| 00:06.54 | *** part/#asterisk jtoy_ (n=jtoy@c-24-60-178-47.hsd1.ma.comcast.net) |
| 00:08.52 | Lann | i'd need to in that case |
| 00:09.09 | *** join/#asterisk lee_is_me (n=chatzill@12-201-102-196.client.mchsi.com) |
| 00:09.11 | Lann | one for ambient sound effects, one for people talking and the system's descriptions, one for interface |
| 00:09.12 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 00:09.49 | lee_is_me | Anyone use the Octware EC? I just installed it on our system and a customer's system. Pretty nice. |
| 00:10.03 | Lann | no |
| 00:12.14 | Vorondil | I see... perhaps you can accomplish that with a MeetMe room. |
| 00:12.41 | *** join/#asterisk pfn (n=pfnguyen@hanhuy.com) |
| 00:15.24 | Teccy | ok, ive sort of narrowed down my tdm400 dialling issue using ztmonitor |
| 00:15.54 | Teccy | if its in its normal state, i can hear a hum on the line, and i can hear stuff if someone else picks up a receiver on the same line |
| 00:16.19 | *** part/#asterisk chronomex (n=duncan@c-24-19-6-204.hsd1.mn.comcast.net) |
| 00:16.33 | Teccy | if i dial out, the hum disappears, and i hear the dtmf dialling, but then nothing, it's just a silent line, however i can hear what the person on the asterisk end is saying |
| 00:16.54 | Teccy | though if i pick up that other phone connected to the same pstn line, i just hear a dialtone |
| 00:17.15 | *** join/#asterisk bcnl (n=mike@S010600131078957c.vc.shawcable.net) |
| 00:17.25 | Teccy | its as if it's in the 'picked up' state to start with, then hangs up before dialing, if ya get me |
| 00:17.28 | Teccy | any thoughts? |
| 00:21.02 | *** join/#asterisk MrTelephone (n=test@bas13-toronto63-1178013079.dsl.bell.ca) |
| 00:21.25 | MrTelephone | hey I got this adit 600 hooked up to a nortstar CICS and the norstar CICS takes forever to release the channel when a caller hangs up |
| 00:21.32 | MrTelephone | is that a CICS configuration issue? |
| 00:28.28 | tzanger | res_csis? |
| 00:28.52 | tzanger | MrTelephone: adit600 FXO has CPD, is the CICS configured to detect it? |
| 00:30.29 | MrTelephone | CPD? |
| 00:30.37 | tzanger | called party disconnect |
| 00:30.42 | tzanger | when the remote end of the call hangs up |
| 00:30.57 | tzanger | bell canada will either drop battery or reverse battery polarity to signal that the far end has hung up |
| 00:31.03 | tzanger | adit600 fxo cards detect this |
| 00:31.08 | tzanger | and signal it appropriately |
| 00:31.16 | tzanger | I know because I've had working setups use that exact hardware |
| 00:31.19 | MrTelephone | adit600 fxs modules im using into the cics |
| 00:31.31 | tzanger | descript your PSTN connection please |
| 00:31.34 | tzanger | er describe |
| 00:31.51 | MrTelephone | SANGOMA PRI -> ADIT 600 -> FXS -> CICS |
| 00:32.14 | MrTelephone | i can make calls fine but when I call the CICS and the auto attendant picks up it takes a real long time for the line to be released |
| 00:32.20 | tzanger | sangoma pri? |
| 00:32.34 | MrTelephone | PRI on my asterisk box |
| 00:32.35 | tzanger | you mean a sangoma card to telco as PRI, and another T1 to the adit600? |
| 00:32.40 | MrTelephone | yeah |
| 00:32.52 | tzanger | are you using some kind of weird PRI card on the adit 600? |
| 00:33.03 | tzanger | or is it CAS T1 between the Sangoma and Adit600? |
| 00:33.17 | MrTelephone | i think call supervision is turned on, is that a ambiguous with CPD? |
| 00:33.32 | MrTelephone | its a CAS |
| 00:33.38 | tzanger | MrTelephone: on the CICS, perhaps... I'm a MICS guy, CICS is a little out of my knowledge base |
| 00:33.41 | tzanger | MrTelephone: ok |
| 00:33.51 | MrTelephone | CAS meaning robbed bit signalling? |
| 00:33.56 | tzanger | so PSTN <PRI> Asterisk <CAS T1> Adit600 <FXS> CICS trunk lines |
| 00:34.00 | tzanger | MrTelephone: yes |
| 00:34.04 | MrTelephone | yeah |
| 00:34.07 | tzanger | ok |
| 00:34.09 | MrTelephone | thats my setup |
| 00:34.24 | MrTelephone | should i turne echo off on the CAS t1? |
| 00:34.29 | tzanger | and when the call on the PSTN side hangs up, does Asterisk see the hangup immediately? |
| 00:34.40 | MrTelephone | im calling from a sip phone to the adit600 |
| 00:34.46 | tzanger | ok, same question |
| 00:34.46 | MrTelephone | asterisk says hangup zap/25 |
| 00:35.05 | tzanger | when the SIP phone hangs up, asterisk sees the hangup and you see the "hangup zap/25" message right at that time? |
| 00:35.11 | MrTelephone | yeah |
| 00:35.21 | tzanger | but the CICS does not see it? What about the Adit600, does the FXS port LED turn from amber to green right away? |
| 00:35.34 | MrTelephone | no |
| 00:35.44 | MrTelephone | the cics holds on to the line |
| 00:35.48 | MrTelephone | it stays orange |
| 00:35.54 | MrTelephone | it eventually hangs up |
| 00:35.55 | tzanger | the Adit600 LED turns the LED from amber to green several seconds afterward? |
| 00:36.10 | tzanger | the CICS isn't in control of that |
| 00:36.16 | MrTelephone | yeah I can phone it after a long time |
| 00:36.24 | tzanger | what channel signaling are you using on the CAS T1, and what is your span config like on the Adit600 T1 port? |
| 00:36.45 | MrTelephone | b8zs, esf |
| 00:36.48 | tzanger | you want fxo_ks for asterisk and ... shit what is it on the Adit600, LSCPD I think |
| 00:36.57 | tzanger | no that's the T1 provisioning, I want the robbed bit signaling |
| 00:37.15 | MrTelephone | fxols |
| 00:37.26 | tzanger | change that to ks |
| 00:37.30 | tzanger | you are not providing CPD |
| 00:37.44 | MrTelephone | ks is cpd and ls isn't? |
| 00:37.51 | tzanger | LS does not provide CPD, KS (idiotic naming, but that's OSS for you) is LS with CPD |
| 00:37.54 | MrTelephone | on the adit600 i only have choice of ls |
| 00:38.12 | tzanger | MrTelephone: that may be okay then, I am sure my Adit600 has LSCPD or something like that |
| 00:38.24 | MrTelephone | i'll change it in asterisk first and see |
| 00:39.06 | tzanger | stop asterisk and restart it, and re-run ztcfg again as well (after changing /etc/zapata.conf) -- I am not sure if it was ever changed so reloads actually changed the signaling or not |
| 00:39.25 | MrTelephone | any zap changes for me required a restart |
| 00:39.29 | tzanger | it used to not be the case, but I've been using asterisk so long now I don't remember if some of hte oldies have been fixed :-) |
| 00:40.09 | Qwell | http://slashdot.org/article.pl?sid=07/06/15/2016246 that freaking sucks |
| 00:40.23 | tzanger | I'm reading /. too heh |
| 00:40.30 | *** join/#asterisk A-Data (n=asd@196.218.18.125) |
| 00:40.46 | MrTelephone | i think sms should be free |
| 00:41.08 | MrTelephone | tell the oil rich arseholes to give them money instead of normal people |
| 00:41.39 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 00:41.53 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 00:42.17 | A-Data | Hello all . What need to be typed in context to run AGI script? |
| 00:43.17 | tzanger | A-Data: a preliminary search on google or voip-info.org would reveal such basic information |
| 00:44.39 | MrTelephone | tzanger, so if it doesn't work I should look for lscpd option in the adit600 config? |
| 00:44.45 | Nuitari | http://yro.slashdot.org/yro/07/06/15/2118213.shtml <-- this is a bit worse then paying for sms |
| 00:44.51 | MrTelephone | im waiting for all the calls to be done so i can restart asterisk |
| 00:45.08 | tzanger | MrTelephone: something along that line yes -- I can't get to my adit600 at this point in time to tell you for sure |
| 00:46.30 | A-Data | exten => 400,n,AGI(my-agi-script) is this the correct format? |
| 00:47.26 | _DAW | MrTelephone: lscpd is under line configuration ie.. set 1:1 signal lscpd |
| 00:47.44 | tzanger | _DAW: yeah that sounds about right |
| 00:49.05 | MrTelephone | tzanger u probably junk around with call pilots too? |
| 00:49.12 | tzanger | MrTelephone: nope never touched one |
| 00:49.22 | MrTelephone | I tried to hook up my laptop to the eth port but i couldn't get a link light but it worked with another laptop |
| 00:49.24 | MrTelephone | really wierd |
| 00:49.32 | tzanger | MrTelephone: try a crossover cable |
| 00:49.46 | A-Data | exten => 6600,n,AGI('asterisk-perl-0.09/examples/agi-sayani.agi') << give me person not found any clue |
| 00:49.52 | MrTelephone | ohhhh |
| 00:50.02 | MrTelephone | yes i guess one netcard might have done an auto crossover |
| 00:50.05 | MrTelephone | never thought of that |
| 00:50.09 | MrTelephone | nice |
| 00:50.43 | A-Data | exten => 6600,n,AGI('asterisk-perl-0.09/examples/agi-sayani.agi') << give me person not found any clue |
| 00:52.51 | MrTelephone | tzanger, have u ever had to play with the gains on the adit600? |
| 00:53.56 | A-Data | exten => 6600,n,AGI('asterisk-perl-0.09/examples/agi-sayani.agi') << give me person not found any clue |
| 00:55.07 | tzanger | MrTelephone: never had to on fxs |
| 00:55.23 | *** join/#asterisk Marshall- (n=Marshall@cpe-76-181-119-87.columbus.res.rr.com) |
| 00:55.30 | *** join/#asterisk Marshall-Laptop (n=eman0n@cpe-76-181-119-87.columbus.res.rr.com) |
| 00:56.52 | A-Data | exten => 6600,n,AGI('asterisk-perl-0.09/examples/agi-sayani.agi') << give me person not found any clue |
| 00:58.17 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 00:58.28 | A-Data | exten => 6600,n,AGI('asterisk-perl-0.09/examples/agi-sayani.agi') << give me person not found any clue |
| 00:58.35 | *** join/#asterisk aksnowman (n=john@rdbck-2701.wasilla.mtaonline.net) |
| 00:58.40 | A-Data | wb [TK]D-Fender |
| 00:58.58 | MrTelephone | i get the odd bit of distortion |
| 00:59.11 | MrTelephone | once every couple minutes or something |
| 00:59.22 | MrTelephone | sounds like an echo cancel problem actually |
| 01:00.32 | [TK]D-Fender | A-Data, how are we to know what in that magical little agi of yours? we're not PSYCHIC |
| 01:01.17 | file | A-Data: repeating your question 5 times is quite... rude |
| 01:01.36 | A-Data | [TK]D-Fender i used the agi example on perl |
| 01:01.41 | A-Data | file sorry was my script |
| 01:02.13 | [TK]D-Fender | file, make that 6... he PM'd me on arrival as well. |
| 01:02.40 | file | A-Data: do you have that disabled now? |
| 01:02.51 | A-Data | yes file |
| 01:03.44 | A-Data | any how ... i know i am noob .. but i feel that every one here thinking that noob is a crime i think all of you were noob and all of you asked more questions that i do |
| 01:03.53 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 01:03.59 | A-Data | if you don`t welcome me here tell me and i will leave for ever |
| 01:04.34 | aksnowman | A-Data: the main way to no longer be a "noob" *is* to ask questions |
| 01:04.58 | aksnowman | the difference is how politely you ask your questions and who happens to be here when you ask |
| 01:04.59 | [TK]D-Fender | A-Data, there's a difference between being a "n00b" and being obnoxious and repeating yourself to EVERYONE over and over and in PM as well. |
| 01:05.08 | A-Data | aksnowman that what i was doing asking and they feel me it crime |
| 01:06.25 | A-Data | [TK]D-Fender beacuse i said wb the script paste it for you in pvt |
| 01:06.45 | [TK]D-Fender | A-Data, asking over and over is DAMN annoying. We heard you the first 5 times. If we had anything to say we'd have said it. You are showing any communications skills or attempts to better explain your problem. You are merely repeating yourself. |
| 01:06.52 | MrTelephone | tzanger, that seemed to solve the problem |
| 01:07.17 | tzanger | MrTelephone: awesome |
| 01:07.45 | MrTelephone | it says starting simple switch after hangup |
| 01:08.18 | aksnowman | so, I've got asterisk installed, I can connect to it as can others, but we can't seem to talk to eachother (other than leaving voicemails for eachother), we're using softphones, was wondering if anyone might know why we can't talk to eachother or in a conf extension |
| 01:09.28 | MrTelephone | aksnowman, use canreinvite=no in your sip.conf for each client |
| 01:10.24 | aksnowman | thanks, will try |
| 01:10.53 | tzanger | MrTelephone: that's perfectly acceptable |
| 01:10.58 | tzanger | the CICs likely does not hang up immediately |
| 01:11.21 | MrTelephone | yeah anything under 60 seconds is good for me |
| 01:12.13 | *** join/#asterisk Math` (n=privmath@modemcable037.229-56-74.mc.videotron.ca) |
| 01:12.49 | Math` | any reason why the 'g' dial option (continue dialplan execution here after call) doesnt work when you Dial() something with chan_local? |
| 01:12.55 | MrTelephone | the people who designed sip meant to make it so complicated with different ports for rtp tx and rx? |
| 01:13.18 | Nuitari | MrTelephone: yes |
| 01:13.30 | MrTelephone | math try using M() option in Dial |
| 01:13.36 | MrTelephone | I used M() with success |
| 01:13.51 | Math` | that doesnt do the same thing |
| 01:14.00 | Math` | g keeps executing when the call is over |
| 01:14.06 | Math` | M() calls a macro when the call is established |
| 01:14.21 | MrTelephone | is an rtp trunk just a link between to sip gateways or does a real sip trunk actually multiplex rtp packets into bigger packets? |
| 01:14.31 | MrTelephone | oops sorry math |
| 01:14.48 | file | Math`: perhaps your Local channels are getting optimized out of the way and the place where you did the Dial with g is stopping because of it? |
| 01:15.10 | Math` | I tought Local would isolate that optimization behavior |
| 01:15.19 | file | isolate? |
| 01:15.32 | MrTelephone | are you guys speaking english? |
| 01:15.43 | file | it's a feature of chan_local to optimize itself out of the bridges, you can disable it by adding /n to the end |
| 01:15.54 | Math` | nice, is that documented anywhere? |
| 01:16.02 | *** join/#asterisk anthm][ (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 01:16.10 | Math` | or its a "vim chan_local.c" kinda thing |
| 01:16.45 | file | I do not know |
| 01:22.14 | *** part/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 01:23.23 | Math` | the inner Local channel does a Dial() outside too, but its still ignoring my 'g' option |
| 01:23.26 | Math` | even with /n even tought I see more Local/ channel so they are not optimized |
| 01:24.08 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 01:24.08 | *** mode/#asterisk [+o mog] by ChanServ |
| 01:25.34 | MrTelephone | anyone have a web address for the RBS bits |
| 01:27.18 | _DAW | Try http://www.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a00800e2560.shtml |
| 01:27.46 | aksnowman | anyone have any suggestions for softphones to use? (on windows) |
| 01:28.28 | _DAW | Xlite is nice |
| 01:29.24 | aksnowman | k |
| 01:29.27 | Math` | try idefisk, it does both SIP and IAX |
| 01:29.39 | Math` | xlite blocks you from doing call transfers if u dont buy their full version |
| 01:29.50 | *** mode/#asterisk [+o anthm] by ChanServ |
| 01:29.53 | aksnowman | k, thanks guys |
| 01:30.41 | [TK]D-Fender | Math`, Pastbin your dialplan so we can take a look... |
| 01:33.03 | Math` | well its pretty straightforward... http://voip.acetix.ca/dialplan.txt |
| 01:33.58 | Math` | and show channel reports the right extension/priority when the Dial(Local/....) is executing |
| 01:34.06 | *** join/#asterisk dijungal (n=kdaniel@64.86.52.254) |
| 01:34.56 | dijungal | hello... will a H.323 call show up in the asterisk CLI debug window even if the channel is not configured properly? |
| 01:35.42 | dijungal | i ask because i've been trying to configure asterisk to recieve outpulse calls from a Cisco 3661 router with ZERO luck |
| 01:35.53 | dijungal | it's like the calls not even getting to the box |
| 01:36.17 | Math` | when u sniff do u see the packets "trying" to make their way through? |
| 01:37.08 | dijungal | how do i do that |
| 01:37.11 | dijungal | i am on centos |
| 01:37.18 | dijungal | i tried netstat but that's no help |
| 01:37.20 | *** join/#asterisk steve___ (n=steve@store-fw.porchlight.ca) |
| 01:37.20 | Teccy | tcpdump |
| 01:37.28 | dijungal | ahhhh |
| 01:37.34 | dijungal | i will try that |
| 01:37.36 | Math` | or tshark (ethereal's new name, console version) |
| 01:37.45 | Math` | its going to actually decode h323 packets so it might be good to have |
| 01:38.25 | dijungal | i've been wondering what i can use to see if the traffic is even getting to the box |
| 01:38.42 | dijungal | i'll try the tcpdump it looks like it comes with centod |
| 01:38.44 | dijungal | centos |
| 01:40.08 | *** part/#asterisk MrTelephone (n=test@bas13-toronto63-1178013079.dsl.bell.ca) |
| 01:40.54 | dijungal | Math: can i do a dcpdump for a specific addres? |
| 01:41.00 | Math` | yeah |
| 01:41.06 | Math` | tcpdump ip host [address] |
| 01:41.11 | dijungal | in other words i need to see if traffic is coming from a specific address |
| 01:41.13 | dijungal | ok thanks |
| 01:41.20 | Math` | you are gonna see traffic to and from that address |
| 01:42.47 | dijungal | ahhh it's monitoring the eth0, let me call in now to see what's going on |
| 01:44.21 | dijungal | ok just tried to call in... dead air then congestion tone |
| 01:44.29 | dijungal | and still no packets from that address |
| 01:45.17 | dijungal | so i guess the cisco router not getting the packets accross |
| 01:45.23 | dijungal | well that's one mystery solved |
| 01:45.28 | dijungal | Math: thanks much |
| 01:45.48 | dijungal | now i get to shout at the network tech guys!! :) |
| 01:45.53 | Math` | haha |
| 01:46.47 | dijungal | i can see the conversation now... "u'r not sending me taffic damit!!"... "no yuh stupid pbx thing... aster..whatever it names is crap.. get a cisco callmanager pbx"... |
| 01:48.21 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 01:48.26 | *** join/#asterisk dlynes_laptop (n=dlynes@d154-20-9-152.bchsia.telus.net) |
| 01:49.54 | *** join/#asterisk [hC] (n=hardcore@S0106000d8891877c.vc.shawcable.net) |
| 02:15.00 | *** join/#asterisk Marshall- (n=Marshall@cpe-76-181-119-87.columbus.res.rr.com) |
| 02:28.03 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 02:31.25 | Nuitari | hey russellb |
| 02:31.46 | Nuitari | Is there a way to get a list of devices set by func_devstate ? |
| 02:32.38 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 02:36.09 | aksnowman | hi, I can reach and talk to myself in a confrence call via softphones on my local network using my external IP, but the other guys I'm trying to get connected show they're connected, but on my asterisk console it shows "User hung up" each time they connect |
| 02:36.50 | russellb | Nuitari: yeah ... there is a CLI command |
| 02:36.56 | russellb | Nuitari: but ... i don't remember what it is |
| 02:37.23 | russellb | Nuitari: "funcdevstate list" |
| 02:37.44 | Nuitari | Thanks |
| 02:37.48 | Nuitari | should have though about that |
| 02:37.57 | Nuitari | hum, all unknown |
| 02:39.32 | Nuitari | inuse shows up though |
| 02:40.38 | Nuitari | not_inuse too, but not idle |
| 02:41.29 | *** join/#asterisk chr05210084 (n=root@203.115.187.97) |
| 02:48.14 | aksnowman | anyone know why I would be able to connect from local network but external users wouldn't? (softphones, and asterisk box is set up as dmz) |
| 02:49.56 | *** join/#asterisk mutilator (n=WebChat@the.drinkproject.com) |
| 02:50.38 | dijungal | when is a gatekeeper needed for H.323? If two PBXs are trying to send calls between them and they're both on public IP addresses do they need a gatekeeper? |
| 02:52.08 | Math` | u dont need any in that case u can just do gw to gw |
| 02:52.52 | dijungal | nice |
| 02:53.41 | dijungal | the cisco router has a public IP and my PBX has a public ip so they should not need a gatekeeper between |
| 02:54.03 | Math` | you mean the cisco voip gw? |
| 02:54.19 | Math` | cant those do sip anyways? |
| 02:56.36 | *** join/#asterisk karleeto (i=karl@gentoo.karlhaines.com) |
| 02:57.05 | karleeto | can anyone reccomend a linux based software iax2 phone? |
| 03:03.34 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:05.07 | *** join/#asterisk PtitGNU (n=ptitgnu@2001:6f8:32c:0:218:f3ff:fe4f:570b) |
| 03:05.47 | PtitGNU | hi |
| 03:07.18 | dijungal | Math: apparently the 3661 can only do H.323 |
| 03:07.20 | dijungal | no sip :( |
| 03:10.30 | PtitGNU | I need to install an asterisk server for production use. (so, it must be stable (really)) Currently, the server is running Gentoo in x86_64 (amd64)... I must make a choice between the 1.2.x or the 1.4.x but the 1.2.x is more mature I think (don't it ?) and I don't know anything about the stability of asterisk on x86_64 arch... What do you think ? |
| 03:10.48 | *** join/#asterisk grndslm (n=grndslm@69.92.23.67) |
| 03:10.52 | Qwell | PtitGNU: x86_64 is fine |
| 03:11.21 | karleeto | Qwell: well, what sources are you using in gentoo? gentoo-sources ? |
| 03:11.37 | Qwell | for kernel you mean? |
| 03:11.40 | karleeto | <--gentoo dev (being mentored atm) |
| 03:11.44 | PtitGNU | I'm using gentoo-sources... 2.6.21-r3 |
| 03:12.02 | Qwell | karleeto: I don't understand the question |
| 03:12.08 | karleeto | PtitGNU: you may want to use vanilla-sources |
| 03:12.09 | Qwell | or, rather, the reason for the question |
| 03:12.19 | karleeto | Qwell: thats cause you are not a gentoo person, obvioisly |
| 03:12.36 | Qwell | Gentoo Base System release 1.12.9 |
| 03:12.39 | Qwell | I most certainly am |
| 03:12.52 | PtitGNU | karleeto: why ? there is a problem with asterisk and dsd patches ? |
| 03:13.27 | Qwell | what I'm asking, is why would kernel version matter? Are there known issues with gentoo-sources and amd64? |
| 03:13.56 | karleeto | PtitGNU: gentoo-sources has all kinds of gentoo patches added to it. if you are experiencing instability, and are concered about it, you might wanna use a kernel that is stable and not patched with preformance patches |
| 03:14.24 | karleeto | cause on a stable server you should not be concerned about improved desktop functionality in the kernl |
| 03:14.34 | karleeto | WHATVER. just a suggestion |
| 03:14.38 | Qwell | karleeto: This is why gentoo needs a server profile :) |
| 03:14.50 | karleeto | Qwell: it does |
| 03:14.53 | karleeto | and a |
| 03:15.02 | karleeto | "hardened |
| 03:15.04 | karleeto | " |
| 03:15.09 | Qwell | meh, that's a use flag |
| 03:15.09 | karleeto | system. at, |
| 03:15.13 | karleeto | yeah |
| 03:15.27 | karleeto | along with the server profile it builds a great secure server |
| 03:15.43 | karleeto | of course they all need their server admin's attention ;) |
| 03:16.25 | *** join/#asterisk coppice (n=chatzill@10.198.17.210.dyn.pacific.net.hk) |
| 03:17.00 | karleeto | anyway, sonsider that PtitGNU |
| 03:17.10 | karleeto | consider |
| 03:17.18 | karleeto | <--- has had some beers |
| 03:17.35 | PtitGNU | performance patches ? It is essentially some bug fixes and backports from -rc (to fix some bugs too)... http://dev.gentoo.org/~dsd/genpatches/releases-2.6.21.htm |
| 03:18.08 | karleeto | well, good then, you would have the best asterisk box you could hope for if you maintain it correctly |
| 03:18.13 | karleeto | IMO |
| 03:18.16 | karleeto | i love gentoo |
| 03:18.24 | karleeto | lol. do a whois on me ;_ |
| 03:18.41 | rob0 | I have a long-running (but low load) * on x86_64. |
| 03:18.55 | karleeto | rob0: what OS? |
| 03:19.30 | rob0 | Linux, slamd64 |
| 03:19.31 | dlynes_laptop | good morning, coppice |
| 03:19.41 | coppice | hi |
| 03:19.47 | dlynes_laptop | rob0: how stable is slamd64? |
| 03:19.57 | dlynes_laptop | rob0: is it as stable as the regular slackware? |
| 03:20.45 | rob0 | It's more stable than my electric service. :) I get outages which outlast my UPS, otherwise I'd have 2 years' uptime. |
| 03:20.48 | Qwell | karleeto: Where's your gentoo cloak? |
| 03:20.56 | Qwell | You should get one |
| 03:21.07 | karleeto | Qwell: i'm in mentor stage arm |
| 03:21.09 | karleeto | err atm |
| 03:21.20 | PtitGNU | in fact, I asked this because I have some friends that say x86_64 is useless and it make * less stable... so :) and for 1.2.x vs 1.4.x ? |
| 03:21.34 | karleeto | Qwell: another 2 weeks or so (when my mentor, UberLord) gets back from his honeymoon |
| 03:21.47 | Qwell | PtitGNU: 1.4, x86_64 is what I'd do, personally |
| 03:21.49 | karleeto | Qwell: i'm gonna be involved in Gentoo FreeBSD |
| 03:22.00 | dlynes_laptop | lord help us |
| 03:22.09 | Qwell | yeah... |
| 03:22.33 | rob0 | x86_64 is the future. |
| 03:22.42 | karleeto | rob0: i hope so |
| 03:22.43 | denon | alpha's the future! |
| 03:22.43 | dlynes_laptop | rob0: I was referring ot Gentoo FreeBSD :) |
| 03:22.45 | Qwell | x86_64 is the...present |
| 03:22.52 | Qwell | x86 is the past |
| 03:22.53 | Qwell | move on |
| 03:23.13 | dlynes_laptop | Qwell: UltraSPARC is the present |
| 03:23.14 | PtitGNU | Qwell: I think so :) |
| 03:23.15 | denon | everyone and their dog already has an x86_64 cpu, half of em dont even realize it |
| 03:23.26 | karleeto | i still am using p3's, p4s, xeon's, etc |
| 03:23.31 | Qwell | dlynes_laptop: UltraSPARC is the future :p |
| 03:23.37 | karleeto | hardware i can get my hands on |
| 03:23.43 | dlynes_laptop | Qwell: The past, the present and the future |
| 03:23.54 | Qwell | mmm...sparc |
| 03:24.08 | Qwell | sparcs probably have some of the highest uptimes |
| 03:25.08 | dlynes_laptop | Qwell: Except for mine....they're collecting dust because I've been too busy doing taxes, moving, and gearing up for a wedding :9 |
| 03:25.08 | Qwell | I have an old ss20...110mhz...it rocks |
| 03:25.08 | Qwell | used it as my router for a good 3 years |
| 03:25.11 | dlynes_laptop | Qwell: Yeah...the Netra T1 is one of the nicest machines I've ever used |
| 03:25.25 | Qwell | T2000 is my current favorite ;p |
| 03:25.42 | dlynes_laptop | Qwell: I've got a Sunfire v250 here that kicks pretty good ass, too |
| 03:25.59 | dlynes_laptop | Qwell: Looking at selling it though...need to be debt free before I get married :p |
| 03:26.08 | Qwell | bad idea |
| 03:26.18 | PtitGNU | Qwell: well, I have the same question to spyroux (I don't know if you know who is he), and he answered exactly the opposite :D (1.2 on x86)... I'm confused :p |
| 03:26.21 | dlynes_laptop | Qwell: The wedding, or the selling of the sparc? |
| 03:26.26 | PtitGNU | asked* |
| 03:26.30 | Qwell | both |
| 03:26.33 | dlynes_laptop | lol |
| 03:26.46 | Qwell | If you sell it now, you'll never get another one...ever :P |
| 03:26.46 | coppice | dlynes_laptop: you'll find it much harder to buy after marriage |
| 03:26.52 | Qwell | exactly |
| 03:26.54 | dlynes_laptop | lol |
| 03:27.10 | Qwell | be in AS MUCH DEBT as you can when you get married. It only gets worse. :p |
| 03:27.10 | dlynes_laptop | Well, I'm hoping things will go differently |
| 03:27.28 | karleeto | LISTEN to Qwell ! |
| 03:27.30 | dlynes_laptop | We've got a number of things on the horizon to make some good coin |
| 03:27.34 | Qwell | heh |
| 03:27.37 | karleeto | i speak of experience |
| 03:27.42 | dlynes_laptop | None of which include computers |
| 03:27.54 | PtitGNU | mmmh, and I don't have any 1.4.x ebuild on gentoo :/ |
| 03:27.56 | coppice | smart move |
| 03:28.08 | Qwell | PtitGNU: don't use the ebuild |
| 03:28.22 | karleeto | PtitGNU: emerge --sync |
| 03:28.25 | Qwell | no offense to the maintainers, but it's very difficult for some reason to get a good package of asterisk |
| 03:28.28 | dlynes_laptop | Yeah...real estate and mlm |
| 03:28.29 | karleeto | PtitGNU: you should do that anyway |
| 03:28.36 | *** join/#asterisk CuriosCat (i=stian@mack.bigrig.org) |
| 03:29.02 | Qwell | I don't even want to know what the USE flags would look like for that... |
| 03:29.11 | dlynes_laptop | Still planning on pursuing voip and asterisk though |
| 03:29.26 | Qwell | a good deal of the modules depend on other things, all of which would need to be a USE flag |
| 03:29.54 | karleeto | Qwell: asterisk stuff will get better real soon in gentoo as well, thats my other group i'm in ;) |
| 03:30.11 | Nuitari | karleeto: thankfully there is also layman |
| 03:30.19 | karleeto | Qwell: as well as aterisk stuff on gentoo freebsd, which has a much stabler kernel than linux |
| 03:31.04 | dijungal | ok thats enuff VOIP for me for one day |
| 03:31.04 | karleeto | Nuitari: whats layman? |
| 03:31.08 | dijungal | gotta go grap a beer |
| 03:31.12 | dijungal | *grab |
| 03:31.13 | dijungal | lata |
| 03:31.33 | karleeto | dijungal: i'm already drinking one ;) glad for you to join |
| 03:31.35 | karleeto | have fun |
| 03:31.39 | Nuitari | karleeto: portage overlays |
| 03:31.49 | dijungal | lol |
| 03:31.49 | dijungal | k |
| 03:31.51 | Nuitari | karleeto: http://gentoo-wiki.com/TIP_Overlays |
| 03:31.51 | *** part/#asterisk dijungal (n=kdaniel@64.86.52.254) |
| 03:32.00 | karleeto | Nuitari: i know about overlays ;) |
| 03:32.05 | karleeto | Nuitari: i see now ;) |
| 03:32.40 | Qwell | You know, there's one thing I've gotta say about gentoo... |
| 03:33.01 | Qwell | a lot of people don't like it for some reason or another, but I think everybody can agree that they've got some of the best documentation available |
| 03:33.32 | karleeto | anyone who wants to help, i need to people to get the project off the ground.. already got servers and domain name |
| 03:33.46 | Nuitari | start a wiki ? |
| 03:33.59 | karleeto | well, put one in place ;) |
| 03:34.27 | karleeto | jsut want to build a team, to help me get people involved |
| 03:34.53 | Nuitari | more up to date info then voip-info.org would help |
| 03:35.01 | *** join/#asterisk Strom_M (n=strom@dsl-202-173-183-69.vic.westnet.com.au) |
| 03:35.04 | Nuitari | most of the examples still use priorityjumping |
| 03:35.06 | karleeto | exactaly what i was thinking |
| 03:35.13 | rob0 | Why not just update voip-info.org ? |
| 03:35.17 | karleeto | Nuitari: and it would attract people to gentoo as well |
| 03:35.17 | rob0 | :) |
| 03:35.47 | karleeto | Nuitari: which IMHO, is the best base for any customized linux server |
| 03:36.00 | *** join/#asterisk Marshall- (n=Marshall@cpe-76-181-167-76.columbus.res.rr.com) |
| 03:36.01 | *** join/#asterisk SwK (n=SwK@m055e36d0.tmodns.net) |
| 03:38.23 | karleeto | anyway, WAYY OT there |
| 03:38.24 | PtitGNU | Qwell: so, you advise me to install asterisk by hands rather than the gentoo way... It's not very clean and I don't like it but if it's really really necessary I will do so... just for information: http://bugs.gentoo.org/show_bug.cgi?id=159013 |
| 03:39.01 | karleeto | but, Nuitari if you think at least that my idea would be a good one, advice would be appreciated, and help even more! Let me know |
| 03:39.15 | *** part/#asterisk SwK (n=SwK@m055e36d0.tmodns.net) |
| 03:42.14 | Nuitari | karleeto: of course it's a good idea |
| 03:43.06 | *** join/#asterisk saftsack (n=saftsack@pD9E04E71.dip.t-dialin.net) |
| 03:47.04 | *** join/#asterisk samarora (i=minesh@203.88.149.166) |
| 03:47.15 | samarora | hi there |
| 03:47.27 | samarora | can anybody help me to define context in extensions.conf |
| 03:47.39 | [TK]D-Fender | [heresacontext] |
| 03:47.44 | [TK]D-Fender | NEXT!@@!@!@ (c) BKW |
| 03:48.05 | samarora | i want to restrict users from making outgoing calls but are able to make internal extensions calls.. |
| 03:48.19 | samarora | hi TK |
| 03:48.23 | samarora | pl help me |
| 03:50.01 | rob0 | Have those users in a context which doesn't include access to the ${TRUNK}, but does include internal extensions? |
| 03:51.02 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 03:52.56 | *** join/#asterisk bmg505 (n=leon@196.209.181.95) |
| 03:54.53 | karleeto | Nuitari: seriously, if there is anything you'd like to contrib, some docs, some advice, etc, i love to talk with you more.. karl@karlhaines.com |
| 03:59.50 | *** join/#asterisk aksnowman (n=john@rdbck-3402.wasilla.mtaonline.net) |
| 03:59.55 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 04:04.47 | *** join/#asterisk `Sean (i=Un1x@CPE000c258d147c-CM000a73a94167.cpe.net.cable.rogers.com) |
| 04:09.17 | russellb | Qwell: other systems don't need as much documentation if they are easier and more intuitive to setup and use ... |
| 04:09.24 | Qwell | :p |
| 04:10.36 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:12.47 | *** join/#asterisk clyrrad (n=darryl@CPE000802212b48-CM0011aea484a4.cpe.net.cable.rogers.com) |
| 04:13.38 | clyrrad | Hey.... anyone famaliar with this message or how to get around it? rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. |
| 04:14.04 | [TK]D-Fender | clyrrad, Disable CNG in yuor client |
| 04:14.06 | [TK]D-Fender | NEXT!@@!@!@ (c) BKW |
| 04:14.29 | clyrrad | TKD: Excellent - do you know if there is such an option on Linksys Sipura? |
| 04:14.37 | clyrrad | I have reviewed the configs and did not notice that setting |
| 04:15.05 | Nuitari | clyrrad: yes there is |
| 04:15.27 | clyrrad | Nuitari: Is it called CNG? |
| 04:15.27 | [TK]D-Fender | clyrrad, Might be under VAD |
| 04:15.53 | clyrrad | Ok this should be under the SIP or under the acutal Extension Line Configuartion? I can check again :) |
| 04:16.04 | [TK]D-Fender | clyrrad, yes :) |
| 04:16.59 | clyrrad | Strange.... I cant locate any item called CNG or VAD...... |
| 04:17.23 | Nuitari | clyrrad: look for Silence Supp Enable |
| 04:17.33 | Nuitari | Under the advanced line configurations |
| 04:18.26 | clyrrad | Nuitari: Thanks checking on that now :) |
| 04:18.55 | Nuitari | took me a while to find it |
| 04:19.32 | clyrrad | Nuitari: Ok that is currently set to "No" |
| 04:20.07 | clyrrad | But I still get that message on the CLI.... |
| 04:23.40 | clyrrad | Nuitari: was there any other setting change you needed to make? |
| 04:24.21 | clyrrad | [TK]D-Fender: Could CNG be labeled something other than CNG, VAD or Silence Supp Enable? |
| 04:24.34 | Juggie | [TK]D-Fender, you got a sec? |
| 04:24.42 | [TK]D-Fender | clyrrad, yes. |
| 04:24.46 | [TK]D-Fender | Juggie, yes. |
| 04:25.33 | Juggie | [TK]D-Fender, i'm having a really weird problem, i shoudnt be stuck on this but none the less, i have a pri->asterisk box->iax2->another asterisk |
| 04:25.35 | Nuitari | not that I remember |
| 04:25.54 | Juggie | [TK]D-Fender, the second asterisk runs an agi which does a playback, but the audio doesnt go through |
| 04:26.03 | Juggie | the iax2 connection gets accepted ok, but no audio. |
| 04:26.12 | *** join/#asterisk BSD_Tech (n=BSDTech@adsl-69-230-169-41.dsl.irvnca.pacbell.net) |
| 04:26.29 | [TK]D-Fender | Juggie, Do you gt audio if you playback with the app & not AGI? |
| 04:26.51 | Juggie | [TK]D-Fender, no i added a background into the dialplan of the second * box and no audio |
| 04:27.01 | Juggie | however, if i do a background on the first * box, i get audio just fine |
| 04:27.06 | [TK]D-Fender | Juggie, Ok, then that rules out AGI. |
| 04:27.16 | clyrrad | [TK]D-Fender: hrm - any idea what else I may need to change? I found an article from google that suggested to have the provider disable CNG/VAD/Silence Supression - howerver the Asterisk warning indicates this can be done client side. I have confirmed the Silence Suppression is set to "No" on the phone - yet I still see this error on the CLI.... |
| 04:27.18 | BSD_Tech | whats going on |
| 04:27.19 | Juggie | so i've narrowed it down to iax. |
| 04:27.34 | Juggie | i copied default configs from the 1.4.4 src/configs |
| 04:27.37 | Juggie | and only added my hosts in |
| 04:27.37 | [TK]D-Fender | Juggie, Got any zaptel cards in there? |
| 04:28.13 | clyrrad | Nuitari: Thanks for your tips.... I still get the warning so must be something else :s - thanks for the info :) |
| 04:28.22 | Juggie | hmmMMm |
| 04:28.24 | Juggie | [TK]D-Fender, i know |
| 04:28.31 | Juggie | i have a zaptel card in the 2nd box |
| 04:28.36 | Juggie | w/ no active trunks |
| 04:28.40 | Juggie | hence no timeing |
| 04:28.47 | Juggie | need to unload the module and load ztdummy |
| 04:28.52 | Juggie | annoying 'gotcha' |
| 04:29.45 | Juggie | ah, it works now of course |
| 04:29.49 | Juggie | thanks for talking me though it :) |
| 04:29.51 | Juggie | dumb bug. |
| 04:29.54 | [TK]D-Fender | :) |
| 04:29.59 | [TK]D-Fender | my work here is done. |
| 04:30.58 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 04:34.37 | clyrrad | Guys I just noticed this warning comes before any dial is even issued to any phone, the warning comes up as soon as your press a KEY on your phone |
| 04:34.43 | clyrrad | it looks to be DTMF related |
| 04:34.50 | clyrrad | any ideas are apprcaited |
| 04:39.46 | [TK]D-Fender | clyrrad, Can't be before dial. |
| 04:40.18 | [TK]D-Fender | clyrrad, Otherwise there'd be no sound to suppress ;) |
| 04:42.48 | *** part/#asterisk samarora (i=minesh@203.88.149.166) |
| 04:43.03 | blitzrage | y0! |
| 04:43.27 | [TK]D-Fender | blitzrage, I DON'T WANT TO STILL BE UP... |
| 04:43.41 | blitzrage | I'D RATHER JUST.... |
| 04:43.44 | [TK]D-Fender | blitzrage, ! ! ! |
| 04:43.54 | blitzrage | you and me both buddy.... you and me both |
| 04:44.21 | [TK]D-Fender | blitzrage, tomorrow night I'm putting myself back out on the market and changing my scene |
| 04:45.08 | blitzrage | [TK]D-Fender: oh? |
| 04:45.32 | [TK]D-Fender | blitzrage, Yup, overdue and I'm working to get back into serious shape and fix up the image. |
| 04:45.40 | clyrrad | [TK]D-Fender: Indeed the warning does appear much before the Dial is executed, that warning appears when you call into the PBX and press a key on your cell phone. In otherwords, as soon as you send the first DTMF into the system is when that warning message appears, and that happens well before any phone starts ringing..... |
| 04:46.04 | [TK]D-Fender | clyrrad, then you ARE in a call.... |
| 04:46.09 | blitzrage | [TK]D-Fender: always a good thing. I've been biking and running a lot |
| 04:46.10 | [TK]D-Fender | clyrrad, just in an IVR. |
| 04:46.49 | [TK]D-Fender | clyrrad, At which point when the phone detects the DTMF you wish to send, it cuts audio for the playback time figuring it'll save on bandwidth |
| 04:47.08 | [TK]D-Fender | clyrrad, There is an option to disable it somewhere there. |
| 04:47.34 | clyrrad | [TK]D-Fender: my guess is this options is an option for Asterisk? I cant be in the phone since the Dial to the phone has not even happned yet |
| 04:47.41 | [TK]D-Fender | but alas I have to get up early and must be off to get some sleep |
| 04:47.42 | Nuitari | is there a way to execute applications in the cli ? |
| 04:47.49 | [TK]D-Fender | later all |
| 04:48.19 | clyrrad | Nuitari: do you mean like sip reload or soemthing like this? |
| 04:48.54 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:49.10 | Nuitari | I mean like DEVSTATE() |
| 04:50.39 | Nuitari | I guess I can just hack the friggin app |
| 04:51.23 | Nuitari | ok it has a cli |
| 04:51.25 | clyrrad | Nuitari: why not just use the Manager Ingerface? |
| 04:51.41 | Nuitari | I use the manager interface, but it's something that just isn't in there |
| 04:51.51 | *** join/#asterisk mindCrime_ (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 04:52.15 | Nuitari | and one of my clients wants to pay me to get the program that I work on to get hints between asterisk servers to get it to work on 1.4 |
| 04:53.08 | Nuitari | of course if I had an idea of how to get func_devstate working on 1.4, despite russellb saying that it will never be ported to it, it would be easier then messing around bristuff's app_devstate |
| 04:57.02 | russellb | Nuitari: how cool would i be if i just ported it for you real quick? |
| 04:57.57 | Nuitari | russellb: immensely cool! |
| 04:58.01 | russellb | heh |
| 04:58.10 | Nuitari | it would save me so much trouble |
| 04:58.39 | russellb | ooh, you're getting paid for this? |
| 04:58.44 | russellb | guess that means you should share :-p |
| 04:59.05 | Nuitari | not much I'm afraid |
| 04:59.15 | russellb | i'm just playing .. |
| 04:59.33 | Nuitari | I'll release my code under GPL anyways |
| 04:59.49 | blitzrage | russellb: I want it too :D |
| 04:59.51 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
| 04:59.54 | blitzrage | codefreeze: !!!!! |
| 05:00.56 | russellb | blitzrage: heh |
| 05:01.03 | russellb | i don't know where to put it, though ... |
| 05:01.10 | russellb | other backports are on svncommunity ... |
| 05:01.15 | blitzrage | svncommunity |
| 05:01.24 | russellb | that means i have to create a repo for myself on there |
| 05:01.25 | russellb | gah! |
| 05:01.27 | blitzrage | just in an app_devstate dir like func_odbc |
| 05:01.34 | blitzrage | blasted! :) |
| 05:01.40 | russellb | and then, set it up to get mirrored ... |
| 05:01.42 | russellb | what a pain |
| 05:01.51 | russellb | i'll do that part next week or something . .. |
| 05:03.00 | Juggie | russellb, got 30 seconds? |
| 05:03.17 | Juggie | want to run a quick problem by u, not for a fix but just curious as to what can be done w/ it. |
| 05:03.35 | Juggie | i'll say it anyways, hopefully you listen :) |
| 05:03.52 | Juggie | lets say i have a single span configured on any given zaptel card... and iax setup as a backup outgoing provider. |
| 05:04.13 | Juggie | if the span goes down, i have no timeing source for iax2 because of course ztdummy isnt loaded |
| 05:04.21 | Juggie | how does one solve this? |
| 05:04.54 | russellb | load ztdummy in addition to the card? |
| 05:05.08 | russellb | and zaptel is not required for iax ... |
| 05:05.19 | russellb | iax trunking will just not work |
| 05:05.22 | russellb | but the calls still will |
| 05:05.25 | Juggie | it is for my setup |
| 05:05.35 | Juggie | without ztdummy loaded i get no audio |
| 05:05.50 | Juggie | if i rmmod my driver (wct4xxp) load ztdummy |
| 05:05.51 | Juggie | works fine |
| 05:05.57 | russellb | can't have them both loaded? |
| 05:06.17 | Juggie | well, you can load them, but loading ztdummy after the hardware driver wont work |
| 05:06.33 | Juggie | you have to rmmod your driver, load ztdummy, then load the hardware driver after |
| 05:06.48 | Juggie | however, i dont know if the order will affect the actual hardware working, there are no t1's on it to check atm |
| 05:06.56 | russellb | well, i guess the important issue is, why is your hardware going down? |
| 05:07.10 | Juggie | because there is just a card in there without any t1's. |
| 05:07.23 | Juggie | at this moment, its a test box. |
| 05:07.35 | Juggie | but in a prod environment you woudnt want to loose iax if your t1 went down |
| 05:07.50 | codefreeze | blitzrage: still there? |
| 05:07.56 | blitzrage | codefreeze: yep |
| 05:07.58 | Juggie | you say iax2 doesnt need a timer, however which ever way its configured, i get no audio w/o ztdummy |
| 05:08.24 | russellb | well, it is only used in chan_iax2 for the trunking stuff |
| 05:08.29 | blitzrage | codefreeze: in your CDRfix5 branch... you're creating dialplan functions, and calling them CDRstart(), CDRabort(), etc... right? |
| 05:08.47 | codefreeze | blitzrage: (my turn) yep. |
| 05:09.32 | blitzrage | codefreeze: well.... dialplan functions are always uppercase... so you actually have CDRSTART(), CDRABORT(), etc.... which is kinda hard to read. Is there not a better syntax we can come up with? |
| 05:09.36 | Juggie | russellb, well i can tell you for 100% certain w/o ztdummy loaded, i get NO audio |
| 05:09.43 | Juggie | with trunk=yes/no |
| 05:09.49 | Juggie | as soon as i load it, it works. |
| 05:10.09 | codefreeze | blitzrage: for you, blitzrage, I can call them anything. What do you prefer? |
| 05:10.38 | Juggie | russellb, i suspect iax2 sees the timeing source (wct4xxp) tries to use it, its broken because there is no t1 up.. and then ends up being broken? |
| 05:10.45 | blitzrage | codefreeze: this is what I was trying to think of in my head......... it feels like it should be like.... CDR(start), CDR(abort), etc.... but we already have CDR() |
| 05:10.45 | Juggie | iax2 does USE timeing if its avail right. |
| 05:11.18 | blitzrage | so I'm trying to abstract the concept and come up with a name for this function.... hrmmmmmmmmmmmmm |
| 05:11.33 | russellb | onnnnnnnnly for trunking |
| 05:11.49 | codefreeze | blitzrage: aye. I had the same internal discussion. Let the creative juices flow! |
| 05:11.55 | russellb | i'm too tired to actually debug something right now |
| 05:12.13 | Juggie | russellb, i'm not delirious, with wct4xxp loaded & a card w/ no active trunks i get no audio... unload the driver and load ztdummy and it works. |
| 05:12.26 | russellb | heh, i don't believe you! |
| 05:12.30 | blitzrage | CDR(<action>|<name>[|options]) ? |
| 05:12.34 | blitzrage | does this make any sense? |
| 05:12.39 | russellb | nah, i do, but it doesn't make any sense |
| 05:12.46 | russellb | and i am too tired to figure out why. |
| 05:12.56 | jql | even with no timing source, wouldn't asterisk use the incoming iax stream as the "trigger" for its own audio packets? odd |
| 05:13.11 | Juggie | russellb, well, i assume chan_iax2 has its own internal timeing, unless a timeing source exists. |
| 05:13.13 | codefreeze | blitzrage: what would be the <action> for just setting the CDR values? |
| 05:13.18 | Juggie | for rtp even? |
| 05:13.33 | Qwell | hmm, I recall ManxPower seeing that issue before |
| 05:13.46 | Qwell | something with playback(), ztdummy, and another zap module |
| 05:14.01 | Qwell | Juggie: You might want to ask him about it tomorrow |
| 05:14.05 | Juggie | i should ask file about it, he's done the most w/ iax |
| 05:14.07 | blitzrage | well... basically, explain to me the points of CDRstart(), CDRstop(), CDRabort() -- this is to control the current CDR ont he channel right? So I can do a bunch of stuff (call setup perhaps), and then I can do like..... |
| 05:14.16 | blitzrage | NoOp(CDRSTART()) |
| 05:14.18 | Qwell | it wasn't channel driver specific |
| 05:14.18 | blitzrage | Dial(...) |
| 05:14.19 | blitzrage | ? |
| 05:14.20 | Juggie | i suspect if i unloaded all the zaptel, iax would use its own internal timeing and be happy. |
| 05:14.45 | Juggie | Qwell, i've seen this happen w/ sip and iax. |
| 05:14.57 | Qwell | yeah, ask him tomorrow... I bet he'll remember the fix |
| 05:15.04 | Qwell | iirc, it was an easy fix |
| 05:15.09 | codefreeze | blitzrage: CDRstart actually returns a handle to an allocated, unattached CDR struct. |
| 05:15.31 | Juggie | Qwell, i'm not sure theres a simple 'fix' |
| 05:15.43 | Qwell | if it's the same problem, there is :D |
| 05:15.44 | Juggie | * is simply trying to use a timeing source it thinks exists (loaded module w/ detected hardware) |
| 05:15.48 | codefreeze | blitzrage: it has its start time set to the time you called CDRstart. It is initialized from the current channel. |
| 05:15.55 | Juggie | but since there are no trunks active, there is no timeing. |
| 05:16.04 | Qwell | shouldn't need any active channels |
| 05:16.12 | russellb | Nuitari: blitzrage ... http://clemsonlinux.org/~russell/func_devstate-1.4.tar.gz |
| 05:16.23 | blitzrage | hawt |
| 05:16.26 | codefreeze | blitzrage: CDRanswer takes the handle returned by CDRstart, and sets the answer time to the current time. |
| 05:16.40 | Juggie | Qwell, well, it is broken :) |
| 05:16.58 | tuxd00d | blitzrage: why are you awake? |
| 05:17.00 | Qwell | is the card configured? |
| 05:17.04 | tuxd00d | Qwell: Hey |
| 05:17.08 | Qwell | tuxd00d: y0 |
| 05:17.13 | codefreeze | blitzrage: CDRclose takes that handle also, and sets the end time, and posts the CDR. |
| 05:17.17 | Juggie | Qwell, no, atm its just two unconfigured spans. |
| 05:17.19 | blitzrage | tuxd00d: yes... just talking to my not-not g/f :) |
| 05:17.22 | Qwell | Juggie: that might be it |
| 05:17.28 | Nuitari | russellb: Thanks!!! |
| 05:17.38 | russellb | Nuitari: you're welcome |
| 05:17.38 | tuxd00d | blitzrage: I'm confused |
| 05:17.40 | blitzrage | codefreeze: ok... that makes sense |
| 05:17.42 | Juggie | Qwell, i guess thats possible if i configured them (even if they were down) |
| 05:17.45 | blitzrage | tuxd00d: it's complicated ;) |
| 05:17.47 | russellb | I didn't test it ... just made it build ... which was a lot harder than i thought it would be |
| 05:17.49 | Qwell | Juggie: yeah, try that :D |
| 05:18.00 | codefreeze | blitzrage: the CDR on the channel, is not touched or affected by the 3 funcs previously mentioned. |
| 05:18.03 | Nuitari | russellb: I'll put it on the server that I'm upgrading to 1.4.5 |
| 05:18.05 | Juggie | Qwell, will do. |
| 05:18.08 | blitzrage | codefreeze: oh really... |
| 05:18.08 | Nuitari | and if it works :) |
| 05:18.16 | blitzrage | codefreeze: so this is a separate CDR then |
| 05:18.20 | blitzrage | so I could do |
| 05:18.23 | blitzrage | NoCDR() |
| 05:18.25 | blitzrage | CDRstart() |
| 05:18.30 | blitzrage | CDRanswer() |
| 05:18.33 | blitzrage | CDRstop() |
| 05:18.37 | blitzrage | and control my own CDRs? |
| 05:18.38 | russellb | Nuitari: you can just tell people how cool of a developer i am :-p |
| 05:18.45 | russellb | Nuitari: and ... buy digium stuff. |
| 05:19.04 | Nuitari | russellb: Will do |
| 05:19.27 | blitzrage | (theoretically) |
| 05:19.37 | codefreeze | blitzrage: totally. I modded the CDR func to take a handle as the 3rd arg. If you supply one, it allows you set almost any CDR field. Otherwise, the same ol' restrictions apply to the channel-attached CDR(s). |
| 05:19.54 | blitzrage | codefreeze: ok, so that makes sense to me then |
| 05:20.01 | blitzrage | (that's hawt btw) |
| 05:20.06 | blitzrage | so I could control my own logic |
| 05:20.15 | blitzrage | and determine if I wanted a CDR or not, based on a condition |
| 05:20.25 | blitzrage | (is this an Internal call, or external call) |
| 05:21.04 | blitzrage | so....... |
| 05:21.08 | codefreeze | blitzrage: yes, you could do that. |
| 05:21.23 | blitzrage | now back to how to name this sucker |
| 05:21.24 | russellb | SUCKER() ? |
| 05:21.27 | blitzrage | :) |
| 05:21.36 | blitzrage | CDR_CONTROL(start) |
| 05:21.56 | blitzrage | no, that is wrong |
| 05:22.05 | blitzrage | Set(CDR_CONTROL()=start) |
| 05:22.11 | blitzrage | that seems not ideal.... |
| 05:22.26 | codefreeze | blitzrage: I was really tempted to call it CDR_ALLOCATE() |
| 05:22.44 | blitzrage | that almost seems like a good name for CDRstart() |
| 05:22.46 | Nuitari | http://nuitari.org/custom_devstate.php.gz |
| 05:23.03 | jql | gah, those prefix names get to me. NEW_CDR() :) |
| 05:23.05 | blitzrage | CDR_ALLOCATE(), CDR_ANSWER(), CDR_CLOSE() ? |
| 05:23.13 | russellb | Nuitari: 404! |
| 05:23.16 | blitzrage | seems like the same problem.... |
| 05:23.17 | jql | namespaces are for the organized |
| 05:23.28 | Nuitari | http://nuitari.org/asterisk/custom_devstate.php.gz |
| 05:23.29 | blitzrage | really feels like it should be a single function |
| 05:23.41 | Nuitari | that works |
| 05:23.58 | blitzrage | would you ever really assign a value to this though........ |
| 05:24.18 | codefreeze | blitzrage: I was tempted to do that; I just hate underscores... but I'm not overly in hate with them. I could do it so. |
| 05:24.26 | Nuitari | I know it's a php program, but I haven't used C/C++ in way too long |
| 05:24.32 | blitzrage | I don't really like it either.... |
| 05:24.59 | Juggie | russellb/blitzrage, i added a fake span definition to zaptel.conf so there would be a configured span (even if the pri physically does not exist) and then everything works |
| 05:24.59 | blitzrage | Juggie: can you send me how you did that? |
| 05:25.02 | blitzrage | just a sample config snippet probably |
| 05:25.06 | Juggie | so it seems if you have a t1 card you must configure a span on it, even if you dont have one, or the timeing goes haywire. |
| 05:25.38 | codefreeze | blitzrage: other func names have _ in them, so it's pretty standard practice... |
| 05:25.43 | blitzrage | ya... |
| 05:25.45 | Juggie | blitzrage, my /etc/zaptel.conf is 3 lines |
| 05:25.45 | Juggie | span=1,0,0,esf,b8zs |
| 05:25.45 | Juggie | bchan=1-23 |
| 05:25.45 | Juggie | dchan=24 |
| 05:25.47 | Juggie | that is all |
| 05:25.52 | Juggie | and that fixed the lack of timeing |
| 05:26.00 | Juggie | (after a ztcfg of course) |
| 05:26.41 | russellb | Nuitari: lol @ the comment at the top |
| 05:26.58 | Nuitari | :) |
| 05:27.18 | Juggie | russellb, so seems simple enough to overcome still kinda lame though, that dropping in a t1 board and restarting your box could cause problems w/ asterisk until you configure a span on the board. |
| 05:27.45 | russellb | Nuitari: well cool, thanks for sharing. i have it saved so that I can try to understand it when i'm more awake :) |
| 05:27.50 | blitzrage | Exec(${IF($[${CALL_TYPE} = EXTERNAL]?Set(CDR_CONTROL(start)):NoOp())}) |
| 05:28.08 | *** join/#asterisk mindCrime_ (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 05:28.12 | Juggie | russellb, i got approved for *-con again... |
| 05:28.14 | blitzrage | codefreeze: I think I like the CDR_CONTROL(start|answer|abort|stop) format.... |
| 05:28.21 | Juggie | visit #3 for free :) |
| 05:28.25 | russellb | nice |
| 05:28.27 | russellb | same here, hehe |
| 05:28.30 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 05:28.46 | russellb | actually, it will be #4, heh |
| 05:28.46 | blitzrage | codefreeze: that kind of dialplan line I wrote is very common in my dialplan |
| 05:28.56 | russellb | first time was on adtran ... second two were digium.. |
| 05:29.03 | Juggie | russellb, mine is care of the canadian tax payer :) |
| 05:29.15 | russellb | heh, nice |
| 05:29.17 | codefreeze | blitzrage: except, that CDRstart returns a handle. You can create several simultaneously. You address them individually. |
| 05:29.28 | Juggie | when blitzrage files his taxes, he pays for my trips hehe |
| 05:29.34 | Juggie | btw, i should flag him for an audit. |
| 05:30.00 | Nuitari | russellb: you know how to reach me |
| 05:30.04 | blitzrage | codefreeze: hrmmm.... ok, I gotta read your blog post so I understand better how you are utilizing those functions |
| 05:30.08 | russellb | Nuitari: indeed |
| 05:30.24 | Qwell | hmm, I wonder if they already have enough speakers |
| 05:31.29 | blitzrage | codefreeze: aha -- I was using it wrong |
| 05:31.34 | codefreeze | blitzrage: forkCDR() allowed you to "stack" multiple CDRs in a channel. Have to allow multiples, or you lose functionality. My version is IMNASHO, is superior, methinks/mehopes. |
| 05:31.40 | russellb | Qwell: don't think there has even been a call for speakers yet ... |
| 05:31.45 | russellb | not that i have seen, anyway |
| 05:31.55 | Qwell | blitzrage: ^^? |
| 05:31.56 | russellb | i need to figure out what i want to talk about this go around ... |
| 05:32.17 | blitzrage | codefreeze: Exec(${IF($[${CALL_TYPE} = EXTERNAL]?Set(mycdr=${CDR_CONTROL(start)}):NoOp())}) |
| 05:32.38 | Qwell | I guess I should run it by Kevin first, heh |
| 05:32.58 | Juggie | codefreeze, you should see the CDR records i work with.. we get like up to 5million seperate records a day from our telco, we have a direct feed for our 1-800 network. |
| 05:33.07 | blitzrage | Qwell: ya... sounds like I gotta do a talk :) |
| 05:33.26 | codefreeze | blitzrage: OK, let's see, CDRanswer returns the same handle it got; but CDRclose doesn't, as that handle is dead. |
| 05:33.28 | Qwell | blitzrage: they already accepting speaker requests? |
| 05:33.37 | Qwell | or, stopped accepting for that matter |
| 05:34.04 | blitzrage | NoOp(CDR answer says: ${CDR_CONTROL(answer|${my_cdr})}) |
| 05:34.28 | russellb | Qwell: ah, there is a link for speaker stuff on the web site |
| 05:34.38 | russellb | if i don't get to speak i will be sad :( |
| 05:34.38 | Qwell | ahh |
| 05:34.51 | Qwell | I've never done a topic at a tradeshow |
| 05:34.54 | Qwell | well, sorta |
| 05:34.58 | blitzrage | Set(nothing=${CDR_CONTROL(end|${my_cdr})}) |
| 05:35.04 | Qwell | not voluntarily, and with notice :P |
| 05:35.07 | blitzrage | codefreeze: that's how I would picture those working |
| 05:35.15 | blitzrage | based on your examples in AEL |
| 05:35.19 | Qwell | well, more than an hours notice, heh |
| 05:35.34 | Qwell | David Rowe kinda ambushed me in Dallas :D |
| 05:36.11 | blitzrage | I could do something on clustering |
| 05:36.33 | Qwell | I could give a talk on Skinny, and have like 3 people show up |
| 05:36.58 | blitzrage | you'd probably be surprised... :) |
| 05:37.03 | Qwell | unlikely |
| 05:37.05 | Juggie | someone needs to do a talk on snmp and monitoring |
| 05:37.15 | Juggie | that would be intreasting |
| 05:37.21 | blitzrage | I wish I had time to work on that |
| 05:37.23 | Juggie | is there still that open snmp mib ticket on the tracker |
| 05:37.29 | Juggie | someone was fixing it up |
| 05:37.30 | blitzrage | I did a bunch of SNMP stuff for school |
| 05:37.35 | Qwell | jeffg I think |
| 05:37.36 | codefreeze | blitzrage: I even played with the idea of making CDRanswer and CDRclose apps instead of funcs, as really, they don't need to return anything. |
| 05:37.48 | Juggie | blitzrage, nic has asterisk monitoring setup via cacti |
| 05:38.08 | Juggie | and then he has his own realtime portal written for like a overview of all servers. and just uses cacti for historical |
| 05:38.08 | blitzrage | codefreeze: ya, but you want to be able to make them functions so you can easier embed them in dialplan logic |
| 05:38.20 | blitzrage | Juggie: nice |
| 05:38.25 | blitzrage | I need something like that |
| 05:39.25 | blitzrage | anyways... I gotta go to bed |
| 05:39.26 | Juggie | blitzrage, so far as i know he just use php snmp to hit all our boxes and then display a lil screen w/ stats, active calls, etc. |
| 05:39.32 | blitzrage | need to catch a train at 7:50am downtown |
| 05:39.41 | blitzrage | Juggie: nice! |
| 05:39.43 | codefreeze | blitzrage: evaluation with side effects embedded in expressions... the world is scheduled to end tomorrow! |
| 05:39.47 | rob0 | Yikes, those things are fast and big. Good luck! |
| 05:40.09 | blitzrage | codefreeze: it's too late for me... you've lost me :) |
| 05:40.42 | codefreeze | blitzrage: sweet dreams; we can pick this up later. |
| 05:40.53 | blitzrage | codefreeze: cool, will do! |
| 05:41.03 | blitzrage | russellb: no hung channels thus far :) |
| 05:41.13 | blitzrage | night all! |
| 05:41.20 | russellb | blitzrage: nice |
| 05:41.21 | russellb | g'night |
| 05:42.28 | Juggie | ah, his changes got merged into trunk for snmp, nice. |
| 05:44.05 | Qwell | bed time |
| 05:50.18 | Nuitari | codefreeze: need a repeater? |
| 05:50.32 | russellb | 130 feet should be fine by far |
| 05:50.38 | Nuitari | ah feet |
| 05:50.42 | Nuitari | then yeah should be fine |
| 05:51.23 | codefreeze | thought ethernet was good for 100 meters, around 300 feet. Very frustrating. |
| 05:51.35 | russellb | codefreeze: yep, supposed to be ... |
| 05:52.32 | codefreeze | the linksys router 4-port switch blinks on/off slowly, the hubs, 3 diff. kinds, blink likewise. What is this, some kind of standards thing? |
| 05:55.10 | Nuitari | wtf |
| 05:55.11 | Nuitari | asterisk: symbol lookup error: /usr/lib/asterisk/modules/cdr_addon_mysql.so: undefined symbol: __dont_use_localtime_r_use_ast_localtime_instead__ |
| 05:55.59 | Nuitari | cdr_addon_mysql.c:132: error: too few arguments to function 'ast_localtime' |
| 05:57.02 | Nuitari | what should be in const char *zone ? |
| 05:58.17 | Nuitari | ok I changed it, passed a null and it's working at least |
| 05:58.25 | jql | that's quite a wtf |
| 05:58.26 | *** join/#asterisk Cabal_ (n=Cabal@unaffiliated/cabal) |
| 05:58.27 | Nuitari | though probably not a clean wway |
| 05:58.39 | Nuitari | so asterisk-addons 1.4.1 will crash asterisk-1.4.5 |
| 06:03.10 | *** join/#asterisk lucidsmog (n=lucidsmo@pool-71-166-71-203.bltmmd.east.verizon.net) |
| 06:04.10 | Nuitari | is Internal RTCP NTP clock skew detected: lsr=4197037742, now=4197037931, dlsr=65500 (0:999ms), diff=65311 something to be worried about ? |
| 06:06.19 | lucidsmog | So out of curiosity, what does asterisk do with a SIP NOTIFY message from a SIP proxy when Asterisk is registered to said SIP proxy as a UA? (Specifically, I'm curious about what happens to a Message Waiting NOTIFY) |
| 06:10.42 | *** join/#asterisk andrew` (n=andrew@69-12-140-101.dsl.dynamic.sonic.net) |
| 06:18.19 | *** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com) |
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| 06:45.53 | *** part/#asterisk kiscokid (n=Ron@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
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| 06:49.18 | *** join/#asterisk bintut (n=bintut@cm145.gamma178.maxonline.com.sg) |
| 06:54.23 | bintut | hello all.. |
| 06:55.05 | bintut | is there a major difference with the configuration files inside the /etc/asterisk/ directory between 1.2.13 and 1.4.4 ? |
| 06:55.32 | bintut | can i make use of my existing asterisk-1.2.13 configs to a new asterisk-1.4.4 setup? |
| 06:55.51 | Nuitari | probably not |
| 06:56.29 | Nuitari | well |
| 06:56.36 | Nuitari | it might work, it depends on your setup |
| 06:56.49 | Nuitari | are you using gentoo? |
| 06:57.04 | bintut | Nuitari: this is my personal home setup with a digium devkit 1fxo and 1fxs |
| 06:57.14 | bintut | Nuitari: debian etch i386 |
| 06:58.09 | Nuitari | do you use priorityjumping? |
| 06:58.16 | bintut | nope |
| 06:58.28 | bintut | very simple setup.. i'm just a newbie |
| 06:58.45 | Nuitari | shouldn't be that hard then |
| 06:58.47 | bintut | i still don't even have any logic on my configs.. :( |
| 06:58.50 | Nuitari | keep backups just in case |
| 06:58.58 | bintut | yeah, i have backup |
| 06:59.25 | bintut | actually, i'm having a problem with echo |
| 06:59.53 | *** join/#asterisk Daniel_Tech (n=danielgo@dsl-220-253-74-163.NSW.netspace.net.au) |
| 07:00.01 | bintut | i don't know.. i can't figure out already. i followed the FAQs of the digium's kb but the echo still remains.. ( |
| 07:03.44 | Nuitari | I just use echocancel = yes |
| 07:03.48 | Nuitari | and have it autotrain |
| 07:07.22 | bintut | Nuitari: and it's fixed already? |
| 07:07.53 | bintut | Nuitari: mine still exist.. |
| 07:12.52 | Nuitari | yeah it's fixed for me |
| 07:13.47 | Daniel_Tech | can any1 give me a clue as to how i would go about choosing the right zaptel card for a network? i never used them before |
| 07:21.30 | *** join/#asterisk steliosk (n=Stelios@62.169.217.209) |
| 07:25.01 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 07:27.44 | tzafrir_laptop | bintut, set opermode? used fxotune? |
| 07:28.34 | bintut | tzafrir_laptop: hello.. what's an opermode? not yet with fxotune.. kinda afraid. :( |
| 07:29.20 | tzafrir_laptop | from which country are you? |
| 07:30.05 | tzafrir_laptop | anyway, try using fxotune. Should help you much with the FXO echo |
| 07:32.36 | aksnowman | I've just gotten an asterisk box set up yesterday (softphones only) and can connect from my home network, but nobody outside of my network can actually stay connected. I've got the asterisk box set up as my dmz so it isn't an access problem. In the console I can see them connecting and whatnot, but then it just says "user hung up". They never even hear the automated messages. Any help would be appriciated |
| 07:35.14 | *** join/#asterisk kiscokid (n=Ron@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 07:47.29 | *** part/#asterisk kiscokid (n=Ron@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 07:49.51 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 07:54.13 | *** join/#asterisk floppp (n=flop@nat-staff.b3g-telecom.com) |
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| 08:30.25 | *** join/#asterisk Shaun2222 (n=shaun@ip68-4-212-221.oc.oc.cox.net) |
| 08:30.35 | Shaun2222 | anybody use the wip330-na phone by linksys? |
| 08:35.19 | bintut | gtg now.. |
| 08:35.20 | bintut | thanks.. |
| 08:43.06 | *** join/#asterisk Cabal (n=Cabal@unaffiliated/cabal) [NETSPLIT VICTIM] |
| 08:43.09 | *** join/#asterisk Cabal (n=Cabal@unaffiliated/cabal) [NETSPLIT VICTIM] |
| 08:49.15 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 08:55.01 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 08:59.04 | *** join/#asterisk A-Data (n=asd@196.218.74.235) |
| 09:00.31 | A-Data | Hello all i put exten => 6666,n,AGI(agi-sayani.agi) in extensions.conf in [context] this is perl AGI when i do reload and try to call the extension it give me the person you have called is unavialable |
| 09:04.30 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00111ae4684c.cpe.net.cable.rogers.com) |
| 09:05.30 | A-Data | any one have suggestion about this proplem? |
| 09:06.01 | tsurko | what is supposed the agi ot do? |
| 09:06.03 | tsurko | *to |
| 09:06.17 | A-Data | tsurko it supposed to say caller ID |
| 09:06.46 | *** join/#asterisk hijacked (i=mSVJ@cerebus.clandestineresearch.com) |
| 09:07.00 | A-Data | but i have tried other scripts to from the perl example all give beep beep beep the person you have called is unavialbel |
| 09:11.40 | A-Data | any help :< |
| 09:14.06 | A-Data | ~agi |
| 09:14.07 | jbot | it has been said that agi is the Asterisk Gateway Interface... similar to CGI for web applications AGI lets you script call control and access databases using your favorite language. AGI wrappers are available for Python (pyst), Perl (astperl?) and other languages |
| 09:15.41 | *** join/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
| 09:20.58 | *** join/#asterisk steliosk (n=Stelios@62.169.217.209) |
| 09:22.29 | *** join/#asterisk EvilGreen_ (n=Miranda@ppp85-140-136-78.pppoe.mtu-net.ru) |
| 09:23.24 | A-Data | any one with AGI experince can help me |
| 09:29.04 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:35.08 | *** join/#asterisk Sargun (n=Sargun@atarack/staff/sargun) |
| 09:35.17 | Sargun | What's the point of a zaptel channel? |
| 09:38.19 | deegan | From my experience and knowledge it's for ISDN connections, you get inbound/outbound calls on a isdn then you need zaptel. |
| 09:42.48 | A-Data | but i have tried other scripts to from the perl example all give beep beep beep the person you have called is unavialbel |
| 09:42.55 | A-Data | Hello all i put exten => 6666,n,AGI(agi-sayani.agi) in extensions.conf in [context] this is perl AGI when i do reload and try to call the extension it give me the person you have called is unavialable |
| 09:48.05 | *** join/#asterisk denke (n=denke@mehess.adsl.datanet.hu) |
| 09:55.49 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.178.29) |
| 10:02.52 | Sargun | deegan, Can I make virtual zaptel channels |
| 10:02.56 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 10:03.02 | Sargun | Like put them SIP calls over them |
| 10:03.09 | Sargun | I mean, I know meetme somehow does it |
| 10:10.24 | aksnowman | I've just gotten an asterisk box set up yesterday (softphones only) and can connect from my home network, but nobody outside of my network can actually stay connected. I've got the asterisk box set up as my dmz so it isn't an access problem. In the console I can see them connecting and whatnot, but then it just says "user hung up". They never even hear the automated messages. Any help would be appriciated |
| 10:14.42 | Daniel_Tech | how do i know what zaptel card i should use???? |
| 10:17.25 | *** join/#asterisk Marshall- (n=Marshall@cpe-76-181-122-108.columbus.res.rr.com) |
| 10:25.06 | *** join/#asterisk coppice (n=chatzill@10.198.17.210.dyn.pacific.net.hk) |
| 10:27.22 | *** join/#asterisk saftsack (n=saftsack@pD9E04368.dip0.t-ipconnect.de) |
| 10:27.23 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 10:39.36 | *** join/#asterisk stoffell_h (n=stoffell@d5153FA28.access.telenet.be) |
| 10:40.19 | stoffell | Daniel_Tech, what are you trying to connect to? |
| 10:44.59 | *** join/#asterisk friedrich| (n=friedric@e177248113.adsl.alicedsl.de) |
| 10:56.03 | Daniel_Tech | well the system i am using is Trixbox. I normally have the internet connectionc moming into the network via modem/router and then into a switch. i then use the NIC onthe trixbox server to make/take calls. i want to do a simialr thing except have a PSTN failover |
| 10:56.08 | Daniel_Tech | if that helps |
| 10:57.12 | stoffell_h | if you want to connect PSTN, you should check out the analogue cards of digium (TDMxxx).. |
| 10:57.59 | Daniel_Tech | thanks. one last question, if im setting up the PBX inthe above way, will i need to have more than 1 FXO port?? |
| 10:58.21 | Daniel_Tech | all the IP phones plug into wall ports connected to the main switch |
| 10:58.23 | DEac- | i think about to try kiax as iax-'client'. my question is: i must type IAX2/User:Password@Host/Ext or it's enough, if i type IAX2/Host ? |
| 10:59.09 | DEac- | ah, i see a mistage. Host is dynamic, so i must use the Extension |
| 10:59.16 | stoffell_h | Daniel_Tech, if you will have only 1 pstn (1 analogue line), only 1 fxo port is needed.. |
| 10:59.40 | Daniel_Tech | stofell_h, thankyou for the help much appreciated |
| 11:01.22 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 11:02.50 | *** join/#asterisk toot (n=toot@84.19.255.123) |
| 11:03.06 | toot | hey folks, as a european snom user - what polycom phones do people recommend? :) |
| 11:03.20 | toot | want to get a couple and test auto provisioning |
| 11:06.11 | stoffell_h | toot, the ip430 is cool, and the 650 should be too (but it's more expensive) |
| 11:14.19 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 11:16.17 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 11:16.19 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 11:37.01 | toot | thanks stoffell_h - bought a couple of the ip430 :) |
| 12:35.21 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 12:54.11 | *** join/#asterisk Gajol (n=mb@port143.ds1-gr.adsl.cybercity.dk) |
| 12:54.14 | Gajol | Hey |
| 12:55.36 | Gajol | i am in need of some help. i have set up a new asterisk server. but have som problems with getting my calls in. i have no problems calling out. it worked fine when i testet it on trixbox. hope that you can help me |
| 12:56.26 | Teccy | is there anyway to initiate dialling on a zap interface from the console? |
| 13:06.31 | stoffell_h | Teccy, not that I know, but you can use a .call file.. |
| 13:07.26 | Teccy | hmm, fair enough, thanks |
| 13:09.49 | Teccy | its certainly a feature that would be useful for testing/setup/debugging |
| 13:15.12 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 13:24.31 | DarKnesS_WolF | wb [TK]D-Fender :-) |
| 13:25.30 | DarKnesS_WolF | Teccy: yes u can using the dial command but u need soundcard / headphones and OSS or alsa installed on that server |
| 13:27.15 | SavageOne | morning folks |
| 13:28.45 | SavageOne | here's my question: I've heard tales warning against having more then 1 phone attached to the same port on an analog adapter, be it an internal fxs port on a pxi card or an external ata like a linksys pap2t or something like that, something about the voltage being drawn from the plain old analog headset having the potention to fry the port or something along those lines. How much truth is there to this? |
| 13:32.46 | [TK]D-Fender | SavageOne, RINGING on and FXO port can fry an FXS port. |
| 13:32.51 | [TK]D-Fender | an* |
| 13:32.56 | SavageOne | k |
| 13:33.00 | SavageOne | but where is the ring generated? |
| 13:33.14 | SavageOne | I'm talking about internal phones not fxo for external, because obviously only one could be connected that way |
| 13:33.15 | [TK]D-Fender | SavageOne, Because your FXS port is not designed to ACCEPT voltage, it tried to put its own. |
| 13:33.18 | *** join/#asterisk _DAW (n=chatzill@adsl-241-93-3.msy.bellsouth.net) |
| 13:33.29 | SavageOne | but does the fxs pass the voltage for the ring from the line |
| 13:33.33 | SavageOne | wait, I see what you mean |
| 13:33.41 | [TK]D-Fender | SavageOne, how else do you think and line would RING? Ever used a phone before? :) |
| 13:34.01 | SavageOne | so asterisk sends the call to the fxs port via zaptel, and then the hardware on the card generates the ring tone voltage and the phone icks and up and ringsp |
| 13:34.44 | SavageOne | so, if there's too many phones, it tries to pull too much voltage from the card and that could fry it right? |
| 13:34.59 | [TK]D-Fender | Still getting it backwards. |
| 13:35.05 | SavageOne | I only ask because I have a dead fxo module on my digium card and everyone is saying it's because I had 2 phones on it |
| 13:35.13 | Teccy | SavageOne: if you're using modern phones, especially ones powered from the mains, like dect phones, they have exteremly low RENs |
| 13:35.26 | Teccy | if it's an fxo module, why did you have phones connected to it? |
| 13:35.32 | SavageOne | these were like 25$ walmart uniden cordless phones |
| 13:35.39 | SavageOne | fxs I mean |
| 13:35.42 | SavageOne | did I say fxo? my bad |
| 13:35.52 | SavageOne | fxs fxs hehe |
| 13:36.16 | [TK]D-Fender | if you plug an FXS (which puts out voltage), to your home PHONE LINE (which does the same), and one end decides to try and ring the other, then you are LOADING the cirecuit with too much power. |
| 13:36.48 | SavageOne | that's not what I mean |
| 13:36.53 | [TK]D-Fender | GTG, back in several hours... |
| 13:36.56 | SavageOne | I'm talking about a regular old asterisk phone system |
| 13:36.59 | SavageOne | not in a home |
| 13:37.15 | SavageOne | just a regular phone system with a digium card in it w/ an fxo port so I can have a single analog extension |
| 13:37.29 | SavageOne | but then I split it to 2 cordless phones via a regular old splitter |
| 13:37.38 | [TK]D-Fender | SavageOne, the only bad thing you can do is plug 2 ports that PUT OUT VOLTAGE INTO EACH OTHER! |
| 13:37.41 | SavageOne | and people are telling me that since I did that I fried the port by putting too much load on it |
| 13:37.45 | [TK]D-Fender | Splitting = fine! |
| 13:37.47 | SavageOne | k |
| 13:37.52 | SavageOne | just verifying |
| 13:38.04 | [TK]D-Fender | what'll happen with overloading REN is your phones won't ring |
| 13:38.20 | SavageOne | so when the phone rings, like on a normal pots line, it's not sending a sound or anything just a certain type of electricity that signals a phone to ring |
| 13:38.21 | [TK]D-Fender | and I'm not sure about any further degradation.... |
| 13:38.29 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:38.29 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:38.36 | [TK]D-Fender | SavageOne, Correct |
| 13:38.46 | SavageOne | so this dude is a retard that told me this |
| 13:38.48 | SavageOne | I had a feeling |
| 13:39.01 | [TK]D-Fender | Ringing is a higher voltage **AC** signal whereas talking on the line is **DC** |
| 13:39.10 | SavageOne | I've got a few homes also setup w/ papt2s w/ teliax accounts who want an alternative to vonage |
| 13:39.17 | *** join/#asterisk _VoiceMeUp_COM (n=_VoiceMe@modemcable159.131-56-74.mc.videotron.ca) |
| 13:39.21 | SavageOne | it's ac? no shit |
| 13:39.25 | [TK]D-Fender | ok, gotta go, best of luck to all. |
| 13:39.25 | _VoiceMeUp_COM | Qwell[] u around ? |
| 13:39.30 | [TK]D-Fender | bbiab |
| 13:39.34 | SavageOne | that's probably why I get a jolt when I'm punching a panel down and a goddamn call comes in rofl |
| 13:39.53 | _VoiceMeUp_COM | wondering if chan_mobile is ok on .18 |
| 13:40.07 | _VoiceMeUp_COM | coz for some reason 1.4 doesnt wor k for me.. audio issues |
| 13:40.18 | _VoiceMeUp_COM | guess i should say .19 now |
| 13:40.23 | Math` | talking on the line is dc? audio on dc doesnt makes sense unless you modulate it, which is not the case with pots |
| 13:42.31 | Corydon76-home | Math`: correct, POTS uses DC current |
| 13:42.44 | Corydon76-home | except for ring voltage, which is AC |
| 13:43.38 | Corydon76-home | It's the only circuit that I know of that uses both DC and AC on the same pair of wires |
| 13:45.24 | _VoiceMeUp_COM | the chan_mobule is on trunk of addons ? or in main |
| 13:46.47 | Corydon76-home | addons |
| 13:47.03 | _VoiceMeUp_COM | yeah i see |
| 13:47.04 | _VoiceMeUp_COM | hmm |
| 13:47.17 | _VoiceMeUp_COM | not in 1.2.6 so its trunk ? |
| 13:49.08 | Corydon76-home | Repeat the mantra: we do not add features to code that has already been released. |
| 13:49.15 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 13:49.22 | _VoiceMeUp_COM | taking to me ? |
| 13:49.29 | *** part/#asterisk _VoiceMeUp_COM (n=_VoiceMe@modemcable159.131-56-74.mc.videotron.ca) |
| 13:49.34 | *** join/#asterisk _VoiceMeUp_COM (n=_VoiceMe@modemcable159.131-56-74.mc.videotron.ca) |
| 13:52.17 | _VoiceMeUp_COM | svn co http://svn.digium.com/asterisk-addons/trunk/ asterisk-addons |
| 13:52.19 | _VoiceMeUp_COM | right ? |
| 13:52.44 | _VoiceMeUp_COM | svn: PROPFIND of '/asterisk-addons/trunk': 405 Method Not Allowed (http://svn.digium.com) |
| 13:56.11 | _VoiceMeUp_COM | yo |
| 13:56.17 | _VoiceMeUp_COM | why a 405 |
| 13:57.34 | tzanger | uh, 405 is resource not allowed, failing a PROPFIND sounds about right |
| 13:57.55 | _VoiceMeUp_COM | ok so im not allowed to get addons trunk ? |
| 13:58.04 | tzanger | no that'd be something else I think |
| 13:58.26 | _VoiceMeUp_COM | ok so how we resolve.. im trying to get the trunk for chan_mobile |
| 13:59.11 | *** join/#asterisk saftsack (n=saftsack@pD9E04368.dip0.t-ipconnect.de) |
| 13:59.21 | russellb | you're missing "svn" before ast-addons |
| 13:59.21 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 13:59.36 | _VoiceMeUp_COM | ah |
| 13:59.46 | _VoiceMeUp_COM | well |
| 13:59.47 | _VoiceMeUp_COM | http://forums.digium.com/viewtopic.php?p=51688&sid=fac5a8195a0173c82b48841cdf56c708 |
| 13:59.48 | tzanger | russelb gets the cookie |
| 13:59.51 | russellb | look at all of the examples all over the internet :-p |
| 13:59.55 | _VoiceMeUp_COM | i guess this mofo is wrong then |
| 14:00.06 | russellb | except the wrong info, of course |
| 14:00.08 | _VoiceMeUp_COM | that where i got the problem in first place lol |
| 14:00.34 | _VoiceMeUp_COM | ok so svn co http://svn.digium.com/svn/asterisk-addons/trunk/ asterisk-addons |
| 14:01.44 | _VoiceMeUp_COM | and i think 1.4 is messed. in terms of compat with 1.2 |
| 14:02.04 | _VoiceMeUp_COM | needs to have all nat=no or else all fails.. but same config on 1.2 works like a charm |
| 14:02.25 | *** join/#asterisk Math` (n=privmath@modemcable037.229-56-74.mc.videotron.ca) |
| 14:08.46 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 14:19.12 | *** join/#asterisk Cyber-Dogg (i=Cyber-Do@24-178-240-97.dhcp.stls.mo.charter.com) |
| 14:27.59 | DarKnesS_WolF | hmm there is no more app_japper.so in asterisk 1.4 ? only chan_gtalk ? |
| 14:28.50 | DarKnesS_WolF | ah it's res_japper |
| 14:29.06 | Qwell | what is japper? |
| 14:29.32 | _VoiceMeUp_COM | japper no idea |
| 14:29.57 | _VoiceMeUp_COM | qwell you approves chan_mobile.. can thi work asterisk -addons trunk with 1.2.19 ? or need 1.4.5 |
| 14:30.06 | Qwell | trunk |
| 14:30.22 | _VoiceMeUp_COM | astersk trunk ? |
| 14:30.23 | _VoiceMeUp_COM | k |
| 14:31.30 | DarKnesS_WolF | Qwell: google talk |
| 14:31.35 | DarKnesS_WolF | hmm chan_mobile :D? |
| 14:31.40 | _VoiceMeUp_COM | yeah |
| 14:31.43 | _VoiceMeUp_COM | trying it out |
| 14:31.45 | DarKnesS_WolF | what is that for ? |
| 14:31.48 | DarKnesS_WolF | bluetooth ? |
| 14:31.51 | _VoiceMeUp_COM | got 10 ericson t68i's |
| 14:31.53 | Qwell | _VoiceMeUp_COM: and, you missed an "svn" in your URL |
| 14:31.55 | Qwell | DarKnesS_WolF: yep |
| 14:31.59 | _VoiceMeUp_COM | but they not supported ;( so im tyring headpeace |
| 14:32.03 | _VoiceMeUp_COM | yeah i know |
| 14:32.10 | DarKnesS_WolF | Qwell: i failed with other bluetooth chanels :-s |
| 14:32.12 | Qwell | pretty much any cell should work with it |
| 14:32.16 | DarKnesS_WolF | so i gave up :-) |
| 14:32.18 | _VoiceMeUp_COM | the 9:59 line has it |
| 14:32.18 | Qwell | DarKnesS_WolF: the other bluetooth channels suck :p |
| 14:32.25 | DarKnesS_WolF | Qwell: oh ya :P |
| 14:32.30 | _VoiceMeUp_COM | well i connect to headset but not cell |
| 14:32.34 | _VoiceMeUp_COM | same pass everything |
| 14:32.36 | _VoiceMeUp_COM | ill see |
| 14:32.44 | DarKnesS_WolF | Qwell: it was supporting some kind of nokia model and motorola only or something like that ... |
| 14:33.01 | DarKnesS_WolF | _VoiceMeUp_COM: keep me updated please :-) |
| 14:33.05 | _VoiceMeUp_COM | david asnwered me |
| 14:33.20 | DarKnesS_WolF | works ? |
| 14:33.24 | _VoiceMeUp_COM | let me pastebin |
| 14:33.44 | DarKnesS_WolF | _VoiceMeUp_COM: and a small fast howto to for voip-info :-) |
| 14:33.52 | _VoiceMeUp_COM | http://pastebin.ca/569989 |
| 14:34.10 | _VoiceMeUp_COM | well headpiece doesnt have rtp with 1.4.4 and chan_cellphone |
| 14:34.20 | _VoiceMeUp_COM | so im redoing it with trunks and chan_mobile |
| 14:34.47 | _VoiceMeUp_COM | basically exten => 1004,2,Dial(CELL/cellphone/15144322343) |
| 14:34.55 | _VoiceMeUp_COM | would bridge from cell |
| 14:34.58 | DarKnesS_WolF | Qwell: works with any bluetook dongol ? or need speciak bluetooth device? |
| 14:35.05 | _VoiceMeUp_COM | any dongle |
| 14:35.05 | Qwell | DarKnesS_WolF: any with linux drivers |
| 14:35.15 | _VoiceMeUp_COM | but needs bluez and libbluetooth |
| 14:35.16 | DarKnesS_WolF | Qwell: i have very cheap one works with linux perfectly |
| 14:35.22 | Qwell | then it'll work fine |
| 14:35.24 | *** join/#asterisk wunderkin (i=wunderki@ip68-104-149-97.ph.ph.cox.net) |
| 14:35.29 | DarKnesS_WolF | hmm may be will start to compile asterisk and try again on my laptop |
| 14:35.35 | _VoiceMeUp_COM | yeah |
| 14:35.36 | _VoiceMeUp_COM | lol |
| 14:35.38 | DarKnesS_WolF | Qwell: chan__mobile or chan_cellphone ? |
| 14:35.46 | _VoiceMeUp_COM | actualy the dongle gives me bluescreens in xp pro |
| 14:35.49 | Qwell | chan_mobile is what it was committed as |
| 14:35.49 | _VoiceMeUp_COM | mobile |
| 14:35.51 | _VoiceMeUp_COM | form trunk |
| 14:35.58 | DarKnesS_WolF | what is chan_cellphone? |
| 14:36.09 | _VoiceMeUp_COM | cellphone is the prototype from david and well it was integrated and tweaked in as mobile |
| 14:36.18 | DarKnesS_WolF | ic |
| 14:36.23 | _VoiceMeUp_COM | its the granfather of mobile |
| 14:36.25 | _VoiceMeUp_COM | ;) |
| 14:36.34 | DarKnesS_WolF | haha okay :-) |
| 14:37.20 | DarKnesS_WolF | _VoiceMeUp_COM: so everything works with u ? |
| 14:37.22 | _VoiceMeUp_COM | 1.4.5 has improved in terms of noobienest |
| 14:37.29 | _VoiceMeUp_COM | well no |
| 14:37.32 | _VoiceMeUp_COM | im still compiling |
| 14:37.41 | _VoiceMeUp_COM | this an old laptop .. p3 1.8 |
| 14:38.12 | DarKnesS_WolF | _VoiceMeUp_COM: u patched aganiest 1.4.5 ? |
| 14:38.13 | _VoiceMeUp_COM | actually trying to ee what this laptop cpu |
| 14:38.14 | DarKnesS_WolF | or 1.4.4 ? |
| 14:38.15 | _VoiceMeUp_COM | saw this |
| 14:38.16 | _VoiceMeUp_COM | hci_scodata_packet: hci0 SCO packet for unknown connection handle 2 |
| 14:38.22 | _VoiceMeUp_COM | that why i had no rtp from headset |
| 14:38.27 | _VoiceMeUp_COM | a motorola hs80 |
| 14:38.34 | _VoiceMeUp_COM | 1.4.5 |
| 14:38.39 | DarKnesS_WolF | mmm |
| 14:38.42 | _VoiceMeUp_COM | no i mean trunk |
| 14:38.46 | _VoiceMeUp_COM | darn you |
| 14:38.47 | DarKnesS_WolF | ah ic |
| 14:38.50 | _VoiceMeUp_COM | giving me choices lol |
| 14:38.50 | DarKnesS_WolF | :P |
| 14:39.44 | *** join/#asterisk purplet (n=purplet@010.041.dsl.concepts.nl) |
| 14:40.04 | _VoiceMeUp_COM | the heck ? codec _zap |
| 14:41.14 | _VoiceMeUp_COM | 1,4,5 brewing... |
| 14:41.34 | *** join/#asterisk bintut (n=bintut@cm145.gamma178.maxonline.com.sg) |
| 14:41.45 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 14:41.45 | *** join/#asterisk jaxon_ (n=chatzill@ip67-91-86-195.z86-91-67.customer.algx.net) |
| 14:41.52 | jaxon_ | hello |
| 14:41.58 | _VoiceMeUp_COM | hi jax |
| 14:42.37 | jaxon_ | i just install trixbox. i need to configure zaptel.conf to use different channels? can i hand edit this? |
| 14:42.44 | Qwell | no |
| 14:43.52 | jaxon_ | when i run zaptelconf program, it just uses default 24 trunks. it won't prompt me for any conf. in my t1, i have trunk 1-15 for voice |
| 14:44.12 | Qwell | !wtf trixbox? |
| 14:44.15 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:44.15 | Qwell | erm |
| 14:44.19 | Qwell | ~trixbox |
| 14:44.20 | jbot | Trixbox is a full linux distro that includes , FreePBX, and other 3rd party add-ons. It is these things on top of which make it seriously painful to support and hence you will find little help here for it. Try asking in #trixbox , or their forums & WIKI at http://www.trixbox.org |
| 14:44.23 | Qwell | wrong trigger :D |
| 14:44.53 | logyati | ok, i made my asterisk server, i learnd about ivr, everything is working fine, using sip internally and calling to pstn phones... now everything is working fine... now im gonna make another step, i dont want asterisk controlling sip accounts, thats why i made my openser sip server... my question is, when i receive a call from pstn, for example, how to i set asterisk to send the call to openser? |
| 14:47.01 | lee_is_me | Is it me or could the AMI used a bit more structured output... |
| 14:47.15 | blitzrage | lee_is_me: oej has a branch that is fixing that |
| 14:47.16 | ber111 | Dial(SIP/NUMBER@SER.SERVER.IP) |
| 14:47.40 | lee_is_me | blitzrage: really? any samples of the output? Parsing the existing stuff is a PIA |
| 14:48.15 | lee_is_me | Its like parsing a page out of a novel ;) |
| 14:48.20 | blitzrage | lee_is_me: http://svn.digium.com/svn/asterisk/team/oej/moremanager/ |
| 14:48.37 | blitzrage | lee_is_me: well, the output isn't gonna change that much, it'll just be more standardized |
| 14:49.14 | lee_is_me | wouldn't be nice to be able to just grab a list of queue names and summary data in xml or tab separated format? |
| 14:49.33 | lee_is_me | but more standardized is nice too |
| 14:50.06 | lee_is_me | the way the output is now, makes it too easy to break existing code bases |
| 14:50.21 | lee_is_me | by changing a bit here and there in the output |
| 14:50.40 | lee_is_me | good to know, that there is improvement being worked on |
| 14:51.00 | blitzrage | lee_is_me: sure, patches are always welcome |
| 14:51.20 | lee_is_me | damn, I knew that was coming ;) |
| 14:51.42 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 14:51.47 | lee_is_me | I know C enough to read it a bit, but mostly from using C# a bit |
| 14:51.57 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00111ae4684c.cpe.net.cable.rogers.com) |
| 14:52.03 | lee_is_me | oops should have been "...dont know c..." |
| 14:54.30 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 14:54.50 | *** join/#asterisk Marshall-Laptop (n=eman0n@cpe-76-181-162-66.columbus.res.rr.com) |
| 14:55.10 | Marshall-Laptop | What are some good IAX2 Providers Besides Vitelity Telix And VoipStreet? |
| 14:55.17 | `Sean | jbot whos your owner |
| 14:55.18 | *** join/#asterisk Math` (n=privmath@modemcable037.229-56-74.mc.videotron.ca) |
| 14:55.35 | _VoiceMeUp_COM | we got a winner |
| 14:55.48 | _VoiceMeUp_COM | Marshall-Laptop we offer iax |
| 14:55.58 | Marshall-Laptop | site? |
| 14:56.03 | _VoiceMeUp_COM | jbot who's your daddy |
| 14:56.04 | jbot | YOU are, Mr Sexy Pants! |
| 14:56.08 | _VoiceMeUp_COM | www.voicemeup.com |
| 14:56.15 | _VoiceMeUp_COM | hehehe |
| 14:56.16 | purplet | Hello, someone who compiled app_odbcexec with asterisk 1.4.2? Can't get it to work :( |
| 14:56.20 | _VoiceMeUp_COM | <PROTECTED> |
| 14:56.21 | jbot | YOU are, Mr Sexy Pants! |
| 14:56.27 | _VoiceMeUp_COM | yeah ! i like the answer.. |
| 14:56.32 | _VoiceMeUp_COM | karma jbot +++ |
| 14:56.37 | Corydon76-home | purplet: email the author |
| 14:56.47 | _VoiceMeUp_COM | why not use 1.4.5 ? |
| 14:56.51 | Corydon76-home | purplet: or migrate to func_odbc |
| 14:57.31 | Marshall-Laptop | VoiceMeUp_COM |
| 14:57.39 | Marshall-Laptop | do you allow outgoing CID spoof? |
| 14:58.11 | _VoiceMeUp_COM | yes |
| 14:58.18 | _VoiceMeUp_COM | we call it CID ... |
| 14:58.27 | _VoiceMeUp_COM | spoof sounds.. evil |
| 14:58.33 | _VoiceMeUp_COM | pm me |
| 14:58.33 | `Sean | _VoiceMeUp_COM have you ever had trouble? |
| 14:59.46 | purplet | Corydon76-home: thx, going to check func_odbc |
| 15:02.36 | blitzrage | func_odbc!!! |
| 15:02.52 | Corydon76-home | ftw!!! |
| 15:03.21 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 15:04.16 | blitzrage | heck ya |
| 15:04.47 | *** join/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 15:05.32 | Math` | kinds kills a box when a drive switches to PIO during a raid1 resync |
| 15:06.14 | Corydon76-home | blitzrage: so have you tried mode=multirow yet? |
| 15:07.05 | blitzrage | Corydon76-home: not yet unfortunately, but I ended up upgrading to the latest func_odbc from svncommunity |
| 15:07.11 | blitzrage | does that have it in there now? or is that only trunk |
| 15:07.24 | blitzrage | I think the upgrade I did was just the Oracle bug you fixed |
| 15:08.51 | *** join/#asterisk EvilGreen (n=Miranda@ppp85-141-155-92.pppoe.mtu-net.ru) |
| 15:10.03 | *** join/#asterisk gerwinin (n=gerwinin@ip5457b30e.direct-adsl.nl) |
| 15:11.07 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 15:13.49 | Corydon76-home | blitzrage: it's in svncommunity, too, I think |
| 15:15.02 | Corydon76-home | blitzrage: the Oracle fix may or may not have done anything... |
| 15:15.23 | _VoiceMeUp_COM | wonder hmm yeah i wonder |
| 15:15.38 | _VoiceMeUp_COM | got this new DB prototype not sure waht it was.. but i got a test account |
| 15:15.43 | Corydon76-home | The box I'm using with Oracle still runs out of cursors after 2 weeks |
| 15:15.52 | _VoiceMeUp_COM | its like.. a new way to see tables etc.. supposed to be 500% faster then anything out there |
| 15:16.13 | *** join/#asterisk littleball (n=littleba@bb220-255-70-91.singnet.com.sg) |
| 15:16.40 | Corydon76-home | blitzrage: I suspect there's a massive memory leak in the Oracle libraries |
| 15:16.42 | *** join/#asterisk Marshall- (n=Marshall@cpe-76-181-162-66.columbus.res.rr.com) |
| 15:16.44 | littleball | hello, how to make asterisk receive sms from fixed line? |
| 15:17.02 | Corydon76-home | littleball: you can't, unless you're in the UK |
| 15:17.29 | Corydon76-home | only BT circuits support that functionality |
| 15:17.59 | littleball | Corydon76-home, i know my operator support fixed line |
| 15:18.01 | Math` | u can sms to a landline in uk? |
| 15:18.03 | littleball | not only UK |
| 15:18.12 | Corydon76-home | Math`: yes |
| 15:18.17 | Math` | if you do here, they text2speech the message lol |
| 15:18.35 | Corydon76-home | littleball: app_sms is the key. See the application documentation |
| 15:19.11 | littleball | actualy, my question: in app_sms.c file, h->opause = 2400; /* initial message delay 300ms (for BT) */ |
| 15:19.23 | littleball | how is 300ms calculated? |
| 15:19.24 | Qwell | oh, speaking of app_sms |
| 15:19.37 | Qwell | Corydon76-home: that mkdir patch you posted had a bunch of formatting fixes in app_sms |
| 15:19.42 | littleball | because i need to increase the initial message delay |
| 15:19.45 | Corydon76-home | littleball: because there are 8000 samples per second |
| 15:19.54 | Corydon76-home | Qwell: I know, it was massively bad |
| 15:20.13 | littleball | Corydon76-home, where 8000 samples per second is defined? |
| 15:20.24 | Corydon76-home | littleball: ITU standard |
| 15:20.35 | littleball | ok. sampling what? |
| 15:20.39 | littleball | voice? |
| 15:20.46 | Corydon76-home | Correct |
| 15:20.53 | littleball | thanks. this is the point i missing |
| 15:20.56 | Math` | its probably modulating the sms as FSK over voice |
| 15:20.57 | *** join/#asterisk elg (n=fugalh@216.31.27.110) |
| 15:21.03 | Corydon76-home | 8000 samples per second is the basis for the entire PSTN |
| 15:21.34 | littleball | thanks a lot |
| 15:22.22 | Corydon76-home | Qwell: did you see that localtime() is still being used in app_sms? |
| 15:22.34 | littleball | Corydon76-home, i did get one sms received. but most times, just get SMS[1] TX 92 01 FF 6E ....SMS[70] TX 92 01 FF 6E |
| 15:22.43 | littleball | what is your recommendation? |
| 15:22.54 | elg | hi, i'm seeing some strange symptoms on this client's asterisk box. TDM400P usually works fine, but from time to time will get in a weird state where the fxs lines are half-dead. they still have a dialtone, but asterisk doesn't recognize the dtmf. they don't ring, though asterisk says it's ringing them |
| 15:22.57 | Corydon76-home | littleball: No idea; being in the US, I've never used it |
| 15:22.58 | littleball | SMS[0] TX 93 00 6D |
| 15:23.05 | littleball | ok. |
| 15:23.20 | littleball | i try to increase the initial message delay |
| 15:23.21 | elg | reloading the modules fixes it |
| 15:23.36 | _VoiceMeUp_COM | BTWqwell |
| 15:23.41 | _VoiceMeUp_COM | Qwell[] ? |
| 15:23.43 | _VoiceMeUp_COM | chan_mobile.c:272: error: `HANDSFREE_AGW_PROFILE_ID' undeclared (first use in this function) |
| 15:23.56 | Corydon76-home | elg: sounds like a low power situation |
| 15:23.57 | _VoiceMeUp_COM | some libs have HANDSFREE_PROFILE_ID |
| 15:24.09 | _VoiceMeUp_COM | and not the AGW part .. FYI.. if you can patch for both |
| 15:24.10 | Corydon76-home | elg: might want to check into a UPS for that system |
| 15:24.24 | _VoiceMeUp_COM | BTW got 4 ups to sell brand new |
| 15:24.28 | _VoiceMeUp_COM | 3500's HP |
| 15:24.29 | Qwell | _VoiceMeUp_COM: You need to upgrade your bluez then |
| 15:24.31 | _VoiceMeUp_COM | R |
| 15:24.31 | elg | it's on a ups, so they say |
| 15:24.36 | elg | maybe needs a beefier PSU? |
| 15:24.43 | Corydon76-home | elg: I've seen that happen before when voltage dropped momentarily on the electrical circuit |
| 15:24.53 | _VoiceMeUp_COM | Qwell on it thanks |
| 15:25.01 | Corydon76-home | elg: possibly that, too |
| 15:25.09 | _VoiceMeUp_COM | maybe TRAP that error and say use latest versions |
| 15:25.14 | elg | Corydon76-home: thanks, that gives me something to go on |
| 15:25.16 | _VoiceMeUp_COM | since you know that thats it |
| 15:25.17 | gerwinin | VoiceMeUp_com: starting your own provider ? |
| 15:25.21 | _VoiceMeUp_COM | no ? |
| 15:25.28 | _VoiceMeUp_COM | starting ? |
| 15:25.39 | _VoiceMeUp_COM | been in this biz for 2 years.. 4 hosting and 16 dev |
| 15:25.45 | _VoiceMeUp_COM | well going on 16 |
| 15:25.54 | gerwinin | VoiceMeUp: ah okay no visited your website |
| 15:26.03 | _VoiceMeUp_COM | ;) |
| 15:26.25 | gerwinin | VoiceMeUp: I used to have a provider as well but sold it 1 year ago |
| 15:26.49 | _VoiceMeUp_COM | actualy mor ehten 2 eyars.. worked witha competitor and got them off the gorund then got sharked into going on my own then got bought and partner with new firm |
| 15:27.12 | _VoiceMeUp_COM | T.....x |
| 15:27.20 | _VoiceMeUp_COM | and where not talking about texas |
| 15:27.36 | gerwinin | VoiceMeUp: I had a pretty big provider we used to do odm for the big telco's here in my country |
| 15:28.02 | _VoiceMeUp_COM | you in de ? |
| 15:28.15 | gerwinin | VoiceMeUp: did pearing , dids and developed a medium and small sized business pabx |
| 15:28.17 | littleball | hello, how to hangup all channels ? |
| 15:28.20 | littleball | through CLI |
| 15:28.26 | _VoiceMeUp_COM | al ? |
| 15:28.28 | _VoiceMeUp_COM | cant |
| 15:28.33 | _VoiceMeUp_COM | but soft hangup <tab> |
| 15:28.39 | _VoiceMeUp_COM | youll see ones interested in |
| 15:28.49 | _VoiceMeUp_COM | unless a real reload as in restart could |
| 15:28.58 | _VoiceMeUp_COM | to bad you cant * a channel name |
| 15:28.58 | gerwinin | VoiceMeUp.com : No I am NL and business was in NL, DE, France, Austria, Taiwan , china |
| 15:29.05 | _VoiceMeUp_COM | like soft hangup sip/bob* |
| 15:29.11 | littleball | i want to hangup all calls, but the hangup AGI will be called properly |
| 15:29.12 | _VoiceMeUp_COM | ah cool |
| 15:29.14 | _VoiceMeUp_COM | like nl |
| 15:29.20 | _VoiceMeUp_COM | they give me rough times on poker tables |
| 15:29.44 | gerwinin | VoiceMeUp_com : if you want to have good peering partners make sure you place a gateway at ancotel in germany |
| 15:31.43 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 15:32.43 | _VoiceMeUp_COM | well our next gw is in 1 wilshire |
| 15:32.52 | _VoiceMeUp_COM | then we have a few E's in belgium etc |
| 15:33.12 | gerwinin | VoiceMeUp_com: cool |
| 15:33.14 | _VoiceMeUp_COM | also asia in the next month or 2 |
| 15:33.20 | _VoiceMeUp_COM | singapore i think |
| 15:33.24 | _VoiceMeUp_COM | not sure yet |
| 15:33.35 | *** join/#asterisk Mez (n=Mez@ubuntu/member/mez) |
| 15:33.42 | gerwinin | VoiceMeUp_com: you can better go in that case for Taiwan / China |
| 15:34.26 | gerwinin | VoiceMeUp_com: singapore = expensive and rates are not so good |
| 15:34.42 | _VoiceMeUp_COM | yeah taiwan |
| 15:34.46 | _VoiceMeUp_COM | then not sure |
| 15:34.49 | Mez | Hi there, I'm looking for a way to make it so that I can make calls and have them linked into jackd, so I can broadcast them on air. atm, the only way I can find is to use a sip phone with aoss and linking alsa into jack - but this degrades the quality of the call massively. Does anyone know the best way to do this |
| 15:35.21 | gerwinin | VoiceMeUp_com: you need to have minutes for china and indonesia and they are difficult to get in singapore |
| 15:35.31 | tzanger | you could do it by conferencing the call into a meetme room and having one of the members being a local channel with oss or alsa |
| 15:35.35 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 15:35.53 | blitzrage | codefreeze: w000000000000t! |
| 15:35.55 | Mez | tzanger, was that at me |
| 15:35.56 | Mez | ? |
| 15:36.16 | tzanger | Mez: yes |
| 15:36.48 | Mez | tzanger - I've no idea how that would work! |
| 15:36.55 | *** part/#asterisk karleeto (i=karl@gentoo.karlhaines.com) |
| 15:37.14 | Mez | if only there was a jack enabled sip client! |
| 15:40.04 | elg | Mez: I've looked into adding jack support to twinkle. shouldn't be too hard, and then it would be closer to working on os x |
| 15:40.23 | *** join/#asterisk hyphen (n=hyphen@c-71-224-214-148.hsd1.pa.comcast.net) |
| 15:40.28 | Mez | elg, jsack support in twinle would be awesome - it's my client of choice |
| 15:40.33 | Mez | elg, do you have anything working ? |
| 15:40.38 | _VoiceMeUp_COM | all i get is Jun 16 11:40:10] DEBUG[28326] chan_mobile.c: rfcomm_write() (frankhead) [ |
| 15:40.38 | _VoiceMeUp_COM | RING |
| 15:40.38 | _VoiceMeUp_COM | ] |
| 15:40.45 | elg | no, I just reviewed the source to see if it would be doable |
| 15:40.45 | _VoiceMeUp_COM | and no ring in the headset |
| 15:40.53 | elg | then I had to focus on finals :( |
| 15:40.57 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 15:41.02 | Mez | elg :( is it possible? |
| 15:41.14 | _VoiceMeUp_COM | and kernel: hci_scodata_packet: hci0 SCO packet for unknown connection handle 1328 |
| 15:41.14 | elg | yeah, I didn't see any major obstacles |
| 15:41.17 | _VoiceMeUp_COM | in llogs |
| 15:41.40 | Mez | elg, I'd love to see that, however, I've no damned clue how I'd go about doing it myself :P |
| 15:42.27 | elg | and I'm not sure when I'll get the chance, but it's on my list :) |
| 15:42.47 | Mez | lol |
| 15:42.52 | Mez | well, please keep me updated |
| 15:44.02 | elg | this is what I told the author in an email: "Adding Jack audio should be straightforward, although I might have some questions about buffer sizes for you later." |
| 15:44.15 | elg | i'm a big help to myself, as you can tell |
| 15:44.40 | Mez | ;) |
| 15:44.52 | Mez | elg, jack support would be uber helpful |
| 15:45.08 | Mez | lol - so I can pipe it into idjc and broadcast incoming/outgoing phone calls to the world :D |
| 15:45.40 | elg | i think more of the things you could do bringin in audio from other apps into the call :) |
| 15:47.15 | Mez | lol, well I work at a couple of online stations, we want to get a phone system setup so that we can take/answer incoming calls |
| 15:47.53 | Mez | I'm just trying to learn to work with jack - by making a "soundboard" to plug into jack (which will also be useful for outgoing calls so I can set up some prank calls to be recorded) |
| 15:50.14 | *** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net) |
| 15:58.43 | _VoiceMeUp_COM | DarKnesS_WolF ? |
| 16:00.30 | _VoiceMeUp_COM | think im getting to it |
| 16:03.51 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 16:08.02 | _VoiceMeUp_COM | Got SIP response 603 "Declined (no dialog)" back from |
| 16:08.14 | _VoiceMeUp_COM | this a bug in the sip ? or its an options with empty packets |
| 16:08.19 | _VoiceMeUp_COM | like a ping |
| 16:08.59 | DarKnesS_WolF | _VoiceMeUp_COM: yes ? |
| 16:09.59 | _VoiceMeUp_COM | still trying to do this bluetooth thing |
| 16:10.03 | _VoiceMeUp_COM | ran into a huge prob |
| 16:10.12 | _VoiceMeUp_COM | upgraded alsa-libs and alsa-utils |
| 16:10.17 | _VoiceMeUp_COM | for the audio |
| 16:10.37 | _VoiceMeUp_COM | BUT.. now i need gtk>208.0 |
| 16:10.41 | _VoiceMeUp_COM | im on 2.0 |
| 16:10.49 | _VoiceMeUp_COM | i mean 2.4.7 |
| 16:10.52 | _VoiceMeUp_COM | weird |
| 16:13.42 | DarKnesS_WolF | gtk !? |
| 16:13.46 | DarKnesS_WolF | what u'll do with GTK !? |
| 16:13.47 | *** join/#asterisk MindX (n=MindX@office.microlink.lt) |
| 16:13.58 | _VoiceMeUp_COM | no idea it needs glibs |
| 16:14.05 | _VoiceMeUp_COM | so back to sqare one |
| 16:14.14 | _VoiceMeUp_COM | maybe some core libs for sound |
| 16:14.20 | DarKnesS_WolF | may be |
| 16:14.22 | gerwinin | VoiceMeUp_COM : make a symbolic link to your libs |
| 16:14.29 | DarKnesS_WolF | i'm still fighting to get gtalk running again with 1.4.5 |
| 16:14.40 | _VoiceMeUp_COM | nah they barfing with errors |
| 16:14.57 | _VoiceMeUp_COM | configure: error: Library requirements (glib-2.0 >= 2.8.0, gthread-2.0 >= 2.8.0, gobject-2.0 >= 2.8.0) not met; consider adjusting the PKG_CONFIG_PATH environment variable if your libraries are in a nonstandard prefix so pkg-config can find them. |
| 16:15.02 | _VoiceMeUp_COM | so i need glib.. atk etc |
| 16:15.37 | _VoiceMeUp_COM | fun part is finding out hte order to compile it all |
| 16:16.26 | DarKnesS_WolF | _VoiceMeUp_COM: what is ur distro ? |
| 16:17.02 | _VoiceMeUp_COM | centos |
| 16:17.08 | _VoiceMeUp_COM | on laptop |
| 16:17.28 | DarKnesS_WolF | mmmm |
| 16:17.30 | DarKnesS_WolF | best luck :-D |
| 16:17.31 | _VoiceMeUp_COM | since my video nuked and AWS on keyboard nuked.. i need a unix flavor on it |
| 16:17.35 | DarKnesS_WolF | i'll try all this tomorrow :D |
| 16:17.49 | _VoiceMeUp_COM | yeah , by thn ill have overcome all problems and can help |
| 16:18.24 | DarKnesS_WolF | _VoiceMeUp_COM: great :-) |
| 16:19.55 | *** join/#asterisk karleeto (i=karl@gentoo.karlhaines.com) |
| 16:20.14 | karleeto | can anyone reccommend a good software (IAX2) phone for linux?> |
| 16:20.20 | _VoiceMeUp_COM | sjphone ? |
| 16:20.23 | _VoiceMeUp_COM | ah |
| 16:20.25 | _VoiceMeUp_COM | no idea |
| 16:20.32 | _VoiceMeUp_COM | think theres one with ubuntu |
| 16:20.46 | karleeto | idefisk works for linux as well, right? |
| 16:21.06 | DarKnesS_WolF | karleeto: yes |
| 16:21.11 | *** join/#asterisk _DAW (n=chatzill@adsl-241-93-3.msy.bellsouth.net) |
| 16:22.31 | _VoiceMeUp_COM | if anyone has a good source for windows sip let me know |
| 16:22.36 | _VoiceMeUp_COM | all i come into is bogus pos |
| 16:22.41 | blitzrage | windows sip? |
| 16:22.42 | _VoiceMeUp_COM | like very bad stuff |
| 16:22.45 | _VoiceMeUp_COM | windows sip client |
| 16:22.48 | _VoiceMeUp_COM | source |
| 16:22.49 | blitzrage | idefisk |
| 16:22.58 | _VoiceMeUp_COM | all SDK's i bought.. well companys bellied up |
| 16:23.07 | codefreeze | blitzrage: w0000t? |
| 16:23.08 | _VoiceMeUp_COM | aint that java ? |
| 16:23.16 | blitzrage | codefreeze: your change to CDR_CONTROL() :) |
| 16:23.54 | codefreeze | blitzrage: my pleasure. So little feedback. I try to please. |
| 16:24.01 | blitzrage | codefreeze: nice! |
| 16:24.02 | _VoiceMeUp_COM | ah ideafisk |
| 16:24.06 | _VoiceMeUp_COM | yeah they want like 20k |
| 16:24.08 | _VoiceMeUp_COM | yeah right |
| 16:24.29 | _VoiceMeUp_COM | i can hire 300 guys in paki/india for 1 year and get like 50 diff phones for that price.. |
| 16:27.28 | *** join/#asterisk shane2k (n=chatzill@ip67-91-86-195.z86-91-67.customer.algx.net) |
| 16:27.39 | shane2k | hi there |
| 16:31.19 | *** join/#asterisk TheBigSpark (n=thebigsp@216.161.248.183) |
| 16:31.26 | _VoiceMeUp_COM | hi shane2k |
| 16:32.03 | TheBigSpark | hello, all. |
| 16:32.13 | _VoiceMeUp_COM | hello back at you |
| 16:33.08 | TheBigSpark | I have a "simple" question. Any one here familure with setting up zap channels, specificly a quad-fxo card. |
| 16:33.12 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 16:33.26 | blitzrage | what does chan_features provide? |
| 16:33.37 | _VoiceMeUp_COM | blitzrage was asking myself the same exact thing |
| 16:33.48 | _VoiceMeUp_COM | will need to scout the .c |
| 16:33.50 | blitzrage | you'd think I should know this.... :) |
| 16:34.58 | *** join/#asterisk gardo (n=gardo@121.97.255.20) |
| 16:35.26 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 16:36.14 | _VoiceMeUp_COM | woiw |
| 16:36.18 | _VoiceMeUp_COM | all this for shitnitz |
| 16:36.25 | _VoiceMeUp_COM | dbus.36 needed and that on centos5 |
| 16:36.27 | _VoiceMeUp_COM | i think |
| 16:37.43 | TheBigSpark | any one? It has been a while (version 1.2) since I set up any zap channels. I am getting the error "Unable to reconfigure channel '1-4'" Any ideas. (btw, I have been using * since 01) |
| 16:40.45 | *** part/#asterisk shane2k (n=chatzill@ip67-91-86-195.z86-91-67.customer.algx.net) |
| 16:41.20 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 16:42.00 | Dr-Linux | huh lumenvox |
| 16:44.28 | _VoiceMeUp_COM | BTW asteirsk is not cutting of the first 15 chars of callerid name sent by Useragent : Sipura/SPA942-4.1.10(e) |
| 16:44.35 | _VoiceMeUp_COM | so its like 50 chars lol |
| 16:51.06 | _VoiceMeUp_COM | anyone good with pkg_config |
| 16:51.14 | _VoiceMeUp_COM | <PROTECTED> |
| 16:51.16 | _VoiceMeUp_COM | iots there |
| 16:51.22 | _VoiceMeUp_COM | but pkg-config dbus cant find it |
| 16:57.17 | *** part/#asterisk karleeto (i=karl@gentoo.karlhaines.com) |
| 16:57.28 | *** join/#asterisk perf3kt (n=sd@cpe-24-166-11-173.indy.res.rr.com) |
| 17:00.03 | _VoiceMeUp_COM | been compiling stuff so long i dont remmeber what i compiled it for |
| 17:01.05 | tzafrir_laptop | perf3kt, here? |
| 17:02.30 | perf3kt | on the cli I get Got SIP response 400 "Bad Request" |
| 17:03.27 | *** join/#asterisk TheBigSpark (n=thebigsp@216.161.248.183) |
| 17:03.33 | _VoiceMeUp_COM | yeah |
| 17:03.37 | _VoiceMeUp_COM | bad |
| 17:03.45 | tzafrir_laptop | that's an error. Related to anything? |
| 17:03.50 | perf3kt | I think I have the registration for my sutom VOIP setup bad |
| 17:04.10 | perf3kt | well no calls are coming in |
| 17:05.07 | tzafrir_laptop | 'sipshow registry' should give you an idea about that |
| 17:05.58 | TheBigSpark | perf3kt: I know this may seem like stupid mistake, but I made it recently, if you can make outgoing calls, but not get incomming, check the context the incommming calls goto. If it does not exsist, or is otherwise malformed, incomming calls will fail. |
| 17:08.25 | perf3kt | there are two entries, one of them is registered |
| 17:09.42 | perf3kt | well, at least that error won't show |
| 17:15.39 | perf3kt | i'm trying to enderstad sip client versus server |
| 17:15.43 | perf3kt | *understand |
| 17:16.36 | perf3kt | in order to setup the correct nat settings |
| 17:21.37 | *** join/#asterisk bintut (n=bintut@cm53.gamma179.maxonline.com.sg) |
| 17:21.37 | bintut | why is it that i always hear a "click" sound either on analog phones or sip softphones? |
| 17:21.37 | TheBigSpark | Does anyone have any ideas about this: |
| 17:21.37 | TheBigSpark | [Jun 16 10:26:07] ERROR[6134]: chan_zap.c:10455 build_channels: Unable to reconfigure channel '1' |
| 17:21.37 | TheBigSpark | I have my zaptel.conf and zapata.conf files set-up correctly. |
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| 17:26.02 | *** join/#asterisk GlobeTrotter (n=eric@190.10.28.145) |
| 17:27.16 | GlobeTrotter | hi all.. i cant get zaptel install on centos 5.. i think that all the compliling goes good,, but modprobe zaptel gives me this error FATAL: Module zaptel not found. |
| 17:27.20 | GlobeTrotter | please help |
| 17:27.23 | DarKnesS_WolF | someting wrong with gtalk register with asterisk ? |
| 17:27.53 | DarKnesS_WolF | GlobeTrotter: i think the compiling didn't went good may be the kernel name is not the same as teh kernelsource tree |
| 17:30.41 | GlobeTrotter | thanks <DarKnesS_WolF> how do i verify that |
| 17:35.38 | *** join/#asterisk nowork (n=jfu2808@216.254.141.97) |
| 17:36.00 | _VoiceMeUp_COM | Packet2Packet |
| 17:36.04 | _VoiceMeUp_COM | can we disable this ? |
| 17:36.09 | _VoiceMeUp_COM | i get no audio im sure coz of it |
| 17:36.29 | nowork | hi, when I call some 1800 numbers, the IVR of select language didn't bill my cell. can we do this kind of ivr on asterisk? how ?thanks |
| 17:36.52 | _VoiceMeUp_COM | and where my sip debug function ? |
| 17:36.54 | _VoiceMeUp_COM | in trunk |
| 17:38.23 | _VoiceMeUp_COM | << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [S |
| 17:38.25 | _VoiceMeUp_COM | foudn why |
| 17:38.49 | _VoiceMeUp_COM | is packet 2 packet trying to bind to an ealier v of ast ? |
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| 17:42.52 | DarKnesS_WolF | GlobeTrotter: http://www.voip-info.org/wiki/view/Asterisk+Zaptel+Installation |
| 17:45.22 | _VoiceMeUp_COM | [ TYPE: Null Frame (5) SUBCLASS: N/A (0) |
| 17:45.27 | _VoiceMeUp_COM | can one let me know wth this is ? |
| 17:45.38 | GlobeTrotter | thanks again' |
| 17:46.39 | nowork | hi, anyone has the answer for my question?? |
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| 17:48.30 | _VoiceMeUp_COM | amaflags |
| 17:48.33 | _VoiceMeUp_COM | check those |
| 17:50.31 | nowork | ok.thanks |
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| 17:52.38 | _VoiceMeUp_COM | ok so anyway to disable packet 2 packet ? |
| 17:55.54 | _VoiceMeUp_COM | ok nm |
| 17:55.56 | _VoiceMeUp_COM | uninstalling |
| 17:58.37 | _VoiceMeUp_COM | 1.4.5 also crashes cisco |
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| 18:12.16 | DarKnesS_WolF | yaaaaaaaaaay my gtalk works :-) |
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| 18:26.11 | GlobeTrotter | hey guys.. im trying to compile zaptel.. the instructions say to use make linus26 but i cant get that to work |
| 18:26.46 | GlobeTrotter | i get an error make: *** No rule to make target `linux26'. Stop. |
| 18:27.02 | GlobeTrotter | can someone please tell me what i am doing wrong |
| 18:27.22 | fadey | what version are you using? |
| 18:27.52 | GlobeTrotter | zaptel-1.4.3 |
| 18:28.31 | fadey | I guess ./configure && make && make install will work |
| 18:29.54 | GlobeTrotter | that looks like it ran ok,, but i get this error when i try modprobe zaptel |
| 18:29.54 | GlobeTrotter | FATAL: Module zaptel not found. |
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| 18:32.50 | DarKnesS_WolF | GlobeTrotter: past uname -a output here |
| 18:33.51 | GlobeTrotter | Linux localhost.localdomain 2.6.18-8.el5 #1 SMP Thu Mar 15 19:46:53 EDT 2007 x86_64 x86_64 x86_64 GNU/Linux |
| 18:34.19 | DarKnesS_WolF | ok now tell me what is in ur Makefile in ur /usr/src/linux ? |
| 18:35.42 | GlobeTrotter | you mean this line EXTRAVERSION = 2.6.18-8.el5 |
| 18:36.16 | GlobeTrotter | ? |
| 18:36.35 | rob0 | What zaptel device do you have? |
| 18:36.49 | GlobeTrotter | none.. i want to use ztdummy |
| 18:38.12 | rob0 | and "modprobe -v ztdummy" does ... ? |
| 18:38.37 | GlobeTrotter | FATAL: Module ztdummy not found. |
| 18:38.57 | rob0 | try "depmod ; modprobe -v ztdummy" now. |
| 18:39.25 | DarKnesS_WolF | GlobeTrotter: check if there is a .version file inside ur kernel tree |
| 18:39.55 | rob0 | I would guess the Makefile detects the running kernel version, but I don't know. |
| 18:40.06 | DarKnesS_WolF | rob0: it's kernel issues not the zaptel |
| 18:40.11 | DarKnesS_WolF | i had the same problems with mandriva :-) |
| 18:40.19 | rob0 | I also would have guessed the "make install" would do depmod. |
| 18:40.21 | DarKnesS_WolF | was pain and in mandriva 2007 i find a .version file |
| 18:40.45 | GlobeTrotter | i ran depmod i got no error then i ran modprobe -v ztdummy |
| 18:40.46 | GlobeTrotter | [root@localhost zaptel-1.4.3]# modprobe -v ztdummy |
| 18:40.46 | GlobeTrotter | FATAL: Module ztdummy not found. |
| 18:40.56 | DarKnesS_WolF | GlobeTrotter: did u look for a .version file ? |
| 18:40.57 | rob0 | Maybe it needs the kernel source with a good .config file? |
| 18:41.28 | DarKnesS_WolF | rob0: sure it neeed the kernel source and i think he did install it |
| 18:42.11 | rob0 | I've only ever installed zaptel on my own custom kernels, so everything works fine. And mine are Slackware / Slamd64. |
| 18:42.24 | DarKnesS_WolF | aww i hate slack :P |
| 18:42.25 | DarKnesS_WolF | brb |
| 18:43.15 | GlobeTrotter | i am looking in the /kernels/2.6.18-8.1.6.el5-x86_64/ there is no file called version in htere |
| 18:43.26 | GlobeTrotter | im i looking for the right thing? |
| 18:44.14 | fadey | did you install kernel-devel package? |
| 18:44.22 | GlobeTrotter | yes |
| 18:45.38 | GlobeTrotter | rpm -q kernel-devel |
| 18:45.39 | GlobeTrotter | kernel-devel-2.6.18-8.1.6.el5 |
| 18:46.18 | GlobeTrotter | what ami doing wrong :( |
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| 18:47.45 | GlobeTrotter | i want to install ztdummy |
| 18:47.55 | GlobeTrotter | i dont have any hardware |
| 18:53.11 | perf3kt | I can't get anything to hit my * box |
| 18:53.22 | perf3kt | I've been working on these nat settings |
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| 18:57.15 | _VoiceMeUp_COM | same here |
| 18:57.40 | _VoiceMeUp_COM | 1.4.5 no audio.. some audio .. no audio.. reload with 1.2.19 all good.. |
| 18:57.58 | _VoiceMeUp_COM | then 1.4.5 killed my cisco.. had to reboot it.. happened 3 times.. bad sip mess |
| 18:58.14 | _VoiceMeUp_COM | so going back to 0.85 |
| 18:58.31 | *** join/#asterisk AvoidingDeadlock (n=brian@ip70-189-76-137.ok.ok.cox.net) |
| 18:58.37 | _VoiceMeUp_COM | unless one knows the prob |
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| 19:05.36 | DarKnesS_WolF | GlobeTrotter: .version |
| 19:05.38 | DarKnesS_WolF | not version |
| 19:05.42 | DarKnesS_WolF | it's a hiden file |
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| 19:05.57 | DarKnesS_WolF | GlobeTrotter: no u need kernel-source as i think |
| 19:06.00 | DarKnesS_WolF | and or kernel-headers |
| 19:06.31 | GlobeTrotter | rpm -q kernel-devel |
| 19:06.31 | GlobeTrotter | <GlobeTrotter> kernel-devel-2.6.18-8.1.6.el5 |
| 19:09.35 | GlobeTrotter | how do i see hidden files.. and if its not there were do i get it from? |
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| 19:13.04 | GlobeTrotter | ok i run ls -a and i dont see the .version file |
| 19:13.19 | GlobeTrotter | do i have to copy it from somewhere/ |
| 19:13.22 | GlobeTrotter | ? |
| 19:14.33 | *** join/#asterisk fadey (n=fadey@116.19.218.87.dynamic.jazztel.es) |
| 19:18.58 | mrdigital | GlobeTrotter: need help? |
| 19:23.40 | *** part/#asterisk BSD_Tech (n=BSDTech@adsl-69-230-169-41.dsl.irvnca.pacbell.net) |
| 19:24.11 | GlobeTrotter | yes i do |
| 19:24.13 | DarKnesS_WolF | GlobeTrotter: don't know i don't use centos :-) |
| 19:24.27 | GlobeTrotter | ah ok |
| 19:29.27 | perf3kt | I can't get anythign ringing an extension from outside |
| 19:29.47 | perf3kt | but I see alot of outside traffic on the box in sip debug |
| 19:31.13 | GlobeTrotter | mrdigital yes i do please |
| 19:31.34 | tzafrir_laptop | GlobeTrotter, the .version file is generated at build time if it's not there |
| 19:32.10 | tzafrir_laptop | if you pulled the source from svn |
| 19:32.31 | tzafrir_laptop | if you got a standard tarball, it should be included |
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| 19:32.46 | tzafrir_laptop | unless you copied * or something... |
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| 19:36.07 | perf3kt | frir I sewe alot of activity with the outside number coming in from sip oroviuder, but no calls dial extensions |
| 19:36.19 | GlobeTrotter | /usr/src/asterisk-1.4.5/.version |
| 19:36.20 | GlobeTrotter | /usr/src/libpri-1.4.0/.version |
| 19:36.20 | GlobeTrotter | /usr/src/zaptel-1.4.3/.version |
| 19:36.20 | GlobeTrotter | /usr/src/zaptel-1.4.3/xpp/.version |
| 19:36.34 | GlobeTrotter | these are all the .versions files that i have on the system' |
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| 20:13.17 | cy303 | Can anyone suggest a way to run something after a meetme conference has ended? Is it even possible? |
| 20:13.35 | cy303 | basically want to do something like this: |
| 20:13.36 | cy303 | exten => s,n,MeetMe(${ARG1},rMs) |
| 20:13.43 | cy303 | exten => s,n,System(cmd) |
| 20:16.58 | sergee | cy303: i'm using 'h' extension to bill MeetMe users |
| 20:18.32 | sergee | exten => h,1,DeadAGI(user-exit.pl) |
| 20:18.38 | sergee | something like that |
| 20:18.47 | `Sean | Anyone here know of any toher service such as tollfreegateway |
| 20:18.51 | `Sean | s/toher/other/ |
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| 20:39.32 | cy303 | sergee: hmm, interesting |
| 20:39.38 | cy303 | thanks dude, I'll have a look at that |
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| 20:56.11 | Dovid | after applying a patch to asterisk do i need to do make and make install or just make install ? |
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| 21:09.06 | Nuitari | Hi |
| 21:11.27 | Nuitari | Why would asterisk try to speak with a device in a codec that it can't translate, ie g722 ? |
| 21:13.24 | Nuitari | hum, there is a g722 translation in trunk |
| 21:13.42 | Nuitari | any chance to get it in 1.4 ? |
| 21:14.29 | ber111 | is the asterisk box supposed to transcode |
| 21:14.35 | ber111 | or is it just passing through the call to another device |
| 21:14.42 | Nuitari | passthru only on 1.4 |
| 21:14.50 | ber111 | 1.2 does passthru |
| 21:14.59 | ber111 | i have done g729 passthru no problem |
| 21:15.00 | Nuitari | 1.4 too |
| 21:15.11 | Nuitari | but the problem is that most devices don't support g722 |
| 21:15.14 | ber111 | if asterisk isnt terminating the call |
| 21:15.25 | ber111 | then it can speak to a device that says g722 supported in the invite |
| 21:15.35 | ber111 | and pass it on to wherever its supposed to terminate the call |
| 21:15.43 | ber111 | hopefully a device that can talk g722 |
| 21:15.46 | Nuitari | yeah that I understand |
| 21:15.49 | ber111 | or else the call will get rejected for no codec |
| 21:16.03 | Nuitari | but if the other device doesn't speak g722, why isn't asterisk trying to use a different codec, instead of failing? |
| 21:16.17 | ber111 | whatever is starting the call |
| 21:16.22 | ber111 | liek say a soft phone |
| 21:16.28 | ber111 | specifies what type of codec it supports |
| 21:16.33 | ber111 | so it can list like 4 codecs |
| 21:16.37 | ber111 | ulaw, alaw 729, 722 |
| 21:16.50 | ber111 | teh codec is negotiated with the endpoint |
| 21:16.57 | ber111 | and if there is one supported by both it will be chosen |
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| 21:17.14 | ber111 | if whatever that far endpoint is doesnt support 722 and the softphone says it only supports 722 |
| 21:17.19 | ber111 | then the call wont work |
| 21:17.50 | Nuitari | I'm checking the trace to see what's happenning |
| 21:18.04 | ber111 | yeah you can look at the invite from whatever is starting the call |
| 21:18.11 | ber111 | u will see what codecs it says its supporting to asterisk |
| 21:18.12 | Nuitari | yeah |
| 21:18.20 | Nuitari | g722, ulaw, alaw, g729 |
| 21:18.38 | ber111 | ok then read the confirm back from the far endpoint |
| 21:18.40 | ber111 | saying what it supports |
| 21:18.46 | ber111 | it _should_ support ulaw |
| 21:18.47 | Nuitari | not g722 |
| 21:18.53 | Nuitari | but all the others and a bunch more are there |
| 21:19.08 | ber111 | the sip debug will tell u where the error is |
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| 21:19.22 | ber111 | if the far end supports ulaw it should be negotiated ulaw |
| 21:19.33 | ber111 | just make sure that asterisk has those codecs supproted for yoru sip login |
| 21:19.40 | ber111 | when you specify the sip account |
| 21:19.58 | Nuitari | yeah it has |
| 21:19.59 | ber111 | ok |
| 21:19.59 | Nuitari | g722 is the new one that I added cause we have 2 phones that support it |
| 21:21.36 | Nuitari | though it looks like asterisk first negociates the g722 link, then notices it can't translate when it starts talking to the 2nd device |
| 21:21.58 | Nuitari | instead of seeing that it can make a passthru |
| 21:23.00 | aksnowman | lolI've just gotten an asterisk box set up yesterday (softphones only) and can connect from my home network, but nobody outside of my network can actually stay connected. I've got the asterisk box set up as my dmz so it isn't an access problem. In the console I can see them connecting and whatnot, but then it just says "user hung up". They never even hear the automated messages. Any help would be appriciated |
| 21:23.12 | aksnowman | err, minus "lol" |
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| 21:24.02 | nextime | anyone using mISDN 1.2.0? |
| 21:25.15 | Nuitari | no |
| 21:30.02 | mvanbaak | no |
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| 21:41.46 | dijungal | hi guys.... |
| 21:42.12 | dijungal | can anyone recommend a good soft phone for linux |
| 21:42.16 | dijungal | ubuntu to be correct |
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| 21:44.30 | mvanbaak | ekiga |
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| 21:46.59 | dijungal | that sucks@ |
| 21:47.02 | dijungal | ! |
| 21:48.50 | GlobeTrotter | hi guys,, are there any special steps that i need to take when installing zaptel with * 1.4 and centos 5 ? |
| 21:49.24 | mvanbaak | GlobeTrotter: install debian first ;) |
| 21:49.29 | mvanbaak | sorry, that was not funny |
| 21:49.35 | GlobeTrotter | :) |
| 21:50.27 | GlobeTrotter | im having mucho problems getting ztdummy going on my box |
| 21:50.49 | Math` | mvanbaak: ahahaha it was :P |
| 21:50.59 | mvanbaak | lol |
| 21:51.00 | mvanbaak | :) |
| 21:52.22 | sergee | aksnowman: seems like you have issues with codecs, try to enable 711 everywhere |
| 21:53.06 | aksnowman | that would be a codec problem? we're using the same softphones |
| 21:54.14 | sergee | aksnowman: i don't know what that would be, i just assume |
| 21:54.41 | aksnowman | lol, ok |
| 21:54.48 | dijungal | do i have to install the zaptel drivers when installing asterisk if i'm only gonna use it for sip voip traffic |
| 21:54.57 | dijungal | ? |
| 21:55.31 | sergee | aksnowman: configure SIP (?) peers in asterisk with next: disallow=all allow=ulaw allow=alaw |
| 21:55.50 | [TK]D-Fender | aksnowman, : |
| 21:55.52 | [TK]D-Fender | ~sipnat |
| 21:55.53 | jbot | i guess sipnat is for for more information about configurtion of Asterisk with SIP behind NAT, see http://voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 21:55.55 | [TK]D-Fender | ^^^^^^ |
| 21:55.58 | sergee | dijungal: no you don't |
| 21:56.04 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 21:56.10 | dijungal | nice |
| 21:56.18 | sergee | dijungal: however |
| 21:56.22 | [TK]D-Fender | dijungal, You'll need it if you intend on doing IAX2 trunking or MeetMe conferences. |
| 21:56.36 | dijungal | ohooo... then i should probably put it in |
| 21:56.53 | dijungal | can i add it after the asterisk has already been installed? |
| 21:56.58 | sergee | dijungal: if you are planing to use conferences, or MeerMe, you will need ztdummy from zaptel |
| 21:57.04 | dijungal | k |
| 21:57.23 | sergee | conferences or IAX trunking :) |
| 21:58.17 | dijungal | k |
| 21:58.27 | dijungal | then i need to put in the ztdummy drivers |
| 21:58.38 | sergee | are you? |
| 21:59.53 | dijungal | i guess |
| 22:00.07 | sergee | from my experience, i wasn't able to get a good results with MeetMe + ztdummy, qulity of sound was very poor, so i baught digium interface card ... |
| 22:00.51 | *** join/#asterisk tosh| (n=tosh@213.219.169.70.adsl.dyn.edpnet.net) |
| 22:00.53 | dijungal | k |
| 22:01.09 | dijungal | so what do u suggest? |
| 22:01.58 | Math` | as far as I can recall MeetMe() only uses zaptel's sound mixing functions so interface card or not... you're using the same libs |
| 22:02.00 | sergee | dijungal: if you wish to use conferencing - buy digium card, or find someone who managed to get good results with ztdummy (i didn't found anybody) |
| 22:02.04 | [TK]D-Fender | dijungal, that you store that last comment in a dark place until you find that YOU'RE dissatisfied with it or not. |
| 22:02.29 | tosh| | I'm trying to use a trunk towards voipjet but when I add the extensions in extensions.conf, they do not show up in freePBX as being extensions. trying to call out give me a no route to destination errormsg. any idea on why freePBX doesn't recognize the conf lines as being a valid extension? |
| 22:02.35 | aksnowman | so my problem is that the outside clients are behind NAT's? |
| 22:02.48 | [TK]D-Fender | aksnowman, yes |
| 22:02.54 | aksnowman | ahh, k |
| 22:02.57 | dijungal | k |
| 22:02.57 | aksnowman | that makes sense |
| 22:03.09 | dijungal | i'll take my bet with the ztdummy |
| 22:03.16 | sergee | Math`: yes, but are there any non-interface cards from digium which provides timer? |
| 22:03.32 | `Sean | [TK]D-Fender do you know of a serivce provider that provides free tollfree termination like... TollFreeGateway?? |
| 22:04.10 | [TK]D-Fender | tosh|, freepbx does not READ your extensions and INTERPRET them. |
| 22:04.29 | [TK]D-Fender | aksnowman, there are a bunch of settings you'll have to make for this to work. |
| 22:04.54 | aksnowman | :( |
| 22:04.56 | [TK]D-Fender | tosh|, and.... |
| 22:04.58 | [TK]D-Fender | ~freepbx |
| 22:04.59 | jbot | freepbx is probably unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 22:05.17 | [TK]D-Fender | aksnowman, its not that big a dea to set up , really |
| 22:05.42 | aksnowman | k, I'm looking into it |
| 22:06.02 | blitzrage | russellb: hiiiiiI!!!!!!!!! |
| 22:06.05 | blitzrage | :) |
| 22:06.07 | russellb | omghi |
| 22:06.15 | blitzrage | omghiyourself |
| 22:06.16 | tosh| | [TK]D-Fender it was my understanding that it was a GUI for editing /etc/asterisk/*.conf |
| 22:06.36 | [TK]D-Fender | tosh|, you are mistaken. |
| 22:06.46 | Strom_M | tosh|: s/editing/murdering/ |
| 22:06.53 | blitzrage | russellb: is chan_mobile supposed to be an option in menuselect in trunk? |
| 22:07.01 | blitzrage | or is it part of chan_features or something... ? |
| 22:07.07 | `Sean | [TK]D-Fender do you know of a serivce provider that provides free tollfree termination like... TollFreeGateway?? |
| 22:07.07 | russellb | blitzrage: it's in addons, silly goose |
| 22:07.11 | blitzrage | ahhhhhh |
| 22:07.15 | blitzrage | for some reason I thought it was in trunk |
| 22:07.24 | russellb | well, trunk of addons |
| 22:07.29 | russellb | for licensing reasons |
| 22:07.35 | russellb | (uses a GPL lib) |
| 22:07.39 | blitzrage | gotcha |
| 22:07.40 | [TK]D-Fender | tosh|, it is a craptastic flaming pile of ^#%$ that lets you jump through hoops and build *'s configs based on its creator's tunnel-vision view of what a PBX should be. |
| 22:08.02 | `Sean | [TK]D-Fender do you know of a serivce provider that provides free tollfree termination like... TollFreeGateway?? |
| 22:08.04 | russellb | [TK]D-Fender: now tell him what you *really* think. |
| 22:08.06 | `Sean | you going to answer :P? |
| 22:08.27 | [TK]D-Fender | `Sean, I have never heard of them and aside from just repeating yourself like a broken record... you are now doing so directed at *ME*. |
| 22:08.29 | russellb | `Sean: it is rude to post the same question over and over ... |
| 22:08.40 | `Sean | i know russellb |
| 22:08.50 | `Sean | trying to find a provider like that asterlink is being gay YET once again |
| 22:08.52 | [TK]D-Fender | russellb, I was too vague... wasn't I? ;) |
| 22:08.57 | `Sean | and dropped support for 18** numbers |
| 22:09.05 | `Sean | so i need to find a provider to make 18** calls |
| 22:09.07 | sergee | only 2 buttons works at his kb: up and enter :) |
| 22:09.22 | russellb | `Sean: give nufone a try. |
| 22:09.35 | `Sean | i will after i finish up my credit with asterlink |
| 22:09.46 | `Sean | bkw is moron how is someone supposed to terminate all there calls |
| 22:09.48 | russellb | or voicepulse |
| 22:09.54 | `Sean | if they cant even get to dail a 18** number |
| 22:11.28 | mvanbaak | heya russellb |
| 22:11.30 | mvanbaak | :) |
| 22:12.30 | russellb | greetings |
| 22:12.44 | mvanbaak | howz u ? |
| 22:13.09 | blitzrage | there isn't an svn repo for addons I guess... ? (apparently I've never, ever used anything from addons :)) |
| 22:13.15 | [TK]D-Fender | ok, I'm out for a bit.... later all |
| 22:13.25 | *** join/#asterisk nephfl (n=nephfl@adsl-070-147-105-151.sip.gnv.bellsouth.net) |
| 22:13.59 | nephfl | hello, i cant seem to get my custom sounds to play in ivr |
| 22:14.10 | nephfl | can someone tell me where i'm going wrong? |
| 22:14.17 | blitzrage | probably the wrong format |
| 22:14.18 | mvanbaak | nephfl: what does the CLI say ? |
| 22:14.21 | nephfl | it will play with default sounds |
| 22:14.28 | blitzrage | make sure it's 8-bit, 8000Hz, mono |
| 22:15.59 | *** join/#asterisk DaveCanoe (n=Dave@belbrrcnas09-1088684470.dial.bell.ca) |
| 22:16.12 | *** join/#asterisk zeeesh (n=aadilism@202.125.143.65) |
| 22:16.13 | zeeesh | hi |
| 22:16.41 | nephfl | it was recorded in windows 16-bit 8khz mono...then i used sox file.wav file.gsm |
| 22:18.54 | *** join/#asterisk `Sauron (n=sauron@dsl001-130-033.aus1.dsl.speakeasy.net) |
| 22:19.14 | tosh| | anyone got any good pointers/tutorials on how to setup asterisk along with the voipjet service? I'm not getting through |
| 22:19.48 | blitzrage | nephfl: you still haven't converted to 8-bit then probably |
| 22:20.36 | Strom_M | blitzrage: 16-bit 8khz mono wav will work just fine |
| 22:20.41 | dijungal | i'm out |
| 22:20.53 | Strom_M | in fact, it's better - ulaw companding sounds better than linear 8-bit |
| 22:21.01 | dijungal | time to go drink some beers down the road :) |
| 22:21.23 | dijungal | and think about that ztdummy drivers :o) |
| 22:21.29 | blitzrage | Strom_M: strange... I had no idea Asterisk could read 16-bit |
| 22:21.36 | Strom_M | nephfl: don't convert to GSM unless you're using the GSM codec |
| 22:21.44 | *** part/#asterisk dijungal (n=kdaniel@209.59.110.15) |
| 22:21.44 | Strom_M | blitzrage: well it's all slin internally, right? |
| 22:21.49 | Strom_M | therefore, 16-bit 8khz |
| 22:22.04 | nephfl | i got it..it was just not set in the dialplan correctly |
| 22:22.10 | blitzrage | I don't pay attention too much to that part since it all "just works" for me :) |
| 22:22.17 | Strom_M | hehehehe |
| 22:22.18 | blitzrage | I spend more time on DB integration and such |
| 22:23.05 | GlobeTrotter | hi guys, im having problems compiling libpri.. * 1.4 /centos 5 |
| 22:23.09 | Strom_M | ok, time for breakfast |
| 22:23.34 | mvanbaak | I'm writing a chan_covide to interact with our webbased CRM-Groupware app |
| 22:23.38 | GlobeTrotter | i get this error CC=gcc ./mkdep -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g `ls *.c` |
| 22:23.41 | mvanbaak | why? "because I can" |
| 22:23.42 | GlobeTrotter | etc... |
| 22:24.41 | blitzrage | that doesn't look like an error |
| 22:25.22 | GlobeTrotter | ok thanks |
| 22:25.34 | blitzrage | why do you think you're getting an error? |
| 22:25.43 | GlobeTrotter | so how can i tell if libpri is installed ? |
| 22:26.01 | GlobeTrotter | well i'm having problems getting zaptel going |
| 22:26.18 | blitzrage | are you using T1/E1's? |
| 22:26.22 | GlobeTrotter | make linux62 will not work for me |
| 22:26.30 | GlobeTrotter | no i want to install ztdummy |
| 22:26.32 | blitzrage | you don't need to use that |
| 22:26.55 | GlobeTrotter | i can use make && make install ? |
| 22:27.01 | blitzrage | what version? |
| 22:27.18 | mvanbaak | 1.0.8 |
| 22:27.19 | mvanbaak | ;) |
| 22:27.48 | zeeesh | trying to run perl script at asterisk-1.4.4 getting too many these kind of lines "Can't locate LWP.pm in @INC (@INC contains: /usr/lib/perl5/site_perl/5.8.8/i386-linux-thread-multi "? |
| 22:28.01 | GlobeTrotter | zaptel-1.4.3 |
| 22:28.15 | blitzrage | make sure you run: ./configure ; make menuselect ; make install |
| 22:28.24 | nephfl | when using an analog trunk is there a way to reduce the amount of time between dialing an outside number and receiving a ringing tone? |
| 22:29.27 | tzafrir_laptop | zeeesh, apt-get install liblwp-perl ? |
| 22:30.12 | mvanbaak | tzafrir_laptop: you did not have that one? |
| 22:30.13 | zeeesh | let me check |
| 22:30.20 | mvanbaak | oh wait |
| 22:30.21 | mvanbaak | sorry |
| 22:30.23 | mvanbaak | ;) |
| 22:33.16 | zeeesh | <tzafrir_laptop>: i tried to find by this command "yum search liblwp-perl" as well as "liblwp" but get response "Reading repository metadata in from local files" "No Matches found"? |
| 22:33.34 | tzafrir_laptop | what distro is it? |
| 22:33.54 | zeeesh | RHEL |
| 22:34.26 | zeeesh | redhat enterprise using ... |
| 22:34.48 | zeeesh | using asterisk-1.4.4 .. |
| 22:34.55 | tzafrir_laptop | so debian package names won't help, I guess ... |
| 22:34.56 | blitzrage | use 1.4.5 |
| 22:35.21 | zeeesh | before that error i was getting msg ... about AGI.pm ... missing ... |
| 22:35.34 | zeeesh | i hv resolved then get this |
| 22:37.14 | *** join/#asterisk ReDNeQ- (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 22:37.26 | *** join/#asterisk iBuMp (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 22:37.37 | GlobeTrotter | he guys i am trying to get ztdummy going,, after i install it i run modprobe and get this error |
| 22:37.39 | GlobeTrotter | modprobe zaptel |
| 22:37.39 | GlobeTrotter | FATAL: Module zaptel not found. |
| 22:39.53 | GlobeTrotter | how can i verify that i have compiled and installed zaptel corectly?? |
| 22:40.04 | cy303 | sergee: you still around? |
| 22:40.04 | aksnowman | is SIP p2p in the server to client sense or like client to client to client sense? |
| 22:40.28 | mvanbaak | GlobeTrotter: did you run: depmod -ae |
| 22:40.58 | GlobeTrotter | yes, i just did,, it ran with no errors |
| 22:45.42 | aksnowman | nevermind, I see what's going on now :/ |
| 22:46.14 | *** join/#asterisk kombi (n=kombi@213.160.14.18) |
| 22:47.57 | kombi | x-lite registers fine but asterisk doesn't (as a sip client for a sip provider) it might be the old "auth" vs. auth but all googlable patches are heavily outdated. Must I now sniff packets and see what the difference realy is? |
| 22:48.19 | kombi | s/realy/really/ |
| 22:48.35 | kombi | holy crap.. bot! |
| 22:48.43 | *** join/#asterisk mazpe (n=email@adsl-074-173-243-244.sip.bct.bellsouth.net) |
| 22:49.29 | mazpe | where can do i find the file recordings? |
| 22:49.34 | kombi | nobody is here anywhere, I'll go get drunk instead then (or is there?) |
| 22:49.45 | kombi | what are the file recordings? |
| 22:50.03 | GlobeTrotter | hi guys,, can anyone help me with my zap problem? |
| 22:50.22 | kombi | drinking it is.. |
| 22:52.18 | mazpe | is there a way to set a setting so that at anytime from any IVR option or even Voicemail i can go back to the main IVR? |
| 22:53.44 | *** part/#asterisk nextime (n=nextime@unaffiliated/nextime) |
| 22:54.33 | cy303 | Ugh |
| 22:54.58 | cy303 | Can anyone suggest a way to run a command after a meetme conference has ended? |
| 22:55.08 | aksnowman | is there a way I could set up asterisk to use softphones with a protocol that would work with clients behind NAT's? |
| 22:55.31 | aksnowman | ie not SIP |
| 22:55.50 | cy303 | something like this: |
| 22:55.51 | cy303 | exten => s,n,MeetmMe($ARG1},rMS) |
| 22:55.51 | cy303 | exten => s,n,System(cmd) |
| 22:56.09 | cy303 | unfortunately the System application is never called once the meetme conference ends |
| 22:57.06 | justdave | the stuff after MeetMe() in the same context/extension is only run if the user is kicked from the conference |
| 22:57.17 | justdave | (or if you have that code enabled to allow people to exit manually) |
| 22:58.57 | justdave | h extension in the same context might get run when the conference ends (since it hangs up on the caller) |
| 23:01.00 | ber111 | is there a 180 minute limit somehow hardcoded in meetme |
| 23:02.34 | cy303 | justdave: not having any luck :( |
| 23:08.40 | justdave | ber111: doubt it. I've had people forget to hang up their phones and stay in a meetme room for a couple days before :) |
| 23:13.34 | *** join/#asterisk antlers (n=antlers@ip70-173-89-173.lv.lv.cox.net) |
| 23:13.48 | tzafrir_laptop | GlobeTrotter, still problems? |
| 23:14.15 | tzafrir_laptop | what error have you got? |
| 23:17.00 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 23:17.00 | *** mode/#asterisk [+o anthm] by ChanServ |
| 23:20.09 | GlobeTrotter | hi tzafrir_laptop |
| 23:20.27 | GlobeTrotter | yes i cant get ztdummy install |
| 23:21.42 | GlobeTrotter | its seems to compile correctly. but when i modprobe i get an error |
| 23:21.43 | GlobeTrotter | modprobe zaptel |
| 23:21.43 | GlobeTrotter | FATAL: Module zaptel not found. |
| 23:25.43 | *** join/#asterisk elg (n=fugalh@216.31.27.110) |
| 23:26.11 | ManxPower | GlobeTrotter: copy it from the directory in /lib/modules that the install script instaled it into and move it to the correct place. |
| 23:26.31 | tzafrir_laptop | find /lib/modules -name zaptel.ko |
| 23:26.50 | tzafrir_laptop | that can copy it to the wrong dir |
| 23:27.07 | tzafrir_laptop | better insmod it directly... |
| 23:29.07 | GlobeTrotter | i found it here |
| 23:29.20 | GlobeTrotter | --/lib/modules/2.6.18-8.1.6.el5/misc/zaptel.ko |
| 23:30.03 | ManxPower | GlobeTrotter: and 'uname -a' is not returning that kernel version. |
| 23:30.37 | GlobeTrotter | uname returns |
| 23:30.39 | GlobeTrotter | Linux localhost.localdomain 2.6.18-8.el5 #1 SMP Thu Mar 15 19:46:53 EDT 2007 x86_64 x86_64 x86_64 GNU/Linux |
| 23:31.11 | *** join/#asterisk christopherwhull (n=50114830@24.111.130.38) |
| 23:31.15 | ManxPower | and I'll bet you have a /lib/modules/2.6.18-8.el5 directory and that is where the module needs to be copied to. |
| 23:31.41 | ManxPower | unless your kernel source tree does not match your current kernel, but in that case you have much bigger problems. |
| 23:32.21 | GlobeTrotter | how do i verify if they match? |
| 23:41.40 | tzafrir_laptop | GlobeTrotter, what's the output of 'uname -r' ? |
| 23:41.55 | *** join/#asterisk trinary (n=dpb@www.daves.net) |
| 23:42.02 | tzafrir_laptop | ah.. ok |
| 23:42.03 | GlobeTrotter | 2.6.18-8.el5 |
| 23:42.30 | tzafrir_laptop | so it has been built for a different kernel |
| 23:43.18 | tzafrir_laptop | rpm -qa | grep kernel |
| 23:43.48 | GlobeTrotter | kernel-2.6.18-8.el5 |
| 23:43.48 | GlobeTrotter | kernel-headers-2.6.18-8.el5 |
| 23:43.48 | GlobeTrotter | kernel-devel-2.6.18-8.1.6.el5 |
| 23:46.01 | GlobeTrotter | you mean that zaptel has been built for a diffrent kernel? |
| 23:47.08 | aksnowman | that version of zaptel is built for the latest dev kernel, you're running the latest stable, that's my understanding |
| 23:48.44 | GlobeTrotter | i downloaded this one |
| 23:48.45 | GlobeTrotter | http://ftp.digium.com/pub/zaptel/zaptel-1.4.3.tar.gz |
| 23:49.00 | GlobeTrotter | do i need this one |
| 23:49.01 | GlobeTrotter | zaptel-1.4.3.tar.gz |
| 23:49.09 | GlobeTrotter | http://ftp.digium.com/pub/zaptel/zaptel-1.4-current.tar.gz |
| 23:49.10 | GlobeTrotter | ? |
| 23:49.25 | tzafrir_laptop | your kernel-devel and kernel packages do not match |
| 23:49.53 | tzafrir_laptop | Note the "8" vs. the "8.1" |
| 23:50.00 | tzafrir_laptop | in the version number |
| 23:50.14 | *** join/#asterisk Cyber-Dogg (n=jphelps@24-178-240-97.dhcp.stls.mo.charter.com) |
| 23:50.22 | GlobeTrotter | ah ok,, i see,,, where can i ge the version that i need? |
| 23:50.37 | Cyber-Dogg | so... I'm trying to get two servers setup to work togethe riwth IAX |
| 23:50.37 | tzafrir_laptop | This is part of the "kernel version string", which is also the name of the directory under /lib/modules/ |
| 23:50.49 | tzafrir_laptop | Modules get loaded automatically from that directory |
| 23:51.04 | Cyber-Dogg | are there any good tutorials on how to setup IAX? |
| 23:51.11 | Cyber-Dogg | I don't need dynamic capabilites |
| 23:51.17 | Cyber-Dogg | I have static addresses on both systems |
| 23:51.32 | tzafrir_laptop | either downgrade the headers package (possible?) or upgrade the kernel package and reboot |
| 23:51.36 | GlobeTrotter | under my /lib/modules and have these three directories |
| 23:51.37 | GlobeTrotter | 2.6.182.6.18-8.el5 2.6.18-8.1.6.el5 2.6.18-8.el5 |
| 23:51.48 | GlobeTrotter | is that incorrect? |
| 23:52.37 | tzafrir_laptop | the sample config files have examples for that, IIRC |
| 23:52.51 | tzafrir_laptop | Cyber-Dogg, google a bit |
| 23:53.01 | tzafrir_laptop | tons of examples |
| 23:53.28 | Cyber-Dogg | I found a lot for dynamic... with users... but I need static peers |
| 23:53.54 | zeeesh | can i make call conference without using zap channel ? |
| 23:53.57 | tzafrir_laptop | so? |
| 23:54.02 | tzafrir_laptop | adjust a bit |
| 23:54.11 | Cyber-Dogg | I don't know enough to do that... yet |
| 23:54.34 | tzafrir_laptop | for one direction you have to specifiy the address anyway |
| 23:55.15 | tzafrir_laptop | leave it dynamic for starters. That's easier to see when the other side has successfully registered and thu is available |
| 23:56.10 | Cyber-Dogg | ok |
| 23:56.29 | Cyber-Dogg | so... I haven't set anything up yet... extensions or iax... |
| 23:56.42 | Cyber-Dogg | I just installed asterisk and setup the zaptel configuration |
| 23:56.57 | Cyber-Dogg | do I only have to configure extensions and iax to get this to work? |