) |
| 22:48.51 | asdx | hi |
| 22:49.03 | JT | blocking a useful packet is good how? |
| 22:49.08 | asdx | how much should i charge someone for installing a pbx |
| 22:49.16 | _x86_ | well if you dont want people doing mass ping sweeps and getting somewhat usable information, it makes sense to block inbound icmp pings |
| 22:49.16 | PodMan99a | all IP traffic but www smtp pop3 is blocked on that part of network... thats inbound... outbound specifies icmp.... lets try removing that |
| 22:49.17 | asdx | is $500 usd a good price? |
| 22:49.28 | _x86_ | JT: justify how it's useful? |
| 22:49.46 | _x86_ | asdx: perhaps in iraqistan ;) |
| 22:50.14 | JT | _x86_: it lets you easily know if a host is up, and the latency |
| 22:50.17 | asdx | _x86_: how much do you charge? |
| 22:50.28 | JT | _x86_: there is no logical reason to blind block it for most hosts |
| 22:50.44 | PodMan99a | JT: still same error with ICMP allowed |
| 22:50.50 | _x86_ | JT: that's the EXACT reason that i blindly block it on my outside-facing hosts |
| 22:51.04 | JT | _x86_: err, if you're worried about your host been discovered, there's 539839359735979753 other ways to detect them |
| 22:51.07 | _x86_ | JT: so people CANT see which IP's are up |
| 22:51.15 | _x86_ | JT: sure... |
| 22:51.26 | _x86_ | JT: but the Most Common(tm) way is a ping scan |
| 22:51.31 | JT | _x86_: that's the thing about using retarded methods to "secure" your network, they're still retarded |
| 22:51.44 | JT | nmaps are much more useful to an attacker anyway |
| 22:52.05 | _x86_ | i'm not "securing" anything... dunno where you got that idea... just trying not to publically advertise "HERE I AM" |
| 22:52.17 | JT | then firewall all ports |
| 22:53.40 | JT | seriously, if your network is so vulnerable that whether a host is detectable by ping or not will make a noticable security difference, you have serious problems with your network |
| 22:54.23 | PodMan99a | JT: we use ignorance is bliss... if we block ICMP then we dont need to worry about security holes.... block anything unless its required |
| 22:54.53 | PodMan99a | network management by sleeping |
| 22:54.58 | JT | PodMan99a: that's bad security policy |
| 22:55.36 | PodMan99a | but cheap.... lol .... lots of IPTABLES and Cisco programming to do something that 5 lines of code will solve |
| 22:56.03 | JT | eh? |
| 22:56.52 | PodMan99a | JT: the policy of block everything unless used ... i.e. only allow http smtp pop3 in/out its cheaper and easier to implement.... and gauranteed |
| 22:57.20 | PodMan99a | yes it makes my job redundant once setup however i can get on with more important things once its working |
| 22:57.48 | brainspiral | Anybody want a pile of Cisco and Linksys phones? I hate them. |
| 22:58.11 | Qwell | brainspiral: I'll give you a buck |
| 22:58.18 | PodMan99a | + reading up on what to do as an alternative is a mission... there are 1000's of lines i can put in to my 515E to do the same as 5 lines |
| 22:58.26 | PodMan99a | simple security policy is easy |
| 22:58.43 | brainspiral | A buck a piece & it's a deal. |
| 22:59.02 | Qwell | brainspiral: alright |
| 22:59.19 | Qwell | assuming by "pile" you don't mean "pile of rubble" |
| 22:59.34 | [TK]D-Fender | brainspiral, count me in :) |
| 23:00.03 | PodMan99a | suppose ill have to go on firewall irc's to compile new plan.... but for now it works |
| 23:00.49 | *** part/#asterisk blq (i=me@1.1.1.vg) |
| 23:01.08 | brainspiral | D-Fender & Qwell: What your phones of choice? |
| 23:01.11 | JT | PodMan99a: this makes no sense |
| 23:01.14 | Qwell | ~phones |
| 23:01.15 | jbot | [phones] 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 like such as |
| 23:01.16 | JT | PodMan99a: your security policy |
| 23:01.37 | PodMan99a | JT: yea?? ... block everything less its needed |
| 23:01.52 | JT | PodMan99a: the only way it works is if you block absolutely everything |
| 23:02.11 | JT | PodMan99a: blocking everything except a couple and "forgetting about it" |
| 23:02.28 | PodMan99a | JT: if i understand the question correctly then yes |
| 23:02.37 | JT | what question |
| 23:02.39 | PodMan99a | JT: the hole does not exist and there is no way to break it |
| 23:02.44 | JT | it was a statement |
| 23:02.53 | PodMan99a | close enough |
| 23:03.00 | JT | if you open ports for services |
| 23:03.10 | JT | and those services are vulnerable, it's all for nothing |
| 23:03.21 | JT | firewalls are simply false senses of security |
| 23:03.34 | PodMan99a | yes... btu 2 things... all software is patched regulary |
| 23:03.42 | PodMan99a | and kernels are up2date |
| 23:03.59 | PodMan99a | but then firewall blocks all else |
| 23:04.15 | PodMan99a | anything thats running is open on firewall but then there is not much running in 1st place |
| 23:04.50 | PodMan99a | i see your point but basic security is still taken care of......... i.e. |
| 23:05.00 | PodMan99a | mod_security and decent rules on apache |
| 23:05.24 | PodMan99a | smtp configured to allow from only authenticated senders normal stuff |
| 23:05.30 | PodMan99a | or am i missing something obvious here |
| 23:06.11 | brainspiral | Thanks, jbot. I've used the Aastra 480i CT with great success. Might stick with those. |
| 23:06.31 | JT | what is the advantage of firewall then? |
| 23:06.42 | JT | brainspiral: polycom are the best, and jbot is a bot |
| 23:06.55 | [TK]D-Fender | brainspiral, PM |
| 23:07.14 | PodMan99a | stopping FU**RS from pinging us to death and the occasional FTP user creating bad CGI etc code banging out traffic to the net |
| 23:07.41 | JT | PodMan99a: yes, buffer overflows that get an attacker root access, they don't care about if your smtp is authenticate or not |
| 23:07.43 | PodMan99a | problem with securing apache is you can secure it too tight.... has to be open enough for them to use |
| 23:08.06 | PodMan99a | hence updates to software regularly |
| 23:08.44 | PodMan99a | although that is not perfect ... it works |
| 23:08.51 | JT | PodMan99a: sorry, and why can't you rate limit ICMP instead of block? |
| 23:09.05 | PodMan99a | my cisco aint that good... lol still learning.... BRB |
| 23:09.28 | JT | PodMan99a: then just open icmp |
| 23:10.23 | asdx | is there a web site or something to see what are the prices for pbx setups? |
| 23:10.38 | asdx | so i can have an idea in how much to charge... |
| 23:13.53 | JT | asdx: how long is a piece of string? |
| 23:14.32 | *** part/#asterisk zerohalo (n=zeroHalo@pool-71-255-162-167.bstnma.east.verizon.net) |
| 23:14.34 | JT | PodMan99a: the reality is your firewall isn't providing any real security |
| 23:15.09 | asdx | JT: it depends? |
| 23:15.31 | PodMan99a | JT: saving money ... level 3 aint cheap.... some idiot pings one of our servers then we stuffed... although i do take your point about ICMP limit |
| 23:15.35 | JT | asdx: right |
| 23:15.42 | PodMan99a | jt: now getting No authority found |
| 23:15.48 | JT | PodMan99a: how are you stuffed if someone pings? |
| 23:15.57 | asdx | JT: so i should ask this guy what they want to do exactly... and base my price on all those factors? |
| 23:15.57 | JT | PodMan99a: so the host is reachable now |
| 23:16.04 | PodMan99a | JT: large ping packets |
| 23:16.07 | JT | asdx: well that's obvious |
| 23:16.18 | asdx | JT: i'm not good with business |
| 23:16.39 | asdx | JT: i'll call this guy |
| 23:16.40 | JT | asdx: you don't charge the same for 3 handset in a house as 10000 handsets across 100 offices of a national company |
| 23:16.47 | JT | asdx: well it's logic ;) |
| 23:17.01 | JT | you need to scope out the project |
| 23:17.03 | PodMan99a | jt: ok host now reachable.... no authority found meaning im not allowed to make calls out on this connection |
| 23:17.05 | JT | number of users |
| 23:17.08 | JT | servers |
| 23:17.11 | JT | features |
| 23:17.25 | JT | PodMan99a: and these large ping packets will do what? |
| 23:17.42 | PodMan99a | JT: no authority found and not posted errors |
| 23:18.01 | JT | PodMan99a: sip or iax2? |
| 23:18.21 | PodMan99a | large ping packets cost bandwith .... 10240 packets being sent also same being replied however limiting should solve that |
| 23:18.34 | PodMan99a | IAX2 |
| 23:18.59 | JT | PodMan99a: you still receive and pay for them, anyway |
| 23:19.14 | JT | PodMan99a: if you have DDoS problems, it has to be dealt with upstream from you |
| 23:19.33 | PodMan99a | no we recieve the request and reject it so we dont send back |
| 23:19.38 | PodMan99a | 50% |
| 23:19.42 | JT | (and you should probably also stop running irc and gaming servers ;) ) |
| 23:19.58 | PodMan99a | game servers rule |
| 23:20.08 | JT | PodMan99a: sure, but the problem is still there anyway |
| 23:20.12 | *** join/#asterisk stubert (n=stu@techtools.actusa.net) |
| 23:20.15 | JT | PodMan99a: if you want to attract kiddies they do |
| 23:20.36 | PodMan99a | most of our players are 30+ ... its a sad area... lol |
| 23:20.46 | PodMan99a | and the cannon fodder 2 lol |
| 23:20.57 | PodMan99a | been researching have we JT lol.... |
| 23:21.09 | PodMan99a | JT: IAX2 is not posting |
| 23:21.10 | JT | kiddie is a state of mind |
| 23:21.17 | JT | PodMan99a: nresearch? |
| 23:21.17 | PodMan99a | true... im one all through |
| 23:21.27 | PodMan99a | knowing we run game servers |
| 23:21.35 | JT | how would i know that |
| 23:21.44 | JT | aside from the fact that you get DDoSes |
| 23:21.48 | *** join/#asterisk ghento (n=ghento@75.155.241.7) |
| 23:21.49 | stubert | Having problems with NexTone and dropped calls |
| 23:21.50 | JT | that's why i said irc and gaming |
| 23:21.53 | PodMan99a | true.... domain not shown.... |
| 23:21.57 | JT | they are the main causes of DDoSes |
| 23:22.20 | JT | hence why a lot of professional datacentres prohibit those services |
| 23:22.49 | PodMan99a | all they want over here is bandwith ... how they get it they not that bothered,... lol |
| 23:22.50 | stubert | NexTone is claiming asterisk is not returning the right CSeq number on the ACK causing it to drop the call |
| 23:23.05 | stubert | Any help would be great... |
| 23:23.32 | PodMan99a | why would IAX2 not be posting? |
| 23:23.50 | JT | posting? |
| 23:24.08 | PodMan99a | [Nov 14 23:23:40] WARNING[4554]: chan_iax2.c:7450 socket_process: Call rejected by 217.14.138.49: No authority found |
| 23:24.08 | PodMan99a | [Nov 14 23:23:40] NOTICE[5696]: cdr.c:434 ast_cdr_free: CDR on channel 'IAX2/voiptalk-1' not posted |
| 23:24.14 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 23:24.36 | JT | ignore the cdr line |
| 23:24.40 | JT | it is a byproduct |
| 23:24.48 | Corydon76-dig | Because the call was not answered |
| 23:24.48 | JT | of having no authority |
| 23:24.51 | JT | and therefore no call |
| 23:24.53 | PodMan99a | k .... |
| 23:25.02 | JT | it's only a NOTICE |
| 23:25.04 | PodMan99a | so authentication with the above ip is required |
| 23:25.11 | JT | a WARNING or ERROR is more important |
| 23:25.13 | JT | PodMan99a: yes |
| 23:25.33 | asdx | JT: i just called this guy and they want to build their business. |
| 23:25.36 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) |
| 23:25.51 | asdx | JT: they want to resell accounts so people can call anywhere |
| 23:26.05 | JT | asdx: so it's not installing PBXes? |
| 23:26.07 | asdx | JT: through a voip provider |
| 23:26.32 | asdx | JT: i think they want only one pbx |
| 23:26.53 | asdx | JT: i'm not sure yet because i'm a noob, but i want to use this experience to learn |
| 23:26.59 | JT | asdx: they want to be a service provider? |
| 23:27.06 | asdx | JT: yes |
| 23:27.57 | PodMan99a | i have an outbound proxy address.... how do i define that |
| 23:28.11 | JT | PodMan99a: eh? |
| 23:28.24 | JT | asdx: do they have datacentre space? |
| 23:28.39 | PodMan99a | type=peer |
| 23:28.39 | PodMan99a | username=HIDDEN |
| 23:28.39 | PodMan99a | secret=HIDDEN |
| 23:28.39 | PodMan99a | host=HOST |
| 23:28.54 | asdx | JT: they bought a dedicated server already |
| 23:29.01 | PodMan99a | thats my block for [voiptalk] |
| 23:29.02 | JT | PodMan99a: type=friend |
| 23:29.07 | PodMan99a | ah |
| 23:29.31 | *** part/#asterisk Seldon75 (n=chatzill@69.77.161.3) |
| 23:29.36 | asdx | JT: but they want to start small |
| 23:29.42 | JT | asdx: sure |
| 23:29.59 | JT | asterisk is a bit of a pain for virtual PBXes |
| 23:30.02 | PodMan99a | JT: still doing same error |
| 23:30.03 | rob0 | In case anyone saw my ipkall issue earlier: it was a PEBKAC. Hostname of the proxy was a homonym of the actual hostname ... "cc.domain" vs. "cici.domain". So that's why they weren't hitting my firewall ... couldn't resolve the hostname! duh. |
| 23:30.15 | JT | as you can't specify different domains in the one instance |
| 23:30.22 | rob0 | I looked at it and cc looked right to me. |
| 23:30.38 | PodMan99a | JT: suppose ssh access is out of question... lol |
| 23:30.42 | JT | PodMan99a: check your authentication details are correct |
| 23:30.48 | PodMan99a | 20 times |
| 23:31.26 | JT | i don't really want to or have time |
| 23:31.32 | JT | iax2 auth is funky |
| 23:31.39 | JT | as in annoying and really anal |
| 23:31.48 | PodMan99a | worth a try lol |
| 23:31.57 | Corydon76-dig | Um, and SIP auth isn't? |
| 23:32.17 | PodMan99a | Corydon76-dig i dont SIP with works ... lol |
| 23:32.21 | PodMan99a | done* |
| 23:32.33 | Corydon76-dig | authentication is anal, period... |
| 23:32.56 | PodMan99a | true |
| 23:33.17 | *** join/#asterisk ghento (n=ghento@75.155.241.7) |
| 23:33.54 | JT | Corydon76-dig: yes SIP is much easier to setup than IAX2, especially auth wise |
| 23:34.15 | JT | iax2 gets all emo about the name of your peer/friend entry and contexts, etc ;) |
| 23:35.28 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 23:35.41 | puzzled | hi |
| 23:37.26 | *** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com) |
| 23:44.07 | PodMan99a | JT: may be similar however unable to recieve in as well for some reason... same errors |
| 23:44.14 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 23:44.55 | PodMan99a | ah!! that may do it ... error other side i think... have no authority both inbound and outbound.... |
| 23:45.08 | JT | ok |
| 23:45.27 | PodMan99a | although if i dial my number from a land line they are talking to me...?? |
| 23:45.43 | *** join/#asterisk coppice (n=chatzill@102.204.17.210.dyn.pacific.net.hk) |
| 23:45.45 | JT | what? |
| 23:46.07 | PodMan99a | well they have to talk to my server so it knows im recieving a call |
| 23:47.26 | b1ch0 | JT: sorry, back again .... yes i will have aroud 50 phones in the same LAN and a few ones in remote branch (going to use IAX to avoid nat problem) |
| 23:48.47 | b1ch0 | and im planning to use G711 in the headquarter, and 723, GSM or other for remote branch phones (to avoid bandwith issues) |
| 23:50.01 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 23:50.04 | generalhan | hey all ! |
| 23:51.10 | generalhan | anyone know of a way to pass variables from one * server to another ? i have a server connecting to my server via IAX2 and i want to be able to set a variable on their end and call it on mine ! |
| 23:51.50 | JT | b1ch0: forget G.723 |
| 23:52.05 | fujin | generalhan: use SQL |
| 23:52.38 | JT | b1ch0: SIP works fine behind most NAT devices |
| 23:54.05 | b1ch0 | JT: yes i know but until now i coudnt meka any test with transversal nat, so if i got troubles , i always got IAX |
| 23:54.41 | asdx | JT: SIP works fine behind NAT? |
| 23:54.47 | generalhan | fujin: explain please |
| 23:54.54 | [TK]D-Fender | b1ch0, if you have bandwidth concerns you'll definitely be wanting to use IAX for trunking |
| 23:54.58 | fujin | USE SQL? ;} |
| 23:55.34 | generalhan | fujin: you have a doc somewhere that might be a bit more in depth about exactly how to "use sql" |
| 23:56.23 | *** join/#asterisk ghento (n=ghento@75.155.241.7) |
| 23:57.16 | asdx | JT: i've tried to setup * with SIP here, but the client i tried X-Lite detected the NAT in my router and couldnt connect to * |
| 23:57.25 | b1ch0 | well, not exactly .... i have IP phnoes that work with SIP and IAX either, so was planning to enable IAX on all external branch and SIP on all internal phones |
| 23:57.28 | asdx | JT: then i did ifconfig eth0 down and connected well, locally. |
| 23:57.40 | TJNII | Bleh, it seems I've picked up a dud fax machine..... |
| 23:58.20 | alrs | ~book |
| 23:58.21 | 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 |
| 23:58.29 | b1ch0 | ... my original question was about what append if i havve all phone using GSM (or other codec) and they need to communicate to pstn world (using a TDM400P card) |
| 23:59.05 | b1ch0 | wich codec it is used on * to transcode the call from IP phones to pstn lines ? |
