| 00:09.53 | *** part/#asterisk asr33 (n=asr33@64.56.252.72) |
| 00:10.00 | *** join/#asterisk RoyKa (n=roy@91.149.3.241) |
| 00:14.31 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
| 00:17.40 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
| 00:21.32 | *** part/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 00:24.19 | *** join/#asterisk thevoipforum_org (n=hex@87-194-8-43.bethere.co.uk) |
| 00:24.26 | *** part/#asterisk mattwj2005 (n=Matt@75-168-139-194.mpls.qwest.net) |
| 00:29.30 | mamep | how can i change my default audio codec for g729a? |
| 00:30.42 | ManxPower | mamep: How many G729 licenses do you have? |
| 00:30.51 | ManxPower | mamep: G729 IS a codec. |
| 00:31.30 | mamep | w many G729 licenses do you have? |
| 00:31.30 | mamep | [02:30am] [ManxPower] mamep: G729 IS a codec. |
| 00:31.37 | mamep | hmm g729a |
| 00:32.31 | ManxPower | g729 is basically the same as g729a |
| 00:34.08 | ManxPower | you need a license to use g729 |
| 00:34.57 | mamep | by default installed with asterisk |
| 00:35.21 | ManxPower | there is no g729 in default asterisk |
| 00:35.50 | Corydon76-dig | You can download it, but you can't use it unless you purchase a set of licenses |
| 00:36.24 | Corydon76-dig | Or more correctly, you can't LEGALLY use it without purchasing licenses |
| 00:37.10 | mamep | ah |
| 00:37.11 | mamep | ok |
| 00:37.16 | mamep | so no need to work with it.. |
| 00:37.26 | mamep | i will concentrate for the callerid for now |
| 00:37.29 | Corydon76-dig | It's only $10/seat |
| 00:39.32 | mamep | yeah i know that |
| 00:39.53 | mamep | but i don't think if g729 is suitable for me.. |
| 00:39.58 | mamep | too much packets |
| 00:40.56 | Corydon76-dig | Heh, too much? |
| 00:41.23 | mamep | greek dsl isp's are not the best ones.. |
| 00:41.24 | Corydon76-dig | It's the second smallest voice codec without going Mr. Roboto |
| 00:42.08 | Corydon76-dig | If it wasn't good, would anybody pay for it? |
| 00:47.13 | nephfl | my php agi script doesnt seem to be reading in the variables from asterisk and im not sure why |
| 00:47.46 | ManxPower | The primary advantage of G729(a) is low bandwidth usage when communicating to a hardphone. |
| 00:48.13 | sbingner | grr. I can't find a good way of determining all the prefixes that are 'local' to me |
| 00:52.08 | sbingner | woo found it |
| 00:54.06 | drmessano | G729 licenses only apply for transcoding, correct? |
| 00:55.37 | ManxPower | drmessano: Correct. |
| 00:57.27 | *** join/#asterisk bkruse (n=bkruse@76.73.154.120) |
| 00:57.27 | *** mode/#asterisk [+o bkruse] by ChanServ |
| 00:57.45 | *** join/#asterisk cl4sh (n=cl4sh@jaracz.ds.pg.gda.pl) |
| 00:57.48 | cl4sh | hi all |
| 00:58.07 | cl4sh | ive got some questions about sipp |
| 00:58.09 | drmessano | Thats what I thought |
| 00:58.59 | cl4sh | when i try to simulate user agent connection |
| 00:59.16 | *** join/#asterisk VJFROMGT (n=vjfromgt@user-387g9sf.cable.mindspring.com) |
| 00:59.26 | cl4sh | there si an error looks like |
| 00:59.39 | cl4sh | The auto_media_port keyword requires PCAPPLAY |
| 00:59.55 | cl4sh | i dont know wht i do wrong |
| 01:00.36 | cl4sh | os anybody able to help me/ |
| 01:00.38 | cl4sh | ? |
| 01:01.25 | hmmhesays | does spandsp really need gsm support? |
| 01:02.09 | *** join/#asterisk cl4sh (n=cl4sh@jaracz.ds.pg.gda.pl) |
| 01:03.16 | hmmhesays | I hope not |
| 01:06.54 | VJFROMGT | i have a trunk with vitelity, all of a suden its giving 404 errors |
| 01:07.05 | VJFROMGT | but if i stick credentials in xlite it works |
| 01:12.54 | hmmhesays | odd |
| 01:12.57 | hmmhesays | my vitelity is working fine |
| 01:12.57 | ManxPower | VJFROMGT: I didn't know vitelity supported IAX2 trunking |
| 01:13.09 | ManxPower | 404 means "I don't know ANYTHING about that number." |
| 01:13.47 | ManxPower | ~trunk |
| 01:13.47 | jbot | hmm... trunk is is a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. |
| 01:14.20 | Qapf | SIP Trunk |
| 01:14.24 | hmmhesays | yeah we know what he means |
| 01:14.35 | hmmhesays | you have a sip account with vitelity and your asterisk isn't registering |
| 01:14.59 | ManxPower | actually he said "404", he did not indicate when he gets the error. |
| 01:15.09 | VJFROMGT | <ManxPower> vitel does support iax but i dont use iax just sip |
| 01:15.20 | hmmhesays | they recommend that you don't |
| 01:15.23 | ManxPower | VJFROMGT: There is no such thing as a "SIP trunk" |
| 01:15.33 | hmmhesays | here we go... |
| 01:15.34 | VJFROMGT | ok,, then i am using sip |
| 01:15.40 | ManxPower | good. |
| 01:16.07 | ManxPower | hmmhesays: I also correct people when they call a web site "the internet". I assume you do not? |
| 01:16.18 | VJFROMGT | when i do a sip show show registry it does nto show up |
| 01:16.42 | ManxPower | VJFROMGT: that is because sip show registry shows devices registerd TO Asterisk. |
| 01:16.46 | ManxPower | you want "sip show peers" |
| 01:17.14 | VJFROMGT | sip show peers gives me Vitelity/[MYUSERNAME] 64.2.142.29 5060 OK (56 ms) |
| 01:17.33 | ManxPower | so you do not have a registration problem. |
| 01:17.49 | ManxPower | chances are you have a dialplan problem |
| 01:17.58 | *** join/#asterisk MooingLemur (n=troy@unaffiliated/mooinglemur) |
| 01:18.08 | VJFROMGT | SIP/2.0 404 Not Found |
| 01:18.08 | VJFROMGT | Via: SIP/2.0/UDP 208.75.208.90:5060;branch=z9hG4bK65f2823e;received=208.75.208.90;rport=5060 |
| 01:18.08 | VJFROMGT | From: "Unknown" <sip:Unknown@208.75.208.90>;tag=as3335a0f4 |
| 01:18.08 | VJFROMGT | To: <sip:inbound1.vitelity.net>;tag=as4745e00f |
| 01:19.29 | VJFROMGT | outbound works but no inbound. when i do a sip debug ip 64.2.142.29 and make a call to my did , i dont see anything in cli |
| 01:27.18 | hmmhesays | well time to try my spandsp frankenstein |
| 01:29.21 | hmmhesays | if this works i'm treating myself to a beer |
| 01:31.06 | VJFROMGT | anyone got suggestions? |
| 01:31.40 | jblack | Woot. My dundi network might be picking up hawaii and switzerland. |
| 01:33.45 | hmmhesays | wtf there is no wget on this blackfin |
| 01:38.43 | hmmhesays | its saturday I'm more laid back on the weekend ManxPower |
| 01:39.31 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-6-214.lns10.syd7.internode.on.net) |
| 02:04.44 | ManxPower | What is the CLI output when the Dial happens |
| 02:14.37 | *** join/#asterisk superwhitea (n=superwhi@ip68-2-227-46.ph.ph.cox.net) |
| 02:37.38 | *** join/#asterisk bkruse (n=bkruse@76.73.154.120) |
| 02:37.38 | *** mode/#asterisk [+o bkruse] by ChanServ |
| 02:38.23 | *** join/#asterisk cesar_CR (n=cesar@celord.ice.co.cr) |
| 02:40.32 | drmessano | yawn |
| 02:46.07 | bkruse | [TK]D-Fender: http://bkruse.com |
| 02:49.21 | drmessano | bkruse.. |
| 02:49.31 | bkruse | drmessano: yes? |
| 02:49.39 | drmessano | Explain to me what youve done there |
| 02:49.49 | drmessano | Im familiar with the OpenMoko, or so I thought |
| 02:49.57 | bkruse | drmessano: yes, and? |
| 02:50.08 | bkruse | I am....very familiar also. |
| 02:50.14 | drmessano | Whats the utility of getting asterisk to install there? |
| 02:50.29 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 02:50.40 | bkruse | drmessano: a sample alsa console channel I am using to base my iaxclient of off |
| 02:50.42 | bkruse | off of* |
| 02:50.49 | drmessano | Ah got ya |
| 02:50.58 | *** join/#asterisk vitaminmoo (n=vitaminm@70.58.177.109) |
| 02:51.10 | bkruse | not to mention, that you can use asterisk as whatever you want if you can route calls from gsm -> asterisk |
| 02:51.23 | drmessano | Interesting |
| 02:51.28 | bkruse | eg, grab calls, voice menu's, meetme |
| 02:51.41 | bkruse | but there is not real reason besides saying you have a pbx in your pocket,. it really isnt practical |
| 02:52.20 | drmessano | I didnt know if you were using as a framework, or if there was another device maybe you were targeting |
| 02:52.23 | bkruse | But I can use chan_alsa to base my alsa hook off of, which is my main reason for seeing if it compiled, passed calls |
| 02:52.40 | bkruse | nah, well, it works for openmoko, so any device that will us OM will be able to install this package |
| 02:52.55 | drmessano | Having asterisk + meetme on the phone blows the shit out of 3 way calling |
| 02:53.10 | bkruse | totally. Performance wise? Who knows, I will try it out tomorrow |
| 02:53.16 | drmessano | lol |
| 02:53.23 | bkruse | I got the GUI working on it, and registered some sip phones to it |
| 02:53.33 | bkruse | totally random and pointless, but so freaking cool |
| 02:53.51 | drmessano | Actually.... |
| 02:54.00 | drmessano | The iPhone stores VM locally, doesnt it? |
| 02:54.13 | bkruse | Actually, for my hacked tmobile iphone, nope |
| 02:54.27 | bkruse | I can actually retreive other peoples voicemail using asterisk on my moko, I tried it..lol |
| 02:54.31 | drmessano | lol |
| 02:54.44 | bkruse | It does not prompt for a password, just checks your callerid |
| 02:54.49 | drmessano | I was just thinking.. with VM on the phone with asterisk, it was comparable |
| 02:54.58 | bkruse | agreed, it would be super cool |
| 02:55.12 | bkruse | you could like do meetme, or even queues, music on hold? |
| 02:55.22 | drmessano | MOH would be hilarious |
| 02:55.42 | bkruse | lol seriously |
| 02:55.50 | drmessano | more I think about it.. queues.. |
| 02:55.51 | bkruse | "hold on, brb" music on hold |
| 02:55.57 | bkruse | wtf? I thought this was a cell phone |
| 02:56.11 | drmessano | Calling a mobile and going into a Queue.... thats very FTW |
| 02:56.16 | bkruse | lol |
| 02:56.51 | drmessano | "Your call is very important to me.." |
| 02:56.56 | drmessano | "Please stand by" |
| 02:56.58 | bkruse | haha! |
| 02:57.04 | florz | *ggg* |
| 02:57.07 | bkruse | at the tone, please leave your number and message |
| 02:57.19 | bkruse | A ticket has been opened in your name |
| 02:57.22 | drmessano | HAHA |
| 02:57.58 | florz | callcenter-in-a-phone |
| 02:58.05 | bkruse | please go to double-u double-u double-u dot bkrusephone door com |
| 02:58.09 | drmessano | OMG.. backup callcenter |
| 02:58.15 | bkruse | ! |
| 02:59.18 | bkruse | Anyways, goal is to get an iaxclient. OpenMoko uses libpulseaudio instead of directly binding to alsa |
| 02:59.48 | drmessano | Hmmm |
| 03:00.09 | drmessano | You could load up a bunch of SIP peers and IAX trunks and have some super flexible calling |
| 03:00.18 | drmessano | Moreso than any other mobile.. ever |
| 03:00.39 | bkruse | Agreed. Not to mention edit the users and trunks on the fly in the web interface :P |
| 03:01.19 | drmessano | Easy rebooting of your asterisk box when youre away from home |
| 03:01.26 | drmessano | Hold Off for 5 seconds |
| 03:01.57 | bkruse | my ultimate goal is so I can walk around and use GPRS to make all my calls through my voip providder |
| 03:02.54 | drmessano | Thats awesome |
| 03:03.38 | drmessano | I have to admit.. I love Linux GUI install routines |
| 03:03.44 | drmessano | Lots of fluff and buttons |
| 03:03.50 | drmessano | Cool graphics |
| 03:03.52 | drmessano | Reboot |
| 03:03.57 | drmessano | Blinky command line |
| 03:04.01 | drmessano | *sigh* |
| 03:04.12 | drmessano | Grantid, not installing X |
| 03:04.25 | bkruse | dpkg-reconfigure xserver-xorg |
| 03:04.30 | bkruse | <3 debian |
| 03:04.33 | *** join/#asterisk adjohn (n=adjohn@pc-a80127.wlan.inet.fi) |
| 03:04.33 | drmessano | lol |
| 03:05.18 | drmessano | Im slowly taking the time to blow up every windows server box I have the power life to have an effect on |
| 03:05.27 | drmessano | power in life* |
| 03:05.45 | bkruse | good |
| 03:05.54 | bkruse | I did the same at my school, overnight, and no one noticed |
| 03:06.09 | bkruse | Being sys admin has its advantages, especially in mid-terms.... |
| 03:06.32 | drmessano | You didnt pull 20 miles of Cat 6 under a tennis court and go to work making french fries later, did you? |
| 03:06.46 | bkruse | Never, ever |
| 03:06.49 | drmessano | lol |
| 03:06.50 | bkruse | I hate laying cat * |
| 03:06.51 | bkruse | lol |
| 03:07.04 | bkruse | I just say "oh....i'll do the configuration" |
| 03:07.09 | drmessano | mrdigital was here earlier.. thats jblacks buddy |
| 03:07.15 | bkruse | gotcha |
| 03:07.20 | drmessano | Reminded me of his presence |
| 03:07.28 | drmessano | The School thing |
| 03:08.08 | bkruse | ugh, school.... |
| 03:08.19 | drmessano | lol |
| 03:08.52 | WilliamK | I remember school daze... |
| 03:09.17 | WilliamK | someone was unhappy at my appreciation to put the entire bldg through a winproxy because they couldn't give me enough ips... |
| 03:09.44 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-86-191.lns10.syd6.internode.on.net) |
| 03:10.09 | drmessano | I remember school... dBase IV |
| 03:10.15 | WilliamK | all they wanted to give me was a /28 for the entire bldg |
| 03:10.51 | WilliamK | oh and we as students did take turns wiring the bldg |
| 03:11.00 | WilliamK | that was fun (not) |
| 03:13.24 | *** join/#asterisk brain- (n=bastard@201-89-68-249.ctame700.dsl.brasiltelecom.net.br) |
| 03:13.54 | WilliamK | the teacher tech tried to give me a 0 on final exam that he wouldn't give me the exam for even though I showed up.... boy oh boy was that an interesting last day of school... next year the dean comes over from my other school that I've known for years as well as the administrators... showed them the nice report card and explained what he did and all hell broke loose... |
| 03:14.09 | WilliamK | that was my most interesting year of school |
| 03:34.29 | jblack | Has anyone else read Ulrich Dreppers "What every programmer should know about memory?" |
| 03:36.08 | troubled | you loose it as you get older? |
| 03:36.08 | *** join/#asterisk wglenncamp (n=wglennca@c-68-63-244-93.hsd1.ky.comcast.net) |
| 03:37.03 | wglenncamp | Why would paging lock up an asterisk appliance. The paging works with my polycom phones, but then locks everything up for about 30 seconds. Any ideas? |
| 03:37.35 | wglenncamp | Actually, it's more like a minute... |
| 03:37.41 | troubled | sounds like its waiting for something to timeout, doesn't it |
| 03:37.57 | troubled | a lookup, dns, something tcp perhaps? |
| 03:38.12 | *** join/#asterisk r0d3nt (i=nobody@pinky.ratman.org) |
| 03:38.18 | wglenncamp | Ya, but why? I would think that as soon as the phone making the page hangs up, it would simply release everything |
| 03:38.27 | troubled | you know how things have like 10sec timeout, with 3 retries and all that stuff |
| 03:38.28 | wglenncamp | DNS? I wouldn't think so. |
| 03:38.44 | troubled | well, there is dundi and e164 stuff |
| 03:39.16 | wglenncamp | It's holding the channels even after the originator hangs up. |
| 03:39.26 | troubled | my other guess was hardware stall like when windows goes to lunch when it cant read a cd/dvd |
| 03:40.28 | troubled | *shrug* what version of * and what does your dialplan look like etc. |
| 03:41.07 | troubled | doubt i can be of much more help myself. then again, reading something new is usually easier than writing it |
| 03:41.24 | wglenncamp | It's running 1.4 (It's an 8FXO appliance)... |
| 03:41.47 | troubled | aww, is that one of those 8port tdm cards from digium? |
| 03:42.01 | wglenncamp | No. An asterisk appliance. |
| 03:42.11 | wglenncamp | With the 8FXO ports |
| 03:42.19 | troubled | usb based? |
| 03:42.30 | wglenncamp | http://www.digium.com/en/products/appliance/ |
| 03:42.42 | troubled | i don't exactly know all the fancy gizmo's out there like you guys remember \:) |
| 03:42.43 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-27-128.lns10.syd7.internode.on.net) |
| 03:42.58 | wglenncamp | My company sells them. I just have a customer that is needing some custom crap done to it. |
| 03:43.22 | wglenncamp | ie.. paging. ;) |
| 03:43.42 | troubled | well, i know 1.2 seems to be ok here. but ive no idea what your setup is like compared to mine |
| 03:44.07 | troubled | lucky me has a built in speaker in the pbx as well <3 IBM |
| 03:44.31 | wglenncamp | This is going in a pharmacy that requires pagins on all of the IP501's |
| 03:44.43 | wglenncamp | Paging is working fine, just having this small issue |
| 03:45.00 | wglenncamp | oh, and a couple of 320's |
| 03:45.26 | troubled | where is the dialplan is it hanging? begin, end, random, other? |
| 03:49.59 | *** join/#asterisk wglenncamp (n=wglennca@c-68-63-244-93.hsd1.ky.comcast.net) |
| 03:50.14 | wglenncamp | duh... I am connected to the appliance, and it locked up. HAHA |
| 03:51.20 | *** join/#asterisk wunderkin (n=wunderki@216-19-219-35.getnet.net) |
| 03:51.32 | troubled | 'my supposed to laugh too? |
| 03:52.13 | wglenncamp | I thought it was funny.... I was trying to reply to something, and couldn't Gotta laugh to keep from going CRAZY! |
| 04:08.34 | *** join/#asterisk LancerZ (n=Anon@CPE-72-135-235-232.wi.res.rr.com) |
| 04:09.47 | LancerZ | Does anyone know if there is a difference in config files in the polycom 320 and the polycom 501??? |
| 04:10.00 | LancerZ | In terms of the FTP aspect of it |
| 04:10.02 | bkruse | LancerZ: I do not believe there is |
| 04:10.34 | LancerZ | bkruse: so a standard config file from asterisktutorials or whatever would be fine? |
| 04:11.05 | bkruse | LancerZ: yes, actually, check out res_phoneprov in asterisk trunk, those config files used (in the documentation) will work for both. |
| 04:13.03 | bkruse | wunderkin: about what? |
| 04:13.14 | wunderkin | i have been gone from here for like forever |
| 04:13.33 | bkruse | wunderkin: agreed, where you been? |
| 04:13.44 | wunderkin | here but not here! |
| 04:14.26 | wunderkin | i had like 18k mailing list messages.. or more.. i forget |
| 04:15.42 | LancerZ | bkruse: sorry for the delay in response. Thanks for your advice. Will check that out now |
| 04:15.56 | bkruse | LancerZ: It is a new thing released in trunk, which seriously rcks... |
| 04:20.13 | LancerZ | Uh, what is trunk? |
| 04:20.32 | jblack | LancerZ: The development version of asterisk |
| 04:23.13 | LancerZ | jblack: is there a url for this? I am trying to access "trunk" that bkruse referred me to on asterisk tutorials. More specifically config files for polycom. Can you advise? |
| 04:23.40 | LancerZ | astrisktutorials.com <--- |
| 04:26.52 | [TK]D-Fender | LancerZ, just stick with your standalone guide |
| 04:27.33 | *** join/#asterisk esaym (n=user@cpe-72-183-198-134.satx.res.rr.com) |
| 04:27.53 | esaym | this is deprecated in 1.4 right? exten => 88,1,VoicemailMain(${CALLERIDNUM}) because it doesn't seem to work |
| 04:28.13 | [TK]D-Fender | esaym, No, that was deprecated in 1.2 <- |
| 04:29.18 | LancerZ | I am actually doing this for my company I just started. I don't have any guides, and am trying to learn about this. |
| 04:29.23 | Corydon76-dig | Try ${CALLERID(num)} |
| 04:30.00 | Corydon76-dig | LancerZ: http://svn.digium.com/view/asterisk/trunk/ |
| 04:30.13 | Corydon76-dig | change "view" to "svn" to do an svn checkout |
| 04:30.25 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-50-170.lns10.syd7.internode.on.net) |
| 04:30.34 | ManxPower | I have polycom configs at http://www.fnords.org/~eric/polycom-config-examples/ |
| 04:30.53 | esaym | thank you Corydon76-dig |
| 04:30.58 | Corydon76-dig | ManxPower: he may be looking for res_phoneprov |
| 04:32.50 | LancerZ | Thanks guys. I'm going to check these config files out now |
| 04:32.59 | LancerZ | I appreciate all the help/advise |
| 04:34.10 | [TK]D-Fender | LancerZ, I advise you provision them yourself as you were intending to. |
| 04:36.23 | LancerZ | Well, I am intending to config them, so if that's what you mean by provisioning then I am planning on doing that. I am basically editing config files. Or that's what I am hoping to do |
| 04:36.28 | phix | hmmm echo on all calls from analog phone -> Supura 3102 -> Internetz |
| 04:38.00 | [TK]D-Fender | LancerZ, Good, then grab the admin guide and your firmware at get to it. Those 2 phones can be configured damn near identically with the execption of how you'll distribute line-keys |
| 04:38.20 | [TK]D-Fender | phix, what exactly is on the other end of "internetz"? |
| 04:39.40 | phix | [TK]D-Fender: SIP service |
| 04:39.52 | [TK]D-Fender | phix, DETAILS please..... |
| 04:40.19 | phix | [TK]D-Fender: what do you want to know about it? |
| 04:40.41 | [TK]D-Fender | phix, what EXACTLY are you connecting to on each end. |
| 04:40.42 | phix | I pay money for an account with some company so I can make calls to landlines via Internet using SIP |
| 04:41.14 | [TK]D-Fender | phix, you are using an analog phone on the FXS port of your SPA and getting echo? |
| 04:41.56 | phix | yes |
| 04:42.17 | phix | I can hear my self talk, at a lower volume but I can still hear the feedback |
| 04:42.25 | jblack | that's echo. |
| 04:42.35 | phix | yup |
| 04:42.53 | [TK]D-Fender | phix, if we're talking the FXS port, odds are it means your provider sucks |
| 04:43.13 | jblack | Can you list every piece of equipment from "what your mouth talks into" to the guy on the other side? |
| 04:43.22 | phix | yes sir |
| 04:43.23 | jblack | [TK]D-Fender: Hmm. Remember how back when I started, I got nasty echo from turning echo cancellation _on_ ? |
| 04:43.52 | jblack | perhaps he turned it on premptively and is canceling something that's not there. |
| 04:44.31 | phix | Analog phone -> Sipura -> Switch -> Asterisk Box -> ADSL Modem -> SIP Provider |
| 04:45.47 | [TK]D-Fender | phix, "sip provider" sucks |
| 04:45.49 | phix | I tried Analog phone -> PBX -> Same Sipura -> Switch -> Asterisk Box -> ADSL Modem -> Same SIP Provider, that had no echo |
| 04:46.12 | jblack | Ok. Go everywhere you can control, and turn echo cancellation off please. |
| 04:46.21 | phix | ok |
| 04:46.36 | phix | Asterisk has no echo cancellation for SIP ay |
| 04:46.38 | jblack | Then, if necessary, we'll one on at a time. |
| 04:46.55 | phix | The only device that has echo cancelation is Sipura |
| 04:47.03 | phix | but I think that is on FXO port |
| 04:47.27 | jblack | I believe the SPAs have an echo cancellation option |
| 04:47.34 | [TK]D-Fender | phix, No, it can AEC the FXS but thats almost never any kind of issue |
| 04:47.41 | phix | ok |
| 04:49.00 | phix | network jitter lvl: High, Jitter buffer adjustmant: up and down |
| 04:49.48 | [TK]D-Fender | phix, no issue probably. jitter != echo |
| 04:50.00 | [TK]D-Fender | phix, Go call somewhere else with your SPA to test |
| 04:50.15 | [TK]D-Fender | phix, remove the obvious suspect from the equation. |
| 04:51.02 | phix | ok this is on line (FXS) Echo Canc Enable: Yes, Echo Canc Adapt Enable: Yes, Echo Supp Enable: Yes |
| 04:52.13 | phix | [TK]D-Fender: ok, it does it on all calls |
| 04:52.26 | [TK]D-Fender | phix, to where else? |
| 04:52.41 | phix | oh a diff SIP provider? |
| 04:52.47 | [TK]D-Fender | phix, DUH! |
| 04:53.10 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-22-180.lns10.syd7.internode.on.net) |
| 04:53.38 | phix | [TK]D-Fender: sure, what's the IP, user name and password? :) |
| 04:55.20 | *** join/#asterisk the_5th_wheel (n=edd@dsl-242-188-170.telkomadsl.co.za) |
| 04:56.27 | the_5th_wheel | hi. i have a cisco 7960 with a sidecar 7914, and it seems to not send or receive audio to a dialed party. im using sscp since the sidecar wont work with sip |
| 04:59.06 | RypPn | what version of asterisk? |
| 04:59.48 | the_5th_wheel | gimme a sec |
| 05:00.14 | the_5th_wheel | its an old one, but the boss wont let me upgrade it |
| 05:00.32 | phix | [TK]D-Fender: hmm ok it does it from sipura -> asterisk -> softphone |
| 05:00.36 | RypPn | I just wondered if it was 1.2 or 1.4 |
| 05:00.56 | the_5th_wheel | 1.0.9 |
| 05:01.39 | [TK]D-Fender | phix, hate to say I wouldn't trust your softphone. |
| 05:01.52 | phix | yeah |
| 05:01.55 | [TK]D-Fender | phix, feel free to try the option I PM'd you |
| 05:02.03 | phix | ok, thank you |
| 05:02.16 | RypPn | the_5th_wheel: is it chan_sccp or chan_skinny? |
| 05:02.18 | phix | just need to add that to asterisk as an extension |
| 05:03.30 | [TK]D-Fender | phix, yes |
| 05:05.08 | *** join/#asterisk BBHoss (n=hoss@c-71-207-173-38.hsd1.al.comcast.net) |
| 05:13.33 | the_5th_wheel | RypPn: Chan_scco |
| 05:13.35 | the_5th_wheel | *sccp |
| 05:14.39 | RypPn | the_5th_wheel: have you tried changing the codec in use to see if it makes any difference? |
| 05:15.27 | the_5th_wheel | i will try now |
| 05:15.51 | the_5th_wheel | where would i change that? |
| 05:16.20 | RypPn | in the sccp.conf file |
| 05:17.19 | RypPn | in the [general] section |
| 05:17.50 | the_5th_wheel | i have alaw ulaw and g729 |
| 05:18.02 | the_5th_wheel | wich has the biggest probability of working? |
| 05:20.04 | RypPn | well, I get no audio over pstn regardless of codec, but you may have more joy, I'm using a different version of chan_sccp from you probably... my issue is with chan-sccp-b from sourceforge, I assume your using the berlios version |
| 05:21.03 | jameswf-home | I heard chan sccp support was a myth.. the .c file simply says hello world |
| 05:21.22 | RypPn | nah, works fine over voip |
| 05:21.25 | BBHoss | heh |
| 05:22.47 | the_5th_wheel | mine doesnt even work over voip |
| 05:22.59 | the_5th_wheel | and yes, it seems im using the berlios one |
| 05:26.42 | jblack | Wow. I didn't know that a cache hit resulting in hitting main meory cost 120 cycles. |
| 05:27.38 | [TK]D-Fender | RypPn, then its not your phone. |
| 05:28.02 | RypPn | [TK]D-Fender: please elaborate |
| 05:28.32 | [TK]D-Fender | RypPn, set for a/u law If either works for one, it'd better work for the other |
| 05:36.14 | the_5th_wheel | the 7914 part of my phone, seems to not want to update its speeddials. where would i change that? since ive chaged it in the sscp thin |
| 05:36.19 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-76-175-71-252.socal.res.rr.com) |
| 05:36.22 | *** join/#asterisk Agrajag- (n=filip@c211-30-185-177.artrmn2.nsw.optusnet.com.au) |
| 05:36.53 | RypPn | [TK]D-Fender: http://tinyurl.com/38hnp7 describes my issue, my personal feeling is there are some hooks missing to zaptel |
| 05:36.54 | Agrajag- | g'day. is it possible to have some sort of database of numbers/names so asterisk can rewrite the caller id from the number to name? |
| 05:37.20 | [TK]D-Fender | Agrajag-, Yes |
| 05:37.56 | Agrajag- | [TK]D-Fender: great. are you able to point me to documentation on how this can be done? |
| 05:37.59 | BBHoss | Agrajag-, there are a lot of CID name lookup scripts out there |
| 05:38.13 | Agrajag- | ok cool, i just wasn't sure of the term to search for |
| 05:39.07 | jblack | You can even make it yourself with a couple lines in your dialplan with embedded sql statements |
| 05:40.03 | [TK]D-Fender | Agrajag-, "show function DB", "show application agi" |
| 05:40.19 | Agrajag- | cheers |
| 05:40.23 | [TK]D-Fender | Agrajag-, or any of the 1/2 dozen other DB access methods |
| 05:40.25 | BBHoss | Agrajag-, this might work http://www.monetra.com/~brad/callerid_shell.agi |
| 05:40.29 | drmessano | Can I make an AGI that gets a dwarf to crawl out of the broom closet and check the phonebook? |
| 05:41.02 | RypPn | sup danny :) |
| 05:41.05 | [TK]D-Fender | drmessano, No, I've enslaved him to manage the little lamp in my fridge! |
| 05:41.08 | Corydon76-dig | I don't know, can you? |
| 05:41.18 | RypPn | still swarf-bashing I see ;) |
| 05:41.24 | RypPn | dwarf* |
| 05:41.37 | [TK]D-Fender | Killer Dwarfs rock! |
| 05:41.57 | Corydon76-dig | "dwarves" |
| 05:43.18 | *** join/#asterisk Strom_C (n=strom@m120e36d0.tmodns.net) |
| 05:43.35 | drmessano | Howdy |
| 05:44.13 | drmessano | There is one trick that works on all Cisco phones |
| 05:44.17 | drmessano | Its a little known "hack" |
| 05:44.36 | RypPn | flash to sip and get on with your life? |
| 05:44.38 | drmessano | Patch the firmware with mod_sellitonebayandgetapolycom |
| 05:46.10 | drmessano | But yeah, the sip flash too |
| 05:46.15 | the_5th_wheel | :-) |
| 05:46.25 | the_5th_wheel | sip wont work for me, bloody sidecart |
| 05:47.00 | drmessano | Dont believe the BS about SIP not being the native format for the phones.. thats bollocks.. The CCM uses SIP now too, so the damn phone should speak it well :) |
| 05:48.28 | RypPn | its missing the nice features that sccp has tho |
| 05:48.53 | the_5th_wheel | ok, next thing. im using a budgettone special, and its Caller ID isnt very clear. is there a way i can display the caller id on my linux box? |
| 05:49.10 | [TK]D-Fender | RypPn, However Cisco lacks all the nice features Polycom has.... like WORKING :p |
| 05:49.26 | [TK]D-Fender | the_5th_wheel, Sure... go code something up :) |
| 05:49.32 | RypPn | [TK]D-Fender: very true, but its hard to beat free when I got my cisco |
| 05:49.52 | [TK]D-Fender | RypPn, You're right... can't argue with free! |
| 05:51.08 | Corydon76-dig | Free isn't free when you find out that Cisco says you can't legally use it without purchasing a license to use their firmware |
| 05:51.29 | RypPn | shhhhhhh |
| 05:51.54 | Corydon76-dig | Polycom gives their firmware away free of charge |
| 05:53.01 | the_5th_wheel | [TK]D-Fender: there is a windblows thing called yac if im not mistaken. |
| 05:53.08 | jameswf-home | http://www.yougetsignal.com/tools/web-sites-on-web-server/ << neat toy |
| 05:53.10 | the_5th_wheel | surely there is something like that for linux? |
| 05:54.09 | Corydon76-dig | Yes, it's that thing you wrote |
| 05:54.37 | Corydon76-dig | but won't share, you evil bastard. ;-) |
| 05:54.43 | drmessano | VoIP on windows rocks.. I am running it... crap.. brb, reboot |
| 05:54.51 | sbingner | lol |
| 05:55.30 | drmessano | Attention in the building, nobody call or or anything, gotta reboot the phone system again.. windows updates |
| 05:56.22 | jblack | Speaking of windows... Why is it 2008 and their filesystems still require defrag? |
| 05:56.44 | drmessano | Because they're using the same one they've been using since 1996? |
| 05:57.35 | jblack | I don't mean technically why. I mean logistically why. |
| 05:57.55 | phix | w00t |
| 05:57.56 | sbingner | because they never updated their file system? |
| 05:57.58 | sbingner | :p |
| 05:58.17 | WilliamK | why so it could turn into a Vista or ME filesystem? |
| 05:58.29 | sbingner | it's not... it's NTFS |
| 05:58.42 | sbingner | oh, right |
| 05:58.47 | WilliamK | it'd be an NTFS-ME filesystem if they borked it |
| 05:58.53 | sbingner | lol |
| 05:59.01 | WilliamK | and I could see the lawsuits climbing rapids quickly |
| 05:59.04 | sbingner | Vista, ME 2.0! |
| 05:59.19 | drmessano | Because they're stuck in NTFS until they get off their duff and actually spend the time building a new FS |
| 05:59.27 | drmessano | Which so far is 13 years behind |
| 05:59.38 | WilliamK | I think I'm going to do what HP says not to do, and that's load XP Pro on a prebuilt sys |
| 05:59.42 | jblack | That's what I mean. Why haven't they bothered to fix this mess yet. |
| 05:59.45 | sbingner | they could always just use ext3 ;) |
| 05:59.47 | phix | sbingner: ok |
| 05:59.54 | drmessano | The replacement for NTFS has been in the works since 1995, before NTFS4 came out |
| 06:00.16 | drmessano | But again, they've just not done it.. they claim its a very large project and can take years |
| 06:00.16 | jblack | Oh, they do have one in the works? What is it? |
| 06:00.18 | sbingner | isn't it that database-fs? |
| 06:00.18 | drmessano | Really? |
| 06:00.20 | WilliamK | I don't really care for the limitations of Windows Media player in vista... unless you buy home premium or ultimate you can't get all the dang codecs |
| 06:00.25 | drmessano | Yes |
| 06:00.26 | drmessano | WinFS |
| 06:00.34 | jblack | Oh, I thought winfs was just a layer on top of ntfs |
| 06:00.36 | drmessano | Which is vaporware.. its been vaporware |
| 06:00.39 | drmessano | Well |
| 06:00.43 | drmessano | Actually, yes |
| 06:00.53 | drmessano | Thats as close as theyve gotten |
| 06:01.04 | drmessano | Not even started on a REAL "new" file system |
| 06:01.12 | drmessano | Just WinFS to extend it |
| 06:01.47 | drmessano | They're waiting for windows executables to run natively on linux so they can dump their core and steal from open source |
| 06:01.53 | jblack | maybe I'm being harsh. It did take them over a decade to get to a consumer-able, fully protected memory model. |
| 06:01.58 | drmessano | then they can use ext3 :) |
| 06:02.17 | jblack | what, now that we're all nearly about to move to ext4? |
| 06:02.23 | drmessano | Of course |
| 06:02.40 | drmessano | Vista is the last of it's kind |
| 06:03.01 | drmessano | They may not dump the NT core in the next release, but I bet money they go to a *nix core |
| 06:03.02 | jblack | lol. That's what they said about 98! And ME, and XP! |
| 06:03.08 | drmessano | No |
| 06:03.16 | drmessano | Not by their choice |
| 06:03.22 | drmessano | By force |
| 06:03.45 | drmessano | Vista came out like it did because the entire windows codebase is an unmanageable mess |
| 06:03.55 | jblack | I don't see them throwing out their kernel. |
| 06:04.02 | WilliamK | I saw the screenshots someone posted of Windows 7(?), looked like vista all over again |
| 06:04.06 | drmessano | They cant take it any further without a major redesign somewhere |
| 06:04.10 | jblack | I think they'd exit the operating system business first. |
| 06:04.30 | drmessano | This isnt speculation, this comes from MS as well |
| 06:04.44 | drmessano | They cant keep kludging on top of the current codebase |
| 06:04.58 | drmessano | So there are plans to tear it apart.. but I dont see them doing it.. |
| 06:05.00 | jblack | Yeah. I've seen that talk sneak out too. |
| 06:05.07 | drmessano | I think they''ll steal from open source first |
| 06:05.14 | drmessano | Add a windows layer to *nix |
| 06:05.20 | drmessano | Something like that |
| 06:05.21 | jblack | But they're not going to throw everything out and emulate on a bsd kernel. |
| 06:05.34 | jblack | At the most, the'll make a brand new kernel, and emulate on top of that. |
| 06:05.45 | sbingner | what do you guys find is a reliable wireless sip phone? |
| 06:05.50 | drmessano | They were using a BSD IP Stack for 15 years.. lol |
| 06:06.33 | sbingner | they're not anymore? |
| 06:06.50 | drmessano | Vista/2008s IP Stack is new |
| 06:07.00 | drmessano | Completely rewritten |
| 06:07.10 | drmessano | They couldnt handle the issues anymore and actually rewrote it |
| 06:07.28 | drmessano | Its shocking it's not caused more issues |
| 06:07.56 | BBHoss | speaking of microsoft, have you guys seen this photosynth thing they have? |
| 06:08.19 | drmessano | Of course.. its possible that not enough ppl are using Vista to really test it.. lol |
| 06:08.37 | sbingner | lol |
| 06:08.52 | sbingner | jblack, well.. MS is good for the economy... |
| 06:09.02 | jblack | I dont think so. |
| 06:09.06 | BBHoss | heh i didnt realize it required vista, but apparently they are working on firefox support |
| 06:09.18 | sbingner | wtf is photosynth |
| 06:09.19 | jblack | I think they're a tax on the naive public |
| 06:09.29 | sbingner | jblack, which is good for the economy lol |
| 06:09.42 | BBHoss | http://labs.live.com/photosynth/ |
| 06:10.01 | drmessano | M$ keeps india employed |
| 06:10.06 | drmessano | Good for THEIR economy |
| 06:10.11 | BBHoss | sbingner, for the sole purpose of them being in the DOW average |
| 06:11.46 | BBHoss | people pay so much attention to one number, its so stupid |
| 06:12.31 | [TK]D-Fender | alrighty... checkout time, later all |
| 06:12.36 | jblack | what would you do? Track 3k stocks manually? |
| 06:12.42 | jblack | [TK]D-Fender: have a good one |
| 06:12.54 | drmessano | Nite TK |
| 06:13.05 | jblack | If you don't like the DOW because it just does 50, there's other indexes too. |
| 06:13.29 | drmessano | jblack: 3k stocks.... 2 words, RSS Reader |
| 06:13.47 | sbingner | or... an acronym and a word! |
| 06:14.36 | jblack | I don't know a better way to track the health of the markets than to observe indexes. |
| 06:15.33 | *** join/#asterisk |Lurch| (n=mike@pool-71-253-56-48.pitbpa.east.verizon.net) |
| 06:16.05 | *** join/#asterisk ice_croft (n=nolan@213.132.86.246) |
| 06:16.14 | drmessano | :( |
| 06:16.17 | drmessano | RSS is a word |
| 06:16.27 | drmessano | So is RLY and KTHXBYE |
| 06:16.29 | sbingner | how is it pronounced? |
| 06:16.32 | sbingner | lol |
| 06:16.39 | jblack | arressess |
| 06:16.39 | sbingner | arsss? |
| 06:16.40 | drmessano | Umm |
| 06:16.45 | drmessano | Yes |
| 06:16.49 | sbingner | lol |
| 06:17.02 | drmessano | Cant call it a "news reader" |
| 06:17.08 | drmessano | Thats NNTP |
| 06:17.13 | jblack | Sync is a word, and it lacks the 5 contemporary vowels. |
| 06:17.16 | drmessano | Not that anyone under 25 would know that |
| 06:17.25 | |Lurch| | hello all. does Asterisk support receiving a modem call? |
| 06:17.29 | sbingner | heh |
| 06:17.33 | drmessano | lol |
| 06:17.34 | jblack | |Lurch|: I believe it does not. |
| 06:17.37 | sbingner | drmessano, unless they like to get their porn from news? |
| 06:17.47 | drmessano | SHHHHHH |
| 06:17.49 | |Lurch| | jblack: do you know of anything that can receive a modem call over SIP? |
| 06:17.50 | drmessano | Dont tell them!!! |
| 06:17.52 | sbingner | lol! |
| 06:17.54 | drmessano | They're RUIN it for us! |
| 06:18.15 | jblack | |Lurch|: Not that I know of. The closest I'm aware of is T.38, and that's for faxing. |
| 06:18.19 | sbingner | |Lurch|, a modem can |
| 06:18.34 | |Lurch| | sbingner: what do you mean? |
| 06:18.45 | sbingner | |Lurch|, terminate the sip call to an ATA or w/e and connect a modem to it |
| 06:18.54 | sbingner | |Lurch|, needs to be ulaw and like NO jitter tho |
| 06:19.04 | sbingner | and you'll probably get like 2400baud |
| 06:19.04 | |Lurch| | sbingner: that seems really ass backwards. |
| 06:19.09 | sbingner | haha |
| 06:19.14 | jblack | Hmm. apparently "cwm" is a word in some dictionaries. |
| 06:19.15 | sbingner | I didn't say it was efficient |
| 06:20.01 | sbingner | it's kinda like TCP SIP over a sat link |
| 06:20.26 | drmessano | Wait |
| 06:20.39 | drmessano | I can run 2400 baud over ulaw |
| 06:20.48 | drmessano | IM PUTTIN MY BBS BACK UP |
| 06:21.04 | sbingner | hahah |
| 06:21.12 | sbingner | you can run a bbs over telnet.... |
| 06:21.12 | Nugget | telnet is eeeeeeevil! |
| 06:21.31 | |Lurch| | drmessano: that's in essence what i need to do |
| 06:21.31 | drmessano | Hmmm |
| 06:21.36 | drmessano | telnet |
| 06:21.39 | sbingner | or you could run excalibur bbs - it's just like html! |
| 06:21.52 | drmessano | HTML BBS...... isnt that a FORUM? |
| 06:22.16 | sbingner | http://en.wikipedia.org/wiki/Excalibur_BBS |
| 06:22.47 | sbingner | some are, apparently, still running |
| 06:22.51 | |Lurch| | drmessano: in essence I need telnet over a SIP connection. |
| 06:23.05 | drmessano | |Lurch| You set your BBS back up over Asterisk, and I will forward so much .gif porn to you, your eyeballs will pop |
| 06:23.24 | sbingner | lol |
| 06:23.30 | WilliamK | so has anyone really figured out which is the better cpu for apps...(just an example) 1.6ghz core dou or a P4 D641 3.2ghz (single core) |
| 06:23.31 | sbingner | no mpg? :( |
| 06:23.37 | drmessano | lol |
| 06:23.55 | drmessano | telnet over sip? |
| 06:24.12 | WilliamK | actually I've dialed in over SIP and * before |
| 06:24.24 | WilliamK | which is what amuses me is that works, but not a freaking fax doing passthrough |
| 06:24.33 | sbingner | I want to do TCP over UDP over TCP |
| 06:25.17 | drmessano | I want to encapsulate my UDP SIP in TCP SSL so I won't have packet loss, ever |
| 06:25.25 | sbingner | then encapsulate the whole thing in an SSH tunnel |
| 06:26.30 | jblack | |Lurch|: What's going on with you, that you'd have sip, but not tcp? |
| 06:26.32 | |Lurch| | WilliamK: can you explain the setup? |
| 06:26.47 | jblack | |Lurch|: if you're just trying to get through obnoxious firewalls, there's things like Tor |
| 06:26.52 | drmessano | |Lurch|, we want to know YOUR setup |
| 06:27.05 | sbingner | jblack, he wants to dial up to a modem.. the rest was a side-track |
| 06:27.25 | |Lurch| | jblack: i am trying to get a dial-up connection (PPP) from a standard old modem, but all my lines are digital PRIs. |
| 06:27.26 | jblack | Got it. He has service somewhere, and he wants to give dialup service to granma. |
| 06:27.30 | sbingner | I think? |
| 06:28.01 | drmessano | This device only has a modem? |
| 06:28.08 | drmessano | Is that it? |
| 06:28.31 | sbingner | so you want something to emulate a modem on your * box, right |
| 06:28.44 | jblack | You'd probably be better off if you just got a pots line and setup mgetty |
| 06:29.13 | |Lurch| | drmessano: no. i want to be able to dial-in with a regular modem. but the incoming lines are PRIs that goes into a cisco box and comes out as SIP + h323. |
| 06:29.14 | jblack | Heh. emulate a modem. That brings soft modems to a whole new level |
| 06:29.24 | |Lurch| | jblack: that's the problem. i don't have a pots line. |
| 06:29.40 | drmessano | You dont have internet service? |
| 06:29.44 | |Lurch| | jblack: at least not an analog pots line. |
| 06:29.48 | jblack | I get what's going on. |
| 06:30.04 | jblack | He has a T1, or some such. He wants to be able to dial in. |
| 06:30.21 | sbingner | |Lurch|, uh goes to a cisco box? ... won't your cisco box handle incoming modem calls? |
| 06:30.25 | drmessano | But no internet? |
| 06:30.29 | jblack | Rather than carving off a channel for an analog modem, he wants * to emulate a modem, so that he can hand it off to ppp. |
| 06:30.36 | drmessano | Hmm |
| 06:30.40 | jblack | drmessano: Sure he has internet. But the person calling in doesn't. |
| 06:31.05 | drmessano | ah |
| 06:31.13 | drmessano | 2400 baud is fast |
| 06:31.26 | |Lurch| | jblack: correct. because carving off an analog modem is done by putting in a 24 channel card into the cisco, which i want to avoid. |
| 06:31.34 | sbingner | |Lurch|, you can probably just assign one of the incoming lines on the PRI to be terminated on your cisco box, it will probably talk to a modem for you |
| 06:31.49 | sbingner | or not |
| 06:31.51 | jblack | |Lurch|: I'd just get the phone company to install an analog line, myself. |
| 06:32.12 | jblack | but if you want to write a app_modem for *, then go for it. :) |
| 06:32.15 | sbingner | |Lurch|, your DSP on your cisco box doesn't support incoming data calls? |
| 06:32.17 | |Lurch| | jblack: in this case, i am the phone company. |
| 06:32.54 | jblack | Then run more lines from your CO. |
| 06:32.59 | |Lurch| | sbingner: i am not aware of such functionality. if you have a pointer to a reference, that would be nice. |
| 06:33.09 | sbingner | |Lurch|, what kinda of cisco box? |
| 06:33.12 | |Lurch| | jblack: they all come in as digital PRIs. |
| 06:33.21 | |Lurch| | sbingner: have a bunch. easiest would be an as5300 |
| 06:33.23 | sbingner | heh |
| 06:33.26 | jblack | I mean run more lines from your central office. |
| 06:33.32 | jblack | Your switching station |
| 06:33.33 | sbingner | AS5300 = what my last job terminated 48-lines of RAS service on |
| 06:33.58 | sbingner | |Lurch|, you can just set them to be a serial line, with ppp encoding |
| 06:34.13 | sbingner | *encapsulation |
| 06:34.14 | |Lurch| | sbingner: what is "them"? |
| 06:34.22 | sbingner | |Lurch|, one of your incoming lines |
| 06:34.44 | sbingner | I may have our old as5300 config somewhere if you want me to look heh |
| 06:34.45 | jblack | see what ATA on the ata gives me |
| 06:35.42 | sbingner | |Lurch|, you may have to have different dsp/modems in it tho... what type of hardware is in your as5300? |
| 06:36.41 | sbingner | |Lurch|, if you want to be able to do 56k you have to terminate it digitally on your end, without converting to analog |
| 06:36.54 | drmessano | jblack.. I got it |
| 06:37.06 | drmessano | You and I... millionaires |
| 06:37.26 | drmessano | IPKALL + some modem app for Asterisk = ISP |
| 06:37.43 | drmessano | TCP/IPKall |
| 06:37.46 | drmessano | YAY |
| 06:37.58 | *** join/#asterisk Strom_C (n=strom@m7e0e36d0.tmodns.net) |
| 06:37.59 | sbingner | lol |
| 06:38.13 | |Lurch| | sbingner: it's an interesting thought. i'll dig around the cisco config. maybe all it takes is a cisco config. |
| 06:38.24 | sbingner | |Lurch|, you may need to have MICA modems in your AS5300 |
| 06:39.11 | |Lurch| | sbingner: yes, i have mica modems. 24 of them. i don't want to waste the cisco slow. |
| 06:39.11 | |Lurch| | slot. |
| 06:39.25 | |Lurch| | sbingner: i need 1 line. |
| 06:39.34 | drmessano | Cisco.. one WIC card away from success |
| 06:42.14 | sbingner | wahahaha solution: http://www.lazylaces.com/56Kmodem/ *grin* |
| 06:42.21 | sbingner | (that's a bit of a joke) |
| 06:42.49 | drmessano | WOW |
| 06:42.50 | drmessano | WOW |
| 06:43.02 | drmessano | I havent heard that sound in YEARS |
| 06:43.33 | drmessano | Crazy |
| 06:43.46 | drmessano | Im redialing.. 19200 my ass |
| 06:44.11 | sbingner | lol |
| 06:44.36 | sbingner | wtf |
| 06:44.41 | sbingner | it isn't connecting at 56k |
| 06:44.46 | sbingner | broken! |
| 06:44.57 | x86 | hahaha |
| 06:45.55 | drmessano | YAY |
| 06:46.00 | drmessano | I connected at 44400 |
| 06:46.32 | drmessano | My short foray into 52800 land was short lived.. but I loved it |
| 06:49.14 | drmessano | I remember having a whole building on WinProxy |
| 06:49.22 | drmessano | 1 modem |
| 06:49.34 | sbingner | lol |
| 06:50.21 | drmessano | We had a T1 in one location |
| 06:50.29 | drmessano | A 10 dialup accounts at the other |
| 06:50.36 | drmessano | Then the dialup accounts went away one day |
| 06:50.44 | drmessano | So we paid for 1 Earthlink account... lol |
| 06:50.47 | drmessano | and would share |
| 06:51.02 | drmessano | Someone needed to dial, other person had to get off the line |
| 06:51.04 | sbingner | really though there should be something to terminate a modem call on * recieved via a T1 |
| 06:51.31 | drmessano | I get WinProxy, slap it on a NT box.. and man.. |
| 06:52.12 | drmessano | I would think so, sbingner |
| 06:53.00 | sbingner | but I haven't found any heh |
| 06:54.15 | jblack | sbingner: There certainly could be. |
| 06:54.55 | jblack | there's already a lot of code out there that's close. |
| 06:55.31 | drmessano | Once you find the code, do we need to find a modem too? :) |
| 06:55.40 | jblack | nah. |
| 06:55.51 | sbingner | I have lots of modems ;) |
| 06:55.55 | jblack | Think about it. soft modems are basically sound cards with the DSP work done in software. |
| 06:56.24 | jblack | take the DSP code, wrapp it into a new app_modem... |
| 06:56.26 | jblack | It's doable |
| 06:56.48 | sbingner | you know what would be funny |
| 06:57.11 | sbingner | modem call routed over sip to * ... then an ATA doing VoIP through that connection |
| 06:57.18 | drmessano | lol |
| 06:57.34 | drmessano | Softphone on the PC dialing in |
| 06:57.37 | sbingner | hehe |
| 06:57.49 | x86 | hehe |
| 06:58.14 | drmessano | Hmm |
| 06:58.19 | jblack | I remember the day I replaced my 300 baud modem with a 1200. |
| 06:58.27 | jblack | It was so... cool |
| 06:58.32 | drmessano | Heres one for ya |
| 06:58.50 | sbingner | slowest I had was either 1200 or 2400 :\ |
| 06:59.16 | jblack | then you never saw an accoustic modem |
| 06:59.17 | drmessano | An on-demand zaptel trunk from box <> box using X100Ps |
| 06:59.17 | sbingner | I was so annoyed when I got 56k modems and found out they only really worked at 28.8 |
| 06:59.43 | jblack | 33.6, and the limitation is 1 way. |
| 06:59.53 | sbingner | but they only REALLY worked at 28.8 |
| 07:00.00 | sbingner | 33.6 once in a while worked |
| 07:00.44 | sbingner | course I suppose I had crappy lines |
| 07:00.48 | drmessano | I used to get much more than that from Bellsouth, before they began limiting it |
| 07:01.05 | sbingner | drmessano, lol illegal! |
| 07:01.17 | drmessano | I got 44400 most days |
| 07:01.24 | sbingner | oh |
| 07:01.26 | drmessano | Then one day it was 28800.. period |
| 07:01.30 | sbingner | thought you meant > 56k |
| 07:01.34 | drmessano | no |
| 07:02.00 | drmessano | Bellsouth started limited it at the switches |
| 07:02.08 | drmessano | I guess to limit crosstalk |
| 07:02.14 | drmessano | limiting* |
| 07:03.30 | *** part/#asterisk |Lurch| (n=mike@pool-71-253-56-48.pitbpa.east.verizon.net) |
| 07:04.43 | drmessano | I had a provider that was the first in the region to bundle phone, cable, internet.. the dialtone was generated from the back of the house.. I used to get 52000 every time.. until they added some noise to the line to keep users from using dialup internet over them |
| 07:04.53 | drmessano | Couldnt get more than 19200 after that.. |
| 07:05.21 | drmessano | Switch to bellsouth.. got 44400 for 6 months, and then they told them they were limiting everyone to 28800 at the switch from then on |
| 07:05.32 | sbingner | gay |
| 07:05.34 | drmessano | Which when I became a DSL users |
| 07:05.35 | drmessano | Which when I became a DSL user |
| 07:05.36 | drmessano | yeah |
| 07:06.01 | sbingner | that noise crap was legal? |
| 07:07.01 | drmessano | From what I remember him explaining, he said to me they put noise on the line to keep users from using dialup internet on their phone lines, which they indicated they didnt support in their TOS |
| 07:07.48 | drmessano | Im sure it was something umm |
| 07:08.02 | drmessano | More technical than "we added noise" |
| 07:08.09 | drmessano | Probably removed some DSP or something |
| 07:08.34 | drmessano | Changed a filter setting or bandpass for the lines |
| 07:09.24 | sbingner | they could have just filtered some audio ranges used by modems |
| 07:10.08 | drmessano | yeah.. and it probably was less technical to them than that.. Some software setting for "allow modem signals" |
| 07:10.18 | drmessano | and they just clicked "no" |
| 07:11.16 | drmessano | I wasnt even paying for internet anyway lol |
| 07:11.22 | drmessano | So the line was the only expense |
| 07:11.51 | jblack | Ast of Jan 2005, fidonet was down to 6800 nodes, of which 1000 were ip based and 1,100 private |
| 07:11.57 | drmessano | Used to hack NetZero, Bluelight, and whatever other free internet service I could find |
| 07:12.20 | drmessano | lol |
| 07:12.43 | drmessano | That was a lot of fun.. They all used Java.. |
| 07:13.24 | drmessano | So all you had to do was replace the exe for the banner, or write a zero length replacement for a .jar |
| 07:13.32 | *** join/#asterisk Damin (n=damin@nucleus.nacs.net) |
| 07:13.36 | drmessano | maybe patch an EXE now and then |
| 07:14.22 | sbingner | heh |
| 07:15.20 | drmessano | THey all checked for idle and would auto disconnect after x minutes |
| 07:15.27 | sbingner | heh |
| 07:15.39 | sbingner | that's why you have a script to do pings or telnets or something ;) |
| 07:15.48 | sbingner | I was conected for > a week straight once |
| 07:15.52 | drmessano | So you would open IRC.. write a script to send zZzZzZzZ in channel when you were away, even 10 mins |
| 07:16.05 | drmessano | every* |
| 07:16.06 | drmessano | lol |
| 07:17.56 | drmessano | Thats why I love Asterisk.... stickin it to the man |
| 07:20.01 | jblack | I spent $900 stickin it to the man. |
| 07:20.07 | sbingner | lol |
| 07:20.17 | sbingner | I've spent at least that much on * |
| 07:20.33 | sbingner | and more on lawyer fees to really stick it to the man ;) |
| 07:20.35 | drmessano | Same here lol |
| 07:20.40 | drmessano | lol |
| 07:21.02 | drmessano | jblack just likes putting his Ex in IVR's from hell |
| 07:21.07 | jblack | bear in mind I came from skype, where I was spending 3 cents a minute and $60/year for a DID. |
| 07:21.09 | drmessano | Thats his motivation |
| 07:21.24 | jblack | Heh. Actually, she helped me run the lines for $100. |
| 07:21.31 | drmessano | lol |
| 07:21.52 | jblack | She has a spending problem, so from time to time I give her money to do stuff. |
| 07:22.20 | drmessano | Dont you wish you kept it that way from the start? |
| 07:22.28 | drmessano | Just pay her when you "need" her... |
| 07:22.47 | jblack | I do. She dragged me down to the brink of financial ruin before she ran off with her new hairy boyfriend. |
| 07:22.59 | drmessano | lol |
| 07:23.07 | drmessano | Not lol in that way |
| 07:23.12 | drmessano | The hairy boyfriend |
| 07:23.29 | sbingner | lol, this is funny |
| 07:23.30 | sbingner | http://www.youtube.com/watch?v=eYzpl6KFPeQ |
| 07:23.36 | drmessano | It's amazing what the next guy looks like |
| 07:24.33 | jblack | Is this gonna give me the exorcist 1/2 way through? |
| 07:24.36 | drmessano | omg |
| 07:24.53 | drmessano | Im LOL'ing |
| 07:24.56 | sbingner | lol, no it's just strange |
| 07:25.28 | x86 | damn this song is awesome |
| 07:25.30 | drmessano | Just needs leekspin as the soundtrack |
| 07:25.31 | sbingner | "tickle the roof of my mouth! oooh yea, that's it baby! |
| 07:25.35 | x86 | AC/DC - Touch Too Much |
| 07:25.43 | x86 | sbingner: wtf? haha |
| 07:25.47 | sbingner | lol |
| 07:29.49 | jblack | man. the only way to kill these koreans in crysis is to shoot them in the face. |
| 07:30.14 | sbingner | ... huh |
| 07:30.32 | drmessano | lol |
| 07:30.52 | jblack | this game is great. |
| 07:32.40 | drmessano | exten => 911,1,Dial(sip/marykate@olsontwins.com) doesn't work :( |
| 07:33.11 | sbingner | perv |
| 07:33.35 | drmessano | lol |
| 07:33.42 | drmessano | No no |
| 07:33.58 | jblack | jfhd |
| 07:34.35 | drmessano | 911 should always dial Mary Kate first, just like Heath Ledgers masseuse did |
| 07:34.40 | jblack | how insulting... after the 5th death in 2 minutes, the game says "you can alter the difficulty" |
| 07:34.51 | sbingner | hahahah |
| 07:35.15 | drmessano | That was me all day friday.. |
| 07:35.36 | drmessano | "Man, my internet is running slow as hell" "hang on, lemme call Mary Kate" |
| 07:36.07 | drmessano | Naturally, I decided she needed to be added to the dialplan |
| 07:41.13 | sbingner | somebody actually called her before 911? |
| 07:41.19 | drmessano | Yes |
| 07:41.26 | drmessano | 9 minutes before |
| 07:41.33 | drmessano | 3 calls |
| 07:41.56 | WilliamK | has anyone had alot of issues with x86-64 bit systems that wasn't a problem on x86-32 bit systems? (linux) |
| 07:41.59 | sbingner | oh that's the guy that died right? |
| 07:42.00 | sbingner | haha |
| 07:42.05 | sbingner | did he die during the happy ending??!? |
| 07:42.07 | drmessano | yeah |
| 07:42.11 | drmessano | lol |
| 07:42.12 | WilliamK | my issues seem to be in the libcrypto's |
| 07:42.16 | *** join/#asterisk oej (n=olle@218.Red-80-34-119.staticIP.rima-tde.net) |
| 07:42.26 | WilliamK | morning oej |
| 07:42.46 | sbingner | WilliamK, just install the -32 libs? |
| 07:43.12 | WilliamK | I've got RH Enterprise 5 |
| 07:43.21 | WilliamK | not sure howto install the 32bit libs |
| 07:43.30 | WilliamK | I thought it was pre-installed with both |
| 07:44.17 | sbingner | that was just a WAG |
| 07:44.38 | drmessano | You have RHEL5? |
| 07:45.42 | WilliamK | ya |
| 07:45.52 | WilliamK | says it installs both at install |
| 07:46.05 | WilliamK | 32-bit libraries are stored in the standard locations: "/lib" and "/usr/lib". |
| 07:46.05 | WilliamK | 64-bit libraries are stored in the following locations: "/lib64" and "/usr/lib64". |
| 07:46.40 | WilliamK | with 64bit compiling being default, but able to pass the flag to gcc to compile in 32bit |
| 07:47.01 | WilliamK | however, my issues are with perl and dbi not functioning properly (md5 hashes) |
| 07:48.22 | sbingner | WilliamK, we'd need the error |
| 07:48.47 | WilliamK | I'm downloading the RH 5.1-32bit right now, but it's a royal pain to reformat, redo the raid, etc.. on the quadcore box it's on |
| 07:49.09 | WilliamK | I wish I knew howto get an error, it's one of my webapps (cgi) |
| 07:50.27 | WilliamK | what bothers me I guess the most about this is it worked on CentOS 5 which is really a Redhat 4.1 core |
| 07:52.46 | sbingner | WilliamK, so you're getting incorrect hashes? |
| 07:53.03 | sbingner | WilliamK, I doubt that's a library error.... |
| 07:53.39 | WilliamK | I believe what's happening is the cgi script isn't functioning well with the &crypt in the perl code |
| 07:54.06 | sbingner | WilliamK, what module are you using for md5? |
| 07:54.16 | WilliamK | all my other files seem to work, and since it won't let me auth, the only thing I can think of is the code isn't x86-64 compat |
| 07:54.21 | sbingner | heh |
| 07:54.25 | WilliamK | I knew you'd say that :) |
| 07:54.26 | sbingner | if it wasn't -64 compat it wouldn't RUN |
| 07:55.06 | WilliamK | unfortunately I have no earthly idea how the script was built to begin with, it used to run on my rh 7.1 box |
| 07:55.15 | WilliamK | it's just straight perl/cgi |
| 07:55.25 | sbingner | WilliamK, I'll fix it for you in exchange for a small donation! hah |
| 07:55.31 | WilliamK | I went into cpan and loaded the modules to mirror the others |
| 07:55.35 | WilliamK | small donation? |
| 07:55.47 | sbingner | I'm bored, let me in and I'll fix it for you |
| 07:55.59 | WilliamK | whatcha want for donation? |
| 07:56.01 | sbingner | since I assume you have no idea where to start fixing it |
| 07:56.01 | WilliamK | :) |
| 07:56.09 | sbingner | I don't really care, that was mostly a joke |
| 07:56.22 | WilliamK | honestly, it's worth $ to me to fix |
| 07:56.40 | sbingner | well, get me access to it... I'll fix it.. and you can send me money if you like |
| 07:57.03 | WilliamK | k, it'll take me a min to set it up for remote |
| 08:00.13 | *** join/#asterisk adjohn (n=adjohn@pc-a80127.wlan.inet.fi) |
| 08:02.24 | *** join/#asterisk steliosk (n=Stelios@athedsl-321178.home.otenet.gr) |
| 08:11.07 | *** join/#asterisk af_ (n=getsmart@88-149-230-64.dynamic.ngi.it) |
| 08:38.00 | the_5th_wheel | has anyone had luck with the asterisk desktop manager? |
| 08:38.31 | *** join/#asterisk PepOSX (n=angeldav@190.72.149.191) |
| 08:48.05 | phix | WilliamK: Do I get access to? :P |
| 08:48.19 | phix | I could supervise :P |
| 09:04.07 | sbingner | lol |
| 09:07.27 | WilliamK | phix, you'd probably be haunting me for fun too :) |
| 09:07.47 | WilliamK | come holloween I could just see it.... painting of ghost screens in ANSI colors |
| 09:07.55 | phix | :P |
| 09:08.31 | WilliamK | luckily for me, I'm having surgery in a week and half and NOT over holloween |
| 09:08.37 | WilliamK | that would really bite over holloween |
| 09:09.35 | WilliamK | having both knees operated on in a week and half (feb 6th) |
| 09:10.18 | WilliamK | everytime I think about SG-1/startrek and my knees are in pain... I think where the heck is those dampners.... |
| 09:15.23 | *** join/#asterisk stew-baby (n=stewart@smckee.plus.com) |
| 09:16.16 | phix | wow |
| 09:16.45 | phix | WilliamK: ummm over holloween? isn't holloween in october? |
| 09:20.04 | WilliamK | close enough |
| 09:20.16 | WilliamK | my mind is sooo far scattered right now you wouldn't believe |
| 09:20.47 | WilliamK | between the meds, the pain, trying to close out projects, getting ready to do a network migration of a datacenter...etc.. |
| 09:21.13 | WilliamK | my mom is also coming here for 2 weeks to watch over me after I have surgery, so that'll kinda help |
| 09:24.54 | WilliamK | oh and both of my parents are going to be at the hospital while I'm in surgery... so this should be interesting |
| 09:25.01 | *** join/#asterisk mvanbaak (n=michiel@vanbaak.xs4all.nl) |
| 09:25.10 | WilliamK | they're telling me so far I'll be in the operating room for 2.5 hours |
| 09:26.05 | WilliamK | and then I've got another 4-12 weeks of physical therapy to do |
| 09:26.25 | WilliamK | anyone wanna trade with me? |
| 09:30.45 | phix | haha |
| 09:31.03 | WilliamK | feeling lucky now phix? |
| 09:31.04 | WilliamK | :) |
| 09:34.38 | phix | heh |
| 09:37.39 | *** join/#asterisk rikardok (n=imodejoy@210.16.52.230) |
| 09:37.49 | rikardok | need help |
| 09:38.06 | tzafrir | ~ask |
| 09:38.06 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 09:38.07 | rikardok | configuring TE410P in zaptel & zapata.conf |
| 09:40.34 | rikardok | how-to configuring TE410P in zaptel & zapata.conf |
| 09:43.03 | *** join/#asterisk los415 (n=los415@cpe-76-171-125-207.socal.res.rr.com) |
| 09:46.40 | tzafrir | One simple thing: genzaptelconf -svdM |
| 09:46.55 | tzafrir | Should work for the basic case |
| 09:47.47 | tzafrir | Though it will put the actual zapata config in /etc/asterisk/zapata-channels.conf |
| 09:48.24 | tzafrir | you'll need to either add '#include zapata-channels.conf' to the end of your /etc/asterisk/zapata.conf, or edit it accordingly |
| 09:52.23 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 09:55.06 | BBHoss | anyone ever played with any Spectralink or KIRK gear in the US? |
| 09:57.52 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 10:02.17 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 10:05.46 | *** join/#asterisk oej (n=olle@218.Red-80-34-119.staticIP.rima-tde.net) |
| 10:11.14 | *** join/#asterisk moa_ (n=moa_@host-69-145-242-178.grf-mt.client.bresnan.net) |
| 10:12.44 | moa_ | Does anyone know what happened to soxmix? |
| 10:13.04 | BBHoss | what happened? its still in apt... |
| 10:14.08 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 10:14.34 | moa_ | I just upgraded our asterisk box and installed sox out of portage and I don't have soxmix anymore. Only sox |
| 10:14.53 | BBHoss | try emerge soxmix, thats what i had to do on debian |
| 10:15.02 | moa_ | no such package |
| 10:15.05 | BBHoss | they must be separate packages |
| 10:15.52 | moa_ | I also downloaded the new source of sox and in the change log it says "Removed soxmix" |
| 10:16.22 | moa_ | So I tried to make a symlink in /usr/bin soxmix -> sox |
| 10:16.48 | moa_ | and now calls aren't getting merged together, the -in is being placed after the -out |
| 10:19.18 | moa_ | So I guess, my next question is. How do I make asterisk call sox -m not soxmix |
| 10:25.23 | *** join/#asterisk Tili (n=tili@cm48.gamma244.maxonline.com.sg) |
| 10:28.51 | tzafrir | moa_, what version of Debian is it? |
| 10:28.59 | moa_ | Gentoo |
| 10:29.36 | tzafrir | BBHoss, what version of Debian was it? |
| 10:29.47 | tzafrir | A separate package for soxmix? |
| 10:30.11 | BBHoss | etch, i think it was debian, could have been my ubuntu machine |
| 10:30.43 | BBHoss | yeah it was debian etch |
| 10:31.17 | tzafrir | A look at the man page shows that six -m mixes files |
| 10:31.29 | BBHoss | yes it does |
| 10:31.44 | moa_ | Right, and I found where soxmix is in res_monitor.c |
| 10:31.54 | moa_ | but there's also MONITOR_EXEC |
| 10:32.06 | moa_ | but I don't know the syntax for it in the dialplan |
| 10:32.13 | BBHoss | heh, that kind of shit shouldnt be hard coded :( |
| 10:32.57 | tzafrir | Etch has soxmix. Lenny doesn't |
| 10:36.16 | *** join/#asterisk guillote_GNU (n=guillote@host155.200-43-37.telecom.net.ar) |
| 10:37.38 | tzafrir | maybe try the following for soxmix: echo '#!/bin/sh' >soxmix; echo 'exec sox -m "$@"' >>soxmix; chmod +x soxmix |
| 10:39.03 | WilliamK | morning tzafrir |
| 10:39.28 | tzafrir | hi |
| 10:39.30 | moa_ | interesting tzafrir, let me try that |
| 10:39.34 | WilliamK | how goes? |
| 10:42.05 | *** join/#asterisk rikardok (n=imodejoy@210.16.52.230) |
| 10:42.29 | *** part/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 10:43.13 | moa_ | tzafrir: awesome, works like a champ! |
| 10:43.32 | *** join/#asterisk Steven_elvisda (n=Steven_E@202.47.107.60) |
| 10:50.08 | rikardok | anyone has a documentation on how to configure(zaptel+zapata) TE410P using E1/isdn pri? need help |
| 10:52.00 | moa_ | also, that might be something to put in a bug report? |
| 10:52.50 | tzafrir | rikardok, have you tried doing what I suggested? |
| 10:53.14 | tzafrir | rikardok, there are generally some pages in voip-info.org |
| 10:53.30 | tzafrir | The page regarding zapata.conf is quite useful |
| 10:53.52 | tzafrir | The one regarding zaptel.conf is probably overly-complicated - just use genzaptelconf |
| 11:03.11 | rikardok | i have tried the configs suggested in voip-info |
| 11:03.11 | rikardok | but no luck |
| 11:03.51 | rikardok | configuring 124 channels.. error says.. ZT_CONFIG**.. span related error something liek that |
| 11:04.26 | rikardok | ill see if i can grab some info about gezaptelconf |
| 11:04.32 | rikardok | thats kinda new |
| 11:06.03 | tzafrir | rikardok, what version of zaptel do you use? |
| 11:06.34 | moa_ | I think that's the same card I've got running |
| 11:13.28 | tzafrir | rikardok, can you answer questions? |
| 11:17.52 | *** join/#asterisk _ys (n=yuri@80.70.236.69) |
| 11:18.18 | BBHoss | rikardok, use pastebin to post your error messages and any relevant information |
| 11:19.08 | phix | hey, I am going to purchase a TDM400p (30b) although I have used one before and one I changed motherboards with this card there was crackeling noises and distorsion over the line. |
| 11:19.14 | rikardok | zaptel 1.2.21 |
| 11:19.49 | phix | how can I ensure that the TDM400p is going to sound good on my mobo? or is it the module that would of caused this? |
| 11:19.53 | tzafrir | zaptel 1.2 does not install genzaptelconf by default. Use xpp/utiols/genzaptelconf from the source directory |
| 11:19.56 | BBHoss | rikardok, i would suggest upgrading everything to 1.4 - latest, it has recently been given the "stable-enough" status |
| 11:20.29 | rikardok | im actually using asterisk 1.2.x for my vicidial |
| 11:20.36 | Mw3 | phix: check hardware compatibility notes on digium's site, and use the latest zaptel modules |
| 11:20.37 | phix | I will be using zaptel from testing (debian) |
| 11:20.45 | phix | Mw3: ok |
| 11:20.50 | tzafrir | zaptel 1.4 has long ago been "stable enough". For a new installation, use latest zaptel, generally. Even if you go for 1.2, use latest zaptel 1.2 |
| 11:20.53 | BBHoss | wtf is with this vicidial shit |
| 11:21.43 | BBHoss | phix, have you checked to see how old the apt version is? sometimes you can get some very old versions |
| 11:21.44 | tzafrir | Something to keep all of the 120 channels he has occupied |
| 11:22.05 | tzafrir | ~vicidial |
| 11:22.33 | tzafrir | The version of Zaptel in Testing is 1.4.7 . 1.4.8 is not in yet, sorry |
| 11:22.48 | Mw3 | phix: personally i dont recommend using precompiled/packaged asterisk or zaptel |
| 11:22.49 | BBHoss | thats good enough |
| 11:22.58 | BBHoss | but i would still go from source |
| 11:23.00 | phix | Mw3: Version: 1:1.4.7.1~dfsg-1 |
| 11:23.42 | phix | Asterisk 1.2.13 |
| 11:24.25 | tzafrir | ~vicidial |
| 11:24.26 | jbot | rumour has it, vicidial is a predictive dialer available from http://astguiclient.sourceforge.net/vicidial.html . |
| 11:24.58 | tzafrir | phix, asterisk 1.4.17 is now in Testing (finally) |
| 11:26.14 | tzafrir | Do upgrade |
| 11:26.21 | phix | tzafrir: Any main diferences? |
| 11:26.37 | tzafrir | Hmm... between asterisk 1.2 and 1.4? |
| 11:27.05 | tzafrir | In the packaging: no more a separate asterisk-bristuffed package (The main package is now bristuffed) |
| 11:30.55 | phix | no I mean asterisk features :) I am happy with the current level of featues, will 1.4 do anything better or include any bug fixes ? |
| 11:31.04 | phix | or it is just implementing new features? |
| 11:31.28 | rikardok | by the way.. error says@ ztcfg -vvvvv: ZT_SPANCONFIG Failed on Span 1; Invalid argument(22) |
| 11:44.55 | *** join/#asterisk worgil (n=USTA@88.254.90.87) |
| 11:45.14 | worgil | hi evryone |
| 11:49.01 | *** join/#asterisk pat_lehem (i=lehem@bzq-79-179-108-143.red.bezeqint.net) |
| 11:49.28 | pat_lehem | Hi all. Is there a way to change the frames per packet ratio? |
| 11:51.23 | pat_lehem | I've seen some questions in various forums, and even a patch from a while ago. But I am not sure this is configurable in sip or iax. Does anyone know? |
| 11:51.41 | pat_lehem | (a while ago = over 2 years) |
| 11:52.54 | pat_lehem | well, I guess not. |
| 11:57.24 | tzafrir | pat_lehem, why would you need that? |
| 11:57.37 | pat_lehem | to support lower bandwidth |
| 11:57.51 | tzafrir | play with mtu? |
| 11:58.12 | tzafrir | rikardok, what is the output of: cat /proc/zaptel/* |
| 11:59.10 | pat_lehem | may help, but the main problem is that the packet header is using my bandwidth instead of the actual data (well, alongside it, anyways) |
| 11:59.41 | pat_lehem | and I won't mind a 40 or 60ms packet, as far as delay is concerned |
| 12:02.32 | pat_lehem | my network delay is about 1-1.5 seconds, so that delay won't matter much |
| 12:09.15 | pat_lehem | Besides that, MTU will not lower the total data sent (except for IP headers, maybe), only send more data each time. My problem is that I pay per kilobyte |
| 12:09.24 | pat_lehem | so I need to reduce the overhead as much as possible |
| 12:13.04 | tzafrir | what codec do you use? |
| 12:13.16 | pat_lehem | either gsm or g729a |
| 12:13.17 | tzafrir | for some codecs you can play with the sample rate |
| 12:13.27 | tzafrir | I think that even with gsm |
| 12:13.37 | pat_lehem | where can I do that? |
| 12:13.41 | tzafrir | hmmm |
| 12:14.03 | pat_lehem | I saw a define in the code, but I am not sure what the effects would be to just change it |
| 12:14.09 | pat_lehem | it appears all around there |
| 12:15.30 | pat_lehem | called GSM_SAMPLES |
| 12:15.52 | *** join/#asterisk d00gster (n=doughant@bas1-toronto12-1177639130.dsl.bell.ca) |
| 12:20.53 | tzafrir | pat_lehem, http://svn.digium.com/svn/asterisk/branches/1.4/doc/rtp-packetization.txt |
| 12:21.24 | pat_lehem | 10x! |
| 12:21.47 | pat_lehem | This means no IAX trunking, but it may be better |
| 12:22.34 | tzafrir | iax already has some optimizatoins with several trunks to the same destination |
| 12:22.41 | tzafrir | (as they all use the same socket) |
| 12:24.04 | pat_lehem | yes, but with 20ms packets, even trunking is wasteful. Also, I will probably not have more than 2-5 calls at the same time, so while the trunk will help, it will not be as significant |
| 12:24.23 | *** join/#asterisk thevoipforum_org (n=hex@87-194-8-43.bethere.co.uk) |
| 12:24.35 | pat_lehem | anyways, great thanks. I will give it a try |
| 12:26.21 | *** part/#asterisk ahilb (n=noone@104.93.187.81.in-addr.arpa) |
| 12:26.22 | *** join/#asterisk ahilb (n=noone@104.93.187.81.in-addr.arpa) |
| 12:39.14 | *** join/#asterisk af_ (n=getsmart@88-149-230-64.dynamic.ngi.it) |
| 12:41.28 | *** part/#asterisk pat_lehem (i=lehem@bzq-79-179-108-143.red.bezeqint.net) |
| 12:56.29 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 12:57.47 | *** join/#asterisk tinloaf (n=tinloaf@d83-189-107-142.cust.tele2.de) |
| 12:58.11 | tinloaf | is there any automated way of havin a testcall from the outside? |
| 12:59.07 | *** join/#asterisk Dovid (n=Dovid@bzq-79-177-102-208.red.bezeqint.net) |
| 13:00.15 | tinloaf | or any other way to test wether my asterisk-server is reachable from the outside? |
| 13:03.39 | *** join/#asterisk esaym (n=user@cpe-72-183-198-134.satx.res.rr.com) |
| 13:04.55 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 13:36.17 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 13:46.08 | *** join/#asterisk fujin (n=aj@unaffiliated/fujin) |
| 13:47.51 | *** join/#asterisk d-k-t (n=dt@60.177.54.141) |
| 13:50.29 | *** join/#asterisk Worgil (n=dGoCbRfX@85.104.111.96) |
| 13:51.11 | Worgil | http://paste.ubuntu-nl.org/53683/ can anyone tell me what is my proglem ? |
| 13:51.33 | Worgil | proglem =problem |
| 13:52.46 | tinloaf | autotools not installed? |
| 13:53.11 | Worgil | what is it mean ? |
| 13:53.50 | tinloaf | well, i think /include/linux/autoconf.h is part of the autotools |
| 13:54.15 | Worgil | tinloaf, how can i solve ? |
| 13:54.44 | tinloaf | aptitude install autoconf |
| 13:55.27 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 13:55.54 | Worgil | tinloaf i did, but looking same error |
| 13:59.37 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:02.56 | *** join/#asterisk GBR_ (n=gbr@201-41-124-55.gnace703.dsl.brasiltelecom.net.br) |
| 14:07.42 | *** join/#asterisk gentoorax (n=gentoora@cpc2-stkn7-0-0-cust476.midd.cable.ntl.com) |
| 14:13.52 | tzafrir | autoconf.h is not related to autotools |
| 14:14.29 | tzafrir | make -C SUBDIRS=/usr/src/asterisk/zaptel-1.4.5.1 clean |
| 14:14.53 | tzafrir | The kernel directory should come after the -C . You don't have a kernel source / headers |
| 14:15.04 | tzafrir | apt-get install linux-headers-`uname -r` |
| 14:15.34 | tzafrir | And you would have recieved a proper error message have you used a more recent version of Zaptel |
| 14:15.57 | tzafrir | oops, Worgil is here no more |
| 14:26.04 | *** join/#asterisk uwe (n=uwe@217.66.234.159) |
| 14:26.51 | uwe | hello, is there a way to debug agi other than doing agi debug on * cli ? im getting invalid or unknown command but im ubable to find out which command is that |
| 14:26.53 | tinloaf | is there any automated way of having a testcall from the outside? |
| 14:28.13 | tzafrir | uwe, in what language do you write? |
| 14:28.38 | tzafrir | tinloaf, Asterisk can't really automate an incoming call :-) |
| 14:29.52 | uwe | tzafrir, its in php ... i tried running the scrip from command line with php -e but it didnt do anything ... its not my agi, its a2billing script |
| 14:31.41 | *** join/#asterisk jmesquita (n=jmesquit@201.20.243.195.user.ajato.com.br) |
| 14:35.53 | esaym | can't I put "meetme_recordingformat=" in the general section of extensions.conf to make it global? It is not working :( Works fine setting it in the extension though |
| 14:38.11 | esaym | oh wait, it is supposed to be in the globals context not general right? |
| 14:40.57 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:41.23 | esaym | ok, fixt |
| 14:44.49 | tinloaf | tzafrir: perhaps i have a problem understanding SIP. If i have an open Asterisk-Box on the net, say on tinloaf.dyndns.org, then it should be reachable via username@tinloaf.dyndns.org, shouldn't it? |
| 14:45.12 | tzafrir | right |
| 14:45.23 | tinloaf | no need to peer with anyone? |
| 14:46.05 | tzafrir | No inherent need |
| 14:46.31 | tinloaf | peering is for "having asterisk listen for incoming calls on username@some.other.provider", right? |
| 14:47.28 | Dovid | anyone know how to factory reset a snom 360 ? |
| 14:49.08 | tinloaf | is there any way to test wether the sip-module of an asterisk box is reachable from the outside? nc is not that useful as asterisk seems not to answer anyway |
| 14:49.47 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 14:56.21 | *** join/#asterisk SteveTotaro (n=root@pool-70-22-26-147.balt.east.verizon.net) |
| 15:15.13 | tinloaf | can there be any reason for the sip-module of asterisk not to answer to "INVITE"-packets? |
| 15:16.13 | SteveTotaro | firewall, NAT |
| 15:19.30 | *** join/#asterisk _ys (n=yuri@236-069.nat.mns.ru) |
| 15:20.02 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 15:21.15 | riddlebox | man SteveTotaro, do you live here now ;) |
| 15:21.27 | riddlebox | crap wait I am always in here too |
| 15:22.00 | SteveTotaro | yeah, i was about to say that |
| 15:22.32 | SteveTotaro | i just configured pidgen with irc so whenever i use this laptop i am signed in |
| 15:22.33 | riddlebox | my boss makes fun of me cause all day long I work on pbx's then go home and play with asterisk |
| 15:23.00 | riddlebox | thats how I have it on my work laptop, but I like xchat better |
| 15:23.07 | SteveTotaro | we share that in common then |
| 15:23.22 | SteveTotaro | ;) |
| 15:23.34 | riddlebox | do you install asterisk systems? |
| 15:24.20 | SteveTotaro | yes, i go after large scale call centers but i am a CCNA and a jack of all trades |
| 15:24.35 | SteveTotaro | 3com, NEC PBX certified too |
| 15:24.55 | riddlebox | wow call centers, if there is anything wrong with your system they will find it |
| 15:25.06 | riddlebox | how do you like the 3com stuff? |
| 15:25.33 | SteveTotaro | the V3000/NBX stuff is really cool and easy |
| 15:25.41 | riddlebox | if you work on NEC, then I guess you work on the ActiveVoice stuff too? |
| 15:25.57 | Qwell | SteveTotaro: installed any of the 3com Asterisk appliances? |
| 15:26.04 | SteveTotaro | certified on activevoice |
| 15:26.09 | riddlebox | 3com is pretty cheap too right? well compared to what we sell, avaya IP Office and Communiation Manager |
| 15:26.16 | SteveTotaro | no i haven't yet |
| 15:26.40 | SteveTotaro | a 3com V3000 is cheap, can compete with an asterisk solution |
| 15:26.42 | riddlebox | SteveTotaro, I am the only one in the company who works on ActiveVoice Repartee LX |
| 15:26.51 | *** join/#asterisk Netyciukas (n=bigman@87-239-84-128.kis.user.lt) |
| 15:27.03 | SteveTotaro | but has only four FXO ports in a 1U chassis |
| 15:27.23 | SteveTotaro | it is when you need to upgrade to a t1 that you pay the real high cost |
| 15:28.37 | riddlebox | The Amanda Company, or The Automatic Answer Company has two asterisk appliances now |
| 15:29.15 | SteveTotaro | my asterisk appliance is the HP DL380 ;) |
| 15:29.38 | SteveTotaro | i like IBM 306s too |
| 15:29.45 | SteveTotaro | for lower end |
| 15:30.19 | riddlebox | I am getting ready to install asterisk at an insurance agency, well after I cable it out |
| 15:30.44 | riddlebox | the guy is real excited, I hope I can deliver for him |
| 15:31.28 | SteveTotaro | i am sure you can deliver |
| 15:32.19 | SteveTotaro | i learned my lesson the hard way to make a firm scope of work with asterisk though |
| 15:34.36 | SteveTotaro | during the sales cycle you can truly say "yes asterisk can do that" |
| 15:35.20 | riddlebox | well I am trying to get the guy used to the idea of parking calls, right now, and trying to explain it will save buttons if he doesnt have to see appearance of the lines |
| 15:35.21 | SteveTotaro | then it is expected, or if you get into all the possible features to get the deal then the customer expects those features |
| 15:36.21 | SteveTotaro | yeah the parking idea is so abstract to key system users |
| 15:37.11 | *** join/#asterisk aksyn (n=aksyn@78-86-127-220.zone2.bethere.co.uk) |
| 15:37.54 | riddlebox | he has a good point he just wants to see if all lines are busy, but I told him I can put appearances on buttons, but I am having a hard time allowing you to press that button and dial out on that line |
| 15:38.08 | riddlebox | it may just be my crappy grandstream test phone |
| 15:38.23 | *** part/#asterisk aksyn (n=aksyn@78-86-127-220.zone2.bethere.co.uk) |
| 15:39.01 | SteveTotaro | i learned about grandstream the hard way too |
| 15:39.25 | SteveTotaro | i pretty much went back to any early granstream installations and upgraded them to polycom for cost |
| 15:40.38 | riddlebox | well this is for my house, which I dont care about, but I am going to have this guy buy me one, he is putting me on the payroll as IT so I figure I will need one to connect to the system remotely so I can test ;) |
| 15:41.03 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 15:41.07 | SteveTotaro | ouch, payroll |
| 15:41.15 | *** join/#asterisk ice_croft (n=nolan@213.132.86.246) |
| 15:41.35 | SteveTotaro | too much taxes, you better get some free goodies |
| 15:44.10 | SteveTotaro | so what version of asterisk are you going with? |
| 15:44.47 | SteveTotaro | 1.6 ;0 |
| 15:45.36 | SteveTotaro | man, i hate it when my online banking is not online |
| 15:45.43 | riddlebox | I am not sure, I havent really looked at 1.6 yet |
| 15:45.51 | riddlebox | I am using the latest 1.4.X here |
| 15:46.58 | SteveTotaro | i am scared to put a 1.4.x box out in the wild |
| 15:47.14 | SteveTotaro | i will try it for a small install |
| 15:47.20 | riddlebox | really, why? |
| 15:47.21 | wwalker | SteveTotaro: why scared? |
| 15:47.27 | SteveTotaro | but certainly not a high volume call center |
| 15:48.29 | SteveTotaro | because i have worked out exactly what works |
| 15:48.46 | dwayne | ActiveVoice !?! |
| 15:49.06 | SteveTotaro | activevoice is rock solid once setup |
| 15:49.24 | dwayne | I visited them once. They had a slide in their engineering department to go from the 3rd to 2nd floor |
| 15:49.40 | riddlebox | I have a Repartee LX that slows to a crawl once a month, I am hoping the next OS version will fix it |
| 15:49.43 | dwayne | this was years ago |
| 15:49.59 | riddlebox | dwayne, did you meet Walter? |
| 15:50.48 | dwayne | possibly, but it was so long ago I don't really remember. It was during my dialogic days |
| 15:50.52 | SteveTotaro | i went to pingtel when they had super cool pastel phones, bean bag chairs the whole dotcom bubble before it popped for them |
| 15:51.12 | SteveTotaro | i was supposed to be learning but knew more about their system than they did |
| 15:51.16 | riddlebox | they informally offered me a job, and I told them they would have to double my pay now and pay for me to move, and Walter said theres no way he could get that to happen |
| 15:53.06 | SteveTotaro | that happened to me once when i had my webstore |
| 15:53.25 | SteveTotaro | i wound up setting up this system |
| 15:53.37 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 15:53.42 | troubled | curious, if "maxmsg" suppose to show up as a voicemail user? |
| 15:54.38 | troubled | s/if/is/ |
| 15:54.38 | *** part/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
| 15:55.10 | riddlebox | haha speaking of that the other day I was playing with my system and took out my hangup() line from my main incoming route, and someone called and I got 106 messages |
| 15:56.11 | troubled | how? |
| 15:56.43 | SteveTotaro | http://translate.google.com/translate?hl=en&sl=fr&u=http://www.osiris.sn/article1636.html&sa=X&oi=translate&resnum=1&ct=result&prev=/search3Fq3Dbeth2Bpayne2Bsengal2Bcomputer2Bfrontiers26hl3Den26client3Dfirefox-a26rls3Dorg.mozilla:en-US:official26hs3Dh9o |
| 15:57.38 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 15:57.50 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 15:58.43 | SteveTotaro | i had to do all the IVRs in French, Wolof, and English |
| 16:01.12 | SteveTotaro | i also had to watch my back because i was taking Africatel |
| 16:01.17 | SteveTotaro | 's business |
| 16:02.03 | SteveTotaro | when Sontatel found out what the E1 was for, they wanted more money, like $3k US a month was not enough |
| 16:05.48 | SteveTotaro | after i pulled off Senegal as a 1 man show with twelve bosses |
| 16:06.24 | SteveTotaro | i was sent to Liberia, Sierra Leone, Ghana, and Guinea as part of USAID WARP |
| 16:06.36 | SteveTotaro | that was an eye opener! |
| 16:10.41 | Corydon76-dig | Monopolies are always nervous when you show them you can beat their pants off |
| 16:11.07 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:11.44 | SteveTotaro | the funny thing is that i was setting up the new monopoly, if you wanted to get a US Visa you had to go through CSC to get an appointment with the Embassy |
| 16:12.08 | SteveTotaro | $15 a pop |
| 16:12.35 | Dovid | lol |
| 16:12.36 | SteveTotaro | or whatever 7,800 CFA was back then |
| 16:12.59 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 16:13.12 | SteveTotaro | i wonder if they have raised the price since the U$D has been tanking |
| 16:13.26 | Dovid | the US dollar is killing me |
| 16:13.49 | SteveTotaro | i hold gold in Zurich |
| 16:14.03 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 16:14.09 | SteveTotaro | well at least that is what they tell me, i haven't seen it yet |
| 16:14.26 | SteveTotaro | but the good news for people in the US is higher exports |
| 16:14.48 | SteveTotaro | and people in other countries actually outsourcing to US! |
| 16:24.53 | [TK]D-Fender | A the birth of a new 2nd-3rd world country.... |
| 16:28.46 | *** join/#asterisk |dennis| (n=dennis@200.32.236.18) |
| 16:29.49 | SteveTotaro | i don't care what the dollar is worth elsewhere as long as prices stay the same here |
| 16:29.57 | *** join/#asterisk br4k3r (n=br4k3r@bas3-ottawa10-1279551404.dsl.bell.ca) |
| 16:31.45 | *** join/#asterisk saint_ (n=saint@c-69-242-118-124.hsd1.nj.comcast.net) |
| 16:32.54 | *** join/#asterisk lemanal (n=lemanal@cpe-066-026-085-055.nc.res.rr.com) |
| 16:33.13 | *** join/#asterisk Assid (n=assid@unaffiliated/assid) |
| 16:33.15 | Assid | heya |
| 16:33.23 | Assid | quick question |
| 16:34.12 | Assid | if i have 2 sip clients behind a lan.. and the asterisk server is on public internet.. and if i call from 1 sip client to another, does it use the lan bandwith? or does it route back and forth |
| 16:35.21 | [TK]D-Fender | Assid, can't do reinvites through NAT. I'll route through your server. |
| 16:36.05 | Assid | damn :( thought i could use that to call the other locations in my house |
| 16:36.08 | [TK]D-Fender | Assid, which is why its a good idea to have at least a minimal * server on-site for multiple phones behind the same NAT, if only to act as a single relay |
| 16:36.25 | [TK]D-Fender | Assid, you CAN, it'll just take out WAN bandwidth |
| 16:36.45 | Assid | [TK]D-Fender: right, i dont have too much of the WAN bandwith.. thats why |
| 16:37.03 | [TK]D-Fender | Assid, go setup another * server there for them |
| 16:37.31 | Assid | too much overheads to keep that running all the time |
| 16:39.13 | [TK]D-Fender | Assid, time for therapy then.... |
| 16:40.37 | tinloaf | damn it. is there absolutely no way of testing wether my server is reachable from the public? some automated test call? |
| 16:41.05 | SteveTotaro | tinloaf, what do you mean? |
| 16:41.41 | tinloaf | SteveTotaro: some web-service e.g. where i enter tinloaf@tinloaf.dyndns.org and which then calls me automatically |
| 16:41.52 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 16:41.58 | tinloaf | my server seems not to be reachable from the outside and i want to test this |
| 16:42.06 | Assid | set up srv records |
| 16:42.07 | real0ne | hello, ihave donwload asterisknow should i run im vmware or burn it into cd and boot from it? |
| 16:42.45 | SteveTotaro | boot from it |
| 16:43.08 | SteveTotaro | tinload, accessible how? |
| 16:44.18 | [TK]D-Fender | Assid, no need for SRV or anything special... |
| 16:45.21 | Assid | i use SRV to use my email as sip address |
| 16:46.38 | tinloaf | SteveTotaro: well, i think if i try to dial into my asterisk at tinloaf.dyndns.org via FWD, it fails at some point |
| 16:47.08 | tinloaf | and i want to analyse where this point is, whether my firewall blocks, asterisk just doesn't answer or my network just is screwed up |
| 16:48.36 | [TK]D-Fender | tinloaf, no, it doesn't work. |
| 16:49.11 | real0ne | SteveTotaro there are aversin for usb stick? |
| 16:49.45 | [TK]D-Fender | real0ne, Just pick whatever distro you want, install it, and slap * on top |
| 16:50.20 | tinloaf | [TK]D-Fender: what message? busy or timed out? |
| 16:50.27 | real0ne | is it os ? |
| 16:51.51 | [TK]D-Fender | tinloaf, "no such host" |
| 16:52.09 | [TK]D-Fender | real0ne, ? |
| 16:52.28 | real0ne | is asterisk like operating system? |
| 16:52.38 | [TK]D-Fender | real0ne, No, * is a piece of SOFTWARE |
| 16:53.42 | real0ne | but if i boot with it he can't destroy all my systems? |
| 16:54.43 | [TK]D-Fender | real0ne, Listen very closely : Install whatever linix disitribution you want in whatever way does not dirupt anything you have installed and just install * on that distro's section of your setup. |
| 16:57.05 | *** join/#asterisk RypPn_OuT (i=TuMbL@rosscom.demon.co.uk) |
| 16:58.15 | tzafrir | real0ne, Asterisk is not a complete operating system. But AsteriskNow is |
| 16:58.37 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 16:59.30 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 16:59.30 | *** mode/#asterisk [+o russellb] by ChanServ |
| 17:01.26 | real0ne | i have donwload asterisknow |
| 17:01.28 | real0ne | plz |
| 17:01.45 | real0ne | what is the diference betwen asterisk and asterisknow |
| 17:02.43 | *** join/#asterisk af_ (n=getsmart@88-149-240-154.dynamic.ngi.it) |
| 17:02.46 | linagee | does anyone know how to get a voip service provider (iax) that has numbers in the 928 area code? |
| 17:03.14 | linagee | i already have an account with voicepulse and sellvoip, but neither has 928 |
| 17:04.41 | russellb | real0ne: asterisk is an application, asterisknow is a full linux distribution that includes asterisk, and a lean web GUI to aid in asterisk configuration |
| 17:04.44 | docelmo | real0ne goto #asterisknow.. Asterisk source is just that. You download and compile.. Asterisk now is a liveCD that installs linux and asterisk all in one |
| 17:05.29 | docelmo | linagee why IAX? |
| 17:05.30 | real0ne | thanks a lot |
| 17:05.44 | docelmo | And I do know of one.. |
| 17:05.55 | real0ne | can any one send me the link Asterisk source |
| 17:06.06 | docelmo | sigh.. newb |
| 17:06.17 | docelmo | real0ne goto http://www.asterisk.org |
| 17:06.28 | real0ne | ok |
| 17:06.57 | linagee | docelmo: because IAX rocks |
| 17:07.40 | linagee | i don't know. i can get one in phoenix, it would just be a bit weird for people. |
| 17:07.50 | linagee | they would all be like, "you're in phoenix?" |
| 17:15.12 | drmessano | wow |
| 17:15.21 | drmessano | If I create a product |
| 17:15.30 | drmessano | Put "Conferencing" in it |
| 17:15.45 | drmessano | I can add $150 to zero right then and there |
| 17:16.07 | drmessano | $5 mic becomes $155 conferencing mic |
| 17:16.20 | errr | lol |
| 17:16.31 | drmessano | $150 phone becomes $300 conferencing phone |
| 17:16.36 | drmessano | Im serious |
| 17:17.26 | drmessano | I can go to Wal Mart and buy a White handsfree skype phone for $20... If I paint it black and sell it as a conferencing solution, $200 |
| 17:18.46 | [TK]D-Fender | linagee, unless you need the bandwidth savings on IAX2 Trunking, IAX2 sucks. |
| 17:19.31 | [TK]D-Fender | drmessano, Nope.. thats be $170 :) Keep with the math! |
| 17:19.31 | drmessano | Anyone know how many concurrent inbound calls you can have with an IPKall number? |
| 17:19.37 | drmessano | HAH! |
| 17:19.40 | linagee | [TK]D-Fender: it rules for firewall type stuff too |
| 17:19.47 | drmessano | BAH |
| 17:19.51 | drmessano | Firewall my ass |
| 17:19.55 | [TK]D-Fender | linagee, irrelevant for most cases |
| 17:19.57 | linagee | [TK]D-Fender: not that i need it as i'm on a dedicated server, but still. :) |
| 17:20.01 | drmessano | My #1 pet peeve |
| 17:20.09 | drmessano | SIP works if you're not lazy |
| 17:20.15 | Assid | very very light grey |
| 17:20.35 | [TK]D-Fender | linagee, And you'll soon realize the mass of audio issues IAX2 has, the risk of one channel crashing ALL calls on your server etc. You do that.... |
| 17:20.38 | linagee | [TK]D-Fender: if i have to run my voip from my home broadband because my colo provider gets stupid or something, i would be stuck with SIP + NAT |
| 17:20.53 | drmessano | So? |
| 17:21.02 | drmessano | Configure it |
| 17:21.05 | drmessano | Make it work |
| 17:21.15 | linagee | drmessano: but iax "just works" :) |
| 17:21.25 | [TK]D-Fender | linagee, half a dozen parm in sip.conf. Big bloody deal. |
| 17:21.27 | drmessano | Yeah, no |
| 17:21.47 | [TK]D-Fender | linagee, Yes, and for one as clearly allergic to "work" as you, must be right up your alley! |
| 17:21.55 | drmessano | HAHAHAHA |
| 17:22.26 | drmessano | Ouch.. but yeah... SIP works if you take the 10 secs to add the few lines |
| 17:23.18 | drmessano | SIP + NAT not working and IAX being the "only answer" is some ugly urban legend that Trixbox users started |
| 17:23.22 | linagee | drmessano: not to mention iax is less well known. :) |
| 17:23.31 | linagee | drmessano: which is a good thing in some cases. :) |
| 17:23.35 | drmessano | I am going to make sure Snopes debunks it |
| 17:23.51 | drmessano | linagee: Still not hearing one good reason to use IAX |
| 17:24.24 | linagee | drmessano: if the govt develops some voip reading thing over the internet, they will probably make it read SIP (and less likely IAX) |
| 17:24.48 | UnixDog | we need a new protocol |
| 17:25.01 | linagee | UnixDog: agreed |
| 17:25.10 | UnixDog | iax/sip/mgcp/skinny/sccp/h323 been there done that |
| 17:25.14 | linagee | UnixDog: make it work with NAT, AND make it have the advantages of SIP |
| 17:25.17 | drmessano | linagee: If you don't know how to make SIP + NAT work now, you WILL end up NEEDING to make it work and your hands will fall asleep from sitting on them |
| 17:25.41 | linagee | drmessano: i suppose i could just DMZ everything to one host inside my network. but that's just dumb as well |
| 17:25.47 | drmessano | Good god |
| 17:25.51 | drmessano | MAKE SIP + NAT WORK |
| 17:25.53 | drmessano | DMZ? |
| 17:25.56 | drmessano | WTFFFFF |
| 17:26.00 | drmessano | 6 lines |
| 17:26.03 | UnixDog | DMZ = network hack |
| 17:26.06 | UnixDog | not |
| 17:26.17 | UnixDog | learn to port forward |
| 17:26.22 | UnixDog | and portmap |
| 17:26.26 | drmessano | DMZ = Lazy admin who cant add lines to sip.conf.. shouldnt admin a toaster |
| 17:26.27 | linagee | UnixDog: sip doesn't use just one port |
| 17:26.42 | drmessano | linagee: maybe Skype is for you |
| 17:27.09 | drmessano | Skype has many nice offerings that you can view at Wal Mart |
| 17:27.21 | drmessano | Perhaps a Hello Kitty Skype phone for the ladies |
| 17:27.28 | drmessano | and a GI Joe one for your CEO |
| 17:27.33 | linagee | drmessano: yes, because things must be hard to configure in order to be drmessano-approved? |
| 17:27.40 | drmessano | hard to configure? |
| 17:27.41 | *** join/#asterisk worgil (n=USTA@88.226.220.76) |
| 17:27.43 | drmessano | 6 lines |
| 17:27.48 | drmessano | 30 seconds of your time |
| 17:27.55 | linagee | drmessano: i'll note you still haven't said them |
| 17:28.01 | linagee | drmessano: it's like you like to flame bait |
| 17:28.02 | drmessano | ~sipnat |
| 17:28.03 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 17:28.08 | drmessano | Go read |
| 17:28.14 | mvanbaak | externip = bla |
| 17:28.17 | mvanbaak | localnet = foo |
| 17:28.22 | mvanbaak | nat = yes |
| 17:28.26 | mvanbaak | canreinvite = no |
| 17:28.28 | mvanbaak | :wq |
| 17:28.30 | worgil | helklo, http://paste.ubuntu-nl.org/53737/ can anyone help me what is this ? |
| 17:28.34 | mvanbaak | asterisk -rx 'restart now' |
| 17:28.34 | worgil | hello, http://paste.ubuntu-nl.org/53737/ can anyone help me what is this ? |
| 17:28.38 | linagee | mvanbaak: externip? hahaha. and with a dynamic ip? |
| 17:28.41 | drmessano | externhost= bah (hostname) |
| 17:28.48 | drmessano | externrefresh = 10 |
| 17:28.53 | mvanbaak | linagee: externhost = bla.domain.com |
| 17:28.54 | drmessano | externrefresh = 10 (if using externhost) |
| 17:29.03 | linagee | mvanbaak: not when i tried to do it. :P |
| 17:29.15 | drmessano | I put a fromdomain too |
| 17:29.16 | linagee | mvanbaak: back in asterisk verion x-1, there was no externhost |
| 17:29.29 | drmessano | ~book |
| 17:29.30 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 17:29.37 | drmessano | Thats TWO URLs |
| 17:29.43 | drmessano | 3 |
| 17:29.45 | drmessano | Sorry |
| 17:29.48 | drmessano | 2 for NAT.. |
| 17:30.31 | linagee | drmessano: traffic shaping, blocking, and inspection will be done by ISPs for SIP. will they know to do the same for IAX? |
| 17:31.03 | drmessano | Paranoid a little? |
| 17:31.08 | linagee | drmessano: always |
| 17:31.15 | drmessano | Skype |
| 17:31.22 | linagee | drmessano: skype uses SIP |
| 17:31.30 | linagee | or some bastardized format of SIP |
| 17:31.31 | mvanbaak | if your ISP is blocking SIP you should switch ISP |
| 17:31.33 | drmessano | ROFL |
| 17:31.38 | drmessano | Its Peer to Peer |
| 17:31.45 | drmessano | They'll never fidn you |
| 17:31.48 | drmessano | They'll never find you |
| 17:32.00 | drmessano | Skype = Sneaky Squirrel Batphone |
| 17:32.06 | drmessano | Fight the man |
| 17:32.17 | mvanbaak | drmessano: skype still needs a central authority to register and put your status on |
| 17:32.22 | drmessano | I know |
| 17:32.34 | mvanbaak | if you block that, skype wont work |
| 17:32.40 | linagee | drmessano: i'd rather just be calling people inside my own encrypted phone network, but occassionally, i must call someone on the POTS network |
| 17:33.01 | drmessano | linagee: If you want secured VoIP, VoIP is not for you |
| 17:33.28 | linagee | drmessano: dry pair of wires is pretty secure |
| 17:33.37 | worgil | hello, http://paste.ubuntu-nl.org/53737/ can anyone help me what is this ? |
| 17:33.43 | drmessano | Sure it is.. Very secure to a butt set |
| 17:33.44 | linagee | drmessano: the only people that know what you are saying is ma bell + govt. |
| 17:34.05 | drmessano | No ones ever cut into a pair of wires before |
| 17:34.05 | linagee | drmessano: with voip, ma bell, govt, and any ISP along your route knows exactly what you're saying. ;) |
| 17:34.17 | *** join/#asterisk gentoorax (n=gentoora@cpc2-stkn7-0-0-cust476.midd.cable.ntl.com) |
| 17:34.53 | linagee | drmessano: it's not like you're handing voip packets to the person with the cutters |
| 17:35.06 | drmessano | I'm pretty sure you're the one spreading the flamebait, so i'll leave my comments with those 2 URLs for SIP+NAT :) |
| 17:35.14 | drmessano | Good luck |
| 17:35.28 | linagee | drmessano: no. you're spreading the flame bait. i'm just spreading the paranoia bat. :P |
| 17:35.34 | linagee | bait too |
| 17:36.10 | *** join/#asterisk gentoorax (n=gentoora@cpc2-stkn7-0-0-cust476.midd.cable.ntl.com) |
| 17:39.30 | *** join/#asterisk wishbone__ (n=wishbone@189.70.89.224) |
| 17:40.29 | linagee | wtf? skype has cheaper international rates than my ITSP |
| 17:41.05 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 17:41.41 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 17:43.54 | *** join/#asterisk AndyGraybeal (n=andy@node229.38.251.72.1dial.com) |
| 17:49.20 | *** join/#asterisk adker (n=chatzill@74-33-202-237.br1.glv.ny.frontiernet.net) |
| 17:51.46 | adker | ~tfot |
| 17:51.46 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 17:53.40 | drmessano | Anyone know if theres a company selling those "Charles Angels" speakerphones? |
| 17:53.57 | drmessano | That single, ugly desktop unit |
| 17:54.44 | drmessano | Radioshack had them for years.. but to quote the Onion "Even CEO Can't Figure Out How RadioShack Still In Business" |
| 17:56.38 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 17:57.10 | *** join/#asterisk LancerZ (n=Anon@CPE-72-135-235-232.wi.res.rr.com) |
| 17:58.28 | SteveTotaro | tunnel your voip over vpn or i will listen |
| 17:59.39 | SteveTotaro | all your a5 are belong to me |
| 18:00.22 | drmessano | ~skype |
| 18:00.23 | jbot | [~skype] Skype is a free VoIP software and service using a closed client and propritary protocol. Only commercial channel drivers exist, all of which are poor implementations of convoluted hacks. In general, forget about using Skype with Asterisk... |
| 18:00.31 | drmessano | BAh |
| 18:00.34 | drmessano | That needs to be updated |
| 18:01.13 | jblack | What's wrong with it? |
| 18:01.22 | *** join/#asterisk braker (n=br4k3r@bas3-ottawa10-1279551313.dsl.bell.ca) |
| 18:01.26 | jblack | Hmm. The service isn't free |
| 18:02.19 | jblack | Unless you wish to consider a SteveTotaro argument that it's free because some aspect of it's free. ;) |
| 18:04.12 | SteveTotaro | http://www.fatwallet.com/forums/messageview.php?catid=18&threadid=758191&highlight_key=y&keyword1=sunrocket |
| 18:04.59 | outtolunc | the software is free <G> |
| 18:06.14 | tinloaf | can asterisk work as outbound SIP-proxy? |
| 18:07.31 | jblack | Not in a way that qualifies as a proxy. |
| 18:07.46 | jblack | It can do some things that get pretty close though. |
| 18:07.49 | SteveTotaro | ~ser |
| 18:07.50 | jbot | it has been said that ser is Sip Express Router - see http://www.iptel.org/ser/, or an old secret method of obtaining a havoc of NAT problems, or at #ser |
| 18:08.49 | tinloaf | jblack: so there is a way of routing all RTP-Traffic through asterisk? can you give me a hint? |
| 18:09.02 | jblack | That's not what I said. ;) |
| 18:09.16 | SteveTotaro | canreinvite=no |
| 18:09.29 | jblack | One thing that you can do, that might be close enough for what you need, is to have calls dial into your * server, and have that dial out. |
| 18:10.13 | jblack | I.E. when I dial 0393514, I'm dialing 0393514 at my * server, and that in turn is dialing 514 on the freeworlddialup network. |
| 18:11.28 | SteveTotaro | many ITSPs are simply acting the way jblack is describing |
| 18:11.49 | jblack | In simpler terms, my phones think they're only calling * (which is true) as an endpoint (which is not true, because it's routing the call out) |
| 18:12.04 | jblack | tinloaf: Is that good enough for your needs? |
| 18:13.12 | drmessano | jblack: The Skype Post needs to be more um... |
| 18:13.16 | drmessano | Final |
| 18:13.18 | drmessano | As in |
| 18:13.45 | drmessano | Skype: If you've been directed here, then Skype is for you! http://www.skype.com and GTFOKTHXBYE :) |
| 18:13.53 | jblack | Skype sux0rs. The day you get rid of it, you'll say "finally" |
| 18:13.58 | jblack | aww. Bad guess. |
| 18:13.58 | tinloaf | hm... |
| 18:14.07 | jblack | drmessano: a dope-sink? |
| 18:14.08 | tinloaf | it seems like it's not enough |
| 18:14.18 | jblack | tinloaf: Ok. Take a look at siproxy then. |
| 18:14.27 | drmessano | yes |
| 18:14.37 | jblack | That's an actual proxy (asterisk is a pbx, not a proxy) |
| 18:14.39 | tinloaf | i have this setup: a xlite behind a NAT-Router, an asterix on the NAT-Router. |
| 18:14.56 | drmessano | I do too, tinloaf |
| 18:14.56 | tinloaf | and it seems like rtp from outside the network is not passing the router to the client... |
| 18:15.02 | jblack | That's what I have here. Phones inside nat, and * that sits on the nat, where it can access both. |
| 18:15.03 | SteveTotaro | dd-wrt? |
| 18:15.10 | drmessano | and I open ports and set the SIP NAT correctly |
| 18:15.12 | *** join/#asterisk ManxPower (n=manxpowe@97-81-241-251.dhcp.lds.al.charter.com) |
| 18:15.13 | drmessano | and it works great |
| 18:15.18 | jblack | Phones dial *, and * dials the calls in turn. |
| 18:15.20 | drmessano | ~sipnat |
| 18:15.20 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 18:15.21 | drmessano | No proxy |
| 18:16.29 | jblack | drmessano: Huh. How did you workout incoming calls from third parties? |
| 18:16.36 | nephfl | anyone write agi scripts in php? |
| 18:16.51 | drmessano | With them on the other side of NAT? |
| 18:16.59 | tinloaf | yeah, the only thing i would need is rtp passing the NAT, but somehow my iptables fail .. reading the guide.. |
| 18:17.00 | jblack | nephfl: Yeah, there's people that write agi scripts in php. There's even an example in the book. |
| 18:17.08 | jblack | drmessano: Yeah. |
| 18:17.22 | nephfl | the online version of the book removed the examples i think |
| 18:17.34 | drmessano | Ports open on this end, SIP settings configged properly |
| 18:17.37 | drmessano | Thats all |
| 18:17.40 | nephfl | i meant does anyone here have experience with it |
| 18:17.44 | drmessano | Works great with them behind NAT |
| 18:17.53 | jblack | are you forwarding the ports to your sip client? |
| 18:18.06 | drmessano | The remote client? |
| 18:18.08 | drmessano | No |
| 18:18.10 | SteveTotaro | sip client should not need port forwarding |
| 18:18.15 | jblack | No, your natted client. |
| 18:18.28 | drmessano | Im lost |
| 18:18.30 | SteveTotaro | nat client should not need port forwarding |
| 18:18.49 | outtolunc | whos on first? |
| 18:18.52 | SteveTotaro | i have a softphone i can use pretty much anywhere and also a wifi phone |
| 18:18.55 | jblack | Ok. can I dial, right now, sip:nonya@c-76-125-26-150.hsd1.ga.comcast.net and get you? |
| 18:19.02 | drmessano | Yes |
| 18:19.20 | drmessano | SIP Client <NAT> <----> Intarweb <----> <NAT + Port forwarting> Asterisk |
| 18:19.24 | drmessano | Thats how I roll |
| 18:19.31 | drmessano | forwarding too |
| 18:19.41 | jblack | Ok, so you are port forwarding? |
| 18:19.48 | SteveTotaro | on the asterisk side |
| 18:19.49 | drmessano | On the asterisk side YES |
| 18:19.50 | SteveTotaro | only |
| 18:19.53 | drmessano | On the client side NO |
| 18:19.56 | ManxPower | Yes, if you have NAT random devices connecting to your softphone won't connect. |
| 18:20.08 | jblack | Steve, is your name drmessano? I'm not asking you questions about his setup. He's right here, so I'm asking _him_. |
| 18:20.30 | SteveTotaro | change the sip port 5060, 5061, 5062 ... |
| 18:20.31 | ManxPower | but very few people let random devices connect directly to their softphone. Normally all calls go thru the server the softphone is registered to. |
| 18:20.32 | drmessano | All my users are behind NATs out in the field |
| 18:20.41 | jblack | drmessano: Ohhh, I see what you're doing. |
| 18:20.48 | SteveTotaro | then take it private |
| 18:20.48 | esaym | in extensions.conf under [global] I have: MEETME_RECORDINGFILE=/var/lib/asterisk/conference/conference-${EPOCH}-${TIMESTAMP}. It puts the file in the correct location but the filename is not using the variables that I defined so what is the correct context? |
| 18:20.50 | jblack | Local nat, dialing into a remote * server. Right? |
| 18:20.53 | SteveTotaro | or do you not know how |
| 18:20.59 | drmessano | yeah |
| 18:21.20 | jblack | Ok, that makes sense |
| 18:21.21 | ManxPower | esaym: I don't think those variables are available in [global] |
| 18:21.29 | SteveTotaro | it's like pulling teeth with jblack |
| 18:21.33 | drmessano | I dont touch the config on any of their routers |
| 18:21.42 | drmessano | My Asterisk box is the bitch, with the ports open |
| 18:21.50 | drmessano | It handles all the work for them |
| 18:22.04 | esaym | hm, yea it works fine in the extension I know |
| 18:23.46 | drmessano | I even have clients on my VPN hitting the * external address because the VPN does ugly things to the UDP stuff |
| 18:24.28 | jblack | drmessano: That seems quite similiar to what I was describing to tinloaf. It's still routing calls through *. I think he said he just wanted a proxy. |
| 18:24.54 | drmessano | I see |
| 18:25.02 | drmessano | Then he wants SER then |
| 18:25.13 | jblack | tinloaf: You catch that? |
| 18:26.04 | ManxPower | Unless you are trying to handle hundreds of calls, SER is not the right solution for most people. |
| 18:26.17 | ManxPower | In fact a SIP proxy is not the right solution for most people. |
| 18:26.52 | jblack | What do you think the right solution is for people in his solution is? |
| 18:27.21 | SteveTotaro | asterisk and dialplan |
| 18:27.23 | jblack | pardon, in his predicament. He's certainly got one, with nat. |
| 18:28.15 | ManxPower | I came in after this guy stated his problem |
| 18:28.27 | ManxPower | He seems to have left his computer, however. |
| 18:28.43 | drmessano | lol |
| 18:28.54 | jblack | Ahhh. For some reason I didn't dig into, doign the typical dialthrough * isn't appropriate for him. Perhaps he's got a linksys or somesuch. |
| 18:29.03 | ManxPower | What is so unique about his problem that he can't use well known, established ways of handling NAT? |
| 18:29.18 | jblack | well, it can't be that, because the n he wouldn't be able to use a proxy either. |
| 18:29.47 | SteveTotaro | openvpn is the ultimate solution |
| 18:29.47 | jblack | I think he's unique. I'm trying to resolve the party line "* is NOT a proxy" when people ask for a proxy. |
| 18:29.54 | ManxPower | A SIP proxy is just a name for a device/program that handles the SIP signaling stuff. |
| 18:30.14 | jblack | Pardon, I think he's not unique. Very typical |
| 18:30.25 | ManxPower | jblack: Asterisk is not a SIP proxy. HOWEVER, from the standpoint of a SIP client, you can call it a proxy. |
| 18:30.50 | jblack | Yeah, I've taken the latter perspective myself, but got yanked on by tk pretty hard for calling it a proxy. |
| 18:30.54 | tinloaf | jblack: well, i think i misunderstood you. if i told asterisk that my clients can not reinvite, asterisk would have to open a listening RTP-port and have the other end connect to it, right? |
| 18:30.57 | ManxPower | A SIP proxy forwards the SIP signaling packets (maybe with modification). Asterisk NEVER EVER forwards a SIP packet. |
| 18:31.18 | jblack | tinloaf: To work with nat, you would have to turn reinvites off for your internal authenticated sip clients. |
| 18:31.28 | jblack | Or turn them off by default. |
| 18:31.35 | tinloaf | right. i have them off. |
| 18:31.41 | file | <PROTECTED> |
| 18:31.48 | ManxPower | tinloaf: no matter what you do, asterisk will have to listen on an RTP port. |
| 18:31.59 | tinloaf | ok. it is not.. damnit |
| 18:32.46 | jblack | tinloaf: Basically, what many people do is (sipclients)<--->(asterisk)<-----(intarweb) |
| 18:32.54 | ManxPower | jblack: Do you understand why Asterisk is not a SIP proxy now? |
| 18:33.04 | jblack | Yeah, I think I can illustrate why |
| 18:33.10 | ManxPower | tinloaf: diagram your setup. |
| 18:33.48 | tinloaf | xlight --> NAT + asterisk --> internet |
| 18:33.56 | jblack | tinloaf: Whereas with a proxy (at least with a transparent one) the clients and the machiens on the intarweb think they're talking directly, when they're not (there's hidden equipment that mangles addresses and ports appropriately) |
| 18:34.03 | tinloaf | my NAT-Router and the asterisk server are running on the same machine |
| 18:34.18 | jblack | tinloaf: That's exactly what I have here. I don't think you need a "proxy" then. |
| 18:34.31 | ManxPower | tinloaf: don't you mean X-Lite <-> NAT <-> Internet <-> Asterisk <-> Internet? |
| 18:34.36 | tinloaf | yeah, ok, you're right, asterisk is not transparent |
| 18:34.39 | ManxPower | And so Asterisk is not behind NAT? |
| 18:34.56 | tinloaf | ManxPower: no, i do not mean that, the machine running asterisk is standing right here at home ;-) |
| 18:35.03 | *** join/#asterisk bkruse (n=bkruse@76.73.154.120) |
| 18:35.03 | *** mode/#asterisk [+o bkruse] by ChanServ |
| 18:35.11 | ManxPower | tinloaf: then why did you not put that in your diagram? |
| 18:35.21 | ManxPower | tinloaf: now diagram it correctly. |
| 18:35.28 | jblack | tinloaf: Here's how you get happy really easy. |
| 18:35.50 | ManxPower | without a good diagram is will be virtually impossible to help you. |
| 18:35.55 | jblack | Log your sip clients into *. Set up outgoing rules so that certain wildcards go to certain networks, and you'll be happy. |
| 18:36.13 | tinloaf | ah, ok, now the machine is listening on a port i assigned RTP to... |
| 18:36.27 | ManxPower | tinloaf: what port is that? |
| 18:36.36 | tinloaf | in my case 20000-20500 |
| 18:36.49 | tinloaf | i've adjusted rtp.conf |
| 18:36.49 | jblack | You're not going to be able to sip to random ips. I though that would be a problem for me, but it ended up onot beign one. |
| 18:36.55 | jblack | tinloaf: linux, right? |
| 18:36.58 | ManxPower | good. You should not have ANY NAT setting on the phone, of course. That will screw up cals. |
| 18:37.02 | tinloaf | right |
| 18:37.09 | jblack | Don't go over 20k |
| 18:37.11 | tinloaf | ah, ok, that could be the problem... |
| 18:37.22 | tinloaf | jblack: it should go up to 65k? |
| 18:37.32 | bkruse | [TK]D-Fender: bkruse.com |
| 18:37.34 | jblack | I mean pick a range that doesn't end over 20k |
| 18:37.43 | ManxPower | tinloaf: you need two RTP ports for each simultaneous call. |
| 18:37.49 | jblack | Say... 10000-10050, or 10000-10100 |
| 18:38.23 | ManxPower | tinloaf: you understand that SIP is all UDP, not TCP, right? |
| 18:38.31 | tinloaf | ManxPower: i do ;-) |
| 18:38.41 | tinloaf | well there are two open ports on the server right now.. |
| 18:39.00 | ManxPower | tinloaf: and those ports are? |
| 18:39.29 | ManxPower | tinloaf: what specific PROBLEM are you experiencing? |
| 18:40.03 | tinloaf | ManxPower: i have no sound transferred |
| 18:40.20 | ManxPower | tinloaf: for all calls or only calls in a specific direction? |
| 18:40.23 | tinloaf | and tcpdump on the server running asterisk doesn't give me any RTP-packets from the outside |
| 18:40.32 | tinloaf | well, i tested two servers... |
| 18:40.41 | ManxPower | tinloaf: you didn't do something stupid like using "bindaddr" did you? |
| 18:40.53 | SteveTotaro | this is why i am so happy to have access to an entire b class subnet |
| 18:40.55 | tinloaf | bindaddr is on 0.0.0.0 |
| 18:41.02 | ManxPower | tinloaf: comment it out |
| 18:41.16 | ManxPower | tinloaf: you really need to be more specific. |
| 18:41.50 | SteveTotaro | bindaddr 0.0.0.0 = all |
| 18:41.53 | ManxPower | you are not getting any "RTP packets" from the outside means "at the client site", "tcpdump running on asterisk" outside asterisk address? outside client address? |
| 18:42.04 | tinloaf | well, obvioulsly... but right now i need something to eat. ;-) |
| 18:42.20 | tinloaf | no, tcpdump is running on my server running the NAT-Firewall and * |
| 18:42.20 | ManxPower | SteveTotaro: yes, it's supposed to be like that, but there have been bugs in Asterisk in the past that made it not work that way. |
| 18:42.34 | ManxPower | tinloaf: turn off NAT and firewall and see if it works. |
| 18:42.36 | SteveTotaro | bugs in asterisk?!? |
| 18:42.37 | tinloaf | and "from the outside" means "from the internet" |
| 18:42.40 | SteveTotaro | must be 1.4 |
| 18:42.54 | ManxPower | I suspect your NAT rules are screwing things up. |
| 18:43.00 | tinloaf | anyway, i'll have to leave for dinner right now... |
| 18:43.02 | tinloaf | thanks a lot, folks |
| 18:45.17 | ManxPower | and this is why it is silly to run Asterisk on a box doing NAT |
| 18:45.25 | drmessano | How does one get iLBC working in * |
| 18:47.30 | KermitTheFragger | hi all; stupid question perhaps but i can't seem to find the answer |
| 18:47.55 | jblack | What's the question? |
| 18:47.58 | KermitTheFragger | where do you put translated things for ivr like the voice for ONE, TWO, THREE ? |
| 18:48.12 | jblack | There's a call for it. please hold |
| 18:48.24 | KermitTheFragger | i dont see them /var/lib/asterisk/sounds |
| 18:48.41 | jblack | they should be there. |
| 18:48.57 | jblack | Let me check my installation |
| 18:49.28 | jblack | look in /usr/share/asterisk/sounds ? |
| 18:49.32 | KermitTheFragger | doh i get it |
| 18:49.44 | KermitTheFragger | i was looking for one and two |
| 18:49.51 | KermitTheFragger | but they are called first and second... |
| 18:50.35 | jblack | Look At SayDigits() and SayPhonetic() |
| 18:50.45 | jblack | (asterisk -r, show applicaiton SayDigits) |
| 18:50.47 | jblack | That what you want? |
| 18:51.28 | KermitTheFragger | yeah thats what im looking for, thx! |
| 18:51.39 | jblack | Welcome |
| 19:04.12 | *** join/#asterisk ChrisMac5 (n=chatzill@87-194-75-247.bethere.co.uk) |
| 19:05.30 | *** join/#asterisk worgil (n=USTA@81.213.236.67) |
| 19:08.55 | *** join/#asterisk iratik (n=iratik@12-226-179-11.client.mchsi.com) |
| 19:09.58 | iratik | Up until now, I've used a client based auto dialer (client requests a number to call from the server, the client triggers the softphone)... the biggest disadvantage is the fact that on 47% of all calls, no answer |
| 19:10.29 | ManxPower | that happens when you do telemarketing |
| 19:10.34 | jblack | predictive dialing? |
| 19:10.36 | iratik | I'm tasked with making a server-based auto dialer, I'm planning to use AMI originate for this.......... has anyone else had any experience making that type of system |
| 19:11.01 | iratik | if i use AMI originate, i can set it up so the agent's extension doesn't ring untill the other party picks up |
| 19:11.33 | iratik | but i'm having trouble getting the exact logic of how it will work figured out |
| 19:12.08 | iratik | so far my logic is, if there is at least 1 agent available for contact, then start calling numbers |
| 19:12.30 | iratik | my biggest obstacle, is if i call numbers asynchronusly (with async).. then what if more than one number picks up |
| 19:12.40 | jblack | are you in the US? |
| 19:12.44 | iratik | at the same time, i can't just call without async, there would be no time savings |
| 19:12.54 | iratik | yes... i'm in the US and familiar with the statutes |
| 19:13.45 | iratik | i could... set it up so the maximum number of dials is equal to the number of available agents |
| 19:14.25 | ManxPower | iratik: there are formulas for all the stuff you are trying to do. |
| 19:14.29 | iratik | anyone have any experience with this? |
| 19:14.38 | ManxPower | established formulas in the predictive dialer industry. |
| 19:14.48 | jblack | I read a couple payers on it in the last couple days. |
| 19:14.57 | iratik | any hints? resources? |
| 19:15.05 | jblack | What I can't get past is how helping you isn't bad for me. |
| 19:15.11 | ManxPower | Much like the ERLANG stuff for figuring out how many PSTN trunks you need for X number of users at X blocking percentage. |
| 19:15.21 | iratik | blocking percentage? |
| 19:15.25 | jblack | THe more successful you get, the more likely anyone here is to get annoyed. |
| 19:15.33 | ManxPower | iratik: you won't find much help with predictive dialing here. |
| 19:15.44 | ManxPower | iratik: blocking == busy or congestion. |
| 19:16.00 | iratik | its not for mass marketing... but it is for business to business telemarketing in the transportation industry |
| 19:16.12 | ManxPower | i.e. for 500 users with %2 blocking, you need Y number of trunks. |
| 19:16.36 | ManxPower | there are similar formulas for the phone spam industry |
| 19:16.52 | ManxPower | And that is ALL the help you will get from me. |
| 19:18.12 | iratik | phone spam? |
| 19:18.37 | nephfl | what variable on an outbound sip call will give me the dialed number? |
| 19:19.46 | ManxPower | nephfl: ${EXTEN}, but that is a system variable and not available via AGI (normally), however it is handed to the AGI when the AGI starts as part of the AGI setup |
| 19:23.47 | *** join/#asterisk lemanal (n=lemanal@cpe-066-026-085-055.nc.res.rr.com) |
| 19:23.54 | nephfl | is EXTEN retrieved with the other variables? |
| 19:27.42 | docelmo | Anyone in here have any experience with the manager API and originate? |
| 19:27.58 | docelmo | How many originates can you send to asterisk before it pukes? |
| 19:28.30 | nephfl | do you know how I retrieve the exten when it is sent as an argument to the php? |
| 19:28.43 | docelmo | yes.. Are you using phpagi? |
| 19:29.12 | nephfl | i am using an agi that is being parsed by /usr/bin/php |
| 19:29.22 | *** join/#asterisk toddejohnson (n=toddejoh@69.220.214.65) |
| 19:29.39 | docelmo | but are you using the class wrapper PHPAGI? or did you code your own stdin and stdout? |
| 19:30.06 | docelmo | nephfl http://phpagi.sf.net |
| 19:30.46 | docelmo | with that class once you initialize the class model you do a call to something like this: $exten = $agi->request['agi_extension']; |
| 19:31.11 | nephfl | im really new to this programming... i used the sample code from voip-info ..i placed an agi file with !/usr/bin/php -q then used $stdin = fopen('php://... then read variables into an array |
| 19:31.56 | docelmo | eh.. |
| 19:32.07 | docelmo | dont use that.. Do you know OOP in PHP with Classes? |
| 19:33.14 | nephfl | nope, I am not much of a programmer, i can only hack code together from examples basically ... intro to c, c, and cobol are the only languages that I have studied (long ago in school) |
| 19:33.38 | *** join/#asterisk rikardok (n=imodejoy@210.16.52.230) |
| 19:33.39 | nephfl | if you have a good reference, I can take a look |
| 19:33.54 | docelmo | check out http://phpagi.sf.net there are examples there |
| 19:34.09 | docelmo | Alternately you can have someone code it for you |
| 19:34.41 | ManxPower | nephfl: NO! PHPAGI is a php library to handle all the dirty details of AGI for you. |
| 19:34.50 | rikardok | any ideas whats worng with codecs? i have g729 installed(working using show g729 @ console); error: chan_sip.c:3775 process_sdp: No compatible codecs! |
| 19:35.02 | ManxPower | EXTEN is handed to your AGI via STDIN, PHPAGI handles all that for you. |
| 19:35.19 | rikardok | no changes in NAT, previously i can make calls without a problem |
| 19:35.22 | ManxPower | rikardok: how many G729 licensees did your purchase? |
| 19:35.25 | rikardok | 29 |
| 19:35.28 | docelmo | sigh |
| 19:35.38 | docelmo | did you tell your peers they are allowed to use g729? |
| 19:35.49 | rikardok | yes |
| 19:35.55 | ManxPower | OK, now we need more lines (pastebin them), usually the few before and after the message |
| 19:36.12 | ManxPower | also, did you confirm your licenses are installed and active. The license is tied to the MAC of the machine. |
| 19:36.18 | docelmo | did you setup your peers (hardware/software not asterisk) to use g729? |
| 19:36.33 | rikardok | when i run /root/registered |
| 19:36.33 | nephfl | how much would it cost for someone to do this all for me? |
| 19:36.35 | rikardok | when i run /root/register |
| 19:36.39 | rikardok | al through out the installation |
| 19:36.44 | rikardok | it doesnt says any wrong |
| 19:36.45 | docelmo | nephfl PM me |
| 19:36.52 | ManxPower | come to think of it, just do a "show translations" (or maybe "core show translations" there should NOT be a - in the G729 field. |
| 19:36.55 | rikardok | when i issue show g729 |
| 19:37.07 | rikardok | it says 0/0 ... 29 channels |
| 19:37.31 | ManxPower | rikardok: good. we don't have to worry about the license not being installed correctly. |
| 19:38.02 | docelmo | ManxPower I would say if possible to get a SIP trace of the call being made.. It will show much more detail than the CLI output |
| 19:38.04 | ManxPower | rikardok: now in sip.conf (I assume you are using SIP) put in [general] disallow=all and allow=g729 |
| 19:38.45 | ManxPower | docelmo: yes, and much, much, much harder to figure out. the CLI output is simple, but shows many issues. If that isn't helpful, then I would recommend a sip debug. |
| 19:38.59 | rikardok | thats my original config |
| 19:39.11 | docelmo | How hard is sip debug peer XXX then copy paste? |
| 19:39.18 | ManxPower | rikardok: put your sip.conf on pastebin. |
| 19:39.40 | ManxPower | docelmo: oh, very easy for the user. It's us that have to wade thru 300 lines of crap looking for that one line that we need to know. |
| 19:39.46 | ManxPower | I don't usually do that for free. |
| 19:40.03 | ManxPower | docelmo: but if you want to talk over helping rikardok, be my guest. |
| 19:40.10 | ManxPower | talk == take |
| 19:40.39 | docelmo | hehe.. Im in the middle of 10 things right now.. I would but no time right now to look at debugs.. maybe in a couple hours |
| 19:43.03 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 19:44.15 | ManxPower | rikardok: I don't see the pastebin of your CLI output or of your sip.conf yet. What is taking so long? |
| 19:44.36 | *** join/#asterisk AndyGraybeal (n=andy@node229.38.251.72.1dial.com) |
| 19:44.38 | rikardok | guys.. anyways.. i made some changes(extensions.conf) in dialing out; the previous parameters is 'o' |
| 19:44.55 | rikardok | replacing it with oTt works |
| 19:44.59 | rikardok | no more one way audio |
| 19:45.03 | rikardok | why is that? |
| 19:45.13 | ManxPower | rikardok: Tt turns off reinvites. |
| 19:45.36 | ManxPower | it does many other things too, many of them you don't want. Just remove them and put canreinvite=no in the relevant sip.conf entries. |
| 19:45.40 | rikardok | and it became one way audio? |
| 19:45.56 | rikardok | that will override in dialling out? |
| 19:46.05 | rikardok | even if i ut 'o' alone |
| 19:46.06 | ManxPower | rikardok: reinvites don't (generally) work in a NAT situation. |
| 19:46.38 | ManxPower | and a broken reinvite is the number one cause of cancer^H^H^H^H^H^Honeway audio |
| 19:46.50 | rikardok | how about this./.. any ideas whats kind of codec taht has reliable dtmf?? |
| 19:47.36 | ManxPower | rikardok: all of them do, if configured correctly. |
| 19:47.54 | rikardok | im using rfc2833, but sometimes it fails |
| 19:48.12 | rikardok | since rfc2833 is well known and recommended |
| 19:48.24 | ManxPower | i.e. inband DTMF only works with the ulaw and alaw codecs, RFC2833 works with all codecs, but some devices don't support ulaw/alaw + RFC2833. DTMF INFO mode is old and should only be used if there is a known issue with the other DTMF methods. |
| 19:48.43 | ManxPower | T and t would screw up some DTMF, # being the most common. |
| 19:53.14 | rikardok | what would you suggest?? current config is exten => _1NXXNXXXXXX,2,Dial(SIP/${EXTEN}@sipaccount,65,tTo) |
| 19:53.21 | *** join/#asterisk CrashSys (n=kumba@62-209.187-72.tampabay.res.rr.com) |
| 19:53.25 | tinloaf | *sigh* if asterisk tries to relay RTP-packets (because the client has canreinvite=no), the connection should be shown in "show channels", right? |
| 19:54.32 | ManxPower | tinloaf: the connection should be shown regardless of canreinvite. "show channels" shows SIP channels, canreinvite only deals with RTP audio. |
| 19:55.21 | tinloaf | so if my client shows "connected" and asterisk shows nothing in "show channels" there is something seriously broken? |
| 19:55.36 | ManxPower | rikardok: I would suggest exten => _1NXXNXXXXXX,2,Dial(SIP/${EXTEN}@sipaccount,65,o) |
| 19:55.43 | *** join/#asterisk grndslm (n=grndslm@24-116-87-97.cpe.cableone.net) |
| 19:55.51 | ManxPower | and add canreinvite=no to the [sipaccount] in sip.conf |
| 19:56.12 | ManxPower | tinloaf: correct, unless your sip client means "registered". |
| 19:56.40 | tinloaf | nope, it thinks it has a connection with another sip client |
| 19:58.18 | ManxPower | If the sip client is a softphone run wireshark or something similar and see where the phone is communicating to. |
| 20:00.53 | rikardok | ManxPower: your suggestion ends up inb one way audio :( |
| 20:01.10 | ManxPower | rikardok: Then you are not putting the canreinvite in the right place |
| 20:02.37 | ManxPower | But since I'm not going to ask for your sip.conf AGAIN, I guess we are at an impass. |
| 20:05.46 | mocker | Woo, all by myself for my asterisk user group meeting. |
| 20:05.48 | mocker | :( |
| 20:06.04 | mocker | Good thing I picked a place where there is beer. |
| 20:08.36 | outtolunc | so start a #asterisk-mocker-aug-meeting room and have it online |
| 20:11.08 | mocker | Heh. |
| 20:13.17 | esaym | anyone got any tips for moh quality? I use only 711 codecs and I have the moh files in 711. They sound alright I think but it seems like the volume fades in and out and there is alittle bit of stuttering every now and then. Same thing with mp3's |
| 20:13.38 | esaym | I don't have any zaptel hardware but I do use the ztdummy module |
| 20:14.57 | esaym | oh and did the timestamp variable change from 1.2 to 1.4? Because the time stamp is not being added to my files: exten => 55,2(record),Set(MEETME_RECORDINGFILE=/var/lib/asterisk/conference/conference-${EPOCH}-${TIMESTAMP}) |
| 20:15.31 | [TK]D-Fender | esaym, upgrade.txt <- |
| 20:15.51 | esaym | [TK]D-Fender: thank you, I will read it again |
| 20:16.42 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:16.42 | *** mode/#asterisk [+o lmadsen] by ChanServ |
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| 20:40.25 | esaym | [TK]D-Fender: Thank you I got a good time stamp now: ${STRFTIME(${EPOCH},GMT+6,%m_%d_%Y-%T)} :-D |
| 20:41.01 | ManxPower | reading the docs frequently does that. |
| 20:41.24 | nephfl | is $EXTEN set by the system or actively in the dialplan? |
| 20:42.10 | nephfl | i just tried a NOOP ${EXTEN} ad it didnt return a value |
| 20:44.33 | Corydon76-dig | Let's try that again. You executed an application designed to do nothing, and you're surprised that it did nothing? |
| 20:44.35 | [TK]D-Fender | nephfl, pastebin <----- |
| 20:46.28 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 20:46.42 | nephfl | I take that back, i just didnt reload the extensions after the change..it looks like it is returning 1 |
| 20:46.47 | nephfl | which really is no better |
| 20:47.56 | nephfl | which variable has the outside phone number that was dialed, is there one? |
| 20:48.02 | ManxPower | nephfl: too bad you are doing this all the hard way |
| 20:48.11 | nephfl | yeah, i know |
| 20:48.22 | nephfl | but all i need to do is pass 2 variables |
| 20:48.25 | ManxPower | nephfl: you keep asking about numbers. EXTEN contains the currently executing extension. |
| 20:48.47 | ManxPower | so if exten => s,1,AGI(your AGI) then EXTEN=s |
| 20:48.54 | nephfl | lol..yeah, i guess it is just 1 then |
| 20:49.14 | ManxPower | if exten => _1NXXNXXXXXX,1,AGI(your AGI) then EXTEN= whatever number matched 1NXXNXXXXXX |
| 20:49.43 | ManxPower | nephfl: your AGI is NOT going to work right anyway if you don't process the STDIN info Asterisk sends your AGI. |
| 20:50.00 | ManxPower | and that info contains the info from EXTEN |
| 20:50.09 | nephfl | i have done that I think, i have it going into an array |
| 20:50.57 | ManxPower | PHPAGI library handles all of that for you. |
| 20:51.04 | nephfl | i am dialing out through a call file, and then connecting to an ivr...i need the ivr to log the extension and the outbound phone number that I dialed |
| 20:51.23 | ManxPower | nephfl: the exten is whatever exten => is for the IVR. |
| 20:51.37 | nephfl | yeah, i realize that now |
| 20:51.39 | ManxPower | Does the call go to a different extension everytime you do a call? |
| 20:52.13 | nephfl | the variable I cant seem to find is the number that was dialed |
| 20:52.43 | ManxPower | nephfl: I doubt that info is available because that is the REMOTE leg. |
| 20:52.59 | ManxPower | I would put a SetVar in your .call file containing a variable and the number dialed. |
| 20:53.21 | nephfl | so there is no system variable that contains that info? |
| 20:53.33 | ManxPower | Sure there is, on the outgoing leg. |
| 20:54.01 | ManxPower | but that is not the leg of your AGI |
| 20:54.09 | ManxPower | Each call has two legs. |
| 20:54.46 | ManxPower | well technically a call has one or more legs |
| 20:55.19 | ManxPower | You can either accept Asterisk's oddities and live a miserable unhappy life, or you can accept Asterisk's oddities and be happy and free. |
| 20:55.21 | ManxPower | You pick. |
| 20:55.54 | nephfl | i see so, i can only see that number from the outbound part of the dialplan, not my ivr |
| 20:56.02 | nephfl | unless i set a global variable |
| 20:57.28 | nephfl | couldnt i set a global variable from the outbound leg of my dialplan and pass it into my agi as an arg or would it already be handled in my stdin? |
| 20:59.13 | *** join/#asterisk many (i=many@213.95.21.30) |
| 20:59.36 | many | does anyone know which modul is required to get a avm b1 running w/ capi? |
| 20:59.41 | nephfl | or would that change it for all calls if i have multiple calls? |
| 21:00.36 | ManxPower | nephfl: I already told you what you need to do. |
| 21:01.03 | nephfl | just set it in the call file? |
| 21:01.21 | ManxPower | Set using a SetVar (or Set, I don't recall which, look at sample.call) |
| 21:02.00 | ManxPower | I pass mailbox and user info via a .call file. |
| 21:09.13 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 21:09.13 | *** mode/#asterisk [+o russellb] by ChanServ |
| 21:14.52 | *** join/#asterisk marioja (n=chatzill@modemcable078.64-70-69.static.videotron.ca) |
| 21:20.24 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 21:21.22 | *** join/#asterisk Docfxit__ (n=Docfxit@ip-64-32-143-214.lax.megapath.net) |
| 21:23.37 | Docfxit__ | Asterisk seems to stop answering calls after 24 hrs. How can I get it to continue answering 24hrs a day every day? |
| 21:24.38 | marioja | docfxit, you question has no context, what version, is this a new install etc... |
| 21:25.00 | marioja | also, what is in your dial plan and what happens when it stops anwering? |
| 21:25.12 | marioja | also what is the hardware configuration |
| 21:26.12 | ManxPower | Usually there is no easy fix for "stops answering calls after 24 hrs". Usually It is not exactly 24 hours, for one thing. |
| 21:26.45 | ManxPower | In my experience most people with "stops answering calls" issue either have bad analog card or a buggy version of Asterisk. |
| 21:26.58 | ManxPower | But without much more information we can't help you at all |
| 21:27.03 | jql | heck, even cron fudges by a few seconds |
| 21:27.50 | marioja | what I suggest is you set verbose to 10, debug to 10, then call in your asterisk box and look at the logs created (there will be lots, better have it to a file) |
| 21:27.53 | Docfxit__ | The version is 1.4.9 |
| 21:27.53 | ManxPower | jql: and it can't be a system script that is breaking it, becuase he did not say a specific time, just "24 hours" (after restarting Asterisk, I assume, but again he did not say) |
| 21:28.10 | jql | 00 4 * * * * rasterisk -rx 'crash when inconvenient' |
| 21:28.16 | Qapf | hey guys, im having to deal with a connection that varies in latency seemingly at will and i have no power to change this, anyone have any recomended settings for jitterbuffer, or anything else, that can help me even out call quality for my iax2 trunk? |
| 21:28.26 | ManxPower | Docfxit__: the first thing to do is upgrade to the latest 1.4. MANY MANY fixes have been put into 1.4 since 1.4.9 |
| 21:29.44 | Docfxit__ | This is a current download. Isn't 1.4.9 later that 1.4? |
| 21:30.35 | ManxPower | current topic is: Asterisk: The Open Source Telephony Application Platform (asterisk.org) -=- Asterisk, -Addons, Libpri 1.6.0-beta1 (2008/01/18), Asterisk 1.4.17 (2008/01/02), *-Addons 1.4.5 (2007/12/1), Zaptel 1.4.8 (2008/01/14), Libpri 1.4.3 (2007/12/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #asterisk-commits to monitor svn/mantis -=- #freepbx (fre |
| 21:30.48 | ManxPower | notice the asterisk version |
| 21:31.08 | ManxPower | I don't know where the hell you download 1.4.9 from. |
| 21:31.33 | ManxPower | But if you downloaded Asterisk, Zaptel, etc from anywhere other than Digium, you downloaded it from the wrong place. |
| 21:32.37 | Docfxit__ | I was told to download it from a backup web site when the Digium web site was down. |
| 21:33.09 | Docfxit__ | I will download it today. |
| 21:33.41 | *** join/#asterisk mvanbaak_ (n=michiel@vanbaak.xs4all.nl) |
| 21:38.10 | Docfxit__ | ManxPower: How can I apply the current download without loosing my configuration? |
| 21:38.59 | ManxPower | Docfxit__: If you do NOT run "make samples" your configs will be preserved. |
| 21:39.13 | ManxPower | but just make a backup of /etc/asterisk if you are worried |
| 21:39.18 | Docfxit__ | Great. Thanks. |
| 21:51.25 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 21:51.57 | BBHoss | anything major changed between 1.4.13 and 1.4.17? |
| 21:52.22 | codejunky | hello, I am using a provider to which I connect my asterisk box with sip and I have some phones connected to the asterisk box with sip. The problem is that I heard some echo when phoning through my provider, any ideas what I can do to get rid of it? |
| 21:52.44 | jql | you hear it, or your callers hear it, or both? |
| 21:52.58 | BBHoss | codejunky, its most likely your on your provider's side |
| 21:53.09 | codejunky | BBHoss: okay. |
| 21:53.28 | BBHoss | i have a few providers like that, and usually its totally random |
| 21:53.32 | codejunky | jql: I do not know whether my callers hear it, but I heard it. |
| 21:53.37 | ManxPower | codejunky: As you know echo removal must be done at the PSTN/VoIP interface |
| 21:53.52 | jql | they LCR you into the blackhole sometimes. shit happens |
| 21:54.07 | BBHoss | codejunky, is it an echo of your voice back, like really delayed echo? |
| 21:54.20 | codejunky | yeah, that is. |
| 21:54.24 | BBHoss | same here |
| 21:54.27 | BBHoss | sucks eh :) |
| 21:54.33 | jql | if you hear it, chances are the echo is so egregious you probably have no chance to cancel it on your end |
| 21:54.34 | codejunky | :-) |
| 21:54.49 | codejunky | hm okay. |
| 21:54.56 | jql | it the echo-causers side's responsibility to cancel, in a civilized world |
| 21:56.03 | ManxPower | jql: the echo causer is Alexander Graham Bell |
| 21:56.07 | jql | the echo-hearer can only "fix" it if the echo delay is really low, like long-tunnel-effect rather than perceptibly delayed echo |
| 21:56.13 | ManxPower | Or at least the 2-wire PSTN network |
| 21:56.16 | lunaphyte | what would you not pay more than for an unlimited did? |
| 21:56.19 | jql | yeah, damn him for letting the wires touch. DAMN! |
| 21:56.28 | jql | wires don't touch, damnit! |
| 21:56.29 | KermitTheFragger | another stupid noob question perhaps, if i declare a variable in an S extension would it be possible that I cant reference it in a I or T extension ? |
| 21:56.47 | [TK]D-Fender | KermitTheFragger, no that makes no sense. |
| 21:57.06 | KermitTheFragger | [TK]D-Fender: why wouldnt it ? :) |
| 21:57.13 | ManxPower | KermitTheFragger: I would have to lookup what the "S" extension is. But for the "s" extension there is no reason you could not unless it was a different call |
| 21:57.23 | [TK]D-Fender | KermitTheFragger, which typically means you're doing somehting wrong. Maybe you should be SHOWING use your code and its CLI output. |
| 21:57.25 | [TK]D-Fender | ~pb |
| 21:57.26 | 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 |
| 21:57.26 | [TK]D-Fender | ^^^^^^^^^^^^^^^^ |
| 21:57.39 | [TK]D-Fender | us* |
| 21:59.50 | KermitTheFragger | [TK]D-Fender: http://pastebin.com/m5e9da094 |
| 22:00.04 | KermitTheFragger | im trying to do a real simple try counter for an test ivr |
| 22:00.07 | *** join/#asterisk soulproxy (n=soulprox@unaffiliated/soulproxy) |
| 22:00.35 | *** join/#asterisk Cedevita (n=ads@86.58.26.60) |
| 22:00.43 | KermitTheFragger | but it seems the TRIES variable cant be referenced from the i extension |
| 22:00.45 | *** part/#asterisk Cedevita (n=ads@86.58.26.60) |
| 22:00.50 | [TK]D-Fender | KermitTheFragger, OUTPUT PLEASE <---- |
| 22:00.59 | *** part/#asterisk soulproxy (n=soulprox@unaffiliated/soulproxy) |
| 22:01.31 | KermitTheFragger | [TK]D-Fender: well the variable just displays -1 in the debug con |
| 22:01.43 | KermitTheFragger | [TK]D-Fender: and not the 4 assigned in the s extension |
| 22:01.53 | [TK]D-Fender | KermitTheFragger, You know I don't want to hear an explanation, I want to see OUTPUT. |
| 22:02.12 | [TK]D-Fender | KermitTheFragger, if I wanted a story I aready know a pathological liar to two... |
| 22:04.31 | BBHoss | KermitTheFragger, pastebin the CLI debug info |
| 22:04.44 | KermitTheFragger | working on it :) |
| 22:05.05 | KermitTheFragger | [TK]D-Fender, BBHoss : http://pastebin.com/m617ebb04 |
| 22:05.15 | KermitTheFragger | so as you can see the variable is displayed as -1 |
| 22:05.41 | lunaphyte | what is the term for calls that are not local and not long distance? |
| 22:06.14 | BBHoss | lunaphyte, international? |
| 22:06.25 | [TK]D-Fender | KermitTheFragger, Executing [i@ivr_sp:2] Set("SIP/siepkes-b54081c0", "_TRIES=-1") in new stack |
| 22:06.27 | lunaphyte | ok, not international either :) |
| 22:06.40 | [TK]D-Fender | KermitTheFragger, exten => i,1,Set(TRIES=$[${TRIES} - 1]) |
| 22:06.53 | [TK]D-Fender | KermitTheFragger, "_TRIES" != "TRIES" |
| 22:07.05 | [TK]D-Fender | KermitTheFragger, Something tells me you haven't applied that dialplan you showed me |
| 22:07.32 | KermitTheFragger | [TK]D-Fender: changed it, doesnt make a difference :( -- Executing [i@ivr_sp:2] Set("SIP/siepkes-b542d9f8", "TRIES=-1") in new stack |
| 22:07.56 | [TK]D-Fender | KermitTheFragger, Show me something COMPLETE that matches. |
| 22:08.07 | [TK]D-Fender | KermitTheFragger, I am trusting you les and less........ |
| 22:08.29 | KermitTheFragger | [TK]D-Fender: why is that ? |
| 22:08.30 | [TK]D-Fender | (best reason to see hard proof) |
| 22:09.06 | [TK]D-Fender | KermitTheFragger, I don't want to see some tiny little clip only to find it doesn't match. If you're going to tell you "it doesn't work", I'd better see the whoel picture and have it match |
| 22:10.16 | [TK]D-Fender | KermitTheFragger, Now I'd want to see "dialplan show ivr_sp" as well.... |
| 22:12.56 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 22:13.58 | KermitTheFragger | [TK]D-Fender: this is it: http://pastebin.com/m18fbb5d4 including my latest attempt... |
| 22:15.13 | [TK]D-Fender | KermitTheFragger, -- Executing [2000@phones:1] Goto("SIP/siepkes-0833a728", "ivr_sp|s|choice_start") in new stack |
| 22:15.26 | [TK]D-Fender | KermitTheFragger, you jump into your IVR and BYPASS setting you silly counter |
| 22:15.40 | KermitTheFragger | thats correct |
| 22:15.44 | KermitTheFragger | thats the idea |
| 22:15.51 | KermitTheFragger | i only once initialize the counter |
| 22:15.57 | KermitTheFragger | and then i substract stuff from it |
| 22:16.27 | KermitTheFragger | and once it hits zero, i want people kicked of the line |
| 22:16.33 | [TK]D-Fender | KermitTheFragger, well I have no prrof that that var should hold anything other that 0 to begin with and the first thing you do is SUBTRACT from it, so -1 sounds right. |
| 22:16.35 | sbingner | and where was it initialized this once? |
| 22:17.15 | [TK]D-Fender | KermitTheFragger, And it does seem to be hanging up |
| 22:17.22 | KermitTheFragger | [TK]D-Fender: how so ? by doing exten => s,n,Set(TRIES=4) I set it to 4 right ? |
| 22:17.23 | *** join/#asterisk UnixDog (n=unixdog@adsl-69-234-188-27.dsl.irvnca.pacbell.net) |
| 22:17.27 | KermitTheFragger | yes but the first time |
| 22:17.34 | [TK]D-Fender | KermitTheFragger, Where the hell is that CALLED in there? |
| 22:17.51 | [TK]D-Fender | KermitTheFragger, You don't even start your IVR at the BEGINNING |
| 22:18.17 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 22:18.18 | [TK]D-Fender | KermitTheFragger, -- Executing [2000@phones:1] Goto("SIP/siepkes-0833a728", "ivr_sp|s|choice_start") in new stack-- <-- your INITIAL goto is BAD |
| 22:18.36 | KermitTheFragger | i dont understand, why ? |
| 22:18.44 | [TK]D-Fender | KermitTheFragger, You aren't starting your IVR at s,1 like you should based on its design. |
| 22:19.21 | KermitTheFragger | [TK]D-Fender: aaah crap |
| 22:19.24 | KermitTheFragger | i see it now |
| 22:19.28 | BBHoss | heh |
| 22:19.30 | KermitTheFragger | a man what stupid |
| 22:19.30 | BBHoss | love it |
| 22:19.30 | [TK]D-Fender | KermitTheFragger, SMRT |
| 22:19.39 | KermitTheFragger | im off shooting myself... |
| 22:19.55 | KermitTheFragger | [TK]D-Fender: thx, really dumb mistake on my part :( |
| 22:24.35 | lunaphyte | how can i tell asterisk to precede all 10 digit calls with a 1 ? |
| 22:25.27 | [TK]D-Fender | lunaphyte, just shove a 1 in front |
| 22:26.53 | lunaphyte | [TK]D-Fender: how do i do that? |
| 22:27.23 | [TK]D-Fender | lunaphyte, go into your dial, and just shove it in front |
| 22:27.46 | BBHoss | lunaphyte, put a 1 before {EXTEN} in your dial command |
| 22:28.01 | [TK]D-Fender | BBHoss, I wasn't going to assume anything as to what vars were being used |
| 22:28.42 | [TK]D-Fender | BBHoss, as that is never the case for my dialing.... |
| 22:29.38 | BBHoss | well actually if you're using EXTEN, you need to put it before ${EXTEN} (I forgot the #) |
| 22:30.28 | *** part/#asterisk many (i=many@213.95.21.30) |
| 22:31.33 | braker | ok |
| 22:31.35 | braker | so i got |
| 22:31.41 | braker | an extension, 100 |
| 22:31.50 | braker | and a SIP connection to a voip provider |
| 22:31.51 | BBHoss | ~ask |
| 22:31.51 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 22:31.53 | braker | dialing in and out |
| 22:31.55 | sbingner | can you hit <enter> a little less? |
| 22:31.58 | braker | or |
| 22:32.01 | braker | oh, ok |
| 22:32.02 | *** join/#asterisk cesar_CR (n=cesar@celord.ice.co.cr) |
| 22:32.15 | BBHoss | thanks |
| 22:32.52 | braker | so i have my extensino up and dialing out, to a termination provider, and incoming calls can come in. i want to set up a basic vmail box, for extension 100 that when the call is not answer, it will go voicemail and that stuff |
| 22:34.02 | BBHoss | ~book |
| 22:34.03 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 22:34.12 | lunaphyte | ah, i understand - thanks [TK]D-Fender, BBHoss. that's works great (yes, i am using EXTEN) :) nothing fancy here (yet). :) |
| 22:34.23 | lunaphyte | err, that works great, even. |
| 22:34.31 | BBHoss | lunaphyte, good luck |
| 22:34.31 | sbingner | braker, look at macro-stdexten in extenstions.conf.sample |
| 22:35.00 | lunaphyte | what does the underscore signify here? http://rafb.net/p/x1TJMv23.html |
| 22:35.27 | BBHoss | sbingner, read starting at the bottom of page 153 of the book |
| 22:35.29 | [TK]D-Fender | lunaphyte, that what follows is a PATTERN |
| 22:35.33 | braker | k |
| 22:35.39 | sbingner | BBHoss, huh? |
| 22:35.45 | lunaphyte | ah - as opposed to a literal? |
| 22:35.53 | [TK]D-Fender | lunaphyte, otherwise its looking for LITERAL "X"'s and "N"'s in what you are dialing |
| 22:36.02 | BBHoss | sbingner, he just wants to setup voicemail right? |
| 22:36.13 | braker | yes just a vmail box |
| 22:36.14 | sbingner | BBHoss, yea but why are you telling me instead of him? lol |
| 22:36.18 | lunaphyte | thanks. |
| 22:36.21 | BBHoss | shit my bad :) |
| 22:36.22 | [TK]D-Fender | braker, voicemail.conf |
| 22:36.34 | [TK]D-Fender | braker, "core show application voicemail" |
| 22:37.17 | KermitTheFragger | [TK]D-Fender: im off hitting the sack, thx again for the help! |
| 22:37.49 | [TK]D-Fender | KermitTheFragger, You're welcome |
| 22:37.51 | *** join/#asterisk remmo (n=junk@203.32.47.250) |
| 22:39.03 | *** join/#asterisk angryuser (n=Miranda@df01t2-62-34-199-11.d4.club-internet.fr) |
| 22:44.56 | braker | got it |
| 22:44.58 | braker | thank you :) |
| 22:45.02 | braker | voice mail is now workin |
| 22:45.08 | braker | that book is pretty neat |
| 22:54.38 | lunaphyte | there is a longer delay when i dial NXXNXXXXXX or NXXXXXX than when i dial 1NXXNXXXXXX. how can i make it shorter? |
| 22:56.39 | sbingner | lunaphyte, make sure there is no other exten that could match it |
| 22:56.51 | sbingner | which will be hard with NXXXXXX |
| 22:57.19 | lunaphyte | yeah, i see. it's waiting for more digits. |
| 22:57.28 | lunaphyte | i need to be more specific with it then, i guess. |
| 22:57.57 | angryuser | good day i am searching info about operator used in * like + - != = ect |
| 22:58.06 | angryuser | *operators |
| 22:58.38 | [TK]D-Fender | lunaphyte, thats the fault of your phone. |
| 22:59.10 | [TK]D-Fender | lunaphyte, its dialplan is not optimized to know when to send a call being dialed to * as fast as can be matched by * |
| 22:59.31 | [TK]D-Fender | angryuser, ... |
| 22:59.33 | [TK]D-Fender | ~book |
| 22:59.33 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 22:59.35 | [TK]D-Fender | ^^^^^^^^^^^^ |
| 23:00.13 | lunaphyte | [TK]D-Fender: i think it's a little of both, maybe? i have made some improvements to my phone's (pap2t) dialplan, but there's room for more. |
| 23:00.44 | lunaphyte | here's is the phone's dialplan: |
| 23:00.51 | lunaphyte | (*xx|11xxS0|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.) |
| 23:01.00 | angryuser | wanted a direct wiki page, coz no found by myself, but anyway thx |
| 23:01.33 | [TK]D-Fender | lunaphyte, xxxxxxxxxxxx. interferes with [2-9]xxxxxx |
| 23:01.41 | [TK]D-Fender | lunaphyte, be careful of overlap <- |
| 23:02.18 | lunaphyte | i added 11xxS0 , which works great for extensions - but when i tried changing [2-9]xxxxxx to [2-9]xxxxxxS0, i get a busy. |
| 23:02.46 | lunaphyte | it does? that was part of the default dialplan. |
| 23:03.44 | lunaphyte | oh - it means that if i dial 212.757.1207, the phone will wait, because i might be matching xxxxxxxxxxxx. - right? |
| 23:05.04 | *** join/#asterisk jmls (n=jmls@81.138.42.77) |
| 23:05.06 | jmls | is there any way of seeing what realtime engines are available from the cli ? |
| 23:06.30 | [TK]D-Fender | lunaphyte, really?!?! :) |
| 23:06.55 | lunaphyte | re: which comment? |
| 23:07.07 | [TK]D-Fender | angryuser, "asterisk expressions" <- on the WIKI |
| 23:07.20 | [TK]D-Fender | <lunaphyte> oh - it means that if i dial 212.757.1207, the phone will wait, because i might be matching xxxxxxxxxxxx. - right? <- yes, duh. |
| 23:07.22 | angryuser | perfect |
| 23:07.31 | lunaphyte | ah. sarcasm ;) |
| 23:07.36 | [TK]D-Fender | lunaphyte, of course it could math that "something bigger" |
| 23:07.44 | [TK]D-Fender | match* |
| 23:07.46 | lunaphyte | right. |
| 23:07.56 | lunaphyte | i'm new to all this. easy on the caning :) |
| 23:09.08 | lunaphyte | but, i get what you're saying - i need to make sure the phone is passing things along to asterisk as fast as it should first, then worry about asterisk passing the call off to the next component. |
| 23:09.50 | [TK]D-Fender | lunaphyte, you PAP2 controls how fast it is to interpret dialing. |
| 23:10.11 | sbingner | lunaphyte, your phone probably lets you hit # if it hasn't reached a known end-of-dialstring |
| 23:10.34 | *** join/#asterisk errr (n=errr@fedora/errr) |
| 23:10.34 | [TK]D-Fender | sbingner, yes, that is typically a defaul for most ATA's |
| 23:10.43 | lunaphyte | sbingner: yeah, i can do that - i'm just working towards making that more transparent. |
| 23:10.58 | sbingner | lunaphyte, you're not going to be able to with 7-digit numbers ;) |
| 23:11.01 | [TK]D-Fender | lunaphyte, then just accept the delay and let * do what it has to |
| 23:11.12 | [TK]D-Fender | sbingner, Sure he can. |
| 23:11.17 | lunaphyte | sbingner: is that one of these universal truths? |
| 23:11.27 | lunaphyte | oh. maybe not. |
| 23:11.31 | sbingner | only if he won't accept 9-digit numbers |
| 23:11.50 | [TK]D-Fender | sbingner, nope.. |
| 23:12.05 | sbingner | NXXNXXXXXXXX <-- how can NXXXXXX not be a possible match for that? |
| 23:12.20 | [TK]D-Fender | sbingner, actually... SOME wil pass through faster if you're lucky :) |
| 23:12.39 | [TK]D-Fender | sbingner, but yeah, not a 100% guarantee. |
| 23:13.10 | [TK]D-Fender | I'm a big fan of (#.T|*.T|x.T) as a dialplan :) |
| 23:13.20 | [TK]D-Fender | your device should have a dialplan... thats what * is for. |
| 23:13.40 | lunaphyte | in north america - what would drawbacks of not accepting 9 digit numbers be? |
| 23:14.06 | [TK]D-Fender | lunaphyte, just strike that comment and move on, ok? |
| 23:14.13 | lunaphyte | oh, ok. |
| 23:14.26 | [TK]D-Fender | lunaphyte, and where DO you have a 9 digit number anyways? |
| 23:14.29 | tzanger | have any of you taken a look at the telepathy project? It looks like an interesting way to bring together any kind of realtime comms |
| 23:14.41 | lunaphyte | i don't think i do. |
| 23:14.48 | tzanger | looking at it from a "hmm I wonder if asterisk should have a channel driver for it" pov |
| 23:14.58 | angryuser | <PROTECTED> |
| 23:15.15 | sbingner | angler, it is true if arg1 is not the same as arg2 |
| 23:15.31 | sbingner | so yes |
| 23:15.40 | angryuser | ok thx |
| 23:16.02 | [TK]D-Fender | angryuser, just like "equals"... only NOT! |
| 23:16.07 | sbingner | hahaha |
| 23:20.33 | *** join/#asterisk mvanbaak (n=michiel@vanbaak.xs4all.nl) |
| 23:21.14 | iratik | Here is what i don't understand about how predictive dialers using asterisk and AMI originate work.... a predictive dialer dials numbers, and if the number picks up - does something like find an agent to take the call... but to use AMI originate, you have to provide a source and a destination.... so you have to know what extension the call is going to connect with ... before the call? |
| 23:22.07 | jmls | iratik: use a local channel |
| 23:22.14 | jmls | in the local channel, find the agent you want |
| 23:22.16 | *** join/#asterisk tompaw (n=tompaw@gw.krakow.tompaw.net) |
| 23:22.17 | tompaw | hello |
| 23:22.21 | jmls | and dial his/her number |
| 23:22.50 | iratik | local channel? |
| 23:23.23 | sbingner | wouldn't you just provide a source of a local hunt-group or something? |
| 23:23.32 | jmls | iratik: http://www.voip-info.org/wiki/index.php?page=Asterisk+local+channels |
| 23:24.02 | tompaw | I have just installed asterisk 1.2.13 on FreeBSD 6.2. Which GUI/Webinterface would you recommend to set it up quickly and without too much pain? |
| 23:24.16 | jmls | instead of zap/xxx and SIP/yyy (for example) |
| 23:24.17 | sbingner | aah |
| 23:24.29 | sbingner | tompaw, vi |
| 23:24.33 | jmls | use zap/xxx and local/yyy@abc |
| 23:25.06 | tompaw | sbingner: and for the accounts management? |
| 23:25.08 | jmls | in context abc, extension yyy, do some "magic" (db query / lookup / list) to find the agent you want |
| 23:25.14 | jmls | and dial that agent |
| 23:25.28 | iratik | Is there a way to get feedback from the asterisk management interface when an originate command is issued ... so that when the person at Channel: picks up ... the AMI spits out something? |
| 23:25.39 | iratik | ahhhhhhhhhhh |
| 23:25.45 | sbingner | I don't know a good account/billing management software... somebody else may |
| 23:25.58 | *** join/#asterisk CCFL_Man2 (i=7cd4a436@pool-71-241-116-7.scr.east.verizon.net) |
| 23:26.26 | jmls | iratik: we use jabber in the local channel to check the presence of an agent |
| 23:26.37 | tompaw | ok, thx. |
| 23:26.53 | jmls | iratik: we also use func_curl a lot as well |
| 23:27.18 | iratik | yyy can be anything (like INB)... and all i have to do is write my own call handler script to send the call to the appropriate agent |
| 23:27.24 | jmls | yes |
| 23:27.27 | lunaphyte | hmm - so i can configure asterisk to match long distance NXXNXXXXXX and dial without requiring a 1, but then i couldn't immediately match NXXXXXX. or, i could always require a 1 for long ditance, and be able to immediately match NXXXXXX. have i got that right? |
| 23:27.31 | iratik | Local/INB@AUTODIAL |
| 23:27.39 | jmls | somethng like that |
| 23:27.40 | iratik | [AUTODIAL]; exten=> .... .... |
| 23:27.43 | iratik | i see |
| 23:27.52 | jmls | we use Local/DiallerConnected@GetAgent |
| 23:28.12 | iratik | so in extensions.conf, you have [GetAgent] and the rules for handling connected calls |
| 23:28.15 | sbingner | lunaphyte, yes |
| 23:28.17 | jmls | yes |
| 23:28.32 | iratik | thats sounds much easier than what I was thinking |
| 23:28.42 | jmls | it is |
| 23:28.59 | jmls | it took me a while for the penny to drop |
| 23:29.09 | jmls | but iit is so simple now |
| 23:29.55 | iratik | can I see your extensions.conf for Local/DialerConnected@GetAgent ? |
| 23:30.12 | [TK]D-Fender | lunaphyte, adding a 1 to a 10-digit local number would eb bad |
| 23:30.50 | sbingner | there are some states that have local numbers in different area codes, aren't there? |
| 23:31.07 | iratik | yes? |
| 23:31.40 | sbingner | lunaphyte, you could have a database of local numbers and REMOVE the 1 for those also... might be simpler for you |
| 23:32.50 | jmls | iratik: wait one |
| 23:34.11 | jmls | iratik: a snippet is here |
| 23:34.13 | jmls | http://www.pastebin.ca/875493 |
| 23:34.40 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 23:34.47 | jmls | in the GetAgent macro we call our application via CURL to get the next agent |
| 23:35.03 | *** join/#asterisk gentoorax (n=gentoora@cpc2-stkn7-0-0-cust476.midd.cable.ntl.com) |
| 23:35.03 | iratik | i'd bow down to you... but you wouldn't be able to see it |
| 23:35.22 | lunaphyte | sure- nyc. |
| 23:35.29 | *** join/#asterisk Daejeo (n=chatzill@211.211.234.81) |
| 23:35.30 | lunaphyte | i'm sure others too. |
| 23:35.41 | sbingner | cali I expect ;) |
| 23:35.42 | jmls | bow down to the guys that write the code, and people like [TK]D-Fender. Not me |
| 23:36.23 | jblack | He does rock. |
| 23:36.27 | iratik | Any other auto-dialer implementations floating around for asterisk besides vicidial, shadydial and gnudialer? |
| 23:36.39 | lunaphyte | when you call a number that doesn't require an area code, it's possible that it might not be considered a "local" call, right? like in alaska, where the whole state is 907? |
| 23:36.42 | Daejeo | anyone aware of CT-C121 IP Phone ? |
| 23:36.57 | [TK]D-Fender | lunaphyte, entirely possible |
| 23:36.57 | jblack | Daejeo: That sounds like a star-wars bot. :) |
| 23:36.59 | Daejeo | http://www.cyber-telecom.net/index.php/cPath/39 |
| 23:37.29 | lunaphyte | how do you know if it's long distance or not, in that case? |
| 23:37.38 | jblack | lunaphyte: There's long distance, there's local distance, then a particularly nasty beast called extended distance, which is between the two. |
| 23:37.39 | sbingner | didn't all the states go to requiring 1-NXX before all long distance, even in same area code? |
| 23:37.46 | [TK]D-Fender | Daejeo, just a rebranding of this stit-for-all PA1688 phone |
| 23:37.55 | [TK]D-Fender | Daejeo, by Atcom (320 I think it was) |
| 23:37.58 | [TK]D-Fender | shit* |
| 23:38.04 | Daejeo | oh |
| 23:38.06 | jblack | lunaphyte: You call the phone company to know. The phone book can help too. Silly, but that's what it is |
| 23:38.26 | sbingner | I found http://www.localcallingguide.com/ to be useful |
| 23:38.32 | lunaphyte | i see. so either risk it, or ask for every new prefix you might be calling. |
| 23:38.58 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 23:39.33 | [TK]D-Fender | Alrighty I'm off for a bit, later all |
| 23:39.37 | sbingner | or, you could just pass on the call the way the customer dialed it and if it's long distance and they didn't put a 1 it won't work... right? |
| 23:40.01 | andrewn | sbingner, no. in california you still can have toll calls 7 digits |
| 23:40.13 | sbingner | 7 digits w/o a 1 before it? |
| 23:40.15 | andrewn | yes |
| 23:40.19 | sbingner | that's evil |
| 23:41.01 | andrewn | and local calls to other area codes are 1+area code+number |
| 23:41.02 | sbingner | california has a telco-local number lookup page too |
| 23:41.06 | sbingner | forgot the website |
| 23:41.17 | sbingner | wow, that's messed up |
| 23:41.33 | andrewn | where do you live? |
| 23:42.19 | *** part/#asterisk jmls (n=jmls@81.138.42.77) |
| 23:43.33 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 23:43.54 | sbingner | hawaii |
| 23:44.08 | lunaphyte | looking at www.localcallingguide.com - see see, for my area code/prefix, 3 entries. one is nxx-xxx = verizon, another is nxx-xxx-0 = some other company, and a third is nxx-xxx-1 = a 3rd company. does that potentially mean i could get service from someone other than verizon? |
| 23:44.21 | lunaphyte | err, i see, even. |
| 23:44.43 | WilliamK | I know of localcallingguide and also telcodata.us |
| 23:44.46 | sbingner | lunaphyte, they could be CLECs? |
| 23:46.26 | lunaphyte | i'm not sure. i was always under the impression that there were no clec in my area, that i had to live with verizon. |
| 23:47.02 | sbingner | what is your area code/prefix then? |
| 23:47.10 | lunaphyte | 269.668 |
| 23:48.20 | sbingner | ACD Telecom provides Phone Service, DSL and Internet Backhaul service to other telecommunications carriers |
| 23:48.34 | lunaphyte | yeah - that's what caught my attention. |
| 23:49.06 | sbingner | I dont think you can really use those other 2 for small stuff based on what I see |
| 23:49.21 | lunaphyte | that's too bad. |
| 23:49.39 | sbingner | why don't you like vz? |
| 23:51.13 | lunaphyte | they don't care about their customers, only their customers' money. |
| 23:51.52 | sbingner | isn't that the definition of any business? |
| 23:52.06 | troubled | curious, does call waiting work while the called phone is at a dial tone? or only when they are connected and talking with someone? |
| 23:52.28 | lunaphyte | they are very much intertwined, certainly, but i think one has to come before the other will. |
| 23:54.21 | lunaphyte | they have grown in size to the point where 1 small, meaningless customer, if unhappy, has virtually 0 impact on their bottom line, unless that customer make it their life's crusade to have an impact, which almost certainly means a huge impact on that customer's life. |
| 23:54.37 | troubled | anyone? |
| 23:57.19 | lunaphyte | that's ok, if i, as a customer, can patronize a different company for the service i require. |
| 23:57.53 | lunaphyte | but when i can't, and the company behaving like this is using the money i give them to make it even harder for competitors to enter the market, i have a problem with that. |
| 23:58.32 | sbingner | troubled, test it... but I think you have to be in a call |
| 23:58.46 | lunaphyte | i end up being forced to relinquish the only power inherent in being a consumer, which is the right to choose. |
| 23:58.59 | sbingner | lunaphyte, you can choose... just voip |
| 23:59.06 | lunaphyte | anyway, blah blah blah an all the other stuff. :) |
| 23:59.24 | lunaphyte | absolutely - hence my presence here ;) |
| 23:59.59 | sbingner | or, you can become a CLEC yourself |