00:00.13 | seanbright | ~weather KMTN |
00:00.25 | jameswf-home | so long and thanks for all the fish |
00:02.43 | *** join/#asterisk anthm (n=anthm@72.60.151.190) |
00:03.25 | rob0 | ~weather phnl |
00:13.16 | frieze | wow, so the documentation is the polycom admin guide has no obvious relationship to the firmware files they distribute does it? |
00:13.24 | frieze | in the admin guide rather |
00:14.06 | mcab | ~weather cyvr |
00:16.15 | mcab | frieze: if you're setting up a boot server for polycoms, the best thing to do is get the firmware zipfile from your reseller, and use the configs from there as a base |
00:18.46 | *** join/#asterisk Test-TDM421BF (n=Test-TDM@62.120.56.11) |
00:18.55 | Test-TDM421BF | hi all |
00:20.19 | Test-TDM421BF | i have a problem with TDM421BF card, cant dial out! |
00:23.19 | frieze | mcab: thanks. actually have to wait for them to authrize me for downloads though. However I realized I was looking at the SIP application part number, not the phone part number. grr I think I can make the polycom website download work now |
00:30.59 | SteveTotaro | any way to make megaco h.248 work with asterisk? |
00:31.02 | SteveTotaro | ~pb |
00:31.02 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
00:32.27 | SteveTotaro | i am tryin mgcp changed the port for megaco http://pastebin.ca/1026521 |
00:38.24 | edibrac | i see these warnings in my logs "channel_find_locked: Avoided initial deadlock" -- yet i have heard no complaints .. anyone familiar with this problem? One page I googled, it said that to get a better clue, enable certain flags to make logs more verbose |
00:38.52 | SteveTotaro | thus freeswitch was born |
00:39.01 | edibrac | could the warning be an indication of a future problem or I'm wondering if it's something I can ignmore. |
00:39.05 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-254d5948dcdb7855) |
00:39.21 | SteveTotaro | read why freeswitch was created |
00:39.31 | SteveTotaro | it was all about the deadlocks baby |
00:45.03 | edibrac | i'm on 1.2.12 though - i figure newer versions will have addressed a lot of deadlock issues, as more people have submitted backtraces? |
00:45.51 | edibrac | i guess there's no way I'll know what's up, until i turn on debugging, and figure out what's going on as indicated in: http://www.voip-info.org/wiki/view/Asterisk+debugging |
00:57.11 | *** join/#asterisk moy (n=moyhu@189.169.69.205) |
00:59.18 | eric2 | what's the easiest way to compare dates in the dial plan? |
01:01.47 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
01:09.54 | edibrac | SteveTotaro: i was reading http://www.freeswitch.org/node/117 which explains a lot about deadlocks -- does this mean that given any asterisk installation, if you have enough traffic you will eventually hit a deadlock situation? |
01:11.39 | seanbright | edibrac: ideally, no. |
01:12.30 | seanbright | edibrac: i've been running 1.4 in production with 30-40 simultaneous calls for a few months with no deadlocks |
01:12.39 | seanbright | edibrac: not exactly a high traffic situation, but... |
01:13.11 | edibrac | well that's a lot more than me -- do you still get the warning "channel.c: Avoided initial deadlock" ? |
01:13.53 | seanbright | edibrac: no |
01:13.54 | *** join/#asterisk existx (i=existx@that.orgasm.made.me.s.cre.am) |
01:14.41 | edibrac | ah wait, i guess i still have to turn on debugging to see what it might be. and i'm on a different version. |
01:14.51 | seanbright | edibrac: 1.2? |
01:14.56 | edibrac | or i can ignore this all and pretend i didn't see it |
01:14.58 | edibrac | yeah |
01:15.55 | seanbright | i've worked with 1.2 and 1.4, and i've had more lock contention problems with 1.2 |
01:16.16 | seanbright | 1.2 was in a higher volume environment, however. |
01:16.28 | seanbright | but they have since upgraded to the latest 1.4, and have no problems at all |
01:17.21 | seanbright | and if you are seeing 'Avoided initial deadlock' you are in much better shape than if you were actually deadlocked |
01:18.04 | edibrac | ha - yeah. i'm just combing through the logs so i know what i'm up against |
01:21.53 | *** join/#asterisk Frogzoo (n=Frogzoo@124.184.33.9) |
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01:44.18 | *** mode/#asterisk [+o russellb] by ChanServ |
01:48.46 | *** join/#asterisk sumodat (n=fred@CPE0004dc0cb5b2-CM0014e8271e96.cpe.net.cable.rogers.com) |
01:49.26 | sumodat | test |
01:50.21 | sumodat | join #asterisknow |
01:50.44 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
01:50.44 | *** mode/#asterisk [+o russellb] by ChanServ |
01:54.06 | sumodat | can anyone take a cisco question |
01:54.14 | *** join/#asterisk murdock_ut (n=chatzill@c-98-202-154-226.hsd1.ut.comcast.net) |
01:54.56 | Strom_M | i can take it as hard as you can give it |
01:55.16 | sumodat | alrighty then |
01:55.30 | sumodat | just installed asterisknow (new to this) and I have a 1721 |
01:55.47 | sumodat | can't figure out how to get them talking |
01:55.54 | sumodat | just looking for someone to point me in the right direction |
01:56.07 | Strom_M | what protocol is the 1721 talking? |
01:56.14 | sumodat | sip I believe |
01:56.27 | sumodat | brand new config |
01:56.43 | SteveTotaro | h.248 megaco |
01:56.45 | sumodat | so whatever I tell it |
01:57.20 | sumodat | 1721 has 2-fxs card and 2-fxo card |
01:58.07 | Strom_M | oh, it's a 1721 router |
01:58.12 | sumodat | yessir |
01:58.55 | *** join/#asterisk mihinomenest (n=argh@cerebus.clandestineresearch.com) |
01:59.52 | Strom_M | god, its been forever since i dicked with my 1720 |
02:00.06 | sumodat | yah well, I'm taking the economical route |
02:00.44 | sumodat | figured it would be a good way to get started |
02:01.36 | sumodat | where would I tell asterisk about connecting to the 1721 |
02:04.17 | *** join/#asterisk drmessano (n=nonya@c-76-125-26-150.hsd1.ga.comcast.net) |
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02:05.08 | [TK]D-Fender | sumodat, sip.conf |
02:05.30 | sumodat | thx |
02:11.43 | *** join/#asterisk fish-bulb (n=cstewart@216.207.245.1) |
02:11.46 | *** join/#asterisk tapirkopi (n=denmasbo@202.154.57.15) |
02:12.19 | tapirkopi | hello.... |
02:14.02 | tapirkopi | hi is there anyone have success load balancing with asterisk ? |
02:14.53 | [TK]D-Fender | tapirkopi, usually SER or something similar sits in front and handles that. |
02:15.30 | SteveTotaro | openser |
02:15.38 | tapirkopi | yeah...i find the solution using oepnser with asterisk |
02:15.51 | tapirkopi | something like this : |
02:16.00 | SteveTotaro | http://www.jeremy-mcnamara.com/2007/03/28/how-to-configure-openser-sip-registar-sip-proxy-and-far-end-nat-traversal-for-media/ |
02:16.14 | SteveTotaro | it will give you a basic setup |
02:16.24 | tapirkopi | [SIP_Client]---------------------------------------->[OpenSER]-----------------------[Asterisk]-------------------- |
02:16.40 | SteveTotaro | http://www.openser.org/docs/modules/1.2.x/dispatcher.html then check this for load balancing and failover |
02:16.54 | tapirkopi | yes steve |
02:17.04 | tapirkopi | like i say yesterday |
02:17.15 | tapirkopi | i've got problem "loop detected" |
02:17.39 | tapirkopi | and then you suggest me to add "canreinvite=no" |
02:17.45 | SteveTotaro | oh, canreinvite=no did not work |
02:18.13 | tapirkopi | yes did not work |
02:18.33 | SteveTotaro | it should |
02:18.41 | SteveTotaro | maybe openser is setup incorrectly |
02:19.00 | tapirkopi | i've already sent my config to your email |
02:19.07 | tapirkopi | if you don't mind |
02:19.13 | tapirkopi | you could look for it |
02:19.20 | *** join/#asterisk mackes (n=root@cpe-24-198-43-238.buffalo.res.rr.com) |
02:19.40 | SteveTotaro | the link you posted was if a call came from pstn -------------> asterisk ------------------> OpenSER ------------------ back to asterisk -------------------> back to pstn |
02:21.05 | tapirkopi | actually i just want to forward the REGISTER / INVITE to asterisk |
02:21.08 | SteveTotaro | what are .69 and .70? |
02:21.28 | SteveTotaro | .70 is openser |
02:21.32 | *** join/#asterisk drmessano (n=nonya@c-76-125-26-150.hsd1.ga.comcast.net) |
02:21.33 | tapirkopi | and if this success, i'll try to balance the call to other asterisk server |
02:22.06 | tapirkopi | 70 - openser |
02:22.11 | tapirkopi | 69 - asterisk |
02:23.25 | SteveTotaro | rewritehostport, are you sure that is right? |
02:24.28 | *** join/#asterisk blinky42 (n=steveb@c-71-230-47-244.hsd1.pa.comcast.net) |
02:25.39 | SteveTotaro | maybe you should pb your config taking out the IPs so someone can help, i can look at it tomorrow but it is late here |
02:26.28 | tapirkopi | i'm sory |
02:26.52 | tapirkopi | it is morning here |
02:28.31 | tapirkopi | <PROTECTED> |
02:28.51 | tapirkopi | openser : 192.168.1.70 |
02:29.09 | tapirkopi | asterisk : 192.168.1.69 |
02:29.18 | SteveTotaro | ok |
02:29.45 | SteveTotaro | use_media_proxy(); |
02:30.01 | tapirkopi | i'll try to load balance VoIP cal, register / invite message |
02:30.02 | SteveTotaro | put that between rewrite and route(1); |
02:30.17 | tapirkopi | but first, i must be able to forward the call to asterisk server |
02:31.10 | tapirkopi | ok so i must use media proxy |
02:31.49 | SteveTotaro | ~pb |
02:31.49 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
02:32.27 | tapirkopi | ok |
02:32.51 | SteveTotaro | http://pastebin.com/m7893052a |
02:33.54 | SteveTotaro | lines 14-19 i think you need to look at |
02:35.52 | tapirkopi | this is my conf |
02:35.54 | tapirkopi | http://pastebin.com/m258756fd |
02:37.48 | SteveTotaro | do you see where use_media_proxy(); is used in the example, you don't have that |
02:38.06 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
02:38.42 | tapirkopi | ok i see that |
02:38.50 | tapirkopi | i'll try that |
02:39.27 | SteveTotaro | and the example has return too. try to copy the examply closely |
02:41.46 | *** join/#asterisk docelmo (n=vircuser@h59.77.75.24.cable.rstb.cablerocket.net) |
02:42.15 | docelmo | anyone know why asterisk would not pass a "ringing" when the 'r' flag is used in the dial command? |
02:43.33 | SteveTotaro | does it pass ringing without the r flag? |
02:43.39 | docelmo | no |
02:43.51 | docelmo | kinda funky.. Its a Dell Dual Xeon 500 |
02:43.53 | SteveTotaro | DAHDI? |
02:43.57 | docelmo | I just cant remember the model |
02:43.59 | docelmo | dahdi? |
02:44.04 | SteveTotaro | zaptel? |
02:44.20 | docelmo | its TDM->SIP->Asterisk |
02:44.32 | SteveTotaro | so which direction doesn't get ringing? |
02:44.37 | docelmo | then from Asterisk->Polycom |
02:44.41 | docelmo | coming to the polycom |
02:44.45 | docelmo | from the pstn |
02:45.03 | docelmo | its weird.. I have never seen this before in my houndreds of installations of asterisk |
02:45.14 | docelmo | Im thinking its either Ubuntu 8.04 or the Dell |
02:45.22 | docelmo | Im also running Ztdummy |
02:45.38 | SteveTotaro | so what is your pstn connectivity? |
02:45.42 | docelmo | PRI |
02:45.44 | SteveTotaro | why ztdummy? |
02:45.49 | docelmo | not TDM card |
02:45.49 | SteveTotaro | what card? |
02:46.04 | docelmo | PRI via Interaction SIP Gateway |
02:46.10 | SteveTotaro | so a bogus pri |
02:46.24 | docelmo | I mean asterisk should at least be able to put the ring on the line and it cant do that |
02:47.32 | docelmo | I also tried injecting music on hold with the 'm' flag and it doesnt do that |
02:48.30 | SteveTotaro | sip.conf |
02:48.32 | SteveTotaro | ~pb |
02:48.33 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
02:48.42 | SteveTotaro | progressinband=no |
02:49.27 | SteveTotaro | http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+progressinband |
02:49.32 | SteveTotaro | play with that a bit |
02:50.49 | docelmo | I have it set to yes right now but it doesnt seem to matter. I will change to no and see what happens |
02:51.37 | SteveTotaro | have you tried no already? |
02:51.45 | docelmo | Thanks steve.. Im under the honest belief that its either the Dell or Ubuntu.. Im gonna install CentOS on it and see what happens.. if same issue time to break down and buy a box.. |
02:51.46 | docelmo | no |
02:52.02 | docelmo | I tried it w/o the directive which I believe defualts to no |
02:52.13 | *** join/#asterisk BeeBuu (n=beebuu@59.38.96.217) |
02:52.52 | BeeBuu | how can i get the num without the last digit? |
02:53.04 | docelmo | can you be more vague? |
02:53.22 | BeeBuu | ${EXTEN:0:-1}? |
02:53.50 | glaz | What number? the one calling you? |
02:54.01 | glaz | there's plenty of number in asterisk. |
02:54.16 | BeeBuu | yes,like 1234 calling me,i just want get the number 123 |
02:54.48 | glaz | (${CALLERID(num):3) |
02:55.36 | xacatecas | loves asterisk. |
02:55.43 | xacatecas | now for some sleep |
02:55.44 | SteveTotaro | <PROTECTED> |
02:55.46 | docelmo | I do when it does what I cant it to |
02:57.23 | BeeBuu | SteveTotaro: how about the callerid num longer than 3? |
02:57.35 | SteveTotaro | docelmo, funnit thing i was about to ask you if you could be more vague? |
02:57.41 | SteveTotaro | a pri is not sip |
02:57.44 | *** join/#asterisk moy (n=moyhu@189.169.69.205) |
02:57.51 | SteveTotaro | you are just buying marketing crap |
02:59.57 | BeeBuu | SteveTotaro: how about the callerid num longer than 3? |
03:00.28 | mackes | Ahhhh. The Thursday night Asterisk IRC party |
03:04.15 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
03:07.26 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
03:11.01 | *** join/#asterisk Micko113 (n=micko113@c-67-183-170-166.hsd1.wa.comcast.net) |
03:11.35 | drmessano | It's just Beebuu |
03:13.21 | BeeBuu | hi,drmessano |
03:13.37 | BeeBuu | nice to meet you. |
03:13.46 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
03:13.57 | BeeBuu | how's the day? |
03:15.09 | drmessano | It's fantastic |
03:15.20 | drmessano | Did you manage to install Asterisk yet? |
03:15.37 | BeeBuu | no. |
03:15.43 | drmessano | :( |
03:16.02 | BeeBuu | my job is not install asterisk,just using it |
03:16.16 | drmessano | Maybe when the Windows version gets more stable you can run it on Windows Vista with an X100{ |
03:16.21 | drmessano | X100p* |
03:16.51 | BeeBuu | i got your point. i don't using X100p anymore. |
03:18.10 | BeeBuu | drmessano: is there a windows version asterisk? |
03:18.25 | BeeBuu | or you are kidding me? |
03:18.33 | drmessano | Yes, there is |
03:18.39 | drmessano | Are you interested? |
03:19.03 | BeeBuu | yeah,may i get the URL from you? |
03:19.32 | drmessano | In my country, that would be considered slander, but you may google for it |
03:19.33 | BeeBuu | just for other guys... |
03:19.50 | drmessano | Not sure if you are familiar with google, but it's Lycos, but without the dog |
03:19.57 | drmessano | It's quite good, nonetheless |
03:21.03 | BeeBuu | is asteriskwin32? |
03:21.28 | BeeBuu | drmessano: thanks .you are a good man. |
03:21.47 | drmessano | Yes, you are quite welcome.. |
03:22.31 | BeeBuu | drmessano: i know i am not lovly,but you still help me alot... |
03:23.12 | drmessano | I do what I can |
03:23.20 | drmessano | ~drmessano |
03:23.20 | jbot | [drmessano] the leading cause of censorship in #asterisk, maybe a Doctor, not really a mess um a no..... The Doctor is accepting new patients, <drmessano> I am not a OB/GYN but I'll have a look anyway, and is earning his reputation daily |
03:23.28 | drmessano | Indeed |
03:23.52 | *** join/#asterisk techie (n=techie@adsl-76-214-30-46.dsl.lsan03.sbcglobal.net) |
03:24.07 | BeeBuu | :-) |
03:24.30 | BeeBuu | Dr. messano? |
03:37.27 | *** join/#asterisk PepOSX (n=angeldav@200.93.28.168) |
03:38.40 | *** join/#asterisk d00gster (n=doughant@bas1-cooksville01-1279552398.dsl.bell.ca) |
03:42.15 | rbd | hey guys, is it safe to use sleep/nanosleep inside of asterisk applications app_*.c ? |
03:42.31 | tzanger | that doesn't seem like a good idea to me |
03:43.16 | rbd | tzanger: yeah...I do see calls to ast_safe_sleep |
03:43.23 | rbd | that's probably the better one to do |
03:45.04 | *** join/#asterisk Yourname` (n=chatzill@unaffiliated/yourname/x-837320) |
03:45.20 | *** join/#asterisk AndyGraybeal (n=AndyGray@128-177-27-78.ip.openhosting.com) |
03:46.00 | TJNII | Nonono, go ahead and use nanosleep and you can send users off on wild goose chases trying to track down the source of lag in their networks when it is really just asterisk stopping. |
03:46.33 | TJNII | (even though that probably wouldn't happen) |
03:50.47 | BeeBuu | DR.messano: how can i make slient when dial? |
03:56.17 | *** join/#asterisk SomethingISODD (n=dancole@S010600a0d1757bfb.cg.shawcable.net) |
03:56.24 | SomethingISODD | Hello all |
03:56.47 | SomethingISODD | question does anyone know where i can find a php script thats been developed to monitor traffic. how long they are connected etc? |
03:57.14 | *** join/#asterisk killmel8tr (n=IceChat7@c-69-244-155-174.hsd1.mi.comcast.net) |
03:57.17 | TJNII | Realtime or as a log reader? |
03:57.39 | *** join/#asterisk mpruett (n=mpruett@24-240-203-84.static.stls.mo.charter.com) |
03:57.58 | SomethingISODD | TJNII realtime |
03:58.03 | SomethingISODD | like through the manager interface |
03:58.14 | TJNII | .....Manager interface? |
03:58.29 | SomethingISODD | ya manager.. |
03:58.44 | TJNII | You're using a GUI, arn't you? |
03:58.58 | SomethingISODD | no right now i do all my asterisk stuff through the configs. |
03:59.29 | SomethingISODD | I have been trying to write a script for it in php but i cant get it to print out correctly it all comes out as a big mess |
03:59.36 | TJNII | Heh |
03:59.40 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
03:59.46 | TJNII | How are you gathering the info? |
04:00.00 | SomethingISODD | do you know php? if so i could post my script up |
04:00.25 | TJNII | I'm not an expert, but I get by |
04:00.30 | SomethingISODD | ok thanks second |
04:01.04 | mpruett | Any ideas guys (not sure if you need more info - i have it if needed) - I am trying to connect to a remote PBX and everything seems to be working fine. Calls in and out but... |
04:01.37 | TJNII | (pause for suspense) |
04:01.59 | mpruett | when I do a wireshark I see the remote PBX send " SIP Request: OPTIONS sip:XXX.XXX.XXX.XXX" |
04:02.10 | mpruett | my box replies 404 Not found |
04:02.28 | mpruett | ;) - I like suspense! |
04:02.35 | TJNII | (The audience gasps) |
04:02.41 | mpruett | lol |
04:04.13 | mpruett | Any ideas? If more info is needed just let me know what you need. |
04:04.18 | SomethingISODD | TJNII http://pastebin.com/d1769d5e8 |
04:06.39 | *** join/#asterisk mackes (n=root@cpe-24-198-43-238.buffalo.res.rr.com) |
04:07.50 | TJNII | SomethingISODD: Let me tinker with it and get back to you..... |
04:09.02 | SomethingISODD | ok thanks TJNII |
04:10.07 | *** join/#asterisk xpeed (n=unknown@unaffiliated/xt-9) |
04:10.30 | SomethingISODD | mpruett if you dont mind me asking what is wireshark? |
04:11.30 | mpruett | Network Analyzer - http://www.wireshark.org/ |
04:11.46 | mpruett | Previously ethereal |
04:11.56 | SomethingISODD | ohhh ok |
04:12.11 | SomethingISODD | i was wondering why i could never find the ethereal packages lol |
04:13.58 | jaytee | wireshark rocks! |
04:14.56 | jaytee | I used it to good advantage yesterday to monitor my traffice between sipX and Exchange 2007 Unified Messaging. |
04:16.07 | SomethingISODD | jaytee can you run it via command line or only through gui? |
04:16.10 | jaytee | I've even got call redirection working from Exchange UM to * clients |
04:17.12 | jaytee | I use a gui because I'm running it on a Windows Server 2003 64 bit Enterprise Edition. |
04:18.09 | SomethingISODD | oh ok. |
04:19.53 | jaytee | so I use a combination of sip debug on the console through an ssh session using Putty, wireshark for Windows, wireshark for linux on RHEL5 64 bit running * 1.4 and sipX running in a CentOS 5.1 VM |
04:22.40 | SomethingISODD | i am trying to build a web application for asterisk where i can add tracing tools like wireshark |
04:22.49 | TJNII | SomethingISODD: Output looks OK on my system..... |
04:23.03 | SomethingISODD | TJNII did you view it through the website? |
04:23.09 | SomethingISODD | or just through command line |
04:23.17 | TJNII | I did throw in a if($socket == FALSE) { echo "fsockopen failed: [" . $errno . "] " . $errstr . "\n"; exit(1);} though |
04:23.21 | TJNII | CLI |
04:23.39 | *** join/#asterisk mpruett (n=mpruett@24-240-203-84.static.stls.mo.charter.com) |
04:23.41 | SomethingISODD | try it from a website and you will see what i mean |
04:23.45 | SomethingISODD | its a nasty mess |
04:23.46 | TJNII | It would probably view odd on a website since there is no HTML |
04:23.50 | TJNII | Okay, one sec |
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04:28.28 | TJNII | SomethingISODD: Yea, the problem is no HTML |
04:28.35 | TJNII | Let me throw a little in there.... |
04:29.02 | SomethingISODD | thank you very much TJNII i have been trying with html but i dont know array very well so i think that could be part of the problem |
04:29.21 | km2 | my * won't start because i took out the PRI card, so it's complaining about lack of zap channels, but i need it to run for another reason. thoughts just to get it limping at least? |
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04:30.43 | km2 | http://pastebin.com/m502ee520 |
04:35.20 | km2 | fixed! i just commented out 'channel => 1-23' in /etc/asterisk/zapata.conf |
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04:41.29 | xpeed | :) |
04:49.29 | SomethingISODD | TJNII sorry to bug ya any luck |
04:51.21 | TJNII | Not yet |
04:51.25 | TJNII | Was distracted |
04:51.39 | SomethingISODD | ok np :-) |
04:52.35 | mackes | ~mackes |
04:52.35 | jbot | http://www.youtube.com/watch?v=P7v7uBA6LW8 |
04:53.33 | mackes | ~mackes |
04:53.35 | CCFL_Man2 | SomethingISODD: something is odd |
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04:55.12 | CCFL_Man2 | from a cisco sip phone to a cisco sip gateway terminated to T1 which is terminated to a channel bank which terminates a fxs station, i get an echo if i yell, and the echo sound like crap |
05:01.02 | SomethingISODD | question in sip, how many host= can you have under one singal account? |
05:01.20 | *** part/#asterisk bagc82 (n=unknown@200.114.59.173) |
05:05.19 | SomethingISODD | CCFL_Man2 did you try to adjust echo training? |
05:09.46 | TJNII | SomethingISODD: Well, I need to go to bed. You're getting the information, but you need to parse it |
05:09.58 | TJNII | I'd suggest not keying the end of your read on EOF |
05:10.05 | TJNII | And reading between commands. |
05:10.12 | CCFL_Man2 | well, i guess the first step is trying to figure out where the echo is from |
05:10.37 | SomethingISODD | i tried that didnt spit out correct |
05:11.05 | SomethingISODD | CCFL_Man2 i cant help you there i dont run asterisk for stuff like that |
05:11.13 | TJNII | Yea, because you can't key on EOF between commands as the socket is still open |
05:11.16 | SomethingISODD | i use class 5 switchs for stuff like that |
05:11.32 | SomethingISODD | TJNII ya |
05:11.43 | TJNII | Perhaps using something like socket_read or watching for newlines in the while loop |
05:12.05 | SomethingISODD | ok let me read up on sockets a bit more |
05:12.16 | SomethingISODD | thanks TJNII for your recommendations |
05:12.28 | SomethingISODD | and ur time |
05:12.35 | TJNII | NP |
05:12.44 | TJNII | I'll have to look into that manager interface |
05:12.47 | TJNII | Looks handy |
05:12.54 | SomethingISODD | it is :-) |
05:13.02 | SomethingISODD | i do alot of my commands through that |
05:13.30 | SomethingISODD | i will be releasing my scrip to the asterisk community once its done |
05:13.40 | SomethingISODD | i believe it will be one of the most advanced setups |
05:13.50 | CCFL_Man2 | SomethingISODD: there is no asterisk in the system, there is a pots phone to fxs card on channel back, to T1 from channel bank to cisco voice gateway, to voip dial peer to sip phone, i suppose the echo is in the T1 interface? |
05:14.33 | SomethingISODD | ya i would think so trying connecting your phone direct to that device |
05:14.54 | SomethingISODD | basically go backwards from your device one no device each test. and see when you find the echo |
05:15.15 | SomethingISODD | i am suspecting it will be on your t1 interface as thats where i found it on our ds3`s |
05:17.45 | CCFL_Man2 | ahh |
05:17.50 | CCFL_Man2 | thanks |
05:19.19 | SomethingISODD | no==new btw sorry tomany things on the go and i am all fsked up because of this php script |
05:19.24 | SomethingISODD | put me behind two weeks so far lol |
05:20.13 | CCFL_Man2 | lol |
05:21.18 | SomethingISODD | you dont happen to know php do you |
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05:41.57 | vector | SomethingISODD, I'm up late and know php if you want some help |
05:43.29 | SomethingISODD | vector i would love some help its killing me lol here is the script http://pastebin.com/d433ce398 |
05:43.55 | SomethingISODD | the problem is i cant figure out how to get each line from the socket to print out in on a browser as a new line |
05:44.42 | vector | does it print to the browser at all? |
05:44.52 | SomethingISODD | ya let me show you that as well |
05:44.57 | SomethingISODD | http://www.airstarcommunications.com/billing/durations.php |
05:45.20 | vector | ah |
05:45.21 | vector | heh |
05:45.33 | SomethingISODD | ? |
05:45.45 | vector | <pre> |
05:45.48 | vector | try that |
05:45.55 | vector | because \n means nothing in html |
05:46.02 | SomethingISODD | ok |
05:46.18 | SomethingISODD | i dont have \n i have <br> |
05:47.04 | vector | well YOU do.. but what comes out of that socket has \n on the end of each line |
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05:47.24 | SomethingISODD | whops :-p |
05:47.29 | vector | so basically everywhere that you need it to drop to the next line but it's not showing that way... that's a \n from the socket :) |
05:47.54 | vector | other way to do it is to simply add <br> to the end of each line as you get it out of the socket |
05:48.35 | SomethingISODD | i tried that |
05:49.10 | vector | oh |
05:49.12 | vector | I see what you did |
05:49.27 | vector | when you send like "show applications" to the socket.. use \n |
05:50.22 | vector | it's the stuff comming OUT of that socket that we want to put <br> at the end of.. OR you can enclose the entire page in a <pre></pre> tag (preformatted) |
05:50.30 | vector | sorry I wasnt very clear about that |
05:50.42 | SomethingISODD | ok let me give that a shot |
05:51.08 | SomethingISODD | do i incluse pre with ""? |
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05:51.46 | SomethingISODD | thank you |
05:51.49 | SomethingISODD | one more question for you |
05:52.05 | vector | k |
05:52.40 | SomethingISODD | easy to remove the headers from it. if you go refresh the durations.php script you will see how it has asterisk call manager blah blah lol and at the end the message saying good buy |
05:53.05 | vector | (http://pastebin.com/d412997fe that's how I would use <pre> .. just FYI) |
05:53.24 | SomethingISODD | ok |
05:54.32 | vector | ok so you just want to take the part at the beginning and at the end out right? |
05:55.09 | SomethingISODD | yes. |
05:55.23 | SomethingISODD | there is alot more to it but think this will get me in the right direction. |
05:56.34 | vector | pm |
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07:17.32 | lun_ | anyone? |
07:18.21 | Strom_M | lun_: the answer is "buy more toner" |
07:18.36 | lun_ | hehe |
07:21.43 | patrick-- | Hey, can someone tell me why this call in my logfile caused the asteriskd to crash? http://phpfi.com/319019 I see its something about an unsupported format... |
07:22.19 | *** join/#asterisk LuisTorres (n=chatzill@bl6-207-60.dsl.telepac.pt) |
07:31.27 | LuisTorres | Hi |
07:31.36 | BeeBuu | hi hi |
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08:06.07 | jim_1 | Hello! |
08:06.16 | LuisTorres | Howdy |
08:07.18 | jim_1 | Asterisk is really powerfull! |
08:07.59 | LuisTorres | Sometimes after a Dial(SIP/100&SIP/101&SIP/102&SIP/103&SIP/104&SIP/105) I get "decline" messages from all the phones. Im using Polycoms ip550 .., does anyone experienced this too? |
08:08.13 | jim_1 | Who ever worked with the function Playtones ?? |
08:08.14 | jim_1 | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Playtones |
08:08.37 | jim_1 | No, LuisTorres |
08:08.42 | jim_1 | i can't help you |
08:08.52 | LuisTorres | thankyou |
08:08.53 | jim_1 | maybe set the register timeout lower |
08:09.03 | LuisTorres | its on 60secs |
08:09.07 | jim_1 | ah okay |
08:09.25 | LuisTorres | also already try with qualify =yes , but still happen |
08:09.30 | *** join/#asterisk fluff (n=dune@snowflake.fluffigt.net) |
08:09.57 | fluff | does the manager-action QueueStatus ever return a Event: QueueMemberStatus? |
08:09.57 | jim_1 | try to upgrade the firmware in your phones |
08:12.24 | jim_1 | exten => _X.,n,Playtones(ring) & Dial |
08:12.45 | jim_1 | can someone help me. I want to use Playtones and dial at the same time ... |
08:14.36 | *** part/#asterisk synthetiq (n=roger@unl201395.nl.customer.alter.net) |
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08:25.00 | BBHoss | jim_1: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Playtones |
08:26.10 | jim_1 | Thanks BBHoss, but Playtones works but before or after dialing my mobile phone number |
08:26.22 | jim_1 | i want to use it While asterisk is dialing my mobile number |
08:26.39 | jim_1 | the caller hears nothing for about 4 seconds now |
08:26.41 | BBHoss | did you read the example titled Playing tones while dialing? |
08:26.46 | jim_1 | yes |
08:26.57 | jim_1 | using & |
08:27.29 | jim_1 | i'm trying for hours now but |
08:27.43 | jim_1 | i can't solve my problem ... |
08:27.49 | BBHoss | try copy/pasting the example, see if it works |
08:28.05 | BBHoss | you have the & going to a no-op context right? |
08:28.16 | jim_1 | yes |
08:28.27 | BBHoss | what version asterisk are you running? |
08:28.31 | jim_1 | 1.2.10 |
08:28.40 | BBHoss | debian? |
08:28.48 | jim_1 | ubuntu |
08:29.09 | BBHoss | yeah, you should really upgrade to 1.4, unless you have good reasons not to |
08:29.19 | jim_1 | yeah i know but |
08:29.32 | jim_1 | i have a good reason to stay with this verison |
08:29.35 | BBHoss | that is a really old ubuntu build anyways, my ubuntu asterisk is 1.4.17 with security backports and patches |
08:29.53 | BBHoss | what would that be? |
08:30.19 | jim_1 | i have a hangup problem with asterisk versions newer than 1.2.14 |
08:30.28 | jim_1 | i found this on the internet: |
08:30.30 | jim_1 | Since version 1.2.14, * was changed so that not receiving an ACK to an OK is |
08:30.30 | jim_1 | considered a FATAL error. |
08:30.30 | jim_1 | The specific change that causes this problem is in sip_answer() in |
08:30.30 | jim_1 | chan_sip.c: |
08:30.30 | jim_1 | res = transmit_response_with_sdp(p, "200 OK", &p->initreq, 2); |
08:30.31 | jim_1 | Changing the 2 to a 1 will probably fix it. Note that this is NOT a bug in |
08:30.33 | jim_1 | * but improper implementations--either caused by latency, or a software bug |
08:30.35 | jim_1 | (not sending an ACK). Perhaps it might be beneficial to have an option in |
08:30.37 | jim_1 | sip.conf to change how * handles not receiving an ACK? I know... it's |
08:30.39 | jim_1 | someone else's problem, but might help those of us stuck with buggy |
08:30.41 | jim_1 | implementations in production environments. :) |
08:31.02 | jim_1 | Many calles a dropped when i use a newer version of asterisk |
08:31.15 | jim_1 | it's a problem with my new voip host |
08:31.36 | jim_1 | so i use this old version to "fix" it |
08:31.46 | BBHoss | im sure if it was that serious "random calls dropped" someone would have fixed it by now |
08:31.59 | BBHoss | do you have a link to a bug on the tracker? |
08:32.12 | jim_1 | i have this problem: |
08:32.12 | jim_1 | http://lists.digium.com/pipermail/asterisk-users/2007-April/184553.html |
08:33.29 | jim_1 | i don't use a billing system just a clean asterisk version |
08:34.05 | oej | jim_1: If a device is not sending an ACK, it doesn't understand the very basics of SIP and should NOT be used |
08:34.46 | jim_1 | yeah i know |
08:34.52 | jim_1 | but i can't solve the problem |
08:35.16 | jim_1 | so i need to stay with asterisk 1.2.10 |
08:35.29 | jim_1 | many calls are dropped if i don't ... |
08:35.34 | oej | They should be |
08:36.07 | jim_1 | how do you mean oej?? |
08:36.07 | BBHoss | what kind of tom-foolery are you using for a provider? |
08:36.07 | oej | At some point, you have to require a basic level of interoperability with the core protocol implemented |
08:36.44 | jim_1 | i'm using a dutch voip provider |
08:37.53 | BBHoss | no other options? its a pretty serious implementation flaw, especially if asterisk doesn't allow it. |
08:38.13 | oej | No working SIP device should allow that |
08:38.28 | oej | Could also be a problem in your nat/firewall handling |
08:38.30 | oej | So check that |
08:38.37 | oej | Before you stop working with that provider |
08:38.57 | jim_1 | no that not a nat problem |
08:38.58 | BBHoss | oej, yeah it really could be that |
08:39.16 | jim_1 | it's 100% sure not a firewall of nat problem |
08:39.26 | BBHoss | jim_1 have you tried it with a public ip, no nat involved? |
08:39.42 | jim_1 | no |
08:40.02 | jim_1 | but i never had problems with my old provider |
08:40.05 | BBHoss | jim_1 then you have one step left before you dump your provider |
08:40.16 | jim_1 | yes i know |
08:40.35 | BBHoss | speaking of debain/ubuntu: http://www.lessaid.net/fun/apt-get-wife.png |
08:42.09 | BBHoss | jim_1, also make sure you take care of that nasty openssl bug that has affected all debian/related distros |
08:42.31 | jim_1 | ok |
08:42.45 | jim_1 | i'm updating my server every week |
08:42.47 | mvanbaak | whehehe |
08:43.17 | jim_1 | BBHos do you think that it's possible to use Playtones |
08:43.25 | jim_1 | and dial a number at the same time |
08:43.33 | mvanbaak | jim_1: did you also regenerate all ssh and dsa keys ? |
08:43.39 | mvanbaak | I know I did, it was a mess |
08:43.55 | jim_1 | no i didn't |
08:44.08 | jim_1 | only apt-get update && apt-get upgrade |
08:44.31 | mvanbaak | that will only regen the hostkey |
08:44.44 | mvanbaak | you should also regen all ssl keys, ssh keys and asterisk keys |
08:44.45 | tengulre | BBHoss: LOL! that 's good ! haha.. |
08:44.47 | mvanbaak | and probably others |
08:44.49 | *** join/#asterisk gr0mit (n=tim@CSC.ge4-0-0.401.ar1.BBS1.gblx.net) |
08:44.49 | tengulre | wife |
08:45.21 | BBHoss | jim_1, it may only work in newer asterisk, but i would think it should work in 1.2. However, it looks as if its a "hack" because it uses side-effects of the dial command to work |
08:46.03 | jim_1 | i really would like to use a newer verion of asterisk but i can't ... |
08:46.19 | jim_1 | getting this error in my log: |
08:46.27 | jim_1 | maximum retries exceeded |
08:46.29 | jim_1 | and then: |
08:46.34 | jim_1 | hanging up call |
08:48.30 | BBHoss | we can't really help if its not an asterisk problem, sorry :( |
08:49.45 | jim_1 | this works but with the silence before dialing my mobile number: |
08:49.49 | jim_1 | exten => 1,3,Dial(SIP/101&SIP/106,25,Ttr) |
08:49.49 | jim_1 | exten => 1,4,Dial,SIP/UITGAANG/MY MOBILE Numer |
08:50.12 | jim_1 | BBHos thank you |
08:50.49 | BBHoss | best of luck finding another provider :) |
08:51.20 | jim_1 | yeah i know |
08:52.16 | jim_1 | my provider wants 6 euro every month for transferring calls to my mobile phone |
08:52.57 | jim_1 | they told my that they're also using asterisk |
08:53.09 | BBHoss | doubt it |
08:53.30 | BBHoss | if they are its probably ages older than yours |
08:54.23 | jim_1 | Server: Sip EXpress router (0.9.6 (i386/freebsd) |
08:54.34 | BBHoss | SERP |
08:54.39 | BBHoss | -p |
08:54.56 | jim_1 | do you know what Sip Express router is ? |
08:55.11 | BBHoss | yeah |
08:55.32 | BBHoss | something asterisk providers use to allow asterisk to scale better |
08:55.46 | jim_1 | okay |
08:55.47 | BBHoss | that version they are running is two years old |
08:56.01 | jim_1 | may that be the problem ? |
08:56.09 | BBHoss | could be |
08:56.59 | jim_1 | they told me that they are using asterisk 1.4 |
08:57.17 | jim_1 | i can see Server: Sip EXpress router (0.9.6 (i386/freebsd) when sip debug is on |
08:57.20 | BBHoss | yeah but since they are using SER as a proxy, it doesn't really matter |
08:57.29 | jim_1 | oh okay |
08:57.51 | BBHoss | what they are doing is using SER to spread load across multiple servers |
08:57.52 | mvanbaak | jim_1: what provider are you using ? |
08:58.31 | jim_1 | may i send you a pm mvanbaak? |
08:58.34 | mvanbaak | sure |
08:58.50 | mvanbaak | I'm in .nl as well, and using a bunch of dutch providers without trouble |
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09:36.23 | Uatec | morning |
09:37.01 | Uatec | can i have a voicemail notification sent to multiple email addresses? |
09:41.07 | jim_1 | hmm, Sip Express router 0.9.6 is from 2006 ... |
09:52.05 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
09:55.57 | Uatec | openser is the wya forward now |
09:56.00 | Uatec | *way |
09:56.02 | Uatec | #openser |
09:57.31 | patrick-- | ... |
09:57.51 | patrick-- | is there anyone around thats fermilliar with spandsp and rxfax? |
09:58.18 | *** join/#asterisk RoyK (n=roy@fw.fortel.no) |
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10:01.46 | mocker | Are priority labels supported in asterisk realtime? |
10:09.38 | *** part/#asterisk BeeBuu (n=beebuu@59.38.96.217) |
10:10.10 | patrick-- | mocker: explain that |
10:10.27 | patrick-- | what do you mean by "realtime"? |
10:13.51 | *** join/#asterisk Faustov (n=faustov@unaffiliated/faustov) |
10:18.29 | gr0mit | hi Faustov |
10:20.30 | LuisTorres | Hey.., any issues on the new release 1.4.19.2 ? is it safe to upgrade? |
10:23.54 | LuisTorres | lool srry didnt realize that 1.4.20.1 is out :P ... |
10:27.03 | *** join/#asterisk stimpie (n=stimpie@84-104-6-10.cable.quicknet.nl) |
10:27.35 | RoyK | http://www.lessaid.net/fun/apt-get-wife.png |
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10:33.28 | Uatec | hmm |
10:34.18 | Uatec | patrick--, maybe he means "not fake time", like not at 4.72pm on the 34th of Notbruary |
10:35.29 | creativx | Nutbruary |
10:35.50 | SteveTotaro | RoyK: funny |
10:36.21 | *** join/#asterisk _ys (i=yuri@91.151.196.254) |
10:37.11 | *** join/#asterisk Kernel_Core (n=I@85.133.155.134) |
10:37.13 | Kernel_Core | hi all |
10:37.47 | *** join/#asterisk kannan (n=admin@121.243.115.129) |
10:37.49 | Kernel_Core | when I issue "iax2 show netstats " in local I have 14% LOSS |
10:38.20 | Kernel_Core | I use IAX2 Trunk and I have the latest asterisk 1.4.20.1 with speex ! |
10:39.06 | kannan | hello , i am having problem using a Cisco phone 7960 with *. When i dial sip peers from the phone its fine, but i cannot call zap lines. The asterisk CLI is not showing anything at all. Zap calls from Grandstram / x-lite are fine |
10:39.09 | Kernel_Core | I have 0 percent Loss Packet ... ! |
10:40.38 | kannan | anyone using cisco 7960 on SIP with * ? |
10:40.58 | Kernel_Core | kannan: check your extensions.conf |
10:41.15 | kannan | Kernel_Core : i have |
10:41.30 | Kernel_Core | something is wrong there |
10:41.43 | kannan | hmm ok will re-check and get back |
10:41.56 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
10:45.24 | Uatec | can i have a voicemail notification sent to multiple email addresses? |
10:56.31 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
11:05.24 | kannan | Kernel_Core : thanks, the extensions was ok, but i had misconfigured Name in the Cisco SIP config menu on the phone |
11:05.47 | kannan | thanks |
11:05.50 | kannan | bbiab |
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12:00.02 | jeremy_g | girls plz behave yourself |
12:04.42 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
12:04.42 | *** mode/#asterisk [+o russellb] by ChanServ |
12:07.41 | *** part/#asterisk djs (n=djs@unaffiliated/djs26) |
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12:09.49 | drzed | little question on extension: how can i "exclude" a number |
12:10.41 | SteveTotaro | i trust you are pattern matching? |
12:11.17 | drzed | or is it possible to use regexp style expreession like _0[5-9]. |
12:11.41 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
12:11.58 | drzed | yes number pattern matching |
12:11.58 | SteveTotaro | are you pattern matching drzed? |
12:12.14 | SteveTotaro | i know one way |
12:12.34 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
12:13.00 | SteveTotaro | you can have a context containing the numbers you don't want to match on, and do whatever you want with them and then include another context with the pattern match you originally had |
12:14.28 | SteveTotaro | the first context will be applied and if no match it will go to the include |
12:15.01 | drzed | hm sounds like a good idea |
12:15.14 | *** part/#asterisk Oy90 (n=ivan@213.187.111.94) |
12:15.24 | SteveTotaro | disclaimer, all my info is based on 1.2 |
12:15.40 | SteveTotaro | so if 1.4 has changed this, it may not work |
12:15.59 | drzed | im also unsing 1.2 so no problem heree |
12:16.01 | drzed | -e |
12:21.51 | *** join/#asterisk shido6 (n=shido6@209.114.208.192) |
12:24.13 | LuisTorres | Hi..., Any Issues known on the 1.4.20.1? |
12:26.14 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
12:26.30 | russellb | nope, it's perfect :) |
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12:34.22 | Nugget | Zaroo bugs! |
12:35.17 | russellb | no blatant regressions that we are aware of, no. |
12:39.55 | LuisTorres | thanks |
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12:48.06 | tuxx- | heya |
12:48.37 | Nugget | moo |
12:49.06 | tuxx- | where does asterisk store the voicemail soundfiles? I can't seem to find them anywhere... I know that when u call VoiceMailmain that u can record a soundfile for your voicemail... But is there a way to store a soundfile in some dir manually? |
12:52.33 | glaz | /var/lib/asterisk/sounds iirc |
12:54.09 | rob0 | no, that should be astspooldir => /var/spool/asterisk |
12:54.28 | rob0 | (from asterisk.conf) |
12:56.07 | *** join/#asterisk fetcher (n=jnh@ip-72-55-165-168.static.privatedns.com) |
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12:57.12 | fetcher | I'm having trouble with supervised transfers between NAT'ed Polycom phones. sip.conf already has nat=yes, canreinvite=no set. Anything else to try? |
12:57.20 | fetcher | chan_sip.c:6930 get_refer_info: Supervised transfer requested, but unable to find callid '424ac69-87c12d67-641128b2@192.168.111.3'. Both legs must reside on Asterisk box to transfer at this time. |
12:57.57 | fetcher | with canreinvite=no, won't Asterisk be in the media path for any call no matter what? |
13:01.14 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
13:01.23 | tuxx- | ah nice, thanks rob0 :-) |
13:03.38 | fetcher | Did 1.2.13 have any known supervised-transfer bugs that were fixed later? |
13:03.52 | *** join/#asterisk iratik (n=itariki@12-226-116-3.client.mchsi.com) |
13:04.23 | [TK]D-Fender | fetcher: Dial options and monitoring will keep it in the path |
13:04.33 | iratik | if i wasn't that concerned with top quality .... who is the cheapest / most reliable sip/iax trunking provider ? |
13:07.01 | *** join/#asterisk eXistenZ (i=existenz@unaffiliated/existenz) |
13:07.40 | fetcher | [TK]D-Fender: hmm, just setting canreinvite=no isn't enough? |
13:08.08 | fetcher | I guess I could try option 'T' to see if that helps |
13:08.12 | [TK]D-Fender | fetcher: all of those give * reason to sit in the middle. |
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13:15.21 | patrick-- | is there anyone around thats fermilliar with spandsp and rxfax? |
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13:15.50 | *** mode/#asterisk [+o lmadsen] by ChanServ |
13:20.43 | tuxx- | hmz. when i put the files unavail.gsm and busy.gsm in the /var/spool/asterisk/voicemail/default/306/ dir it still plays the 'main' voicemail file... and i cant seem to find that either in the spool directory. is there a way in the config files that you can set a global voicemail soundfile or something like that? |
13:21.16 | yang | ~grandstream |
13:21.16 | jbot | i guess grandstream is the Yugo of VoIP hardware. Run. Run away now. |
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13:25.15 | Nugget | heh |
13:26.29 | *** join/#asterisk Sajjad_Ali_Musht (n=Sajjad_A@invite36.enst-bretagne.fr) |
13:26.51 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
13:29.04 | *** join/#asterisk a-s (n=user@89.38.174.194) |
13:29.14 | a-s | hello |
13:29.25 | a-s | does speex codec of asterisk works for somebody? |
13:29.44 | [TK]D-Fender | tuxx-: You are the one responsible for telling voicmail which recording to play in your dialplan. |
13:30.54 | a-s | Even it appears loaded in `show translations`, I always get [May 23 16:21:25] NOTICE[2535]: chan_sip.c:5500 process_sdp: No compatible codecs, not accepting this offer! |
13:30.54 | a-s | <PROTECTED> |
13:32.05 | [TK]D-Fender | a-s: And you are NOT showing us the SIP debug of your failed call attempt along with your configs. |
13:32.52 | [TK]D-Fender | a-s: So clearly you could not possibly have done something wrong and * must be broken. |
13:32.54 | NovceGuru | Hello, anybody have an opinion of the asterisk appliance? |
13:34.26 | a-s | [TK]D-Fender: :) ok, one moment please... |
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13:39.06 | glaz | NovceGuru: what do you want to know? |
13:40.43 | a-s | [TK]D-Fender: that's it |
13:40.55 | a-s | [TK]D-Fender: I enabled sip debug |
13:41.11 | a-s | however, when I call, no sip packet is transmitted |
13:41.31 | *** join/#asterisk ThaProZac (n=DumbWebb@fw.fortel.no) |
13:41.40 | [TK]D-Fender | a-s: "May 23 16:21:25] NOTICE[2535]: chan_sip.c:5500 process_sdp: No compatible codecs, not accepting this offer!" |
13:41.51 | [TK]D-Fender | a-s: how are you getting a warning without getting a packet? |
13:41.52 | a-s | *CLI> [May 23 16:36:37] NOTICE[2635]: chan_sip.c:5500 process_sdp: No compatible codecs, not accepting this offer! |
13:41.52 | a-s | <PROTECTED> |
13:42.25 | NovceGuru | glaz: if it supports presence |
13:42.39 | a-s | [TK]D-Fender: exactly. I got the NOTICE message, ans no sip packet |
13:42.49 | a-s | and |
13:42.59 | [TK]D-Fender | a-s: then debug isn't enabled properly. |
13:43.10 | [TK]D-Fender | a-s: because thats clearly a call attempt |
13:43.15 | Slashman | is there a way to use {SSHA} or {CRYPT} in the sip.conf for the secret ? |
13:43.16 | a-s | ouf! |
13:43.41 | glaz | NovceGuru: http://www.voip-info.org/wiki/view/Asterisk+presence |
13:45.15 | patrick-- | Hey all, im using SpanDSP and rxfax to receive Faxes over my mISDN channels. But in some cases the Tiff Faxes are not viewable. Could anyone think of why this is the case? |
13:45.22 | a-s | [TK]D-Fender: I did sip debug ip ... and I enabled debug |
13:45.30 | NovceGuru | I know asterisk does, but if the asterisk appliance gui had settings for it |
13:45.34 | a-s | afterwards I made the call. |
13:45.57 | *** join/#asterisk hsv-al (n=ccvp@66.0.46.210) |
13:46.01 | glaz | Slashman: you have to have mysql support, sip in mysql. |
13:46.12 | *** join/#asterisk waKKu (n=ugabuga@unaffiliated/wakku) |
13:46.33 | glaz | NovceGuru: I doubt it, I don't like/use the gui. |
13:46.53 | Slashman | glaz : I'll wait for integrating ldap support then |
13:47.24 | *** join/#asterisk railsmunky (n=nick@collaboration.capuk.org) |
13:47.29 | railsmunky | back again :) |
13:47.45 | *** join/#asterisk danlock2 (n=bean@wikipedia/danlock2) |
13:48.12 | railsmunky | i'm getting a No application 'DigitTimeout' for extension ... any ideas? |
13:49.54 | *** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek) |
13:50.21 | Zeeek | Good Morning |
13:51.10 | *** join/#asterisk Ubluzok (n=ubluzok@62.141.89.219) |
13:51.35 | *** join/#asterisk maruz (n=maumar@88-149-241-192.dynamic.ngi.it) |
13:52.15 | danlock2 | morning |
13:52.17 | a-s | [TK]D-Fender: Now it transmitted sip packets, and codecs are incompatibles... |
13:52.22 | maruz | in manager, Event: Newchannel trim 0 on front callerid number |
13:52.33 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
13:52.33 | *** mode/#asterisk [+o putnopvut] by ChanServ |
13:52.36 | maruz | CallerID: 721234567 |
13:52.46 | maruz | instead of 0721123456 |
13:53.03 | maruz | but only this Event, the others has it |
13:53.39 | maruz | can i configure asterisk to get this 0? |
13:53.57 | maruz | nationalprefix=0 doesn't fix it |
13:54.16 | hsv-al | why am I being spammed with this? |
13:54.20 | hsv-al | [May 23 08:53:32] WARNING[10879] config.c: Unknown directive '' at line 231 of /etc/asterisk/../zaptel.conf |
13:54.32 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
13:54.51 | [TK]D-Fender | a-s: "thats nice". |
13:54.58 | Zeeek | hsv-al: what is at line 231 ? |
13:54.59 | danlock2 | hsv-al: somewhere in your zaptel.conf you have something wrong. |
13:55.33 | [TK]D-Fender | I wonder why it is there ARE even that many lines in there... |
13:56.03 | a-s | [May 23 16:49:00] NOTICE[2686]: chan_sip.c:5500 process_sdp: No compatible codecs, not accepting this offer! |
13:56.03 | a-s | <PROTECTED> |
13:56.06 | Zeeek | lines are cheap |
13:56.53 | Zeeek | off the wall guess: a capture of a text that has a bunch of blank lines and a wacky character |
13:57.19 | [TK]D-Fender | a-s: Are you going to keep spamming that useless message over & over? |
13:57.38 | Zeeek | the word spam has now been overused on this channel |
13:57.51 | tzafrir_home | hsv-al, that stupid warning is because something is trying to read zaptel.conf as an asterisk config file |
13:58.10 | patrick-- | Is anyone using tx and rxfay? |
13:58.12 | patrick-- | fax* |
13:58.13 | tzafrir_home | in asterisk everything that begons with a '#' is a special directive |
13:58.26 | [TK]D-Fender | patrick--: just get to your actual question. |
13:58.44 | NovceGuru | glaz: I've always steered clear of GUIs but thought the "official" one might be better |
13:59.16 | hsv-al | tzafrir, ive been trying to make a custom addon, so i can manually specify a youtube URL |
13:59.19 | tzafrir_home | The really funny thing would be a line in the lines of zaptel.conf '#exec echo rm -rf /etc/asterisk' |
13:59.24 | [TK]D-Fender | patrick--: Maybe its just an incomplete transmission. Have you tried other viewers |
13:59.29 | Zeeek | you shouldn't use a GUI behind the wheel. CLI only |
13:59.30 | hsv-al | for hold music. |
13:59.47 | glaz | NovceGuru: I know you're a freebsd admin, I doubt you have any problems running asterisk with the CLI. |
14:00.22 | [TK]D-Fender | NovceGuru: Stop thinking and start trying. |
14:00.26 | sp00kz | the less crap on your asterisk box, the less troubleshooting you'll do :p |
14:00.35 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.207.130) |
14:00.50 | NovceGuru | [TK]D-Fender: well I was considering purchasing the * appliance but wanted to ask around a bit before dropping $1500 :P |
14:01.06 | patrick-- | [TK]D-Fender: its weird, cause it happends only now and then |
14:01.08 | Zeeek | NovceGuru: the appliance works great |
14:01.13 | patrick-- | im using the windows tiff viewer |
14:01.14 | [TK]D-Fender | NovceGuru: What are you hoping to get out of it vs building a normal server? |
14:01.22 | oej | tzafrir_home: '#exec "echo rm -rf /etc/asterisk; echo \"#include /etc/passwd\")" |
14:01.26 | [TK]D-Fender | patrick--: are all faxes bad? |
14:01.39 | Zeeek | fax should be banned forever |
14:02.05 | patrick-- | no |
14:02.24 | patrick-- | [TK]D-Fender: not all... |
14:02.40 | a-s | [TK]D-Fender: No, I won't, but please give me an idea what to do to make speex work... :( |
14:03.03 | [TK]D-Fender | a-s: pastebin the debug & your configs or stop wasting our time. |
14:03.09 | hsv-al | do I have to buy the $10 license, if I get a TDM411e? |
14:03.26 | coppice | patrick--: mISDN is bad news for anything requiring an accurate audio stream |
14:03.27 | [TK]D-Fender | patrick--: I'd bet its just a failed fax. I've gotten those on occasion. |
14:03.59 | [TK]D-Fender | hsv-al: huh? |
14:04.12 | hsv-al | http://store.digium.com/productview.php?product_code=HPECLIC |
14:04.24 | hsv-al | qwell was telling me about this, but iwas confused what he was talking about needing this or not? |
14:04.45 | hsv-al | Im buying a 411e for home use today |
14:04.54 | [TK]D-Fender | hsv-al: that card has HARDWARE EC, no need for the software EC |
14:05.22 | hsv-al | i mis=understood then, ok cool |
14:05.59 | [TK]D-Fender | hsv-al: and HPEC is free for owners of Digium cards under warranty |
14:06.36 | hsv-al | figured I get this card instead of wasting the money on some burberry ties |
14:06.37 | hsv-al | &:^) |
14:07.14 | Zeeek | [TK]D-Fender: this is for the FXO modules? |
14:07.37 | [TK]D-Fender | Zeeek: Applies to any zaptel channel |
14:08.11 | a-s | [TK]D-Fender: http://asterisk.pastebin.ca/1027079 |
14:08.47 | Zeeek | I ask because we don't have much trouble with echo on our old cards but I do not use the FXO modules I own on them |
14:09.15 | [TK]D-Fender | a-s: First, you can't spell : insercure = port, invite |
14:09.34 | [TK]D-Fender | a-s: second you didn't even SPECIFY your allowed codecs in there at all. And you didn't provide the SIP debug. |
14:09.40 | *** join/#asterisk murdock_ut (n=chatzill@70.99.184.194) |
14:10.02 | a-s | [TK]D-Fender: one moment please... |
14:10.10 | [TK]D-Fender | a-s: But before even bothering with SIP debug, you didn't bother setting your codecs in the first place. |
14:10.57 | Zeeek | why doesn't someone write a pre-scanner for the .conf files that outputs stuff like "Did you mean insecure?" |
14:11.49 | jblack | A linter for asterisk config files would be nice. |
14:11.58 | Zeeek | "You have extensions that start with 'n' - that won't cut it" |
14:12.03 | railsmunky | any ideas :) |
14:12.11 | railsmunky | Getting the same for ResponseTimeout too |
14:12.44 | Zeeek | "what's this shit about priorities not being in order the extension [sipusers] ?3 |
14:12.52 | [TK]D-Fender | railsmunky: those 2 apps were deprecated in 1.2 and removed entirely in 1.4 |
14:13.01 | railsmunky | ah :) |
14:13.07 | [TK]D-Fender | railsmunky: you're reading outdated docs |
14:13.15 | [TK]D-Fender | railsmunky: "core show function TIMEOUT" |
14:13.29 | railsmunky | [TK]D-Fender Brilliant thanks! |
14:14.12 | patrick-- | [TK]D-Fender: weve gotten lots |
14:14.42 | Zeeek | I want to know who has the biggest on the channel right now. |
14:14.51 | Zeeek | asterisk installation, I mean |
14:15.53 | Maliuta | Zeeek: size isn't everything |
14:18.10 | Zeeek | I know, mine is very small |
14:18.49 | Zeeek | but in about 100 minutes we have a live conference on asterisk and higher call volumes |
14:19.02 | Zeeek | http://x2z.eu will get you the info |
14:19.46 | a-s | [TK]D-Fender: that's it |
14:19.50 | a-s | [TK]D-Fender: look again |
14:19.52 | a-s | http://asterisk.pastebin.ca/1027084 |
14:19.52 | a-s | <PROTECTED> |
14:20.23 | [TK]D-Fender | a-s: Wheres the call debug? |
14:20.34 | a-s | I did not provide sip debug, because I succedded to make the call and to answer; my new problem is that I hear nothing :( |
14:21.04 | [TK]D-Fender | a-s: And where is your phone relative to *? |
14:21.10 | Zeeek | a-s: is there video though? |
14:21.34 | a-s | [TK]D-Fender: http://asterisk.pastebin.ca/1027089 |
14:21.47 | a-s | I put the message from sjphone too |
14:22.21 | a-s | [TK]D-Fender: what do you mean by `phone relative to *` ? |
14:22.39 | a-s | Zeeek: it's not a video call, just audio |
14:22.39 | [TK]D-Fender | a-s: what networking sits between your phone and *? |
14:22.57 | a-s | [TK]D-Fender: ah, I explain the topology: |
14:23.56 | *** join/#asterisk merkurie (n=merkurie@192.153.163.44) |
14:24.08 | a-s | PBX <- (registered to) * |
14:24.09 | a-s | * <- PHONE1 (ULAW) |
14:24.09 | a-s | * <- PHONE2 (SPEEX) |
14:24.24 | a-s | * registered to pbx |
14:24.34 | *** join/#asterisk apocn (n=apo@unaffiliated/apocn) |
14:24.36 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
14:24.36 | *** part/#asterisk apocn (n=apo@unaffiliated/apocn) |
14:24.45 | a-s | I call from phone 1 to phone2 |
14:25.02 | [TK]D-Fender | a-s: WRONG ANSWER |
14:25.24 | [TK]D-Fender | a-s: I said what NETWORKING sits between * and your phones? that means SWITHES, ROUTERS, NAT, ETC. |
14:25.25 | a-s | [TK]D-Fender: ??? what information do you need in plus ? |
14:25.36 | [TK]D-Fender | sdsfdsfdaasd |
14:26.00 | Nugget | Your hat size and your favorite color. |
14:26.18 | Zeeek | you can leave your hat on |
14:26.37 | a-s | [TK]D-Fender: aaaah! |
14:26.42 | a-s | to look.... |
14:26.57 | Maliuta | Zeeek: no, we don't like robert palmer in here ;) |
14:27.07 | Zeeek | Randy Newman |
14:27.12 | Zeeek | but we digress |
14:27.23 | [TK]D-Fender | Zeeek: Joe Cocker |
14:27.24 | Nugget | joe cocker did a cover of that song too. |
14:27.35 | Zeeek | SJ Phone is like Opera. Some people really like it. I've never gotten it to work on any system |
14:27.35 | a-s | phone 1 -> cisco -> switch |
14:27.35 | a-s | phone 2 sjphone -> my computer -> switch |
14:27.53 | [TK]D-Fender | a-s: what is "cisco:" |
14:28.07 | Zeeek | Seesco |
14:28.13 | a-s | cisco is a sip->analog converter |
14:28.29 | Maliuta | in my case the 7941 sitting on my desk |
14:28.31 | *** part/#asterisk fetcher (n=jnh@ip-72-55-165-168.static.privatedns.com) |
14:28.42 | Maliuta | cisco is all things ta all people |
14:28.50 | Zeeek | in my case, a stock that split at $100 in the heady days before the bubble burst |
14:29.11 | Maliuta | gets all zen on cisco |
14:29.18 | Zeeek | and I even own a phone since they bought linksys who bought sipura |
14:29.44 | Maliuta | my cisco has Che Guevara as the logo |
14:30.06 | Zeeek | for you big cisco phone freaks, get the second or third season of west wing |
14:30.14 | Nugget | http://macnugget.org/stuff/asterisk-cow-real.bmp :) |
14:30.26 | Maliuta | Zeeek: been there, seen that |
14:30.37 | Zeeek | Martin Sheen for prez |
14:30.50 | Maliuta | Zeeek: and did you know that NCIS uses Logitech trackballs? |
14:31.16 | Zeeek | haven't been watching yet but now that I have a proxy that works, I can download them free from Hulu |
14:31.57 | Zeeek | speaking of TV, I hope to transmit some stuff from Asterisk Tag next week, maybe Mark Spencer's keynote. Like Steve Jobs does with Apple :) |
14:32.33 | Zeeek | http://asterisktv.com/ |
14:33.07 | Zeeek | right now there's a cute model doing an asterisk install |
14:33.48 | a-s | [TK]D-Fender: I got it |
14:33.52 | Zeeek | she's actually explaining how to find a goos SIP carrier! |
14:33.58 | a-s | [TK]D-Fender: thanks a lot for help |
14:34.12 | a-s | the microphone from the computer did not work |
14:34.30 | a-s | and I heared nothing |
14:34.48 | NovceGuru | Sorry guys busy morning, [TK]D-Fender: It just seemed like a nice all in one solution and I'm a fan of embedded hardware |
14:35.03 | Zeeek | NovceGuru: the appliance works well |
14:35.14 | Zeeek | mine is voip-only |
14:37.16 | NovceGuru | hmm, i'll need some fxs ports |
14:37.39 | Zeeek | naw, just buy one of these: http://x2z.eu/h |
14:37.43 | *** join/#asterisk drehlecom (i=ircbnc@unaffiliated/drehlecom) |
14:38.01 | danlock2 | any ideas why my system would claim to see a call coming in, claim to answer it, but i just hear ringing on my phone? |
14:39.12 | Zeeek | God I have way too many phones for two people |
14:39.35 | *** join/#asterisk macros73 (n=cs@c-24-131-77-140.hsd1.pa.comcast.net) |
14:41.45 | *** join/#asterisk Kobaz (n=kobaz@its.kobaz.net) |
14:42.18 | tzanger | Zeeek: my neighbour has like 8 tvs for 2 people |
14:42.27 | Zeeek | "Now it is time for another test" I love this video |
14:42.41 | JenniferAkemi | [TK]D-Fender: do you have recommendations for IP in quebec? |
14:43.39 | [TK]D-Fender | JenniferAkemi: www.3menatwork.com |
14:44.03 | tzanger | men at work? isn't that an oxymoron? |
14:44.14 | JenniferAkemi | [TK]D-Fender: what do you like about them |
14:44.26 | sp00kz | yes it is tzanger |
14:44.41 | [TK]D-Fender | JenniferAkemi: Price/performance, and they wholesale to my ISP which I love. |
14:45.04 | [TK]D-Fender | tzanger: No, jsut an alternative-lifestyle 80's band ;) |
14:45.32 | tzanger | heh |
14:45.33 | Zeeek | it's raining! |
14:45.49 | tzanger | looks out the window |
14:45.50 | tzanger | no it's not |
14:46.03 | Zeeek | is so |
14:46.08 | JenniferAkemi | blah, no prices listed. |
14:46.27 | *** join/#asterisk rickross (n=rickross@supporter/active/rickross) |
14:46.28 | JenniferAkemi | any idea on a ballpark for 10 megs? |
14:46.50 | Zeeek | hates the endless number of voip sites with totally opaque pricing and services |
14:47.06 | [TK]D-Fender | JenniferAkemi: What sector? |
14:47.17 | JenniferAkemi | [TK]D-Fender: what do you mean? |
14:47.27 | JenniferAkemi | [TK]D-Fender: business? |
14:47.27 | [TK]D-Fender | JenniferAkemi: Geopgraphically. |
14:47.31 | JenniferAkemi | [TK]D-Fender: montreal |
14:47.38 | [TK]D-Fender | JenniferAkemi: When in town? |
14:47.40 | [TK]D-Fender | where* |
14:47.48 | JenniferAkemi | [TK]D-Fender: downtown |
14:47.57 | [TK]D-Fender | JenniferAkemi: Core = $1000, outer = 1300 +/- |
14:48.01 | Maliuta | ahh canuks |
14:48.15 | Maliuta | my parents live in Fort Macmurray |
14:48.17 | JenniferAkemi | [TK]D-Fender: are you affiliated with them? |
14:48.37 | [TK]D-Fender | JenniferAkemi: nope, jsut researched a lot for the day job. We've got a T1 I'd like to abolish. |
14:48.53 | JenniferAkemi | [TK]D-Fender: thanks |
14:49.00 | *** join/#asterisk spokra (n=spokra@host093-179-178.sea0.speakeasy.net) |
14:49.09 | JenniferAkemi | [TK]D-Fender: just wanted something to go along wiht the telus and bell quotes |
14:49.28 | *** part/#asterisk spokra (n=spokra@host093-179-178.sea0.speakeasy.net) |
14:49.34 | *** join/#asterisk spokra (n=spokra@host093-179-178.sea0.speakeasy.net) |
14:49.47 | JenniferAkemi | the telus and bell sales guys both give us great prices, then go back to draw up the contracts and are told, you can't sell it for that, you have to add X and Y and Z into the cost. it's getting quite annoying. |
14:49.51 | [TK]D-Fender | JenniferAkemi: www.colba.net |
14:50.02 | [TK]D-Fender | JenniferAkemi: They undercut the other guys. |
14:50.34 | [TK]D-Fender | JenniferAkemi: was considering them as well. I'm just outside of the core so they were better for me. Not the best customer service though... PITA to get ahold of someone IMO |
14:50.48 | [TK]D-Fender | JenniferAkemi: Bell are total fuck-offs. |
14:50.51 | JenniferAkemi | heh |
14:51.04 | [TK]D-Fender | JenniferAkemi: too me 2 weeks to get a SALES CALL. |
14:51.17 | JenniferAkemi | yeah i know what you're saying |
14:51.30 | [TK]D-Fender | JenniferAkemi: I've got fiber on-site and they can't deal for shit. |
14:51.52 | JenniferAkemi | yeah |
14:51.55 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
14:52.05 | JenniferAkemi | i'd like it to come in on fibre |
14:52.06 | Zeeek | speaking of Digium, I hope y'all will be on with us in an hour at http://x2z.eu for Mike Trest's spot on "big and fast" with asterisk ? |
14:52.59 | *** join/#asterisk rickross (n=rickross@supporter/active/rickross) |
14:53.17 | Zeeek | <PROTECTED> |
14:53.51 | JenniferAkemi | do you guys ever talk about high availability solutions on teh conference? |
14:54.14 | Zeeek | yes, we hope to. I believe that Mike, today's guest could speak to that |
14:54.28 | Zeeek | Take a look at his profile: http://mike.trest.com |
14:55.10 | Zeeek | ANyone who is ready to help the rest of us learn is welcome to speak. The conference belongs to everyone interested in VoIP and asterisk |
14:55.19 | JenniferAkemi | cool |
14:55.39 | Zeeek | starts in one hour, check it out |
14:55.48 | JenniferAkemi | i'll try to |
14:55.57 | Zeeek | http://x2z.eu (spam seems to be big today, why should I be an excaption) |
14:56.06 | Zeeek | or exception even |
14:56.22 | Maliuta | anyone in .ca have a recommendation for VoIP provider doing DID's in Alberta? |
14:57.06 | Zeeek | by the way, join #voip-users-conference if you are interested in the topics, you can follow there |
14:58.23 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
15:00.40 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
15:02.23 | *** join/#asterisk banzaika (n=banzaika@rrcs-208-105-66-210.nyc.biz.rr.com) |
15:04.51 | banzaika | anybody experiencing problems with Voicepulse service using digium hardware ? |
15:05.18 | [TK]D-Fender | Bananaskin: How many MPG should I be getting on my Segway? |
15:05.58 | Kobaz | do de do |
15:06.15 | Kobaz | okay, so i have an avaya pbx over here, just sitting minding it's own business |
15:06.28 | Kobaz | i have two fxs's configured on it, one to a phone, one to an fxo on asterisk |
15:06.36 | *** part/#asterisk imesper (n=chatzill@200.142.121.162) |
15:06.56 | *** join/#asterisk darviria (n=darviria@87-194-177-180.bethere.co.uk) |
15:06.59 | Kobaz | i call an extension mapped to the asterisk fxo, asterisk will do a pickup and play some tacks |
15:07.02 | Kobaz | tracks... |
15:07.13 | Kobaz | but as far as the avaya is concerned, the circuit is still ringing |
15:07.19 | Kobaz | *but* this doesn't happen all the time |
15:07.40 | Kobaz | one out of every 5-10 calls will just keep ringing even though asterisk picked up... |
15:07.50 | Kobaz | so... is this an asterisk issue or an avaya issue |
15:07.59 | arbuser | (or a hardware issue) |
15:08.12 | Kobaz | this happens with a digium fxo as well as a sangoma fxo |
15:08.24 | Kobaz | with either a nortel pbx in the middle, or an avaya pbx in the middle |
15:08.32 | arbuser | ah |
15:08.32 | arbuser | ok |
15:08.46 | Kobaz | i've duplicated the problem we had with the nortel with our avaya here |
15:09.02 | [TK]D-Fender | Kobaz: And if you plug a cheap-o phone in place of *? |
15:09.15 | Kobaz | good question |
15:09.47 | *** join/#asterisk moy (n=moyhu@nat/ibm/x-ee0be127875797dc) |
15:10.00 | Kobaz | i've called a single line set many times and never noticed an issue |
15:10.03 | Zeeek | Bananaskin: I never have |
15:10.36 | Kobaz | i've never specifically tried to see the behavoir on a consistant basis |
15:11.24 | [TK]D-Fender | Kobaz: On that port? Maybe the avaya port is bad |
15:12.56 | Kobaz | not specificall on that port |
15:13.06 | [TK]D-Fender | Kobaz: then go test it |
15:13.21 | Kobaz | aye |
15:14.24 | *** part/#asterisk Oy90 (n=ivan@213.187.111.94) |
15:15.48 | danlock2 | anyone know what "zt_handle_event: ring/off hook in strange state 6 on channel 1" means? |
15:16.31 | spokra | I've used them.. things worked fine. |
15:16.49 | spokra | Voicepulse that is |
15:17.27 | Zeeek | Voicepulse rocks |
15:17.36 | *** join/#asterisk waKKu (n=ugabuga@unaffiliated/wakku) |
15:17.46 | *** join/#asterisk bsdwarrior (n=mfahey@fahey.enter.net) |
15:17.55 | spokra | there api needs a little work |
15:17.57 | *** join/#asterisk bbryant (n=brett@216.207.245.1) |
15:18.10 | bsdwarrior | For someone reason the pickup group on one phone does not work. Ive tried several things. anyone have any suggestions? |
15:18.18 | Zeeek | I never used it, just the macro |
15:18.37 | Kobaz | [TK]D-Fender: but here's the weird thing, this also happened on a nortel, same exact behavoir |
15:18.48 | Kobaz | completely different hardware, etc |
15:18.58 | [TK]D-Fender | bsdwarrior: pastebin is your friend. |
15:19.02 | spokra | ah i wrote a web site to provision new numbers etc via there api.. |
15:19.05 | Kobaz | basically my question is... is there something on the asterisk side that can be adjusted |
15:19.40 | Kobaz | it seems like asterisk isn't doing a polarity switch when it's picking up the call, sort of thing |
15:19.46 | Maliuta | bsdwarrior: solar flares |
15:19.48 | Kobaz | or which ever the default behavoir of an fxo is |
15:20.10 | Kobaz | what is the default behavoir on pickup anyway? does it just go off hook? |
15:20.13 | Maliuta | bsdwarrior: based on what you have told us, definatly solar flares |
15:21.51 | Zeeek | [TK]D-Fender: no one is your friend here |
15:25.39 | *** join/#asterisk ctaloi (n=ctaloi@nat-66-218-1-233.usadatanet.com) |
15:27.03 | *** join/#asterisk drumkilla (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
15:27.03 | *** mode/#asterisk [+o drumkilla] by ChanServ |
15:28.03 | Kobaz | [TK]D-Fender: it's definitly asterisk or the server |
15:28.12 | danlock2 | *sigh* anyone ever seen this error message chan_zap.c:4135 zt_handle_event: Ring/Off-hook in strange state 6 on channel 1 |
15:28.13 | Kobaz | [TK]D-Fender: i made about 50 calls phone to phone without issue |
15:28.25 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
15:28.29 | Kobaz | [TK]D-Fender: within the first 10 of phone -> asterisk fxo, i get the problem |
15:34.39 | Zeeek | come on over to #voip-users-conference for the call today in 1/2 hour. Right now we're talking about DECT/SIP phones though |
15:35.10 | *** join/#asterisk mukudo (n=jgreen@58.251.97.17) |
15:35.23 | ddunavant | how do we list voip providers in the US? |
15:35.32 | ddunavant | i know there is a way to do it... |
15:35.33 | Kobaz | google |
15:35.40 | ddunavant | ok |
15:36.45 | Faustov | hi, is there any setting to enable indications.conf, other than having that config file in the confdir? |
15:37.32 | *** join/#asterisk sp00kz (i=ilubj00@our.government.is.in.the.dark.bz) |
15:38.19 | *** join/#asterisk sacitec (n=tobi@201.144.211.82) |
15:39.59 | sacitec | good morning, i'm working with asterisk 1.2.x and Aastra ip phones 9133i with autoanswer. Now i'm looking for a polycom solutions in audioconference ip phone models (Soundpoint 4000) |
15:40.28 | sacitec | does anyone has tried to make autoanswer on polycom soundpoint 4000/6000 ? |
15:40.28 | *** join/#asterisk legis (n=legis@unaffiliated/legis) |
15:40.58 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-94-228-tpr-esr-2.dynamic.isadsl.co.za) |
15:44.05 | *** join/#asterisk tobias (n=tobias@cpe-069-134-035-018.nc.res.rr.com) |
15:47.55 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
15:48.28 | *** join/#asterisk kannan (n=admin@121.243.115.129) |
15:48.52 | kannan | hello, back with a fresh new troubles |
15:50.17 | banzaika | is there a way to handle invalid extension/phone numbers dialed ? |
15:50.29 | banzaika | something like DIALSTATUS ? |
15:51.16 | *** join/#asterisk oej (n=olle@ns.webway.se) |
15:52.19 | maqr | btw, why's it called Comedian Mail? did someone think that would be funny? |
15:53.17 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
15:53.53 | banzaika | very |
15:54.19 | Nugget | Just be glad it's not called DAMHI for Digium Answering Machine Human Interface of something dorky like that ;) |
15:54.26 | kannan | It is doing comedy when i try to attach vm to email |
15:54.50 | Zeeek | http://x2z.eu VoIP Users Conference begins now |
15:54.54 | kannan | sendmail is fine, but i think smtp has to be run on the localhost itself? |
15:55.54 | Kobaz | maqr: and the default voices sound very condescending |
15:55.58 | *** part/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek) |
15:56.01 | danlock2 | psh... i wish i could get a call to even come through |
15:56.21 | Kobaz | there should be a "i'm better than you" track added to every prompt of the default sounds |
15:56.53 | Kobaz | the person at... 4....3...2....1... is unavailable.... because i'm better than you |
15:58.02 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584529.dsl.bell.ca) |
15:59.16 | lmadsen | tzanger: ping? |
15:59.21 | tzanger | pong |
15:59.43 | maqr | Kobaz: yeah, i'm going to have to learn how to change those |
15:59.47 | lmadsen | see privmsg |
15:59.49 | Nugget | 1 packets transmitted, 1 packets receive, 0% IRC loss |
15:59.56 | Kobaz | maqr: see vector voice |
16:02.36 | maqr | Kobaz: 'voice vector'? |
16:03.38 | Kobaz | http://voicevector.com/ |
16:03.40 | Kobaz | yeah that's it |
16:04.00 | maqr | Kobaz: i'll probably do my own, i just have to learn more about ulaw so i can optimize the sound files for it |
16:04.06 | Kobaz | well |
16:04.10 | Kobaz | the chick they have sounds really nice |
16:04.23 | maqr | lol |
16:04.23 | Kobaz | hard to beat |
16:04.29 | maqr | what a good business |
16:05.01 | maqr | should hire sexy sounding women and men with british accents and start a asterisk recording business |
16:05.22 | lmadsen | I'm not female or british.. I'm damn sexy |
16:05.27 | maqr | hmm |
16:05.36 | lmadsen | don't listen to whatever tzanger says |
16:12.09 | *** join/#asterisk oej (n=olle@ns.webway.se) |
16:18.14 | coppice | what about us men with sexy sounding british accents? |
16:19.03 | JenniferAkemi | swoon |
16:19.33 | *** join/#asterisk af_ (n=getsmart@88-149-230-31.dynamic.ngi.it) |
16:20.25 | *** join/#asterisk mking_cd (n=mking_cd@pool-72-78-183-123.phlapa.east.verizon.net) |
16:21.07 | mking_cd | has anyone else experienced a problem with chanspy which causes the audio to be out-of-sync after the spied-on caller activates hold? |
16:25.04 | *** part/#asterisk jivco (n=jivco@85.187.217.6) |
16:25.11 | *** join/#asterisk smartlu (n=seblu@fw.sj.tdf-pmm.net) |
16:25.16 | smartlu | hello |
16:26.07 | smartlu | i have a problem on my snom phone and my new asterisk installation with hints and monitoring extension |
16:26.21 | smartlu | asterisk is 1.4.20.1 |
16:26.33 | smartlu | and snom fw is 7.1.30 |
16:27.03 | *** join/#asterisk km2 (n=x@c-24-23-252-175.hsd1.ca.comcast.net) |
16:27.16 | esaym | how do you install asterisk? is it "./configure" "make menuselect" "make" "make install" or "./configure" "make menuselect" "make install"? |
16:27.31 | esaym | I guess the question is do you have to run make after you do a make menuselect or not? |
16:27.33 | smartlu | first |
16:27.37 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
16:27.56 | smartlu | to be true, i have run make -j8 |
16:28.03 | smartlu | after make menuselect |
16:28.31 | smartlu | this can be revelent ? |
16:28.52 | esaym | ok thanks |
16:31.21 | smartlu | my snom key have this config fkey7: dest <sip:it@192.168.3.253;user=phone> |
16:32.04 | smartlu | and my asterisk |
16:32.07 | smartlu | [default] |
16:32.07 | smartlu | exten => 1111,hint,SIP/test1 |
16:32.07 | smartlu | exten => 1111,1,Dial(SIP/test1) |
16:32.07 | smartlu | exten => 2222,hint,SIP/test2 |
16:32.07 | smartlu | exten => 2222,1,Dial(SIP/test2) |
16:32.08 | smartlu | exten => it,hint,SIP/test1&SIP/test2 |
16:36.07 | *** join/#asterisk isamar (n=isamar@200.254.219.17) |
16:36.09 | isamar | hi folks |
16:36.34 | isamar | need help with SET(CDR(anyfield) |
16:37.22 | isamar | I've created a new field in cdr table but cannot set with SET(CDR()) :-( |
16:43.36 | lmadsen | isamar: are you using cdr_adaptive_odbc? |
16:43.51 | lmadsen | isamar: you can't just Set(CDR(anyfield)=foo) in the DB without it |
16:44.21 | isamar | lmadsen: it doesn't work |
16:44.36 | isamar | lmadsen: I am using cdr_mysql from asterisk-addons-1.4.5 |
16:44.43 | *** join/#asterisk Dabba (n=d@ipv6.mfnx.ip6net.net) |
16:44.45 | lmadsen | that isn't cdr_adaptive_odbc |
16:45.04 | isamar | lmadsen: ok.. got you |
16:45.35 | lmadsen | cdr_adaptive_odbc is in 1.6.0-beta, or you can use the backport from http://svncommunity.digium.com/view/tilghman/branches/1.4/ |
16:45.58 | lmadsen | that's the purpose of that module -- is to allow you to set custom fields in the CDR table -- otherwise, you can't |
16:46.15 | lmadsen | use ODBC for connecting to the DB -- it's must more supported |
16:46.22 | lmadsen | goes back to work |
16:46.27 | isamar | lmadsen: okey dokey |
16:46.33 | isamar | lmadsen: thanks dude |
16:47.10 | *** join/#asterisk Trapa (n=no@207.230.238.94) |
16:47.24 | Dabba | anyone got any idea why two identical linksys phones spa-941's can only call between each other in one direction ? one peer can call another but not the other way round , three days of fiddling! |
16:48.06 | Dabba | both can call pstn via the E1 and e1>sip is ok |
16:48.57 | *** join/#asterisk LuisTorres (n=chatzill@bl6-192-3.dsl.telepac.pt) |
16:51.06 | smartlu | nobody use snom phone with this kind of setting ? |
16:52.51 | isamar | Dabba: seems to be route problem in your extensions.conf or some firewall issue |
16:53.04 | isamar | smartlu: which setting? |
16:54.07 | Dabba | isamar, not a route issue as changing the sip device to a different brand cures the problem |
16:54.17 | Trapa | Would anyone be able to look at this log and tell me if there's anything in it that would indicate a problem such that would result in calls getting dropped when the asterisk server has been up for 8 hours http://pastebin.com/d55f90534 |
16:54.32 | *** join/#asterisk jmardonesk (n=jmardone@236-166-18.adsl.din.tie.cl) |
16:55.02 | Dabba | all sip devices on same firmware and every setting in sip.conf same and in phone guis |
16:55.48 | *** join/#asterisk beek (n=klinebl@65.211.106.242) |
16:56.27 | isamar | Dabba: you need to give a look on the sip chat to see what's happenning.. |
16:56.55 | jmardonesk | hi all, I have a litle question, can make actions in the dialplan after hangup, i.e. I make a call in an analog line, then the user in the otrer side hang up, can make an post action like obtain the call duration o the start and end of the call? |
16:57.07 | [TK]D-Fender | Dabba: pastebin the CLI output of the failed attempt with SIP debug enabled |
16:57.08 | Dabba | has done sip debug on both peers but not v helpfuleg |
16:57.09 | [TK]D-Fender | ~pb |
16:57.10 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
16:57.20 | [TK]D-Fender | ^^^^^^^^^^^^^ |
16:57.20 | *** join/#asterisk mmurdock (n=chatzill@mail.kimballequipment.com) |
16:57.31 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
16:57.32 | Dabba | will do a paste debug :-) |
16:57.38 | [TK]D-Fender | jmardonesk: to do what withe xactly? |
16:57.41 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-246-236.balt.east.verizon.net) |
16:59.34 | *** join/#asterisk dlynes (n=chatzill@S01060016b68219f1.vs.shawcable.net) |
16:59.40 | dlynes | I was told that there was a version of linux called 'ATC Linux', or 'ATCLinux' that had a kernel optimised for a specific Intel chipset to run Asterisk more efficiently...would anyone happen to have a pointer to it? |
17:00.18 | jmardonesk | I was thinking in store the start/end date-time of the call, and the dialed numer in a database using dialplan and func_odbc |
17:01.48 | smartlu | isamar: setting which i have posted above |
17:02.06 | smartlu | isamar: a snom phone which monitor a sip extension |
17:02.54 | [TK]D-Fender | jmardonesk: Ever heard of this wonderful thing known as CDR? |
17:03.51 | dlynes | didn't 1.4.20 just come out? and now 1.4.20.1 is out? another iax2 security fix? |
17:04.09 | Qwell | no |
17:04.15 | [TK]D-Fender | dlynes: * 1.4.8.6.7.5.3.0.9! |
17:04.25 | [TK]D-Fender | dlynes: Get yours now! |
17:04.36 | Kobaz | 1.4.3.14.1.5.9 |
17:04.43 | jmardonesk | [TK]D-Fender, I dont know CDR, Im realy new in this, (I work with asterisk 3 years ago... but never see CDR) |
17:04.50 | dlynes | [TK]D-Fender: 1.4.20.3.1415627 |
17:04.50 | drumkilla | it's a silly console cosmetic fix |
17:04.52 | [TK]D-Fender | Kobaz: mmmm now I'm hungry again... |
17:05.03 | [TK]D-Fender | dlynes: You're a little off... |
17:05.14 | dlynes | [TK]D-Fender: rounding error |
17:05.36 | [TK]D-Fender | drumkilla: Oil of Delay ;) |
17:05.44 | Kobaz | quick quick, who can recite pi to 100 digits |
17:05.59 | [TK]D-Fender | "pi to 100 digits" |
17:06.02 | [TK]D-Fender | wins |
17:06.13 | *** join/#asterisk eXistenZ (i=pectic@unaffiliated/existenz) |
17:06.14 | Qwell | ~pi |
17:06.15 | jbot | i guess pi is 3.141592653589793238462643383279502884197169399375105820974944592307816406286208998628034825342117067982148086513282306647093844609550582231725359408128481117450284102701938521105559644622948954930381964428810975665933446128475648233786783165271201909145648566923460348610454326648213393607260249141273724587006606315588174881520920962829254091 |
17:06.15 | Kobaz | heh |
17:06.19 | Qwell | wins |
17:06.35 | Dabba | [TK]D-Fender: i see a few Retransmitting #1 on the INVITES ? |
17:07.00 | dlynes | hrm...I guess 1.4.20.1 is so important, they don't even say why it was released on asterisk.org :) |
17:07.02 | [TK]D-Fender | Dabba: Do You? I see nothing... where's the pastebin? |
17:07.23 | Kobaz | dabba do |
17:07.44 | Dabba | has sanitising to do |
17:10.39 | [TK]D-Fender | HCL for those tough to get out stains... |
17:10.46 | [TK]D-Fender | ~h2so4 |
17:10.47 | jbot | [~H2SO4] "John was here but is no more, for what he thought was H2O was H2SO4" |
17:10.49 | [TK]D-Fender | :D |
17:10.53 | [TK]D-Fender | Still there, heh |
17:15.21 | Trapa | Could somone review http://pastebin.com/d55f90534 and see if they can find any reason why we're getting calls disconnecting after the servers been up for a few hours? |
17:15.34 | *** join/#asterisk rattler_ (n=misha@finly.sats.volia.net) |
17:20.24 | Dabba | [TK]D-Fender: http://pastebin.ca/1027211 |
17:21.37 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
17:22.52 | SteveTotaro | 1.4.20.1 was released to remove the g |
17:22.54 | SteveTotaro | GPL |
17:23.01 | drumkilla | blinks |
17:23.20 | SteveTotaro | :-D |
17:24.50 | [TK]D-Fender | Dabba: pastebin your sip.conf masking only passwords |
17:27.26 | *** join/#asterisk darmock (n=root@c-98-211-225-216.hsd1.fl.comcast.net) |
17:29.27 | Dabba | [TK]D-Fender: http://pastebin.ca/1027220 |
17:31.18 | [TK]D-Fender | Dabba: What is RS, and whats the networking between them? |
17:32.12 | Dabba | both rs and pmr are linksys spa-941 with public ip's and on same lan |
17:33.10 | Dabba | both devices running same firmware and all gui settings on the endpoints are identical except username / pass |
17:34.51 | [TK]D-Fender | Dabba: the retransmit is typically a network issue, be it firewall, or NAT. |
17:35.14 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
17:36.00 | Dabba | mad isnt it, phones not behind nat, in same switch, real ip's /25 mask pbx in same lan real ip etc etc |
17:36.02 | Nugget | http://nugget.livejournal.com/131726.html <-- I need one of these |
17:36.38 | Qwell | Nugget: you know you can get custom stamps made pretty cheap? |
17:36.45 | Nugget | yeah |
17:36.53 | *** part/#asterisk mking_cd (n=mking_cd@pool-72-78-183-123.phlapa.east.verizon.net) |
17:38.20 | *** join/#asterisk nny_2 (n=Scott_My@64.203.239.83) |
17:39.12 | nny_2 | isn't sip show peers still the way to show sip peers? Or have our benevolent gods changed the syntax? |
17:40.19 | nny_2 | nm bad config file, module isnt loaded |
17:42.01 | *** join/#asterisk shinao1 (n=shinao1@smtp.gtbplc.com) |
17:46.53 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
17:47.39 | anonymouz666 | what could case ECHO between two ATAs (SIP-only)? PAP2-A calls PAP2-B. A hears echo. PAP2-A calls PSTN no echo at all. PAP2-A calls a Hardphone no echo at all. |
17:48.56 | anonymouz666 | amazing. |
17:50.15 | anonymouz666 | another ATA in PAP2-A position still produces echo. |
17:50.22 | Dabba | having just purchased the TE121andEC i wonder if digium will assist with this problem i have |
17:50.53 | anonymouz666 | Dabba: why don't you call the Digium support? |
17:50.57 | drumkilla | yes, please contact support@digium.com |
17:51.03 | drumkilla | or call |
17:53.16 | anonymouz666 | just to finish the caes, if PAP2-A calls PAP2-C. A hears echo. |
17:53.47 | *** join/#asterisk golumn (n=acxty@201.220.132.138) |
17:55.16 | golumn | Hi guys, I want to make a Dial outside asterisk. My context is 5018989 in sip.conf so I am trying this. Dial(SIP/5018989/98781232) but it tell me Received response: "Forbidden" from .... I make a test conecting the line with xlite and it can dial outside |
17:55.54 | smartlu | thanks for your help... |
17:57.38 | plik | golumn: DIal(SIP/98781232@5018989) number@context should work |
17:58.40 | golumn | plik the same |
17:58.43 | golumn | result |
18:00.16 | Dabba | doh |
18:01.03 | Dabba | as my issue is not digium hardware related, i need to buy consulting time, but the sales lady transferred me back even though i asked to buy some, lol |
18:01.07 | Dabba | oh well |
18:02.37 | golumn | does someone have that same issue before? |
18:08.04 | *** join/#asterisk techie (n=techie@adsl-76-214-30-46.dsl.lsan03.sbcglobal.net) |
18:08.34 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
18:09.04 | *** join/#asterisk jjshoe (n=jjshoe@72.37.252.50) |
18:11.31 | *** join/#asterisk davidgonzalezh (n=dgonzale@190.26.166.229) |
18:11.46 | davidgonzalezh | Hi frinds from asterisk comunity |
18:15.07 | JenniferAkemi | hi davidgonzalezh |
18:15.43 | Qwell | jameswf-home: 3 now |
18:16.27 | jameswf-home | qua? |
18:16.31 | Qwell | nothing |
18:18.48 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
18:25.16 | davidgonzalezh | Wwell guys very pleased to be on this channel |
18:25.26 | davidgonzalezh | it was long time I used IRC chat. |
18:26.45 | davidgonzalezh | I'd like to tell ypou that I'm been experimenting with Asterisk for the last yr and it's been great, ya know I hated telephony and lal that but * has opened my eyes to a whole new world |
18:27.36 | danlock2 | ... ok |
18:29.07 | jameswf-home | starts humming the little murmaid song.... a whole new world... |
18:30.14 | davidgonzalezh | yeah rite the lil mermaid you don't know a thing |
18:30.23 | davidgonzalezh | that son's from Aladdin |
18:30.50 | jameswf-home | at least I had a disney movie,,,, arent they all the same |
18:31.12 | cpm | It is a most elusive fish! |
18:31.33 | davidgonzalezh | Anyway this is the normal behavior I expect from guys that join these channels and you get yell at if you ask a question that's been answered a thousand times. |
18:31.55 | davidgonzalezh | anyway I'm just offering my help and experience with asterisk tto all o' those that may need it. |
18:33.21 | JenniferAkemi | davidgonzalezh: do you have any experience with making asterisk highly available? (through clustering etc) |
18:35.42 | *** join/#asterisk adeel (n=adeel@c-24-7-132-23.hsd1.ca.comcast.net) |
18:35.50 | davidgonzalezh | JenniferAkemi: well we made a project on the company I worked for and it was quite big, we used PostgreSQL and ODBC to cluster all of the config files we used * realtime feature and it went quite good. |
18:36.05 | davidgonzalezh | ther's load balancing between the servers and all. |
18:36.13 | JenniferAkemi | what did you use do do the load balancing? |
18:36.34 | Qwell | JenniferAkemi: should look into dundi |
18:36.57 | davidgonzalezh | balance I guess it was. |
18:36.58 | JenniferAkemi | Qwell: yeah i have that on my list for the outbound load balancing |
18:37.24 | JenniferAkemi | i'm also trying to figure out what we'll do to make the registration redundant |
18:37.30 | JenniferAkemi | i was looking at using lvs and heartbeat |
18:37.32 | Qwell | for incoming, a lot of people seem to use SER, or even just a load balancer router box |
18:37.36 | davidgonzalezh | dundi should be good, I never used it really I'd like to tho. |
18:37.53 | *** join/#asterisk dkwiebe (n=darren@h66-112-187-16.mcsnet.ca) |
18:38.54 | Qwell | (hardware router, that is) |
18:38.59 | JenniferAkemi | another thing i'm worried about is if one of the asterisk box goes down or freezes up or something, what happens to my PRI. in order for it to work right it will have to take the PRI out of service, or else the PSTN will still send calls to that asterisk box |
18:39.26 | Qwell | or T1 failover boxes |
18:39.38 | JenniferAkemi | well i'm not worried about hte T1 failing |
18:39.47 | JenniferAkemi | because it's coming from a switch that is under my control |
18:39.49 | davidgonzalezh | Aw tehre's no way to solve that but to have two PRI cards for redundancy.. |
18:39.57 | km2 | JenniferAkemi, you could get a trunk bypass switch, though i've only heard of these, never seen one |
18:39.58 | JenniferAkemi | if the T1 fails that's good - i'll just send it to the next asterisk box |
18:40.02 | Qwell | T1 > magic box >> asterisk/asterisk |
18:40.04 | Qwell | like that |
18:40.14 | JenniferAkemi | i can have like 60 T1's |
18:40.25 | JenniferAkemi | and if one goes down, it can just go to the next one |
18:40.43 | JenniferAkemi | the issue is that the T1 has to go down if the asterisk box isn't ready to take the call that will show up on it |
18:40.54 | JenniferAkemi | i'm worried about the sip stack freezing or something |
18:41.03 | davidgonzalezh | hmmmm |
18:41.19 | JenniferAkemi | I seem to remember i've seen random people asking about or other things in my various web meanderings |
18:41.19 | gr0mit | JenniferAkemi, that would _never happen ;-) |
18:41.22 | JenniferAkemi | ok |
18:42.00 | Kobaz | JenniferAkemi: western telematic makes an rj48 8 conductor A/B switch |
18:42.11 | JenniferAkemi | what's that for |
18:42.19 | Qwell | to switch the destination of the T |
18:42.28 | JenniferAkemi | how does it know when to switch it |
18:42.35 | Kobaz | you tell it when to switch it |
18:42.41 | Kobaz | it has a telnet interface |
18:42.41 | Qwell | if it's an A/B switch, it's a button you'd press |
18:42.51 | JenniferAkemi | when is that useful |
18:42.53 | Kobaz | it can also switch on current loss |
18:42.54 | Qwell | rarely |
18:42.57 | JenniferAkemi | hehe |
18:43.07 | Kobaz | Qwell: we use them all over here |
18:43.09 | Qwell | if you're talking HA, you want 0 lost calls. |
18:43.11 | JenniferAkemi | i'm not worried about the T1 going down |
18:43.15 | Qwell | so it would be automatic and immediate |
18:43.20 | Qwell | would need to be* |
18:43.29 | JenniferAkemi | if the T1 is down, the PSTN calls won't go to it, they'll go to the next PRI in the hunt group |
18:43.50 | Kobaz | JenniferAkemi: we don't use it for a t1 dieing, we use it to switch over a single t1 to a failover box |
18:44.14 | Kobaz | ie: we need to change out hardware or bring the box down |
18:44.16 | JenniferAkemi | Qwell is right though, I don't want any interruption of service at all |
18:44.22 | gr0mit | JenniferAkemi, what is the application you are considering? |
18:44.33 | JenniferAkemi | residential voip |
18:44.40 | Qwell | 5 9's? |
18:44.53 | JenniferAkemi | or as close as we can get |
18:45.06 | JenniferAkemi | probably going to be marketed as a "second line" |
18:45.10 | JenniferAkemi | but it can't suck :) |
18:45.19 | gr0mit | i think you probably want a telco-grade switch then |
18:45.25 | gr0mit | if you want it scalable. |
18:45.39 | gr0mit | Asterisk is great but scalability is a bit 'meh' |
18:45.41 | JenniferAkemi | it's your opinion asterisk won't do the job gr0mit? |
18:45.51 | gr0mit | how many subscribers? |
18:46.06 | JenniferAkemi | a few thousand |
18:46.10 | drumkilla | i use asterisk for 1 billion subscriberes |
18:46.16 | drumkilla | but i can't spell |
18:46.20 | JenniferAkemi | heh |
18:46.27 | JenniferAkemi | what is the application drumkilla? |
18:46.28 | jjshoe | JenniferAkemi we have a magic t1 box switcher, it detects when the box goes down |
18:46.39 | Qwell | I'd say Mike proved that Asterisk can most certainly handle that type of volume earlier |
18:46.42 | jjshoe | JenniferAkemi but it's here to try and get us to market it, i don't htink we've ever used it |
18:46.57 | JenniferAkemi | jjshoe: heh |
18:47.00 | Kobaz | jjshoe: what do you use? |
18:47.12 | *** join/#asterisk delparnel (n=delparne@KTNRON06-1168103470.sdsl.bell.ca) |
18:47.15 | JenniferAkemi | i don't think i need a magic t1 box switcher at all |
18:47.20 | jjshoe | Kobaz if our pri's go down we automatically roll to a voip provider. |
18:47.25 | adeel | JenniferAkemi, a lot of people use a hybrid setup to do what you're looking for...typically handle the sip registrations through openser and then let * handle the media transcoding/voicemail/etc |
18:47.36 | *** join/#asterisk sbingner (n=john@pdpc/supporter/sustaining/sbingner) |
18:47.46 | gr0mit | JenniferAkemi, I really don;t know if it can handle thousands of subs. most people use SER in front to load balance |
18:47.48 | JenniferAkemi | yeah i've seen a lot of talking about using openser to handle sip registrations. why use openser for that over asterisk though? |
18:47.50 | jjshoe | Kobaz we have the ability to directly point our number on the pri's to the voip. <3 our telco. |
18:48.06 | gr0mit | and use asterisk as a glorified media gateway |
18:48.09 | adeel | JenniferAkemi, openser's implementation can scale better than *'s |
18:48.17 | JenniferAkemi | what if you just add asterisk boxes |
18:48.17 | Qwell | JenniferAkemi: SER is a SIP proxy. It is not a PBX, so it doesn't have to handle any of that type of thing |
18:48.19 | JenniferAkemi | to scale |
18:48.20 | Kobaz | jjshoe: ah |
18:48.31 | JenniferAkemi | how many registrations can asterisk handle |
18:48.34 | gr0mit | adding asterisk boxes is problematic |
18:48.34 | Qwell | JenniferAkemi: that's perfectly acceptable, as long as you have some way to get the packets between the boxes |
18:48.56 | JenniferAkemi | dedicated gigabit ethernet apart from the voice path? |
18:49.18 | gr0mit | but even then you are not really load sharing |
18:49.35 | JenniferAkemi | i was hoping one asterisk box could handle registrations, and store them in a realtime database |
18:49.36 | gr0mit | because all your traffic then has to go through 2 or more boxes |
18:49.46 | jjshoe | honestly? making money by running a reseidential voip service is almost impossible |
18:49.48 | JenniferAkemi | incoming calls come in through multiple pris |
18:50.08 | gr0mit | residentail voip is a mugs game. |
18:50.11 | JenniferAkemi | which go to multiple asterisk boxes which look up the ip's of the registered sip things |
18:50.17 | JenniferAkemi | why gr0mit. what is the roadblock |
18:50.18 | jjshoe | at $10 a month it takes one 5 minute phone call from the customer before you've spent the $10 they paid you that month |
18:50.20 | gr0mit | customers expect perfection with zero cost |
18:50.25 | Qwell | JenniferAkemi: that's what most people use SER for |
18:50.27 | *** join/#asterisk kannan (n=kann@123.201.60.110) |
18:50.35 | Qwell | most? many? some? whatever |
18:50.38 | JenniferAkemi | i'm just wondering why SER instead of asterisk though? |
18:50.47 | Qwell | SER is only a SIP proxy |
18:50.47 | gr0mit | not instead of, as well as |
18:51.16 | JenniferAkemi | i understand they use asterisk for voicemail and for the voice path etc, but SER for the registration, but what i dont understand is why |
18:51.21 | kannan | hello, i am not able to attach vm to email, i get in /var/log/maillog , connection refused at 127.0.0.1 |
18:51.44 | gr0mit | JenniferAkemi, the problem is load |
18:51.44 | JenniferAkemi | jjshoe: i guess we have an advantage |
18:52.07 | kannan | what about server loads, isnt SER supposed to be very much efficent? also it can switch TCP SIP? |
18:52.09 | JenniferAkemi | jjshoe: we have pstn minutes available to us an incredibly cheap rates. |
18:52.31 | adeel | JenniferAkemi, from my readings, the max number of sip registrations i've seen on an * box is around 4000-5000 (i may be wrong) but i've seen SER boxes (similar in specs or worse) that can handle 10,000-20,000 registrations |
18:53.23 | gr0mit | and once a client has registered on box A, you need to know that client z is registered on box A |
18:53.27 | adeel | kannan, you haven't setup your postfix/sendmail/mail relay properly |
18:53.46 | *** join/#asterisk banzaika (n=banzaika@rrcs-208-105-66-210.nyc.biz.rr.com) |
18:53.48 | JenniferAkemi | using realtime i was under the impression it didn't matter which box the client regisered on. |
18:54.16 | gr0mit | so if a call comes in for client z on an E1 on box B, then box B has to send it to Box A over IAX or SIP |
18:54.22 | *** part/#asterisk drzed (n=drzed@synflood.homelinux.org) |
18:54.31 | Qwell | JenniferAkemi: to be honest, I wouldn't use realtime in a setup like that |
18:54.47 | JenniferAkemi | can't box b just look it up in the sip table in realtime and send it to the ip |
18:54.58 | JenniferAkemi | Qwell: do you have a reason? |
18:55.10 | Kobaz | JenniferAkemi: asterisk doesn't work like that |
18:55.12 | gr0mit | JenniferAkemi, have you used Asterisk yet ? |
18:55.21 | Qwell | it adds several extra layers of potential failure |
18:55.23 | JenniferAkemi | not in the scenario i'm describimg : |
18:55.26 | JenniferAkemi | :) |
18:55.32 | JenniferAkemi | i am in the middle of setting it up though |
18:55.38 | Qwell | requires extra database servers, etc, etc |
18:55.42 | gr0mit | well, |
18:55.47 | JenniferAkemi | i've got one asterisk box using realtime attaching to 2 database servicers which are mirrored master master |
18:55.57 | JenniferAkemi | i'm just setting up the other |
18:56.01 | Strom_M | servicers? sigh |
18:56.06 | Kobaz | realtime doesn't gain you much of anything |
18:56.07 | JenniferAkemi | and am going to attemp to call from one to the other |
18:56.08 | kannan | adeel : how to install the sendmail? |
18:56.15 | JenniferAkemi | servers sorry typo |
18:56.23 | gr0mit | realtime only helps with automateing adding sip clients |
18:56.34 | kannan | /usr/sbin/sendmail -d0 < /dev/null | grep -i version gives me the version, and the X package manager shows sendmail is installed |
18:56.35 | Kobaz | you can do an insert into a db rather than generating config files, that's about it |
18:56.35 | gr0mit | does not help you scale an operation. |
18:56.46 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
18:56.52 | JenniferAkemi | the thing i've liked about realtime is that there is only one config for allthe boxes |
18:56.56 | JenniferAkemi | they all access it |
18:57.08 | Qwell | JenniferAkemi: yeah, but you could do that with rsync and the likes |
18:57.11 | gr0mit | and you can achieve the same with a big file and nfs export to multiple asterisk boxes |
18:57.12 | Kobaz | you can have a master configuration server that rsyncs your configs to your dest |
18:57.15 | Kobaz | yeah exactly |
18:57.27 | JenniferAkemi | hm. |
18:57.47 | adeel | JenniferAkemi, i don't mean to be rude, but if it was as simple as you expect it to be, everyone would be doing it |
18:57.58 | JenniferAkemi | coming from a non linux background i am at a disadvantage as I didn't even know what rsync was |
18:58.09 | jjshoe | if you have super cheap pstn minutes available sell them to other voip provider. |
18:58.12 | JenniferAkemi | adeel: which? providing residential voip, or using realtime to provide HA |
18:58.13 | jjshoe | providers. |
18:58.15 | adeel | kannan, yeah, that can take a lot of work, and the best way to get the answer is to google it... |
18:58.26 | Kobaz | JenniferAkemi: so you're comming from a non-linux background and you want to use asterisk to sell phone service to 1000+ people? |
18:58.35 | Kobaz | JenniferAkemi: i suggest you try and different business |
18:58.39 | JenniferAkemi | jjshoe: once we get this voip stuff running that's another market to tap |
18:58.43 | Kobaz | s/and/a |
18:58.45 | gr0mit | JenniferAkemi, I don't want to put you off |
18:58.47 | JenniferAkemi | well i'm coming from a telco background |
18:58.52 | gr0mit | asterisk is GREAT |
18:58.54 | gr0mit | but..... |
18:59.14 | adeel | JenniferAkemi, providing residential voip isn't too hard, but scaling it efficiently is the hard part...if you have 200 * boxes to service 10,000 clients, i don't think you'll be able to cover your expenses while keeping your prices low |
18:59.17 | gr0mit | you don't use an MD110 as an internaltional SS7 gateway, |
18:59.33 | JenniferAkemi | asterisk boxes are cheap though |
18:59.34 | gr0mit | and you don't use an AXE10 for a 100-seat pbx/call centre |
18:59.35 | adeel | JenniferAkemi, and using realtime * typically involves some custom app's and whatnot... |
18:59.42 | JenniferAkemi | the big cost is the digium card |
18:59.43 | adeel | JenniferAkemi, but electricity and connectivity isn't |
18:59.52 | JenniferAkemi | connectivity is cheap |
18:59.56 | Kobaz | JenniferAkemi: and you don't want digium either |
18:59.59 | JenniferAkemi | if you already have it |
19:00.04 | gr0mit | JenniferAkemi, i dont thing anyone here is saying 'No' |
19:00.14 | Qwell | Kobaz: back that statement up with facts. |
19:00.28 | gr0mit | but I think you should consider playing in a small-scale depoloyment before you scale up |
19:00.31 | Kobaz | Qwell: i have a higher pile of dead digium cards than any other |
19:00.34 | Strom_M | Qwell: watch these facts all be from five years ago |
19:00.41 | JenniferAkemi | gr0mit which is what i'm doing right now |
19:00.49 | JenniferAkemi | obviously we'll open up for beta testing first |
19:00.50 | gr0mit | JenniferAkemi, fine! |
19:00.56 | [TK]D-Fender | Qwell: fact : * isn't what most telcos would consider "stable", this relying on that to get to the PSTN = not so great |
19:01.02 | Kobaz | i find sangoma much more solid |
19:01.14 | gr0mit | uses Sangoma cards |
19:01.25 | JenniferAkemi | we're going to get sangoma card to test |
19:01.28 | adeel | JenniferAkemi, my biggest cost right now is the bandwidth i need |
19:01.33 | JenniferAkemi | we have a digium and a t100p clone right now |
19:01.38 | JenniferAkemi | adeel how much do you pay? |
19:01.39 | gr0mit | JenniferAkemi, which country is this for? |
19:01.41 | JenniferAkemi | adeel i was pricing that out today |
19:01.42 | JenniferAkemi | Canada |
19:01.50 | Qwell | JenniferAkemi: avoid the clones |
19:01.52 | gr0mit | ok |
19:02.02 | gr0mit | well Sangoma are in Toronto |
19:02.12 | kannan | is x100p.com a clone? |
19:02.13 | *** join/#asterisk A500mg (n=x@ACaen-156-1-2-126.w90-17.abo.wanadoo.fr) |
19:02.17 | gr0mit | their support is great |
19:02.20 | adeel | JenniferAkemi, depends...i'm in california, and depending on where i go, i get different rates....t1 from 300 bucks to a t3 for just under 3,000 or so....could get into a carrier hotel, but the costs are still pretty high |
19:02.27 | [TK]D-Fender | kannan: No, its just shit :) |
19:02.35 | kannan | oh ok |
19:02.54 | JenniferAkemi | i figured i'd start with business dsl for beta testing |
19:02.54 | [TK]D-Fender | You wanna run an ITSP, use serious back-end gear like AudioCodes gateways. |
19:03.06 | gr0mit | or Teles |
19:03.07 | kannan | hmm , is there a friendly GUI fror sendmail heh? |
19:03.14 | gr0mit | my wholesaler uses Teles kit |
19:03.23 | [TK]D-Fender | kannan: webmin |
19:03.29 | JenniferAkemi | yeah i just noticed sangoma is local |
19:03.32 | jjshoe | um hahaha |
19:03.37 | jjshoe | if you think the cards are expensive |
19:03.41 | [TK]D-Fender | kannan: For as "frienldy" as that can be. when in doubt, JFGI |
19:03.42 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
19:03.43 | gr0mit | 'local' as in Canada, ! |
19:03.44 | jjshoe | you're in the wrong line of work |
19:03.46 | adeel | kannan, unless you need a FULL MTA, don't use sendmail...just setup SSMTP for relaying |
19:03.48 | kannan | [TK]D-Fender : thanks |
19:03.48 | *** part/#asterisk techie (n=techie@adsl-76-214-30-46.dsl.lsan03.sbcglobal.net) |
19:03.50 | gr0mit | (tis a big place!) |
19:03.51 | JenniferAkemi | i'm about 30 minutes from markham |
19:03.56 | gr0mit | ah ok |
19:04.05 | kannan | adeel : thanks , whats that? heh |
19:04.07 | A500mg | mmh |
19:04.15 | A500mg | I've a question |
19:04.16 | kannan | i just need to attach the vm to email, |
19:04.16 | adeel | kannan, a simple mail relay |
19:04.19 | *** join/#asterisk techie (n=techie@adsl-76-214-30-46.dsl.lsan03.sbcglobal.net) |
19:04.24 | gr0mit | are you with an ISP JenniferAkemi |
19:04.24 | JenniferAkemi | jjshoe: the computers are cheaper than the cards. minutes are pracitally free |
19:04.39 | JenniferAkemi | long distance provider |
19:04.41 | [TK]D-Fender | kannan: You shouldn't HAVE to configure anything for sendmail for that to work in most cases |
19:04.42 | kannan | we have to replace the cmd for sendmail in voicemail.conf after building that? |
19:04.44 | A500mg | when I do a "core show channels" during a call is ringing |
19:04.44 | gr0mit | aaaah ok |
19:04.53 | kannan | or can we hust build and link to sendmail |
19:04.55 | gr0mit | hence 'free' minutes ;-) |
19:05.02 | JenniferAkemi | we have 20 ds3s around the country |
19:05.05 | A500mg | I've a "ring" and a "ringing" |
19:05.07 | JenniferAkemi | so yeah :) |
19:05.18 | A500mg | what's the difference between "ring" and "ringing" ? |
19:05.25 | jjshoe | JenniferAkemi that's it? |
19:05.32 | [TK]D-Fender | kannan: jsut install * like normal, * just uses it to send the message. shouldn't have anything to configure there unless your ISP forces you to do SMTP through their gateway |
19:05.33 | kannan | [TK]D-Fender : i thought so, it worked in other boxes automatically, now it is not working, dunno howto figure it at all |
19:05.38 | gr0mit | well, iiwy I would look at getting some telco-grade media gateway |
19:05.54 | JenniferAkemi | like a cisco something? |
19:05.59 | gr0mit | nah |
19:06.04 | gr0mit | Teles |
19:06.07 | [TK]D-Fender | JenniferAkemi: AudioCodes Mediant series |
19:06.09 | gr0mit | or Audiocodes |
19:06.10 | kannan | i am not getting any, any at all, messages at root@localhost |
19:06.27 | jjshoe | Kobaz you're %100 correct on your recommdantion btw. |
19:08.19 | A500mg | I think "ring" is the caller and "ringing" the callee, for example: |
19:08.20 | A500mg | SIP/tech02-09210c70!context-tech02!701!1!Ringing!AppDial!(Outgoing Line)!701!!3!1!(None) |
19:08.20 | A500mg | SIP/tech01-091eca00!context-tech01!701!2!Ring!Dial!SIP/tech02|60|tT!700!!3!1!(None) |
19:08.31 | A500mg | tech01 is calling tech02 |
19:09.02 | JenniferAkemi | what sorts of rates would you guys pay for voip minute |
19:09.06 | JenniferAkemi | s |
19:09.14 | Qwell | 2c/min is common |
19:09.23 | JenniferAkemi | to us ? |
19:09.32 | kannan | lol |
19:09.34 | Qwell | us, canada, some places in europe, mexico |
19:09.51 | JenniferAkemi | what sort of quality? |
19:10.03 | Qwell | toll? |
19:10.04 | kannan | oh to USA , i thought to JenniferAkemi |
19:10.14 | JenniferAkemi | :P |
19:10.15 | Strom_M | for 2c/minute, I expect toll quality |
19:10.19 | Qwell | Strom_C: yeah.. |
19:10.21 | JenniferAkemi | what's toll quality |
19:10.26 | JenniferAkemi | is that the best? |
19:10.36 | Strom_M | JenniferAkemi: i thought you said you came from a telco background :P |
19:10.37 | delparnel | Like a normal phone line |
19:10.42 | JenniferAkemi | yeah but we just buy the best |
19:11.00 | Strom_M | toll quality is the same quality you expect when placing a standard toll call over the PSTN |
19:11.04 | JenniferAkemi | ok |
19:11.07 | Qwell | well, what's the best? surely you aren't offering 44khz stereo audio |
19:11.18 | JenniferAkemi | i mean |
19:11.44 | JenniferAkemi | when a company says they have "gold rates" "platinum rates" or sorry to say what they sometimes call "voip rates" we only buy the platinum minutes |
19:12.00 | JenniferAkemi | it hasn't been worth the customer service nightmares to buy the cheapest |
19:12.10 | Strom_M | JenniferAkemi: uh, so exactly what kind of "telco background" do you have? |
19:12.18 | jjshoe | I'm beginning to think this is one very good troll |
19:12.31 | JenniferAkemi | the kind that you get from working at a telco for the last 10 years |
19:12.35 | [TK]D-Fender | JenniferAkemi: I use cubic zirconium minutes. The cost much less but last just as long |
19:12.43 | JenniferAkemi | a troll? |
19:12.52 | delparnel | bahahaha D-Fender |
19:13.20 | Strom_M | JenniferAkemi: so wait, how can you work at a telco for a decade and not know what "toll quality' means |
19:13.22 | JenniferAkemi | i'm offended even though you made me laugh |
19:13.28 | A500mg | lol d-fender |
19:13.29 | jjshoe | JenniferAkemi so what do you plan to do with 10 years of experience of mopping the floors of a telco for 10 years? |
19:13.43 | jjshoe | ooo that was an awful sentence. |
19:13.51 | delparnel | yes, yes it was. |
19:13.58 | [TK]D-Fender | JenniferAkemi: pwned |
19:14.08 | Qwell | girl + irc + troll. impossible |
19:14.26 | JenniferAkemi | toll quality could mean what people commonly accept on a long distance call - as in a toll call, vs what one might expect on a local call - which in my opinion is a little different |
19:14.41 | JenniferAkemi | you expect the person you call on the other side of the world to sound a little fuzzier than the person next door |
19:14.42 | jjshoe | Qwell hahaha +10 points to you |
19:14.51 | Qwell | no, I meant, she really is a girl :p |
19:14.54 | jjshoe | JenniferAkemi no I don't. |
19:15.00 | JenniferAkemi | not anymore |
19:15.04 | JenniferAkemi | but historically people did |
19:15.10 | delparnel | I think for 2c a minute you're going to get toll quality or better if your connection is good. |
19:15.11 | *** join/#asterisk CaTtleyA (n=DDoS@bas1-montrealak-1128581558.dsl.bell.ca) |
19:15.30 | JenniferAkemi | i'm not disputing your 2c/minute number for toll quality |
19:15.38 | jjshoe | Qwell that would explain a lot |
19:15.47 | Qwell | JenniferAkemi: this is true. these days "toll" doesn't really apply |
19:15.52 | delparnel | Sometimes you can get even better, depending on the volume of calls you do. |
19:15.58 | JenniferAkemi | are you being a little sexist here jjshoe? |
19:16.06 | Qwell | JenniferAkemi: ignore him |
19:16.14 | JenniferAkemi | what about for inbound minutes on a DID |
19:16.15 | delparnel | I have received as low as $0.015/min |
19:16.15 | JenniferAkemi | same rate? |
19:16.27 | JenniferAkemi | thanks Qwell. |
19:16.28 | Qwell | JenniferAkemi: pretty much |
19:17.17 | JenniferAkemi | it's true that the telco world is a little overrun by men. |
19:17.51 | JenniferAkemi | i've only ever talked to one or two other women at other telcos |
19:17.57 | delparnel | JenniferAkemi: Some ITSP's offer unlimited incoming for a fixed rate per DID.. something like $5.95 in the USA or $6.95 in Canada |
19:18.17 | Qwell | delparnel: those aren't actually unlimited though |
19:18.21 | Qwell | it's "unlimited*" |
19:18.36 | delparnel | Or alternatively $0.99/DID + $0.01/min |
19:18.38 | [TK]D-Fender | (tm)(r)(oac)(apr) |
19:18.40 | Qwell | (where "unlimited" means x minutes * xc/min) |
19:18.57 | A500mg | what's the difference between "ring" and "ringing" when I do "core show channels" ? |
19:19.03 | Qwell | A500mg: incoming vs outgoing |
19:19.04 | delparnel | It usually ends up being more profitable though if you do a high volume. |
19:19.07 | Qwell | I don't know which is which though |
19:19.29 | A500mg | mmh |
19:19.35 | A500mg | I think this also |
19:19.54 | A500mg | but i've test an inbound call on misdn |
19:20.23 | A500mg | and I've ringing for both (misdn line and sip/phone) |
19:20.30 | A500mg | maybe a little difference with misdn ? |
19:20.33 | jjshoe | any res. voip company that gets big enough is just going to get slapped around by the major telco's like vonage did anywyas. it's a stupid market to enter imho. |
19:20.42 | A500mg | I will test with zap channel for see ... |
19:20.51 | *** join/#asterisk Hydrant (n=aj@CPE0011950c737b-CM0012c90d1420.cpe.net.cable.rogers.com) |
19:20.52 | Qwell | vonage was stupid on their own |
19:21.20 | Qwell | you don't spend $150 per customer in advertising for a customer you're only going to have for 3 months |
19:21.22 | Strom_M | is that barrel of monkeys still around? |
19:21.35 | *** join/#asterisk M1s3ry (n=M1s3ry@216.207.245.1) |
19:21.41 | Strom_M | hey M1s3ry |
19:21.51 | M1s3ry | howdy |
19:22.16 | *** join/#asterisk axisys (i=iqbala@otaku.freeshell.ORG) |
19:22.17 | JenniferAkemi | i don't get the big price difference between like vonage and vbuzzer |
19:22.32 | JenniferAkemi | ones $20 a month and ones $2 a month |
19:22.38 | JenniferAkemi | and they seem to offer the same service pretty much |
19:22.44 | Qwell | like I said - you don't spend $150 per customer :) |
19:22.50 | A500mg | mmh or this is inbound/outbound and not callee/caller, call is inbound on misdn and inbound also on the phone who is ringing |
19:23.15 | M1s3ry | if the quality and everything else is the same, I guess the difference would only be $18 |
19:23.23 | JenniferAkemi | heh M1s3ry |
19:23.24 | Strom_M | I would like to strangle whoever thought up the word "callee" |
19:23.30 | kannan | yippee You have new mail in /var/mail/root |
19:23.31 | Strom_M | it's "called party" |
19:23.39 | Qwell | and caller? |
19:23.43 | Strom_M | "calling party" |
19:23.44 | Qwell | (calling party) |
19:23.48 | jjshoe | Strom_M I'd agree. |
19:23.50 | Qwell | but, are you okay with caller? |
19:24.04 | kannan | heh, after a thorough googling , i had to do a chmod a+x rc.sendmail for that one, heheh |
19:24.04 | Strom_M | Qwell: moreso than I am with 'callee' |
19:24.17 | Strom_M | but i don't like it in that context |
19:24.19 | davidgonzalezh | which would be a good voip provider tthat goves me free 8000 calls |
19:24.22 | JenniferAkemi | why is it scln and scdn |
19:24.25 | M1s3ry | or what about "calling agent" and "agent called"? |
19:24.35 | A500mg | mmh |
19:24.37 | kannan | but at least it gave me a start on MTA |
19:24.40 | *** join/#asterisk angom (n=angom@201.170.65.143) |
19:24.40 | A500mg | 3 test: |
19:25.18 | A500mg | SIP/phone1 place an internal call on SIP/phone2: SIP/phone1 is "ring" and SIP/phone2 is "ringing" |
19:25.53 | A500mg | SIP/phone1 place an external call on misdn/(number): SIP/phone1 is "ring" and misdn-e54ezf54 is "ringing" |
19:26.00 | Hydrant | anyone have experience using a USB phone? I want to setup a USB phone with my laptop so that when I'm travelling and on wireless I can have my laptop phone on a ring group and I can get calls |
19:26.36 | jjshoe | Hydrant never used one. I'm not sure I see the dificulty though. I'm sure things like x-lite might have support for it. |
19:26.54 | A500mg | inbound call from external (isdn line), on SIP/phone1: misdn-ezf251ze5 is "ringing", SIP/phone1 is "ringing" |
19:27.03 | Hydrant | jjshoe: just used to the good ol' Linux days where the word "USB" made you cringe |
19:27.07 | lmadsen | what am I doing wrong with this STRPTIME() function? |
19:27.08 | lmadsen | http://www.pastebin.ca/1027321 |
19:27.12 | A500mg | the last example trouble me :) |
19:27.17 | Qwell | Hydrant: USB "phones" are just soundcards |
19:27.42 | A500mg | misdn can be the "caller" but is "ringing" |
19:27.48 | Hydrant | Qwell: cool... so they should be fine with Linux then... I'll check to see what I can do |
19:27.56 | lmadsen | nevermind |
19:27.58 | lmadsen | TYPO! |
19:28.25 | A500mg | if we think "misdn receive a call from external" and "SIP/phone1 receive also a call from misdn" , "ringing" for both is logic |
19:28.28 | lmadsen | ${start_time} != ${start_date} |
19:28.49 | A500mg | I just need to understand :) |
19:29.05 | jjshoe | Hydrant it still does, what os are you using? |
19:29.22 | Hydrant | jjshoe: Linux |
19:29.26 | Hydrant | jjshoe: 2.6.x |
19:31.17 | jjshoe | Hydrant oh, best of luck. |
19:31.26 | davidgonzalezh | Aw cool that'¡s agood list of Voip providers very comperhensive |
19:31.42 | jjshoe | Hydrant why not use your existing sound card? |
19:31.54 | Hydrant | jjshoe: I just like the feeling of a phone |
19:32.04 | Hydrant | jjshoe: although I guess I could get a USB headset or something |
19:32.12 | davidgonzalezh | hehe, the old days of a phone. |
19:32.21 | davidgonzalezh | get over it use xlite |
19:32.52 | Hydrant | only other nice thing is a ringer |
19:33.03 | Hydrant | although I can setup something on my screen or something else |
19:33.07 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
19:34.54 | Hydrant | I guess the best bet is to get a headset of some sort after all |
19:35.03 | Hydrant | researching it seems to be less of a headache |
19:37.07 | JenniferAkemi | i have a headset that i use that works well when i'm not at home. it's not usb though, it just has two jacks, one for the mic and one for the headphones |
19:37.50 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
19:39.57 | Hydrant | JenniferAkemi: I think that's the way I'm gonna go |
19:40.23 | Hydrant | I just have to figure out a way to setup something that will work with Asterisk, so that I can setup my laptop as a SIP device or something... |
19:40.42 | Hydrant | Maybe setup something with openvpn for incoming calls, outgoing shouldn't be a problem |
19:40.50 | JenniferAkemi | i just use it with xlite connected to asterisk as davidgonzalezh suggested. |
19:42.12 | *** join/#asterisk niZon (n=niZon@tande.voinetworks.net) |
19:43.02 | JenniferAkemi | well damn :( you guys were right that didn't work |
19:43.36 | JenniferAkemi | i just setup the second * box sharing the sip_buddies table using realtime and registered to one and tried to call an account on the other one |
19:48.33 | JenniferAkemi | maybe i need to use DUNDI to make this part work too |
19:52.15 | kannan | hello, i just realize sendmail enables me to set the from address as anything i desire, isnt that a risky thing in general? |
19:52.15 | *** join/#asterisk didz_ (n=hoje@201.19.207.130) |
19:52.15 | didz_ | the counter of "Bipolar Viol" is increasing on zttool... anyone knows what it means? |
19:52.30 | kannan | i mean there is no sanctity for a email adress? |
19:52.56 | *** part/#asterisk rattler_ (n=misha@finly.sats.volia.net) |
19:53.04 | kannan | i know this snt the rrom for it, just set me wondering a lot |
19:54.49 | lanning | not really a sendmail issue. it extends from SMTP |
19:54.53 | *** join/#asterisk kai4711 (i=psybnc@h1395155.stratoserver.net) |
19:55.19 | [TK]D-Fender | kannan: Seriously... get a clue. |
19:55.42 | lanning | you can telnet to TCP port 25 (SMTP port) and send anything you want. |
19:55.46 | Bananaskin | hey has anyone got iaxmodem/hylafax working on a sip trunk would like to carry out a test of faxing over sip ? |
19:55.55 | [TK]D-Fender | kannan: And you WONDER why your "friends" send you ads for Viagra. Guess what, it isn't "them", they we spoofed |
19:56.11 | *** join/#asterisk A500mg (n=x@ACaen-156-1-104-54.w90-17.abo.wanadoo.fr) |
19:56.12 | A500mg | raaaaaaaa |
19:56.13 | [TK]D-Fender | Bananaskin: .... WGLWAT |
19:56.21 | A500mg | sorry disconnected and pseudo in use, grrrr |
19:56.21 | Bananaskin | hey [TK]D-Fender huh ? |
19:56.23 | [TK]D-Fender | were* |
19:56.27 | [TK]D-Fender | ~wglwat |
19:56.28 | jbot | [wglwat] well, good luck with all that |
19:56.30 | [TK]D-Fender | :p |
19:57.09 | [TK]D-Fender | Bananaskin: the send you say "SIP over internet", thats akin to saying "automatic fax transmission failure" |
19:57.21 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
19:57.21 | [TK]D-Fender | Bananaskin: for faxing that is... |
19:57.35 | A500mg | Qwell have you try to answer my question ? ("ring" and "ringing") |
19:57.36 | Bananaskin | lol, well tbh, I tested an outgoing last night to an actual fax wired to pstn and accidentally sent it over sip and it was as good if not better than the one I sent out over the zap channel |
19:58.05 | [TK]D-Fender | Bananaskin: fax is digital... should have absolutely no impact |
19:58.07 | Bananaskin | ironically the one over zaptel took longer as well |
19:58.24 | *** join/#asterisk golumn (n=golumn@201.220.132.138) |
19:58.30 | Bananaskin | Digital until it meets the telco |
19:59.24 | golumn | Hi guys, I am getting handle_response_invite: Received response: "Forbidden" From "user"....... every time I try to make an outbound call |
20:01.08 | Bananaskin | seems there was a bit of jitter logged on the outbound over sip but nothing which affected the fax transmission |
20:09.52 | *** join/#asterisk grantm (n=grant@66.29.180.194.static.utahbroadband.com) |
20:18.41 | Trapa | Has anyone here had major issues with XLITE? |
20:18.43 | Trapa | I' |
20:18.50 | nny_2 | using it at home Trapa |
20:19.07 | Trapa | I'm getting a problem that the phone will get a half-ring and then it gets a hangup |
20:19.24 | nny_2 | is this on the local network? |
20:19.36 | Trapa | Yeah |
20:20.05 | nny_2 | hmm not sure, have you tried Ekiga or other to confirm it isn |
20:20.16 | nny_2 | 't an issue with the dialplan? |
20:20.41 | Trapa | So the calls come into the asterisk server and then go to the xlite phones which are local to the asterisk server |
20:20.52 | nny_2 | sounds about right |
20:20.54 | Trapa | I don't know that it's NOT a issue with the dialplan .. I'm using queue's... |
20:21.05 | Trapa | and i thought i tested .. and when i test i can't re-create the problem |
20:21.20 | Trapa | And of course i have some agents .. who are giving me not very useful information about what is happening ... |
20:21.57 | Trapa | This is a pastebin of the verbose output from asterisk. This was from last night, until this morning ... |
20:22.12 | Trapa | This morning the agents said that we needed to restart the server, and when we did it was resolved http://pastebin.com/d55f90534 |
20:22.28 | Trapa | But i don't see any errors in the log there. So i am pretty confused as to whats happening |
20:23.32 | *** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net) |
20:23.51 | nny_2 | yeah being able to not recreate the issue is a PIA eh? |
20:24.05 | Trapa | yeah no kidding |
20:24.10 | nny_2 | personally never had that issue unless asterisk suddenly thought the sip peer was gone |
20:24.25 | Trapa | I mean i kinda wonder if people are just hanging up |
20:24.32 | nny_2 | ha |
20:24.37 | nny_2 | check the CDR? |
20:24.46 | Trapa | CDR? |
20:24.56 | *** join/#asterisk hmmhesays (n=hmmhesay@24-119-176-74.cpe.cableone.net) |
20:25.00 | hmmhesays | what up folks |
20:25.20 | nny_2 | /var/log/asterisk/cdr-custom/Master.csv |
20:25.51 | Trapa | don't have that log |
20:25.57 | nny_2 | that and the asterisk logs should at least tell you some more info about the past |
20:25.58 | nny_2 | hmm? |
20:26.07 | nny_2 | strange |
20:26.20 | nny_2 | is on all my installs by default, without any extra input |
20:26.52 | outtolunc | check in the cdr-csv (vice the cdr-custom you had him look in) |
20:27.10 | nny_2 | yeah heh |
20:27.58 | nny_2 | the whole "they could just be hanging up" thing is possible, although i hate guessing myself.. is it frequent or once in a while? |
20:29.41 | *** join/#asterisk pikachu2000 (n=pikachu2@pikachu.cyberdesigns.co.za) |
20:29.54 | Trapa | We only get calls like once in a long while |
20:31.38 | nny_2 | well the cdr records may help with tracking down who what etc, but (correct if i am wrong, peeps) i would think the event would show up as ANSWERED once your IVR or if an Answer() is in before it goes to an agent, so tracking it down may be a bit hard... alternatively, if the sip client was doing the hang-up vs the caller, I don't think it would show up differently in the cdr |
20:31.53 | nny_2 | i am a bit green myself, so take all that with a grain of salt :) |
20:37.57 | JenniferAkemi | does anyone know what the sip field fullcontact is for? |
20:38.44 | *** join/#asterisk datachomper (n=russ@75.146.194.59) |
20:39.08 | datachomper | Is there a "dialtone" sound file? If I want to play a dialtone to a user, how would I normall go about doing it? |
20:39.18 | Strom_M | playtones() |
20:39.51 | datachomper | great, thanks |
20:43.00 | kannan | bye all , c u tomorrow |
20:43.32 | bbryant | JenniferAkemi: it stores the contact header sent when a sip call is registered |
20:45.07 | Ritzerisk | is there a good gui to install for like doing a easy configuration |
20:45.21 | Ritzerisk | on top of the core asterisk already |
20:46.31 | Trapa | nny_2 thanks for the help i'm just busy i'll be back in a momne |
20:46.34 | JenniferAkemi | looks like i could do a mysql lookup for fullcontact, and then send the call directly via sip instead of going through the original sip registration server |
20:46.41 | JenniferAkemi | i guess that's too much overhead though |
20:46.49 | nny_2 | Trapa: heh not sure if its help, but i try |
20:47.18 | *** join/#asterisk ddunavant (n=David@75.145.240.14) |
20:51.20 | Trapa | So ... My queue has several seconds of silence before it says anything to the customer .. Is there some way of changing that? |
20:51.43 | Trapa | I would like the customer to always hear somthing .. be that the "Your next in line" or the music on hold, or ringing or anything other than silence .. |
20:53.23 | *** join/#asterisk war59312 (n=war59312@unaffiliated/war59312) |
20:53.50 | *** join/#asterisk unstable (i=unstable@tor/regular/sid) |
20:57.01 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
20:58.19 | JenniferAkemi | so i'm beginning to understand that asterisk is more like a pbx and not a tandem switch |
20:58.24 | JenniferAkemi | it's good at the features |
20:58.36 | *** join/#asterisk revengervn (n=test_tes@static-96-226-59-205.dllstx.dsl-w.verizon.net) |
20:58.39 | JenniferAkemi | but not so good at the rock solid huge volume just switching calls |
20:58.45 | revengervn | hi |
20:58.45 | JenniferAkemi | is that correct? |
21:00.28 | unstable | yes |
21:00.55 | *** join/#asterisk qdk_ (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
21:00.56 | unstable | revengervn: what is your question? |
21:01.37 | revengervn | does anyone know what extension MP3Player application support ? |
21:01.37 | revengervn | can it play .wma or .asf |
21:01.51 | revengervn | ? |
21:01.56 | revengervn | anyone can help me? |
21:02.56 | revengervn | hi |
21:02.59 | revengervn | nice to see you here |
21:03.04 | revengervn | because |
21:03.18 | revengervn | I want asterisk to play streaming media |
21:03.19 | Ritzerisk | is there a simple way to configure a wildcard T1 i tried doing a ztcfg -vv |
21:03.24 | revengervn | so I wonder |
21:03.25 | Ritzerisk | but it saw no channells |
21:03.43 | revengervn | if MP3 player can handle other extensions like .asf |
21:03.58 | revengervn | or wma |
21:04.11 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
21:04.42 | Strom_M | Ritzerisk: did you set the appropriate settings in zaptel.conf? |
21:05.18 | Ritzerisk | so i would have to go there first to set it up as like 24 channel t1 |
21:05.27 | Strom_M | well, yeah |
21:05.38 | Strom_M | it doesnt just magically autoconfigure itself |
21:05.55 | Ritzerisk | ohh haha geesh |
21:06.48 | jameswf-home | our channel banks automagicly configure tham selves :) |
21:06.52 | Ritzerisk | it automatically did it for me in the elastix version so i was kinda huh |
21:07.19 | Ritzerisk | but theres no asterisk gui on this so hopefully i can install a overlapping gui on top of the vicidial gui too |
21:09.39 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
21:10.00 | Trapa | I need some helps with queues .. anyone out there a queue guru? |
21:10.18 | jjshoe | Trapa I would recommend asking a real question |
21:10.26 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
21:10.28 | Trapa | So ... My queue has several seconds of silence before it says anything to the customer .. Is there some way of changing that? |
21:10.46 | *** join/#asterisk l2cache (n=l2cache@117.178.101.97.cfl.res.rr.com) |
21:11.39 | datachomper | AGI Rx << EXEC Playtones dial |
21:11.40 | Ritzerisk | am i able to just do a yum -y update asterisk its currently at 1.2 |
21:11.44 | datachomper | <PROTECTED> |
21:11.59 | datachomper | Any reason why it would execute playtones, but no actually generate any sound? |
21:14.08 | *** join/#asterisk jmardonesk (n=jmardone@236-152-110.adsl.din.tie.cl) |
21:15.58 | Yourname` | Is there a good Canadian online store where we can buy VoIP products that anyone knows about? Preferably in Toronto? |
21:16.52 | *** join/#asterisk Nasra (n=maxshipp@190.166.71.163) |
21:17.01 | revengervn | Does anyone know PYTHON AGI framework except pyst? |
21:17.52 | jmardonesk | hi, all.. in a IVR system when I have a backgroung sound waiting for an extension number, the priority of this jump to the extension is always the lower as possible? |
21:18.14 | *** join/#asterisk delparnel (n=delparne@CPE001839f2889c-CM00137185dfb6.cpe.net.cable.rogers.com) |
21:24.36 | Trapa | Does anyone know what Spawn extension (gssphones, 604, 1) exited non-zero on 'Local/604@gssphones-7902,2' means |
21:27.41 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-94-228-tpr-esr-2.dynamic.isadsl.co.za) |
21:30.09 | *** join/#asterisk watchy (n=watchy@adsl-69-152-41-251.dsl.ltrkar.swbell.net) |
21:30.23 | watchy | whats the best way to pipe in FM radio to * for on hold music? |
21:31.48 | *** join/#asterisk angom_w (n=angom@201.170.65.143) |
21:31.52 | maqr | this whole "follow me" thing is very clever |
21:32.04 | Strom_M | watchy: bad idea unless you've got all the licensing and such already set up |
21:33.54 | maqr | Strom_M: i doubt anyone's ever got sued for that, i'd imagine you'd get c&d first anyway |
21:34.18 | maqr | watchy: i'd pirate a higher quality stream though, if you're going that route... don't want commercials playing on your hold music |
21:37.25 | *** join/#asterisk mvanbaak (n=michiel@vanbaak.xs4all.nl) |
21:38.42 | *** join/#asterisk fish-bulb (n=cstewart@216.207.245.1) |
21:39.57 | maqr | fish-bulb: i lol'd, good nick |
21:39.57 | *** join/#asterisk jembo_ (n=mannje@217.114.52.2) |
21:42.52 | *** join/#asterisk xipi (n=oliver@91-65-57-226-dynip.superkabel.de) |
21:42.55 | xipi | hi |
21:43.33 | *** part/#asterisk beek (n=klinebl@65.211.106.242) |
21:44.55 | *** join/#asterisk mercutioviz (n=chatzill@66-17-33-47.biz.visl.arrival.net) |
21:44.57 | xipi | i am looking for a way to allow users to call from a different account. example: user1 wants to use the line of user5. how can this be done? |
21:45.55 | xipi | if i am not mistaken, the setting would need to be made in the extensions.conf file |
21:47.03 | *** join/#asterisk l2cache (n=l2cache@97.101.178.117) |
21:47.45 | xipi | is there any place, where i can look it up? like some tutorial? |
21:48.17 | *** part/#asterisk Hydrant (n=aj@CPE0011950c737b-CM0012c90d1420.cpe.net.cable.rogers.com) |
21:48.50 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
21:50.24 | watchy | maqr: they want the local radio station |
21:50.49 | watchy | maqr: its a super small radio station + small company, aint no one gonna care |
21:51.28 | watchy | i just wanna know the best way to pipe in a FM station |
21:51.34 | maqr | watchy: well, i'm not 100% sure how to do it, but any sound card should work as a source... |
21:51.39 | maqr | you'd have to hook up a radio, of course |
21:51.42 | watchy | yea |
21:51.44 | maqr | it might look a little silly |
21:51.44 | maqr | heh |
21:51.49 | watchy | but i'm wondering how to get * to see it |
21:53.16 | watchy | can * play streaming mp3s? |
21:53.38 | riddlebox | watchy, yes |
21:54.19 | spokra | anyone have any wisdom on getting festival working |
21:54.24 | watchy | i guess put a soundcard in it, have some software stream from the soundcard to a mp3 |
21:54.35 | spokra | the howto on voip-info doesn;t work |
21:54.47 | riddlebox | watchy, you could do that |
21:54.52 | riddlebox | spokra, which distro? |
21:55.28 | spokra | <PROTECTED> |
21:55.28 | spokra | lenny/sid |
21:55.42 | watchy | riddle: is that the most sane/simplest way? |
21:55.53 | riddlebox | spokra, you could apt-get install festival |
21:55.58 | spokra | i did |
21:56.15 | spokra | and tried the change to scn file |
21:56.26 | watchy | anyone wanna write a XM radio internet streaming addon for *? |
21:56.29 | watchy | i'll pay you $25 |
21:56.29 | riddlebox | watchy, well the other was is to have a sound card with an input, then plug a cord from there to the headphone jack of the radio and use some app to play it |
21:57.05 | riddlebox | watchy, there are xm and sirius apps for mythtv, I am sure you can find one if you search for it |
21:57.24 | Qwell | watchy: can't just use madplay or whatever? |
21:57.29 | riddlebox | spokra, I use it on ubuntu and just apt-get install it and it works fine, whats the problem you have? |
21:57.34 | watchy | dunno what madplay is. |
21:57.39 | watchy | but ill goog it |
21:57.40 | jbeez | . |
21:57.40 | Qwell | surely it's just a "standard" stream. does it require a proprietary player? |
21:57.58 | watchy | hmm, it streams from the web in their own player thing |
21:58.09 | Qwell | so then it's likely a standard stream |
21:58.11 | watchy | no idea what type of steam it is. i know it requires a l/p |
21:58.20 | Qwell | check the source |
21:58.20 | spokra | pbx*CLI> |
21:58.20 | spokra | x7f; -- Attempting call on sip/14252810448@sip.broadvoice.com for 5555@local:2 (Retry 1) |
21:58.20 | spokra | x7f; == Using SIP RTP CoS mark 5 |
21:58.20 | spokra | x7f; > Channel SIP/sip.broadvoice.com-0829e990 was answered. |
21:58.20 | spokra | x7f; -- Executing [5555@local:2] Festival("SIP/sip.broadvoice.com-0829e990", "mary had a little lamb") in new stack |
21:58.22 | spokra | x7f; == Parsing '/etc/asterisk/festival.conf': x7f; == Found |
21:58.24 | spokra | x7f; == Spawn extension (local, 5555, 2) exited non-zero on 'SIP/sip.broadvoice.com-0829e990' |
21:58.32 | jbeez | hi watchy |
21:58.38 | watchy | sup mr jbeez |
21:58.39 | riddlebox | spokra, please pastebin it |
21:58.46 | spokra | sorry for the cut and paste |
21:58.51 | Ritzerisk | is there like a config tool for a zapata i cant seem to get the te122 T1 to work.... |
21:59.59 | *** join/#asterisk murdock_ut (n=chatzill@70.102.148.44) |
22:00.21 | riddlebox | Ritzerisk, there is zttest, zttool |
22:00.33 | maqr | could anyone recommend an SMS/MMS provider for the US? |
22:01.07 | spokra | http://pastebin.com/d15e4a39b |
22:01.33 | Ritzerisk | hmm |
22:02.08 | *** join/#asterisk l2cache (n=l2cache@117.178.101.97.cfl.res.rr.com) |
22:02.36 | watchy | hmm i guess * don't do mp3s anymore. |
22:02.41 | riddlebox | spokra, so you dial 5555 and want it to say mary had a little lamb? |
22:02.53 | riddlebox | watchy, I am using mp3 as moh |
22:03.05 | watchy | well it says 1.4 went to wav |
22:03.19 | maqr | ~sms |
22:03.20 | jbot | extra, extra, read all about it, sms is Stop Making Sense, the greatest concert film ever, starring Talking Heads and directed by Jonathan Demme. Send message to mobile phones via the internet for free. |
22:03.21 | maqr | tries the bot |
22:03.35 | maqr | nope, not useful |
22:04.07 | Corydon76-dig | maqr: T-Mobile ? |
22:04.08 | spokra | yep.. |
22:04.28 | Corydon76-dig | I'd go with them and a GSM modem |
22:04.41 | spokra | it tries to be festival doesn't like the config.. for asterisk |
22:04.54 | maqr | Corydon76-dig: err, yeah, i actually am on t-mobile.... but i mean, isn't there an SMS gateway or something i could just use? |
22:05.01 | maqr | Corydon76-dig: it'd be mighty silly to wire a phone up to the pbx |
22:05.11 | Corydon76-dig | maqr: for what direction? |
22:05.12 | *** join/#asterisk thepacmanfan (n=thepacma@12-218-140-147.client.mchsi.com) |
22:05.21 | Corydon76-dig | Sending? |
22:05.31 | spokra | are you in the us? |
22:05.42 | Corydon76-dig | 2342342345@tmomail.net |
22:06.00 | Qwell | I assume he means cheaper than $0.15c/min |
22:06.09 | maqr | Corydon76-dig: sending and receiving, ideally |
22:06.13 | Qwell | erm, -c |
22:06.22 | Corydon76-dig | You're going to need a GSM modem for receiving |
22:06.23 | maqr | Corydon76-dig: you'd think if someone sends an SMS to my DID, i could get it... somehow |
22:06.33 | Corydon76-dig | Not in the US you can't |
22:06.34 | spokra | there are cellular cards for PC.. that look like a modem.. and you can send sms thru them |
22:06.36 | Qwell | actually, chan_mobile supports SMS |
22:06.43 | *** join/#asterisk fish-bulb (n=cstewart@216.207.245.1) |
22:06.46 | Corydon76-dig | Oh, right, that... |
22:07.04 | maqr | seriously? you need a physical gsm connection to do ti? |
22:07.05 | maqr | *it |
22:07.18 | Qwell | there are some sms services I've seen people talk about on the asterisk-biz list |
22:07.25 | Corydon76-dig | maqr: you know what GSM modems are, don't you? |
22:07.28 | Qwell | doing it yourself is non-trivial though |
22:07.50 | Corydon76-dig | Little box, hooks up via serial port |
22:08.11 | maqr | Corydon76-dig: well, that implies you get cell service in the data center, right? |
22:08.12 | Corydon76-dig | Takes the same SIM card as a cell phone |
22:08.29 | maqr | Corydon76-dig: i'm on a VPS for my hosting anyway, i can't send them a modem to hook up to a serial port :p |
22:08.44 | Corydon76-dig | maqr: I've actually run one of those little things in an office, off an old Pentium 200MMX |
22:08.53 | spokra | why not send it via email to your cell phone? |
22:09.02 | maqr | hmm |
22:09.10 | Corydon76-dig | maqr: and then I ran openvpn between the two machines |
22:09.18 | maqr | Corydon76-dig: that's pretty clever |
22:09.34 | maqr | Corydon76-dig: how about if i wanted one of those short codes, like 6 digits or something? any idea how you get those? |
22:09.43 | Corydon76-dig | and then I passed messages via HTTP and a CGI script |
22:09.57 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
22:10.01 | Corydon76-dig | maqr: Are you willing to pony up the big bucks? |
22:10.02 | spokra | I worked for ATT in the SMS group good luck getting a short code |
22:10.14 | Qwell | spokra: oh? do tell |
22:10.20 | spokra | let alone a 6 digit |
22:10.23 | Corydon76-dig | maqr: I don't think they let them go for under $100k/mo |
22:10.24 | maqr | Corydon76-dig: no, i'd much prefer to pony up the very tiny bucks |
22:10.26 | maqr | wtf? |
22:10.28 | spokra | what do you want to know |
22:10.34 | Qwell | spokra: why? |
22:10.41 | maqr | Corydon76-dig: that's insane |
22:10.46 | Qwell | why good luck? why not 6-digit? |
22:10.48 | Yourname` | <PROTECTED> |
22:11.01 | Corydon76-dig | maqr: that's the amount of traffic that you need to generate in order to make it worth your while |
22:11.02 | Qwell | polycom |
22:11.23 | Yourname` | Qwell: One with backlist display and no PoE? |
22:11.23 | spokra | they don;t sell short codes but it's a pain in the A$$ to get cerified to connect to ther enetwork.. |
22:11.25 | Corydon76-dig | maqr: otherwise, you get yourself a phone number |
22:11.31 | maqr | Corydon76-dig: phone number it is! |
22:11.48 | maqr | Corydon76-dig: those scam services must make a lot of money selling jokes and ringtones though, to be able to afford that plus the tv commercials |
22:11.52 | *** join/#asterisk macros73 (n=cs@c-24-131-77-140.hsd1.pa.comcast.net) |
22:12.12 | Corydon76-dig | maqr: that is precisely why it is so expensive |
22:12.42 | Qwell | besides, there are only 1 million 6 digit codes (maximum) |
22:12.47 | maqr | Corydon76-dig: oh |
22:13.17 | spokra | the only way into the cell companies is to be a good o boy.. |
22:14.33 | spokra | the smpp protocal is available on the internet.. good luck writing it!! |
22:14.34 | Corydon76-dig | It's a simple matter of putting the price up to the point where the demand drops to only want what's available |
22:17.04 | maqr | makes sense to me |
22:17.17 | maqr | Corydon76-dig: still, you'd think that tmo or whoever could route sms to any DID |
22:17.31 | Corydon76-dig | maqr: they can |
22:17.32 | maqr | scammers can't be responsible for that not happening |
22:18.20 | Corydon76-dig | but there's a central SMS authority |
22:18.34 | maqr | that should make it even easier then |
22:23.26 | *** join/#asterisk bsaxon (n=bsaxon@71-8-14-108.dhcp.leds.al.charter.com) |
22:26.08 | watchy | aparently mplayer will stream xm |
22:26.22 | watchy | from console |
22:29.14 | Qwell | watchy: if mplayer can, it's likely that madplay can (see musiconhold.conf) |
22:30.08 | watchy | thanks |
22:30.16 | *** join/#asterisk exothermc (n=miles@74.85.89.146) |
22:30.20 | watchy | i gotta log off but i'm gonna check that out |
22:30.22 | watchy | im heading home |
22:30.39 | exothermc | what is a recommended SIP client that does video for linux? |
22:32.08 | Ritzerisk | ahhhh i did a modprobe and it said fatal error not found |
22:32.23 | jjshoe | when shortcodes first came out they where roughly 10k a month.. |
22:32.56 | jjshoe | or that was my experience anywho |
22:33.01 | jjshoe | dunno now, it's been years. |
22:33.47 | *** join/#asterisk bsaxon (n=bsaxon@71-8-14-108.dhcp.leds.al.charter.com) |
22:34.07 | jjshoe | but if you're just looking to send sms's just sign up to one of many available pay per message sms gateways, many even accept emails |
22:34.29 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
22:35.33 | jjshoe | I'm pretty sure most of them will handle responses as well |
22:41.54 | *** join/#asterisk [TK]D-Fender (i=Joe@64.235.218.194) |
22:43.34 | *** join/#asterisk bsdwarrior (n=mfahey@fahey.enter.net) |
22:43.44 | bsdwarrior | how do you re register with your iax server? |
22:43.49 | bsdwarrior | I forget the command |
22:46.11 | bsdwarrior | iax2 reload ? |
22:48.30 | bsdwarrior | anyone ? |
22:49.08 | seanbright | bsdwarrior: when you type 'iax2 reload' into the asterisk CLI and hit enter... what happens? |
22:49.31 | bsdwarrior | it loads the template, but I dont see it register |
22:49.49 | seanbright | ok, then iax2 reload isn't the command you want |
22:49.54 | seanbright | hope that was helpful |
22:49.56 | seanbright | :) |
22:50.09 | bsdwarrior | I have the config in iax.conf |
22:50.42 | drmessano | asterisk -rx reload |
22:50.43 | drmessano | Done |
22:50.52 | seanbright | service asterisk restart |
22:50.53 | seanbright | heh |
22:51.18 | drmessano | service trixbox restart |
22:51.29 | seanbright | halt |
22:51.54 | bsdwarrior | its not asterisk, its my crappy provider |
22:52.23 | drmessano | so? |
22:57.44 | [TK]D-Fender | Darn just a little slow to tell him a trick to get it to re-register.. |
22:58.09 | seanbright | share the wealth |
22:59.29 | seanbright | [TK]D-Fender: ? |
23:00.07 | [TK]D-Fender | seanbright, Where do you think * stores its registry info? |
23:00.22 | seanbright | dbm |
23:00.27 | jjshoe | Qwell bahahaha, what a bozo. |
23:00.28 | [TK]D-Fender | AstDB <-- |
23:00.36 | *** part/#asterisk exothermc (n=miles@74.85.89.146) |
23:00.48 | Qwell | I should've had drmessano join.. |
23:00.55 | [TK]D-Fender | seanbright, So relatively easy to kill off the key, issue a reload and watch it rereg |
23:01.19 | seanbright | [TK]D-Fender: indeed. good call. |
23:05.28 | [TK]D-Fender | Kernel upgrade. Stepping out to recompile stuff. |
23:05.30 | [TK]D-Fender | BBIAB |
23:07.12 | *** join/#asterisk Frogzoo (n=Frogzoo@124.184.33.9) |
23:08.15 | jjshoe | Qwell you've got me thinking you sound like you know wtf is going on |
23:08.32 | Qwell | jjshoe: GOOGLE |
23:08.34 | Qwell | :) |
23:14.00 | Qwell | jjshoe: you didn't think I could pull that off, did you? |
23:14.05 | *** join/#asterisk RoyK (n=roy@ip-26-13-149-91.dialup.ice.no) |
23:16.01 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
23:16.50 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-40-133.lns10.syd7.internode.on.net) |
23:18.22 | jjshoe | Qwell I didn't realize you were going to be so dedicated into fooling him :D |
23:18.37 | jjshoe | although it's friday, and I've had enough of javascript, so I am looking at houses in SD |
23:20.39 | jjshoe | anything on friday before a long weekend but work.. |
23:21.50 | *** join/#asterisk mackes (n=root@cpe-24-198-43-238.buffalo.res.rr.com) |
23:22.15 | mackes | Mackes is in. |
23:29.13 | *** join/#asterisk [TK]D-Fender (i=Joe@64.235.218.194) |
23:40.12 | *** join/#asterisk anthm (n=anthm@mbf0736d0.tmodns.net) |
23:51.34 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
23:56.31 | *** join/#asterisk angom (n=angom@201.170.65.143) |