| 00:02.08 | *** join/#asterisk Agrajag- (n=filip@c211-30-185-177.artrmn2.nsw.optusnet.com.au) |
| 00:03.34 | *** join/#asterisk shazaum (n=889uJWER@unaffiliated/shazaum) |
| 00:03.52 | ScarEye_ | Quick question I downloaded asterisk-1.6.0.10.tar.gz but I don't see asterisk-addons-1.6.0.10.tar.gz ? Anyone know where I can download this from? |
| 00:04.38 | Qwell | ScarEye_: the versions don't match exactly. |
| 00:04.54 | Qwell | the latest asterisk-addons for 1.1.0 is 1.6.0.3 |
| 00:05.02 | Qwell | s/1.1/1.6/ |
| 00:05.57 | ScarEye_ | thank you guys I appreciate your time. |
| 00:08.06 | Agrajag- | g'day. is there a common way that people use to let the answerer of a call know what ivr buttons the caller has pressed to get to them? e.g. sending a SIP message or something - but i can't see how to easily do this, especially when in Dial() i have made it ring multiple SIP phones |
| 00:22.40 | *** join/#asterisk howie (n=howie@71-95-220-206.dhcp.mtpk.ca.charter.com) |
| 00:22.46 | *** join/#asterisk Farfadet[Matteo] (n=xinstinc@bas1-montreal02-1096727332.dsl.bell.ca) |
| 00:23.27 | [TK]D-Fender | Agrajag-: A(), M(), or mod the CID |
| 00:24.11 | Farfadet[Matteo] | hello. i search a numbers did canadian for my ipbx and skype. |
| 00:25.39 | [TK]D-Fender | Farfadet[Matteo]: voip.ms |
| 00:27.11 | *** join/#asterisk plundra (i=404@article.se) |
| 00:27.30 | Farfadet[Matteo] | thanks |
| 00:27.42 | Orbixx | Why would Asterisk stop at parsing the Festival config and go no further? (it connects to the Festival server but immediately disconnects without issuing any command) |
| 00:36.00 | *** join/#asterisk moy (n=moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com) |
| 00:41.35 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 00:42.08 | *** join/#asterisk bobJR (n=bob@adsl-150-237-62.tys.bellsouth.net) |
| 01:01.35 | *** join/#asterisk Juggie (n=Juggie@CPE001601df17fb-CM001ceac25ada.cpe.net.cable.rogers.com) |
| 01:02.56 | *** join/#asterisk coppice (n=chatzill@193.194.17.210.dyn.pacific.net.hk) |
| 01:04.55 | *** join/#asterisk e4 (n=e4@76.84.81.72) |
| 01:05.06 | e4 | Any recommendations on in-depth Asterisk books? |
| 01:05.16 | [TK]D-Fender | ~book |
| 01:05.17 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 01:05.49 | e4 | Book's awesome, but already read it :) |
| 01:07.52 | e4 | Looking for something that goes into further depth, if anything further exists in booko form. |
| 01:08.17 | [TK]D-Fender | e4: * doesn't have "depth". * has tons of extraneous bits that aren't that special, just different from each other |
| 01:08.32 | [TK]D-Fender | e4: What are you expecting? |
| 01:08.48 | mwalling | where do you want to go today? |
| 01:08.52 | mwalling | cues the drum intro |
| 01:10.53 | e4 | Just wondered if there's more out there. Reading through that book was enough to get a system installed and working, but doing things like debugging degrading call quality, troubleshooting, that sort of thing. |
| 01:14.39 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 01:17.26 | e4 | Just not sure where to go to take it to the next step, want to learn more and not sure how to do it. |
| 01:18.53 | *** join/#asterisk jfsd (n=jfsd@66.241.82.207) |
| 01:22.05 | jfsd | hey there. who supports libiax2? |
| 01:25.36 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/session) |
| 01:25.43 | *** join/#asterisk Faiz (n=otakucon@c-98-221-51-177.hsd1.nj.comcast.net) |
| 01:27.52 | drmessano | Holy crap, Riker was here! |
| 01:28.23 | Faiz | issues with res_fax.so not being loaded by asterisk, "undefined symbol: ao2_lock", is there anything within the .conf files that need to be configured to resolve this? |
| 01:30.00 | *** join/#asterisk MikeJ (n=MikeJ@freeswitch/developer/mikej) |
| 01:30.09 | MikeJ | ~thebook |
| 01:30.11 | infobot | methinks thebook is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://astbook.asteriskdocs.org |
| 01:32.08 | *** join/#asterisk TrentCreek (n=kvirc@adsl-99-20-31-130.dsl.hrlntx.sbcglobal.net) |
| 01:34.24 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 01:38.42 | *** join/#asterisk sjobeck (n=Adium@208-151-246-17.dq1sn.easystreet.com) |
| 01:39.34 | sjobeck | any one a master on openSUSE 11.1 out there? i have compiled like a million times on this thing before but tonight for some i cant, what am i missing? i am getting this: make[4]: *** [/usr/src/zaptel-1.2.27/zaptel-base.o] Error 1 |
| 01:41.07 | WindowsUser | paste bin that and a few lines above it |
| 01:41.25 | WindowsUser | make does omg omg omg omg omg omg the world has ended |
| 01:41.40 | WindowsUser | make[1]: ** error wil robinson |
| 01:41.47 | MikeJ | sjobeck: what is the real error |
| 01:41.49 | MikeJ | should be above that |
| 01:41.54 | sjobeck | windowsuser: coming right up |
| 01:42.56 | sjobeck | http://sjobeck.pastebin.com/d25c41e0 |
| 01:43.34 | sjobeck | thx for the hand guys. appreciate it. i think my eyes are crossed at this point or something. i am just not seeing it. |
| 01:43.40 | *** join/#asterisk OrNix (n=ornix@l151-249-47.static.cn.ru) |
| 01:45.59 | WindowsUser | looks like the version of zaptel isn't matching with the version of kernel includes |
| 01:47.04 | sjobeck | yeah, old version of zaptel (1.2.27) & asterisk (1.2.6) on this box. that is my next project on this box is to get those updated, but was hoping to not do that now, since we're down. |
| 01:48.03 | sjobeck | am i jammed? must i go to 1.4 ? |
| 01:48.52 | *** join/#asterisk Deeewayne (n=dwayne@76.29.245.9) |
| 01:48.52 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 01:48.53 | WindowsUser | well why did the kernel get upgraded? |
| 01:49.06 | sjobeck | an unrelated storage issues |
| 01:49.08 | russellb | the problem is that version zaptel is freaking ancient |
| 01:49.21 | russellb | and good lord, asterisk too |
| 01:49.22 | russellb | heh |
| 01:49.22 | sjobeck | :) |
| 01:49.24 | sjobeck | well said |
| 01:49.29 | sjobeck | i know . i know. |
| 01:49.40 | sjobeck | so, it sounds like i am forced to do 1.4, eh? |
| 01:49.55 | WindowsUser | might be able to get away with a newer 1.2 |
| 01:49.57 | sjobeck | which i am delighted to do, and was going to, just wanted to get home tonight |
| 01:50.07 | russellb | well, and if you're doing that, you should use DAHDi |
| 01:50.10 | sjobeck | dont i have latest zaptel 1.2.x? |
| 01:50.10 | russellb | and not zaptel at all ... |
| 01:50.20 | WindowsUser | ~zaptel |
| 01:50.21 | infobot | hmm... zaptel is zapata telephony interface. A low level interface designed to abstract hardware access to a variety of devices for BRI, PRI or analogue access. a phone card company |
| 01:50.22 | sjobeck | dahdi is ast1.4.x only though yes? |
| 01:50.24 | WindowsUser | awwwww |
| 01:50.35 | sjobeck | someone turn off that bot |
| 01:50.36 | russellb | sjobeck: that is correct. but i don't think you can use zaptel 1.4 with asterisk 1.2 |
| 01:50.37 | sjobeck | :) |
| 01:50.55 | WindowsUser | sjobeck: i was hoping it'd be like "lastest zaptel is xxxx" |
| 01:50.55 | sjobeck | i dont either, but i have no idea, never tried |
| 01:51.02 | russellb | or ... well ... you can try, but i won't be surprised if it doesn't work i guess |
| 01:52.41 | sjobeck | errrr. grrrrr. i stepped in it. i think. bummer. i think i might have to do 1.4 & hope & pray & finger cross. i dont enjoy that. |
| 01:53.00 | sjobeck | what the heck, i'll try dahdi for grins |
| 01:53.58 | [TK]D-Fender | I'd recommend 1.6.0 |
| 01:54.17 | sjobeck | zoinks ! from 1.2 to 1.6 in one night. you have a lot of faith . |
| 01:54.20 | [TK]D-Fender | Save yourself watching 1.4 get EOL'd like it's due very soon |
| 01:54.21 | sjobeck | (in me?) |
| 01:54.41 | sjobeck | well, there is a LOT going on, on this box, i'll break zads of stuff for sure |
| 01:54.49 | sjobeck | but i get your pt |
| 01:55.20 | WindowsUser | horray for a does everything bo |
| 01:55.58 | WindowsUser | actually today at a bubble tea place i saw a desktop pc on a shelf near the cielin apparently acting as a security camera recorder |
| 01:57.06 | *** join/#asterisk ArchGT (n=ArchGT@unaffiliated/archgt) |
| 01:59.37 | sjobeck | hmmm, looks like it all went on ok but the daemon startup script. there is no /etc/init.d/dahdi |
| 01:59.44 | sjobeck | ah, right, suse ..... |
| 01:59.45 | sjobeck | right ? |
| 01:59.58 | sjobeck | how i start this turkey by hand ? |
| 02:00.19 | sjobeck | who chose suse on this box any way 5 yrs ago ?! |
| 02:00.26 | sjobeck | :) |
| 02:07.30 | Faiz | issues with res_fax.so not being loaded by asterisk, "undefined symbol: ao2_lock", is there anything within the .conf files that need to be configured to resolve this? |
| 02:09.05 | Deeewayne | Faiz, you are probably using the wrong version of res_fax.so for the version of asterisk that you are using |
| 02:09.37 | Faiz | indeed i didn't leave out any possible options |
| 02:09.41 | Faiz | and tested even with version 1.4 |
| 02:09.53 | Faiz | and tested with various other processor optimized builds |
| 02:09.55 | Deeewayne | are you using the 1.4 version of res_fax ? |
| 02:10.23 | Faiz | no, my version of asterisk is 1.6.1.1 |
| 02:10.43 | Deeewayne | ok, where did you get the res_fax for asterisk 1.6.1.1 ? |
| 02:11.08 | Faiz | using: http://www.digium.com/en/docs/FAX/faa-download.php |
| 02:11.15 | Faiz | Asterisk Version: 1.6.1.0 - 1.6.1.2 |
| 02:11.25 | *** part/#asterisk MikeJ (n=MikeJ@freeswitch/developer/mikej) |
| 02:11.50 | Faiz | for Flavor (recommended by Bandfax) for Pentium4m |
| 02:13.04 | Deeewayne | try http://downloads.digium.com/pub/telephony/fax/res_fax/asterisk-1.6.1/x86-32/res_fax-1.6.1_1.0.11-x86_32.tar.gz |
| 02:13.16 | [TK]D-Fender | ~book |
| 02:13.17 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 02:13.19 | Faiz | will do, thanks |
| 02:13.24 | [TK]D-Fender | ~jerjerguide |
| 02:13.25 | infobot | [~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * : http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
| 02:13.37 | Deeewayne | Faiz, I dont know what Bandfax is ... sorry |
| 02:16.13 | Faiz | ah, same problem sir |
| 02:16.38 | Faiz | Bandfax is the utility discussed in the admin's manual to find the best optimum module to run the fax service within asterisk |
| 02:16.53 | *** join/#asterisk Defraz (n=T0tal@24-117-156-215.cpe.cableone.net) |
| 02:16.54 | Deeewayne | oh, I think its benchfax. I have never used it |
| 02:17.35 | Faiz | oh im sorry |
| 02:17.39 | Faiz | my apologies |
| 02:17.42 | Deeewayne | no problem |
| 02:18.00 | Faiz | but regarding the file you just linked, i still have the same issue |
| 02:18.02 | Faiz | any suggestions? |
| 02:18.21 | *** join/#asterisk jeff (i=jeff@unaffiliated/jeff) [NETSPLIT VICTIM] |
| 02:18.21 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) [NETSPLIT VICTIM] |
| 02:19.43 | Deeewayne | Faiz, one sec |
| 02:19.52 | Faiz | sorry for rushing you, thanks :) |
| 02:20.40 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 02:21.00 | Deeewayne | do you require 1.6.1 ? |
| 02:21.15 | Faiz | i believe so, my asterisk version is 1.6.1.1 and dahdi is 2.2.0.2 |
| 02:21.31 | Deeewayne | I mean, do you require asterisk 1.6.1 |
| 02:21.34 | Faiz | i haven't updated asterisk with respect to svn (forgot how to :/) |
| 02:21.35 | Faiz | oh, no |
| 02:22.44 | Deeewayne | Faiz, http://downloads.digium.com/pub/telephony/fax/res_fax/asterisk-1.6.0/x86-32/res_fax-1.6.0_1.0.11-x86_32.tar.gz loads for me w/ Asterisk 1.6.0.6 |
| 02:24.45 | Faiz | argh, same issue.. |
| 02:24.54 | Faiz | do you think its asterisk 1.6.1.1 that's causing the issue? |
| 02:24.58 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 02:24.58 | leifmadsen | Deeewayne: !!! |
| 02:25.29 | leifmadsen | Faiz: what is the issue with benchfax? |
| 02:25.35 | Deeewayne | leifmadsen, howdy |
| 02:25.47 | Faiz | hey leif, it's not benchfax, its res_fax.so in particular |
| 02:26.00 | leifmadsen | what is the issue? |
| 02:26.03 | Faiz | when I load res_fax.so by issuing "module load res_fax.so" in the CLI, it says: |
| 02:26.07 | leifmadsen | can't scroll up |
| 02:26.08 | Deeewayne | Faiz, did you switch to Asterisk 1.6.0.6, rebuild Asterisk, download http://downloads.digium.com/pub/telephony/fax/res_fax/asterisk-1.6.0/x86-32/res_fax-1.6.0_1.0.11-x86_32.tar.gz, and try to load it ? |
| 02:26.18 | Faiz | loader.c:375 load_dynamic_module: Error loading module 'res_fax.so': /usr/lib/asterisk/modules/res_fax.so: undefined symbol: ao2_ref |
| 02:26.28 | Faiz | i haven't tried that |
| 02:26.31 | leifmadsen | note that a LOT of changes with faxing have occured lately |
| 02:26.43 | Faiz | oh |
| 02:26.44 | leifmadsen | try latest 1.6.0 or 1.6.1 from branches |
| 02:27.00 | Faiz | update asterisk branch via svn? |
| 02:27.03 | leifmadsen | RC's will be out very soon |
| 02:27.15 | leifmadsen | svn co http://svn.asterisk.org/svn/asterisk/branches/1.6.0 |
| 02:27.17 | Deeewayne | svn switch http://svn.digium.com/svn/asterisk/tags/1.6.0.6 |
| 02:27.29 | Deeewayne | or listen to leifmadsen :-) |
| 02:27.32 | leifmadsen | heh :) |
| 02:27.37 | Faiz | ah, i figured as much |
| 02:27.44 | Faiz | i'll load that |
| 02:27.44 | Deeewayne | I'm a svn switch fan |
| 02:28.12 | Faiz | "svn switch" changes the current branch to another branch? |
| 02:28.39 | Orbixx | Ok, why isn't Asterisk paying attention to DTMF when input is greater than one digit? |
| 02:28.51 | leifmadsen | I have no idea |
| 02:29.04 | leifmadsen | provide some information which may allow people to help debug the issue |
| 02:29.14 | Orbixx | What sort of information would be required? |
| 02:29.15 | leifmadsen | has never heard of svn switch |
| 02:29.40 | leifmadsen | Orbixx: console output, configuration, what you're trying to do, technologies you're using, DTMF output, etc...... |
| 02:29.45 | Faiz | heheh |
| 02:29.50 | Deeewayne | Faiz, yeah it changes your working copy to another branch/tag/URL |
| 02:29.55 | leifmadsen | you haven't really given us anything other than a vague statement |
| 02:30.37 | Faiz | ok, i may try that for the time being and possibly switch back to the current ast. branch |
| 02:30.49 | Faiz | i assume it doesn't harm or reconfigure any existing .conf files, correct? |
| 02:31.10 | leifmadsen | correct |
| 02:31.16 | leifmadsen | unless you run 'make samples' |
| 02:31.23 | Faiz | hah, certainly not |
| 02:31.31 | Deeewayne | you can try a more recent tag, but I just happen to have 1.6.0.6 built and I tried the 1.6.0 res_fax and it loaded ... that's the only reason I'm suggesting that version |
| 02:31.56 | sirus | Hey anyone got a sip phone I can test? I want to see if I can add you to my dialplan and be able to call you |
| 02:32.00 | Faiz | yeah, i'll give it a try and then switch back possibly when the new tags come out |
| 02:32.08 | Faiz | thanks a million :) |
| 02:33.07 | Deeewayne | leifmadsen, doing anything interesting ? |
| 02:35.41 | *** join/#asterisk sjobeck (n=Adium@32.158.34.121) |
| 02:36.25 | sjobeck | sorry, guys, had to change locations, what did i miss? |
| 02:37.30 | sjobeck | did you guys know it was 104 in Portland today ?! omg. |
| 02:37.46 | leifmadsen | Deeewayne: right now? not really :) |
| 02:37.54 | leifmadsen | it was like 20C here |
| 02:38.09 | sjobeck | true my northen friend |
| 02:38.20 | sjobeck | scorching |
| 02:38.34 | Deeewayne | I heard it was 107 when I called a friend and he was on his kayak |
| 02:39.10 | leifmadsen | sounds fun |
| 02:39.23 | Deeewayne | yeah. I worked all day :-( |
| 02:39.45 | jaytee | yeah, but it's a "dry" heat :-) |
| 02:40.24 | Deeewayne | Elaine is talking about her diaphragm to Jerry's virgin girlfriend |
| 02:40.25 | leifmadsen | heh |
| 02:40.34 | leifmadsen | nice, good episode |
| 02:41.06 | jaytee | I love the episode where Elaine dances at her office party |
| 02:41.27 | Deeewayne | yeah. great dance :) |
| 02:41.41 | jaytee | actually I love all the Seinfeld episodes. |
| 02:45.11 | *** part/#asterisk jfsd (n=jfsd@66.241.82.207) |
| 02:46.30 | Deeewayne | trips file |
| 02:47.08 | file | falls |
| 02:49.27 | leifmadsen | kicks file |
| 03:06.14 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 03:08.25 | Faiz | Hey, Deeewayne, are you still around? |
| 03:10.25 | Deeewayne | yeah, what's up ? |
| 03:15.50 | Faiz | hey, |
| 03:15.57 | Faiz | i'm trying to switch over tags |
| 03:16.11 | Faiz | but when i run svn sw http://svn.digium.com/svn/asterisk/tags/1.6.0.6-rc1/ |
| 03:16.24 | Faiz | i get this message: "svn: '.' is not a working copy" |
| 03:29.24 | Deeewayne | Faiz, 1) are you in the working directory when you do the switch? 2) don't use an -rc when the non-rc (1.6.0.6) is available |
| 03:29.40 | Deeewayne | if you are having a problem doing a switch, just do a new checkout |
| 03:30.37 | Faiz | ok |
| 03:30.44 | Faiz | the working directory would be, i assume |
| 03:30.59 | Faiz | cd /usr/src/asterisk.... |
| 03:31.36 | Deeewayne | whenever you previously did a svn checkout |
| 03:33.03 | Faiz | hm, |
| 03:33.14 | Faiz | which is what i had initially thought, but it keeps giving me the same error |
| 03:33.19 | Faiz | i'll make a new co |
| 03:37.41 | Faiz | hey, by any chance will you be around tomorrow? |
| 03:39.47 | *** join/#asterisk propellerhead (n=yogurt2u@host66.190-31-153.telecom.net.ar) |
| 03:42.20 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 03:44.14 | *** join/#asterisk trnzmeta (n=bleh@secure27.lnk.telstra.net) |
| 03:46.47 | *** join/#asterisk velusip (n=velusip@65.38.42.178) |
| 03:53.43 | *** join/#asterisk raden (n=chatzill@68-191-169-170.dhcp.stpt.wi.charter.com) |
| 03:54.34 | raden | anyone had problems with netgear FVX538 ???? as far as nat |
| 03:56.01 | [TK]D-Fender | radenDon't know that one specifically, but I've heard that several have posed NAT issues |
| 03:56.25 | [TK]D-Fender | raden: Verify your forwarding and * settings are correct and that's what's left... |
| 03:59.29 | raden | i forwarded 5060 and 10000 - 20000 udp right to asterisk box |
| 03:59.43 | raden | created a service group |
| 04:00.03 | raden | still having same issue i can call out fine besides the call quality is freaking crap |
| 04:00.26 | raden | and when i call in the phone rings pick it up no one there on the phone calling in it keeps ringing |
| 04:02.46 | raden | very frusterated |
| 04:04.16 | [TK]D-Fender | raden : that sounds like a bad NAT setup. PB a failed call w/ SIP debug enabled and your sip.conf masking only passwords |
| 04:04.24 | raden | [TK]D-Fender: when a inbound call comes in asterisk hands the connection off to the ITSP and the phone the connection doesnt go through asterisk does it ? |
| 04:04.30 | [TK]D-Fender | ~pb |
| 04:04.31 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/, or apt-get install pastebinit |
| 04:04.47 | [TK]D-Fender | raden : Phone should be passing through * if you did it right |
| 04:04.49 | raden | [TK]D-Fender: yes been through all this I not at office at home brainstormin |
| 04:05.12 | raden | how could the nat setup be bad |
| 04:05.20 | [TK]D-Fender | raden :Then you might want to consider asking for help when you're capable of folling it ;) |
| 04:05.27 | raden | i have had 3 people in here help me go ever everything and everyone says must be nat |
| 04:05.29 | [TK]D-Fender | following* |
| 04:05.52 | [TK]D-Fender | raden: Well I'm not "3 people", and I don't comment on things I don't see. |
| 04:06.10 | raden | i understand im just trying to get some logic to this |
| 04:06.34 | [TK]D-Fender | radenShowing me real details would be logical.... |
| 04:09.07 | raden | is there a way to search for pastebin posts |
| 04:15.19 | raden | guess not |
| 04:19.58 | *** join/#asterisk iamamoron (n=iamamoro@210.238.181.188) |
| 04:20.01 | iamamoron | hi there |
| 04:20.14 | iamamoron | any ideas where to buy SIP speaker? |
| 04:20.37 | iamamoron | I am planning to implement paging system on asterisk |
| 04:21.35 | iamamoron | ? |
| 04:21.39 | trnzmeta | sip speaker = tts (text to speech)??? |
| 04:23.17 | [TK]D-Fender | iamamoron: Describe where you'd put it |
| 04:23.48 | iamamoron | just like a paging system |
| 04:24.06 | iamamoron | wherein i could setup meetme then it will connect to all speakers |
| 04:24.14 | iamamoron | and broadcast the voice |
| 04:24.21 | iamamoron | would it be possible? |
| 04:25.11 | drmessano | Have you googled for it? |
| 04:25.14 | [TK]D-Fender | iamamoron: of course, and plenty of options. |
| 04:25.20 | drmessano | Theres quite a few out there |
| 04:25.30 | [TK]D-Fender | iamamoron: and I asked WHERE you install it. Describe the physical room. |
| 04:25.50 | iamamoron | it is a big bricktile factory |
| 04:26.01 | iamamoron | thats why I will be needing amplifiers |
| 04:26.04 | drmessano | [TK]D-Fender: Padded walls, no sharp objects, bright colors |
| 04:26.15 | [TK]D-Fender | iamamoron: ok, thats the OUTER walls. where inside? In a large production floor? |
| 04:26.26 | [TK]D-Fender | drmessano: walllllllssss SOFTTTTTTTT!!!!! |
| 04:26.27 | iamamoron | bottom line is, the main office can broadcast voice to any points of speakers |
| 04:26.31 | [TK]D-Fender | pets the walls |
| 04:26.40 | iamamoron | i need to hang it |
| 04:26.41 | drmessano | licks the walls |
| 04:26.49 | iamamoron | factory is already there |
| 04:27.03 | *** join/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
| 04:27.09 | [TK]D-Fender | iamamoron: Please answer the question... |
| 04:27.20 | iamamoron | i just create a place for speakers to be hang and amplified so that voice can get through |
| 04:27.22 | iamamoron | any ideas? |
| 04:27.39 | iamamoron | large production floor |
| 04:27.43 | iamamoron | it mix actually |
| 04:28.13 | [TK]D-Fender | iamamoron: Go lookup Viking Pagings amplifiers. Tehy have an FXS or FXo interface you'd attact to an ATA |
| 04:33.27 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 04:34.37 | iamamoron | does it have a reseller in asia? |
| 04:34.46 | iamamoron | i am in japan actually |
| 04:35.46 | raden | thats why your a moron |
| 04:37.13 | *** join/#asterisk voxter (n=voxter@190.241.15.56) |
| 04:38.39 | raden | lol j/k anyway just import it |
| 04:41.19 | raden | I dont understand how a inbound call can ring a phone but not make a bi-directional audio connection |
| 04:41.41 | *** join/#asterisk lolo2 (n=bob@c-69-180-160-4.hsd1.mn.comcast.net) |
| 04:42.52 | lolo2 | can someone tell me what to add to exctentions to have context call the demo context? |
| 04:43.22 | Faiz | ~book |
| 04:43.23 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 04:44.01 | lolo2 | i have the book open can you tell me what to look under? |
| 04:44.20 | [TK]D-Fender | lolo2: You don't call a context. You call LANDS in a contex targeting an extension that matches what you dialed |
| 04:45.30 | lolo2 | how can i "trnsfer" to the demo context form a diffrent context. I am mostly likly using the wrong vocabluary sorry for my bad english |
| 04:48.56 | lolo2 | My goal is to be in [default] and somehow go back to [demo] ? |
| 04:51.49 | lolo2 | do i use Goto? |
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| 04:55.09 | carrar | http://tinyurl.com/nqyy8q |
| 04:58.03 | WindowsUser | lolo2: sounds right |
| 04:58.13 | lolo2 | thanks :) |
| 05:06.57 | [TK]D-Fender | lolo2: Goto is a though.. but you don't jsut goto a context, you go to a specific extension & priority |
| 05:16.55 | lolo2 | exten => _17474,n,Goto(demo,s,1) worked for me thanks! |
| 05:17.22 | lolo2 | i dont know if i did it "correct" but it worked |
| 05:17.41 | [TK]D-Fender | lolo2: Thats a valid looking Goto |
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| 05:35.16 | iamamoron | does grandstream phones have speaker out? |
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| 06:02.32 | Micc | I'm having problems with some ploycom phones behind a sonicwall |
| 06:02.56 | Micc | I've set the expire to 120 and it still ends up unreachable for 10-20 minutes at times. |
| 06:08.02 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 06:13.38 | iamamoron | is there any ip phone that has speaker out? |
| 06:14.26 | *** join/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 06:14.29 | kaldemar | is trying to think of one without a speaker |
| 06:14.41 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 06:21.31 | dwery | is there a way to have ringing tone instead of the music in the queues? |
| 06:23.48 | dwery | found: Queue(..., r) |
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| 06:43.51 | DigitalDaz21 | Hi all, I have seen some sip logs where asterisk contains Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO in the REGISTER and INVITE requests, I have also seen some logs were it is not sent can anyone tell me how to force it to be sent? |
| 06:44.45 | jojo^ | DigitalDaz21, sip set debug on ? |
| 06:46.21 | DigitalDaz21 | no, that just turns on the logging |
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| 06:59.46 | kaldemar | DigitalDaz21: i doubt you can force asterisk to send the allow header with those, as it is not required by the protocol. |
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| 07:03.06 | DigitalDaz21 | really, thats interesting. I must have another problem then. I have to different softphones and an IP phone that register and receive calls fine, all three send the allow: and work fine, asterisk registers and receives the OK but when you dial in no sip traffic is received, that was the only difference I could see between the different REGISTER logs |
| 07:04.14 | kaldemar | with what messages do they send the allow header? OPTIONS is really meant for querying message capabilities in SIP, and an OK in response to an OPTIONS is afaik the only message which MUST contain the allow header. |
| 07:04.53 | DigitalDaz21 | They send it with the REGISTER and INVITE |
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| 07:06.19 | DigitalDaz21 | I'm pretty new to looking at stuff at thuis level and I was making the assumption that this provider was being fussy and not sending the traffic because asterisk had not said what it "allowed" |
| 07:06.37 | kaldemar | how is the lack of allow header actually causing problems? |
| 07:07.21 | DigitalDaz21 | I was just guessing its the allow but I can neither send or recieve calls |
| 07:07.29 | kaldemar | the allow header can be sent with an invite to reduce the amount of dialog between user agents, but it is not required. |
| 07:07.43 | DigitalDaz21 | right |
| 07:08.19 | kaldemar | the problem must be something else if you can't send or receive calls. if you can pastebin a failed call attempt, someone will surely take a look at it and take an educated guess on the issue. :) |
| 07:09.30 | DigitalDaz21 | I get the OK back to the asterisk and I also see it as registered in the sip show registry but just receive nothing. |
| 07:09.41 | DigitalDaz21 | I'll try that :) |
| 07:10.17 | *** join/#asterisk mintos (n=mvaliyav@209.132.188.254) |
| 07:10.22 | DigitalDaz21 | Also if I fire up a softclient after registering the asterisk, asterisk now starts to receive calls too |
| 07:10.56 | kaldemar | so you're registering to an ISP but not getting calls from them? |
| 07:11.24 | DigitalDaz21 | Yes but only from the asterisk, softclients work fine |
| 07:13.36 | kaldemar | mmkay.. that's weird. i've heard of some ISP's that don't accept asterisk clients for some reasons. changing the useragent string (param useragent= in sip.conf under [general]) has helped in those, but start with showing the sip debug. |
| 07:15.42 | DigitalDaz21 | yep, I have already changed the user agent, on an outbound I get the 480 Temporarily Unavailable |
| 07:17.15 | DigitalDaz21 | Here is a failure log http://pastebin.com/d33e6c9b9 |
| 07:18.17 | DigitalDaz21 | Just noticed too, it DOES send the allow, with the invite so that theory is out the window :) |
| 07:18.46 | *** join/#asterisk mikkel (n=mikkel@130.226.36.170) |
| 07:20.00 | kaldemar | is the number you're calling legit? it's only four digits. |
| 07:20.55 | TommyBotten | Maybe he lives in a small country? :> |
| 07:21.36 | DigitalDaz21 | yes, its BT call minder, I've just fired up the softphone and dialled it too and its fine, now though, for the next hour, until the softphones registration times out (I'm assuming its that), the asterisk box will send and receive calls fine |
| 07:21.42 | DigitalDaz21 | lol |
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| 07:52.26 | BeeBuu | how can i sent voicemail to my email? |
| 07:53.58 | tamiel | BeeBuu: You just need to configure voicemail.conf (with attach options) and use Voicemail application in your extensions.conf . |
| 07:54.47 | tamiel | BeeBuu: http://www.voip-info.org/wiki/view/Asterisk+config+voicemail.conf is a good starting point =) |
| 07:57.36 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 07:58.36 | BeeBuu | thanks,i had done like that,but still got nothing in my email |
| 07:59.31 | *** join/#asterisk Dovid (n=annon@213.8.121.90) |
| 07:59.34 | BeeBuu | is the server must install a mail program? |
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| 08:01.03 | Dovid | huh ? |
| 08:01.21 | Dovid | Well you would want sendmail if you wanted voicemail2email |
| 08:05.51 | BeeBuu | Dovid: yes,that's what i need |
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| 08:12.35 | Han | Can a 3g/UMTS gateway + * replace a femtocell? |
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| 08:16.29 | flohack | Hi! When using the 1.6 SendFAX, is it just normal that asterisk starts transmitting the fax with g711a and the T38 gateway then reinvites to t38? My fax calls drop with 'Error transmitting fax. result=49: The call dropped prematurely.'... |
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| 08:37.51 | flohack | quit |
| 08:38.02 | flohack | ups, wrong window |
| 08:44.36 | ltd_wk | Anyone know what it means when * is sending back a 404 to a sip invite but there is absolutely nothing in the logs/verbose/debug? |
| 08:45.01 | Dovid | 404 means the extension was not found |
| 08:45.05 | Dovid | try setting the verbosity to 9 |
| 08:45.13 | Dovid | and turn on full logging |
| 08:45.29 | ltd_wk | with full debug logging, and full verbose, there is no indication of why the 404 occurred |
| 08:45.45 | ltd_wk | if i do a sip debug on the peer, i get nothing too. it's as if it isn't recognising the peer |
| 08:45.46 | Dovid | then look at ur dialplan |
| 08:46.07 | Dovid | so when the call comes in it just sends the remote machine a 404 ? |
| 08:46.15 | ltd_wk | yep. |
| 08:46.20 | Dovid | can u pb your sip.conf and extensions.conf ? |
| 08:46.26 | Dovid | with a sip debu |
| 08:46.30 | Dovid | dbug* |
| 08:46.33 | Dovid | debug* |
| 08:47.11 | *** join/#asterisk Shinsaku (n=Shinsaku@chello084010157123.chello.pl) |
| 08:47.14 | ltd_wk | that's the weird thing, sip debug shows nothing about the call |
| 08:47.22 | ltd_wk | shows no invite coming in and no response |
| 08:47.31 | Dovid | try using ngrep |
| 08:47.54 | Dovid | http://ngrep.sourceforge.net/ |
| 08:47.56 | ltd_wk | i can capture the packets using tcpdump, i see them, but i'm confused as to why sip debug shows nothing. |
| 08:48.00 | kaldemar | that's usually a case of client misconfiguration |
| 08:48.07 | Dovid | like i said ;) |
| 08:48.39 | Dovid | but i have seen ITSP's send send traffic not from the registar IP and Asterisk has no configuration for that IP |
| 08:49.01 | Dovid | people think that if I register with X I will only get calls from X and not Y |
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| 08:50.50 | Dovid | brb |
| 08:52.00 | ltd_wk | http://pastebin.ca/1511993 |
| 08:56.25 | *** join/#asterisk Stese (n=Someone@adsl.ntsols.com) |
| 08:56.31 | Stese | Morning all |
| 09:02.04 | Stese | I'm having an issue dialing, well anything from my * dialplan.... .conf files here... http://pastebin.com/d10f118b3 |
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| 09:02.50 | Stese | if I dial an 'internal' or external number, it says it can't find that extension... |
| 09:03.00 | TommyBotten | It sounds like you are in the wrong context |
| 09:03.38 | Stese | the phones all start in the [LocalSip] Context, |
| 09:08.04 | *** join/#asterisk M1s3ry (n=M1s3ry@boromir.api-digital.com) |
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| 09:13.43 | kaldemar | ltd_wk: there's no User-Agent or Server header in the 404 that would suggest that it is actually asterisk sending the message. |
| 09:14.57 | kaldemar | Stese: remove the whitespace in dialplan |
| 09:15.53 | kaldemar | if that doesn't help it, pastebin a cli output with sip debug of a failed call |
| 09:16.00 | Stese | ok |
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| 09:21.56 | Stese | kaldemar > http://pastebin.com/d79cd48d1 |
| 09:23.18 | kaldemar | Stese: that's not a call, but a mere SIP capability check between the client and asterisk. paste a whole call with the invite and the response to it. |
| 09:23.52 | Stese | ok...I thought it did have the notice in there... |
| 09:24.55 | kaldemar | that has nothing to do with the call. a dialog that is started with an INVITE is what is needed. |
| 09:25.13 | Stese | ok, must have missed it in my C&P |
| 09:26.51 | Stese | http://pastebin.com/d2c287dc2 |
| 09:28.48 | kaldemar | still no invite there. did you remove the whitespaces and do a dialplan reload? |
| 09:29.37 | Stese | yes, I did... sorry, for some reason my SSH isn't letting me c&p everything |
| 09:32.12 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 09:33.55 | pif | there is no more "core reload" in asterisk-1.6.2 ? |
| 09:34.21 | kaldemar | when was there? |
| 09:35.53 | pif | 1.6.1 had it |
| 09:36.16 | kaldemar | mine doesn't. :P |
| 09:36.32 | pif | ok, so how do you "reload" ? |
| 09:36.41 | kaldemar | just a reload without the "core". |
| 09:37.00 | pif | No such command 'reload' (type 'core show help reload' |
| 09:37.10 | Stese | kaldemar > http://pastebin.com/dba5b470 |
| 09:37.39 | flohack | Hi! When using the 1.6 SendFAX, is it just normal that asterisk starts transmitting the fax with g711a and the T38 gateway then reinvites to t38? My fax calls drop with 'Error transmitting fax. result=49: The call dropped prematurely.'... |
| 09:39.37 | kaldemar | Stese: now we're talking. it's looking for it in LocalSip alright. do a "dialplan show LocalSip" and pastebin it. |
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| 09:40.35 | Stese | kaldemar > http://pastebin.com/d10f118b3 (before i closed down the whitespace) |
| 09:40.52 | Stese | kaldemar > but I will do a new one for the dialplan show now |
| 09:40.55 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 09:41.07 | kaldemar | pif: "module reload" will do it in 1.6.2 |
| 09:41.21 | pif | ok |
| 09:41.26 | Stese | kaldemar > http://pastebin.com/d33986c05 |
| 09:42.20 | kaldemar | Stese: oh, your extensions don't have priority 1. should've spotted that earlier... |
| 09:42.34 | Stese | doh... |
| 09:42.53 | Stese | I thought the needed to step, but thats only for processes, |
| 09:42.57 | Stese | thanks... |
| 09:43.05 | kaldemar | same in context Incoming |
| 09:44.01 | Stese | Are there any other silly things I've missed in there? |
| 09:45.26 | kaldemar | macro-Incoming won't work |
| 09:45.49 | jojo^ | If you use Dial with G() to "split" the two parties of a call, is there a way to join the parties again later? |
| 09:45.53 | kaldemar | "OutBoundMain/" in the Dial is extra |
| 09:46.22 | kaldemar | and macros are deprecated in favor of GoSub nowadays |
| 09:46.41 | Stese | ack... ok |
| 09:47.16 | Stese | I'll have to read up on that one... |
| 09:47.37 | Stese | as for incoming... i'll pastebin the changes, could you have a look at it again? |
| 09:47.42 | kaldemar | sure |
| 09:48.26 | Stese | http://pastebin.com/d45646405 |
| 09:48.30 | kaldemar | jojo^: app Bridge might be suitful |
| 09:49.10 | Stese | basically I transposed the two dial() apps (the variable is supposed to be on the outbound macro) |
| 09:49.39 | kaldemar | this is something strange: Set(CALLERID(all)=INT:$[CALLERIDNAME]) |
| 09:50.21 | Stese | Ahh yes... I may have missed the part where that variable is set... |
| 09:50.29 | kaldemar | CALLERIDNAME variable has [] around it, and actually the function should be used, ${CALLERID(name)}. |
| 09:51.05 | kaldemar | but INT:${CALLERID(name)} doesn't make sense to me either. what exactly are you trying to set there? |
| 09:53.02 | Stese | it's supposed to be adding INT: to the imcoming caller ID |
| 09:53.16 | kaldemar | to the name part? |
| 09:53.23 | Stese | yes |
| 09:53.49 | Stese | so on the phone it will say INT : <number> |
| 09:53.51 | kaldemar | the it's supposed to be Set(CALLERID(name)=INT:${CALLERID(name)}) |
| 09:55.42 | Stese | Ok, done... |
| 09:55.55 | kaldemar | then macro-OutCallUFO. you're passing a number to it as ARG1. however, the Dial doesn't have a tech in it. |
| 09:55.56 | Stese | what about on that other macro, have I done that correctly? |
| 09:56.08 | *** join/#asterisk csiadmin (n=csiadmin@81.144.152.52) |
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| 09:56.39 | kaldemar | the mixmonitor also has [ (but no ], should have {}) |
| 09:57.09 | kaldemar | and the Set possibly needs a space between Financial and the <...> |
| 09:57.30 | bbt | hi, just after some advice on which dyndns service is better, as i do not have a static ip and have to manually restart asterisk to get the sip to register :/ |
| 09:57.53 | *** join/#asterisk qdk (n=qdk@0x573c23bb.bynqu1.dynamic.dsl.tele.dk) |
| 09:59.07 | Stese | ok, think i've got all those sorted ^_^ |
| 10:01.03 | Stese | With Reference, is the whitespace a particular issue? |
| 10:01.20 | Stese | try again, |
| 10:01.38 | Stese | Would having whitespace in the dialplan cause an specific issues? |
| 10:05.43 | kaldemar | depends on where you have it. try to avoid whitespaces in the middle of exten lines. |
| 10:06.13 | Stese | ok, means I can still have them for readability between the contexts |
| 10:06.25 | Stese | thanks for your help :) |
| 10:07.23 | kaldemar | yes, between contexts and exten lines spaces don't matter. |
| 10:15.05 | jojo^ | kaldemar, Bridge seems to work actually. But I fail to set a variable which is available to the new channel after the G()-split |
| 10:15.22 | jojo^ | kaldemar, I mean available to the called party |
| 10:23.19 | kaldemar | jojo^: if you set a channel variable for a calling channel, it won't be available for the callee channel. use a global variable. |
| 10:26.34 | *** part/#asterisk bbt (n=sam@ppp121-44-38-199.lns10.syd7.internode.on.net) |
| 10:26.57 | jojo^ | kaldemar, Hmm. I'm thinking unique global variables then.. But then I will not know in the called leg which global variable to use :) |
| 10:30.45 | kaldemar | U() would also be one choice, you could pass arguments. |
| 10:33.45 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 10:35.15 | Stese | sip reload |
| 10:35.19 | Stese | ack! |
| 10:35.27 | kaldemar | Reloading SIP |
| 10:35.37 | Stese | lol |
| 10:36.06 | Stese | We need a bot that parses all the current commands and outputs the approriate responce |
| 10:37.21 | Stese | or just slaps the user for being an idiot |
| 10:37.29 | Stese | ~idiot Stese |
| 10:37.30 | infobot | Stese: you are an idiot. |
| 10:38.47 | *** join/#asterisk decimalz (n=pbxk1064@203.171.199.190) |
| 10:38.50 | Stese | ~idiot infobot |
| 10:38.50 | infobot | infobot: you are an idiot. |
| 10:39.08 | Stese | ok, i'll stop playing with the bot now |
| 10:48.39 | jojo^ | kaldemar, Hmm. Is U() is newer than 1.4? |
| 10:49.46 | kaldemar | M() is the older equivalent, if U() doesn't exist in your version |
| 10:50.21 | jojo^ | Ah |
| 10:51.00 | kaldemar | U() seems to be 1.6 only. it uses GoSub where G() uses Macro. |
| 10:51.24 | jojo^ | But you can't give an argument or variable to the macro specified with M(), right? |
| 10:52.08 | kaldemar | you can give arguments, core show application Dial: "Arguments can be specified to the Macro using '^' as a delimiter." |
| 10:52.17 | jojo^ | Ah, ^arg |
| 10:52.21 | jojo^ | Thanks :) |
| 10:52.31 | grEvenX | chicken mao pow! |
| 10:54.38 | *** join/#asterisk Shinsaku (n=Shinsaku@chello084010157123.chello.pl) |
| 11:00.36 | Stese | ok, another issue calls coming in from my sip provider can't be found in the dialplan... Ie I call a number that is pointed to * via my sip provider, but asterisk can't find the extension. http://pastebin.com/d13653376 |
| 11:00.48 | Stese | the number I am dialing is 01252 413073 |
| 11:04.26 | *** join/#asterisk Alfio (n=amunoz@75.112.88.200.m.sta.codetel.net.do) |
| 11:07.31 | *** join/#asterisk iewebguy (n=mark@c-75-68-250-121.hsd1.vt.comcast.net) |
| 11:13.22 | Stese | Hmmmm |
| 11:14.36 | kaldemar | Stese: you don't have context defined for [84414746], so it uses default. |
| 11:15.10 | Stese | I thought I'd set it to head to Imcoming instead, Via the register... |
| 11:15.44 | Stese | s/Imcoming/Incoming |
| 11:15.46 | *** join/#asterisk aurax (n=aurax@bzq-179-76-199.static.bezeqint.net) |
| 11:15.46 | kaldemar | btw, you're using a lot of users.conf syntax for [84414746]. is it really in sip.conf? |
| 11:16.41 | kaldemar | register => [transport://]user[:secret[:authuser]]@domain[:port][/extension][~expiry] <--- /EXTENSION, not context |
| 11:17.19 | Stese | yes... it's in Sip.conf... |
| 11:17.34 | Stese | not used users.conf before |
| 11:18.20 | Stese | so that [84414746] context is in there to allow the insecure = port,invite settings |
| 11:18.20 | kaldemar | hasiax, registeriax, hassip, registersip, trunkstyle and hasexten are not sip.conf parameters. you can remove those. |
| 11:18.27 | Stese | ok |
| 11:18.51 | kaldemar | keep it that way with users.conf. :) |
| 11:24.29 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff-emeritus/bhaal) |
| 11:26.01 | BhaalWK | Hey guys, quick question, I have an misdn card plugged in and working, but for some reason when a sip call comes in from my voip provider the misdn drivers seem to steal it, it gets present with a incoming route number which belongs to the isdn trunk |
| 11:26.08 | BhaalWK | How do I stop this from happening? |
| 11:26.32 | BhaalWK | Sorry, its an incoming IAX call |
| 11:26.41 | *** join/#asterisk J4zen (n=j4zen@a82-95-153-17.adsl.xs4all.nl) |
| 11:27.45 | J4zen | I was wondering, in the ASterisk Manager API theres a function called ExtensionState. It returns a status-code when given an extension id |
| 11:27.55 | J4zen | now, on http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+ExtensionState some of those status codes are listed |
| 11:27.56 | *** join/#asterisk Stese (n=Someone@adsl.ntsols.com) |
| 11:28.06 | J4zen | but not nearly all im afraid |
| 11:28.07 | Stese | Ack, PC locked up |
| 11:28.21 | J4zen | Does anyone know a complete documentation of this function? |
| 11:33.14 | jojo^ | kaldemar, M() didn't seem to work with G(), but I found out that there are inherited variables which seem to work fine. |
| 11:36.35 | Stese | kaldemar > I'm guessing all I need to do is to set a 'context =' for that sip trunk, and it will go there, instead of default |
| 11:36.54 | *** join/#asterisk J4zen (n=j4zen@95-36-106-201.dsl.alice.nl) |
| 11:40.13 | kaldemar | Stese: you guess right |
| 11:40.20 | Stese | :) |
| 11:40.41 | Stese | after my PC fell over, I realised what you were trying to say |
| 11:48.10 | *** join/#asterisk Polysics (n=Luca@host59-14-dynamic.24-79-r.retail.telecomitalia.it) |
| 11:48.12 | Polysics | hello |
| 11:48.17 | Polysics | i'm just starting out |
| 11:48.30 | Polysics | do you think i should start from something like pbx in a flash or from scratch? |
| 11:49.02 | Polysics | also, in case of from scratch, which linux distro do you recommend? i'm familiar with ubuntu and debian, slightly less with centos |
| 11:49.40 | Gnarfy | I bought the asterisk book from o'reilly, started reading and just messed around with the examples... its awesome to get the first call done |
| 11:49.54 | Stese | ~boook |
| 11:49.59 | Stese | ~book |
| 11:50.00 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 11:50.17 | Polysics | so you say start from scratch following book? |
| 11:50.33 | Polysics | i will not be using hardware, justa voip provider |
| 11:50.55 | Polysics | the book also has no specific distro listed, i just want to avoid gettign stranded because of some kernel quirk |
| 11:51.42 | Gnarfy | I think its the best way to get to know how asterisk works... to make it easy i'd suggest starting with a distro that has a package manager or something and just install the package to get familiar with it all, but thats just my 2 cents |
| 11:51.44 | Stese | Polysics > I used this webpage http://www.voip-info.org/wiki/index.php?page_id=5608 |
| 11:51.45 | kaldemar | ubuntu or debian are fine. most people just recommend the distro you are most familiar with yourself. |
| 11:52.48 | Polysics | then i'll go with ubuntu, at least i know where to look :-) |
| 11:54.56 | *** join/#asterisk raimund (n=raimund@port-92-198-15-96.static.qsc.de) |
| 11:55.13 | *** join/#asterisk ik_5 (n=ik@95.211.21.45) |
| 11:55.34 | ik_5 | hello |
| 11:56.00 | Polysics | and about starting from scratch? |
| 11:56.06 | Polysics | is it better than a premade install? |
| 11:56.29 | Stese | it depends what you are trying to acheive really |
| 11:57.17 | Polysics | i'm trying to unserdstand asterisk :-) goal is having a simple extension-based PBX, so there is a phone number to call and then you choose the office you want to talk to with the dialtones |
| 11:57.30 | ik_5 | i have a freepbx 2.5.1.5 with asterisk 1.4.26 and i want to customize the voicemail prompt prior of the recording, but when i place a recording at the voicemail user at busy unavail and anunc it does not play the file at all, how can i do it properly ? |
| 11:58.44 | kaldemar | ik_5: have you asked in #freepbx ? |
| 11:59.05 | ik_5 | kaldemar, nop, didn't know there is such channel |
| 11:59.51 | kaldemar | ik_5: start there, freepbx is not supported here. :) |
| 12:00.04 | ik_5 | kaldemar, thanks |
| 12:00.37 | *** join/#asterisk kombi (n=kombi@195.158.185.205) |
| 12:05.59 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:07.13 | *** join/#asterisk propellerhead (n=yogurt2u@host183.190-136-104.telecom.net.ar) |
| 12:07.16 | kombi | can only make one call at a time with dahdi/isdn/ptp, two should be possible. Someone care to look? config: http://pastebin.se/198609 cli-output: http://pastebin.se/198608 |
| 12:08.46 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 12:10.07 | [TK]D-Fender | kombi: -- Executing [01778435622@bureau:2] Dial("SIP/Detlev-0825c498", "DAHDI/1/01778435622,,") in new stack <-- you are dialing out one specific channel, not a group. |
| 12:10.33 | kombi | geee, thanks! |
| 12:10.45 | [TK]D-Fender | kombi: And you'r group directives are all the same, and in the wrong place |
| 12:11.02 | [TK]D-Fender | reaches for his ClueBat (tm) |
| 12:11.59 | Polysics | now with more clue? |
| 12:12.07 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 12:12.21 | kombi | don't hit me just yet (actually someone at digium did all the configs..;) |
| 12:12.55 | Polysics | it's probably OT, but in case, does Didgium provide install/configuration services? |
| 12:13.30 | [TK]D-Fender | Polysics: Config, sure, install.... doubt it... |
| 12:14.02 | Polysics | i should be able to get a basic install going |
| 12:17.24 | [TK]D-Fender | Polysics: PolySix = great analog synth... |
| 12:18.22 | Polysics | it's probably related, but nock comes from a japanese electronic music band |
| 12:18.37 | [TK]D-Fender | Polysics: Probably |
| 12:20.45 | *** join/#asterisk Zhad (n=tom@server30261.uk2net.com) |
| 12:21.16 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
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| 12:28.28 | Stese | Polysics > did you have a look here http://www.voip-info.org/wiki/index.php?page_id=5608 |
| 12:28.51 | Polysics | Stese, wow, i lost your above msg |
| 12:28.56 | Polysics | looks pretty thorough |
| 12:29.02 | Polysics | and i can sorta negotiate centos |
| 12:29.03 | Polysics | thanks |
| 12:30.01 | *** join/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 12:30.11 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 12:34.51 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 12:35.35 | stacker | hello guys, I've asterisk for calling between 2 sip clients. it's working like charm, but I can not reach PTSN. At work I've Avaya pbx and I'd like to redirect my asterisk pbx to reach PTSN via my h323 avaya extenison at work. Can you please give me hint, how to register asterisk pbx with avaya extension? |
| 12:35.52 | *** join/#asterisk kaptengu (n=kaptengu@unaffiliated/kaptengu) |
| 12:36.39 | ariel_ | Is the Avaya a gatekeeper? |
| 12:37.17 | stacker | yes it is |
| 12:37.21 | ariel_ | you can install h323 support via the asterisk-addon |
| 12:37.29 | stacker | and then? |
| 12:37.51 | kaldemar | stacker: http://rblondon.blogspot.com/2008/02/set-h323-trunk-between-asterisk-and.html |
| 12:38.17 | stacker | kaldemar, it's called trunk even only one extension? |
| 12:39.41 | kaldemar | just throwing a link at you. |
| 12:40.09 | stacker | kaldemar, I've seen this link yesterday, but it requires to configure avaya system for trunk, this is not possible for me :( |
| 12:42.11 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 12:42.22 | ariel_ | stacker: it still might work, but on the asterisk side it's a trunk, |
| 12:42.58 | ariel_ | Avaya = $$$ and they don't like supporting non Avaya systems/setups |
| 12:43.32 | Stese | Polysics > No worries, I've used that in the last week to build a box... :) |
| 12:43.33 | stacker | ariel_, ok, so it should be enough to add something like this: register => 192:123456@10.59.99.40:1720/192 into h323.conf ? |
| 12:44.10 | stacker | I know that avaya=$$$ therefore I have no sip trunk :)) |
| 12:44.17 | ariel_ | if the avaya works correctly as a gatekeeper yes it should. But please note there is very little support for this |
| 12:44.26 | *** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl) |
| 12:45.02 | ariel_ | Most of the time when we setup connections to Avaya we either connected via a pri or analog ports. |
| 12:45.58 | stacker | ok, thank you for your help |
| 12:46.58 | *** join/#asterisk ramindia (n=balajibh@202.63.96.10) |
| 12:48.05 | jaytee | Nortel just announced they are selling off their Enterprise Systems stuff to Avaya. |
| 12:50.29 | ariel_ | One less major PBX supplier |
| 12:51.16 | *** join/#asterisk mfilipe (n=michel@189.26.205.3.dynamic.adsl.gvt.net.br) |
| 12:55.37 | Dovid | wow |
| 12:55.42 | Dovid | Nortel was known for their phones |
| 12:56.15 | mfilipe | Hi! I have 4 channels and I want send 50 files to outgoing. When I tested, the asterisk was trying using a channel busy and many calls don't works. Is there any way to create a queue to send a call to a channel only if it is active? |
| 12:58.31 | mchou | I have an opportunity to get a (used) polycom soundpoint IP 500 for $50. Is that a fair price? |
| 13:01.38 | [TK]D-Fender | mfilipe: You need to make your own. |
| 13:01.49 | [TK]D-Fender | mchou: Sure |
| 13:01.57 | *** join/#asterisk keyp (n=chatzill@66.184.128.98) |
| 13:02.30 | *** join/#asterisk iewebguy (n=mark@65.19.81.253) |
| 13:02.34 | angryuser | mchou, yes, but the production of this phone is stopped and parts until 2011 |
| 13:02.40 | mchou | [TK]D-Fender: cool. thx |
| 13:03.12 | angryuser | mchou, "The SoundPoint IP 500 has reached End of Life. Parts, support will no longer be available for this product after 5/01/2011" |
| 13:03.18 | mchou | [TK]D-Fender: that unit has lcd that's backlit, correct? |
| 13:03.35 | [TK]D-Fender | mchou: nope |
| 13:03.49 | mchou | angryuser: yeah, beggars cant be choosers :) |
| 13:03.53 | [TK]D-Fender | mchou: Only models in the last 2 years have that. |
| 13:03.54 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:04.19 | mchou | [TK]D-Fender: oh well, that sucks :) |
| 13:05.09 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:05.15 | [TK]D-Fender | mchou: Yeah it sucks that the Chevy Malibu I have wasn't a Mercedes at the same price too.. |
| 13:05.17 | mchou | angryuser: as long as the sip stack in firmware works well I'm not too concerned |
| 13:05.48 | bobJR | . |
| 13:06.05 | [TK]D-Fender | mchou: IP 500 stops at 2.0 firmware IIRC |
| 13:06.18 | [TK]D-Fender | mchou: Stable enough... |
| 13:06.33 | *** join/#asterisk dajhorn (n=dajhorn@206.16.96.160) |
| 13:11.35 | mchou | angryuser: hmm, maybe I should wait until after may next year :) |
| 13:12.13 | angryuser | mchou, wait for what ? |
| 13:12.15 | mchou | angryuser: prices ought to be even lower then :) |
| 13:12.35 | [TK]D-Fender | mchou: How about 10 years? It'll be free! |
| 13:12.41 | *** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be) |
| 13:12.48 | angryuser | 50$ for a phone is cheap allready |
| 13:12.52 | mchou | [TK]D-Fender: free is good |
| 13:13.52 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
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| 13:29.56 | kombi | read Dial() thoroughly, can't find how to dial out on a group rather than a channel. How is it done? |
| 13:31.42 | stacker | ok, thank you for your help |
| 13:31.59 | *** join/#asterisk stix (n=stix@exchange2003.corporate.billetkontoret.dk) |
| 13:32.31 | stix | Hi guys. Will wav-soundfiles consume more bandwidth than gsm-files or is this defined by the codec used? |
| 13:32.56 | kombi | the latter, might consume more cpu though |
| 13:33.25 | *** join/#asterisk Greek-B0y (n=greek@41.188.154.137) |
| 13:33.27 | stix | okay |
| 13:34.35 | [TK]D-Fender | kombi: DAHDI/g1/1234567890) . <- g/G depending on order |
| 13:34.43 | [TK]D-Fender | Kobaz: and the "1" is the group # |
| 13:34.50 | [TK]D-Fender | kombi: and the "1" is the group # |
| 13:35.16 | [TK]D-Fender | stix: only consume different CPU |
| 13:36.03 | kombi | thanks fender! |
| 13:40.31 | kombi | ;) |
| 13:46.56 | [TK]D-Fender | kombi: these are basics that are documented in tons of places... go read the samples |
| 13:48.19 | *** join/#asterisk jmacz (n=mcorb@190.144.75.22) |
| 13:49.17 | aurax | I'm having this annoyinh problem with callback, when my pbx callback to my cellphone it hang ups immediately |
| 13:50.42 | keyp | Does anyone know if it would be possible to have asterisk set up as an extension on an NEC NEAX pbx on a digital telephone port? |
| 13:50.48 | keyp | I'm out of analog ports and don't know if I can get approval to buy another analog card for the NEC system. |
| 13:54.46 | *** join/#asterisk ruben23 (n=RPL@122.55.48.243) |
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| 13:55.18 | *** join/#asterisk wellwell (n=wellwell@189.4.243.249) |
| 13:55.33 | csiadmin | hi all |
| 13:59.07 | *** join/#asterisk coppice (n=chatzill@193.194.17.210.dyn.pacific.net.hk) |
| 13:59.41 | [TK]D-Fender | keyp: If the port is digital only, then no. |
| 13:59.50 | *** join/#asterisk xpot-mobile (n=james@71-213-54-186.slkc.qwest.net) |
| 14:00.57 | keyp | [TK]D-Fender: That's what I was afraid of, thanks. |
| 14:03.33 | Stese | Does anyone know of an addon which would allow me to set up email notifications if a peer goes unreachable, or a sip registration fails? |
| 14:04.33 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 14:04.33 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:06.57 | guax | !nat |
| 14:07.01 | guax | ~nat |
| 14:07.02 | infobot | rumour has it, nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 14:09.37 | [TK]D-Fender | ~sipnat |
| 14:09.37 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 14:10.01 | guax | [TK]D-Fender, that is it =P thank you |
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| 14:18.50 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:19.35 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 14:20.49 | Gnarfy | I still have the problem with my sip provider dialing to me with s@me, even tho i have several numbers connected (so i can't see which number is calling me)... sip debug: http://pastebin.com/mbc601b Now my question is this: is there any chance that my side (as in asterisk) is responsible for forcing it to call there (at register time or so), instead of just whatever number? because the provider insists it should just work and that it is working at other pe |
| 14:22.08 | wellwell | any1 here use Avantfax? |
| 14:23.14 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:d05e:20a:71e9:3e55) |
| 14:23.29 | [TK]D-Fender | Gnarfy: As we discussed yesterday, if they send multiple DID's to that peer, they do not indicate WHICH is being used anywhere. this is THEIR fault. |
| 14:24.02 | [TK]D-Fender | Gnarfy: The fact is goes to "s" is your fault because of the formatting of your register statement, but either way you're never going to know when # was originally dialed |
| 14:25.15 | Gnarfy | [TK]D-Fender: oke, i'll go shout at the tech helpdesk some more to transfer me to a tech guy :) |
| 14:25.39 | [TK]D-Fender | Gnarfy: I don't know why you would expect a different answer than yesterday.... |
| 14:25.50 | Gnarfy | Well, because they insist i'm doing something wrong :/ |
| 14:26.40 | Gnarfy | They even send me a "basic asterisk" guide they wrote, so ye... kinda had me doubting myself :x |
| 14:28.20 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 14:28.20 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:29.18 | Stese | Gnarfy, is the incoming caller ID in the sip debug? |
| 14:30.00 | [TK]D-Fender | Gnarfy: e-mail them the SIP debug and ask "Where's the #&^$ing number? Huh? WHERE?" |
| 14:30.14 | Stese | dw, I'll read it (I have the same setup with my SIP provider (one username, multiple possible DID's) |
| 14:30.41 | [TK]D-Fender | Gnarfy: You have the equivalent of video footage of a murder with a signed confession and 50 witnesses. |
| 14:30.53 | *** join/#asterisk thieums (n=mathieu@LPuteaux-156-16-101-43.w80-12.abo.wanadoo.fr) |
| 14:31.08 | *** part/#asterisk ik_5 (n=ik@95.211.21.45) |
| 14:31.08 | Gnarfy | [TK]D-Fender: hehe, thats a good idea :) was just making sure, sorry :p |
| 14:32.12 | *** join/#asterisk riksta (n=rick@92.63.131.41) |
| 14:32.20 | riksta | Hi, I have been using asterisk 1.6.1 and 1.6.2 and I am encountering problems with CDRs not being set to BUSY correctly. Has anyone else seen the issue? (SIP->SIP calls) From the sip debug information i see asterisk gets the hangup cause like so: CSeq: 102 BYE X-Asterisk-HangupCause: User busy X-Asterisk-HangupCauseCode: 17 |
| 14:33.12 | *** join/#asterisk LapTop006 (n=laptop00@203.55.222.22) |
| 14:33.52 | *** part/#asterisk LapTop006 (n=laptop00@203.55.222.22) |
| 14:39.11 | *** join/#asterisk hesco (n=hesco@24.99.160.121) |
| 14:39.50 | Gnarfy | riksta: I might be wrong, but if the busy signal is from after a Dial - it wont show up because the CDR is from how the dialplan/incoming call was handled... |
| 14:40.21 | riksta | Gnarfy: Is there a way i can capture that busy event and do something manually? |
| 14:40.34 | riksta | Gnarfy: I can show you the dialplan if it helps? |
| 14:41.02 | Gnarfy | i have a dialmacro, which after the dial does a: exten => s,n,AGI(/home/gnarf/storeit.php,storecall,${NUMBER},${ANSWEREDTIME},${DIALSTATUS}) |
| 14:41.05 | hesco | The TFOT2d talks a bit about how STDERR from an .agi script will only go to the FIRST *CLI console invoked. Working remotely, how can I assure myself that I am connecting to that console. I used the -rc switches but am not seeing my debug output. Any direction available? |
| 14:41.22 | Gnarfy | Then in that storeit, i have the dialstatus from the last call and store it in mysql... |
| 14:41.41 | *** join/#asterisk moos3 (n=richard@pool-72-73-113-232.ptldme.east.myfairpoint.net) |
| 14:42.10 | moos3 | can anyone tell me whats with my dail line in this macro http://pastie.org/565349 |
| 14:42.18 | moos3 | it never calls my phone |
| 14:42.24 | riksta | Gnarfy: so you get DIALSTATUS rather than DISPOSITION , interesting |
| 14:42.52 | Gnarfy | riksta: i just messed around till it worked, this is just how i did it :p |
| 14:43.04 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 14:43.12 | riksta | Gnarfy: thanks, it could prove to be very useful...and it will maybe allow me a quick fix |
| 14:43.23 | Gnarfy | :) |
| 14:43.24 | riksta | Gnarfy: it sux that you cannot override the disposition, as it's read only |
| 14:44.21 | [TK]D-Fender | moos3: It doesn't look like config right now... more like a status issue. thing is you're lokoing at code, and not execution. Pastebin a complete call attempt at verbsoe 10, SIP DEBUG enabled<- |
| 14:44.23 | [TK]D-Fender | ~pb |
| 14:44.24 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/, or apt-get install pastebinit |
| 14:44.25 | [TK]D-Fender | ^^^^^^^^^^^^^ |
| 14:44.27 | riksta | i think i can use this info Gnarfy |
| 14:45.04 | moos3 | how do you enable verbose 10 and SIP DEBUG |
| 14:45.06 | riksta | Gnarfy: mind if i PM you with something? |
| 14:45.17 | Gnarfy | Sure go ahead, kinda new to asterisk tho :p |
| 14:45.50 | [TK]D-Fender | moos3: Go to * CLI : core set verbose 10", "sip set debug" |
| 14:45.57 | moos3 | k |
| 14:47.17 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 14:47.51 | *** join/#asterisk eric_hill (i=eric_hil@204.94.175.11) |
| 14:48.40 | *** join/#asterisk astassistant (n=skip@65-126-63-1.dia.static.qwest.net) |
| 14:49.06 | mfilipe | does the asterisk outgoing have a "pre-processor"? I'm thinking that if the file exists the asterisk will call... :( |
| 14:49.11 | *** part/#asterisk astassistant (n=skip@65-126-63-1.dia.static.qwest.net) |
| 14:50.25 | moos3 | [tk]d-fender: heres the debug out put http://pastie.org/565365 |
| 14:51.39 | *** join/#asterisk EiNSTeiN_ (n=einstein@unaffiliated/einstein/x-615171) |
| 14:54.20 | moos3 | ok now it have complete debug informaiton http://pastie.org/565365 |
| 14:58.17 | *** part/#asterisk raimund (n=raimund@port-92-198-15-96.static.qsc.de) |
| 15:01.24 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:01.52 | *** part/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 15:02.17 | *** join/#asterisk bongfrog (n=quassel@c-98-245-211-157.hsd1.co.comcast.net) |
| 15:02.41 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 15:03.51 | csiadmin | anyone know if I should be concerned about "B-channel 0/1 successfully restarted on span 1" messages? |
| 15:05.00 | *** join/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 15:05.06 | [TK]D-Fender | moos3: Ok, so which is not getting called? |
| 15:05.17 | *** part/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 15:05.20 | moos3 | the last number |
| 15:05.27 | moos3 | the 2077900973 |
| 15:05.29 | *** join/#asterisk propellerhead (n=yogurt2u@host66.190-31-153.telecom.net.ar) |
| 15:05.42 | [TK]D-Fender | moos3: actuall none of : SIP/8092&SIP/8100&IAX2/brenthome |
| 15:06.05 | moos3 | yeah |
| 15:06.36 | *** join/#asterisk thansen (n=thansen@76.27.110.194) |
| 15:06.50 | [TK]D-Fender | moos3: I see multiple calls in there |
| 15:07.05 | moos3 | yeah my asterisk box is a busy busy box |
| 15:07.05 | [TK]D-Fender | moos3: -- Executing [2077900973@default:2] Dial("SIP/9042-08ef0f88", "SIP/2077900973@EASYOFFICEPHONE1,70") in new stack |
| 15:07.24 | moos3 | so what did i do wrong |
| 15:07.32 | [TK]D-Fender | moos3: -- Executing [7027@incoming-did:1] Dial("DAHDI/3-1", "SIP/8091&SIP/8092&SIP/8100&IAX2/brenthome,20,tT") in new stac |
| 15:07.38 | [TK]D-Fender | moos3: Which call are we debugging? |
| 15:08.02 | moos3 | -- Executing [2077900973@default:2] Dial("SIP/9042-08ef0f88", "SIP/2077900973@EASYOFFICEPHONE1,70") in new stack |
| 15:08.15 | [TK]D-Fender | moos3: that DID call out it seems |
| 15:08.31 | [TK]D-Fender | moos3: 101+ |
| 15:08.54 | [TK]D-Fender | moos3: 214 = progress |
| 15:09.22 | [TK]D-Fender | moos3: 296 the other side gave up trying |
| 15:09.31 | moos3 | k |
| 15:09.42 | [TK]D-Fender | moos3: So your calling server seems fine. The callee gave up |
| 15:09.52 | moos3 | its wierd i can call my cell directly but not in the chain |
| 15:10.15 | moos3 | 20 seconds should be long enough for the call right |
| 15:10.43 | *** join/#asterisk Heretic (n=BuRn@ZA1-securenode.echelon.co.za) |
| 15:11.21 | [TK]D-Fender | moos3: The callee decided to stop trying. What are you calling with that dial? It SAYS its an X-Lite, but I'm not believing that because of the rest of the formatting |
| 15:11.41 | moos3 | no its xlite |
| 15:12.14 | moos3 | see it calls the extension right away but never rings my cell |
| 15:12.19 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
| 15:13.50 | [TK]D-Fender | moos3: Why would that dial ring 2 things? Look at it. you are dialing 1 peer |
| 15:14.47 | moos3 | ok |
| 15:15.02 | moos3 | but it rings my deskphone and them softphone but not the cellphone |
| 15:15.11 | moos3 | heres more logs http://pastie.org/565391 |
| 15:15.32 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 15:16.13 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 15:16.44 | Kobaz | do de do |
| 15:18.32 | moos3 | [tk]d-fender can you give more details |
| 15:19.15 | *** join/#asterisk Defraz (n=T0tal@corp.fuzecore.com) |
| 15:21.14 | Gnarfy | about that DID problem: hurray, the tech guy 3rd line support said: yes sir, we'll fix it right away sir :) |
| 15:21.27 | Stese | Gnarfy > Congrats |
| 15:22.14 | Kobaz | i got a little 500mhz small form factor box for testing, is there a minimum cpu limit for meetme and transcoding and stuff? As soon as i launch a meetme, asterisk goes up to 100% cpu and kills the box |
| 15:22.38 | [TK]D-Fender | moos3: I see it calling 1 device, not multiple |
| 15:22.51 | KavanS | Kobaz, I run a 900mhz p3 for a small office...and it does the job quite well, including multiple meetme sessions... |
| 15:22.55 | moos3 | its calling to devices |
| 15:23.07 | moos3 | whats the best way to handle this then |
| 15:23.28 | moos3 | becuse 8051 is my desk, 9042 is my softphone, 2077900973 is my cell |
| 15:23.32 | Stese | moos3 > have you pastebin'd your extentions.conf? |
| 15:24.04 | Kobaz | KavanS: i've seen this many times on slower machines |
| 15:24.09 | moos3 | k |
| 15:24.13 | Kobaz | KavanS: around 1ghz is usually find |
| 15:24.14 | Kobaz | fine |
| 15:24.18 | KavanS | ahh yes... |
| 15:24.28 | KavanS | I assumed 500mhz would be fine for a small setup, but not sure myself, never tested |
| 15:24.39 | Kobaz | especially with transcoding |
| 15:24.49 | Kobaz | even on my 1ghz, if it's transcoding, it completely kills the box |
| 15:24.50 | KavanS | Kobaz, transcoding from what....wav to ogg/mp3? |
| 15:24.57 | Kobaz | no like gsm to ulaw |
| 15:25.01 | KavanS | Kobaz, there's a utility called "nice" |
| 15:25.03 | kn0x | anyone using Openser dispatcher? |
| 15:25.08 | Kobaz | yeah i know about nice |
| 15:25.12 | KavanS | Kobaz, man nice...is it to set a low priority on your encoding processes |
| 15:25.23 | Kobaz | yeah i know |
| 15:25.25 | *** join/#asterisk xpot-mobile (n=james@71-213-54-186.slkc.qwest.net) |
| 15:25.29 | [TK]D-Fender | moos3: [2009-07-30 11:12:39] VERBOSE[14953] logger.c: -- Executing [2077900973@default:2] Dial("SIP/9042-09002948", "SIP/2077900973@EASYOFFICEPHONE1,70") in new stack <-- this is not calling 2 devices |
| 15:26.00 | moos3 | thats from when i dailed the phone directly |
| 15:26.14 | [TK]D-Fender | moos3: * is not placing calls to 2 devices... |
| 15:26.26 | *** join/#asterisk telnettech (i=telnette@gw.percipia.com) |
| 15:27.15 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
| 15:28.46 | jaytee | telnettech, hi guy! how are things? |
| 15:28.47 | Nugget | telnet is eeeeeeevil! |
| 15:29.12 | jaytee | Nugget, do you just lurk in here waiting for him to logon? |
| 15:29.12 | *** part/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 15:29.16 | telnettech | hey jaytee......not bad still looking for a new job |
| 15:29.19 | [TK]D-Fender | prods the Nugget-bot |
| 15:29.24 | telnettech | tomorrow is mylast day |
| 15:29.44 | jaytee | telnettech, ouch! |
| 15:30.23 | jaytee | telnettech, time to do your own startup selling inexpensive * solutions for small businesses |
| 15:30.47 | telnettech | so Monday i get to become part of the 11 percent unemployed here in Ohio |
| 15:31.00 | Stese | telnettech > if you do that, becareful what you name your company... |
| 15:31.02 | telnettech | I do have a couple leads....1 in TN and another in FL |
| 15:31.32 | Stese | Ahh, no matter, since you are in the US, I doubt it would be an issue :P |
| 15:31.32 | jaytee | my friend who lives outside Youngstown is worried about losing her job at the library because Ohio is closing several library branches. |
| 15:31.59 | jaytee | ah, crap! gotta go fix a Sharepoint "issue" |
| 15:32.02 | jaytee | bbiab |
| 15:33.08 | *** join/#asterisk cusco (n=tralala@2001:0:53aa:64c:3ce6:eab9:2ac0:762d) |
| 15:33.09 | Chainsaw | jaytee: Oh noes :( |
| 15:33.15 | moos3 | [tk]d-fender: here you go http://pastie.org/565426 |
| 15:34.01 | mbrevda | anyone know if you can use subscriptions with Skype for Asterisk Beta? |
| 15:35.00 | cusco | before we get the licences for the g729 codec, is there a softphone that we can implement that does not require paying for more fees? |
| 15:35.54 | cusco | for both linux and windows |
| 15:38.33 | Stese | moos3 > You are setting args for your macro, but you aren't using them... |
| 15:38.41 | flohack | cusco: I wrote opensoftphone: http://code.google.com/p/opensoftphone/ |
| 15:38.58 | moos3 | what you mean i'm using them |
| 15:39.13 | UQlev | cusco: twinkle for linux seems to support g729 |
| 15:39.16 | moos3 | exten => 8051,1,Macro(moose,${EXTEN},SIP/${EXTEN},70) |
| 15:39.32 | Stese | yes, you are setting arguments for use in the macro |
| 15:40.04 | Stese | but your macro doesn't call them at all |
| 15:40.08 | moos3 | so wha should that line look like |
| 15:40.29 | cusco | UQlev: we use twinkle right now, on twinkle webpage does not reffer g729 |
| 15:40.35 | flohack | cusco: I may have misunderstood you, opensoftphone does not support g729 |
| 15:40.52 | cusco | flohack: ok.. |
| 15:40.57 | flohack | cusco: Why do you need g729? |
| 15:41.16 | cusco | flohack: we need to optimize our bandwidth usage |
| 15:41.32 | Stese | moos3 > well, technically, > exten => 8051,1,Macro(moose), I think |
| 15:41.37 | moos3 | k |
| 15:41.54 | UQlev | cusco: I have seen it in audio settings of twinkle |
| 15:42.03 | Stese | i'm not sure that will make a difference, |
| 15:42.21 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 15:42.22 | cusco | if we get licenses for asterisk to use g729, does it transcode for us..? |
| 15:42.36 | moos3 | we will find out |
| 15:42.37 | cusco | UQlev: let em check |
| 15:42.54 | flohack | cusco: why not speex or gsm? |
| 15:43.03 | UQlev | cusco: I had latest from portage on gentoo |
| 15:43.19 | moos3 | ok that rings both my desk and my softphone but not he celly |
| 15:43.35 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:44.11 | Stese | ok, in the macro, temporarily ditch the desk and soft phones, and see if it still fails |
| 15:44.41 | cusco | UQlev: what version is it |
| 15:44.59 | cusco | flohack: g729 is only 8kb |
| 15:45.22 | *** join/#asterisk dajhorn (n=dajhorn@206.16.96.160) |
| 15:45.30 | UQlev | cusco: 1.4.2 |
| 15:45.30 | moos3 | here is the log from that |
| 15:45.55 | *** join/#asterisk SkramX (n=SkramX@cpe-70-112-25-138.austin.res.rr.com) |
| 15:46.16 | [TK]D-Fender | moos3: It isn't calling multiple devices. We see this in your output |
| 15:46.26 | *** join/#asterisk captiancrash (n=jonmoore@70.159.118.70) |
| 15:46.30 | [TK]D-Fender | moos3: Where do you see something that makes you believe it should do otherwise? |
| 15:47.06 | [TK]D-Fender | moos3: [2009-07-30 11:12:39] VERBOSE[14953] logger.c: -- Executing [2077900973@default:1] Set("SIP/9042-09002948", "CALLERID(all)=Symplicity <7033510200>") in new stack <-- this line does not originate from a Macro |
| 15:47.19 | moos3 | Stese: It hung up on me when i calling my extension |
| 15:47.21 | Stese | [TK]D-Fender > he pasted his extensions.conf... he has a macro that starts the calls |
| 15:47.30 | [TK]D-Fender | Stese: Look at the CALL. |
| 15:47.37 | [TK]D-Fender | Stese: the CALL is not using a Macro! |
| 15:48.00 | [TK]D-Fender | Stese: Who cares whats in the top shelf of my cabinet when all you do is look on my DESK? |
| 15:48.59 | cusco | ok I will install it UQlev |
| 15:49.09 | WindowsUser | cusco: the license is only needed for transcoding, if you call from one device to another that both support g729 asterisk just routes packets or the sip conversation |
| 15:49.14 | moos3 | ok so I tried in the macro just the cell number and it says trying extension then hangs up |
| 15:49.26 | Stese | moos3 > You are dialing 8051 aren't you |
| 15:49.32 | moos3 | yeh |
| 15:49.56 | *** join/#asterisk zeeesh (n=zeeesh@203.215.179.43) |
| 15:50.06 | moos3 | once I put it back to the old way bam it works, but no celly |
| 15:50.41 | cusco | WindowsUser: yes i noticed that... so we don't need a softphone with g279 if we have lices and the codec installed on asterisk? |
| 15:51.03 | moos3 | ideas? |
| 15:51.27 | [TK]D-Fender | moos3: Nowhere do I see you calling an extension that dials multiple devices. |
| 15:51.50 | moos3 | look here |
| 15:51.51 | moos3 | http://pastie.org/565444 |
| 15:52.18 | Stese | [TK]D-Fender > as far as his extenisons.conf is concerned, he is... |
| 15:52.43 | Stese | [TK]D-Fender > as far as I can see, anyway... |
| 15:52.52 | [TK]D-Fender | Stese: Who cares what dialplan he shows when it clearly isn't what is getting executed? |
| 15:53.20 | moos3 | yeah but why isn't it getting executed |
| 15:53.24 | Stese | moos3 > how long does your cell take to connect when you dial it from a landline |
| 15:53.39 | moos3 | about 4 to 5 seconds |
| 15:53.49 | moos3 | should i extended the time |
| 15:53.54 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 15:53.59 | Stese | it's 30 now, right? |
| 15:54.03 | [TK]D-Fender | moos3: Ok, in this last one we are dealing with a completely different call. |
| 15:54.20 | moos3 | yeha |
| 15:54.36 | [TK]D-Fender | moos3: and I see 3 distinct calls going out |
| 15:54.43 | moos3 | when i call it from my software it takes about 10 seconds to connect |
| 15:55.10 | [TK]D-Fender | moos3: So it places 3 calls and you didn't show how it ended <- |
| 15:56.31 | *** join/#asterisk ethicx (n=ethicx@adsl-074-169-015-252.sip.mia.bellsouth.net) |
| 15:56.37 | moos3 | k |
| 15:56.38 | ethicx | sup sup you all. |
| 15:57.23 | WindowsUser | cusco: exactly |
| 15:57.31 | leifmadsen | [TK]D-Fender: Asterisk Open Beta is now available |
| 15:57.43 | Qwell | Asterisk Open Beta?! |
| 15:57.45 | [TK]D-Fender | \o/! |
| 15:57.49 | Qwell | WHERE DO I SIGN UP? |
| 15:57.55 | [TK]D-Fender | Yay, I've been waiting to try Asterisk for YEARS! |
| 15:58.02 | leifmadsen | errr.... :) |
| 15:58.02 | leifmadsen | lol |
| 15:58.04 | ethicx | TCP is when she expects you to show you're listening by interjecting the occasional "I love you", or "I told you that chick was crazy", or maybe even making a point of your own |
| 15:58.15 | leifmadsen | s/Asterisk/Asterisk for Skype/ |
| 15:58.20 | Qwell | other way around |
| 15:58.22 | ethicx | UDP is when you come home and your girlfriend runs her mouth for an hour. |
| 15:58.34 | leifmadsen | whatevah!!!! |
| 15:58.43 | leifmadsen | Qwell: stop harshing my buzz |
| 15:58.48 | [TK]D-Fender | ethicx: Depends where she runs it ;) |
| 15:58.51 | Stese | ? |
| 15:59.01 | ethicx | haha.. |
| 15:59.31 | WindowsUser | asterisk for skype? |
| 15:59.38 | KavanS | heh |
| 15:59.42 | KavanS | whenever they release it! |
| 15:59.48 | KavanS | hasn't it been in beta for awhile now? |
| 15:59.49 | Qwell | WindowsUser: leifmadsen is dyslexic |
| 16:00.05 | KavanS | for some reason I doubt it's the technical aspect that's slowing down skype/asterisk....moreso a business side |
| 16:00.08 | leifmadsen | KavanS: it's been in private beta |
| 16:00.23 | scoof | leifmadsen: but skype is......evuhl! |
| 16:00.26 | KavanS | leifmadsen, exactly...to me it sounds like they will want some sort of limitation of functionality or features |
| 16:00.29 | leifmadsen | what isn't evil? |
| 16:00.30 | moos3 | http://pastie.org/565477 |
| 16:00.33 | KavanS | scoof, just another way to communicate |
| 16:00.35 | Qwell | leifmadsen: Google. |
| 16:00.35 | moos3 | here you go |
| 16:00.36 | scoof | leifmadsen: *? |
| 16:00.44 | leifmadsen | Qwell: they are the MOST EVIL! |
| 16:00.52 | WindowsUser | leifmadsen: so now any old idiot can download a beta for chan_skype? |
| 16:00.53 | Stese | Where can I have a look @ * for Skype? |
| 16:00.58 | leifmadsen | please actually type Asterisk instead of typing * |
| 16:01.06 | leifmadsen | (general announcement) |
| 16:01.15 | WindowsUser | starts surfing around digium.om |
| 16:01.16 | scoof | leifmadsen: deal |
| 16:01.21 | scoof | leifmadsen: asterisk? |
| 16:01.27 | Stese | if I must... |
| 16:01.27 | WindowsUser | s/digium.om/digium.com/i |
| 16:01.45 | leifmadsen | try the store |
| 16:02.19 | leifmadsen | hmm... odd, I can't find it :) I got it via twitter |
| 16:03.09 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 16:03.13 | Qwell | leifmadsen: twitter from who? |
| 16:03.41 | WindowsUser | http://store.digium.com/productview.php?product_code=804-00019 |
| 16:03.51 | WindowsUser | jsgoecke |
| 16:03.52 | leifmadsen | there you go |
| 16:03.54 | leifmadsen | ya |
| 16:05.10 | *** join/#asterisk dajhorn (n=dajhorn@206.16.96.160) |
| 16:05.25 | Stese | thankies... |
| 16:05.27 | WindowsUser | huh i wonder if i'll need a g.729 codec license, my spa3102 has g.729 |
| 16:05.51 | WindowsUser | i'll try this and get back to y'all if it whines, for informational purposes |
| 16:06.19 | *** join/#asterisk voxter (n=voxter@190.241.15.56) |
| 16:08.07 | moos3 | stese what i'm i doing wrong |
| 16:09.21 | Stese | moos3 > i'll look @ the latest paste you;ve got up, but I [TK]D-Fender is better versed in this than I. |
| 16:10.52 | *** join/#asterisk mort_gib (n=mjensen@adsl-2-234.gibnet.gi) |
| 16:13.05 | *** join/#asterisk many (n=many@dslb-092-075-124-152.pools.arcor-ip.net) |
| 16:13.30 | Stese | that pastebin doesn't dial your cell (2077900973) as far as I can see... (searched the entire pastebin for that number string) |
| 16:16.07 | moos3 | k |
| 16:16.36 | moos3 | look at line 116 |
| 16:18.23 | moos3 | fixed it |
| 16:18.40 | moos3 | I miss typed 207290 |
| 16:18.54 | moos3 | but the macro(moose) is the fix |
| 16:19.00 | *** join/#asterisk Whitor (n=Whitor@cpe-74-76-185-31.nycap.res.rr.com) |
| 16:19.05 | moos3 | in the extension thanks for all the help |
| 16:20.08 | Stese | I did that yesterday, spelt extensions.conf wrong |
| 16:20.45 | moos3 | lol |
| 16:21.34 | *** join/#asterisk shazaum (n=889uJWER@unaffiliated/shazaum) |
| 16:21.49 | moos3 | anyone know what this is |
| 16:21.50 | moos3 | [2009-07-30 12:18:09] WARNING[3346]: chan_sip.c:2803 retrans_pkt: Maximum retries exceeded on transmission YmQ1Mzc3ZWRkYTgyNDdmNGFlZWExYjJiZmExYmZmN2U. for seqno 133 (Non-critical Request) -- See doc/sip-retransmit.txt. |
| 16:22.13 | russellb | did you read the document referenced in the warning message? |
| 16:22.16 | shazaum | "See doc/sip-retransmit.txt." |
| 16:22.26 | *** join/#asterisk mykhyggz (n=mykhyggz@evolone.org) |
| 16:32.32 | cusco | WindowsUser: so if we get a softphone that supports taht g729 codec, we only need one channel per conversation instead of two, thus requiring only one licence instead of two? |
| 16:35.03 | *** join/#asterisk yogesh1 (i=yogesh1@122.160.173.92) |
| 16:36.33 | yogesh1 | asterisk 1.6.1 getting error unable to create channel of type SIP . can anyone please help |
| 16:36.56 | Defraz | Having a weird issue that just started, when a call comes in from my sip provider, the phone rings and we take the ip phone off hook and it doesn't pick up. You still see the call there but no pickup. |
| 16:37.16 | [TK]D-Fender | yogesh1: Go look at your peer status |
| 16:37.17 | *** join/#asterisk AlmightyOatmeal (n=jamie@68-185-102-22.dhcp.mdsn.wi.charter.com) |
| 16:39.05 | AlmightyOatmeal | is it possible to dial a phone number from the cli? |
| 16:39.50 | WindowsUser | cusco: i thought a channel was both ways |
| 16:39.50 | yogesh1 | [TK]sip show peers displays 0 sip peers |
| 16:40.00 | [TK]D-Fender | yogesh1: Can't call to a peer that doesn't exist... |
| 16:40.02 | cusco | not with g729, I think |
| 16:40.33 | WindowsUser | if you have g729 just passing through asterisk you dont need a license |
| 16:40.38 | [TK]D-Fender | * G.729 is licensed per channel and includes 1 ENCODER, 1 DECODER |
| 16:41.02 | yogesh1 | But i am able to register my phones |
| 16:41.52 | [TK]D-Fender | yogesh1: Oh really? Show us. PASTEBIN the SIP DEBUG from a registration and a "sip show peers" along with the failed call attempt. |
| 16:41.53 | WindowsUser | <PROTECTED> |
| 16:41.53 | [TK]D-Fender | ~pb |
| 16:41.54 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/, or apt-get install pastebinit |
| 16:41.56 | [TK]D-Fender | ^^^^^^^6 |
| 16:43.18 | AlmightyOatmeal | @#)@&#%(*$ |
| 16:43.23 | AlmightyOatmeal | giggles |
| 16:45.11 | *** join/#asterisk robito (n=robito@190.41.153.85) |
| 16:46.07 | robito | hi guys someone knows this warning |
| 16:46.09 | robito | WARNING[18847]: file.c:1227 waitstream_core: Unexpected control subclass '-1' |
| 16:51.28 | yogesh1 | [TK] http://pastebin.ca/1512418 --- sip debug with registrations for users 2000 and 2001 |
| 16:51.53 | yogesh1 | [TK] http://pastebin.ca/1512421 --- sip debug for sip show peers |
| 16:52.12 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:1234:211:2fff:fe18:d003) |
| 16:54.05 | yogesh1 | [TK] http://pastebin.ca/1512425 --- sip debug with chanunavail error |
| 16:54.17 | WindowsUser | crap i just made a skype account for my asterisk the wrong way |
| 16:56.08 | *** join/#asterisk manxpower (n=eric@57.sub-70-222-222.myvzw.com) |
| 16:56.55 | *** join/#asterisk generalhan (n=asd@about/windows/staff/generalhan) |
| 16:57.27 | leifmadsen | oh snap |
| 16:57.39 | AlmightyOatmeal | crackle |
| 16:57.42 | AlmightyOatmeal | and POP! |
| 16:57.57 | leifmadsen | btw: the user I create for the beta will be available on the non-beta right? |
| 16:58.16 | leifmadsen | I presume it will be :) |
| 16:58.28 | Qwell | leifmadsen: hmm, best ask Terry, but...probably |
| 16:58.52 | leifmadsen | hrh |
| 16:58.55 | leifmadsen | heh |
| 16:59.24 | yogesh1 | [TK] any help |
| 17:02.48 | WindowsUser | leifmadsen: the skype business user?, it shouldn't lose its businessness |
| 17:03.28 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 17:03.43 | *** join/#asterisk ingenius (n=alektro@186.136.6.218) |
| 17:04.08 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 17:06.30 | JerJer | hey digium ppl there is a bug on your site |
| 17:06.48 | JerJer | Once you have received your password, please [link]proceed to login[/link] |
| 17:06.52 | JerJer | that link does not work |
| 17:06.57 | JerJer | from https://www.digium.com/en/login/registered.php |
| 17:07.03 | WindowsUser | thats kind of a minor bug |
| 17:07.12 | JerJer | still a damn bug |
| 17:07.19 | WindowsUser | so report it |
| 17:07.31 | WindowsUser | but not to random people on irc ;) |
| 17:07.48 | JerJer | so digium folks can't hang out here? |
| 17:07.51 | WindowsUser | damn, segfault |
| 17:09.40 | JerJer | keep being lame and i'll make it segfault even more |
| 17:12.25 | yogesh1 | asterisk 1.6.1 getting error unable to create channel of type SIP . can anyone please help |
| 17:12.49 | ariel_ | what errors |
| 17:12.52 | JerJer | yogesh1: lots of reasons |
| 17:12.58 | JerJer | is the peer registered? |
| 17:13.06 | JerJer | is chan sip even loading ? |
| 17:13.16 | yogesh1 | But i am able to register my phones |
| 17:13.32 | ariel_ | well that is just part of it |
| 17:14.38 | yogesh1 | jerjer i have pastebined the sip debugs. will that be of nay help to you? |
| 17:14.46 | hesco | !paste |
| 17:14.51 | hesco | ~paste |
| 17:14.52 | infobot | i heard paste is http://rafb.net/paste/, or see also pb, or http://bin.cakephp.org/ |
| 17:14.58 | JerJer | pastebin.ca |
| 17:15.30 | yogesh1 | [jerjer] http://pastebin.ca/1512425 --- sip debug with chanunavail error |
| 17:15.48 | yogesh1 | [jerjer] http://pastebin.ca/1512418 --- sip debug with registrations for users 2000 and 2001 |
| 17:16.04 | yogesh1 | [jerjer] http://pastebin.ca/1512421 --- sip debug for sip show peers |
| 17:16.37 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:16.43 | JerJer | there isn't any sip peers? |
| 17:16.48 | WindowsUser | leifmadsen: when you type skype show version do you get UNKNOWN_and_probably_unsupported for the Library version? |
| 17:17.13 | leifmadsen | JerJer: you need to report it to webmaster@digium.com I think, or through the form |
| 17:17.22 | yogesh1 | [jerjer] yes i also saw that but i am able to register my phones |
| 17:17.23 | leifmadsen | the people in charge of the website don't hang out here |
| 17:17.24 | [TK]D-Fender | yogesh1: "sip show users" |
| 17:17.31 | JerJer | leifmadsen: yeah i did that before I came here |
| 17:17.35 | *** join/#asterisk kannan (n=kannan@59.163.207.97) |
| 17:17.38 | leifmadsen | WindowsUser: yes I do |
| 17:17.55 | JerJer | WindowsUser is just being a 'windows user' aka lame |
| 17:18.09 | yogesh1 | [TK] sip show users doesnot also show any info about my phones |
| 17:18.18 | [TK]D-Fender | yogesh1: And I do not see you showing us a successful registration like I asked |
| 17:18.26 | JerJer | grr - there is no sip 'user' |
| 17:18.29 | [TK]D-Fender | yogesh1: Are you using realtime peers? |
| 17:18.37 | WindowsUser | k |
| 17:18.37 | leifmadsen | hmmm... I can't seem to get the skype user to register for some reason |
| 17:18.47 | leifmadsen | and the debugging is kind of cryptic |
| 17:18.49 | yogesh1 | [TK] yes realtime peers. |
| 17:18.58 | leifmadsen | NOTICE[29095]: core.cpp:522 transport_log: SKYPE ERROR - SESocketTransport::push(), not connected |
| 17:18.59 | WindowsUser | leifmadsen: you're using a skype user made via the bcp? |
| 17:19.12 | leifmadsen | WindowsUser: yes, I just signed up for BCP and there is 1 user in there |
| 17:19.18 | [TK]D-Fender | yogesh1: Show me that your target phone is registered. because right now nothing seems cached locally |
| 17:19.19 | leifmadsen | I didn't use my existing account |
| 17:19.26 | yogesh1 | [tk] i have pastebined the sip with successful registration http://pastebin.ca/1512418 |
| 17:20.05 | *** join/#asterisk buttons840 (n=buttons8@c-24-10-149-58.hsd1.ut.comcast.net) |
| 17:20.55 | WindowsUser | I'm registering but asterisk is segfaulting when i hit voicemail, i guess i'll a Dial() to a handset |
| 17:21.38 | leifmadsen | aha! they initial account is not a business account apparently |
| 17:21.39 | *** join/#asterisk bobJR (n=bob@adsl-150-237-62.tys.bellsouth.net) |
| 17:23.21 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 17:23.59 | DagMoller | hi folks, anyone has sucessful instaled/configured chan_skype on asterisk 1.6.0.10? |
| 17:24.09 | yogesh1 | [tk]i have done a pastebin again which is after sip set debug off. http://pastebin.ca/1512455 |
| 17:24.41 | yogesh1 | [tk] it shows the successful registration of two phones 2000 and 2001 |
| 17:24.53 | hesco | I'm encountering a snag in my attempt to use Asterisk::AGI. Details at: http://pastebin.com/d20760fc9 Any ideas what might move me forward would be appreciated. I'm not seeing the output of a ->noop() call, much less anything more complicated than that. |
| 17:25.08 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 17:25.09 | buttons840 | I'm issuing an originate command, the channels is just a number on the PSTN. My connection to the PSTN is done though a PCI card, sangoma fxo. The dialplan I connect to is wait(3); playback(tt-weasles); wait(3). Preaty simple dial plan. The problem is the dial plan begins imediately. I always though the dial plan would be started once the connection was made, but often when calling my cell phone the dial plan has completed |
| 17:25.09 | buttons840 | <PROTECTED> |
| 17:26.07 | hesco | buttons840: is there an Answer() in the mix? |
| 17:26.15 | buttons840 | no, should there be? |
| 17:26.24 | manxpower | buttons840: FXO ports are considered "answered" as soon as dialing is finished. |
| 17:26.26 | hesco | perhaps, given what little I know |
| 17:26.35 | hesco | not familiar with that |
| 17:27.13 | manxpower | this is one of the major limitations of FXO ports |
| 17:27.13 | buttons840 | manxpower, your saying this is expected behavior? |
| 17:28.07 | manxpower | buttons840: If I understand what you are saying, yes. One way around that that I've used is play the file over and over and wait for the user to press "1" to accept the call or something like that, then continue in the dialplan |
| 17:28.53 | buttons840 | i'll have to consider that, |
| 17:29.14 | hesco | anyone here use Asterisk::AGI? |
| 17:29.26 | manxpower | PRI (and virtually all VoIP providers use PRI or similar) you don't have that issue |
| 17:29.46 | [TK]D-Fender | buttons840: Or you can do "callprogress=yes" to get audio-based call progress. this is also regrettably synonymous with "disconnectmycallsatrandom=yes" however |
| 17:29.50 | yogesh1 | [tk]dow the pastebins provide any insight? |
| 17:30.03 | buttons840 | my ultimate objective is to place an outbound automated message (i'm issuing originates using a ami script), i wont be able to do this effectively using fxo, but should i use fxs (?) or a viop provider will these work better? |
| 17:30.03 | manxpower | so if you will be using a PRI or SIP for this when it's in production get yourself a cheap VoIP provider for testing. |
| 17:30.22 | manxpower | buttons840: FXS no. |
| 17:30.50 | [TK]D-Fender | yogesh1: Your SQL setup looks broken |
| 17:30.51 | *** join/#asterisk voxter (n=voxter@190.241.15.56) |
| 17:30.52 | buttons840 | the final environment will use SIP provider |
| 17:31.13 | [TK]D-Fender | buttons840: That also depends on your SIP provider <- |
| 17:31.14 | buttons840 | so that should work as I desire? that is, not starting the dial plan until the call is answered? |
| 17:31.26 | manxpower | buttons840: test with a SIP provider then. |
| 17:31.29 | yogesh1 | [tk] can you please elaborate? |
| 17:31.41 | buttons840 | [TK]D-Fender, what is callprogress = yes, what does it do? |
| 17:31.43 | manxpower | that will solve all the issues you will have with analog lines |
| 17:31.48 | [TK]D-Fender | yogesh1: http://pastebin.ca/1512455 <-- errors everywhere, field mismatches, etc. BAD |
| 17:32.01 | manxpower | buttons840: callprogress should be called randomlydisconnectmycalls=yes |
| 17:32.22 | buttons840 | what is it's purpose then? |
| 17:32.33 | buttons840 | or is it just a left over feature? |
| 17:32.45 | [TK]D-Fender | buttons840: to provide audio progress. Its also hit-or-miss, and occasionally it might drop an active call |
| 17:32.48 | manxpower | buttons840: it was supposed to solve these FXO issues, but it has never worked well. |
| 17:33.26 | buttons840 | i see |
| 17:34.10 | buttons840 | i suppose i could also listen for silence to indicate the phone has stoped ringing and someone/something is listening? |
| 17:34.24 | yogesh1 | [tk]yes there are attempts made by odbc to connect and it fails intitally. But the odbc connections do get connected to asterisk. (the last line of pastebin 1512455) |
| 17:34.55 | [TK]D-Fender | yogesh1: Your setup is still not proper. Fix that first. I don't deal with realtime as it is, and seeing those errors does not instill any faith in me at all |
| 17:35.58 | bmoraca | yogesh1: your ODBC is connecting, yes, but asterisk's queries are failing, meaning that your database likely isn't set up properly. you need to fix that or not use realtime. |
| 17:36.06 | buttons840 | what can i ask potential SIP/VoIP providers to determine if they have this feature i want? something about PRI? |
| 17:36.14 | yogesh1 | [tk] thanks for your input. |
| 17:37.18 | [TK]D-Fender | buttons840: They use PRI or SS7 themselves. Some ITSP's also report ANSWER immediately and pass ringing as audio, not OOB progress. |
| 17:37.25 | [TK]D-Fender | buttons840: which would leave you equally screwed |
| 17:37.35 | [TK]D-Fender | buttons840: So test with your provider first |
| 17:37.50 | DagMoller | anyone has sucessful installed/configured chan_skype on asterisk 1.6.0.10? |
| 17:38.55 | *** join/#asterisk leejohn (n=hellrace@124.106.206.203) |
| 17:39.30 | yogesh1 | [bmoraca]Can you point out what i need to fix. the odbcinst -q -s command returns without any errors |
| 17:40.31 | WindowsUser | anyone else in here testing skype for asterisk? |
| 17:41.17 | yogesh1 | [bmoraca] i have the dsns and res_odbc.conf excatly the way various tutorials have indicated. Is there any specific how to that i can follow? I will reinstall if required. Will recompiling * help |
| 17:41.42 | [TK]D-Fender | yogesh1: its *'s interaction that is the problem. You could have a perfect configured server, but it will still fail if you try using it as a HAMMER |
| 17:41.45 | bmoraca | yogesh1: it's not your asterisk config that's messed ut, it's the format/layout of your database. |
| 17:42.07 | [TK]D-Fender | yogesh1: Its your * ODB configs that don't match the expected specs. recompiling is not going to help |
| 17:43.10 | yogesh1 | [bmoraca]I will recreate the DB tables. Can you point me to a sample DB table structure on postgresql |
| 17:43.26 | yogesh1 | [tk]I will recreate the DB tables. Can you point me to a sample DB table structure on postgresql |
| 17:44.36 | [TK]D-Fender | yogesh1: Its in the docs, go read them. |
| 17:44.57 | [TK]D-Fender | 9tarball) |
| 17:44.59 | [TK]D-Fender | and |
| 17:45.00 | [TK]D-Fender | ~book |
| 17:45.01 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 17:45.24 | yogesh1 | [tk]Thanks |
| 17:45.37 | yogesh1 | [tk]will do |
| 17:47.06 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 17:52.11 | buttons840 | anyone know where you can get the source (is it available?) PHP for nerdvittles teleyapper? |
| 17:56.22 | [TK]D-Fender | buttons840: Go check NerdVittles |
| 17:56.37 | [TK]D-Fender | thinks that sounds like a dog-food site... |
| 17:56.49 | buttons840 | i have been, but no success, only install scripts |
| 17:57.18 | *** join/#asterisk yziquel (i=53acc979@gateway/web/freenode/session) |
| 17:57.36 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:57.36 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 17:57.37 | buttons840 | i'm told teleyapper is able to detect when call has been answered over fxo, i want to see their code |
| 17:59.29 | WindowsUser | buttons840: the install script has to get the .php unless they're shipping precompiled php |
| 18:00.55 | *** join/#asterisk sjobeck (n=Adium@c-71-193-223-71.hsd1.or.comcast.net) |
| 18:02.17 | *** part/#asterisk sjobeck (n=Adium@c-71-193-223-71.hsd1.or.comcast.net) |
| 18:02.22 | kaii | buttons840: if you read the first 5 lines of the install script carefully, you will eventually notice the following URL: |
| 18:02.25 | kaii | buttons840: http://pbxinaflash.net/scripts/teleyapper.tgz |
| 18:02.36 | kaii | in it, you will find the PHP sources |
| 18:02.43 | buttons840 | thank you |
| 18:02.47 | kaii | (php is mostly always source, though) |
| 18:03.15 | *** join/#asterisk borga454 (n=alex@80.172.25.71) |
| 18:03.43 | borga454 | a |
| 18:05.08 | kaii | reading the teleyapper install script, i hope the app itself is not as badly written as the installer :) |
| 18:06.52 | borga454 | Hello all, I am quite new in asterisk and I am trying to create a dialplan that executes the following steps: |
| 18:06.53 | borga454 | 1.A sip friend dials 102 extension. |
| 18:06.53 | borga454 | 2.Asterisk PBX responds with some beeps. |
| 18:06.53 | borga454 | 3.The sip friend hangs up the phone. |
| 18:06.53 | borga454 | 4.Asterisk PBX calls back to the sip friend after 30 seconds with the application music on hold. |
| 18:06.54 | borga454 | I tried to implement this using h extension but I got the following message: |
| 18:06.56 | borga454 | Spawn extension (internal, h, 1) exited non-zero on 'SIP/bt100-083b8e60' |
| 18:06.58 | borga454 | I used also dial local but the result was the same. :( |
| 18:07.00 | borga454 | I suppose that when I pick up the phone asterisk creates thread and when I hang up the thread stops. |
| 18:07.04 | borga454 | Am I think correctly?? |
| 18:07.06 | borga454 | Is there any way of executing those steps only using the dialplan?? |
| 18:10.17 | WindowsUser | borga454: make a callfile |
| 18:14.58 | WindowsUser | borga454: http://www.voip-info.org/wiki/view/Asterisk+Documentation+1.6.1+callfiles.txt give that a read |
| 18:15.14 | leifmadsen | WindowsUser: get your SFA all setup? |
| 18:15.38 | WindowsUser | borga454: you could use system() to make, touch, move a file |
| 18:15.59 | *** join/#asterisk Chinorro (n=Chino@54.225.117.91.dynamic.mundo-r.com) |
| 18:16.01 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 18:16.14 | *** join/#asterisk hardwire (n=hardwire@216-67-98-253.static.acsalaska.net) |
| 18:16.22 | WindowsUser | borga454: i'm typing this on channel as thats what its for and I might get distracted for awhile :) |
| 18:16.33 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:16.33 | hardwire | so I have like 20 thousand lines of "We're Zap 60 not" in my asterisk full log |
| 18:16.39 | hardwire | and the high load to prove it. |
| 18:16.42 | hardwire | any idea what causes that? |
| 18:16.43 | WindowsUser | leifmadsen: its set up but i cant call to playback or voicemail |
| 18:16.45 | borga454 | oki :) |
| 18:17.00 | leifmadsen | WindowsUser: really? hmm.... mine calls into the IVR and rings through to my phone no problem |
| 18:17.12 | leifmadsen | asterisk 1.4 with x86_64 1.4 version of SFA |
| 18:17.51 | WindowsUser | ah |
| 18:17.54 | WindowsUser | I'm on 1.6 |
| 18:18.46 | leifmadsen | I haven't tried the 1.6 SFA yet. Haven't had time to upgrade my home system |
| 18:19.13 | sirus | wants to be able to dial other sip phones |
| 18:19.19 | sirus | for right now I can only do extensions |
| 18:20.07 | borga454 | ok, so from what I understood, using just the dialplan for call back is impossible. I will use deadAgi and a script. What do tou think? |
| 18:20.59 | leifmadsen | borga454: what version of asterisk? |
| 18:21.05 | borga454 | 1.4.26 |
| 18:21.28 | leifmadsen | borga454: impossible with that version. I think 1.6.1.x first introduces Originate() dialplan application |
| 18:21.40 | leifmadsen | you'll need to use an external AGI() script to create a callfile |
| 18:22.17 | borga454 | ok, thanks ;) |
| 18:25.32 | WindowsUser | sirus: what do you mean? |
| 18:25.35 | *** part/#asterisk swaj (n=scott@unaffiliated/swaj) |
| 18:27.36 | *** join/#asterisk outtolunc (n=me@c-98-248-96-110.hsd1.ca.comcast.net) |
| 18:29.29 | [TK]D-Fender | You dial extensions. Extensions dial devices... |
| 18:31.03 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 18:31.56 | hardwire | turns off threewaycalling and advanced dtmf phone features |
| 18:37.58 | Kobaz | hmmmm |
| 18:38.30 | Kobaz | doing a TrySystem in an h extension, will cause asterisk to take down the box it's on |
| 18:40.35 | *** join/#asterisk mackarry (n=servitro@200.58.208.157) |
| 18:40.36 | *** join/#asterisk Greek-Boy (n=greek@41.188.154.137) |
| 18:40.51 | *** part/#asterisk mackarry (n=servitro@200.58.208.157) |
| 18:40.52 | Kobaz | actually... doing a TrySystem anywhere, will cause asterisk to take down the box |
| 18:41.24 | Kobaz | <PROTECTED> |
| 18:41.33 | Kobaz | and then hard lock |
| 18:43.03 | [TK]D-Fender | \o/ |
| 18:44.00 | Qwell | Kobaz: issues.asterisk.org |
| 18:44.01 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 18:44.10 | Kobaz | yeah |
| 18:44.22 | sirus | WindowsUser: I need to configure my extensions.conf to allow me to dial other people with sip phones not on my network |
| 18:45.03 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 18:45.41 | WindowsUser | like SIP/bob@google.com? |
| 18:48.42 | [TK]D-Fender | ^^^^Sir |
| 18:50.04 | Katty | plops |
| 18:52.39 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 18:54.50 | WindowsUser | http://pastebin.ca/1512570 looks like the 1.4.21 debian package is a no go for SFA |
| 18:55.10 | *** join/#asterisk ArchGT (n=ArchGT@unaffiliated/archgt) |
| 18:55.33 | WindowsUser | leifmadsen: what version of asterisk are you using? (sorry if I've asked this already, internets been crapping out) |
| 19:01.31 | *** join/#asterisk stack_ (n=sgerstac@mail.edpaymentsystems.com) |
| 19:02.37 | buttons840 | what is the meaning of this GotoIf($["foo${MSG}" = "foo"]?s,6:s,8) particularly the foo${MSG} part, i'm rusty on variable syntax, i know var is the references and $var is the value, but what is foo${MSG} ? |
| 19:03.04 | stack_ | I put together an Asterisk box with PRI a long time ago. I don't know much about Asterisk, just enough to get it running. We are moving to a new building. Our PRI is currently set to 5ess for the switch type. The phone company says I'll have to switch to DMS100 for the new building. Is this a problem or do I just have to make the change in zapata.conf to dms100 |
| 19:03.06 | buttons840 | reads the book while waiting for a response |
| 19:03.13 | [TK]D-Fender | buttons840: same thing with 3 letters in front |
| 19:03.37 | [TK]D-Fender | stack_: Latter |
| 19:03.54 | stack_ | [TK]D-Fender: cool, thanks |
| 19:03.59 | buttons840 | [TK]D-Fender, so the logical condition is if MSG is blank? |
| 19:04.06 | leifmadsen | WindowsUser: np -- I'm using a recent 1.4 checkout from svn |
| 19:04.11 | leifmadsen | on a 64-bit proc |
| 19:04.35 | leifmadsen | buttons840: it's meant to check to see if ${MSG} is null |
| 19:04.37 | [TK]D-Fender | buttons840: replace the text raw... $["foo${MSG}" = "foo"] |
| 19:04.56 | buttons840 | ok, thanks |
| 19:05.01 | [TK]D-Fender | buttons840: empty var = no mor chars = substitute and see that you are comparing 2 eual things |
| 19:05.06 | leifmadsen | although that syntax is unnecessary, because just $["${MSG}" = ""] would have been sufficient |
| 19:05.23 | buttons840 | leifmadsen, my thoughts exactly |
| 19:05.31 | [TK]D-Fender | leifmadsen: Yup, a guy who remembered non-quote add-ons and decided to embellish it... |
| 19:05.42 | leifmadsen | alternatively, could have done: $[foo${MSG} = foo] |
| 19:05.48 | [TK]D-Fender | kicks *'s expression parser in the nads |
| 19:05.52 | *** part/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 19:06.04 | leifmadsen | the foo is because you can't have $[${MSG} = jimmy] if ${MSG} could be blank |
| 19:06.09 | buttons840 | btw, that code is from nerdvittles dial plan logic |
| 19:06.10 | [TK]D-Fender | leifmadsen: At least "" feels more natural... |
| 19:06.19 | leifmadsen | yep |
| 19:06.31 | *** part/#asterisk stack_ (n=sgerstac@mail.edpaymentsystems.com) |
| 19:06.39 | [TK]D-Fender | buttons840: Nervittles does code for FreePBX which I take as an immediate sign of their IQ..... |
| 19:07.03 | buttons840 | opinion noted ;) |
| 19:07.49 | buttons840 | my initial impression (as someone who's never used it) of FreePBX was high, but i decided to avoid the use of such tools and work with asterisk directly |
| 19:09.58 | [TK]D-Fender | buttons840: the tool is OK as long as you understand what you are working with and don't try to fight the system. |
| 19:10.21 | buttons840 | yes, i was afriad i woudl rely on the tool if i started learning by using it |
| 19:10.23 | [TK]D-Fender | buttons840: Key problem is that when you cater to morons, you attract them. And they are without boundaries. |
| 19:11.15 | [TK]D-Fender | "Isn't the just some sort of script (generic term meaning anything) that I can click and just work? NO!??! WHY!?!?! PLAESE!@ :S" |
| 19:11.51 | [TK]D-Fender | I **LOVE** the ":S" blank-faced smilies the best. the sign of nothing going on upstairs... |
| 19:12.15 | buttons840 | also the line t,1,Goto(talk,1) has got me wondering, i think this means "just sit there until they hang up", but is this good practice :S |
| 19:12.17 | [TK]D-Fender | And then when they run at the mere prospect of even having to touch jsut 1 config file a little |
| 19:12.39 | [TK]D-Fender | ~cluebat buttons840 |
| 19:12.40 | infobot | ACTION pulls out a ClueBat (tm) and thwaps buttons840. |
| 19:12.57 | buttons840 | gets a clue, but not sure what the clue is about |
| 19:13.10 | [TK]D-Fender | buttons840: horrible practice and since we don't see where it points to... |
| 19:13.16 | leifmadsen | buttons840: t is the 'timeout' extension |
| 19:13.42 | buttons840 | ohh, i though t was talk |
| 19:13.51 | buttons840 | braces for cluebat |
| 19:14.06 | *** join/#asterisk Micc (n=dotirc@c-76-121-255-52.hsd1.wa.comcast.net) |
| 19:15.06 | *** join/#asterisk Micc (n=dotirc@c-76-121-255-52.hsd1.wa.comcast.net) |
| 19:16.47 | Micc | I have a customer with a sonicwall firewall and some ip450 polycom phones having some issues. It doesn't seem to keep the connection. I'm guessing its something to do with NAT or something else on the sonicwall. I've tried setting the nat interval to 25 second and setting register="1" and expire="120" but the phones still go unregistered for 10 or 20 minutes sometimes. |
| 19:17.23 | [TK]D-Fender | Micc: Disable the SonicWALL's SIP transform features, and set "qualify=yes" for the peer |
| 19:17.52 | *** join/#asterisk Failrar (n=Failrar@coffee.ipv6.goatse.be) |
| 19:19.21 | Micc | TKD-Fender, transform was originally off but we had some other problem I think. I'll try it again and see what happens. |
| 19:19.54 | Micc | I didn't have qualify on at the time. |
| 19:21.35 | *** join/#asterisk errotan (n=errotan@62.201.123.35) |
| 19:22.34 | *** join/#asterisk jmacz (n=mcorb@190.144.75.22) |
| 19:23.03 | leifmadsen | I've never had success with a sonicwall + voip |
| 19:23.59 | Micc | Thats disappointing. |
| 19:24.52 | *** join/#asterisk voxter (n=voxter@201.200.82.226) |
| 19:27.02 | [TK]D-Fender | I run one NOW |
| 19:27.55 | Katty | just got natting working between an el crapo linksys RVO082 and an elcrapper WRT54G router. |
| 19:27.58 | bmoraca | i've never been able to get SIP to work properly with a sonicwall |
| 19:28.19 | Kobaz | s/with a sonicwall// |
| 19:29.01 | Katty | bandwidth -> firewall -> server -> firewall -> router -> polycom -> router -> firewall -> server -> bandwidth -> me |
| 19:29.23 | Katty | FUN |
| 19:29.50 | bmoraca | wow...what moron designed that network :P |
| 19:29.57 | Katty | yours truly. |
| 19:30.03 | eric_hill | Sonicwall tip: http://your.sonicwall.ip.address/diag.html then Advanced Prefs |
| 19:30.30 | Katty | i was just describing a call from a cell phone, to their network, transfered to a remote extension, blind transfered out to my mobile. |
| 19:30.44 | *** join/#asterisk vgster (n=vgster@94-194-190-189.zone8.bethere.co.uk) |
| 19:30.52 | Katty | mobile -> did -> ext -> ext -> mobile |
| 19:31.04 | bmoraca | eric_hill: i've found that it's quicker, easier, and cheaper to just plain get rid of the sonicwall and replace it with something else, like a Cisco ASA or an Adtran NetVanta |
| 19:31.15 | Katty | with a firewall and a router between ext and ext. |
| 19:31.17 | bmoraca | Katty: ahhh. |
| 19:31.25 | Katty | out in the BOONIES. |
| 19:31.26 | Katty | on charter. |
| 19:31.39 | eric_hill | bmoraca: Indeed. We replaced our SonicWall with a Cisco ASA. It's still in the rack though... :) |
| 19:31.47 | bmoraca | *shutters* all bad experiences with charter |
| 19:31.54 | Katty | yes. |
| 19:31.57 | Katty | i am still amazed it works. |
| 19:32.04 | Katty | at first it was porked |
| 19:32.12 | Katty | and the firewall kept spewing CONNECTION POLICY VIOLATION |
| 19:32.35 | Katty | so i had to give the firewall a cookie. |
| 19:32.44 | bmoraca | we've had charter shut down a few circuits to people because they said the routers' DHCP wasn't "working properly" |
| 19:32.45 | Katty | and it was appeased. |
| 19:33.17 | Katty | that's not cool. |
| 19:33.31 | buttons840 | so, as we talked about earlier, when running an originate command in asterisk over a FXO line, it will imediately consider the line as answered and begin dial plan logic before the receiving end actually answers... is this a behavior unique to asterisk, would other PBX software be able to work around this, or is it an inherent part of FXO lines? |
| 19:33.35 | bmoraca | yeah...it's weird |
| 19:33.39 | Katty | so. |
| 19:33.45 | Katty | just wondering |
| 19:33.53 | Katty | who makes faxes with sip registrations |
| 19:34.08 | Katty | if not. someone should invent it. |
| 19:34.23 | eric_hill | Katty: T38 over SIP? :) |
| 19:34.27 | bmoraca | faxes with sip registrations? you mean faxes over sip trunks? |
| 19:34.37 | Katty | yeah |
| 19:34.41 | Katty | like a voip...fax... |
| 19:34.42 | Katty | machine |
| 19:34.43 | Katty | thing |
| 19:34.47 | bmoraca | or do you mean a fax machine with a built in sip client |
| 19:34.50 | bmoraca | that'd be sweet |
| 19:35.24 | eric_hill | Fax over a clear 64k IP voice channel still takes longer than T38 on both ends. |
| 19:35.41 | Katty | i don't care what it takes. |
| 19:35.43 | eric_hill | I think a fax machine with a built in SIP client is called a computer with a scanner and email. :) |
| 19:35.43 | Katty | it would be neat. |
| 19:35.56 | Katty | too complicated for idiots |
| 19:35.59 | Katty | it has to look like a fax machine |
| 19:36.05 | [TK]D-Fender | buttons840: I've already answered that in full. Scroll up |
| 19:36.16 | bmoraca | Katty: why not just use an ATA? |
| 19:36.21 | Katty | that's not the point |
| 19:36.32 | Katty | the point is to use a sip registration like from bandwidth, with a fax machine. |
| 19:36.47 | Katty | that's what i want. |
| 19:37.19 | Katty | i should call kyocera and tell them to design one |
| 19:37.56 | bmoraca | apparently NEC makes a SIP-FAX module for one of their fax machines |
| 19:38.02 | Katty | orly |
| 19:38.04 | Katty | googles |
| 19:38.31 | bmoraca | hmmm...looks like an ATA |
| 19:39.08 | Katty | yeah i was looking at the spec sheet |
| 19:39.31 | Katty | oh well. |
| 19:39.52 | Katty | OH! |
| 19:39.52 | Katty | dangit |
| 19:39.52 | bmoraca | it's a step forward, though, eh? |
| 19:39.52 | Katty | where's seanmh when i need him. |
| 19:39.59 | buttons840 | [TK]D-Fender, indeeed, you said "FXO lines are considered answered immediately" now i'm asking if this is a consideration determined by asterisk or by the nature of fxo itself, that's all |
| 19:40.11 | Katty | bmoraca: yesh. |
| 19:40.12 | [TK]D-Fender | buttons840: KEEP SCROLLING UP. |
| 19:40.15 | Katty | bmoraca: someone will make it eventually. |
| 19:40.19 | [TK]D-Fender | buttons840: I explain this crap in full |
| 19:40.25 | Katty | [TK]D-Fender: you're crap. |
| 19:40.27 | Katty | [TK]D-Fender: in full. |
| 19:40.31 | Katty | [TK]D-Fender: <3 |
| 19:40.32 | *** join/#asterisk wasserja (n=wasserja@corporate.volstate.net) |
| 19:40.35 | [TK]D-Fender | goes to the bathroon |
| 19:40.41 | [TK]D-Fender | s/n/m |
| 19:40.42 | Katty | *hee* |
| 19:41.13 | [TK]D-Fender | Katty: Yesterday's vegetarian Indian lunch took a full day to pass through the system and none of us had dinner :) |
| 19:41.27 | Katty | i didn't need to know that. |
| 19:41.53 | Katty | SHUUUUUUUUUUUUUUSH |
| 19:41.59 | [TK]D-Fender | Katty: First hit's free ;) |
| 19:42.10 | manxpower | buttons840: it is the nature of FXO. |
| 19:42.20 | manxpower | The TELCO does not signal an answer on the far wnd |
| 19:42.22 | manxpower | and end too |
| 19:42.38 | [TK]D-Fender | buttons840: its the defult nature of chan_dahdi.so / chan_zap.so |
| 19:42.50 | [TK]D-Fender | buttons840: and is selectable. NOW READ. |
| 19:43.05 | buttons840 | i've read that convo above twice |
| 19:43.30 | manxpower | buttons840: rather than wasting your time on FXO, just get a SIP line. Cheap, easy, simple. |
| 19:44.22 | buttons840 | manxpower, i'm happy with that answer, but my client isn't |
| 19:44.25 | manxpower | You're trying to make a Ford Pinto go 100Mph. Sure you might be able to do it, but you will spend more (in time and money) than you would spend to get a decent car. |
| 19:44.37 | eric_hill | Speaking of SIP: Anyone have any suggestions for a good SIP provier that could port a cell number over? |
| 19:44.47 | Katty | eric_hill: bandwidth has for us. |
| 19:44.48 | [TK]D-Fender | manxpower: Gas up and get rear-ended. You'll hit 300mph ;) |
| 19:44.51 | Katty | eric_hill: they're sometimes cranky tho |
| 19:44.59 | [TK]D-Fender | manxpower: or at least little bits of you will ;) |
| 19:45.00 | manxpower | buttons840: They are what I call a "Wisher". They want something so much they won't accept it can't be done. |
| 19:45.18 | jaytee | Katty!!!! where ya been? |
| 19:45.24 | [TK]D-Fender | manxpower: But why?!?! PLAESE HALP!?! :S |
| 19:45.28 | manxpower | buttons840: turn on callprogress accept that calls will randomly not wokr |
| 19:45.39 | Katty | jaytee: hiding :> |
| 19:45.51 | Katty | jaytee: sick yesterday :< |
| 19:45.51 | manxpower | (usually happens with people that talk loud or have a higher pitched voice) |
| 19:46.24 | manxpower | But I don't think any of us are going to spend any more time on this question. You have your answer. If you or your client don't like the answer, it does not make the answer false. |
| 19:46.37 | Katty | optionally, hire some help. |
| 19:47.22 | buttons840 | manxpower, i'm not claiming it's false, i asked 1 additional question, which was "is this the nature of FXO itself" and that has now been answered |
| 19:48.39 | Katty | excellent! |
| 19:48.43 | Katty | let's go have drinks now. |
| 19:48.56 | jaytee | you buyin? |
| 19:49.09 | buttons840 | manxpower, part of my frustration is that the client believes teleyapper from nerd vittles is able to detect when a call is answered on FXO, but from my investigation into the source, this is not true |
| 19:49.14 | Katty | jaytee: for you i will. |
| 19:49.43 | buttons840 | well, sorry, i've nothing more to say besides idle chit-chat if anyone cares |
| 19:49.53 | jaytee | buttons840, perhaps [TK]D-Fender will loan you his ClueBat(tm) so you can beat some sense into your client |
| 19:49.58 | Katty | idle chitchat is all i do. |
| 19:50.00 | Katty | anymore. |
| 19:50.11 | manxpower | buttons840: No, it can determine the NEAR END answer (asterisk side), not the FAR END (remote side) |
| 19:50.12 | jaytee | hugs Katty |
| 19:50.43 | manxpower | so telezaper knows when ASTERISk answers, not when the far end answers when calling out. |
| 19:50.45 | telnettech | Katty: I am searching for a new home to ply my trade..... |
| 19:50.46 | jaytee | Katty, well just so you know.....you were missed |
| 19:50.57 | buttons840 | manxpower, he believes it can determin the far end though some sort of program logic, which might be true, i can't see how something like that can be achieved in a 400 line php script |
| 19:51.01 | Katty | telnettech: baroo? |
| 19:51.03 | Katty | jaytee: :> |
| 19:51.23 | manxpower | buttons840: it can't. all that stuff is DSP stuff. |
| 19:51.28 | jaytee | it hasn't been very lively in here all week. I don't think eppigy has said TRABAJO! once even. |
| 19:51.37 | Katty | :< |
| 19:51.38 | buttons840 | what is DSP? |
| 19:51.44 | *** join/#asterisk jack_blunt_ (n=ssharma@ool-18bb1208.dyn.optonline.net) |
| 19:51.50 | Katty | eppigy: PING |
| 19:51.54 | Katty | he is idle on messenger. |
| 19:52.07 | buttons840 | signal processing, got it |
| 19:52.17 | jack_blunt_ | folks I need some help on ubuntu 9.04 .. running asterisk 1.6 .. new install |
| 19:52.19 | manxpower | it's digital signal processing. It's the part of telecom hardware that Asterisk does in software that other companies do in hardware (and much more expensive) |
| 19:52.33 | Katty | jack_blunt_: orly |
| 19:52.41 | jack_blunt_ | <PROTECTED> |
| 19:53.16 | jaytee | ya gotta love dem packages! |
| 19:53.21 | buttons840 | jack_blunt_, huh? can't you just sudo nano /etc/asterisk/asterisk.conf ? |
| 19:53.30 | jack_blunt_ | i know .. I can su .. but should i |
| 19:53.31 | manxpower | jack_blunt_: you did NOT install Asterisk. You installed an ASTERISK GUI. Calling an Asterisk GUI "Asterisk" is like calling Windows 95 "DOS". Find a channel to support your Asterisk based GUI |
| 19:53.37 | Katty | su is good. |
| 19:53.44 | Katty | i would never "Log In" as root |
| 19:53.46 | jack_blunt_ | no i did install asterisk .. it runs as a daemon |
| 19:53.47 | buttons840 | yes, su or change permissions on the files |
| 19:54.09 | jack_blunt_ | now where the heck is the password for asterisk account stored? how does one change that! |
| 19:54.13 | jaytee | he just installed asterisk as a debian type package instead of compiling from source |
| 19:54.23 | bmoraca | i wouldn't recommend changing the permissions...but sudo would be good...and you don't need to touch the asterisk account |
| 19:54.28 | Katty | oh :< |
| 19:54.30 | Katty | skeery. |
| 19:54.30 | jack_blunt_ | jaytee: you are right .. |
| 19:54.36 | Katty | jack_blunt_: i have a better idea! |
| 19:54.45 | jack_blunt_ | go ahead katty -- |
| 19:54.50 | buttons840 | why is su "skeery" ? |
| 19:54.51 | manxpower | jack_blunt_: your question is a distro question |
| 19:54.52 | bmoraca | it's not stored anywhere you access or change it...nor should you. |
| 19:54.53 | jaytee | and asterisk as a package sucks rancid rhino diarrhea in most any distro |
| 19:55.16 | jack_blunt_ | so are you saying short cut was a long way to home ? |
| 19:55.17 | [TK]D-Fender | jack_blunt_: "su asterisk -", or jus chown the files |
| 19:55.58 | jaytee | jack_blunt_, do what [TK]D-Fender just suggested. |
| 19:56.00 | manxpower | jack_blunt_: No, we are saying the shortcut put you in a Louisiana swamp instead of Chicago. |
| 19:56.14 | manxpower | you can see how that could be a problem. 9-| |
| 19:56.43 | jaytee | the shortest distance between two points is a straight line but since according to Relativity spacetime is curved it follows that there are no shortcuts. |
| 19:56.50 | jack_blunt_ | I don't know if being dead in chicago is any better than sucking water in a swap! |
| 19:56.59 | Katty | jaytee: that's only true in a 3d universe. |
| 19:57.15 | Katty | jaytee: the shortest distance between two points is folding the paper in half :P |
| 19:57.16 | jaytee | Katty, what? you live in the Land Of Point? |
| 19:57.20 | manxpower | jack_blunt_: I spent 10 years in New Orleans. Trust me, Chicago is better than the swamp |
| 19:57.22 | [TK]D-Fender | jaytee: With folder space two points cound be in the same space. |
| 19:57.34 | [TK]D-Fender | hits Warp 10 |
| 19:57.38 | jack_blunt_ | i don't want to be at either place -- bad politicians in both towns |
| 19:57.41 | Katty | MAKE IT SO |
| 19:57.50 | [TK]D-Fender | TEA |
| 19:57.52 | [TK]D-Fender | EARL GREY |
| 19:57.54 | [TK]D-Fender | HOT |
| 19:57.54 | Katty | HOT |
| 19:58.03 | buttons840 | ha ha ha! |
| 19:58.10 | manxpower | thinks it is now time for him to go work on his tan |
| 19:58.20 | Katty | manxpower: don't forget sunscreen and goggles. |
| 19:58.36 | Katty | ManxAway: and wipe the bed down!!!! |
| 19:58.41 | Katty | those things are nasty. |
| 19:58.46 | jack_blunt_ | ok .. folks it looks like my big issue is non issue -- any recommendations on keeping it going -- i am just playing around no production pressure |
| 19:58.51 | iflux | gives fender some peppermint and some ice |
| 19:59.00 | ManxAway | Katty: Out by the pool, not under some UV lamp |
| 19:59.09 | Katty | ManxAway: still take sunscreen. |
| 19:59.16 | bmoraca | jack_blunt_: read the book... |
| 19:59.27 | bmoraca | ~book |
| 19:59.28 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 19:59.28 | jack_blunt_ | i read the book .. it's not "that" good |
| 19:59.57 | Katty | jack_blunt_: i'm sure you can find someone who will come out on site and train you (= |
| 20:00.09 | Katty | jack_blunt_: or contract with smoeone for support |
| 20:00.17 | bmoraca | i'd do it, except that i'm in maui next week on my honeymoon |
| 20:00.27 | bmoraca | oh, and it'd be expensive :P |
| 20:00.34 | Katty | honeymoon |
| 20:00.38 | jack_blunt_ | for playing around katty .. i don't need support for that .. my hands are good :-) |
| 20:00.42 | KavanS | jack_blunt_, goto digium's website, I'm sure they have some good info there ;) |
| 20:00.51 | KavanS | lol irc flirting...nerds. |
| 20:00.56 | sirus | Do you have to assign extension to a sip phone off your network in order to call it? |
| 20:00.56 | KavanS | :P |
| 20:01.17 | Katty | sirus: how else would you call it? |
| 20:01.24 | Katty | sirus: polycoms can call via IP address |
| 20:01.31 | Katty | sirus: no server required |
| 20:01.48 | sirus | hrmm ok |
| 20:01.50 | bmoraca | i don't even know where to start with that question...yikes |
| 20:02.06 | sirus | Katty: you have a phone you can test with me? |
| 20:02.13 | Katty | umm |
| 20:02.17 | bmoraca | lol |
| 20:02.23 | [TK]D-Fender | sirus: And where would we call? |
| 20:02.55 | eric_hill | [TK]D-Fender: Pretty sure his number is 911. |
| 20:02.56 | sirus | The number I assign |
| 20:02.57 | Katty | i have to go now. |
| 20:02.59 | Katty | my uhhh |
| 20:03.01 | Katty | house is on fire |
| 20:03.02 | Katty | and my dog |
| 20:03.03 | Katty | just |
| 20:03.07 | Katty | ate a ferret |
| 20:03.07 | Katty | yeah |
| 20:03.13 | Kobaz | hmm |
| 20:03.16 | sirus | yeah yeah |
| 20:03.25 | Kobaz | i can't find how to convert stereo to mono with sox |
| 20:03.26 | Katty | and jaytee needs coffee :P |
| 20:03.28 | Katty | disappears |
| 20:03.30 | sirus | goes back to the pdf |
| 20:03.33 | Katty | Kobaz: oh! |
| 20:03.36 | Katty | Kobaz: i have a wiki on that. sec |
| 20:04.17 | Kobaz | it's not anywhere in the man page |
| 20:04.31 | eric_hill | Kobaz: sox -c 2 stereo.wav -c 1 mono.wav -avg -l |
| 20:04.45 | jack_blunt_ | thanks folks .. ciao for now |
| 20:04.51 | Kobaz | k |
| 20:05.39 | Katty | well nevermind. my dydns is doomy. |
| 20:05.39 | Kobaz | sox sox: Volume value `g' is not a number |
| 20:05.51 | Kobaz | doesn't seem that -avg is a valid parameter |
| 20:05.53 | Katty | but i did it recently, itmight be in my back log |
| 20:05.54 | Kobaz | -a is though |
| 20:06.26 | Katty | sox foo-in.wav -r 8000 -c 1 -s -w foo-out.wav resample -ql |
| 20:06.34 | Kobaz | also -l is not a valid option either |
| 20:06.36 | Kobaz | Katty: k |
| 20:06.36 | *** join/#asterisk RaNaji (n=WinNT@rrcs-76-79-99-213.west.biz.rr.com) |
| 20:06.47 | Katty | optionally |
| 20:06.50 | Katty | sox foo.wav -r 8000 -c1 foo.gsm resample -ql |
| 20:07.06 | jaytee | word of advice, never buy a Point of Sale software system that's written in Delphi by Amish programmers. |
| 20:07.15 | Kobaz | invalid option -w |
| 20:07.29 | eric_hill | jaytee: What's wrong with Amish Delphi users? |
| 20:07.30 | Katty | do you have libsox-fmt-gsm installed |
| 20:07.35 | jaytee | why not just use file convert from the CLI? |
| 20:07.38 | Kobaz | maybe |
| 20:07.40 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 20:07.44 | Kobaz | jaytee: because it needs a mono input file |
| 20:07.46 | eric_hill | jaytee: That sounds like the intro to a really bad joke. |
| 20:07.48 | jaytee | ah |
| 20:07.59 | RaNaji | hi new to astrisk i want to take my house analog line and set it up with my asterisk box what card do i need i want something cheap just for one line |
| 20:08.04 | jaytee | eric_hill, if it is then I'm living the bad joke as we speak |
| 20:08.04 | Katty | if you're using it for asterisk, just turn it into a gsm |
| 20:08.06 | Katty | it will be easier. |
| 20:08.14 | Kobaz | that never works for me |
| 20:08.21 | Kobaz | any time i save as a gsm, asterisk can't play it |
| 20:08.36 | Kobaz | but if i save as wav, and convert to gsm/ulaw, it works fine |
| 20:08.44 | Katty | yeah. |
| 20:08.45 | Katty | uhmm. |
| 20:08.54 | Katty | i've had some issues, but usually it was me being retarded. |
| 20:09.00 | Katty | like Playback(foo.gsm) <- dumb |
| 20:09.11 | Katty | playback(foo) <- with the wav and gsm in the same directory |
| 20:09.13 | Katty | ^- also dumb |
| 20:09.20 | Kobaz | heh, half of these options to sox you guys gave me are valid |
| 20:09.20 | [TK]D-Fender | [16:02]<sirus>The number I assign <- smrt... |
| 20:09.28 | Kobaz | i'll just convert to mono with audacity |
| 20:09.36 | Katty | kay |
| 20:09.43 | bmoraca | RaNaji: you need an FXO interface card. there are several kinds...i'd recommend Sangoma or Digium...Sangoma also makes a 2-port FXO USB card...never used it though. |
| 20:10.07 | Katty | RaNaji: when it comes to business phone systems and 911...we don't do cheap usually. |
| 20:10.23 | Katty | RaNaji: cheap comes back to haunt. |
| 20:10.31 | Katty | OoooOOOO HOLY GHOST |
| 20:10.34 | Katty | holy ghost!!! |
| 20:10.36 | [TK]D-Fender | rajiv: Linksys SPA-3102 |
| 20:11.03 | Katty | holy ghost, this is not an episode of Scooby Doo! |
| 20:11.20 | *** join/#asterisk Mango (n=Mango@S01060016b6853255.vc.shawcable.net) |
| 20:11.40 | Katty | i would have suceeded if it wasn't for those pesky God and Jesus fellows! |
| 20:12.19 | Katty | we will now sing hymn 405. oh god, what on earth is my hairdo all about. |
| 20:12.25 | KavanS | does digium run this channel? or is this run by the userbase? |
| 20:12.35 | Katty | this channel is run by monkeys. |
| 20:12.39 | bmoraca | wow...never realized that Bombay Sapphire was 94 proof...that seems high for gin... |
| 20:12.42 | Kobaz | rabbits too |
| 20:12.47 | Kobaz | don't forget the rabbits |
| 20:12.52 | Katty | Kobaz: with pocket watches. |
| 20:12.59 | KavanS | nice, seriously. |
| 20:13.01 | Katty | oh! and the mice. |
| 20:13.05 | Katty | we can't forget the mice. |
| 20:13.11 | KavanS | I'll guess I'll just look on the web |
| 20:13.14 | Katty | KavanS: it's run by an IRC administrator. |
| 20:13.16 | KavanS | since apparently we are not on the same topic... |
| 20:13.19 | Katty | KavanS: go to freenode.net and figure out who |
| 20:13.19 | KavanS | Katty, that is clear.. |
| 20:13.19 | eric_hill | Katty, send me your address and I'll send you some medication. You need it. |
| 20:13.33 | KavanS | Katty, nevermind... |
| 20:13.34 | Katty | eric_hill: oh please. |
| 20:13.36 | *** join/#asterisk Anth8708 (n=Anth8708@client105.jdcc.edu) |
| 20:13.36 | KavanS | I'll ask someone else... |
| 20:13.39 | KavanS | yeah |
| 20:13.41 | KavanS | indeed eric_hill |
| 20:14.19 | Katty | ya'll need to lighten up (= |
| 20:14.26 | bmoraca | the channel is neither run by nor endorsed by Digium, as far as I can tell. it's just a forum where people who like and use Asterisk can hang out and discuss issues. |
| 20:14.30 | Katty | else life isn't worth living. |
| 20:14.34 | KavanS | thank you bmoraca |
| 20:14.56 | Mango | I give up Googling. How do I set a SIP header in extensions.conf with Asterisk 1.4? I am trying to set up Distinctive Ring on some Linksys devices and need to set the ALERT_INFO header. |
| 20:14.57 | bmoraca | what difference does it make, though? people are here that will answer the questions...whether they're from digium or not makes no difference |
| 20:15.08 | KavanS | bmoraca, no just wondered....I was curious |
| 20:15.18 | Katty | Mango: first, you need to know what Keyword you need for your linksys device. |
| 20:15.20 | seanbright | Mango: core show application SIPAddHeader |
| 20:15.26 | Mango | seanbright: Thanks! |
| 20:15.42 | Mango | Wonder why I couldn't find that with Google. |
| 20:15.54 | Katty | Mango: and i will send you a blog link. |
| 20:15.56 | KavanS | Katty, I appreciate the "lighten up", heh...but you could lighten up by not being sarcastic with people...it reflects negatively on your character...a joke is better if both sides are willing participants |
| 20:16.24 | jaytee | whoa! check out the tude on that dude! |
| 20:16.31 | bmoraca | here you go: http://www.lmgtfy.com/?q=asterisk%20add%20sip%20header |
| 20:16.37 | scoof | KavanS: it should be pretty obvious from "It is an excellent place for Asterisk users to meet to discuss Asterisk and receive support from knowledgeable users." on http://www.asterisk.org/support |
| 20:16.52 | KavanS | right on scoof! |
| 20:17.24 | Katty | i'll have to keep it up then! |
| 20:17.28 | KavanS | where's [TK]D-Fender? |
| 20:17.29 | KavanS | lol |
| 20:17.30 | Katty | keep all those negative folks away from me! |
| 20:17.33 | KavanS | normally he's down to business |
| 20:17.39 | Katty | emites negativity the rest of the afternoon |
| 20:17.39 | scoof | KavanS: also from the support-prices from digium :) |
| 20:17.47 | KavanS | scoof, heh reading on their site now |
| 20:17.53 | scoof | Katty: e-mites? |
| 20:17.59 | KavanS | emits? |
| 20:18.00 | Katty | yes. |
| 20:18.01 | Katty | emites. |
| 20:18.04 | Katty | no, not emits. |
| 20:18.05 | Katty | emites. |
| 20:18.14 | Katty | nevermind. long story. |
| 20:18.23 | scoof | Katty: mites that eat your structured cabling? |
| 20:18.32 | Katty | not quite. |
| 20:18.38 | Katty | but you can pretend if you want! |
| 20:18.43 | bmoraca | no, they're the mites that live in sonicwall firewalls that eat sip headers |
| 20:18.56 | KavanS | bmoraca, sonicwalls do that? |
| 20:18.56 | Katty | and eat intexticated drivers. |
| 20:19.11 | bmoraca | intexticated? rofl |
| 20:19.18 | Katty | roflmao |
| 20:19.24 | Katty | Qwell: ! |
| 20:19.27 | Katty | Qwell: i'm lvling a druid. |
| 20:19.36 | Katty | infobot: roflmao? |
| 20:19.37 | infobot | somebody said roflmao was rolling on the floor laughing my arse off, or painful, or http://www.youtube.com/watch?v=iEWgs6YQR9A |
| 20:19.54 | Katty | excellent. |
| 20:20.03 | Katty | whoever taught infobot that. |
| 20:20.05 | Mango | Thank you everybody, it is working well now! |
| 20:20.05 | Katty | i love you forever. |
| 20:20.35 | KavanS | bmoraca, I thought sonicwalls were just iptables/netfilter boxes? |
| 20:20.53 | bmoraca | the best WoW animated song is "The Internet Is For Porn" from Avenue Q |
| 20:21.04 | Katty | sooooo uberuber leeeet so uberuber leeeeet |
| 20:21.20 | *** join/#asterisk sjobeck (n=Adium@c-71-193-223-71.hsd1.or.comcast.net) |
| 20:21.24 | Katty | bmoraca: yes, i am fond of that one too. |
| 20:21.28 | Katty | bmoraca: and crank dat druid |
| 20:21.30 | Dovid | haha |
| 20:21.31 | *** part/#asterisk sjobeck (n=Adium@c-71-193-223-71.hsd1.or.comcast.net) |
| 20:21.34 | Dovid | that is a great clip |
| 20:21.48 | Katty | yesh. |
| 20:21.50 | bmoraca | Katty: have you ever seen Avenue Q? highly suggest it if you haven't. |
| 20:21.52 | Katty | and i just got moonkin form. |
| 20:22.01 | Katty | bmoraca: k, i will see if i can find some stuff. |
| 20:22.01 | RaNaji | im doing it for my house i want my house analog line to go in to my asterisk box i just want to play with it thats all im new to it |
| 20:22.37 | bmoraca | it's like Seseme Street on meth for adults. it's a great show |
| 20:22.48 | Katty | yeah i just found some. gonna give it a watch |
| 20:23.27 | KavanS | christ... |
| 20:23.33 | KavanS | I'll give this channel a day to rebound |
| 20:23.34 | jaytee | manamana |
| 20:23.37 | *** part/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 20:23.50 | bmoraca | wow |
| 20:23.58 | bmoraca | he wasn't a very happy person |
| 20:24.08 | jaytee | what an uptight sphincter |
| 20:24.42 | [TK]D-Fender | He's just trying to keep insertions to a minimum ;) |
| 20:25.20 | telnettech | the funny thing is he will come back and find that the room is normaly like this when we are all slow with work |
| 20:25.45 | *** join/#asterisk rawko (n=rawko@adsl-070-155-127-242.sip.asm.bellsouth.net) |
| 20:25.55 | rawko | i got $50 via paypal for someone who can help me setup my trunks and dialing rules with a tdm800 with elastix |
| 20:26.21 | jaytee | people should not expect the same level of decorum from a chat room populated by "volunteers" as they would from a pay per call tech support line. |
| 20:27.11 | telnettech | so anybody out there hiring? |
| 20:27.12 | jaytee | rawko, why not ask in #elastix? |
| 20:27.45 | [TK]D-Fender | checkout time, later all |
| 20:28.11 | jaytee | telnettech, not much in Indy although it's not as bad here as in Ohio......yet. |
| 20:28.24 | telnettech | it is really bad here |
| 20:28.34 | Katty | i could use a house keeper. |
| 20:28.44 | telnettech | I have watched the same 7 jobs advertised daily on dice for ohio as a whole |
| 20:28.47 | rawko | ill hire ya for a 30 min job |
| 20:28.49 | Katty | i don't think that's your line of work tho. |
| 20:28.49 | telnettech | i have applied to all of them |
| 20:28.57 | rawko | just talk to me on the phone about a tdm800 and trunks and dialplans |
| 20:29.18 | jaytee | telnettech, what city are you closest to again? Dayton? |
| 20:29.23 | telnettech | Columbus |
| 20:29.33 | *** join/#asterisk propellerhead (n=yogurt2u@host183.190-136-104.telecom.net.ar) |
| 20:29.39 | rawko | no asterisk guru wants $50? |
| 20:29.53 | telnettech | Katty: If it pays 60k to 70k a year, I will clean your house |
| 20:30.16 | Qwell | rawko: elsewhere. |
| 20:30.49 | bmoraca | central CA is really bad for jobs right now. i've got a backlog of 2 hosted PBX, 1 CPE PBX, and 2 SIP-to-PRI trunking installs, though...so that'll last me a while. plus I need to add 4 more offices to one current hosted PBX customer... |
| 20:31.41 | rawko | i already got elastix working and it sees my channels |
| 20:32.01 | Qwell | rawko: FreePBX is not supported here. |
| 20:32.25 | rawko | i could be wrong but isnt asterisk freepbx |
| 20:32.32 | bmoraca | no... |
| 20:32.38 | bmoraca | freepbx is built on top of asterisk |
| 20:32.47 | bmoraca | or, rather, it's a 3rd party utility for asterisk |
| 20:32.50 | *** join/#asterisk voxter (n=voxter@201.200.86.134) |
| 20:32.57 | rawko | ah |
| 20:33.02 | rawko | its the front end? |
| 20:33.06 | rawko | a front end |
| 20:33.08 | Qwell | No. It's *A* front end. |
| 20:33.21 | Qwell | Completely separate project. |
| 20:33.55 | *** part/#asterisk rawko (n=rawko@adsl-070-155-127-242.sip.asm.bellsouth.net) |
| 20:34.01 | bmoraca | he left |
| 20:35.14 | Katty | telnettech: http://www.realtor.com/realestateandhomes-detail/Cape-Girardeau_MO_63701_1110617440 |
| 20:35.19 | jaytee | he went and asked in #elastix and then left there |
| 20:36.18 | Katty | telnettech: not my house :P |
| 20:36.37 | Katty | that's the house i went to today. |
| 20:37.21 | eric_hill | Katty: HFS! That's *huge* |
| 20:38.05 | Katty | eric_hill: did i mention that's not my house? |
| 20:38.09 | eric_hill | Katty: Lol, single-family property, 7 bedroom, 8 bathroom.... Riiiight. |
| 20:38.35 | Katty | that's where i installed the other phone and router today (= |
| 20:38.54 | eric_hill | You should have just made the butler do it. |
| 20:40.35 | Katty | no doubt. |
| 20:40.39 | bmoraca | we installed a $10,000 point to point wireless bridge for one of our customers from their business to their home (7 miles as the crow flies) because he wanted faster internet. rich people annoy me. |
| 20:43.42 | Mango | Oh my. |
| 20:44.35 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 20:46.26 | *** join/#asterisk [TK]D-Fender (n=zsirc@161.216.151.187) |
| 20:46.40 | [TK]D-Fender | \o// |
| 20:47.20 | jaytee | wow, the guy has two left arms |
| 20:47.27 | Mango | waves to [TK]D-Fender |
| 20:48.12 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 20:48.41 | clemahieu | I like rich people, they pay me to do simple things. |
| 20:51.32 | bmoraca | i hate people who live in alaska. frickin morons. never sell a PBX to someone from alaska. they're too dumb to figure out anything but a key system. |
| 20:51.52 | Qwell | hardwire: can you confirm? |
| 20:51.57 | *** join/#asterisk mort_gib (n=mjensen@77.208.67.181) |
| 20:51.57 | bmoraca | lol |
| 20:52.27 | jaytee | bbiab |
| 20:52.58 | bmoraca | i shit you not, i just had a phone call from one customer...here's how it went: "yeah, we're not able to transfer calls...when i hit blind transfer, all I get is a dialtone" *facepalm* |
| 20:56.53 | [TK]D-Fender | bmoraca: I shit peoople like that all thhe time. Usually shortlly after biting their heads off :p |
| 20:57.03 | *** part/#asterisk wasserja (n=wasserja@corporate.volstate.net) |
| 20:57.13 | eric_hill | bmoraca: Tell them it's an IP phone so they need to take their whole phone to the other party. |
| 20:57.51 | bmoraca | they'd been using the phone system for like 6 months, too...it made me die a little inside |
| 21:03.00 | hardwire | bmoraca: agreed |
| 21:03.11 | hardwire | bmoraca: also.. hows life coming along? you sound tense! |
| 21:03.12 | hardwire | :P |
| 21:03.17 | bmoraca | lol |
| 21:03.36 | hardwire | Qwell: bmoraca and I go way back.. he knows not to fuck with Alaska unless he really means it. |
| 21:03.44 | hardwire | fuck/screw |
| 21:03.49 | bmoraca | brought up a 28 phone hosted PBX this week...never again will i do that many... |
| 21:04.04 | hardwire | bmoraca: doh. Anchorage? |
| 21:04.07 | bmoraca | no |
| 21:04.14 | bmoraca | down in California where i'm at |
| 21:04.17 | *** join/#asterisk Downchuck (n=chatzill@c-66-235-10-200.sea.wa.customer.broadstripe.net) |
| 21:04.28 | bmoraca | the customer I was talking about is in Fairbanks |
| 21:04.36 | hardwire | bmoraca: sweet. |
| 21:04.39 | bmoraca | they've got a 6-phone CPE install |
| 21:04.41 | hardwire | we're putting services in at SOME POINT |
| 21:04.43 | hardwire | sigh. |
| 21:04.50 | bmoraca | still no service up there? |
| 21:04.52 | hardwire | well |
| 21:04.55 | Downchuck | I'm trying to use the AddQueueMember command in a way similar to AgentLogin, but I can only get AgentCallbackLogin |
| 21:04.56 | bmoraca | i thought you had that all wrapped up |
| 21:04.57 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 21:04.59 | [TK]D-Fender | hardwire: Who's naillin' Palin? |
| 21:05.09 | hardwire | [TK]D-Fender: your dvd reader. |
| 21:05.17 | Downchuck | How do I put the dynamic member into music on hold, and log them out when they hang up ? |
| 21:05.24 | hardwire | bmoraca: we have capacity.. but no PRI until we sign on more pre-clients |
| 21:05.28 | hardwire | and we don't have any pre-clients yet. |
| 21:05.39 | bmoraca | ick |
| 21:05.42 | Downchuck | AgentLogin works nicely, but I want to use dynamic queue |
| 21:06.03 | hardwire | bmoraca: let me know if you need cheaper term/orig to fbx.. and then maybe we can work soemthing out |
| 21:06.17 | hardwire | also.. how many laws are you breaking atm? that's the best part about working in Alaska. |
| 21:06.19 | hardwire | :P |
| 21:06.45 | bmoraca | well, they have copper from ACsomething or other, can't remember the provider |
| 21:06.49 | hardwire | bmoraca: ACS |
| 21:06.52 | hardwire | DSL |
| 21:06.53 | bmoraca | that's the one |
| 21:07.01 | hardwire | I;m having MASSIVE rtp issues with ACS |
| 21:07.03 | bmoraca | they have a T1 for data and then copper for voice |
| 21:07.04 | hardwire | MASSIVE |
| 21:07.10 | hardwire | ah.. T1 is different. :) |
| 21:07.12 | Downchuck | I was kind of wondering if I should do AddQueueMember, then transfer that member to another queue |
| 21:07.22 | hardwire | their BXB product can eat my shorts atm. |
| 21:07.26 | bmoraca | do you think it's packet shaping or just terrible quality of service? |
| 21:07.32 | hardwire | I'm having so many RTP issues I don't know what to do. |
| 21:07.35 | WindowsUser | does the SFA open beta mean they'll probably release SFA to be purchased soon? or are we getting a little taste to keep us interested? |
| 21:07.50 | hardwire | bmoraca: it's 20:1 on BXB products.. 4:1 on MPLS to Seattle/Portland |
| 21:07.56 | hardwire | it's just all oversubscribed |
| 21:08.06 | bmoraca | yikes |
| 21:08.10 | hardwire | I can't go BXB to BXB within the same town without going in and out of a bunch of useless routers. |
| 21:08.31 | bmoraca | what about AT&T? they any better? |
| 21:08.36 | ManxAway | sound like a job for Point to Point T-1 Man! |
| 21:08.41 | hardwire | so.. be warned.. if it's a BXB product (business class SHDSL) or the less expensive ADSL.. you're somewhat hosed. |
| 21:08.53 | ManxAway | He's faster than a speeding DSL, more powerful than a channelbank! |
| 21:08.54 | hardwire | ManxAway: I'm starting to agree more and more |
| 21:09.06 | hardwire | PTP from FBX to ANC is $670/mo |
| 21:09.28 | ManxAway | hardwire: I love T-1s. However, the MPLS service I helped a customer order claimed to have QoS options. |
| 21:09.29 | hardwire | I can get 1:1 5M GOLD!! MPLS to Portland for around $5000/mo |
| 21:09.52 | hardwire | ManxAway: I hate that term |
| 21:09.54 | hardwire | "Have QoS" |
| 21:10.03 | hardwire | it means "Some of your packets will be less suck ass than others." |
| 21:10.07 | *** join/#asterisk tdg911 (n=tdg911@75-131-246-217.static.slid.la.charter.com) |
| 21:10.16 | ManxAway | hardwire: That was not their exact words 8-) |
| 21:10.18 | bmoraca | ManxAway: MPLS has QoS tagging built in to the packet header...whether or not providers listen to it, though, is another story |
| 21:10.22 | hardwire | ACS has yet to tell me what DiffServ/Tag to use. |
| 21:10.40 | hardwire | bmoraca: listen/can queue excess traffic |
| 21:10.46 | bmoraca | that too |
| 21:10.49 | ManxAway | hardwire: try EF tag and see what happens |
| 21:11.01 | hardwire | ManxAway: are you being silly or for realsies? |
| 21:11.20 | hardwire | There could be some crazy RTP packet inside joke with 0xEF |
| 21:11.26 | hardwire | and I'd be the goose! |
| 21:11.30 | ManxAway | hardwire: I'm being serious. I don't expect them to HONOUR the EF tag, but it would be silly not to try. |
| 21:11.40 | hardwire | wonders how EF breaks down |
| 21:11.46 | ManxAway | no, Express Forwarding is EF. |
| 21:11.52 | ManxAway | I think the hes is 0xB8 |
| 21:11.53 | hardwire | seriously? a hex acronym? |
| 21:11.56 | hardwire | oh |
| 21:11.56 | hardwire | haha |
| 21:12.05 | ManxAway | maybe 0x8B, it's been a year or so since I looked at it. |
| 21:12.52 | hardwire | bmoraca: they claim not to throttle BXB but I've seen one too many occasions where WTF happened. |
| 21:12.55 | hardwire | sigh. |
| 21:12.58 | ManxAway | hardwire: do you have any SIPuras around? Their default Diffserv tag for RTP is whatever the hex is for "Express forwarding" |
| 21:13.08 | hardwire | and it could all be a problem with my routers.. who knows. |
| 21:13.22 | hardwire | ManxAway: indeed.. I'll check it out. |
| 21:13.32 | hardwire | All of our phones are Linksys SPA942 |
| 21:13.43 | ManxAway | hardwire: it can't hurt to try. If that fixes it you owe me big time. |
| 21:14.04 | hardwire | ManxAway: like.. mail order bride owe you? |
| 21:14.07 | *** join/#asterisk TestUser12321 (n=puppy@user-514c4af3.l3.c5.dsl.pol.co.uk) |
| 21:14.23 | ManxAway | hardwire: I went thu learning and implementing QoS w/Cisco routers and Point to Point T-1s like 5 years ago. |
| 21:14.41 | ManxAway | hardwire: as long as he is big, beefy, and likes leather. |
| 21:14.41 | hardwire | ManxAway: sounds promising. |
| 21:14.51 | hardwire | 0xb8 |
| 21:15.07 | hardwire | ManxAway: looks like it's been enabled for a while |
| 21:15.15 | hardwire | I'll make sure asterisk is marking it correctly on our side. |
| 21:16.12 | ManxAway | hardwire: any way to make sure the packets exiting your network still have the tag (i.e. not reset by some router). You might want to do a tcpdump on the far end if you can and see if the tag is still there or if it was reset in the cloud/backbone (bellsouth resets the Diffserv info for packets entering their DSL service, IIRC. |
| 21:16.29 | *** join/#asterisk yziquel (i=53acc979@gateway/web/freenode/session) |
| 21:16.36 | TestUser12321 | Hi. May I ask a question? |
| 21:16.41 | hardwire | no |
| 21:16.48 | hardwire | you're banned from asking that for another 4 minutes. |
| 21:16.53 | ManxAway | ultimately it is all up to the carrier if they honour diffserv tags or not. |
| 21:16.56 | TestUser12321 | :( ok ... sorry ;) |
| 21:17.01 | ManxAway | ~ask |
| 21:17.02 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 21:17.03 | hardwire | ManxAway: I'll check |
| 21:17.16 | ManxAway | hardwire: it's a long shot |
| 21:17.18 | TestUser12321 | Ok. Thanks. |
| 21:17.22 | WindowsUser | TestUser12321: he means you're not allowed to ask if you're allowed to ask a question for 4mins ;) |
| 21:17.35 | TestUser12321 | WindowsUser, I understand... Thank you. :) |
| 21:18.43 | yziquel | hello. http://paste.lisp.org/display/84462 i'm trying to install an asterisk box. it seems i can get throup my VoIP service provider, and that it queries some other server. however, it seems to me that this other server refuses the call. could someone explain to me how to read these sip debugging messages? |
| 21:18.54 | TestUser12321 | Anyone able to advise or suggest a Live Asterisk CD ? |
| 21:19.37 | [TK]D-Fender | TestUser12321: nothing "liv", all install |
| 21:19.47 | [TK]D-Fender | brb |
| 21:19.53 | ManxAway | yziquel: srvlookup=no in sip.conf |
| 21:20.10 | ManxAway | see the sip.conf.sample is the Asterisk source dir |
| 21:20.12 | TestUser12321 | [TK]D-Fender - Ok Thanks. |
| 21:20.15 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 21:20.24 | yziquel | ManxAway: thanks. looking into it. |
| 21:20.55 | TestUser12321 | Another question is does Asterisk have native support for external serial Voice Modems? |
| 21:21.09 | TestUser12321 | (admitting / recognising - their shakey quality!) |
| 21:21.13 | Qwell | TestUser12321: no |
| 21:21.30 | [TK]D-Fender | TestUser12321: ditto |
| 21:21.49 | *** join/#asterisk andres833 (n=andres83@190.144.102.122) |
| 21:22.30 | WindowsUser | TestUser12321: in my experience those modems barely work as answering machines |
| 21:22.53 | WindowsUser | half a second to switch from playing a beep sound to actually recording is silly |
| 21:23.59 | eppigy | Katty: hitya |
| 21:24.10 | [TK]D-Fender | eppigy: ... |
| 21:24.13 | [TK]D-Fender | TRABAJO |
| 21:24.23 | eppigy | DONDE |
| 21:24.29 | [TK]D-Fender | ESTA |
| 21:24.31 | eppigy | ME ESTOY MUCHO |
| 21:24.42 | eppigy | POR QUE |
| 21:25.07 | TestUser12321 | Ok. Thanks :) :( wwwaaaaaaaaahhhhhhhh!!!! ..... Woe is me and my poor external serial voice modems... :(.... |
| 21:25.17 | ariel_ | TestUser12321: there is a live cd from xorcom.com check there web site |
| 21:25.25 | Downchuck | How do I use AddQueueMember() to achieve the same functionality as AgentLogin? |
| 21:25.49 | [TK]D-Fender | Downchuck: Make dialplan that prompts for some sort of ID and auths it, then adds the member |
| 21:25.51 | TestUser12321 | I agree... accept and recognise... that voice modems are often ranging from poor to diabolical... lol :) ... but still potentially useful for preliminary testing... |
| 21:25.58 | TestUser12321 | ariel_ - Ok Thanks :) |
| 21:26.05 | Downchuck | [TK]D-Fender: That part I have down. I don't know how to set the agent on hold, though. |
| 21:26.13 | TestUser12321 | Also... some voice modems almost work!!! :) |
| 21:26.21 | [TK]D-Fender | Downchuck: "pausequeumember" |
| 21:26.39 | ariel_ | TestUser12321: no work |
| 21:26.52 | TestUser12321 | oh ok ... :(.... |
| 21:27.03 | TestUser12321 | I love (poorly) working external serial modems... :) |
| 21:27.10 | Downchuck | [TK]D-Fender: That temporarily stops them from receiving calls (which is good when they're on a call) |
| 21:27.46 | Downchuck | AddQueueMember() WaitForTransferWithHoldMusic() OnHangup(RemoveQueueMember) |
| 21:28.01 | *** join/#asterisk T` (n=total@pdpc/supporter/student/T) |
| 21:28.09 | Downchuck | Typically, addqueuemember is used for agent call backs. that is, the agent is added to the queue.. then hung up on, and they wait for a call back when a new call comes in |
| 21:28.41 | T` | hi.. i am a voip noob and want to setup a way to receive calls from gizmo5 SIP |
| 21:29.14 | T` | does anyone have experience with using a Grantstream HT486 for this? or any other suggestion is also welcome |
| 21:29.23 | ManxAway | TestUser12321: The only people with the SKILLS to turn a modem into a telco card are the people that have no interest in doing so because they know how hard and pointless it is. |
| 21:29.38 | [TK]D-Fender | ManxAway: ...... |
| 21:29.41 | T` | i thought of buying MagicJack, but i dont think they have software for linux and also dont want to have a computer ON always to receive/make calls |
| 21:29.42 | [TK]D-Fender | ~savemoney |
| 21:29.43 | infobot | <Gremlin> It's okay, but I was hoping to save some money with a soldering iron and a bunch of sound cards. |
| 21:30.01 | [TK]D-Fender | :D |
| 21:30.01 | TestUser12321 | ManxAway, That I can appreciate and understand. Thank you. |
| 21:30.10 | ManxPower | A SIPura ATA is less than $60 |
| 21:30.19 | TestUser12321 | Luckily... occasionally and fortunately... a modem company relase a good working voice modem ... e.g. SiteCom... |
| 21:30.33 | TestUser12321 | However I think those days and the volume sales of analogue modems are largely over... |
| 21:30.40 | TestUser12321 | :( ... <<< weeping tears.... |
| 21:30.42 | Downchuck | [TK]D-Fender: am I completely off on this? |
| 21:31.13 | T` | ManxPower, i think most of these devices are discontinued unfortunately |
| 21:31.17 | [TK]D-Fender | Downchuck: You shouldn't need to pause an active member when they take a call, only when you don't want them taking any. |
| 21:31.18 | TestUser12321 | infobot, I am shocked... how dare you - lol :) |
| 21:31.19 | infobot | it is my pleasure to meet you, shocked... how dare you - lol :) |
| 21:31.25 | T` | ManxPower, do you use any of them? |
| 21:31.43 | [TK]D-Fender | TestUser12321: pwned |
| 21:31.48 | TestUser12321 | YAYAYAYAYAY!!!! |
| 21:31.55 | Downchuck | [TK]D-Fender: AddQueueMember() Hangup() works like AgentCallbackLogin, I believe. I want it to work like AgentLogin. |
| 21:31.59 | TestUser12321 | :) :D * _ * ^ _ ^ yayayayay! :) |
| 21:32.17 | Downchuck | After AddQueueMember, what do I run to keep the line active, with the agent listening to hold music until a call comes in? |
| 21:32.17 | [TK]D-Fender | Downchuck: then USE agentLogin |
| 21:32.23 | [TK]D-Fender | Downchuck: It still exists |
| 21:32.34 | [TK]D-Fender | Downchuck: and AQM + Hangup != AgentLogin |
| 21:32.35 | Downchuck | AgentLogin is used for static agents I believe |
| 21:33.03 | Downchuck | AQM + Hangup == AgentCallbackLogin (sort of). |
| 21:33.10 | [TK]D-Fender | Downchuck: Agents yes, static... says who? |
| 21:33.23 | [TK]D-Fender | Downchuck: Callback yes, regular no. |
| 21:33.34 | [TK]D-Fender | Downchuck: Be clear |
| 21:33.59 | Downchuck | [TK]D-Fender: Trying. |
| 21:34.24 | Downchuck | I'm trying to use AQM + Something instead of AgentLogin, because I thought it'd give me better control. |
| 21:35.27 | Downchuck | You're saying, I should use AgentLogin and dynamic agents. Instead of AQM |
| 21:36.28 | yziquel | ManxPower: I still get the exact same message as http://paste.lisp.org/display/84462 |
| 21:36.32 | Downchuck | ... I was trying to replace our current AgentLogin() with an AQM-based macro |
| 21:37.32 | Downchuck | AddQueueMember, UserEvent(Agentlogin), Queue(logged-in-agents) RemoveQueueMember() Hangup(); |
| 21:37.33 | [TK]D-Fender | Downchuck: I'm saying that AQM is like AgentCALLBACKlogin. AgentLogin keeps you on the call. AQM will not work that way |
| 21:37.52 | [TK]D-Fender | Downchuck: there is no bridge mechanism to just sit around and DTMF to accept a call, etc |
| 21:37.59 | Downchuck | ok |
| 21:38.38 | Downchuck | What about using two queues? Where the agent is in a queue, essentially waiting for an inbound transfer? |
| 21:38.52 | [TK]D-Fender | Downchuck: huh? |
| 21:38.58 | [TK]D-Fender | Downchuck: what "transfer"? |
| 21:39.08 | [TK]D-Fender | Downchuck: you get CALLED. |
| 21:39.21 | Downchuck | sorry for the bad usage |
| 21:40.31 | Downchuck | I will look into adding agents dynamically. |
| 21:41.37 | Mango | Hmm. |
| 21:42.01 | Mango | What could cause audio to skip during Playback() or music on hold? It doesn't skip much, probably about once per minute. |
| 21:42.24 | Mango | Server load is 0.04, and this single test call plus IRC are the only things on the network. |
| 21:42.38 | Mango | I tried adjusting the jitter biffer on the phone but that didn't help |
| 21:42.40 | Mango | biffer/buffer |
| 21:43.10 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 21:44.58 | T` | is there a way way to get a ATA which can work with my existing phone line + have SIP capability? |
| 21:45.00 | *** join/#asterisk voxter (n=voxter@201.200.71.34) |
| 21:45.01 | T` | is that even possible? |
| 21:45.21 | *** join/#asterisk snowboarder04 (n=un@94-193-80-153.zone7.bethere.co.uk) |
| 21:45.26 | Mango | T`: I believe the SPA3102 can do that. However, everyone I have spoken to says the PSTN quality is terrible. |
| 21:46.07 | T` | Mango, does PSTN mean when i call non-SIP numbers? |
| 21:46.26 | T` | i want to use SIP to dial international numbers.. for local i want to go thru my usual phone company |
| 21:46.56 | Mango | Yeah. Unforutnately, if you use that device to go through your usual phone company, it will work, but not very well. |
| 21:47.00 | Mango | I hear reports of echo and whatnot. |
| 21:47.22 | Mango | Does anyone else here have good luck with the 3102 or similar? |
| 21:47.27 | yziquel | Could someone be kind enough to explain to me the meaning of my 401 error? |
| 21:47.45 | T` | Mango, ok i see.. i will then just go with a two phone setup |
| 21:48.08 | Mango | T`: Or perhaps a two-LINE phone? |
| 21:48.17 | T` | Mango, i have a Uniden DECT 6.0 phone |
| 21:48.42 | Downchuck | [TK]D-Fender: One more crazy idea. Can I have two queues, and bridge Calls from one queue to Calls sitting idle on the second queue? |
| 21:49.28 | T` | Mango, is this the best i can do with respect o getting free incoming/outgoing calls to US:Google Voice + Gizmo5 -> ATA |
| 21:49.32 | [TK]D-Fender | Downchuck: perhaps with a shit-ton of AMI trickery, etc. |
| 21:49.48 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 21:50.07 | Downchuck | [TK]D-Fender: alright, i was hoping for less than a shit-ton. |
| 21:50.14 | *** join/#asterisk voxter (n=voxter@201.200.71.34) |
| 21:50.14 | T` | Mango, the gizmo5 support for GV was announced recnetly i think.. the outgoing calls are free, but are limited to 3minutes.. but as long as i can receive calls it should be fine imo.. ia m setting this up at my parents.. so i can call them back |
| 21:50.26 | Downchuck | Many thanks, you answered all of my questions. My asterisk guy just said "You're making it too complicated", and refused to chat further. |
| 21:50.40 | Mango | T`: If nobody else here has an answer, you may want to look at http://www.dslreports.com/forum/voip |
| 21:50.42 | [TK]D-Fender | Downchuck: You COULD fake this with ValetParking since you could target slots, and then use the pickup in dialplan you add dynamically. On hangup it'd loop the member back to the parking slot... |
| 21:50.50 | Mango | There are a few big threads about Gizmo5 and GV there. |
| 21:51.01 | T` | ah nice.. will do! |
| 21:51.21 | Downchuck | [TK]D-Fender: I like to say "implement", instead of "fake"/"emulate" |
| 21:51.32 | T` | this stuff is pretty cool.. it seems like i can just use gizmo5 as a way to recieve calls for free |
| 21:51.55 | T` | Mango, do you you use GV? |
| 21:52.04 | Downchuck | I'll look into it, parking, instead of using some AMI nastiness |
| 21:52.39 | Mango | T`: Nah, I'm too privacy paranoid |
| 21:52.51 | Mango | and $0.0125/min is cheap anway :) |
| 21:53.15 | T` | Mango, actually i just came across that in this thread |
| 21:53.25 | T` | now.. it does seem like a stupid idea for me to do this.. |
| 21:53.30 | T` | hmmm |
| 21:54.41 | Mango | Ok. I've learned how to install Asterisk, set up SIP devices, paging, distinctive ring, Caller ID, failover, recording, and music on hold. |
| 21:54.50 | Mango | Maybe I'll go for queues next |
| 21:55.18 | Mango | imagines his wife calling and thinking to herself, "WTF? My call has been placed in priority sequence?" |
| 21:56.28 | [TK]D-Fender | Downchuck: Oh we've passed "fakery" and gone right along to "fugly hacks / inbred non-solutions" |
| 21:56.34 | Downchuck | hehe |
| 21:56.40 | Downchuck | https://issues.asterisk.org/view.php?id=15135 - dynamic parking is almost in |
| 21:56.41 | [TK]D-Fender | And on that note I'm off to martial arts & stage time... |
| 21:56.45 | Downchuck | thanks TK |
| 21:57.07 | [TK]D-Fender | goes to pwn some more newbs... |
| 21:57.41 | *** join/#asterisk Micc (n=dotirc@c-76-121-255-52.hsd1.wa.comcast.net) |
| 21:58.12 | Micc | so on the sonicwall do I want to enable consistent NAT? |
| 21:58.38 | Mango | Dynamic parking? |
| 21:59.15 | bmoraca | Micc: no, you want to get rid of the sonicwall |
| 21:59.30 | Micc | TKD-Fender, do I want consistent NAT? |
| 21:59.32 | buttons840 | i've enabled callprogress=yes and it's working perfectly for the few tests i've put it through, can anyone suggest further reading so i can learn how it works, what it's potential fallbacks are, and perhaps how i could improve upon it? |
| 21:59.42 | Micc | bmoraca, thats not my choice. my customer wants the sonicwall. |
| 22:00.06 | bmoraca | then he isn't going to be able to do voip. that's the breaks. you get one or the other, but not both |
| 22:00.33 | Micc | TKD-Fender said he made it work through a sonicwall. |
| 22:01.13 | ManxPower | buttons840: the documentation is the code. |
| 22:01.24 | *** join/#asterisk pdmmm (i=pdm@pdm.sh) |
| 22:01.45 | pdmmm | anyone have any luck with using # (pound key) in an asterisk extensions.conf? |
| 22:01.55 | pdmmm | i'm using an avaya phone and its sending the # key, but asterisk wont match it |
| 22:02.03 | buttons840 | ManxPower, ohh boy, shall i venture into the code for the first time? i suppose i should start in dsp.c? |
| 22:02.08 | ManxPower | pdmmm: works just fine unless you override it in features.conf |
| 22:02.17 | pdmmm | [Jul 30 17:01:54] NOTICE[6709]: chan_sip.c:18160 handle_request_invite: Call from '1001' to extension '1#' rejected because extension not found. |
| 22:02.25 | pdmmm | but i have 1# |
| 22:02.32 | ManxPower | buttons840: I'd look in chan_zap/chan_DAHDI. |
| 22:02.41 | pdmmm | edge01*CLI> dialplan show extensions |
| 22:02.41 | pdmmm | [ Context 'extensions' created by 'pbx_config' ] |
| 22:02.41 | pdmmm | <PROTECTED> |
| 22:02.41 | pdmmm | <PROTECTED> |
| 22:02.41 | pdmmm | <PROTECTED> |
| 22:02.42 | pdmmm | <PROTECTED> |
| 22:02.45 | pdmmm | its there ;) |
| 22:02.52 | pdmmm | do i need to escape it? |
| 22:02.55 | ManxPower | pdmmm: paste just the exten line for 1#, the actual extensions.conf |
| 22:03.02 | bmoraca | Micc: you can certainly try, but I have never once been able to make it work. Sonicwalls monkey with the packets. perhaps you could try relying on their ALG and setting NAT=no on your sip peers and in your phone configs...but i seriously doubt that it will work, and you will spend more time and money trying to make it work than if you had just replaced it with a less shitty device in the... |
| 22:03.04 | bmoraca | ...first place |
| 22:03.08 | ManxPower | and if you flood the channel nobody will help you. |
| 22:03.10 | ManxPower | ~pb |
| 22:03.10 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/, or apt-get install pastebinit |
| 22:03.14 | pdmmm | exten => 1#,1,Answer() |
| 22:03.14 | pdmmm | exten => 1#,n,SayDigits(${CALLERID(num)}) |
| 22:03.49 | ManxPower | ya know it sure would be nice if chan_sip provided the damn context in it's error messages |
| 22:03.58 | pdmmm | its in the context |
| 22:04.05 | buttons840 | pastebinit ? if that's what i think it is i'll weep because i didn't learn of it sooner |
| 22:04.05 | pdmmm | cuz i got the same extensions line w/ 11 |
| 22:04.06 | pdmmm | and it works |
| 22:04.07 | ManxPower | pdmmm: which context? |
| 22:04.11 | pdmmm | [extensions] |
| 22:04.37 | bmoraca | # doesn't work all that well in dialplan, from what i recall... |
| 22:04.37 | pdmmm | ya |
| 22:04.37 | pdmmm | i'm gathering that |
| 22:04.38 | pdmmm | ;) |
| 22:04.44 | ManxPower | bmoraca: only because phones like to eat the # |
| 22:04.52 | pdmmm | well the phone is sedning the # |
| 22:04.52 | ManxPower | and features.conf like to eat the # |
| 22:04.54 | pdmmm | in the sip invite |
| 22:05.02 | pdmmm | and i redefined anything remotely # in features.conf |
| 22:05.07 | bmoraca | i imagined it had something to do with the tone generation or something |
| 22:05.24 | ManxPower | pdmmm: maybe the phone is URL encoding it? I had to use the URL encoded string for #, IIRC. |
| 22:05.29 | ManxPower | let me check a reference. |
| 22:06.31 | ManxPower | pdmmm: exten => 1%23,1,Answer() |
| 22:06.46 | bmoraca | gross |
| 22:07.25 | pdmmm | ooo |
| 22:07.26 | ManxPower | bmoraca: disgusting, I agree. I vaguely recall late 1.4 or maybe 1.6 makes it work as expected. It's been like 2 years since I had to process # in the dialplan |
| 22:07.27 | pdmmm | lemme try |
| 22:07.51 | pdmmm | negative |
| 22:08.00 | ManxPower | pdmmm: you did an extensions reload |
| 22:08.28 | *** join/#asterisk buttons840 (n=buttons8@c-24-10-149-58.hsd1.ut.comcast.net) |
| 22:09.14 | pdmmm | ya |
| 22:09.24 | ManxPower | pastebin a sip debug of a failed call |
| 22:09.29 | *** join/#asterisk TSM2 (n=the_soft@87-194-32-212.bethere.co.uk) |
| 22:09.47 | ManxPower | if your extensions.conf is not too long feel free to pastebin that too |
| 22:10.11 | WindowsUser | i hope i dont get in trouble for making a bajillion posts on the SFA bug tracker |
| 22:10.14 | pdmmm | well |
| 22:10.20 | pdmmm | lemme run home (i'm at the office) -- and i'll be back |
| 22:10.25 | pdmmm | thanks a million for your help |
| 22:10.30 | pdmmm | i'll work it from home ;) |
| 22:12.46 | buttons840 | what are some of the main differences between asterisk 1.4 and 1.6? |
| 22:14.42 | timeshell_atwork | Hi. Need some help with how asterisk 1.6 reads dahdi channels from users.conf |
| 22:14.52 | ManxPower | buttons840: all documented in the doc/ directory in the Asterisk source. UPGRADE*.txt and Changes (or maybe ChangeLog, I don't recall the exact file name). Some of those files may be in the root of the Asterisk source, rather than the doc/ dir |
| 22:15.16 | ManxPower | timeshell_atwork: users.conf is mainly for GUIs, not many people use it. |
| 22:15.26 | timeshell_atwork | I have 3 channels defined under one trunk in the users.conf as trunk_1 |
| 22:15.27 | buttons840 | right, in the doc's as usual |
| 22:15.47 | *** join/#asterisk ingenius (n=alektro@host215.200-45-165.telecom.net.ar) |
| 22:15.48 | timeshell_atwork | I have a global trunk_1 = DAHDI/trunk_1 in the extensions.conf |
| 22:15.58 | ManxPower | timeshell_atwork: I wish you the best of luck. |
| 22:16.00 | timeshell_atwork | However, asterisk can't find a channel based on that definition. |
| 22:17.05 | timeshell_atwork | The message is "Unable to determine channel for data trunk_1/my_number |
| 22:17.56 | timeshell_atwork | By this message, I presume that asterisk should be able to resolve trunk_1 in the users.conf to extract the channels that are supposed to be available. |
| 22:18.21 | grandpapadot | Word! |
| 22:18.37 | timeshell_atwork | That, or this is an incorrect configuration and I need some guidance on the correct to configure it. |
| 22:19.04 | TSM2 | ive today tried my * install connected to our E1 line, ive get the incomming route to catch all and direct it to an internal extention, problem is that i just get a network busy tone, if i direct the incomming call to the VM then it goes though, any thoughts |
| 22:20.31 | ManxPower | TSM2: in the priority after the dial, put Noop(HANGUPCAUSE is ${HANGUPCAUSE} and DIALRESULT is ${DIALRESULT}) That will tell you the reason for the dial failing. |
| 22:21.09 | ManxPower | HANGUPCAUSE is the most specific and accurate. You can find the meanings of the values by googling for Q.931 result causes or something similar |
| 22:22.07 | *** part/#asterisk BhaalWK (i=bhaal@freenode/staff-emeritus/bhaal) |
| 22:25.04 | TSM2 | ManxPower: will this be in any of the DEBUG or VERBOSE message i have in the full log |
| 22:25.14 | TSM2 | i was running asterisk -r -vvv while i tested it |
| 22:25.53 | ManxPower | TSM2: verbose, no debug required |
| 22:27.07 | TSM2 | part of what i got that seemed odd was.... |
| 22:27.08 | TSM2 | [Jul 30 19:46:07] DEBUG[6376] chan_dahdi.c: Set option AUDIO MODE, value: ON(1) on DAHDI/1-1 |
| 22:27.08 | TSM2 | [Jul 30 19:46:07] DEBUG[6376] chan_dahdi.c: Not yet hungup... Calling hangup once with icause, and clearing call |
| 22:27.09 | TSM2 | [Jul 30 19:46:07] DEBUG[6376] chan_dahdi.c: Set option AUDIO MODE, value: OFF(0) on DAHDI/1-1 |
| 22:28.42 | ManxPower | TSM2: What is the value of HANGUPCAUSE |
| 22:29.29 | TSM2 | ManxPower: unforuntatly i cant test the circuit at the moment, im out of the testing window |
| 22:30.20 | ManxPower | come back when you can |
| 22:33.58 | yziquel | http://paste.lisp.org/display/84462 i've set the fromuser to some specific string (which isn't 'yziquel'). However it doesn't show up in the SIP headers, when sending outbound calls. why could this be? |
| 22:34.53 | *** join/#asterisk sjobeck (n=Adium@38.100.211.40) |
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| 22:39.16 | *** part/#asterisk sjobeck (n=Adium@38.100.211.40) |
| 22:40.42 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 22:41.02 | TSM2 | ManxPower: just checking, you want me to put that noop line just after to Goto clause? |
| 22:41.18 | ManxPower | TSM2: HANGUPCAUSE may be the single most important thing to know when you have DAHDI/Zaptel issues |
| 22:41.27 | ManxPower | no, right after the Dial that fails |
| 22:42.22 | ManxPower | it tells you WHY the dial failed |
| 22:44.01 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 22:44.29 | buttons840 | wanpipe install overwrote all my chan_dahdi.conf files, ManxPower suggested in look at it, but i can't find an original to look at |
| 22:44.36 | TSM2 | this is not a dial out call, its an incomming call to asterisk |
| 22:45.16 | ManxPower | TSM2: so where are you trying to send the call? |
| 22:45.18 | buttons840 | i presume this was the original contents http://www.voip-info.org/wiki/view/chan_dahdi.conf? |
| 22:45.25 | buttons840 | http://www.voip-info.org/wiki/view/chan_dahdi.conf |
| 22:45.43 | ManxPower | buttons840: looks like the sample config |
| 22:46.09 | buttons840 | ManxPower, you suggested i study that for a beginning to dsp? am i right? |
| 22:46.23 | ManxPower | buttons840: no chan_dahdi.c |
| 22:46.45 | TSM2 | ManxPower: its a freepbx config, so i guess its a little more complicated, but i set it to go to an extention, the last line of the script is Goto(from-did-direct,2200,1) |
| 22:46.55 | ManxPower | you are not going to learn anything about DSP without using the source code. You should also pick up a DSP programming reference. I hope you are good with math. |
| 22:47.15 | ManxPower | TSM2: I cannot help with a FreePBX install. Did you try the correct channe? |
| 22:47.27 | ManxPower | Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-bugs #asterisk-dev #asterisk-commits |
| 22:48.05 | TSM2 | TSM2: its a E1 line, i know it works partialy as if i set it to go to the VM of that extention its fine, just to the extention itself it does not work |
| 22:48.15 | TSM2 | ile try over at #freepbx mabey |
| 22:48.30 | ManxPower | ~freepbx |
| 22:48.31 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 22:48.47 | buttons840 | ManxPower, i'm not too good with math, but i understand the current dsp has some shortcomings? i suppose these are beyond what a amature could fix though? |
| 22:49.22 | ManxPower | buttons840: if it was easy to fix it would have been fixed years ago. |
| 22:49.23 | buttons840 | well, i've taken through calculus, is that "good with math"? |
| 22:49.34 | ManxPower | DSP is ALL math. |
| 22:52.48 | buttons840 | a mere 12,000 lines i see :\ |
| 22:53.25 | ManxPower | buttons840: if it was easy it would have been done years ago by people much smarter than you or me. |
| 22:53.27 | yziquel | buttons840: know what a Fourier transform is? |
| 22:53.43 | buttons840 | nope |
| 22:53.49 | yziquel | buttons840: i guess it's a prerequisite. |
| 22:53.52 | buttons840 | what math class will i learn that in? |
| 22:54.06 | yziquel | buttons840: depends on your country :) |
| 22:54.14 | buttons840 | united states |
| 22:54.20 | yziquel | dunno |
| 22:54.36 | buttons840 | what is the class in your country? |
| 22:54.50 | buttons840 | nm, i'll google it |
| 22:54.57 | yziquel | yziquel: in France, "classes préparatoires" or "licence" |
| 22:55.20 | ManxPower | buttons840: http://www.dsptutor.freeuk.com/intro.htm |
| 22:55.22 | yziquel | buttons840: in France, "classes préparatoires" or "licence" |
| 22:56.25 | ManxPower | "mains" means "power grid power" |
| 22:56.46 | TSM2 | is it possable to simulate a call comming in from dahdi |
| 22:57.13 | buttons840 | i looked up a fourier transform, the first clue that it was over my head was that there wern't any numbers in the equations |
| 22:57.41 | ManxPower | TSM2: not really. you could DACS the far end channels if you have a T-1/E-1/ you could plug an fxs port into an fxo port if using analog |
| 22:58.10 | ManxPower | you could use a crossover T-1/E-1 cable if your card has more than one port. |
| 22:58.42 | yziquel | buttons840: you get the idea. |
| 22:59.06 | TSM2 | naa dont have that,ahh well |
| 23:00.21 | hesco | I'm debugging an AGI script written in perl. STREAM FILE gives me in the *CLI: 'Playing /var/lib/asterisk/sounds/custom/path/to/my/file', but I never hear the audio. Then 'AGI Script my_script.agi completed, returning 0', then the next priority in the dialplan after the return says: 'Playing '/var/lib/asterisk/sounds/custom/path/to/same/audio.gsm' and I do hear it on the phone. |
| 23:01.02 | hesco | What am I missing that STREAM FILE says its behaving in the *CLI, but not delivering the audio to my extension? |
| 23:01.50 | ManxPower | hesco: your sound files need to be 8Khz Mono |
| 23:02.04 | ManxPower | can you play the same file in the dialplan? |
| 23:02.19 | hesco | my sound file was created using Record() though my dialplan |
| 23:02.35 | hesco | yes, that is what I just reported above. |
| 23:02.49 | hesco | it plays through the dialplan, but not from the agi script |
| 23:03.22 | ManxPower | paste just the single streamfile command in your script |
| 23:03.44 | hesco | print "STREAM FILE $file '9' \n"; |
| 23:04.06 | hesco | I understand the offset parameter is optional |
| 23:04.13 | ManxPower | that was useful. 8-) does $file contain the filename with the file extension or without the file extension? |
| 23:04.36 | hesco | I also tried it using Asterisk::AGI as: $result = $agi->stream_file($file,"",""); |
| 23:04.55 | hesco | $file = "$custom_audio/welcome_to_validation_line"; |
| 23:05.19 | ManxPower | good. You should never specify the file extension with playing files in Asterisk |
| 23:05.23 | hesco | whether with or without the ::AGI module, I get the same result, i.e no result |
| 23:05.26 | ManxPower | if you do so it won't work |
| 23:05.53 | hesco | and it throws warnings showing me the error of my ways, as well |
| 23:07.54 | hesco | so, ManxPower: any clues? |
| 23:08.10 | ManxPower | hesco: nope. you avoided the most common mistakes |
| 23:09.12 | hesco | no pointers to the least common ones? |
| 23:09.19 | *** join/#asterisk yziquel (i=53acc979@gateway/web/freenode/session) |
| 23:10.33 | buttons840 | so just how problematic is callprogress=yes |
| 23:10.33 | ManxPower | hesco: It's been at least 2 years since I've done any AGI scripting |
| 23:10.33 | buttons840 | ? |
| 23:10.43 | ManxPower | buttons840: depends on the voices of your customers |
| 23:10.49 | WindowsUser | stream file may have issues |
| 23:10.56 | hesco | how about some idea of coaxing more informative debug information. |
| 23:11.07 | ManxPower | loud people and women with higher pitched voices tend to cause hangups more often |
| 23:11.13 | hesco | would CONTROL STREAM FILE work any better? |
| 23:11.21 | WindowsUser | dunno |
| 23:11.24 | ManxPower | hesco: agi debug in the CLI? |
| 23:11.29 | WindowsUser | I can tell you what I've used tho |
| 23:11.36 | hesco | please |
| 23:11.39 | buttons840 | ManxPower, is it possible to disable callprogress once the call is answered? |
| 23:11.40 | ManxPower | that should at least get STDERR sent to the console |
| 23:11.40 | WindowsUser | EXEC playback $file |
| 23:11.46 | ManxPower | buttons840: nope. |
| 23:11.52 | WindowsUser | and I tried BACKGROUND $file |
| 23:12.42 | buttons840 | does asterisk have the ability to detect a answering machine tone/beep? |
| 23:13.14 | hesco | I'm seeing none of my STDERR in the *CLI, I've directed all my debug outpit to a DEBUG filehandle I opened myself |
| 23:13.20 | yziquel | buttons840: in the book, it seems so. |
| 23:13.30 | hesco | buttons840: see amd() |
| 23:14.22 | hesco | WindowsUser: did either of those work for you? |
| 23:14.47 | WindowsUser | EXEC Playback /usr/share/asterisk/vm-intros/eng-unavailable |
| 23:14.54 | WindowsUser | is in use |
| 23:15.30 | WindowsUser | works on 1.4.26 and 1.6.1.1 |
| 23:15.50 | hesco | thanks, trying that now |
| 23:16.20 | ManxPower | buttons840: I strongly doubt it. Nobody uses callprogress |
| 23:16.39 | ManxPower | Well nobody sane. I'm sure there are nutjobs out there that use it. |
| 23:17.04 | buttons840 | ManxPower, so far, it is my only solution for preventing immediate dial plan execution over a FXO channel. |
| 23:17.30 | ManxPower | I would not call randomly disconnecting calls a "solution". |
| 23:17.38 | ManxPower | Can't you just fire the customer? |
| 23:17.43 | WindowsUser | calling people up and telling them not to light anything up? |
| 23:17.53 | buttons840 | fire the customer? |
| 23:18.00 | ManxPower | *nod* |
| 23:18.13 | buttons840 | meaning? |
| 23:18.23 | buttons840 | are you refering to my client? |
| 23:18.28 | WindowsUser | sneaks off |
| 23:18.31 | ManxPower | Consultant to customer: "You're fired" aka "I quit" |
| 23:18.45 | ManxPower | You can't usually do that if you are an employee of course. |
| 23:19.01 | ManxPower | In any case, I'm done with the topic of callprogress. |
| 23:19.13 | ManxPower | I wish you the best of luck, buttons840 |
| 23:19.35 | buttons840 | wait, you were the one talking to me about it earlier? |
| 23:19.35 | hesco | well, I tried all three strategies and one of them worked. Next step is to figure out which one. |
| 23:19.44 | hesco | thanks WindowsUser |
| 23:19.45 | buttons840 | did you change your nick since then? |
| 23:20.19 | buttons840 | ohh, you are the same person |
| 23:20.25 | buttons840 | my appologies |
| 23:22.27 | buttons840 | guess i'll tell em to get some channels with signalling or accept the random hangups |
| 23:23.01 | buttons840 | i didn't even know about signalling earlier today, so thank for your patience and help, with that, i end my work for today |
| 23:23.45 | hesco | well, it seems it was: `print "EXEC Playback $file \n";` which did the trick. Thanks. Finally some progress. |
| 23:24.06 | hesco | does using EXEC create any security risks? |
| 23:24.34 | WindowsUser | watch your inputs |
| 23:26.13 | WindowsUser | setting $file to "beep\nEXEC Dial 1900xxxxxxx" is a problem ;) |
| 23:28.16 | WindowsUser | and away! |
| 23:32.38 | hesco | thanks for your help |
| 23:34.03 | *** join/#asterisk howie (n=howie@71-95-220-206.dhcp.mtpk.ca.charter.com) |
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| 23:45.58 | mltlnx | Why wont my Polycom blf flash when its subscribed device is ringing! |
| 23:47.14 | ManxPower | mltlnx: is Asterisk SENDING the ringing to the subscribed phone? |
| 23:48.20 | mltlnx | ManxPower: Yes, It does show that the line is in use. Meaning that the blf is solid (as opposed to flashing) |
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| 23:54.30 | ManxPower | mltlnx: you confirmed this in sip debug? |
| 23:55.29 | ManxPower | since astersk would have to send different messages to indicate ringing .vs. in use. I suspect asterisk does not support this. |
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| 23:59.23 | exsync | is there no generic voip channel? |