| 00:00.10 | a1fa | i may have to add a small pbx |
| 00:02.07 | [TK]D-Fender | a1fa: Could have sworn I told you that... |
| 00:02.15 | [TK]D-Fender | a1fa: Oh wait... I DID |
| 00:02.34 | coreyf1513 | is it harmful to run StopMusicOnHold if it's already stopped, or StartMusicOnHold if it's already Started? |
| 00:02.59 | [TK]D-Fender | daog: It would help if you actually showed the configs we need to look at.... |
| 00:04.53 | a1fa | [TK]D-Fender : i know.. i know.. i could've got away with it on teliax, but with this other provider I am going to need to be more creative |
| 00:06.08 | a1fa | [TK]D-Fender : what's your opinion on freeswitch? |
| 00:07.45 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 00:08.13 | daog | i put the config file for dahdi http://pastebin.com/r1vF0MbA please let me know if you need other more |
| 00:08.23 | [TK]D-Fender | a1fa: Its.. there.... |
| 00:09.25 | [TK]D-Fender | daog: add "busydetect=yes". Restart *. Retest |
| 00:12.12 | daog | ok let me do |
| 00:12.20 | Draiven | hi, I am using a ManagerConnection and OriginateAction for make a call and the class can connect successful with the asterisk server, but the originateResponse.getResponse() is equal to 'ERROR'. How I can show what is the error? in the messages logs only show me == Manager 'asterisk' logged on from 192.168.0.223 and == Manager 'asterisk' logged off from 192.168.0.223 |
| 00:12.32 | Draiven | it is the class |
| 00:12.35 | Draiven | http://pastebin.com/XfRwiqgd |
| 00:14.11 | a1fa | [TK]D-Fender : now if I can find a compact * box |
| 00:14.12 | a1fa | :) |
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| 00:15.30 | Draiven | the result is originateResponse.getResponse(): Error |
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| 00:17.13 | Draiven | somebody can help me, please? |
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| 00:18.49 | miamiseb | a1fa, wrt54g , although it'd be wierd using that for production |
| 00:20.07 | pabelanger-lap | Draiven: #asterisk-java |
| 00:20.58 | [TK]D-Fender | a1fa: WRT would do for basical forwarding. |
| 00:21.02 | a1fa | miamiseb : how about something supported |
| 00:21.07 | daog | don't work log http://pastebin.com/mEFjaX5E |
| 00:21.09 | a1fa | SwitchWox is a overkill |
| 00:21.16 | a1fa | and its sex-pensive |
| 00:21.18 | WIMPy | miamiseb, a1fa: Or the D-Link Horst-Box which natively runs Asterisk? (although a older version). |
| 00:21.25 | a1fa | err. switchvoc |
| 00:21.55 | a1fa | WIMPy : where do you buy that |
| 00:22.18 | Draiven | pabelanger, ok, thx |
| 00:22.31 | miamiseb | http://limeylinux.org/ - A Linux distribution tailored to run Asterisk on VIA Mini-ITX boards and which is small enough to fit on a 128MB or 512MB, or 1GB compact flash card and 512MB of RAM. |
| 00:22.37 | WIMPy | a1fa: Some online store of your choice? |
| 00:22.48 | a1fa | whats the model name? |
| 00:23.06 | WIMPy | Ugm. let me see. |
| 00:23.11 | a1fa | does it come with FXS ports? |
| 00:23.20 | Draiven | pabelanger, #asterisk-java is empty |
| 00:23.39 | miamiseb | DVA-G3342SB |
| 00:23.44 | WIMPy | DVA-G3342SB |
| 00:23.57 | a1fa | not for sale |
| 00:24.16 | a1fa | in the US |
| 00:24.17 | miamiseb | built it. |
| 00:24.23 | miamiseb | s/built/build/ |
| 00:25.07 | daog | tk : don't work test call her http://pastebin.com/mEFjaX5E |
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| 00:28.57 | a1fa | [TK]D-Fender : so it is not possible to configure SPA8000 1 SIP -> multiple FXS? |
| 00:29.45 | [TK]D-Fender | a1fa: unsure but doubtful |
| 00:30.01 | a1fa | too bad.. wasted hardware :( |
| 00:30.20 | a1fa | is a good looking appliance.. i am going to have my cisco rep bring me one to test |
| 00:30.36 | [TK]D-Fender | a1fa: Doesn't guarantee a "no". Go download the manual or something |
| 00:31.41 | a1fa | wgets |
| 00:31.52 | a1fa | http://support.globalink.us/download/cisco/spa8000_quick.pdf |
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| 00:32.47 | miamiseb | you should be able to setup the same sip account on multiple lines, and just have ONE of them register |
| 00:33.15 | miamiseb | if your looking to deliver calls in a ring group type setup where it hunts linearly, it's unlikely you'll get what you want with the SPA alone. |
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| 00:34.44 | a1fa | ok |
| 00:34.54 | a1fa | but what if someone else picks up.. does it go to a different channel? |
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| 00:35.23 | miamiseb | that would be a hunt group. No. |
| 00:35.48 | miamiseb | you'd need a pbx to do that, at least going from the spa's I've worked with (spa2102) |
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| 00:36.37 | miamiseb | I only had two ports, but had the same issue, both could dial out, but only would deliver to one. You could always try it and see though, as that'll give you a nice definite anwser. |
| 00:37.29 | a1fa | alright |
| 00:37.30 | a1fa | thanks |
| 00:40.32 | miamiseb | np |
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| 00:42.22 | coil_ | is there anyway to make my google voice account work with asterisk |
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| 00:48.57 | nix8n82 | probably if your a good enough hacker. |
| 00:49.39 | russellb | if you have a gizmo account, then yeah ... |
| 00:50.01 | russellb | of course, you have to define "work with" |
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| 00:51.00 | miamiseb | Goognight all. |
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| 00:55.31 | coil_ | russellb, well, make calls with it through * |
| 00:58.08 | *** join/#asterisk devdvd (~myemail@173-31-171-48.client.mchsi.com) |
| 00:58.17 | devdvd | hi all. |
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| 01:04.01 | devdvd | ive got a system setup where calls come in and just get forwarded to an external number. I want to allow those external numbers to be able to transfer calls between each other. But what im finding is each time they try to transfer a call it goes into the default context and wont let them complete the transfer cuz it says 91 (9 is the number to dial outside) is not a valid extension. Now, the quick and dirty solution is obviously to incl |
| 01:04.43 | devdvd | so my question to you all is..is there a way to detect that a call coming in off the trunk channel is a transferred call |
| 01:04.53 | devdvd | so it gets dumped into a different context |
| 01:05.02 | devdvd | or can you all offer up a better way to do what im trying |
| 01:11.23 | a1fa | [TK]D-Fender : how about trixbox with asterisk on top of it :P |
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| 01:17.10 | jsgoecke | hola |
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| 02:25.17 | ChannelZ | devdvd: you might look at the channel var TRANSFER_CONTEXT |
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| 02:25.32 | ChannelZ | not sure if it does what you want it to |
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| 03:29.52 | sudhir492 | Hi all |
| 03:30.29 | sudhir492 | When I set forwarding on the cisco phone, the caller id of the original caller is not passed in the forwarded call |
| 03:30.56 | sudhir492 | Is there a way to set that so that forwarded call has original caller's id? |
| 03:31.24 | WIMPy | Forward it on the pbx instead of the phone. |
| 03:31.59 | sudhir492 | I know, but some of the users are so used to doing that on the phone, that I cannot help it. |
| 03:32.39 | sudhir492 | Forwarding on the PBX has not problem at all |
| 03:33.14 | WIMPy | Maybe you can tell the phone to tell the pbx? |
| 03:33.40 | sudhir492 | The question is HOW? |
| 03:34.24 | WIMPy | only knows the other cisco ones, the sipura shit. |
| 03:34.47 | sudhir492 | how do you do that in Sipura? |
| 03:35.43 | WIMPy | I'm not sure, as I don't really use it for much mor than catching dust, but I think it had a setting to send codes on forwarding requests. |
| 03:37.19 | WIMPy | Or maybe you could do a browser interface and put that on a button? |
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| 03:45.48 | sudhir492 | WIPMy, thanks for your support. I figured out how to do that |
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| 03:50.29 | photographe | hi, ne1 here know how to setup SLA with linksys spa942 and asterisk |
| 03:57.04 | devdvd | hey, trying to get video working over a sip channel. here is my sip.conf, output from the cli and the macro-extensions context that i use to dial http://pastebin.com/ZZxXw3as |
| 03:57.36 | devdvd | I am using x-lite 4 beta phones (xlite3 crashes on x64 win7) |
| 03:57.46 | devdvd | what happens is i click start video |
| 03:57.52 | devdvd | but none comes through on the other end |
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| 03:59.11 | devdvd | any thoughts? |
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| 04:12.19 | knot | How can I get google voice working with Asterisk? I can't find the SIP details |
| 04:14.17 | WIMPy | photographe: Maybe you should take a look at issue #11688 |
| 04:16.59 | jsgoecke | knot you need to use SIPPPhone/Gizmo |
| 04:17.04 | jsgoecke | Problem is, if you don't have it already |
| 04:17.10 | jsgoecke | sol |
| 04:17.20 | jsgoecke | As Google closed the account sign-up when they acquired Gizmo |
| 04:17.28 | knot | There is no oppurtunity for success at all? |
| 04:17.30 | jsgoecke | http://www.voip-info.org/tiki-index.php?page=Asterisk+settings+Gizmo |
| 04:17.42 | jsgoecke | Only one way in and out of GV with SIP, Gizmo/SipPhone |
| 04:17.47 | jsgoecke | Otherwise, it is PSTN |
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| 04:24.00 | photographe | WIMPy were i check issue 11688 |
| 04:26.06 | photographe | thanks WIMPy i will try that |
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| 05:49.23 | iamthelostboy | hello, i am trying to have my dialplan enter a queue just once, then on timeout, exit, though i cant make it do it.. |
| 05:50.03 | iamthelostboy | i have timeout set in queue.conf, as well as using Queue(test,n,,,8) and it just sits in the queue |
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| 06:10.54 | p3nguin | iamthelostboy: I'm not seeing any evidence. |
| 06:11.41 | p3nguin | I don't even us the n option and mine exits after the timeout. |
| 06:11.50 | p3nguin | s/us/use/ |
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| 06:16.15 | p3nguin | And now I'm leaving, so it doesn't really matter anyway. |
| 06:17.13 | iamthelostboy | hmm.. seems i had too many options in my configuration |
| 06:17.20 | iamthelostboy | removed them and it worked.. thanks.. |
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| 07:04.45 | azizLIGHTS | not sure if this the place to ask you. can anyone recommend sip service that i can use to receive inbound calls for free (i can hook upto ipkall if no # is provided) not interested in outbound (as that charges money) |
| 07:20.47 | [TK]D-Fender | IPKALL <- |
| 07:34.39 | azizLIGHTS | yes ipkall |
| 07:35.07 | azizLIGHTS | i know of it, but im looking for sip service :) |
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| 07:40.28 | Godfather_ | hi |
| 07:42.04 | [TK]D-Fender | azizLIGHTS: Thats what they are |
| 07:42.34 | [TK]D-Fender | azizLIGHTS: IPKALL provides a free DID |
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| 07:43.30 | azizLIGHTS | yes but i need a sip provider to give to ipkall you see |
| 07:43.51 | [TK]D-Fender | azizLIGHTS: No, you don't |
| 07:44.43 | azizLIGHTS | well im not familiar with all everything sip related. so let me explain what im doing :) |
| 07:44.48 | azizLIGHTS | maybe then you can recommend what i do? |
| 07:44.58 | [TK]D-Fender | maybe... |
| 07:45.38 | azizLIGHTS | i use the sip client on my nokia phone over wifi |
| 07:45.49 | azizLIGHTS | it asks me for username/pass for a sip service |
| 07:46.14 | azizLIGHTS | so what i have now i have voipbuster (i paid $), and connected that to ipkall to receive incoming calls |
| 07:46.18 | *** join/#asterisk raj-darkmystery (~suraj@203.212.24.28) |
| 07:46.44 | azizLIGHTS | basicall im looking for alternates to voipbuster :) |
| 07:46.59 | raj-darkmystery | hi friends... getting an error in debug mode.. dont know how to troubleshoot this.. please help me with this "Received SIP subscribe for peer without mailbox" |
| 07:47.24 | raj-darkmystery | i know i am supposed to do something in voicemail.conf but not sure what to do |
| 07:48.09 | [TK]D-Fender | azizLIGHTS: IPKALL offers a DID. Point it to your server. THE END |
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| 07:48.30 | azizLIGHTS | ok thanks :) |
| 07:48.34 | [TK]D-Fender | raj-darkmystery: Create a mailbox, and specify it in your SIP peer |
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| 07:49.07 | raj-darkmystery | [TK]D-Fender, thats what I am asking.. how i can create a mailbox? |
| 07:49.24 | raj-darkmystery | [TK]D-Fender, i have specified that peer in sip.conf |
| 07:49.51 | *** join/#asterisk athom (~ho@ip-70-160.dobrich.net) |
| 07:50.02 | raj-darkmystery | [TK]D-Fender, user us also able to make calls but what about this error.. 'm not sure what exactly to do :( |
| 07:50.05 | [TK]D-Fender | rajgo read the voicemail.conf sample config. Its 1 LINE |
| 07:50.16 | [TK]D-Fender | raj-darkmystery: go read the voicemail.conf sample config. Its 1 LINE |
| 07:50.18 | athom | Please help: I install AsteriskNOW 1.7, updated FreePBX to 2.8 to get started but I had an error with MOH module and uninstalled it, it's still on my system but in "Module Admin" shows me "Not Installed (Locally available)". Now when I try to install it I get error "Cannot write to file". Maybe the solution is to remove it from my system and then download the missing packages from "Modules Admin"? |
| 07:51.19 | ChannelZ | see #freepbx |
| 07:51.35 | athom | ok, I'm posting it there |
| 07:51.37 | athom | :) |
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| 07:52.00 | [TK]D-Fender | ChannelZ: Actually He multi-cast it to all 3 channels |
| 07:52.15 | Godfather_ | why diguim sell analogs cards without the echo canceller module? i mean, who will need it without the echo canceller? This is just for making more money? |
| 07:52.15 | ChannelZ | Oh, how lovely for everyone. |
| 07:52.32 | ChannelZ | Godfather_: I use software EC and it works fine |
| 07:52.42 | Godfather_ | ChannelZ, mg2? |
| 07:52.53 | Godfather_ | or oslec? |
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| 07:53.50 | [TK]D-Fender | Godfather_: Stock SWEC with DAHDI isn't very good. |
| 07:54.26 | [TK]D-Fender | Godfather_: OSLEC is a much better option in the "free" side, and Digium's cards under warranty are entitled to HPEC. |
| 07:54.38 | [TK]D-Fender | Godfather_: Or can be added for $10/channel |
| 07:55.21 | Godfather_ | [TK]D-Fender, whats Stock SWE? |
| 07:56.09 | ChannelZ | I'm using mg2. On rare occasions I will answer a call with some echo but it figures it out after a few seconds. Maybe I'm just getting lucky with my telco and whose calling |
| 07:56.22 | Godfather_ | Well, really i tried mg2 and i get a big echo on one side. |
| 07:57.03 | Godfather_ | I'm a newbie and thats why i'm thinking SF echo cancellers doesnt work :| |
| 07:57.16 | [TK]D-Fender | Godfather_: SoftWare Echo Cancellation |
| 07:57.49 | [TK]D-Fender | Godfather_: No... they don't work because they don't work |
| 07:58.09 | [TK]D-Fender | Godfather_: Some are hit&miss depending on your line conditions. |
| 07:58.15 | [TK]D-Fender | Godfather_: so DEAL WITH IT |
| 07:58.34 | [TK]D-Fender | ok.. bed time.. |
| 07:58.36 | [TK]D-Fender | later all |
| 07:58.49 | Godfather_ | :s |
| 07:59.26 | Godfather_ | I'm no going to deal with it, im forced to buy the fu*****-echo-canceller module. |
| 08:00.20 | Godfather_ | ChannelZ, then you are a lucky guy |
| 08:09.59 | coppice | Godfather_: MG2 is pretty useless, but OSLEC with an analogue card that has just a few channels works fine. That is why Digium and Sangoma and everyone else sell smaller cards without hardware echo cancellation. |
| 08:11.08 | Godfather_ | coppice, i understand: oslec > mg2? |
| 08:11.24 | *** join/#asterisk grEvenX (~even@16ldjst.ip.ssc.net) |
| 08:11.38 | coppice | no. OSLEC works. MG2 doesn't |
| 08:12.43 | Godfather_ | mg2 isnt supposed to work with dahdi analog cards? |
| 08:13.07 | coppice | its pretty useless. |
| 08:13.36 | Godfather_ | ok |
| 08:13.51 | Godfather_ | coppice, do you have an analog card with oslec? |
| 08:14.35 | coppice | what relevance does that have? |
| 08:14.57 | Godfather_ | yesterday night i compiled with dahdi, but i cant enable it, i'm doing something wrong |
| 08:15.42 | coppice | if tzafrir is around, he is the best person to ask about configuration problems with that |
| 08:15.47 | *** join/#asterisk BrendanMcc (~BrendanMc@CPE-60-229-252-81.static.nsw.bigpond.net.au) |
| 08:16.03 | Godfather_ | here is a pastebin , http://pastebin.com/Y5XQk8Rp |
| 08:16.26 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 08:16.40 | Godfather_ | Hum.. ok, he told me about oslec yesterday i think. |
| 08:25.46 | *** join/#asterisk linuxcentos (~linuxcent@rhelbox.uio.no) |
| 08:26.39 | *** join/#asterisk TimeRider (~steve@109.224.131.68) |
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| 08:35.40 | tzafrir | Godfather_, I generally use the patch http://svn.debian.org/viewsvn/pkg-voip/dahdi-linux/trunk/debian/patches/dahdi_linux_extra |
| 08:35.57 | Godfather_ | hi tzafrir |
| 08:36.12 | Godfather_ | hum, did you see my pastebin? |
| 08:37.32 | Godfather_ | I restarted dahdi and i noticed this -> "/usr/share/dahdi/waitfor_xpds: Missing astribank_is_starting", i'm not sure if it could be related with "[Jun 25 10:27:38] WARNING[20374]: chan_dahdi.c:2005 dahdi_enable_ec: Unable to enable echo cancellation on channel 1 (No such device)" |
| 08:38.40 | Godfather_ | in fact, i googled it, and its a svn-commit by you |
| 08:40.44 | tzafrir | Godfather_, if you don't have an Astribank, you don't really need that, so you can ignore that warning |
| 08:40.52 | tzafrir | It was fixed later on |
| 08:41.16 | Godfather_ | yes, i dont, then i'll ignore it. |
| 08:41.33 | Godfather_ | tzafrir, how can i apply the patch? |
| 08:42.39 | *** join/#asterisk mikkel (~mikkel@84-238-113-66.u.parknet.dk) |
| 08:43.35 | tzafrir | cat file | patch -p1 --dry-run # test |
| 08:43.38 | tzafrir | cat file | patch -p1 |
| 08:43.45 | tzafrir | in the asterisk source tree |
| 08:43.58 | tzafrir | that is: in the dahdi-linux source tree |
| 08:44.16 | Godfather_ | asterisk source tree or dahdi-linux? |
| 08:44.39 | tzafrir | dahdi-linux |
| 08:45.03 | tzafrir | you'll also need to remove the module oslec, as it uses the name 'echo' for that module there |
| 08:45.25 | tzafrir | But it should be pulled automatically |
| 08:45.47 | Godfather_ | rmmod oslec? |
| 08:46.21 | Godfather_ | he next patch would create the file drivers/staging/echo/echo.c, |
| 08:46.21 | Godfather_ | which already exists! Assume -R? [n |
| 08:46.55 | *** join/#asterisk JAMMAN2110 (~James@unaffiliated/jamman2110) |
| 08:47.35 | Godfather_ | http://pastebin.com/XCT4YbV5 |
| 08:51.44 | Godfather_ | http://pastebin.com/hD1xVWMi |
| 08:53.29 | tzafrir | hmm... start off with a clean dahdi-linux tree |
| 08:53.58 | Godfather_ | i tar.gz again my dahdi-linux-current.tar.gz and now seems ok |
| 08:56.11 | *** join/#asterisk mallchin (~mallchin@mail.dataproservices.co.uk) |
| 08:57.01 | Godfather_ | tzafrir, done, i have compiled dahdi-linux (with your patch) and dahdi-tools, now, should i try it now making a call? |
| 08:57.01 | Godfather_ | or i have to do something previous |
| 08:57.53 | tzafrir | what's the output of lsdahdi ? Do you see a SWEC? |
| 08:58.32 | Godfather_ | tzafrir, no |
| 08:59.02 | Godfather_ | vitto:/etc/dahdi# lsdahdi |
| 08:59.02 | Godfather_ | vitto:/etc/dahdi# |
| 08:59.55 | tzafrir | so you don't have any dahdi channel ATM. You can't make calls like that |
| 09:00.06 | tzafrir | what's the output of: lsmod | grep dahdi |
| 09:00.47 | Godfather_ | http://pastebin.com/WMY5psVQ |
| 09:01.12 | Godfather_ | lol |
| 09:01.17 | Godfather_ | now throws output |
| 09:01.32 | Godfather_ | http://pastebin.com/LC6FcbWi |
| 09:01.45 | Godfather_ | maybe cause i restarted dahdi? |
| 09:02.38 | mallchin | hi guys, how to pass comma in an argument to a macro? they get interpreted as delimeters and the string is split into two arguments |
| 09:02.42 | tzafrir | looks OK now |
| 09:02.56 | *** join/#asterisk elwinformsma (~elwinform@145.222.138.139) |
| 09:02.57 | tzafrir | Now you just need to make sure asterisk is properly configured. |
| 09:03.02 | Godfather_ | Echo Cancellation: |
| 09:03.02 | Godfather_ | 32 taps |
| 09:03.02 | Godfather_ | currently ON |
| 09:03.10 | tzafrir | What do you see on 'dahdi show channel 1' in asterisk? |
| 09:03.17 | tzafrir | ah, ok. Great |
| 09:03.20 | Godfather_ | :) |
| 09:03.27 | Godfather_ | what number of "taps" should be ok? |
| 09:03.37 | Godfather_ | more taps means more echo cancellation? |
| 09:04.38 | tzafrir | Basically. A tap is a sample. The more samples the echo canceller looks at, it is able to cancel sources of echo that are farther |
| 09:04.47 | tzafrir | But also spends more CPU cycles on it |
| 09:05.41 | tzafrir | "X taps" means "a tail of X/8 milliseconds" |
| 09:05.53 | elwinformsma | Hello, i have a issue with queues in combination with reinvite. Sometimes this results in onesided speech. Does anyone know if setting the option 'w' in the queue app results in reinvite being off for all calls to that queue? |
| 09:06.48 | Godfather_ | tzafrir, i tried with 256 taps now |
| 09:07.02 | Godfather_ | i have some echo in one side (on my ip-phone) |
| 09:07.26 | Godfather_ | but muuuch better than mg2. |
| 09:08.51 | Godfather_ | tzafrir, thx! |
| 09:08.51 | *** join/#asterisk frk2 (~faraz@zivios/member/fkhan) |
| 09:13.19 | *** join/#asterisk daemon (~daemon@cpc1-linc11-2-0-cust594.12-1.cable.virginmedia.com) |
| 09:14.00 | daemon | hey guys got an odd problem, I have just deployed asterisk with a couple of SIP tunnels (remote providers) set up alq etc... got my sound quality absolutely perfect, infact everything works apart from a really odd bug |
| 09:14.06 | daemon | sometimes like first in a mornin |
| 09:14.19 | daemon | it will cut the first call made off after 1 minute ~ 20 seconds |
| 09:14.22 | daemon | after that its fine |
| 09:14.31 | daemon | if its left 4/5 hours not in use it will do the same, one again next call is fine |
| 09:14.40 | daemon | not exactly sure what the problem is |
| 09:14.52 | daemon | im using: |
| 09:14.56 | daemon | <PROTECTED> |
| 09:15.11 | daemon | we dont generally make many more calls than two at a time |
| 09:15.13 | mallchin | hi, how can I get asterisk-1.6 to pass this string to the macro without pasing the commas please? |
| 09:15.26 | mallchin | exten => _X.,n,Macro(MySQL-select,SELECT prefix,idcode FROM foo) |
| 09:15.52 | daemon | no idea about acros but my et would be you need to turn it into a string before you send it |
| 09:15.54 | daemon | or escape the commas |
| 09:15.59 | daemon | tried \ '' or "" |
| 09:16.29 | mallchin | escaping works, but the escape character gets passed to the macro too, which confuses the MySQL command |
| 09:17.19 | daemon | try ' ' that is the normal 'do not interpolate' quote |
| 09:17.19 | mallchin | tried quoting the string but that doesn't appear to work, I'll try single quotes, not tried that |
| 09:17.34 | mallchin | I use single quotes in the string too, I guess I'll need to escape them then |
| 09:18.06 | daemon | that or use " in the string if you can and ' as the surround |
| 09:18.14 | Godfather_ | maybe you could cut your arguments, ${ARG:1} or somethink like that |
| 09:19.09 | mallchin | how would cut help? :) I want to pass the entire string as one argument |
| 09:19.25 | Godfather_ | ahhh |
| 09:19.37 | Godfather_ | it wouldnt sorry |
| 09:19.56 | Godfather_ | mallchin, maybe if you set a variable? |
| 09:20.05 | Godfather_ | with the complete string |
| 09:20.17 | daemon | thats a good idea |
| 09:20.24 | mallchin | Godfather_: good idea, I'll try that next |
| 09:20.28 | Godfather_ | ty |
| 09:20.30 | mallchin | thanks :) |
| 09:20.33 | daemon | a variable should not be double interpreted |
| 09:20.56 | daemon | anyone have any ideas about mine :) |
| 09:21.37 | mallchin | single quotes didn't work, I'll try the string |
| 09:23.06 | Godfather_ | daemon, your prob is veeery odd |
| 09:23.09 | Godfather_ | sorry xD |
| 09:23.13 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 09:23.15 | daemon | :) np |
| 09:23.56 | mallchin | variable didn't work :/ |
| 09:25.35 | mallchin | "SQL_QUERY="SELECT prefix,idcode FROM AVSProxySIPFilter LIMIT 0,1"") in new stack |
| 09:25.47 | mallchin | "MySQL-select,"SELECT prefix,idcode FROM AVSProxySIPFilter LIMIT 0,1"") in new stack |
| 09:25.52 | mallchin | "Query MYSQL-RESULT 1 "SELECT prefix") in new stack |
| 09:25.54 | *** join/#asterisk mrchrisadams (~Adium@87-194-125-43.bethere.co.uk) |
| 09:28.04 | elwinformsma | Hello, i have a issue with queues in combination with reinvite. Sometimes this results in onesided speech. Does anyone know if setting the option 'w' in the queue app results in reinvite being off for all calls to that queue? |
| 09:29.47 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 09:29.51 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 09:37.49 | *** join/#asterisk Mw3 (mw3@mw3.hu) |
| 09:38.01 | krion | hi guys |
| 09:38.33 | mallchin | having real problems here :/ must be possible to pass a string containing commas without passing escape sequences, to a macro? |
| 09:38.50 | krion | is a mean jitter of 1 ms could explain low quality of voip calls (apology for my englsih) ? |
| 09:39.37 | krion | http://pastebin.com/LxCM0cu3 here is an analysys of a call whith poor quality, i don't know what's supposed to be correct value for max delta, max skew etc... |
| 09:40.11 | troffasky | 1ms is very low, so I doubt it |
| 09:41.50 | AAA | mallchin single quotes like --> ' |
| 09:41.52 | *** join/#asterisk BANSAL (~bansal@117.207.80.191) |
| 09:42.57 | krion | troffasky: and what about a max delta of 250ms ? |
| 09:43.14 | troffasky | I don't know what that means |
| 09:44.34 | AAA | krion delta is the time transpired. 250ms is about a quarter of a sec |
| 09:45.49 | mallchin | AAA: doesn't appear to work for me |
| 09:46.05 | mallchin | exten => _X.,n,Macro(MySQL-select,'SELECT prefix,idcode FROM AVSProxySIPFilter LIMIT 0,1') |
| 09:46.25 | mallchin | "ARG1 = 'SELECT prefix") in new stack |
| 09:46.35 | AAA | mallchin oh, asterisk stuff. why not ask on #asterisk? |
| 09:46.50 | mallchin | erm... |
| 09:47.43 | mallchin | this isn't woodshop class? |
| 09:47.48 | AAA | mallchin and double quotes don't do it either? |
| 09:48.09 | AAA | mallchin seems like you are misisng an ( |
| 09:48.26 | AAA | mallchin nm, I'm blind |
| 09:48.56 | AAA | "ARG1 = 'SELECT prefix") in new stack |
| 09:49.04 | AAA | ^-- mismatched quotes |
| 09:49.18 | mallchin | that's part of the logs |
| 09:49.37 | mallchin | exten => s,1,NoOp(ARG1 = ${ARG1}) |
| 09:49.56 | mallchin | it must have passed the single quote to the macro |
| 09:50.24 | mallchin | double quote a no-go |
| 09:51.03 | mallchin | it seems quotes and double quotes are passes to the macro, and escaping works, but the argument contains the escape characters |
| 09:51.21 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 09:51.33 | mallchin | I'm pretty adept with syntax and suprised it doesn't work, it seems almost impossible to get the desired result |
| 09:51.39 | *** join/#asterisk jayprakash (~jay@121.247.146.70) |
| 09:52.03 | mallchin | hugs PHP |
| 09:52.09 | krion | AAA: ok, thanks |
| 09:53.39 | troffasky | that's got to be a bug then if the escape chars 'work' but get passed on |
| 09:53.46 | mallchin | agreed |
| 09:53.58 | AAA | mallchin grrr. I know what you mean. in the time you could put \ in front of stuff, you seek the real answer with no results |
| 09:55.01 | mallchin | I wouldn't mind escaping, it's the correct solution, but the escape character should be removed |
| 09:55.20 | AAA | mallchin with single quotes too? |
| 09:55.51 | mallchin | AAA: single and double quotes get passed to the macro |
| 09:56.11 | *** join/#asterisk evangelion (~manzy_zet@grumello.interac.it) |
| 09:56.30 | AAA | hack in a sed -e 's/\\//g' at the end or some'n? |
| 09:56.57 | evangelion | hello, how can i force asterisk to avoid transcoding at all? |
| 09:57.03 | mallchin | "Query MYSQL-RESULT 1 "SELECT prefix\") in new stack |
| 09:57.36 | mallchin | AAA: seems the only way, I'll try it now |
| 09:57.40 | evangelion | i mean how can i force asterisk to negotiate the _same_ codec in both side of a bridged call? |
| 09:57.49 | AAA | just realized he was on the wrong channel |
| 09:57.53 | mallchin | AAA: ;) |
| 09:58.09 | mallchin | AAA: I wondered about your #asterisk statement |
| 09:58.21 | AAA | mallchin hehe, just hit me... |
| 09:58.34 | *** part/#asterisk azizLIGHTS (deltron@silenceisdefeat.com) |
| 09:59.57 | troffasky | evangelion, set only one codec as allowed for the peer in sip.conf |
| 10:00.39 | *** join/#asterisk jayprakash (~jay@121.247.146.70) |
| 10:03.00 | *** join/#asterisk Delido1983 (~Delido198@212.144.236.196) |
| 10:03.18 | *** join/#asterisk debuggerboy (~anish@121.247.146.70) |
| 10:04.58 | Delido1983 | Hello, i have an problem with the caller id Num from some incomming calls i can see only the base number from it. If i call from my handy to asterisk all is okay but 2 customers who called in the nummber is incorrect |
| 10:05.39 | Delido1983 | i have debug the call: [Jun 25 11:44:01] VERBOSE[20481] chan_dahdi.c: [Jun 25 11:44:01] < Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) [Jun 25 11:44:01] VERBOSE[20481] chan_dahdi.c: [Jun 25 11:44:01] < Presentation: Presentation permitted, user number not screened (0) '221XXXX966' ] [Jun 25 11:44:01] VERBOSE[20481] chan_dahdi.c: [Jun 25 11:44:01] < [6c 08 |
| 10:05.43 | Delido1983 | i see only 221XX0 |
| 10:06.23 | *** join/#asterisk Tim_Toady (~moi@178.128.16.115.dsl.dyn.forthnet.gr) |
| 10:06.59 | evangelion | troffasky: i can't |
| 10:07.09 | *** join/#asterisk sp4rc (~sp4rc@178-83-239-81.dclient.hispeed.ch) |
| 10:07.52 | sp4rc | guys, can someone point me to a website/document which contains information about codecs (bit-robustness, packet-loss tolerance)? |
| 10:09.39 | sp4rc | which codecs are the most widely spread? |
| 10:10.12 | sp4rc | g.711 / speex / ilbc / ... ? |
| 10:11.05 | Chainsaw | g711 here. |
| 10:11.24 | Chainsaw | g722 means I get half-speed distorted calls pretty much immediately. |
| 10:11.31 | Chainsaw | So I disallow that explicitly. |
| 10:11.39 | sp4rc | Chainsaw: can you say something about the packet loss tolerance? |
| 10:11.46 | Chainsaw | sp4rc: <1% for G711. |
| 10:12.09 | Chainsaw | sp4rc: Generally worse for codecs that compress more. |
| 10:12.10 | sp4rc | Chainsaw: hm, do you have any sources? i need those for my thesis |
| 10:12.23 | jayprakash | Hi, i am facing problem to getting incoming caller id from my PSTN line. I am using Sangoma sangoma AF200 card with my asterisk box. can u tell me how to get the caller id of indian PSTN line |
| 10:12.57 | Chainsaw | sp4rc: http://www.psytechnics.com/downloads/VoIP_benchmarking_report.pdf |
| 10:13.08 | sp4rc | Chainsaw: so you mean, the higher the compression ratio the lower is the packetloss tolerance? |
| 10:13.31 | Chainsaw | sp4rc: That is correct. It makes sense mathematically. |
| 10:13.42 | Chainsaw | sp4rc: Here is how I acquired my sources, I would recommend you try something similar: http://www.google.co.uk/search?sourceid=chrome&ie=UTF-8&q=G711+ulaw+packet+loss+benchmark |
| 10:14.26 | sp4rc | Chainsaw: thank you very much, tried googling myself before... but couldnt find anything useful except: http://speex.org/comparison/ |
| 10:14.43 | Chainsaw | sp4rc: G711 tends to be yardstick that everything else is compared to. |
| 10:14.48 | Chainsaw | sp4rc: So that will need to be in your search string. |
| 10:15.09 | sp4rc | Chainsaw: G711 is the codec used by isdn, right? |
| 10:15.37 | mallchin | what would be the regexp command to set a string removing backslashes please? |
| 10:17.34 | Chainsaw | sp4rc: That is correct. |
| 10:17.52 | Chainsaw | sp4rc: And as always, there is a "rest of the world" and a "United States" version of it. |
| 10:18.17 | sp4rc | Chainsaw: which means a-law and u-law |
| 10:19.16 | troffasky | America, Fuck Yeah! |
| 10:20.22 | Chainsaw | sp4rc: Indeed. |
| 10:21.29 | daemon | mallchin, s/\\// will remove one |
| 10:21.38 | daemon | or replace it for // nothing |
| 10:22.02 | daemon | mind you, you should if its perl re be able to use a different regex char |
| 10:22.11 | daemon | so s#\## should also work |
| 10:22.46 | mallchin | daemon: thanks, I was planning on using RegExp in asterisk, unsure of the syntax |
| 10:23.06 | mallchin | exten => s,1,Set(SQL=${REGEX("[s/\\//]" ${ARG1})}) |
| 10:23.07 | mallchin | ? |
| 10:23.30 | daemon | should work if its using perl re, try it :) |
| 10:24.03 | mallchin | wil do :) |
| 10:24.06 | mallchin | *will |
| 10:24.48 | mallchin | SQL = 1 :-/ |
| 10:26.26 | daemon | <daemon> eval: my $str = 'some\stuff'; ($str) =~ s/\\//; return $str; |
| 10:26.27 | daemon | <buubot> somestuff |
| 10:26.28 | *** join/#asterisk Trixboxer (~Trixboxer@115.124.115.69) |
| 10:27.10 | mallchin | do I need to do an exec to perl? |
| 10:27.33 | daemon | mallchin, you do know from earlier that im a total noob with asterisk and im only advising you on what happens in different apps and languages right lol |
| 10:27.44 | mallchin | lol |
| 10:27.52 | daemon | im steeping out now before I tell you to try something and your pc turns into nuclear missile launcher and kills the world ;) |
| 10:27.53 | *** join/#asterisk micols (~mio@rlogin.dk) |
| 10:28.15 | mallchin | it says skynet active? |
| 10:28.21 | mallchin | :D |
| 10:28.21 | daemon | haha |
| 10:29.07 | daemon | are you still trying to ge asterisk to send comma |
| 10:29.08 | Godfather_ | i getting Junk at the beggining of frame each time the mp3 on musiconhold is repetead (duration 8 secs) |
| 10:29.09 | daemon | without processing it |
| 10:29.10 | Godfather_ | [Jun 25 12:24:06] WARNING[26259]: mp3/interface.c:216 decodeMP3: Junk at the beginning of frame 49443304 |
| 10:29.19 | mallchin | daemon: yes |
| 10:29.21 | daemon | hmm |
| 10:29.47 | mallchin | I just want to pass a string to a macro as one argument, and containing commas |
| 10:30.25 | mallchin | pretty basic stuff :( |
| 10:31.38 | daemon | hmm |
| 10:31.40 | jayprakash | quit |
| 10:31.42 | jayprakash | exit |
| 10:32.09 | jayprakash | exit |
| 10:32.13 | jayprakash | quit |
| 10:35.00 | daemon | mallchin, just a test |
| 10:35.02 | daemon | not sure what this actually does |
| 10:35.07 | daemon | try setting ya crap in one of these |
| 10:35.19 | daemon | Set(LOCAL(someVar)=Sql,Sucks,Ass); |
| 10:35.29 | daemon | and send it in your macro as: NoOp(${someVar}); |
| 10:38.37 | *** part/#asterisk debuggerboy (~anish@121.247.146.70) |
| 10:39.19 | daemon | oh wait |
| 10:39.20 | daemon | I see |
| 10:39.31 | Godfather_ | <PROTECTED> |
| 10:39.32 | Godfather_ | <PROTECTED> |
| 10:39.34 | Godfather_ | whats the problem? |
| 10:39.41 | daemon | no-op means debug |
| 10:39.44 | daemon | drop the message to console |
| 10:39.52 | Godfather_ | Yes |
| 10:39.58 | daemon | but simply (${someVar}) |
| 10:40.01 | daemon | should now work |
| 10:40.14 | daemon | Im reading from a bug report about comma's I did not read the bit that said noop was debug purpose ;p |
| 10:40.46 | daemon | exten => s,1,Set(SQL=(${someVar}) |
| 10:40.52 | daemon | exten => s,1,Set(SQL=(${someVar})) |
| 10:40.54 | daemon | should be the ticket |
| 10:41.36 | daemon | maybe do not need extra ( ) |
| 10:42.29 | mallchin | exten => s,1,GotoIf(${REGEX("[s/\\//]" ${ARG1})}?foo) |
| 10:42.33 | mallchin | exten => s(foo),n,NoOp(Deprecated - Your query contains deprecated features.) |
| 10:42.44 | mallchin | [Jun 25 11:40:41] NOTICE[4155]: pbx.c:3744 pbx_extension_helper: No such label 'foo' in extension 's' in context 'macro-MySQL-select' |
| 10:42.47 | mallchin | :( |
| 10:43.28 | daemon | mhmm |
| 10:43.56 | daemon | i wonder, this is probably not the right way to do this |
| 10:44.01 | daemon | but can you use ^ instaed of , |
| 10:44.07 | daemon | and in macto-MYSQL-select |
| 10:44.10 | daemon | regex change ^ to , |
| 10:44.33 | daemon | or some other really weird character |
| 10:44.45 | evangelion | can i force asterisk to negotiate the _same_ codec in both sides of a bridged call even if multiple codecs are allowed? |
| 10:44.45 | mallchin | I don't use ^ |
| 10:45.34 | daemon | exten => _X.,n,Macro(MySQL-select,SELECT prefix,idcode FROM foo) |
| 10:46.01 | daemon | mallchin, as the command is MYSQL-select do you need to send SELECT in the query? |
| 10:46.02 | troffasky | why allow multiple codecs if you want to restrict it to one? |
| 10:46.52 | mallchin | daemon: MySQL-select is a macro which runs the command MYSQL() |
| 10:46.57 | daemon | right |
| 10:47.27 | *** join/#asterisk jetlag (jetlag@pool-173-61-204-106.cmdnnj.east.verizon.net) |
| 10:47.31 | mallchin | I don't understand why, when I have a label in the context, it says I do not |
| 10:47.37 | mallchin | can you not use labels in macros? |
| 10:47.44 | evangelion | troffasky: i don't want to restrict it to one! i want to avoid transcoding and be free to choice the codec time by time |
| 10:48.22 | daemon | mallchin, weird according to the docs |
| 10:48.39 | daemon | exten => _X.,n,Macro(MySQL-select,SELECT prefix\,idcode FROM foo) |
| 10:48.43 | daemon | should work flawless |
| 10:49.37 | mallchin | daemon: it passes the whole string as one argument to the macro, but doesn't strip the backslash, which invalidates the SQL |
| 10:51.41 | daemon | mallchin, how is MYSQL() defined |
| 10:51.43 | daemon | is it an internal command |
| 10:52.11 | mallchin | yes, part of asterisk-addons I believe |
| 10:52.48 | daemon | exten => s,3,MYSQL(Query resultid ${connid} SELECT\ name\ FROM\ sip\ WHERE\ name\ LIKE\ "'${ARG1}%'") |
| 10:52.57 | mallchin | can anyone tell me why jumping to foo fails in this macro please? |
| 10:52.58 | mallchin | http://pastebin.com/j1ihpMeT |
| 10:53.23 | daemon | apparently you can only fetch one |
| 10:53.24 | mallchin | daemon: asterisk 1.4 and below require the backslashes |
| 10:53.27 | daemon | and you do not use comma |
| 10:53.32 | daemon | oh |
| 10:53.43 | mallchin | you can fetch multiple fields |
| 10:54.16 | mallchin | this worked on asterisk-1.4, but broke with the MYSQL changes on asterisk-1.6 |
| 10:55.52 | daemon | hmm |
| 10:56.17 | daemon | this is annoying |
| 10:56.23 | daemon | its such a stupid little problem |
| 10:56.43 | mallchin | yep :( |
| 10:58.18 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 11:00.16 | Gugge | mallchin: what is wrong with Macro(macroname,"string,with commas") |
| 11:00.18 | *** join/#asterisk stix (~stix@firewall.o4.dk) |
| 11:00.32 | daemon | Gugge, we tried that earlier tried '' too |
| 11:00.55 | mallchin | Gugge: it passes the double-quotes and the backslash to the macro |
| 11:01.01 | Gugge | what backslash? |
| 11:01.19 | mallchin | Gugge: sorry, it passes a backslash if one is in there |
| 11:01.54 | mallchin | Gugge: using your example, without the backslash, it splits the string at the comma |
| 11:02.06 | Gugge | i guess you would have to strip the " from the arg in the macro, if the MYSQL() cmd wont accept them |
| 11:02.19 | Gugge | hmm, strange |
| 11:02.55 | daemon | mallchin, I would be tempted to make a perl script call that instead of MYSQL() |
| 11:02.56 | mallchin | frustrating :( |
| 11:03.04 | daemon | then you can do any processing you want in the script |
| 11:03.40 | mallchin | daemon: it's a solution, I would use a PHP AGI rather than perl, but I'd rather fix the issue as I will encounter it elsewhere in the futurte |
| 11:04.14 | daemon | so you are saying |
| 11:04.16 | daemon | " " works |
| 11:04.18 | daemon | but it sends the " as well |
| 11:04.28 | daemon | so |
| 11:04.42 | daemon | "some,random,crap gets passed as litrally "some, random, crap" |
| 11:06.07 | mallchin | it gets passes as "some |
| 11:06.16 | mallchin | the comma splits the string |
| 11:06.22 | daemon | ok |
| 11:06.27 | daemon | I think I have an idea |
| 11:06.43 | daemon | Set(LOCAL(someVar)=Sql,Sucks,Ass); |
| 11:08.18 | mallchin | I tried that, but I'll give it another go |
| 11:08.21 | mallchin | :) |
| 11:08.43 | daemon | exten => _X.,n,Macro(MySQL-select,$["${someVar}"]) |
| 11:09.07 | daemon | I need a moment to type up the idea |
| 11:09.10 | daemon | I think that could work |
| 11:09.10 | daemon | lol |
| 11:09.30 | Godfather_ | c'mon daemon go |
| 11:10.26 | mallchin | LOL |
| 11:10.29 | mallchin | kk :D |
| 11:11.23 | mallchin | exten => _X.,n,Set(SQL=SELECT prefix,idcode FROM AVSProxySIPFilter LIMIT 0,1) |
| 11:11.24 | mallchin | exten => _X.,n,Macro(MySQL-select,${SQL}) |
| 11:11.33 | mallchin | "ARG1 = SELECT prefix") in new stack |
| 11:11.40 | mallchin | "ARG2 = idcode FROM AVSProxySIPFilter LIMIT 0") in new stack |
| 11:11.45 | mallchin | "ARG3 = 1") in new stack |
| 11:11.59 | daemon | damn it |
| 11:13.02 | mallchin | I tried $["${SQL}"] too but the double quotes get passed inside the string |
| 11:14.16 | daemon | it should not interpolate a comma in a variable |
| 11:14.17 | daemon | this is freaking nuts |
| 11:15.08 | daemon | mallchin, this is a really random observation and probably will just flat out not work |
| 11:15.11 | daemon | what happens if you use $SQL |
| 11:17.23 | *** join/#asterisk mpe (~mpe@pD95F4BCA.dip.t-dialin.net) |
| 11:18.51 | Faithful | asterisk says it's answering the pstn-to-voice gateway on this spa3102 but it isn't it just keeps ringing on the pstn side. I have tried both a spa3000 and spa3102 with the same problem now |
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| 11:29.06 | mallchin | daemon: I don't think $SQL is valid syntax |
| 11:29.18 | daemon | ah :( did not think so |
| 11:29.27 | daemon | right im going to go out and get drunk |
| 11:29.40 | daemon | I admiteddly was gonig to do the garden today, but after trying to figure this out |
| 11:29.44 | daemon | the pub sounds a better bet |
| 11:29.48 | mallchin | haha |
| 11:29.48 | daemon | sorry I could not help mallchin |
| 11:29.54 | mallchin | it's a lovely day for the pub here |
| 11:30.01 | daemon | yeah its nice here to (england) |
| 11:30.07 | mallchin | daemon: no problem, thanks for trying :) |
| 11:30.17 | mallchin | Essex :) |
| 11:30.21 | daemon | oh just a side though.. I have no IDEA if this is even possible |
| 11:30.36 | daemon | but can you bind together the value of two different commands |
| 11:30.41 | daemon | the thought is you could maybe do something like |
| 11:30.50 | daemon | exten => _X.,n,Set(SQL=SELECT prefix FROM AVSProxySIPFilter LIMIT 0,1) |
| 11:30.56 | daemon | exten => _X.,n,Set(SQL=SELECT idcode FROM AVSProxySIPFilter LIMIT 0,1) |
| 11:31.01 | daemon | and call them seperately then bind the values |
| 11:31.06 | daemon | not pretty but no comma's so it should work |
| 11:31.21 | daemon | you do not need limit 0,1 btw limit 1 does the same thing ;) |
| 11:32.09 | mallchin | daemon: I could, but I use commas elsewhere in the statement -- I simplified the statement when testing, thus the limit 0,1 |
| 11:32.20 | mallchin | daemon: the proper statement only returns 1 row |
| 11:32.28 | daemon | ah :( |
| 11:32.49 | daemon | ok buddy im off to wetherspoons :P good luck and if it does not work, ill buy you a drink at the bar ^_- |
| 11:32.58 | mallchin | haha |
| 11:33.01 | mallchin | have one on me mate! |
| 11:33.06 | mallchin | enjoy :) |
| 11:34.12 | sp4rc | Chainsaw: can you make a statement about the quantitativ (percentage) packet-loss of ilbc, amr-nb and gsm efr? |
| 11:34.28 | sp4rc | Chainsaw: i mean packet-loss-tolerance |
| 11:35.06 | Chainsaw | sp4rc: I can make educated guesses, but if you want hard numbers and "sources", you're going to have to ask Google. |
| 11:35.58 | sp4rc | Chainsaw: all i can find are statements like 'better then...' |
| 11:36.28 | Chainsaw | sp4rc: It is highly dependent on what packet loss compensation algorithms are in use at both ends (and even the type of packet loss, intermittent high packet loss is a lot worse then loss that is consistent over time). |
| 11:37.07 | Chainsaw | sp4rc: I found benchmarks with graphs and well defined testing conditions. Please try a little harder. |
| 11:37.07 | sp4rc | Chainsaw: okay i see this is big studying field for itself... |
| 11:37.22 | Chainsaw | sp4rc: I must remind you that it is your thesis, not mine. |
| 11:38.42 | sp4rc | Chainsaw: i am aware of that, thank you. the main focus is not on codec's and packet loss but handoff technics over different network technologies |
| 11:39.56 | coppice | ilbc has better tolerance of packet loss than AMR-NB, but its bit rate is so much higher, you could send AMR-NB with redundancy, still be at the ilbc bit rate, and have better packet loss tolerance than ilbc. :-) |
| 11:40.00 | Chainsaw | sp4rc: I can tell you that the packet loss tolerance of SIP can be lower then G711. |
| 11:40.18 | Chainsaw | sp4rc: We had no discernable voice quality problems, but a faulty switch caused intermittent transfer failures. |
| 11:40.44 | Chainsaw | sp4rc: Packet loss does funny things to UDP-based protocols. |
| 11:43.39 | sp4rc | Chainsaw: i need to define different criteria for a mobility system, one of this is to keep the packet-loss as small as possible... but this value mainly depends on the choice of codec |
| 11:44.09 | Chainsaw | sp4rc: Lowest possible loss, but don't underestimate consistency. |
| 11:44.52 | sp4rc | Chainsaw: ...which goes hand in hand with jitter |
| 11:44.54 | Chainsaw | sp4rc: A consistent 120ms latency with 2% loss can be more reliable then wildly varying 20-100ms latency and no loss at all. |
| 11:44.59 | Chainsaw | sp4rc: Indeed. |
| 11:45.23 | sp4rc | Chainsaw: i see... |
| 11:45.27 | Chainsaw | sp4rc: Just be sure to include it in all your explanations, because they'll have you on the hook for oversimplification otherwise. |
| 11:49.04 | sp4rc | Chainsaw: so one should _always_ look at the three main parameters: packet-loss, jitter, latency |
| 11:49.18 | sp4rc | Chainsaw: i mean looking at them at the same time |
| 11:49.50 | Chainsaw | sp4rc: Indeed. That defines how well a network link will perform with VoIP. |
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| 12:10.32 | ZeXr0 | In my callflow, I'm doing SendDTMF(#) then Read(data). The automated device that is calling only pause for 100ms before sending a series of DTMF, but it seems that the Read isn't fast enough, and there's some keys that are missed. Is it possible to play a DTMF tone in the background and start listening right when the tone is played ? Or is there any alternative solution, like doing Background(PoundSound) ? |
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| 12:16.53 | Delido1983 | Hey, i have a problem to see the correct callernumber from 2 customer: chan_dahdi.c: [Jun 25 11:44:01] < Calling Number (len=14) [ Ext: 0 TON: National Number (2) Presentation: Presentation permitted, user number not screened (0) '2XXXX19966' ] Presentation: Presentation allowed of network provided number (3) '2XXXX0' ] The degug show this: i want to see the real Numer '2XXXX19966' but i see only '2XXXX0' |
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| 12:24.56 | matagou_ | hello |
| 12:29.33 | rotherad | hmmm.... |
| 12:29.40 | rotherad | having an issue with my setup this morning |
| 12:29.49 | rotherad | ive added a new trunk to a secondary sip provider |
| 12:30.04 | rotherad | and now whenever my SCCP devices call out there is no audio being received |
| 12:30.17 | rotherad | audio is being sent as the other person can hear the sccp device |
| 12:30.22 | rotherad | but not the other way |
| 12:30.41 | rotherad | they are dialling out with a prefix that ensures it goes out 1 particular sip trunk |
| 12:30.44 | matagou_ | having issue when upgrading asterisk 1.4 to 1.6 |
| 12:31.26 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 12:33.15 | matagou_ | after upgrading asterisk to latest 1.6, when trying to start asterisk, it crashes and keep restarting |
| 12:33.57 | ZeXr0 | matagou_ : I can't really help because I don't know a lot, but you can try to stop asterisk, and then run asterisk -vvvvgc |
| 12:34.17 | ZeXr0 | you might receive more information about why asterisk isn't starting |
| 12:36.52 | pabelanger | m17555#last |
| 12:37.17 | pabelanger | matagou_: Read my instructions on your issue |
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| 12:39.27 | matagou_ | ZeXr0: started asterisk -vvvvgc, it complains with errors on loading modules and unknown functions |
| 12:39.27 | Delido1983 | matagou_: its happend when asterisk load old moduls (asterisk 1.4) i must clear the modules folder than is all okay :D |
| 12:39.45 | Delido1983 | then reinstall asterisk..^^ |
| 12:40.27 | matagou_ | ZeXr0: loader.c:429 load_dynamic_module: Error loading module 'func_curl.so': /usr/lib/asterisk/modules/func_curl.so: undefined symbol: ast_custom_function_register |
| 12:41.18 | matagou_ | ok, i will try to remove the /usr/lib/asterisk/modules |
| 12:41.42 | matagou_ | and issue command make install again |
| 12:42.59 | matagou_ | pabelanger: where i can read your instructions? |
| 12:43.32 | pabelanger | matagou_: In the mantis issue you reported this morning |
| 12:44.30 | matagou_ | i will check it right now |
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| 12:54.58 | matagou_ | ok, i will execute all the steps in mantis |
| 12:55.05 | matagou_ | will report later |
| 12:55.11 | matagou_ | thanks for support |
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| 13:20.34 | Katty | hi |
| 13:20.55 | chuckf | lo |
| 13:21.26 | knctrnl | Does anyone know of any good articles or any opensource software to exploit the features of fax for asterisk? |
| 13:21.34 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 13:21.35 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 13:21.39 | knctrnl | or does one need to write something custom? |
| 13:23.07 | Naikrovek | not a lot of us use fax for asterisk, as i understand it |
| 13:23.14 | Naikrovek | not a whole pile of fax expertise in here |
| 13:23.56 | knctrnl | I have not looked deeply into it. I was just wondering if it was even worth exploring. |
| 13:24.21 | knctrnl | email to fax and fax to email look like a huge cost savings to a large enterprise |
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| 13:28.21 | Katty | ooo! squirrely visitors this morning! |
| 13:28.41 | Katty | http://www.ustream.tv/channel/squirrel-critter-cam |
| 13:29.17 | Katty | squirrels are funny like that. the feeders have been empty for a couple months. but i fill them up and in 30 minutes POOF. breakfast time. |
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| 13:40.15 | mallchin | how can I remove backslashes from a string please? |
| 13:40.53 | pabelanger | sed? |
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| 13:43.10 | mallchin | kk, thanks |
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| 13:58.45 | Delido1983 | can someone help me with my callerid Num? i have a problem.. |
| 13:59.18 | WIMPy | ~ask |
| 13:59.19 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 13:59.42 | Delido1983 | Hey, i have a problem to see the correct callernumber from 2 customer: The degug show this: chan_dahdi.c: [Jun 25 11:44:01] < Calling Number (len=14) [ Ext: 0 TON: National Number (2) Presentation: Presentation permitted, user number not screened (0) '2XXXX19966' ] Presentation: Presentation allowed of network provided number (3) '2XXXX0' ]i want to see the real Numer '2XXXX19966' but i see only '2XXXX0' |
| 14:01.01 | WIMPy | What you see IS the real number. |
| 14:01.18 | WIMPy | But I dont think, you have a choice, which number you see. |
| 14:01.56 | Delido1983 | if the customer called me to my handy i see the full number |
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| 14:06.51 | mallchin | how can I pass a string containing commas to a macro? |
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| 14:13.49 | ZeXr0 | mallchin : Have you tried with "" |
| 14:13.56 | ZeXr0 | "string,,,,,withcomma" |
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| 14:14.28 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:15.27 | mallchin | ZeXr0: yes, the double quote is passed as part of the string, and the comma is still interpreted as a delimeter |
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| 14:20.06 | mallchin | I am suprised this doesn't work |
| 14:20.15 | mallchin | seems like a basic feature |
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| 14:26.48 | russellb | mallchin: I think you escape it with a backslash |
| 14:27.51 | russellb | checks |
| 14:27.59 | russellb | mallchin: version? |
| 14:28.48 | pabelanger | Delido1983: Call your telco, they are blocking your ANI |
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| 14:30.33 | Godfather_ | hi |
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| 14:30.49 | Delido1983 | pabelanger: outgoing callerid num is everytime correct (i think so) only incoming is the problem |
| 14:31.28 | pabelanger | Delido1983: Yes, I understand. Tell your telco that. |
| 14:31.52 | pabelanger | Delido1983: The problem is outside of your (asterisk's) control. |
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| 14:32.05 | Delido1983 | pabelanger: ah okay i missunderstood this |
| 14:33.23 | Delido1983 | pabelanger: okay and the option prilocaldialplan oder pridailplan can not be the problem? |
| 14:34.33 | pabelanger | Delido1983: You'd need to pb a debug log for your PRI so we can see the IE |
| 14:35.28 | Delido1983 | pabelanger: chan_dahdi.c: [Jun 25 11:44:01] < Calling Number (len=14) [ Ext: 0 TON: National Number (2) Presentation: Presentation permitted, user number not screened (0) '22XXX66' ] Presentation: Presentation allowed of network provided number (3) '22XXX0' ] |
| 14:35.38 | Delido1983 | pabelanger: or you need more? |
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| 14:36.07 | mallchin | russellb: Asterisk 1.6.2.8 |
| 14:36.26 | pabelanger | Delido1983: If that is the information you get when you enable pri debug, then the problem is with your telco. |
| 14:37.12 | Delido1983 | pabelanger: thanks for this information i will call vodafone |
| 14:37.49 | russellb | mallchin: app_macro doesn't use the argument parsing API which properly handles this ... so you can't do it |
| 14:37.51 | russellb | :-( |
| 14:37.58 | russellb | kicks app_macro.c |
| 14:38.22 | mallchin | kicks app_macro.c too |
| 14:38.24 | mallchin | :( |
| 14:38.30 | mallchin | can I patch it to make it so? |
| 14:38.41 | russellb | sure, it's just software :-) |
| 14:39.10 | mallchin | hrm, maybe a monday morning thing |
| 14:39.28 | russellb | it's not a one liner or anything, though |
| 14:39.41 | mallchin | I had a feeling that might be the case, hehe |
| 14:39.54 | mallchin | is there a workaround? |
| 14:40.01 | russellb | but FWIW, line 352 in apps/app_macro.c is where you would start looking |
| 14:40.15 | mallchin | excellent, thanks, I'll have a look now |
| 14:40.22 | russellb | yeah, there's some workarounds |
| 14:40.38 | russellb | instead of a macro argument, just set a channel variable and read it in your macro |
| 14:40.47 | russellb | that's all arguments to Macro() get turned into anyway |
| 14:40.57 | mallchin | using Set()? |
| 14:41.00 | *** join/#asterisk Pidgeon (~Pidgeon@78-141-41-9.rdns.as8401.net) |
| 14:41.02 | russellb | Set(MYARGWITHACOMMADAMNIT=,,,,,,) |
| 14:41.03 | russellb | yes |
| 14:41.24 | mallchin | Okay, great, I'll give it a go |
| 14:41.28 | russellb | k. |
| 14:41.39 | Pidgeon | Hello, before I start using asterisk I would like to know if there is a TAPI driver for it to appear in Windows Telephony - is there one? |
| 14:43.29 | *** join/#asterisk mog (~mog@c-68-62-169-225.hsd1.al.comcast.net) |
| 14:44.04 | drmessano | Pidgeon --> http://www.google.com/search?q=Asterisk+TAPI |
| 14:44.08 | drmessano | Quite a few hits there |
| 14:46.03 | mallchin | russellb: works for me, thanks |
| 14:46.12 | *** join/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 14:47.28 | mallchin | I'm sending a call via IAX but get no audio |
| 14:47.34 | drmessano | Pidgeon: and FWIW, Asterisk supports a range of protocols and methods for initiating calls, if there is a TAPI driver for that particular protocol, it should work. As you can see there are quite a few SIP TAPI apps, which seems to be the common Windows TAPI target for Asterisk |
| 14:47.43 | mallchin | russellb: been bugging me all day that macro problem, thanks again :) |
| 14:48.37 | russellb | mallchin: you're welcome, that'll be $9.95 |
| 14:48.54 | drmessano | irussellb ? |
| 14:49.09 | mallchin | Macro(Pay-for-help,russellb,$9,95) |
| 14:49.13 | drmessano | I thought the russellb wasn't approved for the iPhone store |
| 14:49.16 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 14:49.24 | ZeXr0 | mallchin : That doesn't parse ... |
| 14:49.26 | russellb | drmessano: was denied :-( |
| 14:49.34 | Pidgeon | thanks drmessano... I'm just specifically wondering if something will put a provider in the Telephony mmc snapin. I have done nothing with TAPI before so my question is probably stupid but there isn't anything that clearly explains how this all works (for any switch, I've been playing with an avya IP400 so far today for this) |
| 14:49.47 | mallchin | ZeXr0: escape the $? :) |
| 14:49.51 | Pidgeon | I'll be trying to use http://www.traysoft.com/addtapi_features.htm with it |
| 14:50.03 | ZeXr0 | <PROTECTED> |
| 14:50.12 | ZeXr0 | :P |
| 14:50.22 | mallchin | ZeXr0: you mean, payment failed? >.< |
| 14:50.33 | *** join/#asterisk Sorcier_FXK (~sorcierfx@unaffiliated/sorcierfxk) |
| 14:51.16 | *** join/#asterisk roe (~roe___@unaffiliated/roe) |
| 14:51.17 | drmessano | Pidgeon, you'll just have to play with it. There's numerous links in that Google search, and I have experimented with one or two of them in the past under XP |
| 14:51.35 | WIMPy | Delido1983, pabelanger: I wonder if it would't make sense to actually handle this relatively common situation and put the two numbers into callerid(num) and callerid(ani). |
| 14:51.50 | roe | What is the recommended way to have clients record IVR messages? Is there a recommended way? |
| 14:51.52 | drmessano | I don't do a lot of Windows anymore, so not sure what StillWorks(TM) |
| 14:52.24 | drmessano | roe: I always find that a $20,000 audio production studio works well |
| 14:52.47 | roe | thanks. I'll take that under advisement |
| 14:52.52 | drmessano | roe: Don't get cheap on the mic's.. RE20's are the way to go, even at $500 each |
| 14:53.14 | roe | luckily we are installing a new phone system at a sound studio |
| 14:53.22 | drmessano | roe: You can also use any audio editor you like, since we're not talking high quality audio here |
| 14:55.18 | roe | drmessano, so from that I can surmise that a computer+mic is preferred/recommended as compared to some kind of direct from phone setup? |
| 14:56.04 | drmessano | You can go either way, but it's only going to sound as good as the microphone. If you have a decent phone to record the prompts, go for it |
| 14:56.55 | drmessano | roe: This is open source, there is no "preferred".. only "patches", "bugs", and "feature requests"... ("oh my!") |
| 14:57.49 | drmessano | oh man |
| 14:57.58 | roe | drmessano, I work with a lot of OS projects, most of them while robust and flexible have a 'preferred' way. Thanks for the info. |
| 15:00.38 | *** join/#asterisk rocksfrow (~kyle@74-92-146-97-WashingtonDC.hfc.comcastbusiness.net) |
| 15:00.40 | *** part/#asterisk rocksfrow (~kyle@74-92-146-97-WashingtonDC.hfc.comcastbusiness.net) |
| 15:00.42 | *** join/#asterisk retentiveboy (~pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
| 15:02.08 | drmessano | No one ever created anything better or new going with "preferred", unless we're talking about simply being an end user. |
| 15:02.39 | jsgoecke | Those of you who want an update on Adhearsion 0.8.4 join the VUC today http://twitter.com/adhearsion/status/17020308244 |
| 15:03.32 | *** join/#asterisk sourcode (~code@ppp-115-87-214-43.revip4.asianet.co.th) |
| 15:06.02 | mallchin | any reasons why IAX2 might not plau audio? |
| 15:06.18 | *** join/#asterisk jasonwert (~jasonwert@99-27-170-70.lightspeed.cicril.sbcglobal.net) |
| 15:10.51 | pabelanger | mallchin: Codec issue? |
| 15:11.17 | *** join/#asterisk dmz (~dmz@64.203.207.101.dyn-cm-pool-54.hargray.net) |
| 15:13.05 | xbp | morning |
| 15:13.13 | xbp | mallchin: nat? |
| 15:13.30 | mallchin | xbp: both on internal subnet |
| 15:13.37 | mallchin | pabelanger: trying some different codecs |
| 15:13.49 | mallchin | hope another codec will work :) |
| 15:14.31 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 15:14.46 | pabelanger | xbp: NAT and IAX2 usually not a problem |
| 15:15.25 | mallchin | I'm dialling via SIP, so it could be a SIP issue |
| 15:15.35 | mallchin | SIP call across net, then internal IAX link |
| 15:15.53 | drmessano | ~sipnat |
| 15:15.54 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 15:16.32 | pabelanger | mallchin: Don't reinvite RTP on SIP if you want to use IAX2 trunks. |
| 15:17.20 | mallchin | would directmedia=nonat do? |
| 15:18.06 | mallchin | trying directmedia=no too |
| 15:18.44 | *** join/#asterisk wam (~wam@unaffiliated/wam) |
| 15:19.39 | drmessano | There's more than that involved. Follow the link |
| 15:19.49 | mallchin | reading the guide :) |
| 15:19.51 | *** join/#asterisk raj-darkmystery (~suraj@122.169.16.248) |
| 15:20.01 | wam | Hi, I'm trying to install asterisk gui with asterisk 1.6. Now after the login in the gui the interfaces reloads over and over and over again. The reason seems to be (after a wireshark session) that the rawman command "dialplan%20reload" doesn't work. Asterisk responds with "No such command". Then the interface reloads. |
| 15:20.18 | wam | Any hints? Must I configure the gui for this asterisk version explicitly? |
| 15:21.01 | raj-darkmystery | hey friends.. i was wondering if i can use my asterisk voip configured at my office from my home network.. is it possible? |
| 15:21.16 | russellb | raj-darkmystery: yes. |
| 15:21.48 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 15:22.23 | raj-darkmystery | russellb, can u tell me how i can use that... i have configured my client side soft with all the details 'm using in office but its throwing error :( |
| 15:23.35 | ChannelZ | Humm. Anyone have issues with ChanSpy totally ignoring you hitting the '*' key to switch channels? |
| 15:23.40 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 15:24.18 | drmessano | raj-darkmystery: Assuming you are using SIP, see here: |
| 15:24.26 | drmessano | ~sipnat |
| 15:24.27 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 15:24.38 | drmessano | Oh |
| 15:25.20 | drmessano | Ok, ports are mentioned |
| 15:25.45 | mallchin | works yay :) |
| 15:25.53 | mallchin | not on speakerphone though |
| 15:26.05 | drmessano | Softphone? |
| 15:26.22 | mallchin | Cisco VoIP phone (hard) |
| 15:26.38 | [TK]D-Fender | Cisco phones + NAT handling = PAIN |
| 15:27.01 | mallchin | I'm only using for testing, hopefully in real world it'd work |
| 15:27.22 | mallchin | strange it doesn't work on speakerphone though |
| 15:27.22 | troffasky | yeah, cos if it doesn't work in testing, it's usually fine in the real world |
| 15:27.23 | mallchin | lol |
| 15:27.46 | *** join/#asterisk lost_soul (shackett@devio.us) |
| 15:27.59 | troffasky | use IPv6 then you won't need NAT |
| 15:28.10 | mallchin | if I could, I would |
| 15:28.10 | troffasky | see, simple solution :-) |
| 15:28.20 | drmessano | mallchin: See "Van Halen - Running with the Devil" and anything by Ozzy for some inspiration dealing with Satanic cultures such as the one surrounding Cisco |
| 15:28.33 | mallchin | fires up Spotify |
| 15:28.58 | mallchin | so, do you think this might just be a problem with the Cisco phone? let me try another |
| 15:29.03 | drmessano | mallchin: Also, play the Beatles White Album backwards |
| 15:29.07 | troffasky | satanic culture? is that like evil yoghurt? |
| 15:29.15 | drmessano | lol |
| 15:29.16 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 15:29.45 | niekie | Semi-offtopic question here: Caller ID can't get blocked to US toll-free numbers, or can it? |
| 15:30.01 | niekie | Read something about Wikipedia suggesting it doesn't get blocked to toll-free numbers. |
| 15:30.29 | mallchin | hrm, only works on the Cisco |
| 15:30.34 | drmessano | niekie: That's entirely baseless, considering it's your provider who determines what CID gets passed |
| 15:30.42 | mallchin | [Jun 25 16:26:06] ERROR[28657]: rtp.c:3438 ast_rtcp_write_sr: RTCP SR transmission error |
| 15:30.45 | mallchin | bah |
| 15:31.08 | *** join/#asterisk Raden (~Raden@71.89.121.119) |
| 15:31.18 | [TK]D-Fender | drmessano: I found the simple life... ain't so simple.. |
| 15:31.37 | drmessano | niekie: Flowroute allows me to be a telephone tough guy, tollfree or not |
| 15:31.47 | niekie | Hm. |
| 15:32.22 | niekie | Well, seems I get passed caller ID from Skype calls (which I thought usually have blocked CID) on toll-free lines. |
| 15:32.30 | troffasky | mallchin, what only works on the cisco? |
| 15:32.45 | niekie | drmessano: this was the quote from Wikipedia: "In the U.S. the FCC requires the number to be transmitted to toll-free numbers regardless of whether the number is blocked." |
| 15:33.13 | *** join/#asterisk freezey (~trees@static-64-61-84-174.isp.broadviewnet.net) |
| 15:33.14 | freezey | <PROTECTED> |
| 15:35.40 | drmessano | niekie: That would be great if I was blocking my number |
| 15:35.48 | mallchin | troffasky: internal phone dialling a sip extension to a remote box |
| 15:37.50 | drmessano | niekie: What if I am presenting no number at all? |
| 15:38.10 | *** join/#asterisk Bartockbatz (~chatzilla@c-24-62-161-95.hsd1.nh.comcast.net) |
| 15:38.42 | *** join/#asterisk BANSAL (~bansal@117.207.80.191) |
| 15:39.27 | Bartockbatz | Hello folks - I am using Asterisk 1.4x and I would like to know where to look to disable the default VM messages - ie , "the user at extension 155 is not available...." |
| 15:40.05 | Bartockbatz | not looking for a hand-out - what application/part of the dialplan/config would I look into. - thanks |
| 15:40.45 | *** join/#asterisk idespinner (~idespinne@cpe-76-93-115-243.socal.res.rr.com) |
| 15:41.00 | drmessano | niekie: "In the U.S., if you're one of the 3 people still using analog lines from AT&T, or are using one of the few ITSPs that ridiculously ties your termination to some DID you purchased, the FCC requires the number to be transmitted to toll-free numbers regardless of whether the number is blocked." <-- Sounds like we need a revision |
| 15:42.58 | *** join/#asterisk Hurky (~ykruH@175.50.60.213.static.mundo-r.com) |
| 15:43.07 | Hurky | Hi |
| 15:43.08 | niekie | drmessano: heh. |
| 15:43.35 | Hurky | anybody tryed time based includes ? |
| 15:44.17 | Hurky | include = DID_trunk_1_timeinterval_oficina-tarde|16:30-19:30|mon-fri|*|* |
| 15:44.30 | Bartockbatz | anyone?? |
| 15:45.13 | Hurky | something must be wrong with the time, cause it does not work, but if I replace 16:30-19:30 with an * it will |
| 15:45.26 | drmessano | niekie: Also see the next paragraph on "CallerID Spoofing", which is inaccurate. Again we're working off the assumption that some DID I purchased can and should be tied to my termination.. which is a wrong that a lot of providers are making "right". |
| 15:47.05 | Bartockbatz | Hello folks - I am using Asterisk 1.4x and I would like to know where to look to disable the default VM messages - ie , "the user at extension 155 is not available...." |
| 15:47.07 | Bartockbatz | not looking for a hand-out - what application/part of the dialplan/config would I look into. - thanks |
| 15:47.09 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 15:47.32 | drmessano | Bartockbatz: You don't need to repeat every 8 minutes |
| 15:48.00 | niekie | Bartockbatz: http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+VoiceMail |
| 15:48.15 | Bartockbatz | Sorry - just a little impatient - I figured someone would have answered by now |
| 15:48.58 | Bartockbatz | Thank you niekie - :) |
| 15:49.02 | niekie | drmessano: *nod* |
| 15:49.25 | niekie | drmessano: my VoIP provider seems to pass the caller ID I provide on calls I originate. |
| 15:49.43 | mort_gib | Hurky: Are you jumping to another context?? |
| 15:50.58 | drmessano | niekie: There is a big difference in providers. Some are nothing more than "AT&T over SIP", charging only slightly less for calls and operating under the same primative rules and assumptions" |
| 15:51.33 | drmessano | niekie: I use Flowroute. Mitnick uses Flowroute. I could pwn Mitnick. Enuff said. |
| 15:51.44 | *** join/#asterisk oldhack (~jfincher@cpe-24-27-56-221.austin.res.rr.com) |
| 15:52.18 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 15:53.27 | niekie | drmessano: how does Flowroute bill you? E.g. what payment methods? |
| 15:54.12 | Hurky | mort_gib, yeah, but it only should jump to it at certain date |
| 15:54.38 | Hurky | it is this but it does not work as it should i think |
| 15:54.39 | Hurky | http://www.voip-info.org/wiki/view/Asterisk+tips+openhours |
| 15:55.11 | niekie | Flowroute's FAQ entry about billing doesn't seem that informative. All it says is "You'll be billed from your prepaid account credit.". It doesn't say how to top it up. |
| 15:55.20 | niekie | Oh, never mind. |
| 15:55.22 | niekie | I already found it. |
| 15:55.26 | niekie | Amazon payments :\ |
| 16:00.52 | Bartockbatz | hey - that did the trick - thanks for your time , folks. |
| 16:02.08 | *** join/#asterisk roni (~roni@190.196.71.206) |
| 16:02.50 | *** join/#asterisk Mhaddog_ (~Mhaddog@z65-50-118-232.ips.direcpath.com) |
| 16:07.27 | *** join/#asterisk t_dot_zilla (~chatzilla@firewall-a.buf.ny.i-evolve.net) |
| 16:08.18 | roni | i everybody .. i got 2 sip providers connected to my asterisk , i need one to make local calls , and the other to make international calls .. is there a way tu use the same dial number to call ? |
| 16:09.40 | roni | 9(00) international and 9 (local) , is there a way to do that ? |
| 16:10.00 | roni | sorry if my english is not clear |
| 16:10.22 | Qwell | ~book |
| 16:10.23 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 16:10.24 | [TK]D-Fender | roni: Yes... make your PATTERN for one, use provider A, and the other pattern use provider B |
| 16:10.25 | Qwell | look at pattern matching |
| 16:11.01 | drmessano | and please look into not using 9 as a dial prefix. Pattern matches are your friends. 9 is so "1997ish" |
| 16:11.13 | roni | thanks .. |
| 16:11.17 | Qwell | drmessano: I bet Mitnick uses 9 |
| 16:11.26 | *** join/#asterisk cesar_CR (~cesar@201.196.220.82) |
| 16:11.52 | drmessano | Qwell: Probably. BRB, I need to go pwn him again. |
| 16:12.10 | Qwell | Free drmessano! |
| 16:12.40 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 16:12.44 | [TK]D-Fender | HIS NAME WAS ROBERT PAULSON! |
| 16:12.57 | freezey | whos attending HOPE this year? |
| 16:13.00 | elred_ | :) |
| 16:13.29 | elred_ | freezey: liberty is when you have lost all HOPE |
| 16:13.35 | freezey | haha |
| 16:14.45 | drmessano | I get so tired of the mainstream "please, give me a book deal" "hacker" community. It's almost painful to read 2600 anymore |
| 16:14.58 | freezey | ahhh come on |
| 16:15.00 | drmessano | "Toaster heating elements: EXPOSED" |
| 16:15.06 | freezey | some of the content in 2600 is whack i will admit that |
| 16:15.12 | freezey | but the conferences are pretty cool |
| 16:15.19 | freezey | granted most of them are writing books but still |
| 16:15.35 | freezey | money is money my friend however you decide to obtain it |
| 16:15.49 | drmessano | Do we really need another whitepaper on the Pre-Vista Windows IP stack? |
| 16:15.49 | freezey | if somebody came to you today and asked you to write a book for a few million you would do it in a heartbeat |
| 16:16.27 | freezey | if it gets you rich why not? |
| 16:16.48 | freezey | you would probably spend less time in here and more time on some boat with a group of chicks |
| 16:16.59 | drmessano | Do we really need a 3 page article on using Wireshark to sniff out _____ that can be done in your first 5 mins of using the app? |
| 16:17.09 | freezey | no |
| 16:17.15 | freezey | BUT |
| 16:17.17 | freezey | money is money |
| 16:18.14 | drmessano | There's more to life than money. Give me enough to pay my bills and enjoy a semi-humble lifestyle, and I will keep my dignity |
| 16:18.32 | freezey | to each their own |
| 16:18.54 | drmessano | Writing a book on how to get the chick at Wendy's to tell me how to get a free Frosty is just pathetic. |
| 16:19.03 | *** join/#asterisk Martinblr (~Miranda@61.12.17.170) |
| 16:19.18 | Martinblr | is there any pstn tone generator device..? |
| 16:19.50 | jsgoecke | Yes |
| 16:19.52 | drmessano | Martinblr: Asterisk |
| 16:20.20 | Qwell | blue box |
| 16:20.30 | *** join/#asterisk imox1234 (~imox1234@p4FC5C519.dip0.t-ipconnect.de) |
| 16:20.57 | Martinblr | drmessano: to set in Asterisk we need some tone generator to test with different country profile |
| 16:20.59 | [TK]D-Fender | Martinblr: An ATA |
| 16:21.01 | drmessano | I would Red Box, but I scratched up the front of my BlackBerry trying to figure out where to attach the coke can tab |
| 16:21.12 | Qwell | drmessano: USB port |
| 16:21.43 | drmessano | Qwell: SOB, that's brilliant. I wasted weeks on that damn coke tab :( |
| 16:22.10 | drmessano | So do I just jam it in there? |
| 16:22.16 | Martinblr | But ATA will have all country tones? |
| 16:23.13 | [TK]D-Fender | Martinblr: Many. |
| 16:23.24 | *** join/#asterisk RobH (~robh@wikimedia/RobH) |
| 16:23.44 | *** join/#asterisk emora (~emora@213.236.9.114) |
| 16:23.46 | [TK]D-Fender | Martinblr: Of get yourself an FXS card. * does just about everything |
| 16:24.19 | *** join/#asterisk emora (~emora@213.236.9.114) |
| 16:27.53 | *** join/#asterisk Deeewayne (~dwayne@75-150-14-49-Atlanta.hfc.comcastbusiness.net) |
| 16:27.53 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 16:28.40 | *** join/#asterisk ccesario (~ccesario@189-29-61-213-ac.cpe.vivax.com.br) |
| 16:31.39 | *** join/#asterisk nolook4542 (~dsm@adsl-065-015-127-091.sip.jax.bellsouth.net) |
| 16:37.56 | *** join/#asterisk wcselby (~wcselby@216.110.88.194) |
| 16:37.57 | wcselby | o/ |
| 16:39.20 | [TK]D-Fender | \o |
| 16:39.27 | [TK]D-Fender | High-5! |
| 16:40.02 | *** join/#asterisk nolook4542 (~dsm@adsl-065-015-127-091.sip.jax.bellsouth.net) |
| 16:41.11 | wcselby | lol |
| 16:41.12 | wcselby | indeed |
| 16:41.23 | wcselby | \o/ |
| 16:41.27 | wcselby | <o> |
| 16:41.40 | wcselby | <o< |
| 16:41.48 | russellb | <PROTECTED> |
| 16:41.54 | wcselby | <PROTECTED> |
| 16:42.01 | wcselby | really bad ymca |
| 16:42.09 | *** join/#asterisk guilhermebr (~Guilherme@ns2.aser.com.br) |
| 16:42.13 | [TK]D-Fender | Lokks more like "headache" to me... |
| 16:42.20 | wcselby | [TK]D-Fender - pretty much |
| 16:42.25 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 16:43.38 | wcselby | sorta like the 266mb Master.csv file I'm trying to open in excel, that won't open because it contains more than 1048576 lines |
| 16:44.21 | p3nguin | introduces split to wcselby |
| 16:44.53 | wcselby | man split |
| 16:44.57 | wcselby | ewwww |
| 16:47.10 | drmessano | Excel? |
| 16:47.16 | drmessano | That's one of those Windows apps, right? |
| 16:47.34 | wcselby | indeed |
| 16:47.58 | p3nguin | Don't worry, OpenOffice.org has a counterpart. |
| 16:48.03 | WIMPy | From the Microsoft Notwork collection |
| 16:48.06 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 16:48.23 | wcselby | i've used openoffice before |
| 16:48.27 | wcselby | it's calc, right? |
| 16:48.32 | wcselby | or spreadsheet, or something |
| 16:48.35 | drmessano | I was reading about Office 2010.. How Excel and Project fully utilize 64-bit, but ACCESS DOESNT |
| 16:49.00 | niekie | Bartockbatz: you're welcome :) |
| 16:49.09 | drmessano | and M$ recommends that unless you REALLY need 64-bit Excel and Project to install the 32-bit version of Office |
| 16:49.16 | drmessano | IN 2010 MIND YOU |
| 16:49.31 | wcselby | sounds about right |
| 16:49.36 | drmessano | R U SIRIUS BALLMER? YES I R |
| 16:49.44 | wcselby | lol |
| 16:50.46 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 16:51.19 | *** join/#asterisk aidinb (~Aidin@ip70-187-172-87.oc.oc.cox.net) |
| 16:52.02 | drmessano | I reloaded this machine back in January with XP x64 and Ubuntu Karmic.. My first Ubuntu desktop, and so far haven't needed to boot into Windows in 3 months |
| 16:52.25 | wcselby | hmmmm, maybe I should tell this client I can't open the file in their program, so they need to go ahead and upgrade to storing their CDRs in a database, like I recommended to begin with....... |
| 16:52.54 | wcselby | drmessano - i preferred linux mint over ubuntu, it's ubuntu with all the stuff ubuntu should have shipped with but didn't |
| 16:53.14 | wcselby | although I will say back in january I reloaded my laptop with windows 7 (removing a dual boot vista/mint install) and I've loved it |
| 16:53.38 | drmessano | I've had too many issues with 7 on my work laptop |
| 16:53.38 | *** join/#asterisk nny (~Scott@cpe-071-076-058-253.sc.res.rr.com) |
| 16:54.41 | drmessano | I prefer Ubuntu over Mint.. Mint was too.. KDE |
| 16:55.01 | wcselby | ....did you install mint kde? I think default mint is gnome.... |
| 16:55.08 | nny | I had asked this question previous, and the overall response ended up stating that I would have to rewrite some code to make this happen. I am trying to eliminate or supress the "State: Ringing" from hints. Any advice greatly appreciated |
| 16:55.15 | wcselby | but then again, it's been two versions since I last installed, so who knows |
| 16:55.29 | nny | i am willing to pay someone who knows the code base to help |
| 16:55.38 | [TK]D-Fender | nny: vi chan_sip.c |
| 16:55.45 | drmessano | wcselby: I demo'ed it on a friends machine and didn't like it |
| 16:55.52 | [TK]D-Fender | nny: :) |
| 16:55.58 | nny | [TK]D-Fender: :D |
| 16:56.08 | drmessano | wcselby: I can install additional packages as needed, if that's the real motivation |
| 16:56.25 | drmessano | nano -w chan_sip.c |
| 16:56.25 | *** join/#asterisk jasonwert (~jasonwert@97-83-98-83.dhcp.trcy.mi.charter.com) |
| 16:58.54 | *** join/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 16:59.23 | nny | #define DEFAULT_NOTIFYRINGING TRUE /*!< Notify devicestate system on ringing state */ |
| 16:59.34 | nny | is this what I want to change? |
| 17:00.10 | *** join/#asterisk hugorebelo (~hugo@200-171-132-124.completo.com.br) |
| 17:03.36 | *** join/#asterisk sulex (~sulex@firewall.blindata.ch) |
| 17:04.14 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.180.70.dsl.dyn.forthnet.gr) |
| 17:07.05 | [TK]D-Fender | nny: No.This was looked at before and it was in the other sense IIRC |
| 17:07.18 | *** part/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 17:07.22 | wcselby | f'in a |
| 17:07.35 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 17:08.16 | wcselby | 1.5 million records since september of last year, this client wants to know how many times 200 separate numbers called in to the system, and went into one of the queues... |
| 17:08.20 | wcselby | for april and may |
| 17:08.31 | wcselby | and the client didn't want to pay for a database |
| 17:08.46 | wcselby | i think they're getting a database, whether they want it or not |
| 17:09.05 | [TK]D-Fender | wcselby: Remarkably easy to import into SQL |
| 17:09.27 | [TK]D-Fender | wcselby: Assuming you tagged the record in the first place |
| 17:09.31 | [TK]D-Fender | (as to hitting a queue |
| 17:09.47 | [TK]D-Fender | wcselby: because it's quite possibly not the last app they hit. |
| 17:10.05 | [TK]D-Fender | wcselby: Go look. You're either completely screwed, or doing a 1-off import. |
| 17:10.10 | wcselby | haha |
| 17:10.28 | [TK]D-Fender | wcselby: then again you could jsut as easily such it into a spreadsheet and do a lookup on it |
| 17:10.59 | [TK]D-Fender | wcselby: That's what I would do if I wasn't expecting to do this regularly... well maybe |
| 17:11.13 | [TK]D-Fender | oh wait.. 1 million records. |
| 17:11.34 | [TK]D-Fender | wcselby: Minor script... or SQL would do better... |
| 17:11.45 | [TK]D-Fender | 1.5* |
| 17:12.01 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 17:12.02 | [TK]D-Fender | wcselby: Not horrible. and hopefully "billable" |
| 17:12.19 | wcselby | definately billable :) |
| 17:13.23 | wcselby | i think i'll import the Master.csv into a mysql database, then parse through that to pull any and all references for those src numbers, then filter on date |
| 17:13.40 | wcselby | should be easy enough |
| 17:13.44 | wcselby | once it's in the database |
| 17:16.20 | nny | [TK]D-Fender: any suggestions on the nomenclature used in chan_sip.c to define ringing states? |
| 17:16.37 | nny | seems like they use the same for both hints and what we discussed before :\ |
| 17:18.35 | nny | so odd this isn't a simple flag..think of what a sidecar looks like when the dialplan rings all phones at once >< |
| 17:22.03 | *** join/#asterisk Pimmetje (~Pimmetje@83.119.156.35) |
| 17:22.33 | nny | anyone willing to help me track this down I'll be happy to paypal you some dough. |
| 17:24.27 | [TK]D-Fender | nny: look for a "NOTIFY" heading. |
| 17:24.33 | [TK]D-Fender | nny: lemme fire something up |
| 17:25.15 | pabelanger | nny: http://svnview.digium.com/svn/asterisk?revision=271868&view=revision |
| 17:26.14 | nny | pabelanger: not sure I follow. Are you saying asterisk manager is responsible for the subscribed state changes? |
| 17:26.55 | pabelanger | nny: no, you can expand on the filtering of events however. |
| 17:27.31 | nny | pabelanger: so if I was to apply this patch, how would I make it so the phones did not know of the State: Ringing |
| 17:28.15 | [TK]D-Fender | nny: Found it |
| 17:28.21 | pabelanger | nny: It wouldn't, this patch only applies to AMI. |
| 17:28.28 | nny | pabelanger: ahh |
| 17:28.32 | nny | [TK]D-Fender: damn you speedy |
| 17:29.09 | [TK]D-Fender | nny: http://pastebin.com/TEzM6Lav |
| 17:29.27 | [TK]D-Fender | nny: Just knock out that state from the case |
| 17:29.33 | daemon | hey guys my asterisk works perfectly audio quality excellent, one weird problem, first call of the day disconnects after 10-50 seconds if the voip is not used for 3-4 hours the next call placed will do the same |
| 17:30.08 | *** join/#asterisk grEvenX (~even@1mldjsj.ip.ssc.net) |
| 17:30.38 | nny | [TK]D-Fender: so pidfnote = "Ringing";   from  case AST_EXTENSION_RINGING:   ? |
| 17:32.07 | *** join/#asterisk patrick^ (~patrick_@2001:470:1d:349:219:21ff:fe4e:f5de) |
| 17:32.21 | pabelanger | daemon: NAT? |
| 17:32.21 | [TK]D-Fender | nny: switch (state) { <- under this |
| 17:32.28 | [TK]D-Fender | <PROTECTED> |
| 17:32.31 | daemon | pabelanger, the following calls are ok |
| 17:32.55 | pabelanger | daemon: Yes, but are you behind a NAT? |
| 17:33.26 | daemon | pabelanger, yes |
| 17:33.27 | *** join/#asterisk WWGD (~WWGD@208.79.14.130) |
| 17:33.51 | nny | [TK]D-Fender: ok i'll remoev both, recompile and test, thanks |
| 17:34.46 | pabelanger | daemon: Either way, we would need a SIP trace to see what is going on. I would guess something with your NAT table expiring after 3-4 hours. |
| 17:35.49 | *** join/#asterisk Alagar (~Administr@122.164.35.9) |
| 17:36.20 | daemon | pabelanger, the asterisk is on the head gateway |
| 17:36.22 | daemon | not on a nat'd box |
| 17:36.31 | daemon | even though that box does nat |
| 17:36.36 | daemon | the asterisk its self has a clear path |
| 17:37.37 | pabelanger | ~collectdebug |
| 17:37.38 | infobot | collectdebug is probably a method of collecting logs allowing others help troubleshoot an issue. Refer to doc/HOWTO_collect_debug_information.txt or http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt |
| 17:37.42 | pabelanger | daemon: ^^^ |
| 17:37.47 | daemon | ty ^_^ |
| 17:42.21 | nny | [TK]D-Fender: I removed those states, recompiled and tested. CLI shows 190@hints : SIP/190 State:Ringing Watchers 0 |
| 17:42.42 | nny | when i call it. did i miss something? I'll PB my chan_sip.c section |
| 17:43.55 | nny | [TK]D-Fender: http://pastebin.com/BCxmuKDL |
| 17:48.04 | *** join/#asterisk AlHafoudh (~AlHafoudh@158.195.218.110) |
| 17:50.12 | Katty | http://www.ustream.tv/channel/squirrel-critter-cam <- squirrely has damage to his left ear )= what shall we name him? |
| 17:50.33 | nny | Holyfield |
| 17:51.17 | nny | just need a mike tyson squirrel now |
| 17:51.32 | Katty | i don't get the holyfield reference |
| 17:51.54 | nny | Katty: http://en.wikipedia.org/wiki/Holyfield-Tyson_II |
| 17:52.10 | keith4 | "...in which Tyson infamously bit off a portion of Holyfield's ear." |
| 17:53.13 | Katty | that is an excellent name sir. |
| 17:54.53 | Katty | holyfield he shall be named! |
| 17:55.02 | nny | heh nice |
| 17:55.26 | nny | an alternative owuld have been van gogh gor your classier typers |
| 17:55.28 | nny | types* |
| 17:56.03 | Katty | oooh |
| 17:56.08 | Katty | i like that one better |
| 17:56.17 | Katty | but i wonder if sir squirrel took that ear to a local brothel |
| 17:56.23 | Katty | and if he did, which lovely lass of a squirrely got it |
| 17:57.46 | Faithful | I can not get * to answer my SPA3000. It picks the call up but the call does not pass through. |
| 17:59.44 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 18:00.31 | nny | Faithful: can you post the CLI output to pastebin? |
| 18:00.58 | [TK]D-Fender | nny: From what I saw in the code... if should prevent the SIP packet from being sent out. Is that not the functional end of your goal? |
| 18:01.08 | [TK]D-Fender | nny: You should still see the HINT track it though |
| 18:01.18 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:01.18 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 18:01.28 | nny | [TK]D-Fender: it is. I didn't see the hint track it, was wondering if I removed the appropiate offending bits |
| 18:01.53 | nny | [TK]D-Fender: is the hint changes state in CLI, but the packet is not sent, this would suffice. |
| 18:02.47 | [TK]D-Fender | nny Under this : static int transmit_state_notify(struct sip_pvt *p, int state, int full, int tim |
| 18:04.11 | [TK]D-Fender | nny: You could drop a very quick CASE / IF right at the start to do a "break;" |
| 18:04.16 | Faithful | nny, http://pastebin.com/Kk8KzwKP |
| 18:05.06 | nny | Faithful: is this vanilla asterisk? |
| 18:05.21 | Faithful | no it is a trixbox |
| 18:05.25 | nny | ahh |
| 18:05.27 | *** join/#asterisk sulex (~sulex@dynamic-adsl-78-14-171-203.clienti.tiscali.it) |
| 18:05.29 | nny | ~trixbox |
| 18:05.29 | infobot | somebody said trixbox was SH1TB0X. Basically a CRAPPY, closed-source distro. STAY AWAY! |
| 18:05.35 | nny | lol |
| 18:05.48 | drmessano | ~trashbox |
| 18:05.50 | Faithful | opensource gpl |
| 18:06.05 | nny | er I dont remember the ~command, but trixbox isn't supported here |
| 18:06.20 | [TK]D-Fender | Faithful: Trixbox FORKED FreePBX, etc. |
| 18:06.30 | Faithful | who cares? |
| 18:06.38 | Faithful | * is * |
| 18:06.49 | Faithful | it's an * issue not trixbox |
| 18:07.00 | nny | basically it's not supported here because the 90 million pounds of stuff added on top of asterisk make it very hard to diagnose issues |
| 18:07.20 | [TK]D-Fender | Faithful: Try showing a COMPLETE call. |
| 18:07.31 | [TK]D-Fender | Faithful: And * doesn't have a problem "picking up". |
| 18:07.43 | pabelanger | ~freepbx |
| 18:07.44 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 18:07.58 | nny | thanks heh |
| 18:08.07 | drmessano | [TK]D-Fender: Tarbaj the Egyptian Magician. "I punch my chest and it disappears into the crowd" |
| 18:08.25 | nny | has a new client entrenched in freepbx who wants simple changes made (DID routing, etc).. wish me luck |
| 18:08.27 | Faithful | * doesn't have a problem picking up... it is the SPA-3000 |
| 18:08.48 | Faithful | which... is a pain to configure with * as I understand it... |
| 18:09.04 | drmessano | spa-3xxx is easy to config with * |
| 18:09.16 | Faithful | the FXS port is just fine... but the PSTN-VOIP gateway is a pain. |
| 18:09.28 | drmessano | Not really, no |
| 18:09.29 | drmessano | http://www.2l2o.com/how-to/spa-3102 |
| 18:09.42 | drmessano | I wrote a guide on it, feel free to have a look |
| 18:09.44 | drmessano | it's simple |
| 18:10.01 | Faithful | Ok... I will look there... if I haven't already seen... |
| 18:10.12 | drmessano | I doubt you have if you're still having a problem |
| 18:10.37 | drmessano | That's the thing about guides. The goal is usually "end up with this thing working" |
| 18:10.52 | [TK]D-Fender | [14:08]<Faithful>which... is a pain to configure with * as I understand it... <- no |
| 18:10.53 | *** join/#asterisk cesar_CR (~cesar@201.196.220.82) |
| 18:11.07 | devdvd | anyone using x-lite to video conference with asterisk. I've tried just about everything i can think of and read but cant seem to get video to go through. I'm using asterisk 1.6.2.9. |
| 18:11.09 | devdvd | Here is a paste with entries from my sip.conf general, and the 2 extensions im calling between, cli output of the call and my extensions macro. http://pastebin.com/ZZxXw3as The problem im having is that I cant get video to display on the other end (ex. 873 calls 871 but 871 never receives video |
| 18:11.33 | devdvd | was wondering if anyone had any thoughts. |
| 18:11.40 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 18:11.42 | nny | hahaha |
| 18:11.45 | nny | shoot me |
| 18:11.59 | nny | 2:10:46 PM) nny: if I add DID routing in extensions_custom to the existing from-trunk context, will they work and presist through updates/ reloads etc? |
| 18:11.59 | nny | (2:10:58 PM) Defraz_ [~Defraz@c72co-edge-router.fuzecore.com] entered the room. |
| 18:11.59 | nny | (2:11:08 PM) fauxalliance: ?handedited @ nny |
| 18:11.59 | nny | (2:11:08 PM) FreepbxBot: nny: Das machine ist nicht fur der fingerpoken und mitzengrabben. Ist easy schnappen der springenwerk, mit spritzensparken und flitzenflamen. Ist nicht fur der wanstaseein und rubbernecken kinder. Keep das hands in der pockets. |
| 18:11.59 | drmessano | ~shoot nny |
| 18:12.00 | infobot | ACTION shoots nny in the ear with a frozen turkey cannon! |
| 18:12.05 | nny | sorry for spam |
| 18:12.17 | nny | shoot me again |
| 18:13.24 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 18:13.26 | [TK]D-Fender | reaches for Mr. Pointy |
| 18:13.33 | Faithful | drmessano, this is the * channel you shouldn't give ~freepbx help in here ;-) |
| 18:13.46 | drmessano | I wasn't |
| 18:14.28 | *** join/#asterisk spenguin[work] (~penguin@59.162.86.164) |
| 18:14.31 | spenguin[work] | test |
| 18:14.33 | spenguin[work] | ing |
| 18:14.51 | Faithful | drmessano, the howto you posted (which is excellent by the way) |
| 18:15.36 | drmessano | Faithful, contains both FreePBX GUI config and vanilla Asterisk config |
| 18:16.36 | Faithful | crazy thing is I am certified digium :) I know how to root around in there |
| 18:16.43 | [TK]D-Fender | It was more of a GTFO (over THERE) anyway... |
| 18:16.57 | [TK]D-Fender | Faithful: You shouldn't * as root. FAIL |
| 18:17.26 | Faithful | not that sort or root |
| 18:21.13 | drmessano | Certified doesn't mean "know how" |
| 18:21.25 | *** join/#asterisk guilhermebr (~Guilherme@ns2.aser.com.br) |
| 18:22.09 | *** part/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 18:22.12 | nny | [TK]D-Fender: (2:03:51 PM) [TK]D-Fender: nny: You could drop a very quick CASE / IF right at the start to do a "break;". can you elaborate? |
| 18:22.51 | pabelanger | ~dcap |
| 18:22.52 | infobot | it has been said that dcap is Digium Certified Asterisk Professional. See http://www.voip-info.org/tiki-index.php?page=Asterisk+dCAP |
| 18:23.23 | Faithful | drmessano, actuall I do know how... I was just fishing for someone who might have the heads up on it. |
| 18:23.41 | drmessano | ~msce |
| 18:23.42 | infobot | rumour has it, msce is a Minesweeper Consultant and Solitaire Expert. http://www.leftmind.net/~adb/asr/mcse.txt |
| 18:24.01 | drmessano | ~ccna |
| 18:24.02 | infobot | rumour has it, ccna is cisco certified network associate |
| 18:24.07 | drmessano | boo |
| 18:24.43 | *** join/#asterisk DrDamnit (~michael@173-165-161-161-atlanta.hfc.comcastbusiness.net) |
| 18:24.45 | drmessano | ~ccna |
| 18:24.46 | infobot | from memory, ccna is Can't Comprehend Network Administration |
| 18:25.21 | DrDamnit | Other than using file convert in the CLI, how can I convert SLN files back to WAV? |
| 18:26.14 | DrDamnit | Of course... once I give up searching google, and decide to come ask the experts, I run into this: sox -t raw -r 8000 -s -w -c 1 {inputfile}.sln {outputfile}.wav |
| 18:26.21 | DrDamnit | Is that right? |
| 18:26.26 | [TK]D-Fender | nny: Right at the start of that function copy the case that checks the state code isolating the ringing types and do a break. |
| 18:27.05 | drmessano | ~A+ |
| 18:27.06 | infobot | hmm... a+ is more like D- |
| 18:27.54 | DrDamnit | ~MCP |
| 18:35.40 | nny | [TK]D-Fender: hrmm. This looks like something I may have to find someone who is fluent in C to work on. My skills are very limited |
| 18:36.57 | [TK]D-Fender | nny: I don't do C but the syntax is close enough to PHP/Pascal that it shouldn't be a big dea for me... |
| 18:37.01 | [TK]D-Fender | deal* |
| 18:52.48 | *** join/#asterisk t_dot_zilla (~chatzilla@firewall-a.buf.ny.i-evolve.net) |
| 18:55.50 | *** join/#asterisk timholum (~chatzilla@64-91-67-5.stat.centurytel.net) |
| 18:56.21 | timholum | does anyone know of a good tutorial for editing softkeys on a polycom phone. I have been trying with no sucsess |
| 19:03.43 | Naikrovek | timholum: i believe polycom has one |
| 19:03.49 | Naikrovek | digs up URL |
| 19:05.50 | Naikrovek | believe it's in the second half of this document: http://knowledgebase.polycom.com/knowledgebase/End%20User/Tech%20Alerts/Audio/Enhanced_Feature_Keys_TB42250.pdf |
| 19:11.08 | *** join/#asterisk ddickenson (~ddickenso@166.205.11.108) |
| 19:13.25 | *** join/#asterisk niekie (~niek@CAcert/Assurer/niekie) |
| 19:20.59 | *** join/#asterisk ccesario (~ccesario@189-29-61-213-ac.cpe.vivax.com.br) |
| 19:31.25 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 19:37.17 | *** join/#asterisk timholum (~chatzilla@64-91-67-5.stat.centurytel.net) |
| 19:39.50 | *** join/#asterisk jmls (~Julian@host217-36-208-155.in-addr.btopenworld.com) |
| 19:39.56 | jmls | evening all |
| 19:40.25 | jmls | does anyone know someone I can contact in order to test a 999 service ? I *really* don't want to call and say "just testing" ... ;) |
| 19:40.34 | jmls | (sorry, 999 in the UK) |
| 19:41.02 | *** join/#asterisk daog (danolga@201.210.109.250) |
| 19:43.55 | daog | hello all please i need some idea in order to fix a problem with the disposition field for the outgoing call going to dahdi channel becouse always set and answered when the call steal in process or is not answered that issue created a cdr with billsec value that must be 0 |
| 19:44.26 | daog | the card that i have is tdm410p with 4 fxo port |
| 19:56.56 | *** join/#asterisk bio-tty (~c@109.3.34.95.customer.cdi.no) |
| 19:57.13 | bio-tty | is there a good sip client for blackberry? |
| 19:58.27 | *** join/#asterisk AlHafoudh (~AlHafoudh@158.195.218.110) |
| 19:59.23 | xuser | I doubt it. |
| 20:17.55 | Katty | well now i have van gogh AND holyfield |
| 20:18.08 | Katty | something is happening to my squirrels |
| 20:18.24 | ChannelZ | Turn off the microwave |
| 20:18.25 | Katty | http://www.ustream.tv/channel/squirrel-critter-cam <- another notched ear victim |
| 20:19.13 | *** join/#asterisk florz (nobody@2001:1a50:503c::1) |
| 20:19.15 | jblack | I hate squirrels. |
| 20:19.16 | *** part/#asterisk jblack (~jblack@pool-71-173-1-106.sctnpa.east.verizon.net) |
| 20:19.22 | *** join/#asterisk florz (nobody@2001:1a50:503c::1) |
| 20:19.53 | Katty | but they have such cute little noses |
| 20:19.57 | Katty | and they tuck their paws under |
| 20:20.35 | Katty | is it pronounced holyfield or hollyfield |
| 20:21.12 | *** join/#asterisk DelphiWorld (~Delphi@41.200.30.85) |
| 20:21.40 | pabelanger | holy* |
| 20:23.18 | Katty | k |
| 20:25.14 | DelphiWorld | hi Katty;) |
| 20:25.41 | wcselby | o/ Katty |
| 20:25.48 | wcselby | playing with furry animals again? |
| 20:26.29 | *** part/#asterisk DelphiWorld (~Delphi@41.200.30.85) |
| 20:27.54 | *** join/#asterisk lhz (~shrekz@c-dba672d5.021-158-73746f34.cust.bredbandsbolaget.se) |
| 20:28.37 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 20:28.55 | Katty | wcselby: mhmm. |
| 20:28.58 | Katty | hugs wcselby |
| 20:29.29 | bio-tty | shure theres no sip client for blackberry around? |
| 20:29.35 | bio-tty | would be so uber cool |
| 20:29.37 | Katty | i hear gizmo is |
| 20:29.45 | Katty | but i've never had much luck with it |
| 20:29.58 | Katty | most providors aren't keen on the idea of allowing SIP |
| 20:30.16 | bio-tty | provider gimme internet |
| 20:30.47 | bio-tty | internet is a service givving access and transport. |
| 20:31.02 | wcselby | a lot of cell providers will block SIP traffic on their networks though |
| 20:31.08 | bio-tty | it gives the freedom to use what the heck i want on top of that |
| 20:31.20 | wcselby | just depends on which cell provider you use |
| 20:31.21 | bio-tty | thats why pple want inet. |
| 20:31.41 | bio-tty | ah, i just thinking about home-netw |
| 20:31.55 | wcselby | bio-tty - I meant, they block SIP traffic on their data internet networks |
| 20:32.27 | wcselby | although AT&T started allowing SIP traffic late last year I think |
| 20:32.57 | bio-tty | ok. |
| 20:35.04 | timholum | Im sorry my browser crashed befor I could get the answer last time i asked but does anyone know of a good tutorial for editing softkeys on a polycom phone. I have been trying with no sucsess |
| 20:35.12 | drmessano | Gizmo on the BB isn't a real SIP client |
| 20:35.29 | drmessano | It's an XMPP client for the IM, and it bridges calls like the GV app does |
| 20:35.43 | bio-tty | can i have a xmpp server? |
| 20:35.50 | bio-tty | and set the bb to use it? |
| 20:35.51 | p3nguin | Only if you ask nicely. |
| 20:35.55 | drmessano | Of course you can |
| 20:36.02 | p3nguin | (if you ask nicely) |
| 20:36.09 | drmessano | (if you ask nicely) |
| 20:36.35 | timholum | I have been reading the manual on my phone, and they give me an idea as to what to do but I just keep failing :( |
| 20:36.36 | p3nguin | I recently deployed openfire. It's pretty simple to set up and use. |
| 20:36.55 | drmessano | I used Openfire until I realized how horrible it is |
| 20:37.00 | drmessano | Ejabberd and never looked back |
| 20:37.18 | p3nguin | I considered ejabberd, but it seemed more difficult. |
| 20:37.37 | p3nguin | I had been using jabberd, but it's a wreck. |
| 20:38.01 | p3nguin | It died on two boxes, so I had to replace it with something that worked. |
| 20:38.13 | drmessano | ejabberd is a little messy, but once you get it, it's not much more cryptic than asterisk |
| 20:38.17 | bio-tty | so bb can do jabber and xmpp to do voip with my server? |
| 20:38.28 | drmessano | bio-tty: No, |
| 20:38.30 | [TK]D-Fender | [15:03]<Naikrovek>timholum: i believe polycom has one |
| 20:38.32 | [TK]D-Fender | [15:03]* Naikrovekdigs up URL |
| 20:38.33 | [TK]D-Fender | [15:05]<Naikrovek>believe it's in the second half of this document: http://knowledgebase.polycom.com/knowledgebase/End%20User/Tech%20Alerts/Audio/Enhanced_Feature_Keys_TB42250.pdf |
| 20:38.35 | [TK]D-Fender | timholum: ^^^^^^^ |
| 20:38.40 | [TK]D-Fender | timSave the link and download it |
| 20:38.49 | drmessano | bio-tty: BB can do Jabber/XMPP CHAT |
| 20:39.02 | bio-tty | i want voip |
| 20:39.04 | drmessano | bio-tty: No BB SIP voice and no XMPP Jingle |
| 20:39.10 | drmessano | bio-tty: I want a pony |
| 20:40.04 | bio-tty | i want a rope |
| 20:40.08 | drmessano | bio-tty: This isn't McDonalds, BTW.. Questions should be in complete sentences, not orders shouted through a cartoon characters face |
| 20:40.55 | drmessano | ~now |
| 20:40.56 | infobot | rumour has it, now is a good time to tell you that I have 6 gigabytes of data |
| 20:41.13 | bio-tty | ~now |
| 20:41.14 | infobot | [now] a good time to tell you that I have 6 gigabytes of data |
| 20:45.56 | timholum | Thanks, I think I have it now, I did not have "enhanced-feature-keys" enabled |
| 20:46.43 | *** join/#asterisk g_r_eek (~g_r_eek@ppp-94-67-71-147.home.otenet.gr) |
| 20:54.20 | wcselby | any regressions with 1.4.33.1 ? |
| 20:55.07 | drmessano | Probably |
| 20:55.15 | drmessano | Nothing thats made the papers though |
| 20:55.30 | drmessano | So "No serious issues" |
| 20:55.43 | wcselby | heh |
| 20:56.08 | drmessano | You may as well have asked if there were any bugs in 1.4.33.1 |
| 20:56.13 | drmessano | "Most Likely" |
| 20:56.17 | wcselby | well, are there? |
| 20:56.20 | wcselby | :P |
| 20:56.24 | drmessano | Without a doubt |
| 20:56.24 | wcselby | sorry |
| 20:56.29 | wcselby | it's been a long week |
| 20:56.31 | p3nguin | already a bugfix release on 1.4.33, hmm? |
| 20:56.51 | wcselby | and I just found out that when I thought I was going to be done at 8 tonight, is more like 1 or 2 am tonight |
| 20:56.57 | wcselby | so I'm kinda glad I slept in a little |
| 20:57.50 | pabelanger | p3nguin: a regression with FXS |
| 21:00.43 | p3nguin | Doesn't appear to be anything that will affect me, so I guess I can forget about it. |
| 21:01.30 | *** join/#asterisk JAMMAN2110 (~James@unaffiliated/jamman2110) |
| 21:09.24 | *** join/#asterisk tacvbo (~tacvbo@187.152.96.128) |
| 21:10.31 | *** join/#asterisk wokkad (~wokka@99-6-237-5.lightspeed.rcsntx.sbcglobal.net) |
| 21:11.26 | *** join/#asterisk AlHafoudh (~AlHafoudh@adsl-dyn190.78-98-189.t-com.sk) |
| 21:12.29 | wokkad | is this a good place to ask for help with a new setup, with a novice asterisk user? |
| 21:12.40 | *** join/#asterisk AlHafoudh_ (~AlHafoudh@adsl-dyn190.78-98-189.t-com.sk) |
| 21:13.04 | [TK]D-Fender | wokkad: Depends on the question. Try to be specific |
| 21:13.27 | wokkad | asterisk 1.4.21.2 version, i have sip phones and sip trunks working for inbound and outbound calls just fine |
| 21:13.47 | wokkad | trying to get a 7960 sccp phone going... it registers, but can't call any extensions our outbound, nor can be called |
| 21:13.57 | wokkad | trying to figure out how to do some debugs,etc to resolve it |
| 21:14.22 | wokkad | using chan-sccp-b |
| 21:14.41 | wokkad | sccp show looks normal for devices, lines, etc |
| 21:15.11 | wokkad | so i think its a problem in my extensions.conf |
| 21:15.33 | [TK]D-Fender | wokkad: What do you see in CLI when you try a call out? |
| 21:16.12 | wokkad | not a lot, even when i set a core debug, but i do see some interested info from an sccp debug, let me post to pastebin |
| 21:17.25 | wokkad | http://pastebin.com/Hqts5iwT |
| 21:18.14 | wokkad | it talks about line 601-0000008 not found |
| 21:18.14 | wokkad | and the 7960's extension is 601 |
| 21:18.14 | *** join/#asterisk lanning (~lanning@208.87.235.224) |
| 21:18.14 | wokkad | trying to call ext 702 |
| 21:18.14 | wokkad | which is a sip softphone and working normally |
| 21:19.52 | [TK]D-Fender | SEP00152BFF663D: Finish to indicate state SCCP (InvalidNumber), SKINNY (Proceed) on call 601-00000008 |
| 21:19.57 | [TK]D-Fender | Invalid # it says |
| 21:20.16 | wokkad | the 601 is invalid? |
| 21:20.23 | [TK]D-Fender | -- SCCP: exten: "702" -- SCCP: context: "sccp" |
| 21:20.37 | [TK]D-Fender | wokkad: indeed check your dialplan |
| 21:20.39 | wokkad | ext 701 can dial 702 with no issues |
| 21:20.51 | [TK]D-Fender | wokkad: indeed check your dialplan <-------------------- |
| 21:21.55 | wokkad | see, this is where i'm lost... if 701 can dial 702, why can't 601, the dialplan doesn't restrict except on calls going to googlevoice |
| 21:22.24 | devdvd | wokkad: do a pastebin of your dialplan |
| 21:22.26 | *** join/#asterisk emora (~emora@213.37.33.105.dyn.user.ono.com) |
| 21:22.55 | [TK]D-Fender | wokkad: ........... |
| 21:22.58 | [TK]D-Fender | wokkad: indeed check your dialplan <-------------------- |
| 21:23.22 | wokkad | ok, another dumb question, where is that kept in /etc/asterisk? |
| 21:23.32 | [TK]D-Fender | EXTENSIONS.CONF |
| 21:23.38 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 21:23.59 | wokkad | i've only modified routes, etc through freepbx, but i'm finding that it isn't the best for indepth work |
| 21:23.59 | wcselby | except not in all caps |
| 21:24.03 | wokkad | :) |
| 21:24.10 | [TK]D-Fender | wokkad: Then indeed your context is WRONG <- |
| 21:24.18 | [TK]D-Fender | wokkad: and time for you to head to |
| 21:24.20 | [TK]D-Fender | ~freepbx |
| 21:24.21 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 21:24.43 | [TK]D-Fender | [17:20]<[TK]D-Fender>-- SCCP: exten: "702" -- SCCP: context: "sccp" <--- phones yshould point to [from-internal] |
| 21:24.56 | wokkad | gotcha, i'll muck around in there, sorry to have botthered you |
| 21:25.02 | wokkad | thanks for the pointers tho |
| 21:25.17 | p3nguin | Sounds like the default sccp.conf is being used. |
| 21:25.34 | wokkad | yeah, copy/pasted from a site that had a howto |
| 21:26.18 | [TK]D-Fender | wokkad: Don't do that with bomb-difusing instructions. |
| 21:26.20 | [TK]D-Fender | Just sayin' |
| 21:26.33 | *** join/#asterisk azlon (~demo@78.154.206.110) |
| 21:26.39 | azlon | !googlevoice |
| 21:26.41 | wokkad | sorry, just trying to learn this stuff |
| 21:26.53 | azlon | can i use my google voice account with asterisk? |
| 21:28.23 | [TK]D-Fender | azlon: http://www.google.ca/#hl=en&source=hp&q=asterisk+google+voice+howto&aq=f&aqi=&aql=&oq=&gs_rfai=&fp=6cf9f243e2b5b480 |
| 21:28.28 | [TK]D-Fender | azlon: JFGI <- |
| 21:28.43 | [TK]D-Fender | azlon: Took longer for you ask than Google to answer. |
| 21:29.23 | wokkad | [TK]D-Fender: btw, it was the context, changed it to match the other phones and it can now dial... thanks very much for the point in the right direction |
| 21:29.35 | wokkad | i'll bug the fpbx folks for any further questions |
| 21:29.44 | [TK]D-Fender | wokkad: Excellent |
| 21:37.39 | carrar | hrmm |
| 21:37.42 | carrar | Friday already |
| 21:39.42 | azlon | [TK]D-Fender, yeah, those links didnt work |
| 21:40.04 | azlon | can i even use google voice to dial out on asterisk? |
| 21:40.11 | azlon | isnt gv inbound only? |
| 21:41.17 | *** join/#asterisk bjhaid (~IceChat7@41.220.68.2) |
| 21:42.50 | [TK]D-Fender | alzYes these guides work, and there are ways of using GV for outbound IIRC |
| 21:43.14 | [TK]D-Fender | azlon: Which one of those guides even tells you haw. |
| 21:43.57 | azlon | hrmm |
| 21:44.04 | wcselby | azlon - check out the ultimate pbx from nerd vittles, they have info on using gv with asterisk |
| 21:44.07 | azlon | maybe im getting ahead of myself... i just isntalled asterisk today |
| 21:44.29 | azlon | wcselby, i was going to try that but they want me to download an iso and install it as my os... |
| 21:44.35 | carrar | maybe |
| 21:44.49 | wcselby | azlon - but if you read their website and guides, you'll get a good idea of how to do it |
| 21:45.23 | wokkad | to some extent, you have to download their scripts that do a lot of it, they don't explain a lot of what is going on |
| 21:45.30 | [TK]D-Fender | azlon: "They"? Who the hell is "they"? |
| 21:45.52 | wokkad | <PROTECTED> |
| 21:45.56 | [TK]D-Fender | Oh... Nerd Vittles SuperCrapAllInOneISO. |
| 21:45.57 | [TK]D-Fender | LOL |
| 21:46.45 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 21:47.04 | [TK]D-Fender | [17:44]<azlon>maybe im getting ahead of myself... i just isntalled asterisk today <- not that much more than anything else if you actually set up a minimal dialplan and soft-phone by now. But in vanilla *... I'm dubious |
| 21:53.35 | wokkad | is it worth starting out with something like asteriskNOW, using freepbx, or am i better off learning everything from the ground up, not relying on the gui... the conf files are a bit daunting |
| 21:55.58 | [TK]D-Fender | wokkad: Depends if yuo want actual control over what happens or if you can be happy enough hacking what you need into FreePBX's cookie cutter format |
| 21:57.08 | *** join/#asterisk troy42 (troy@fitzroy.yort.com) |
| 21:57.32 | wokkad | understood... i'm a network engineery by trade... doing cisco network and voip, cisco CM and CME is no problem for me |
| 21:57.36 | wokkad | er, engineer |
| 21:57.42 | wokkad | i'm not a typists, damnit! |
| 21:57.56 | fenrus | =) |
| 21:58.13 | wokkad | and i play with bsd/linux as a hobby, so that part is no problem... just trying to get my head around how asterisk is all tied together |
| 21:59.11 | [TK]D-Fender | wokkad: Define your goals and the means will announce themselves |
| 21:59.35 | troy42 | my voice is my passport, verify me |
| 22:01.37 | Chainsaw | troy42: You should listen to our hold music. You'll like it :D |
| 22:02.01 | troy42 | nice =) |
| 22:02.16 | wokkad | [TK]D-Fender: to learn... i'm ditching vonage at home, have asterisk in a vm at my colo |
| 22:02.28 | troy42 | our demo number uses the matrix sample ("the phone booth at..") |
| 22:02.32 | wokkad | i'll have 2 or maybe 3 phones and did's setup to learn with, and use at home |
| 22:02.55 | wokkad | i'm not sure what i'll do past that... maybe integrate it in the future at work for shits and giggles |
| 22:09.36 | wokkad | i'd like to find a really good guide... take a freshly installed asterisk... teach you the simple basics of setting up a phone, a trunk, dial out, and in, and then expand upon that with more features, etc |
| 22:09.57 | p3nguin | ~book |
| 22:09.58 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 22:10.04 | p3nguin | There it is. ^^^^ |
| 22:10.11 | wokkad | but most of the guides i've found online all point you to iso's or gui's |
| 22:10.22 | wokkad | p3nguin: thanks! |
| 22:10.41 | wokkad | i'll dump the pdf on my ereader and peruse it |
| 22:10.43 | p3nguin | Grab the PDF. Spend a few days reading through it. |
| 22:11.27 | p3nguin | I first read the book from a cold start. You've at least got a little bit of familiarity with asterisk already. |
| 22:12.22 | wokkad | yeah, lots of reading, trial and error |
| 22:12.33 | wokkad | my asterisk bookmark folder is getting quite large |
| 22:12.42 | p3nguin | It's good for you. |
| 22:12.52 | wokkad | oh yeah, i learn by tinkering with it |
| 22:17.41 | wcselby | later folks |
| 22:26.06 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 22:49.44 | wokkad | thanks again for everyone's help, have a great weekend |
| 22:49.45 | *** part/#asterisk wokkad (~wokka@99-6-237-5.lightspeed.rcsntx.sbcglobal.net) |
| 22:57.20 | *** join/#asterisk pabelanger-lap (~pabelange@CPE0013f7abc09a-CM0013f7abc096.cpe.net.cable.rogers.com) |
| 23:02.56 | *** join/#asterisk jks (jks@193.189.93.254) |
| 23:17.03 | *** join/#asterisk svm_invictvs (~patrick@unaffiliated/svminvictvs/x-938456) |
| 23:17.07 | rustyclarkson | Is it possible to track the amount of time someone is put on hold during a call? I know it's possible when put in a queue, but what about outside of a queue?? |
| 23:17.11 | svm_invictvs | Hola |
| 23:17.46 | svm_invictvs | I'm having trouble getting my asterisk installation to forward voicemail to email. Does anybody know of places I shoudl star tlooking to figure out what's going wrong? |
| 23:21.00 | p3nguin | voicemail.conf |
| 23:21.48 | [TK]D-Fender | rustyclarkson: no |
| 23:22.18 | [TK]D-Fender | svm_invictvs: Whatever MTA you are having * use |
| 23:25.32 | svm_invictvs | ssmtp |
| 23:28.15 | *** join/#asterisk cweagans (~cweagans@67.42.166.69) |
| 23:29.13 | cweagans | I have two Asterisk PBX systems behind my firewall. Right now, 5060 and all the RTP ports are forwarded to pbx1 and the calls are working, but I'm not sure what to do about the second pbx (pbx2), because obviously the ports cannot be forwarded to both machines. What's the best way to handle this? |
| 23:31.28 | jsgoecke | you may customize the second machine's asterisk/sip.conf and asterisk/rtp.conf to use alternative ports |
| 23:31.30 | jsgoecke | And then forward there |
| 23:32.15 | cweagans | jsgoecke: okay. that second machine is using speakeasy sip trunks. Will it matter what the sip port is or can I just change it on the PBX and call it good? |
| 23:39.49 | drmessano | You're only changing the LISTEN port |
| 23:40.53 | *** join/#asterisk aidinb (~Aidin@71-95-223-217.dhcp.mtpk.ca.charter.com) |
| 23:41.30 | cweagans | so do I need to tell speakeasy to signal on a different port then drmessano? |
| 23:41.36 | cweagans | or is it okay? |
| 23:42.40 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 23:43.39 | drmessano | cweagans: This is TCP/IP 101. Changing your listen port is fine. You do NOT need to tell them anything |
| 23:44.47 | cweagans | I guess I just don't understand what actually happens when a SIP call comes in. It was my understanding that the SIP Trunk provider sent a packet on port 5060 that says 'hey, you have a call coming in. do something with it' and then the call audio was handled on a different port. |
| 23:45.22 | cweagans | drmessano: so, continuing my (incorrect) understanding, if the pbx was not listening on 5060, it wouldn't get that message and the call would never happen |
| 23:46.06 | cweagans | at what point do I start being wrong there? =D |
| 23:46.41 | *** join/#asterisk sjobeck (~sjobeck@65.102.45.89) |
| 23:47.17 | sjobeck | hey, hi, all, hope all is well. may i fire off a quick question about "asterisk desktop assitant"? |
| 23:47.26 | WIMPy | cweagans: Nowhere. The point of registering a SIP account is to tell the other party where you can be reached. |
| 23:47.54 | sjobeck | I have ADA installed & seemingly running fine but it fails to originate the call. i *think* its configured correctly. just no call after user hits dial on the button on ADA> |
| 23:48.05 | cweagans | WIMPy: okay. So when my pbx registers, it will say 'I'm listening on this port. please signal me there'? |
| 23:48.36 | WIMPy | cweagans: That's the idea |
| 23:48.53 | cweagans | WIMPy: hmm. well that's easy =D |
| 23:48.56 | cweagans | WIMPy: thanks :) |
| 23:49.24 | WIMPy | And if you have a connection tracking firewall, you don't need to forward more than the SIP port, if at all. |
| 23:49.35 | drmessano | You don't need to forward the SIP port |
| 23:49.57 | drmessano | if you have a need to forward the SIP port (external clients) you can't without the RTP ports. They go hand in hand. |
| 23:50.42 | cweagans | drmessano: well, I know it didn't work when the ports weren't forwarded. |
| 23:51.09 | drmessano | cweagans: Asterisk should work fine as a CLIENT behind a NAT |
| 23:51.21 | WIMPy | Works foir me, but it depend upon your firewall. |
| 23:51.24 | drmessano | cweagans: Connection to your ISP doesn't require open ports. |
| 23:51.32 | drmessano | ITSP |
| 23:51.45 | sjobeck | any one out there get ADA working? |
| 23:52.10 | drmessano | You only open ports if your Asterisk box is being connected TO by external clients (phones, admin interfaces) |
| 23:52.31 | drmessano | and you DONT need to change the SIP and RTP ports for two Asterisk boxes behind a NAT |
| 23:53.27 | drmessano | They're connecting to your ITSP just as two phones or ATA's would, neither of which require port forwarding |
| 23:55.29 | *** join/#asterisk g_r_eek (~g_r_eek@ppp-94-67-71-147.home.otenet.gr) |
| 23:58.29 | *** join/#asterisk CoderForLife (~Miranda@cpe-174-101-155-51.cinci.res.rr.com) |