| 00:00.34 | ManxPower | I like my job, thanyouverymuch |
| 00:01.31 | CrashHD | not a bad thought |
| 00:01.38 | CrashHD | I like the jb for sip in 1.4 though |
| 00:01.40 | blitzrage | it's all about testing |
| 00:01.53 | blitzrage | 1.4 works for the stuff I use (after testing, and getting several bugs fixed) |
| 00:02.15 | *** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com) |
| 00:03.51 | ManxPower | CrashHD: we do not currently need a SIP jitter buffer. 8-) |
| 00:04.28 | CrashHD | :) |
| 00:04.47 | *** part/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
| 00:05.01 | ManxPower | all our IP links that have SIP traffic have strict QoS |
| 00:09.43 | CrashHD | we have lots of client connections over the internet |
| 00:09.50 | CrashHD | so the jb really comes in handy for us |
| 00:10.07 | CrashHD | hints just plain are not working in 1.4 |
| 00:10.58 | ManxPower | it seems to me that it would be easier to port one of the 1.2.x RTP jitterbuffer patches to 1.2.15 then it would be to test 1.4 |
| 00:11.13 | CrashHD | that's not a bad point |
| 00:11.59 | ManxPower | it took until 1.2.10 for me up upgrade from 1.0 |
| 00:12.06 | CrashHD | heh |
| 00:14.00 | *** join/#asterisk coppice (n=chatzill@13.168.17.210.dyn.pacific.net.hk) |
| 00:14.01 | russellb | yep, at this point, we still run 1.2 :) |
| 00:14.18 | CrashHD | actually |
| 00:14.19 | CrashHD | hah |
| 00:14.22 | CrashHD | this is the kicked |
| 00:14.29 | CrashHD | it's 1.2.14 that is having the issue |
| 00:14.38 | CrashHD | (kicker) |
| 00:14.40 | CrashHD | I just saw that |
| 00:14.47 | CrashHD | we have like 10 asterisk installs on one machine |
| 00:14.52 | CrashHD | various versions running |
| 00:14.53 | CrashHD | for testing |
| 00:15.00 | CrashHD | 1.2.14 is having the hint issue |
| 00:15.23 | CrashHD | 1.2.14-r48468 |
| 00:15.33 | CrashHD | 1.2.13-r45413 |
| 00:15.37 | CrashHD | is working fine |
| 00:17.20 | file | svn diff and see what the differences are? |
| 00:17.59 | CrashHD | can you svn diff a whole dir tree? |
| 00:18.09 | file | sure. |
| 00:18.27 | file | svn diff http://svn.digium.com/svn/asterisk/branches/1.2@45413 http://svn.digium.com/svn/asterisk/branches/1.2@48468 | more |
| 00:18.59 | CrashHD | ok I'll do my local copies |
| 00:19.28 | CrashHD | anything in particular I'm looking for? |
| 00:19.45 | file | if I knew I would have told you what to look for already |
| 00:19.54 | JT | anyone have some hints for some linux networking utilities that measure jitter? |
| 00:19.56 | CrashHD | very good point |
| 00:20.02 | JT | like an enhanced ping? :P |
| 00:21.13 | CrashHD | file: looks like some dev state stuff |
| 00:21.23 | CrashHD | chan_local stuff as well |
| 00:22.24 | CrashHD | devstate to extensions (for parking mostly) |
| 00:24.04 | CrashHD | tons of parking/state/hints stuff between the versions |
| 00:24.55 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
| 00:27.19 | CrashHD | *crickets* |
| 00:27.24 | Dovid | lol |
| 00:35.57 | *** join/#asterisk elriah (n=johnny@adsl-072-149-159-016.sip.bhm.bellsouth.net) |
| 00:36.16 | elriah | Greets, asteriteers! |
| 00:38.09 | *** part/#asterisk ctooley (n=ctooley@rrcs-71-42-115-242.sw.biz.rr.com) |
| 00:39.11 | elriah | Any clue when 1.4.1 is coming out? |
| 00:39.38 | Dovid | lol |
| 00:39.43 | Dovid | y is that ? |
| 00:40.16 | Dovid | my guess is when the bug tracker becomes a bit less busy |
| 00:40.56 | elriah | Oh, no reason. Just keep an eye on your local news channel for the story about "IT guy kills bosses with Cisco phones, story at 11" |
| 00:41.28 | DrukenLPY | elriah: i'm taking donations of cisco phones... :) |
| 00:41.38 | *** join/#asterisk foxxtrot (n=craig@67.185.241.244) |
| 00:42.08 | JT | the homicide evidence lockup don't do donations |
| 00:42.21 | elriah | DrunkenLPY: Ever seen Office Space (the movie)? You know the fax machine scene? |
| 00:42.54 | DrukenLPY | what is this paper jam, WHY DOES IT SAY PAPERJAM WHEN THERE IS NO PAPERJAM! |
| 00:43.37 | JT | WHAT IS PC LOAD LETTER |
| 00:43.55 | DrukenLPY | wtf is pc loadletter :) |
| 00:44.01 | Iamnacho | thats one of the best movies of all time |
| 00:44.03 | elriah | lol, now replace the fax machine with a pile of Cisco phones ... |
| 00:44.26 | DrukenLPY | elriah: i'd like a few cisco's to play with.... |
| 00:44.39 | DrukenLPY | but well, they are a bit outside my price range... |
| 00:44.42 | JT | they probably have ink bombs if you open or break them, to stop you stealing their megahurtz |
| 00:45.18 | DrukenLPY | elriah: i have office space on both vhs and dvd :) |
| 00:45.51 | JT | and LD |
| 00:45.53 | JT | and 16mm |
| 00:46.04 | Dovid | teehee: i luv that seen |
| 00:46.06 | DrukenLPY | nah... no laser disc :) |
| 00:46.07 | Dovid | in office space |
| 00:46.13 | JT | no film either? |
| 00:46.26 | DrukenLPY | nope... no projector... |
| 00:48.40 | ManxPower | elriah: you are in Birmingham, AL |
| 00:48.42 | ManxPower | ? |
| 01:00.19 | elriah | ManxPower: Yes, cool place. You? |
| 01:00.59 | ManxPower | Split my time between Steele, AL (about 1 hr NE of Birmingham off of I-59), Tuscaloosa, AL, and New Orleans, LA |
| 01:01.20 | elriah | Cool. What company do you work for? |
| 01:01.29 | ManxPower | Technically Northport, AL not Tuscaloosa |
| 01:01.41 | ManxPower | elriah: I do consulting work for companies in the New Orleans region |
| 01:01.48 | MaartenB | I have a problem with service asterisk start :( |
| 01:02.01 | ManxPower | MaartenB: then run "asterisk -cvvv" and see what the error is. |
| 01:02.02 | MaartenB | the safe_asterisk keeps restarting asterisk because it is terminated with code 1 |
| 01:02.19 | elriah | Did you compile then run as root for the first time? |
| 01:02.19 | MaartenB | ManxPower, that is the problem, asterisk itself runs without any problems |
| 01:02.35 | MaartenB | running 'asterisk' from command line works |
| 01:02.55 | MaartenB | it is the /etc/init.d/asterisk script that does something together with safe_asterisk |
| 01:02.56 | ManxPower | MaartenB: as the SAME user as Asterisk is running as? |
| 01:03.16 | ManxPower | MaartenB: both are just shell scripts, take a look at them. |
| 01:03.22 | MaartenB | root |
| 01:03.42 | elriah | MaartenB: but the init script is starting it as another user, right? So it's a permissions problem. |
| 01:04.10 | ManxPower | elriah: by default the init script starts it as root |
| 01:04.29 | elriah | MaartenB: What user is the init script starting asterisk as? |
| 01:04.31 | ManxPower | unless he used some evil pre packaged asterisk |
| 01:04.46 | MaartenB | you are right |
| 01:05.00 | MaartenB | it was trying to run as a 'asterisk' user |
| 01:05.01 | MaartenB | not as root |
| 01:05.10 | MaartenB | changing it too root solved the problem |
| 01:05.10 | elriah | Right, that's correct. Don't change it - fix your permissions. |
| 01:05.15 | MaartenB | you guys are great :) |
| 01:05.17 | elriah | NO NO NO |
| 01:05.20 | elriah | lol |
| 01:05.23 | elriah | Don't do it man! |
| 01:05.24 | puzzled | MaartenB: from a security perspective it makes sense to run it as non-root |
| 01:05.29 | MaartenB | yes |
| 01:05.31 | MaartenB | I understand |
| 01:05.35 | puzzled | ok |
| 01:05.47 | elriah | That's the typical "I understand, but I'm going to run it as root anyway". |
| 01:05.51 | MaartenB | but I know figured out what to look for :) |
| 01:05.57 | MaartenB | no no |
| 01:06.11 | elriah | Generally, that situation happens when you run it by just typing asterisk after you compile as root. |
| 01:06.12 | MaartenB | at first, we get it to work, second, we make it perfect :) |
| 01:06.15 | ManxPower | elriah: I do Asterisk, WAN, Linux admin |
| 01:06.28 | elriah | In the future, run it with asterisk -U asterisk -G astersk -vvvgc |
| 01:06.35 | elriah | Of course, after you create the user/group asterisk. |
| 01:06.40 | ManxPower | Cisco routers, Polycom phones, Catalyst switches |
| 01:06.47 | elriah | And if you're using POTS lines, add the asterisk user to your dialout group. |
| 01:07.01 | elriah | ManxPower: What company do you work for? |
| 01:07.21 | ManxPower | elriah: BTEL Consulting, a 1 person consulting company - me. |
| 01:07.51 | JT | elriah: POTS or any type of hardware |
| 01:08.03 | elriah | ManxPower: Cool. |
| 01:08.04 | MaartenB | it is now running as user asterisk |
| 01:08.14 | ManxPower | I'm pretty picky about accepting new clients, but I do accept a new client every couple of years. |
| 01:08.17 | MaartenB | thanks elriah, ManxPower |
| 01:08.27 | elriah | np, happy asterisking |
| 01:08.47 | JT | ManxPower: you get enough business to make a living off just a handful of clients? |
| 01:09.27 | ManxPower | JT: Yes. I've had 1 client for 10 years, another client for 5 years. Those are my two main clients. My occasional clients I've all had for at least 10 years. |
| 01:09.53 | *** join/#asterisk tb0301s (n=sowa@brln-4db135fd.pool.einsundeins.de) |
| 01:09.54 | puzzled | ManxPower: nice one |
| 01:10.24 | ManxPower | I also tend to (usually) keep my expenses low, that way I don't have to work all the time. |
| 01:10.56 | JT | ManxPower: wow, they must have really big setups? |
| 01:12.08 | ManxPower | JT: my largest client is a $550 million/yr company has 20 or so offices in 2 states. I manage the routers, cisco switches, VLANs, QoS, T-1s, PRIs, Frame Relay, Firewall (such as it is), spam filtering, virus filtering, e-mail servers, asterisk, polycom phones |
| 01:12.20 | JT | ah ok |
| 01:12.35 | J4k3 | nice client |
| 01:12.35 | puzzled | heh you'll be busy alright |
| 01:12.35 | JT | so almost everything but the desktop crap |
| 01:12.35 | ManxPower | I don't do Windows, backups, or politics. Other people deal with those issues |
| 01:12.35 | JT | :P |
| 01:12.44 | ManxPower | or the AS/400 |
| 01:12.57 | puzzled | that one should need little intervention |
| 01:13.18 | puzzled | it dials home and big blue delivers replacing parts before you know it's (almost) broken |
| 01:13.20 | ManxPower | They do most of the day-to-day admin for Asterisk starting a couple of months ago and the day to day linux server admin |
| 01:15.03 | *** join/#asterisk drako (n=ljd@unaffiliated/luisjose) |
| 01:15.45 | ManxPower | I work my ass off about 1 week per month, do an average amount of work another week of the month, and do very little work the rest of the month. It's not a bad life. 8-) |
| 01:16.09 | *** join/#asterisk y3hsDotBiz (n=w00t@124.6.153.226) |
| 01:16.17 | ManxPower | If I find a potential client that I like, I'd consider taking on another client, but I do not actively seek new clients |
| 01:16.24 | y3hsDotBiz | hi guys |
| 01:16.34 | y3hsDotBiz | i would like to ask what error is this |
| 01:16.35 | y3hsDotBiz | Feb 20 16:04:56 DEBUG[26209]: app_dial.c:1587 dial_exec_full: Exiting with DIALSTATUS=CONGESTION. |
| 01:17.05 | ManxPower | y3hsDotBiz: that means the call could not be completed because of a congestion condition. |
| 01:17.13 | y3hsDotBiz | ok |
| 01:17.16 | y3hsDotBiz | what does it mean? |
| 01:17.27 | y3hsDotBiz | ? |
| 01:17.28 | ManxPower | y3hsDotBiz: I'd have to know what the Dial line that caused that message is |
| 01:22.33 | y3hsDotBiz | ok.. |
| 01:22.45 | y3hsDotBiz | does anyone know of a good iax2 voice prodiver |
| 01:26.25 | y3hsDotBiz | Feb 20 16:04:56 DEBUG[26209]: app_dial.c:1587 dial_exec_full: Exiting with DIALSTATUS=CONGESTION. --> does this mean theres a bandwidth problem? |
| 01:27.40 | JerJer | congestion can mean many things |
| 01:28.11 | JerJer | for whatever reason your provider could not or would not complete that call for you |
| 01:28.28 | *** join/#asterisk HockeyInJune (n=HockeyIn@pool-141-157-252-251.ny325.east.verizon.net) |
| 01:31.39 | y3hsDotBiz | ok |
| 01:31.54 | y3hsDotBiz | ive got an E1 with just 50 seats |
| 01:31.58 | y3hsDotBiz | on g729 |
| 01:35.41 | *** join/#asterisk Fr0zen_ (i=Fr0zen_@67.175.92.171) |
| 01:35.45 | Fr0zen_ | anyone here use cisco ip phone? |
| 01:35.47 | Fr0zen_ | 79xx ? |
| 01:39.01 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 01:39.30 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 01:39.31 | *** join/#asterisk helpmeimjustagir (n=philip@203.63.126.9) |
| 01:39.47 | [hC] | did something happen to the g729 registration server at digium? |
| 01:39.52 | [hC] | I cant connect to it with the registration utility |
| 01:40.26 | helpmeimjustagir | can anyone help with queue_log rolling? |
| 01:44.37 | JerJer | is that anything like cow tipping ? |
| 01:45.37 | helpmeimjustagir | strangely yes |
| 01:45.43 | *** part/#asterisk russellb (i=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 01:45.55 | Fr0zen_ | anyone? |
| 01:46.02 | helpmeimjustagir | i cant seem to over ride the default log rolling, |
| 01:46.10 | helpmeimjustagir | keeps killing my log analyzer |
| 01:46.19 | helpmeimjustagir | by rolling in the middle of the day |
| 01:49.34 | *** join/#asterisk hematitec (n=cratz@adsl-71-159-206-4.dsl.pltn13.sbcglobal.net) |
| 01:53.14 | Qwell | What's a gir? |
| 01:53.32 | Qwell | or did he mean "girl"? |
| 01:53.46 | Qwell | Unless Philip is a girls name... |
| 01:54.09 | JT | who/what? |
| 01:54.19 | Qwell | --> helpmeimjustagir (n=philip@203.63.126.9) has joined #asterisk |
| 01:54.20 | *** join/#asterisk pdt (n=ptinsley@68.53.40.50) |
| 01:54.23 | JT | gir is a character on invader zim, for what it's worth :P |
| 01:54.24 | JT | ah |
| 01:56.04 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 01:56.27 | hellojoe | Guys, how to set a Variable in a Macro and use it outside of the Macro. I am noticing that the value of the Variable inside the Macro is not getting transported the the dialplan when I try to access it elsewhere |
| 01:56.33 | hellojoe | am I missing something here? |
| 01:56.50 | hellojoe | Wondering how to pass values to and from the Macro inside of a dialplan |
| 01:57.01 | hellojoe | any help would be appreciated. Need a quick solution to a critical issue |
| 01:57.05 | hellojoe | thanks in advance |
| 01:59.17 | hellojoe | Anyone ???? |
| 01:59.22 | *** part/#asterisk mogorman (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 01:59.41 | JerJer | how are you setting the variable? |
| 02:00.07 | hellojoe | Set(VARIABLE=0) |
| 02:00.46 | JerJer | i don't think i have set variables in a macro |
| 02:01.08 | hellojoe | GotoIF($[${VARIABLE}=0]?50:40) |
| 02:01.31 | hellojoe | i want to branch out based on what happens within the macro |
| 02:01.45 | hellojoe | any other way to pass the results back to the calling routine?" |
| 02:02.37 | hellojoe | The set command is working (atleast that's what the CLI shows) but I am not sure if the variable pointer remains the same as I come out of hte macro |
| 02:04.11 | hellojoe | Anyone else tried this? How do we get back the results/variables from the Macro? |
| 02:04.24 | ez` | why my sipura 3000 does not detect correctly dtmf ; i try to enter my bank password ; it look like it ; dtmf to go out correctly ,,, |
| 02:04.40 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 02:08.25 | pingboy12 | having probs compiling asterisk 1.4 |
| 02:08.39 | pingboy12 | <PROTECTED> |
| 02:08.49 | pingboy12 | says configure script must be run, before running make |
| 02:09.00 | pingboy12 | but i did that |
| 02:10.12 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 02:14.41 | *** join/#asterisk techie (n=techie@67.181.184.170) |
| 02:16.54 | ManxPower | . |
| 02:16.55 | ManxPower | \ |
| 02:24.53 | Juppers | ManxPower - Got him to update my sip config. everything works happy now. Thanks again |
| 02:30.18 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 02:30.18 | *** mode/#asterisk [+o anthm] by ChanServ |
| 02:32.54 | *** join/#asterisk etfonhome (n=etfonhom@74-140-213-69.dhcp.insightbb.com) |
| 02:33.02 | *** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
| 02:33.09 | FuriousGeorge | hey all |
| 02:34.23 | FuriousGeorge | i need a better way to fallover from my providers.. ive tried four over the past two years and im not blaming them as much as my routes to them via my regular old business cable/dsl isps, but i wouldnt describe voip over optimum online in nj as reliable |
| 02:35.05 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 02:35.15 | FuriousGeorge | today my main provider sounded terrible, but my old/backup provider was great. currently i just jump based on dialstatus and keep the qualifys under 250 |
| 02:35.19 | FuriousGeorge | but that didnt help today |
| 02:37.24 | *** join/#asterisk lowlevel (n=Stuart@CPE000e0c057fad-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 02:39.06 | ManxPower | Juppers: you're welcome. |
| 02:39.26 | *** join/#asterisk sumasuma (n=kurukko@61.14.86.23) |
| 02:39.37 | ManxPower | FuriousGeorge: How do you keep your job with so many voice quality issues? |
| 02:40.19 | sumasuma | what does "bpviol" in the zap show status mean ? with a PRI line |
| 02:40.44 | Strom_C | bipolar violation |
| 02:40.44 | JT | FuriousGeorge: ideally you need to meter packetloss and jitter as well as latency |
| 02:41.16 | sumasuma | Strom_C: it shows 2698 on my PRI, is that OK or abnormal ? |
| 02:41.35 | ManxPower | sumasuma: no it is not normal. you have a serious issue with the line. |
| 02:41.35 | Strom_C | you should be getting as close to 0 as possible |
| 02:42.01 | sumasuma | Strom_C: What should I do to get those ones ? |
| 02:42.57 | *** join/#asterisk lowlevel (n=Stuart@CPE000e0c057fad-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 02:43.11 | lowlevel | should have got an imac :/ |
| 02:44.52 | FuriousGeorge | JT: yeah im running my traceroute |
| 02:45.07 | ManxPower | sumasuma: I'd call my telco tech and say "I'm getting bipolar violations, I have my equipment set to B8ZS/ESF." The tech will say "Yup, that's correct. I'll call you back shortly." And shortly the tech will call back and it will be fixed. |
| 02:45.16 | FuriousGeorge | ManxPower: because i get paid to get told by verizon that they have issues, which i then relay to the client |
| 02:45.37 | ManxPower | FuriousGeorge: must be nice. |
| 02:45.50 | ManxPower | I get paid to make it work as close to %100 as possible. |
| 02:45.59 | JT | FuriousGeorge: traceroute does not tell you jitter or packet loss issues really |
| 02:46.04 | FuriousGeorge | ManxPower: describe your isp |
| 02:46.12 | JT | it can show packet loss a little |
| 02:46.17 | JT | not the ideal tool for it |
| 02:46.27 | JT | i'm after a jitter detection utility |
| 02:46.39 | FuriousGeorge | verizon said something popped in new york and i have 10% ping loss on a hop in nyc between me and my provider |
| 02:46.50 | *** join/#asterisk znoG_ (n=gs@97-228-126-200.fibertel.com.ar) |
| 02:46.52 | JT | argh 10% |
| 02:46.57 | JT | that's death for most codecs |
| 02:47.10 | ManxPower | FuriousGeorge: Um, I don't route calls via an ISP! I route them via PRIs from a CLEC. |
| 02:47.18 | FuriousGeorge | ManxPower: lucky you |
| 02:47.22 | sumasuma | ManxPower: thanks. |
| 02:47.37 | JT | i'd go so far as saying 10% packet loss equates to a 90% loss in Internet connection utility |
| 02:48.11 | FuriousGeorge | JT: how do you mean? 90% of people use the web for web surfing, not phone calls |
| 02:48.14 | ManxPower | the only time we route calls via an ISP is if for some odd reason we are having problems with the CLEC and that doesn't happen very often, then we route them via a IP over a QoS'd T-1 to another office with a different telco |
| 02:48.21 | JT | FuriousGeorge: web surfing too |
| 02:48.28 | JT | 10% is REALLY HIGH packet loss |
| 02:48.28 | JT | imho |
| 02:48.39 | JT | it makes web pages take an incredibly long time to load |
| 02:48.43 | FuriousGeorge | i dont know that i have 10% packet loss anyways, though. i just know 10% of my pings get dropped |
| 02:48.47 | FuriousGeorge | at a certain hop |
| 02:48.58 | ManxPower | sorry, only of BOTH the telco routes fail would we route them via an ISP and that has not happened since Katrina |
| 02:48.59 | JT | it's a pretty good indication |
| 02:49.18 | JT | it's hard to measure udp packet loss, you need application level access for that |
| 02:49.20 | DrkShdw | FuriousGeorge: some routers are setup to drop icmp requests above a certain load or throughput. |
| 02:50.05 | FuriousGeorge | ManxPower: i always advise people, "you pay 300 for 6 landlines and no phone system, why not pay 250 for some dedicated loop isdn" |
| 02:50.12 | JT | yeah some ratelimit |
| 02:50.25 | JT | DrkShdw: that's why it's best to ping against a known peer |
| 02:50.30 | FuriousGeorge | DrkShdw: yeah, i just take it to indicate congestion, not packetloss |
| 02:50.32 | JT | if you know it's normally 0% loss.... |
| 02:50.47 | JT | packetloss is an indication of congestion |
| 02:50.52 | FuriousGeorge | JT: hadnt thought of that |
| 02:50.53 | DrkShdw | FuriousGeorge: right, pinging the endpoints is usually a better indication |
| 02:51.16 | JT | and generally against a KNOWN ping partner, ping is a good indication of packetloss |
| 02:51.24 | DrkShdw | agreed |
| 02:51.29 | FuriousGeorge | ManxPower: if youre referring to my voice quality issues from yesterday, i think these are related |
| 02:52.01 | FuriousGeorge | .so-6-1-0.XL3.NYC4.ALTER.NET 10.0% 12172 32.1 22.8 22.2 135.3 4.1 |
| 02:52.46 | FuriousGeorge | thats loss, pings, last, average, best, worst, sigma |
| 02:54.45 | ManxPower | A UDP "ping" would be a better measure of latency and packet loss as many routers handle ICMP (normal ping) as a very low priority and will drop them before any other type of packet |
| 02:55.56 | FuriousGeorge | yeah weve been saying that |
| 02:57.07 | Bobthehunter | crickets ? |
| 02:57.13 | Bobthehunter | thats like trolls but smaller ;) |
| 02:57.39 | Bobthehunter | oups. bad post.. my scroll bar as half way up.. otught was just said ;) |
| 02:58.09 | FuriousGeorge | heh |
| 03:04.32 | fetcher | Is it possible to do TBCT / 2BCT when the PRI comes into an outboard SIP gateway box? (Cisco AS5300) |
| 03:06.14 | ManxPower | fetcher: Sure...if the SIP/PSTN gateway supports it |
| 03:06.37 | *** join/#asterisk ltd (n=z@202-161-1-26.dyn.iinet.net.au) |
| 03:07.09 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 03:07.09 | fetcher | ManxPower: Seems that newer IOS builds do, but how would * request the 2B transfer over SIP? |
| 03:07.37 | Qwell | it wouldn't. |
| 03:07.49 | Qwell | all asterisk cares about is SIP in that case... |
| 03:08.00 | Qwell | the box should be smart enough to figure it out |
| 03:08.48 | fetcher | Qwell: so, enabling SIP Reinvites, to get * out of the hairpinning RTP path might be enough? |
| 03:09.10 | Qwell | one would hope |
| 03:11.11 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 03:11.35 | fetcher | could be a moot point, if we can't find the magic words to make telco turn this feature on. |
| 03:11.52 | fetcher | assuming their switch even supports it |
| 03:15.25 | sumasuma | Is there is any GSM (mobile) device with IAX in a box ? |
| 03:15.33 | *** join/#asterisk J4k3 (n=jsuter@dhcp-12-197-128-58.intrastar.net) |
| 03:16.55 | fetcher | sumasuma: probably not. It's hard enough to find a normal IP phone w/ IAX :( |
| 03:17.20 | ManxPower | fetcher: I don't think SIP has that concept. The gateway would need to determine how to do the actual transfer. |
| 03:17.41 | *** join/#asterisk ippup (n=chatzill@129.94.190.121) |
| 03:17.47 | ippup | hi all |
| 03:18.51 | ippup | do i need to add anything to my extentions.conf if i have just setup an * server with one account and i then want to call a friend on a softphone is the US ? |
| 03:19.26 | fetcher | ManxPower: what about "call deflection" -- redirecting a call before it's picked up? |
| 03:19.28 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 03:19.29 | ippup | i can call local extentions that i setup on my server but not remote sip users |
| 03:19.30 | JT | SIP lacks a LOT of telco concepts :) |
| 03:19.32 | ManxPower | ippup: that depends on so many things |
| 03:19.39 | ippup | Manx |
| 03:19.42 | fetcher | Would a SIP "Refer" event deflect? |
| 03:20.12 | ippup | ManxPower: my * server is not FW or NAT'ed and has a public ip |
| 03:20.30 | JT | fetcher: hmm, it's been said before, but it is the job of the gateway to be able to facilitate it anyway, so any talk without knowing the gateway's capabilities is going to be guessowrk |
| 03:20.34 | JT | guesswork |
| 03:20.37 | ManxPower | ippup: your friend's softphone would either have to register to your server or your server would have to route the call to whatever VoIP company your friend uses |
| 03:21.21 | ippup | ManxPower: correct, how would my server route the call to VOIP company my friend uses ? |
| 03:21.37 | ippup | ManxPower: Do i need to add entries in extentions.conf ? |
| 03:21.49 | ManxPower | ippup: you would have to have an account of your own with that company or with someone that company peers with. |
| 03:22.06 | fetcher | JT: Cisco AS5300 here, IOS 12.2. I'm digging through the Cisco docs... |
| 03:22.36 | ippup | ManxPower: i was under the impression that i could call any sip user anyhwere from ym * server ? |
| 03:22.47 | JT | fetcher: is it a function you really need? 2BCT/ECT etc always seems to be "too hard basket" material |
| 03:23.31 | ManxPower | ippup: You should be able to if you know the person's IP and the person does not have NAT or FW. IF they have NAT or FW enabled then you have to portforward on their NAT box. |
| 03:23.57 | ManxPower | and the person's phone accepts calls from random servers |
| 03:24.43 | fetcher | JT: not really... just trying to conserve B channels. |
| 03:24.46 | *** join/#asterisk linagee (n=linagee@unaffiliated/linagee) |
| 03:24.50 | linagee | wtf? :( "chan_iax2.c:9183 iax2_exec: No dial application registered" |
| 03:25.07 | ippup | ManxPower: i understand the FW and NAT issues but my question was rather, why can't any * user call any other remote * user ? |
| 03:25.08 | linagee | i'm trying asterisk to asterisk. trying to send the calls to another box using switch => |
| 03:25.29 | ManxPower | ippup: Dial(SIP/ip.add.ress.of.remote.device) |
| 03:25.45 | ManxPower | ippup: then you have to figure out why the call is not completing |
| 03:25.59 | ManxPower | and that would not be an asterisk issue, that would be an "everything except aserisk" issue |
| 03:26.06 | *** join/#asterisk Strom_M (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 03:26.17 | ippup | ManxPower: i'm confused ! |
| 03:26.27 | ManxPower | linagee: is that an actual copy and paste of the message? |
| 03:26.28 | fetcher | ippup: remote server would need to allow your call to connect, also |
| 03:26.56 | ippup | fetcher: wouldn't a public VOIP provider allow that ? |
| 03:27.09 | ManxPower | ippup: many of the free ones do. |
| 03:27.34 | ManxPower | but they still require an account |
| 03:27.34 | ippup | why would i allow anyone to call my internal sip accounts ? |
| 03:27.36 | ippup | would=wouldn't |
| 03:27.52 | ManxPower | ippup: telemarketers for one thing |
| 03:28.12 | ManxPower | if you think email spam is bad, just wait until telemarketers can call you for FREE. |
| 03:28.15 | ippup | ManxPower: so is sip to sip possible at all ? |
| 03:28.30 | techie | i hate telemarketers |
| 03:28.42 | ManxPower | ippup: of course. Heck you could dial the friend using a softphone without any other device/server involved. |
| 03:28.43 | ippup | ManxPower: i'm really missing somwthing here ! |
| 03:29.12 | ippup | ManxPower: as long as i have a softphone ? |
| 03:29.20 | *** join/#asterisk froguz (n=froguz@pc-6-103-104-200.cm.vtr.net) |
| 03:29.35 | froguz | hi |
| 03:29.38 | ippup | ManxPower: and a sip account somewhere |
| 03:29.54 | froguz | somebody has experience with ranch network appliance and Asterisk?? |
| 03:30.03 | ManxPower | ippup: In theory two SIP phones can dial each other directly without any server or service provider. |
| 03:30.11 | ManxPower | ippup: what service provider does your friend have? |
| 03:30.24 | froguz | i need to know if i can configure ranch network appliance as active-passive mode |
| 03:30.26 | ippup | ManxPower: aarnet.edu.au |
| 03:30.47 | ManxPower | ippup: I didn't know aarnet was an internet telephone company |
| 03:31.02 | *** join/#asterisk Moobius (i=Moobius@www2.techcavalry.com) |
| 03:31.02 | ManxPower | ippup: What SIP device/softphone does your friend have? |
| 03:31.11 | ippup | ManxPower: xlite |
| 03:31.13 | froguz | i mean, having just one server running and when it comes down, the secundary server (identically configures) take call control |
| 03:31.26 | ManxPower | ippup: what happens if you Dial(SIP/ip.add.ressoffriend) in asterisk |
| 03:31.27 | Moobius | is there a release date for 1.14.1 announced? |
| 03:31.37 | JT | heh, aarnet |
| 03:31.50 | froguz | what i can see in the RN documents is that works in active-active mode |
| 03:31.50 | JT | ManxPower: aarnet run the national inter-university network in australia |
| 03:31.52 | ippup | ManxPower: SIP/2.0 404 User not found |
| 03:31.54 | ManxPower | Moobius: I don't think there has ever been a release date for any verison of asterisk ever |
| 03:32.04 | JT | they have a few gbit/s connecting most unis |
| 03:32.06 | puzzled | Moobius: there never is. you will see it when it is released |
| 03:32.08 | Moobius | ManxPower: heh. |
| 03:32.10 | ManxPower | ippup: that is what your friend's phone is sending back to asterisk. |
| 03:32.44 | ippup | ManxPower: but i get that from any sip i try, ANY ! |
| 03:32.50 | ManxPower | ippup: so your friend's softphone is rejecting the call. You might try Dial(SIP/1234@ip.add.ressoffriend) |
| 03:33.00 | ippup | ManxPower: except local sip on my * box |
| 03:33.02 | froguz | does somebody have an idea of what am i asking? |
| 03:33.30 | ManxPower | ippup: I have NO idea what "user" X-Lite is expecting you to send. maybe the phone number he put in the X-Lite config. |
| 03:33.47 | ManxPower | Dial(SIP/userontheremotephone@ip.add.ressofremotephone) |
| 03:33.51 | puzzled | froguz: yes but I have no idea. how about calling ranche? |
| 03:34.40 | froguz | my english is very poor :( |
| 03:34.50 | ManxPower | froguz: what you are asking is so complex it would require at least a day of my time to even design something that might have a chance to work. I don't do that kind of work for free. |
| 03:34.53 | froguz | shane on me to call range |
| 03:35.13 | froguz | i've send them an email, but i need a quick answer |
| 03:35.23 | froguz | shame* |
| 03:36.48 | froguz | ManxPower, i think this is not a development question, is about knowing the range network features products, or experience with them |
| 03:38.03 | froguz | but i know it isn't an easy to answer question |
| 03:39.11 | froguz | so, i'll probably have to wait until geting an answer from ranch |
| 03:39.31 | froguz | :( |
| 03:40.44 | *** join/#asterisk dseeb_ (n=dcb@CPE-58-169-80-108.vic.bigpond.net.au) |
| 03:41.07 | froguz | or maybe start thinking on a alternative solution to get 2 redundant IP-PBX |
| 03:43.03 | ManxPower | froguz: any redundant system will be so complex that it will fail more than a single system |
| 03:44.08 | JT | Thanks for the inquiry. We are currently working on adding onboard DSPs |
| 03:44.09 | JT | with echo can capabilities but are still a few months away from production. |
| 03:44.19 | JT | that was the reply from redfone about hardware EC |
| 03:44.25 | froguz | maybe i just don't need a passive-active system... i'm courious about how asterisk look for an unbusy channel on a PRI interface |
| 03:45.24 | *** join/#asterisk froguz (n=froguz@pc-6-103-104-200.cm.vtr.net) |
| 03:45.47 | froguz | maybe a T1/E1 Multiport Repeaters will do the work for me |
| 03:46.01 | JT | froguz: you're looking for PRI failover? |
| 03:46.09 | froguz | having an active-active cluster model |
| 03:46.33 | froguz | i need to have a redundant IP-PBX using the same 4 PRIs |
| 03:46.40 | *** join/#asterisk pdt (n=ptinsley@c-68-53-40-50.hsd1.tn.comcast.net) |
| 03:47.22 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 03:47.25 | JT | froguz: there's an article on options on the wiki |
| 03:47.30 | JT | but basically |
| 03:47.52 | JT | there are a few posibilities, none of them allow you to keep CURRENT CALLS if a machine fails |
| 03:48.06 | JT | but they allow another to take over and start handling calls again if 1 fails |
| 03:48.06 | puzzled | froguz: how about this one: http://www.junghanns.net/en/ISDNguard_produkt.html |
| 03:48.15 | froguz | i'd read that i can use a T1/E1 multiport repeater while one server would not be running Asterisk until the other server is dead |
| 03:48.32 | JT | that's simply an L1 failover switch with asterisk heartbeat, puzzled |
| 03:48.39 | JT | most of the solution is in software |
| 03:48.47 | JT | and you still lose current calls |
| 03:48.53 | ManxPower | froguz: Asterisk automatically hunts for the first available PRI channel, if it cannot find one it will exit Dial and set the value of DIALSTATUS |
| 03:48.59 | JT | it's unavoidable with current hardware and software |
| 03:49.05 | puzzled | JT: there is no stateful failover with Asterisk afaik |
| 03:49.14 | JT | puzzled: yes it is not possible |
| 03:49.28 | puzzled | only thing that can possibly do it is freeswitch |
| 03:49.37 | JT | is it even there? |
| 03:49.46 | JT | it's a heavy task |
| 03:50.10 | puzzled | or you can use the stratus/nec fault tolerant solution. that one works for sure but adds the cost of double the hardware |
| 03:50.48 | JT | puzzled: is that to work with asterisk or any open source B2BUA? |
| 03:50.55 | JT | you can get hardware fault tolerant solutions |
| 03:51.02 | puzzled | or just get a carrier class MaxTNT or APX8000 that pretty does 5 nines |
| 03:51.05 | JT | but they're carrier grade, and don't involve asterisk :) |
| 03:51.11 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 03:51.21 | puzzled | JT: the stratus/nec box runs RHEL & Asterisk |
| 03:51.25 | puzzled | or CentOS off course |
| 03:51.28 | JT | 5nines isn't actually much from a telco standpoint |
| 03:51.47 | JT | puzzled: how on earth do they do parallel comparative computing with that? :P |
| 03:52.38 | puzzled | the software does not know there are actually two servers in the chassis. The moment one of the servers gets a faulty component it switches to the 2nd server which has been running in the exact same state as the first one |
| 03:52.49 | JT | the ITU standard for switch availability is 99.999985% iirc |
| 03:53.03 | puzzled | that's the avg of the stratus boxes |
| 03:53.03 | *** join/#asterisk bmg505 (n=leon@c1-173-11.rndf.isadsl.co.za) |
| 03:53.06 | ippup | ManxPower: do you have a sip number ? |
| 03:53.15 | JT | puzzled: how does it run them both in the same state? |
| 03:53.25 | ManxPower | ippup: not with a public server. |
| 03:53.25 | JT | sounds like a highly modified asterisk |
| 03:53.50 | puzzled | JT: special software at the OS level & some hardware glue. Asterisk runs unchanged |
| 03:54.02 | Qwell | What can it do, 3 calls? |
| 03:54.11 | JT | puzzled: must be extremely special |
| 03:54.30 | puzzled | Qwell: yeah the specs are a bit old so I don't expect too much from it. Would still get an APX8000 anyway |
| 03:54.49 | puzzled | JT: the concept is pretty common in telco country |
| 03:55.17 | ManxPower | puzzled: what verison of Asterisk and how much does the box cost? |
| 03:55.21 | froguz | ManxPower, what are you telling me is that asterisk will succefully find an available PRI channel connected to a PRI repeater, even if some of them are being used for the other PBX? |
| 03:55.29 | JT | puzzled: i don't understand how asterisk could be unmodified, so if there are 2 servers say trying to dial sip, does the hardware catch one of the requests, or combine them or something? |
| 03:55.31 | puzzled | ManxPower: from $10k and afaik any version of asterisk you like |
| 03:55.36 | ippup | ManxPower: so i have to get an account with a voip provider, or setup my asterisk server with a trunk to a voip provider so i can sip call you ? |
| 03:55.42 | ManxPower | froguz: no. |
| 03:55.51 | froguz | buuh :( |
| 03:55.53 | JT | puzzled: yeah i know it can be done with the right hardware solution, it just doesn't usually involve asterisk + linux |
| 03:56.12 | ManxPower | ippup: all SIP to any device I have access to is behind a corporate firewall. |
| 03:56.13 | JT | after all, you got to get it right for only 2 hours downtime every 20 years |
| 03:56.25 | puzzled | JT: don't think of them as two separate servers. It's one logical box consisting of two hardware entities that are always in the same state. |
| 03:56.29 | puzzled | like RAID1 |
| 03:56.39 | JT | puzzled: yes but it's asterisk :) |
| 03:56.44 | JT | and linux |
| 03:56.50 | JT | neither were designed for this |
| 03:56.51 | ippup | ManxPower: so how do i sip call you ? |
| 03:57.01 | ManxPower | ippup: nobody can sip call me. |
| 03:57.02 | froguz | so, it doesn't "listen" the channel like analog ones, it just keep an status flag for every channel it had used? |
| 03:57.07 | JT | how can you guarantee both will give exactly the same state? |
| 03:57.19 | puzzled | JT: that's the cool part about it. Just a little OS sauce and off you go. It's already part of RHEL or even the kernel |
| 03:57.25 | Qwell | puzzled: what happens when you accidently delete a file on RAID1? |
| 03:57.31 | JT | puzzled: open source? |
| 03:57.41 | Qwell | faster than you can blink, it's deleted...on all of the disks |
| 03:57.46 | puzzled | Qwell: same as always :) |
| 03:57.52 | JT | that's not the problem Qwell |
| 03:57.54 | Qwell | now think about that with "stateful failover" |
| 03:57.55 | ippup | ManxPower: VOIP/SIP Asterix does not work like i thought it did, it looks like smtp but not |
| 03:58.02 | JT | we're talking failover solutions, not backup solutions |
| 03:58.10 | ippup | Man |
| 03:58.12 | puzzled | JT: yeah since it's part of RHEL and that stuff is OSS |
| 03:58.13 | Qwell | JT: I was drawing an analogy :P |
| 03:58.13 | ManxPower | ippup: asteirsk is working just fine, it is X-Lite that is refusing your call |
| 03:58.26 | JT | puzzled: any name for this glue? |
| 03:58.40 | JT | and does it only work with esoteric hardware? |
| 03:58.42 | ManxPower | ippup: ALL voice over IP that I do is behind a corporate firewall and has no access to the internet. |
| 03:59.02 | puzzled | Qwell: go to the stratus website and read about it. If e.g. a cpu on server1 dies, the logic (soft/hard) switches over to server1 without loss at the OS/application level |
| 03:59.10 | ippup | ManxPower: do you have a VOIP gateway ? |
| 03:59.11 | puzzled | to server2 |
| 03:59.14 | Qwell | I don't buy it |
| 03:59.24 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 03:59.26 | Qwell | if the CPU dies, we're talking milliseconds... |
| 03:59.31 | *** join/#asterisk ping2921 (n=marc3234@206-248-177-192.dsl.teksavvy.com) |
| 03:59.31 | ManxPower | ippup: yes, it is also behind the corporate firewall. I am not the person to talk to if you want to make test calls. |
| 03:59.47 | ManxPower | ippup: we send no calls over the internet ever. |
| 03:59.49 | JT | Qwell: it does work |
| 03:59.49 | puzzled | Qwell: a few years back I sold the big carrier grade Stratus boxes and have seen it happen |
| 03:59.56 | JT | Qwell: most telco switches do this |
| 04:00.01 | ping2921 | how can I do a: goto to previous menu in an ivr? |
| 04:00.12 | JT | Qwell: comparative parity or so, is one technique |
| 04:00.17 | Qwell | JT: telco switches don't do all that in linux :P |
| 04:00.20 | [TK]D-Fender | ping2921 : "show application goto" |
| 04:00.30 | ippup | ManxPower: i don;t want to make test calls, i just want to understand why i can't call 123456@remotedomain.com from my asterisk box :-( |
| 04:00.30 | JT | Qwell: true, which is what i find the intriguing bit |
| 04:00.32 | froguz | so.. ranch network solution is good for my needs, except for the fact that those two servers will use the same PRI links |
| 04:00.43 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 04:00.55 | puzzled | Qwell: maybe a good idea to take all the devs to a telco datacenter and see how this stuff works. Perhaps it will put stateful failover higher (if at all) on the roadmap :) |
| 04:00.57 | [TK]D-Fender | ippup : is your server behind NAT? |
| 04:01.14 | ippup | [TK]D-Fender: no NAT or FW |
| 04:01.23 | ippup | [TK]D-Fender: public address space |
| 04:01.37 | [TK]D-Fender | ippup : pastebin your [general] section of sip.conf as your dialplan segment that places the call. |
| 04:01.49 | ippup | [TK]D-Fender: sec... |
| 04:01.56 | ManxPower | ~pb |
| 04:02.00 | jbot | pb is, like, a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 04:02.09 | froguz | well. too late for me, i need some rest |
| 04:02.17 | froguz | thanks for your eternal orientation |
| 04:02.28 | ping2921 | the goto command can jump to a context, but how do I find the context of the previous menu? |
| 04:02.33 | puzzled | JT: the stratus solution is too expensive given its specs and how many calls it can handle. At the end of the day an AS5xxx or APX8000 prolly makes more sense |
| 04:02.34 | JT | puzzled: asterisk is aimed at the pbx market |
| 04:02.53 | [TK]D-Fender | ping2921 : You mean being in a given menu you don't know necessarily where youc ame from? |
| 04:02.55 | froguz | nites everybody |
| 04:02.55 | JT | puzzled: ah ok, does the other stuff have stateful failover? |
| 04:03.00 | ping2921 | yes |
| 04:03.27 | [TK]D-Fender | ping2921 : Yes you SHOULD know, or Yes I am correct in that you could end up there form any of 10 places? |
| 04:03.27 | puzzled | JT: sure but of all the companies with > 1000 employess I talked to, 100% asked questions about resilience and were not very comfortable with the absence of stateful failover |
| 04:04.12 | puzzled | JT: not really unless you use them with SS7. they are carrier class boxes and although at some point something will break they have a much higher mtbf than your avg linux server |
| 04:04.25 | ippup | [TK]D-Fender: http://pastebin.ca/365691 |
| 04:04.36 | ping2921 | example: in menu3 I do a goto to menu5, in menu5 how can I jump back to previous menu (menu3 in this case). |
| 04:04.39 | JT | puzzled: yeah, if you have 1000 people in 1 building, you might be able to afford a stateful failover solution, but usually it's in the relm of carriers |
| 04:05.00 | puzzled | maybe that's were the real money is :) |
| 04:05.08 | JT | puzzled: a lot of telco systems have components that don't have failover |
| 04:05.14 | ping2921 | for example, I want the caller to go back to previous menu if they press on *. |
| 04:05.14 | JT | like line cards |
| 04:05.17 | puzzled | but I agree, Asterisk is not suited for the carrier market |
| 04:05.36 | JT | but they run on a big backplane, and everything else keeps working, and an alarm relay and snmp message is triggered |
| 04:05.43 | puzzled | yup |
| 04:06.05 | puzzled | and the backup linecard kicks in 50ms after the 1st one died and everything is automagically routed to card#2 |
| 04:06.14 | JT | surely most businesses don't have pbxes that do line card failover |
| 04:06.25 | JT | hah, i dunno if they always have backup linecards |
| 04:06.30 | JT | especially for POTS |
| 04:06.39 | [TK]D-Fender | ippup : you setup is royally mixed up. |
| 04:06.42 | puzzled | if they have nortel passports, the ones I have seen do have that |
| 04:06.44 | [TK]D-Fender | ippup : you have : |
| 04:06.46 | [TK]D-Fender | nat=yes |
| 04:06.47 | JT | cool |
| 04:06.52 | [TK]D-Fender | and |
| 04:06.54 | [TK]D-Fender | externip=voice.mydomain.org.au |
| 04:07.03 | [TK]D-Fender | ippup : And you said * was NOT behind NAT. |
| 04:07.10 | ippup | [TK]D-Fender: ok will fix that |
| 04:07.24 | ping2921 | tkd: solution? |
| 04:07.29 | [TK]D-Fender | ippup : and "externip" must be assigned a NUMBER, not a domain. thats what "externhost" is for. |
| 04:07.47 | ManxPower | ping2921: exten => *,1,Goto(menu1,1) |
| 04:08.05 | ippup | [TK]D-Fender: since i'm not NAT'ed i don;t need to |
| 04:08.09 | [TK]D-Fender | ping2921 : are you guranteed that there IS always a previous menu? |
| 04:08.11 | ManxPower | [TK]D-Fender: ippup is on a public IP with no FW and no NAT |
| 04:08.19 | ping2921 | manx: that goes back to mainmenu, not the previous menu. |
| 04:08.36 | ManxPower | ping2921: then your menudesign is flawed. |
| 04:08.39 | [TK]D-Fender | ManxPower : Hes confirmed that he could ahve come from one of several other places. its the variable nature thats a tad more complex |
| 04:08.48 | ippup | [TK]D-Fender: my mistake re the 2 entries in the sip.conf, i was testing a few things, removed now |
| 04:08.51 | ManxPower | ping2921: you want each menu and subment in it's own context. |
| 04:09.00 | [TK]D-Fender | ping2921 : use "gosub" and "return" to jump around your menus |
| 04:09.48 | [TK]D-Fender | ippup : Ok, and kill the "domain" line as well. I also asked to see the line you are dialing out over the internet. |
| 04:11.24 | *** join/#asterisk pdt (n=ptinsley@68.53.40.50) |
| 04:20.04 | ippup | [TK]D-Fender 0290297101@iinetphone.iinet.net.au |
| 04:20.15 | [TK]D-Fender | ippup : where/how are you dialing that? |
| 04:20.16 | ippup | [TK]D-Fender is that what your after ? |
| 04:20.38 | ippup | [TK]D-Fender dialing that from my softphone |
| 04:21.19 | [TK]D-Fender | ippup : thats the problem. You softphone is bihint your NAT, and the call is NOT passing through *. Your phone is too dumb to know how to deal with its circumstances. |
| 04:21.48 | ippup | [TK]D-Fender: my sofphoen is not behind a NAT |
| 04:22.39 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 04:24.21 | [TK]D-Fender | ippup : where is it then? |
| 04:24.38 | [TK]D-Fender | ippup : either way, the call has nothing to do with *. |
| 04:24.56 | ippup | [TK]D-Fender: on the same non-FW'ed non-NAT'ed subnet |
| 04:24.59 | [TK]D-Fender | ippup : You gave it a full domain and everything... its going there direct (or atleast going to TRY to) |
| 04:25.23 | [TK]D-Fender | ippup : So your problem is your softphone, not *. |
| 04:26.03 | ippup | [TK]D-Fender: mm... what sofphone do you recommend i use then ? |
| 04:28.26 | *** join/#asterisk pdt (n=ptinsley@c-68-53-40-50.hsd1.tn.comcast.net) |
| 04:28.51 | [TK]D-Fender | ippup : Don't know what the problem is with it, I just know WHERE the problem is. Nothing to suggest besides adding an entry to your * dialplan to dial it for you. |
| 04:30.11 | ManxPower | [TK]D-Fender: that's pretty much what I told him. |
| 04:32.45 | ippup | [TK]D-Fender/ ManxPower: thanks guys, will investigate the softpphone issues |
| 04:40.47 | *** join/#asterisk angom_h (n=Angel@www.aysco.com.mx) |
| 04:42.30 | [TK]D-Fender | ManxPower : Can lead a horse to the water.... but the SPCA won't let you hold its head under.... |
| 04:42.39 | *** join/#asterisk bkw_ (i=brian@adsl-70-143-62-84.dsl.tul2ok.sbcglobal.net) |
| 04:43.06 | ManxPower | [TK]D-Fender: unfortunatly |
| 04:43.20 | ManxPower | I know a few users^H^H^H^H^Hhorses I'd like to do that to |
| 04:52.45 | *** join/#asterisk VampBoi (n=VampBoi@68-187-206-043.dhcp.ahvl.nc.charter.com) |
| 04:57.44 | Qwell | what did the horses ever do, to be likened to users? |
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| 05:00.56 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 05:01.11 | hellojoe | hi, has anyone figured out a way to return value from a macro. I tried setting up a VARIABLE within the macro but the values are NOT getting tranported back to the dialplan |
| 05:01.39 | *** join/#asterisk topping (n=topping@adsl-68-122-119-108.dsl.pltn13.pacbell.net) |
| 05:06.10 | *** join/#asterisk glm2k (n=glm@rrcs-24-199-11-46.west.biz.rr.com) |
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| 05:06.17 | Snake-Eyes | Is there a variable that holds what trunk was dialed? (eg xyz call came in on trunk Zap 1, 2, 3 or 4 ?) |
| 05:06.36 | k-man | is the spa901 ip phone ok? |
| 05:08.37 | [TK]D-Fender | k-man : Too super low-end. May as well get an ATA |
| 05:08.53 | k-man | [TK]D-Fender, except that I would then have to buy an analogue phone also |
| 05:08.57 | *** join/#asterisk InHisName (n=Administ@c-68-38-105-1.hsd1.pa.comcast.net) |
| 05:09.02 | k-man | as i don't have a spare analogue phone |
| 05:09.03 | [TK]D-Fender | Snake-Eyes : Think of your current CHANNEL. |
| 05:09.16 | [TK]D-Fender | k-man : And an analog phone would cost you HOW MUCH? |
| 05:09.33 | k-man | well... a halfway decent one would be $50 or so |
| 05:09.37 | JT | lol |
| 05:09.38 | JT | $50 |
| 05:09.52 | k-man | plus $66 for an ATA |
| 05:09.53 | JT | as if, second hand you can get an awesome one for $20 |
| 05:09.59 | k-man | jt, really? |
| 05:10.05 | JT | yeah ebay |
| 05:10.08 | k-man | jt, please tell me what brand to look for |
| 05:10.28 | JT | analogue phone brands? i dunno, good ones :P |
| 05:10.34 | JT | high end telstra stuff |
| 05:10.34 | k-man | see |
| 05:10.36 | JT | etc |
| 05:10.52 | k-man | i could just get the spa901 and only have to buy one thing |
| 05:10.59 | [TK]D-Fender | k-man : Go find a budget :) |
| 05:11.16 | JT | i got an Access 200 which has a full screen, for $20, i know telstra sold it for $200 |
| 05:11.47 | k-man | nice |
| 05:12.18 | [TK]D-Fender | a 10$ el-cheapo pharmacy phone + ATA would be better than an IP 901, plus more flexible, and you'd have another port on it to boot. |
| 05:12.36 | [TK]D-Fender | k-man : SPA-901 is SUB 10$ phone + ATA combined. |
| 05:12.46 | [TK]D-Fender | k-man : At least you can DO SOMETHING with the ATA |
| 05:12.59 | [TK]D-Fender | k-man : Oh.... and stop being such a cheap-ass |
| 05:13.01 | [TK]D-Fender | :) |
| 05:13.11 | sumasuma | [TK]D-Fender: aware of any GSM gateway ? |
| 05:14.22 | [TK]D-Fender | sumasuma : I'm aware they exist. Thats about all. |
| 05:14.39 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 05:15.05 | sumasuma | I checked in google nothing does with SIP, thought of taking one with IAX |
| 05:15.20 | k-man | sumasuma, i bought one from ebay, its a piece of crap, but it does work |
| 05:15.20 | k-man | sumasuma, its just an analogue output though |
| 05:15.46 | sumasuma | k-man: you mean GSM --- Ethernet Interface ? |
| 05:15.55 | k-man | [TK]D-Fender, ok, so which is the lowest end linksys you would get? |
| 05:16.16 | k-man | sumasuma, no, just an old analogue phone interface |
| 05:16.45 | k-man | and it does the american sounding dial tone and engaged signal which is fine except we are in australia |
| 05:16.55 | sumasuma | k-man: i c, i bought one with Ethernet interface $170+ |
| 05:17.03 | k-man | sumasuma, nice... |
| 05:17.16 | k-man | how do you use it? does it do sip or something? |
| 05:17.34 | [TK]D-Fender | k-man : What kind of call volume? |
| 05:17.44 | sumasuma | yes it has sip, but SIP is blocked by our ISP |
| 05:18.06 | [TK]D-Fender | sumasuma : So put * in front of it and go IAX2 |
| 05:18.25 | k-man | [TK]D-Fender, err... medium |
| 05:18.50 | [TK]D-Fender | k-man : Is this going to be your primary phone (personal)? |
| 05:18.58 | sumasuma | [TK]D-Fender: asterisk in a small box can ? |
| 05:19.08 | [TK]D-Fender | sumasuma : sure |
| 05:19.29 | k-man | [TK]D-Fender, no, work phone for an employee who will be working from home, we want to give her a sip account so she can make calls that are billed to us |
| 05:20.57 | k-man | [TK]D-Fender, maybe i should just get a spa941? |
| 05:21.57 | [TK]D-Fender | k-man : Screw it, get her an SPA-3102 and sh can plug it in-line with her home phone line & an existing phone. |
| 05:22.37 | [TK]D-Fender | k-man : Gets you an FXO later, and her and FXS you can use for her primarily. that way she only has to have 1 phone at her desk 9the one she already has) |
| 05:23.01 | k-man | hmm |
| 05:23.13 | k-man | ok |
| 05:23.20 | k-man | tahnks for the advice |
| 05:23.45 | [TK]D-Fender | k-man : I'll simplify her life & yours |
| 05:24.06 | k-man | yeah |
| 05:24.09 | k-man | your probably right |
| 05:24.09 | *** join/#asterisk kavit (n=kavit@eth2238.nsw.adsl.internode.on.net) |
| 05:25.14 | kavit | does any one know what the Alert-Info header for Linksys SPA-942 should be set to for custom ringing information? |
| 05:25.26 | k-man | [TK]D-Fender, thanks |
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| 05:34.02 | *** mode/#asterisk [+o russellb] by ChanServ |
| 05:40.19 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqkq.cable.mindspring.com) |
| 05:43.13 | InHisName | who has succeeded in connectiing icall.com into * ? |
| 05:49.59 | *** part/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 05:54.55 | *** join/#asterisk ManxPower (n=manxpowe@117.sub-75-200-33.myvzw.com) |
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| 06:05.16 | InHisName | amazingly quiet last 1/2 hour, is it just me and 287 robots or ppl asleep at the terminal ? |
| 06:05.52 | zamba | i'm here :) |
| 06:05.52 | Snake-Eyes | sleep! its 4 in the afternoon here |
| 06:06.05 | JT | it's 5pm here |
| 06:06.09 | InHisName | 1:08am here |
| 06:06.17 | Snake-Eyes | then again afternoon snooze :P |
| 06:06.32 | InHisName | Some places have Siesta. |
| 06:06.43 | linagee | why can i not get switch => going to one of my ivr's? :( |
| 06:06.47 | florz | InHisName: You are in a weird time zone =:-) |
| 06:06.50 | florz | 07:06 < InHisName> 1:08am here |
| 06:06.52 | linagee | i don't understand this. :( |
| 06:07.05 | InHisName | who has succeeded in connectiing icall.com into * ? |
| 06:08.06 | zamba | 7 am here |
| 06:08.24 | sumasuma | zamba: so what ? |
| 06:08.29 | JT | linagee: :( |
| 06:08.47 | zamba | sumasuma: huh? |
| 06:08.57 | sumasuma | it is 02:08 P.M here |
| 06:08.58 | JT | sumasuma: we're all discussing timezones and channel activity, no need to be rude |
| 06:09.06 | sumasuma | JT: i c |
| 06:09.42 | InHisName | linagee: I know ivr but what exactly is switch => mean ? |
| 06:10.10 | sumasuma | InHisName: switch is connecting two points ! |
| 06:11.10 | ManxPower | InHisName: switch => option in extension.conf is used for looking up extensions on remote Asterisk servers. |
| 06:11.14 | InHisName | linagee like connecting up two * together ? ( or rather two extens from two * ) |
| 06:11.18 | ManxPower | unless you are using AEL, of course. |
| 06:11.33 | linagee | InHisName: yes |
| 06:11.50 | linagee | InHisName: i want to call up the ivr on another asterisk box |
| 06:12.01 | linagee | InHisName: calling extensions works great with switch => |
| 06:12.51 | linagee | InHisName: i've tried dialing 7777, but i guess that's a local thing..... |
| 06:12.59 | ManxPower | Generally DUNDi is designed to do what switch => does. I would not be surprised if it was remove in a future release of Asterisk |
| 06:13.02 | InHisName | linagee does the other * have its ivr on an extension that you can access ? Mine just got fixed two days ago. |
| 06:13.21 | linagee | InHisName: i don't think it does. it has it at a context... |
| 06:13.37 | linagee | InHisName: i've tried to make an extension that goes to that context, but it didn't work either |
| 06:13.47 | linagee | errr. |
| 06:13.50 | linagee | did a exten => to it |
| 06:13.56 | linagee | exten => Goto() |
| 06:14.05 | InHisName | linagee Mine was that way too, until I added an exten with goto the ivr context. |
| 06:14.10 | linagee | hrm... |
| 06:14.34 | linagee | aha! maybe i'm not including it from the parent context!! :) |
| 06:14.35 | linagee | nice |
| 06:14.37 | linagee | ok |
| 06:14.40 | *** join/#asterisk zeeesh (i=zeeesh@202.38.55.125) |
| 06:14.42 | zeeesh | hi |
| 06:17.22 | InHisName | any Stanaphone users around ? |
| 06:18.39 | CrashHD | anyone have issues compile 1.2.15? |
| 06:18.43 | CrashHD | codec_zap is failing |
| 06:18.47 | CrashHD | to be compiled |
| 06:19.24 | InHisName | Sunrocket.com calls my stanaphone.com account and I get their vm. Another sip account calls and it rings through to my *. Can any one explain why? |
| 06:24.32 | linagee | i can dial extensions using switch =>, but not ring groups, ivrs, :( |
| 06:25.22 | ManxPower | CrashHD: works for me. |
| 06:25.31 | ManxPower | linagee: as long as they are an extension it should work. |
| 06:25.44 | ManxPower | and IVR is just an extension |
| 06:27.51 | linagee | got it working! :) |
| 06:27.53 | linagee | woo |
| 06:28.00 | linagee | i had to add it to a follow me. LOL |
| 06:28.02 | linagee | oh well. ;) |
| 06:28.07 | linagee | it work! woohoo! :) |
| 06:28.56 | InHisName | linagee wonderful, but what is the follow me part ? |
| 06:31.51 | [TK]D-Fender | Screams "FreePBX" to me... |
| 06:37.00 | JT | Connection reset by FreePBX |
| 06:38.02 | sumasuma | Is there is any PHP library for asterisk AGI ? |
| 06:38.35 | [TK]D-Fender | sumasuma PHP-AGI yes. look it up on the WIKI |
| 06:38.44 | [TK]D-Fender | well... way too late here.. I'm off |
| 06:38.47 | [TK]D-Fender | Good luck all. |
| 06:39.23 | sumasuma | http://phpagi.sourceforge.net/ |
| 06:39.26 | sumasuma | looks like it is too old |
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| 06:39.44 | sumasuma | is there is anything new out there for php version 5+ |
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| 06:57.52 | hellojoe | Hi guys, I am recording a file using the Record command and I want to mail the recording. However, the user hangs up the call after recording and the dialplan executes Hangup. Is there a way to force executing a few instructions after the Hangup???? |
| 06:58.09 | hellojoe | i am sorry, before the Hangup |
| 06:58.43 | sumasuma | hellojoe: I guess voicemail should work ! |
| 06:59.11 | hellojoe | Sumasuma: I am trying to implement a simple scheme for voicemail to email |
| 06:59.33 | hellojoe | full blows voicemail requires realtime setup etc if I know the email |
| 06:59.38 | hellojoe | full blown |
| 06:59.53 | hellojoe | In my dialplan, I have the email address of the called party |
| 07:00.02 | hellojoe | I wanted the caller to record the message and ship it out |
| 07:00.05 | hellojoe | that's all. |
| 07:00.17 | hellojoe | Wish there was a way to specify email in the Voicemail application |
| 07:02.34 | *** part/#asterisk zeeesh (i=zeeesh@202.38.55.125) |
| 07:03.12 | sumasuma | hellojoe: yes, there is a way o specify to which email address it need to send |
| 07:03.27 | hellojoe | really...!! Please suggest how? |
| 07:04.03 | hellojoe | sumasuma, I have dynamic users coming and going so there is a no fixed mailbox for each user |
| 07:04.35 | hellojoe | I can not set a context for each user. In my dialplan, I have called party's email address and that's all |
| 07:04.57 | sumasuma | i c |
| 07:05.00 | hellojoe | is there a way for the caller to record and vmail and for us to ship out the .wav attachment to the called-party's emails address |
| 07:05.03 | *** join/#asterisk aadilismail (i=zeeesh@202.38.55.125) |
| 07:05.04 | aadilismail | hi |
| 07:05.37 | sumasuma | hellojoe: possible with agi |
| 07:05.43 | hellojoe | so i need a way to (force) execute a few instructions after the recordings. |
| 07:05.54 | hellojoe | before the hangup |
| 07:06.08 | hellojoe | yeah.. AGI may work. |
| 07:06.21 | hellojoe | Unless there is a simple way to force execute a few instrucitons |
| 07:06.31 | hellojoe | i might add some stuff to the custom hangup |
| 07:06.34 | hellojoe | let me try something |
| 07:06.39 | sumasuma | ok |
| 07:06.44 | *** join/#asterisk aadilismail (i=zeeesh@202.38.55.125) |
| 07:06.46 | aadilismail | hi |
| 07:06.54 | sumasuma | i don't think the record will do record and send email |
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| 07:22.27 | Fr0zen_ | when you run a pbx at your home and also have a voip provider. Are you just running your calls thru your pbx and from your pbx to their out to pots lines? is that how it works in an over simplified form. |
| 07:24.17 | *** join/#asterisk angryuser (n=Miranda@i03v-213-44-169-43.d4.club-internet.fr) |
| 07:24.34 | angryuser | good morning/day/night |
| 07:25.08 | angryuser | where the delay of "Auto congestion" is set? |
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| 07:33.30 | *** mode/#asterisk [+o denon] by ChanServ |
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| 07:51.17 | JT | Fr0zen_: to put it simply, there are many ways you can set it up, but if you have a voip provider, asterisk can connect to it, and at the voip provider, they have digital PRI lines to the PSTN, not POTS |
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| 07:56.14 | Snake-Eyes | just sent * into loop by put ! in the wrong places |
| 07:56.20 | JT | hah |
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| 08:01.50 | angryuser | Snake-Eyes: i remember putting on loop by Goto() badly set;) |
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| 08:11.04 | Snake-Eyes | well as it turns out the ! was not to blame, damn goto (added line, so when some where else went to that line .....) |
| 08:16.02 | *** join/#asterisk active_si (n=AcTiVe@84.52.176.86) |
| 08:17.41 | active_si | hi all! I have a question, I am a new asterisk user, and I have managed a setup with two HFC-S cards TE/NT mode and everything works great, must say this is an astonishing product, now I have one question, is it possible to add a GSM modem connected on COM port of the computer so the users could make voice calls out to the GSM network using this modem? |
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| 08:41.59 | JT | active_si: you cannot make voice calls with a gsm modem. |
| 08:44.05 | active_si | JT: even if it supports voice calls? |
| 08:44.24 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 08:45.53 | JT | it's not really acting as a modem then |
| 08:46.33 | JT | the device must either talk a voip protocol over tcp/ip, or it must be a piece of hardware specifically supported by asterisk |
| 08:46.58 | JT | i'm pretty sure i haven't heard of anything called a "gsm modem" that can talk to asterisk in a voip capacity |
| 08:47.03 | JT | s/voip/voice/ |
| 08:47.13 | JT | you can get gsm pci cards for asterisk |
| 08:47.17 | JT | they are not cheao |
| 08:47.18 | JT | cheap |
| 08:47.40 | JT | you can also get gsm gateways, that either talk POTS (analogue - undesirable) or SIP |
| 08:47.45 | JT | they're not cheap either |
| 08:48.21 | JT | there's also a new bluetooth driver called chan_cellphone, that's a pretty cheap option, but requires a bluetooth mobile phone in proximity and a blue tooth dongle on the asterisk server |
| 08:48.39 | denon | JT: cheap is relative - people used to whine about the cost of X100Ps |
| 08:48.51 | active_si | so unless there is a drivers that would support Siemens TC35 there is no chance I can use it. |
| 08:48.59 | JT | yeah, clowns, denon :) |
| 08:49.13 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:49.18 | denon | all the world's a stage |
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| 09:04.40 | Strom_C | holy catsex |
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| 09:14.09 | kippi | what is the best way just to reload the extenstion config? |
| 09:14.25 | Strom_C | extensions reload |
| 09:19.47 | kippi | is there a easy way to check if a string has been set? I am using SET to setup night service but it keeps on going to voicemail, can I check that string to make sure its setting it? |
| 09:22.50 | JT | kippi: Verbose() or NoOp() |
| 09:23.05 | Strom_C | kippi: you can use gotoiftime() to set time-dependent night service |
| 09:23.17 | Strom_C | also, persistent values should probably be stored in the astdb |
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| 09:25.17 | kippi | it needs to be user set |
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| 09:34.59 | kippi | so this should work? exten => 5010,1,Set(DB(family/key)=${0}) |
| 09:35.02 | angryuser | is there any way to change a delay of auto congestion (if peer is not responding it take 5-6 sec) i need it to set to 2 sec |
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| 09:46.52 | ik_5-w | how can i know the time period that the QueueStatus's "Completed" value is pointed to ? |
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| 10:06.04 | kippi | is there away to get your ringtone for internal to be differnet to a outside call? |
| 10:08.28 | Strom_C | what do you mean by "ringtone" |
| 10:08.54 | kippi | on the handset when it rings, so the handset will have 2 differnet rings |
| 10:09.16 | Strom_C | handsets themselves ring? and here i thought it was the desk set that had the speaker in it :) |
| 10:09.24 | Strom_C | it's going to depend on your telephone set |
| 10:09.33 | Strom_C | what are you using? |
| 10:09.46 | MaartenB | hi everyone |
| 10:09.53 | Strom_C | dogballs |
| 10:09.55 | MaartenB | I keep getting an error in Asterisk: No private structure for packet? |
| 10:10.06 | MaartenB | anyone knows what it is caused by? |
| 10:11.30 | Strom_C | kippi: what kind of telephone set are you using? |
| 10:11.50 | kippi | grandstream |
| 10:11.56 | Strom_C | ugh |
| 10:12.03 | Strom_C | I have no experience with those :) |
| 10:12.30 | kippi | can I send a INVITE with Alert-Info header from the dial plan? |
| 10:14.19 | Strom_C | possibly |
| 10:16.34 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 10:17.58 | [hC] | how should i go about converting wav to g729? |
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| 10:27.21 | aadilismail | mysql |
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| 10:44.15 | phearless | what is the trick to pickup a ringing phone connected to the same asterisk ? |
| 10:44.19 | phearless | I forgot the trick.. |
| 10:45.17 | [hC] | *8 |
| 10:45.30 | [hC] | dunno if id call it a trick though |
| 10:45.36 | Strom_C | assuming, of course, that both phones are in the same pickup group |
| 10:45.42 | Strom_C | [hC]: I'd call it "Centrex" |
| 10:46.45 | phearless | seems to be "pickupexten" |
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| 10:53.19 | kippi | anyone used the grandstream phones? and ring tones? |
| 10:53.48 | Strom_C | long story short: I don't like grandstream phones |
| 10:54.24 | ThoMe | hallo |
| 10:54.25 | ThoMe | :-) |
| 10:54.28 | kippi | same as, I have got it working with one extenstion, but dont think you can have a range |
| 10:56.48 | *** part/#asterisk r0d3nt (n=RatMan@punk.valuetel.net) |
| 10:58.15 | ThoMe | kann hier wer deutsch? |
| 11:00.01 | Strom_C | ThoMe: nein. sprechen englisch? |
| 11:00.03 | angryuser | is it possible to use variable in MATH() function? |
| 11:00.18 | Strom_C | angryuser: sure, why not |
| 11:00.29 | angryuser | Strom_C: ok thank |
| 11:01.22 | *** join/#asterisk r0d3nt (n=RatMan@punk.valuetel.net) |
| 11:01.22 | ThoMe | Strom_C: ok. |
| 11:02.40 | ThoMe | cdr DB: how i can see which phone take of the call? |
| 11:03.28 | ThoMe | example: i have a incomming call: |
| 11:03.29 | ThoMe | exten => 14334610,n,Dial(Local/gruppe_alle@gruppen) |
| 11:03.49 | ThoMe | exten => gruppe_alle,1,Dial(SIP/82&SIP/11&SIP/21&MISDN/2/99&MISDN/3/99,30,rtT) |
| 11:04.16 | ThoMe | but i can't see in the CDR db where of this group has take of if i have a incoming call |
| 11:04.46 | Strom_C | ThoMe: instead of looping back into your dialplan with Local, just dial all those phones directly |
| 11:04.56 | Strom_C | or use a Goto() |
| 11:05.31 | angryuser | is there any way ti reduce the delay of auto congestion on sip peers? (currently 5-6 sec) |
| 11:05.37 | *** join/#asterisk merbzt (n=banan@136.240.13.217.in-addr.dgcsystems.net) |
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| 11:08.45 | ThoMe | Strom_C: hmm. |
| 11:09.22 | ThoMe | Strom_M: synchronistically dialing?? how? |
| 11:09.37 | phearless | I did put "*8" in the phone Dialplan, but when I press *8 it does not autodial |
| 11:09.41 | phearless | is it normal? |
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| 11:12.56 | ThoMe | Strom_M: ? |
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| 11:18.35 | r0d3nt | neat. |
| 11:25.29 | *** join/#asterisk tutt9876 (n=tut123@cvl92-2-82-228-144-230.fbx.proxad.net) |
| 11:25.32 | tutt9876 | hi |
| 11:26.11 | tutt9876 | some know why i have an opened port called 'callbook' on an asterisk server? |
| 11:26.53 | tutt9876 | someone |
| 11:26.53 | tutt9876 | knows |
| 11:27.48 | JT | tutt9876: what is the port number? |
| 11:28.03 | *** join/#asterisk lorinc (n=ang@217.20.136.141) |
| 11:28.30 | tutt9876 | 2000 |
| 11:28.40 | JT | angryuser: qualify=yes |
| 11:28.51 | tutt9876 | and it looks like I can telnet on this port? |
| 11:28.51 | Nugget | telnet is eeeeeeevil! |
| 11:29.25 | JT | tutt9876: use lsof to determine what is opening the port |
| 11:29.33 | tutt9876 | asterisk |
| 11:29.41 | JT | well there you go |
| 11:29.54 | JT | so it can only be one of a few things |
| 11:30.01 | JT | manager interface, sip, iax |
| 11:30.15 | JT | not iax if it's tcp |
| 11:30.33 | tutt9876 | it's not manager: can i disable it? |
| 11:31.14 | JT | it's not manager? |
| 11:31.40 | tutt9876 | no I launched manager on another port and i failed to log on this port |
| 11:32.00 | JT | well you have to know what it is before you can disable it |
| 11:32.16 | tutt9876 | you said it's sip? |
| 11:32.36 | JT | no, i did not |
| 11:32.43 | JT | i listed possible things |
| 11:33.03 | JT | anyway a quick google has revealed to me it's sccp/chan skinny |
| 11:33.15 | JT | you would've found that in 2 seconds had you googled :) |
| 11:33.45 | tutt9876 | not i found nothing on this port: what did you look for? |
| 11:33.51 | kippi | ok, I have internal numbers, is there away to set the CID for internal calls so like 1000 |
| 11:33.59 | angryuser | JT: i cant use qualify=yes because of strange sip * qualify bug, i have sip peers unreachable after 30 min or so, but in reality they are reachable |
| 11:34.34 | JT | tutt9876: asterisk port 2000 |
| 11:34.34 | tutt9876 | JT: what did you look for asterisk 2000 opened port on google? |
| 11:34.39 | JT | is what i googles for |
| 11:34.44 | JT | googled |
| 11:34.55 | JT | angryuser: i see |
| 11:35.46 | angryuser | JT: so i need to reduce auto congestion to 2 sec;) |
| 11:35.53 | JT | qualify working |
| 11:36.24 | JT | hrm |
| 11:36.33 | JT | maybe there's a timeout parameter |
| 11:36.49 | tutt9876 | JT: any idea of what is Sccp? |
| 11:37.24 | JT | yes, it's the cisco skinny protocol |
| 11:37.42 | tutt9876 | so no use of it? |
| 11:37.56 | JT | i'm guessing if you don't know what it is, then no |
| 11:38.25 | tutt9876 | i have no cisco device: so no use of it? |
| 11:38.50 | angryuser | JT: gm, not for me then, i have postes this on forums |
| 11:39.20 | JT | gm? |
| 11:39.28 | JT | tutt9876: didn't i really just answer that? |
| 11:40.06 | tutt9876 | Yes , but be nice just confirm? |
| 11:40.21 | angryuser | http://forums.digium.com/viewtopic.php?t=13727 here it is JT |
| 11:43.23 | tutt9876 | so any use of cisco skinny protocol? |
| 11:45.23 | *** join/#asterisk coppice (n=chatzill@13.168.17.210.dyn.pacific.net.hk) |
| 11:46.40 | phearless | I did put "*8" in the phone Dialplan, but when I press *8 it does not autodial |
| 11:46.41 | phearless | is it normal? |
| 11:46.53 | phearless | * + something is always strange |
| 11:50.00 | Strom_C | maybe because *8 conflicts with vertical service codes *80-*89 |
| 11:51.38 | phearless | ok :) |
| 11:56.10 | *** join/#asterisk kaldemar (n=kalde@vipunen.hut.fi) |
| 11:57.21 | tutt9876 | any idea o nthe way to disable skinny/sccp protocol? |
| 11:57.40 | Strom_C | put noload statements in modules.conf |
| 12:04.11 | tutt9876 | ok it works: thanks to everyone, special thalns to JT |
| 12:04.15 | tutt9876 | thanks |
| 12:04.16 | *** part/#asterisk tutt9876 (n=tut123@cvl92-2-82-228-144-230.fbx.proxad.net) |
| 12:10.26 | kezza491 | i think i have set up asterisk but i dont know what i need to do after seting it up do i need a GUI for picking up the calls? |
| 12:14.09 | *** join/#asterisk UlbabraB (n=salama@host241-43-static.72-81-b.business.telecomitalia.it) |
| 12:19.26 | *** join/#asterisk saftsack (n=oliver@pD9E068AA.dip.t-dialin.net) |
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| 12:24.45 | xheliox | Anyone use Teliax? And if so, did they change their incoming caller ID format? |
| 12:30.08 | ThoMe | Strom_C: have u a answer of my question? |
| 12:31.25 | ThoMe | 12:04:40 < Strom_C> ThoMe: instead of looping back into your dialplan with Local, just dial all those phones directly |
| 12:32.23 | ThoMe | Strom_C: how i can multiple ringing phones, at the same time? without group? |
| 12:44.02 | Druken | ThoMe: it's a wonderful thing called & :) |
| 12:44.18 | Druken | dial(SIP/1&SIP/2,,,) |
| 12:45.04 | JT | ThoMe: google asterisk cmd dial |
| 12:45.05 | JT | :) |
| 12:45.28 | kippi | can any give me a tip of why exten => 650137,1,SIPAddHeader(Alert-Info: something) dosn't seem to work with my grandstream? |
| 12:45.32 | Druken | morning JT |
| 12:46.50 | ThoMe | Druken / jt: identic with: |
| 12:46.52 | *** part/#asterisk kezza491 (n=opera@c211-28-159-50.brasd1.vic.optusnet.com.au) |
| 12:47.02 | ThoMe | exten => 14334610,n,Dial(Local/gruppe_alle@gruppen) |
| 12:47.05 | ThoMe | exten => gruppe_alle,1,Dial(SIP/82&SIP/11&SIP/21&MISDN/2/99&MISDN/3/99,30,rtT) |
| 12:47.05 | ThoMe | ? |
| 12:48.03 | Druken | icky... a dial local... |
| 12:48.28 | ThoMe | icky ? |
| 12:48.38 | ThoMe | ah |
| 12:48.42 | ThoMe | "eklig" (german= |
| 12:48.42 | ThoMe | ;) |
| 12:48.56 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 12:50.07 | ThoMe | Druken: jojo ;) |
| 12:52.00 | *** join/#asterisk macTijn (i=martijn@linda.net.insecure.nl) |
| 12:56.20 | ThoMe | Druken: why not local? |
| 12:58.17 | *** join/#asterisk friedrich| (n=friedric@e177250126.adsl.alicedsl.de) |
| 12:58.33 | ThoMe | mal nen deutscher.. symatisch, tag friedrich ;) |
| 12:58.35 | ThoMe | +p |
| 12:58.44 | friedrich| | Hallo ThoMe :) |
| 12:59.04 | friedrich| | Du kannst mir gleich helfen :P |
| 12:59.36 | ThoMe | Hab ich nicht genug wissen und schon garkeine Lust. :-) |
| 12:59.39 | ThoMe | Was gibts denn? |
| 13:00.54 | friedrich| | Nichts.. :) |
| 13:01.23 | friedrich| | Muss ich mich mal wann anders auseinandersetzen mit. Idle hier bloß rum, um es nicht zu vergessen. |
| 13:01.45 | ThoMe | aha. ;) |
| 13:02.36 | kippi | can I change internal calls so that it does not show the phone number and says a name? |
| 13:02.45 | ThoMe | Druken: ? |
| 13:03.18 | Druken | ThoMe: just screws of cdr's... i use goto |
| 13:03.33 | *** join/#asterisk ahattar (i=ahattar@218.sub-75-193-180.myvzw.com) |
| 13:03.34 | ThoMe | Druken: do u have a example? |
| 13:03.48 | ThoMe | Druken: please. |
| 13:04.43 | ahattar | Question guys, does asterisk has an intercom capability? |
| 13:05.06 | Druken | exten => _NXXXXXX,1,goto(outgoing,705${EXTEN},1) |
| 13:06.40 | ThoMe | Druken: hm, no.. i mine if i have a incomming call to 1234567 then ringing, 11, 12, 13, 14 etc. interncal |
| 13:06.44 | ThoMe | -c |
| 13:08.39 | Druken | so it would be exten => 1234567,1,goto(gruppen,gruppe_alle,1) |
| 13:08.52 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 13:09.33 | ThoMe | gruppen, gruppen_alle <<two groups? |
| 13:10.17 | Druken | no... context,extension,priority |
| 13:11.27 | ahattar | Any idea, if asterisk has an intercom capability? |
| 13:11.52 | Druken | ahattar: it can be built in given the right phones... |
| 13:12.35 | ahattar | like wut phones? |
| 13:13.14 | Druken | i belive polycoms have the page function.... |
| 13:13.30 | ahattar | kool |
| 13:15.06 | ahattar | I have a client interested on asterisk (12 ext office), he has to sys, intercom and phones, i guess he will be happy if he can compined both system |
| 13:15.17 | ThoMe | Druken: ah, ok. sorry. I'm a idiot ;) |
| 13:15.45 | ThoMe | Druken: and gruppen_alle r |
| 13:15.46 | ThoMe | exten => gruppe_alle,1,Dial(SIP/82&SIP/11&SIP/21&MISDN/2/99&MISDN/3/99,30,rtT) |
| 13:15.46 | ThoMe | ? |
| 13:16.25 | ahattar | thnx Druken, bye for now |
| 13:22.55 | *** join/#asterisk queuetue (n=scott@70.54.254.134) |
| 13:24.20 | queuetue | Hi. Some time ago, someone sent me to a fantastic tool for measuring the real-time connection quality between two points - the cli app was called m-something-somthing... I'm sorry to be so wishy-washy, but I've lost the logs of the conversation - can anyone think of what I'm talking about? |
| 13:24.32 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 13:25.23 | queuetue | It was sort of a constant traceroute, top-style... |
| 13:25.39 | JT | sounds useful to have\ |
| 13:28.46 | ThoMe | Druken: hmmmm |
| 13:28.46 | active_si | I have one question, what is the most used solution for receiving/sending faxes, is it the app_rxfax/app_txfax or any other solution? |
| 13:28.46 | queuetue | It was http://www.bitwizard.nl/mtr/ I think. |
| 13:29.13 | angryuser | active_si:i am happy with hylafax |
| 13:29.28 | *** join/#asterisk `Sauron (i=sauron@dsl001-130-033.aus1.dsl.speakeasy.net) |
| 13:29.54 | *** join/#asterisk etfonhome_ (n=etfonhom@210.116.119.164) |
| 13:31.02 | ThoMe | Druken: |
| 13:31.03 | ThoMe | http://tm.muc.de/temp/a.txt |
| 13:31.14 | queuetue | It was mtr - great app. |
| 13:31.21 | ThoMe | Druken: this is a entry of a incomming call |
| 13:31.54 | ThoMe | Druken: i can't not see to which number has calling and i can't see which internal number has take of |
| 13:35.03 | *** join/#asterisk yonahw (n=yonahw@genie03-173-74.inter.net.il) |
| 13:35.35 | ThoMe | Druken: any ideas? |
| 13:35.43 | Druken | src is the caller id of the incoming call |
| 13:36.08 | Druken | and if you want to know which phone took each call, you need to use a queue |
| 13:37.57 | ThoMe | hmm. a queue? but how? please help me :-) |
| 13:38.43 | ThoMe | Druken: i have non erase the local call :-) and have your idea with goto implement |
| 13:38.47 | ThoMe | non = now |
| 13:39.12 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 13:40.08 | yonahw | I am trying to install zaptel drivers for a te110p on a trixbox installation. it seems that I have the source code for 2.6.9-34.0.2.EL-i686 while it is looking for 2.6.9-34.0.2.ELsmp. any ideas for a complete noob? |
| 13:40.57 | Druken | ThoMe: well, a queue is alot diffrent |
| 13:41.00 | ThoMe | ah. |
| 13:44.13 | *** join/#asterisk IPmonger (n=ipmonger@c-68-84-208-206.hsd1.pa.comcast.net) |
| 13:45.58 | mendol | could anybody help me with 2xnat sip problem? :-/ |
| 13:48.17 | *** join/#asterisk lilwookie (n=lilwooki@mendax.enter-net.com) |
| 13:49.10 | *** join/#asterisk axisys (i=vadud3@anapnea.net) |
| 13:59.50 | *** join/#asterisk n0n0x (n=n0n0x@201-212-168-53.net.prima.net.ar) |
| 14:00.04 | n0n0x | Hi, someone using auris as inbound provider? |
| 14:05.32 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 14:11.08 | *** join/#asterisk svanlund (n=dasv@slave.rixport80.se) |
| 14:12.06 | tzafrir | yonahw, look for kernel-devel-smp-VERSION |
| 14:12.45 | tzafrir | yonahw, trixbox does really ugly things with the kernel. I really don't recommend you to use it |
| 14:13.01 | mendol | can I use one asterisk installation for few diff bussiness clients? |
| 14:14.29 | mendol | i mean if its possible to add few groups each one with own trunk/extension settings? |
| 14:14.58 | Ahrimanes | mendol: yes, thats what context's are for |
| 14:15.39 | *** join/#asterisk lowlevel (n=Stuart@CPE00145176d140-CM000f9f1e356e.cpe.net.cable.rogers.com) |
| 14:17.03 | yonahw | thanks tzafrir |
| 14:18.01 | tzafrir | yonahw, get debian etch, apt-get install asterisk |
| 14:19.36 | yonahw | tzafrir, what are the advantages of getting debian etch? |
| 14:20.12 | mendol | but then i cant make same extension numbers for all of clients |
| 14:20.16 | tzafrir | debian etch is my favorite distro, so it has to be the best ;-) |
| 14:20.28 | mendol | each one of them has to get his own ext group? |
| 14:20.32 | tzafrir | mendol, include => |
| 14:21.26 | mendol | hmm |
| 14:21.44 | tzafrir | yonahw, well: well maintained, and with many packages in the main tree |
| 14:22.19 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 14:24.14 | yonahw | tzafrir how long would you imagine it takes to get debian, asterisk and freepbx installed? |
| 14:24.37 | tzafrir | we have freepbx debs in the works |
| 14:25.02 | tzafrir | asterisk works out of the apt-get |
| 14:25.27 | tzafrir | I hope to have freepbx working out-of-the-aptget within a few days |
| 14:25.59 | yonahw | does that mean that i can not currently install freepbx on a debian distro? |
| 14:27.16 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:27.18 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqkq.cable.mindspring.com) |
| 14:28.08 | *** join/#asterisk Modcuts (n=bob@lan.proporta.com) |
| 14:28.08 | eald | (I just means that installation of freepbx is like in any other distro when you install from tarball) |
| 14:28.27 | tzafrir | yonahw, not from official debs. You'll need the standard procedure |
| 14:28.30 | eald | ups not I, is It |
| 14:28.46 | mendol | sorry for my noob questions :-) tzafrir using include i can make same extensions for diff groups of users? |
| 14:29.33 | tzafrir | mendol, you can include the same context in several cotexts used by different users |
| 14:29.58 | *** join/#asterisk dhill (i=dhill@fog.mindcry.org) |
| 14:30.06 | mendol | sounds complicated but got the idea, thanks a lot :-) |
| 14:30.26 | dhill | oej around |
| 14:30.27 | dhill | ? |
| 14:30.35 | tzafrir | ~seen oej |
| 14:30.57 | jbot | oej <n=olle@apollo.webway.se> was last seen on IRC in channel #asterisk, 7d 19h 21m 15s ago, saying: 'cpatry: Then you have a misconfiguration or a bug'. |
| 14:30.57 | *** join/#asterisk [Airwolf] (n=airwolf@89.205.176.67) |
| 14:31.08 | dhill | doh :) |
| 14:31.18 | yonahw | tzafrir excuse my lack of knowledge but what is the difference between installing from debs and standard procedure |
| 14:32.17 | tzafrir | two differences: 1. the installation from a package is easier. If it is a well-made package it will actually save you time |
| 14:32.51 | tzafrir | 2. when you install it yourself you actually have to read some minimal documentation and gain a bit of knowledge |
| 14:32.59 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 14:34.58 | yonahw | tzafrir my main fear is that I do not think I currently have enough knowledge to get a full blown pbx up and working in a reasonable amount of time without the trixbox |
| 14:35.13 | *** join/#asterisk giasai68 (n=giasai@ip-240-130.sn2.eutelia.it) |
| 14:35.20 | tzafrir | sure you do |
| 14:35.46 | tzafrir | trixbox is actually over-complicated |
| 14:35.58 | giasai68 | hello i need help to fix this worning: [Feb 21 15:35:49] WARNING[20832]: pbx.c:2460 __ast_pbx_run: Timeout, but no rule 't' in context 'default' |
| 14:36.15 | yonahw | most of the trixbox is actually stuff i have no use for interestingly enough |
| 14:36.34 | tzafrir | if you want to understand how things work for you, you'll encounter a very complicated configuration |
| 14:37.08 | yonahw | my problem is that i am quickly finding out i am way over my head but do not want to back out at this point |
| 14:37.25 | yonahw | i think i am going to take your advice and see how it goes with debian |
| 14:37.38 | tzafrir | do you have any asterisk-related hardware? |
| 14:37.57 | [TK]D-Fender | Trixbox will teach you NOTHING. It is NOT a "launching pad" to learning *. It is a cop out that does not lead to anything except either utter surrender to it, or its removal. That is unless you are one fo the .01% that actually end up DEVELOPING Zit, but then again.... WGLWAT |
| 14:37.57 | *** join/#asterisk mavior (n=Miranda@88-149-162-164.f5.ngi.it) |
| 14:38.38 | yonahw | i have a digium te110p |
| 14:39.25 | kippi | should this set 0 exten => 5011,1,Set(TEST=0) in TEST |
| 14:39.51 | [TK]D-Fender | kippi: Yes |
| 14:40.18 | *** join/#asterisk JoNate (n=noone@mail.wmelec.com) |
| 14:40.29 | mavior | hello...got one question: i have the register--> comand in the iax.conf file registering to an no-ip server address...when it refresh its registration, it look-up every time and solve the no-ip address or just update and try to re-register to the first ip it resolved ? |
| 14:41.38 | yonahw | tzafrir, d-fender: you guys have me convinced I am starting the debian download now |
| 14:41.55 | giasai68 | hello i need help to fix this worning: WARNING[20832]: pbx.c:2460 __ast_pbx_run: Timeout, but no rule 't' in context 'default' |
| 14:41.56 | yonahw | tzafrir i truly appreciate your assistance |
| 14:42.05 | [TK]D-Fender | yonahw: Is Debian an OS you are relatively comfortable administering? |
| 14:43.03 | yonahw | i know nothing about debian |
| 14:43.10 | yonahw | i am extremely new to linux in general |
| 14:44.05 | kippi | [TK]D-Fender: getting this error |
| 14:44.06 | kippi | <PROTECTED> |
| 14:44.06 | kippi | <PROTECTED> |
| 14:44.21 | *** join/#asterisk synaptic (i=nite@nat/digium/x-28622d66319ba2d4) |
| 14:45.14 | [TK]D-Fender | yonahw: Ok well then go read up on what it takes to run Debian (the basics that make its experience different than other Distros), and make sure to install * from the SOURCE tarballs and not the Debian provided packages. |
| 14:45.48 | [TK]D-Fender | kippi: That isn't an error. thats just saying that you have no priority 2 for that exten and execution is stopping. |
| 14:46.01 | yonahw | d-fender: what are the advantages of installing from the source and not the packages? |
| 14:46.19 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 14:46.19 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:46.20 | [TK]D-Fender | giasai68: Go read up on *'s Standard Extensions. |
| 14:46.34 | ManxPower | mavior: Asterisk only tries to resolve hosts on the first register, as far as I know. |
| 14:47.21 | [TK]D-Fender | yonahw: Packaged versions are often (especially with Debian) outdated, and you lose the ability to customize the experience. The install instructions are very simple and straight forward as far as fdoing it all by source is concerned. You should have no issue with this. |
| 14:47.35 | mavior | weird! my ip could change and asterisk will try to register to the wrong ip |
| 14:47.50 | kippi | [TK]D-Fender: so that means it has taken that, so there will be 0 sitting in TEST |
| 14:47.53 | ManxPower | mavior: Asterisk's support for DNS sucks. |
| 14:48.00 | ThoMe | Feb 21 15:47:32 WARNING[24979]: chan_agent.c:1849 __login_exec: Extension '51' is not valid for automatic login of agent '1511' |
| 14:48.03 | ThoMe | how? |
| 14:48.05 | mavior | i just noticed this behaviour |
| 14:48.05 | yonahw | i dont have a problem with the * source install i tried it once and it wasnt that difficult im just wondering why not use the packages |
| 14:48.33 | mavior | Manx, do you know where in the source this could be modified? this is really really weird! |
| 14:48.38 | [TK]D-Fender | kippi: No. Variable only survive as long as your channel does. as soon as you fall through *poof* |
| 14:48.43 | ManxPower | yonahw: the ONLY software I build myself is Asterisk, ClamAV, and Courier IMAP |
| 14:48.57 | kippi | so then i need to use a AstDB? |
| 14:49.06 | [TK]D-Fender | ThoMe: Have you confirmed that its VALID? |
| 14:49.16 | ManxPower | mavior: if it was easy to fix it would have been fixed years ago. search the mailing list archives for more information |
| 14:49.17 | [TK]D-Fender | kippi: Yes, or some other persistant source |
| 14:49.19 | ManxPower | ~mailinglist |
| 14:49.20 | jbot | Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.htmm |
| 14:49.26 | ThoMe | [TK]D-Fender: emm, why not valid? |
| 14:49.40 | ThoMe | [TK]D-Fender: in the queues.conf i have member => Agent/1511 |
| 14:49.51 | ThoMe | [TK]D-Fender: and in agents.conf agent => 1511,4494950,Thomas Mebes |
| 14:49.52 | *** join/#asterisk anthm (n=anthm@64.241.37.140) |
| 14:49.52 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:49.59 | giasai68 | hello i need help to fix this worning: [Feb 21 15:35:49] WARNING[20832]: pbx.c:2460 __ast_pbx_run: Timeout, but no rule 't' in context 'default' |
| 14:50.11 | [TK]D-Fender | ThoMe: Irrelevent. it said the EXTENSION is not valid. Go check your dialplan. |
| 14:50.27 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 14:50.31 | [TK]D-Fender | giasai68: I've already answered you on this. Go back and read the chapter on *'s Standard Extensions. |
| 14:50.33 | [TK]D-Fender | ~book |
| 14:50.34 | jbot | from memory, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 14:50.36 | *** join/#asterisk vgster (n=vgster@h14690.navonline.net) |
| 14:50.38 | ManxPower | giasai68: create an exten => t,1,Noop(Timeout on expected DTMF input) |
| 14:50.51 | ThoMe | [TK]D-Fender: hm. |
| 14:51.02 | giasai68 | ok i will try to do this thank yu |
| 14:51.05 | giasai68 | ou |
| 14:51.06 | *** join/#asterisk sivana (n=sivana@208.68.91.47) |
| 14:51.14 | ThoMe | [TK]D-Fender: i have http://tm.muc.de/temp/a.txt |
| 14:51.51 | [TK]D-Fender | ThoMe: THAT is not the problem. Its the extension where you LOG IN |
| 14:52.29 | [TK]D-Fender | ThoMe: A clue : look for where the '51' came from.... |
| 14:52.32 | ThoMe | [TK]D-Fender: refresh a.txt |
| 14:52.34 | ThoMe | http://tm.muc.de/temp/a.txt |
| 14:52.45 | *** join/#asterisk fr33bi3 (n=fr33bi3@59.144.4.55) |
| 14:53.09 | [TK]D-Fender | ThoMe: NO. thats just a device description. it has to LOG IN to the queue. |
| 14:53.32 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 14:53.37 | ThoMe | [TK]D-Fender: ok. u mine queues.conf ? |
| 14:53.39 | [TK]D-Fender | ThoMe: the problem is in your dialplan. |
| 14:53.53 | [TK]D-Fender | ThoMe: I've said it TWICE now. |
| 14:54.22 | *** join/#asterisk Cyon (n=cyon@216.179.31.170) |
| 14:54.26 | ThoMe | [TK]D-Fender: ok. in the extensions.conf yes? |
| 14:54.42 | [TK]D-Fender | ThoMe: Yes. |
| 14:55.11 | giasai68 | <PROTECTED> |
| 14:55.11 | giasai68 | <PROTECTED> |
| 14:55.31 | *** join/#asterisk m0nst3r (n=kvirc@c-71-206-116-36.hsd1.mi.comcast.net) |
| 14:55.36 | ThoMe | [TK]D-Fender: u mean: exten=> 2011,1,AgentCallbackLogin(||${CALLERIDNUM}@queues) |
| 14:55.39 | ThoMe | <PROTECTED> |
| 14:55.46 | giasai68 | i got this but i think that the call not work rigth? |
| 14:55.49 | ManxPower | giasai68: good. It is working,. |
| 14:56.18 | ManxPower | giasai68: WHAT specifically is not working? |
| 14:57.16 | giasai68 | i see that the number not dialed from my gateway and the call was destroy |
| 14:57.59 | ManxPower | giasai68: I cannot help you with H323 |
| 14:58.44 | giasai68 | ah.. ok.... other suggest? |
| 14:58.53 | ManxPower | giasai68: don't use H323 |
| 14:58.53 | [TK]D-Fender | ThoMe: pastebin your entire diaplan please. |
| 14:59.23 | ManxPower | giasai68: very few people use H323 with Asterisk because it is very difficult to correctly configure. |
| 14:59.52 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-221.ph.ph.cox.net) |
| 15:00.10 | [TK]D-Fender | ManxPower: Funy thing is we don't even have proof he's trying to run an IVR.... |
| 15:00.11 | giasai68 | any suggest for read about ooh323? |
| 15:00.22 | [TK]D-Fender | ~pb |
| 15:00.23 | jbot | extra, extra, read all about it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 15:00.29 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 15:00.33 | ThoMe | [TK]D-Fender: http://tm.muc.de/temp/a.txt |
| 15:00.42 | ManxPower | giasai68: the mailing list archives |
| 15:00.49 | ThoMe | [TK]D-Fender: refresh http://tm.muc.de/temp/a.txt |
| 15:01.03 | *** join/#asterisk mercestes (n=merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 15:01.27 | [TK]D-Fender | ThoMe: See this - exten=> 2011,1,AgentCallbackLogin(||${CALLERIDNUM}@queues|) ? |
| 15:01.40 | ThoMe | [TK]D-Fender: is it wrong? |
| 15:01.54 | angryuser | so noone know how to reduce the autocongestion delay when dialing out with non qualifyed sip peer? |
| 15:01.59 | ThoMe | [TK]D-Fender: i have this of a example... google ;) |
| 15:02.15 | [TK]D-Fender | ThoMe: Your agent is logging in with it using his CallerID as the extension, and looking for it ('51') in context [queues]. there IS no "exten => 51,1,whatever" in there. |
| 15:02.33 | ThoMe | hm. ok |
| 15:02.36 | [TK]D-Fender | ThoMe: Stop trying to cut& paste code off the web, and go read the book, then go read the WIKI. |
| 15:03.00 | ThoMe | [TK]D-Fender: yes ok. is this korrekt?: AgentCallbackLogin(${CALLERIDNUM}) ? |
| 15:03.22 | [TK]D-Fender | ThoMe: You need to specify an extension & context. Make sure it is VALID |
| 15:03.33 | mercestes | Good morning, Fender. |
| 15:03.44 | angryuser | ThoMe:if you use aster 1.4 ${CALLERIDNUM} is deprecated its ${CALLERID(num)} |
| 15:03.45 | *** join/#asterisk GaVak (n=denniso@adsl-074-228-124-003.sip.sav.bellsouth.net) |
| 15:03.48 | [TK]D-Fender | mercestes: Really? Less than stellar so far... |
| 15:03.59 | ThoMe | i have 1.2 |
| 15:04.00 | *** join/#asterisk lupino3 (n=lupino3@217-133-45-108.b2b.tiscali.it) |
| 15:04.06 | lupino3 | hello guys |
| 15:04.10 | giasai68 | another think i got another warning : [Feb 21 16:02:19] WARNING[22591]: src/chan_h323.c:977 ooh323_indicate: Don't know how to indicate condition -1 on ooh323c_4187 |
| 15:04.14 | [TK]D-Fender | angryuser: You'll want to leave this one.... he's on 1.2- and has REAL dialplan issues |
| 15:04.18 | lupino3 | I think that I've found a bug in Asterisk, and I'd like to debug it |
| 15:04.20 | giasai68 | hope you can help me |
| 15:04.24 | lupino3 | have you got some indications for me? |
| 15:04.48 | ManxPower | lupino3: README.backtrace in the asterisk source docs directory |
| 15:04.52 | [TK]D-Fender | ManxPower: No thanks. I'm developing intolerance to all sorts of drug, and now foods I used to eat. New diet is causing a major shift... |
| 15:04.59 | lupino3 | ManxPower, thaaanks :) |
| 15:05.09 | mercestes | Fender: I was being optimistic..=/ |
| 15:05.46 | [TK]D-Fender | mercestes: Well... rhetorically speaking : good morning. Sorry for being a downer. |
| 15:06.27 | mercestes | Fender: no, please. Abuse me all you like. I kinda like it. |
| 15:06.32 | mercestes | :) |
| 15:08.45 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 15:09.02 | iCEBrkr | That is all.. |
| 15:09.12 | ThoMe | [TK]D-Fender: i give up. thx! |
| 15:09.27 | angryuser | ThoMe: really read wiki |
| 15:09.29 | giasai68 | <giasai68> another think i got another warning : [Feb 21 16:02:19] WARNING[22591]: src/chan_h323.c:977 ooh323_indicate: Don't know how to indicate condition -1 on ooh323c_4187 |
| 15:09.30 | [TK]D-Fender | mercestes: Yeah I know we've enjoyed our bit of verbal sparring, and I would like to think its in the interest of humour, personal development/correctio, etc, but I know I can be over-persistant on it and don't want to make any bad long-term habits |
| 15:09.39 | ThoMe | angryuser: hm. ok. |
| 15:10.00 | angryuser | ThoMe: dialplan application |
| 15:10.02 | iCEBrkr | [TK]D-Fender: Over persistant, that's an understatement :-D |
| 15:10.24 | mercestes | Aww, I love you too, Fender. :D |
| 15:10.26 | [TK]D-Fender | ThoMe: Do you understand what the login application is looking for? You tell it what exten in your dialplan to call when that agent is called (if they log in properly). |
| 15:10.57 | *** join/#asterisk Crescendo_ (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 15:11.01 | mercestes | hey! that reminds me. Ppl forwarding voicemails, it gives an email notification listing a "zero duration" voicemail when it's not really zero duration. can that be fixed or should I just remove duration from the email template? |
| 15:11.17 | kippi | Has anyone got a good howto on AstDB ? need to be able to put somthing in the database, and then get it back and if ture need to Goto |
| 15:11.21 | [TK]D-Fender | ThoMe: That means that whatever you enter on that line has to exist. you might make a context called [queue] in there and add a line like "exten => 51,1,Dial(IAX2/51)" in it |
| 15:11.49 | mercestes | kippi: the wiki has some astDB stuff on it. |
| 15:11.51 | [TK]D-Fender | kippi: "show fucntion DB" , "show application dbdel" |
| 15:12.02 | [TK]D-Fender | kippi: "show function DB" , "show application dbdel" |
| 15:12.22 | [TK]D-Fender | kippi: an look up "asterisk functions" on the WIKI. |
| 15:12.29 | *** join/#asterisk fr33bi3 (n=fr33bi3@125.22.65.165) |
| 15:12.41 | *** join/#asterisk l3chuck_ (n=lk@62-101-102-100.ip.fastwebnet.it) |
| 15:12.42 | angryuser | core show function DB in aster 1.4;) |
| 15:12.53 | lupino3 | ManxPower, is it normal that asterisk doesn't leave a corefile even if it's called with the -g option? |
| 15:13.06 | lupino3 | there isn't any core file in /tmp |
| 15:13.15 | iCEBrkr | kippi: exten => s,1,Set(DB(callerid/number)=${ARG1}) |
| 15:13.37 | ManxPower | lupino3: is asterisk deg faulting? |
| 15:13.43 | ManxPower | or seg faulting |
| 15:13.55 | lupino3 | don't know.. to my eyes, it simply crashes |
| 15:14.12 | [TK]D-Fender | ThoMe: Or change your login like this - exten=> 2011,1,AgentCallbackLogin(||${CALLERIDNUM}@intern|) |
| 15:14.30 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 15:14.34 | [TK]D-Fender | ThoMe: That way it will use the already defined context that seems to have the dial statement for your internal extensions. |
| 15:15.16 | [TK]D-Fender | iCEBrkr: y0, long time..... |
| 15:15.47 | ThoMe | tzafrir: i have it but: Feb 21 16:15:26 WARNING[26064]: chan_agent.c:1849 __login_exec: Extension '51' is not valid for automatic login of agent '51' |
| 15:16.07 | lupino3 | it happens when I try to check voicemail from external calls that come in from BRI ISDN, through mISDN drivers |
| 15:16.40 | ThoMe | <PROTECTED> |
| 15:16.40 | ThoMe | <PROTECTED> |
| 15:16.40 | ThoMe | <PROTECTED> |
| 15:16.42 | [TK]D-Fender | ThoMe: Ok, you need to fix the formatting of your login line. do "show application AgentCallbacklogin" and verify your parameters |
| 15:16.44 | ThoMe | [TK]D-Fender: i don't know why. |
| 15:17.57 | iCEBrkr | [TK]D-Fender: Been really busy |
| 15:18.12 | *** part/#asterisk l3chuck_ (n=lk@62-101-102-100.ip.fastwebnet.it) |
| 15:18.15 | giasai68 | <giasai68> another think i got another warning : [Feb 21 16:02:19] WARNING[22591]: src/chan_h323.c:977 ooh323_indicate: Don't know how to indicate condition -1 on ooh323c_4187 |
| 15:18.22 | [TK]D-Fender | ThoMe: Go read the instustoms for the application like I showed you |
| 15:18.26 | iCEBrkr | [TK]D-Fender: Been working hard on a personal project of mine and this damn day job of mine actually requires me to do work. |
| 15:18.38 | [TK]D-Fender | iCEBrkr: Oh noes! |
| 15:18.42 | iCEBrkr | and LumenVox speech recognition um, rocks. BTW |
| 15:19.13 | *** join/#asterisk dshrout (n=chatzill@c-76-18-159-29.hsd1.tn.comcast.net) |
| 15:19.39 | [TK]D-Fender | iCEBrkr: Approx cost? |
| 15:21.35 | lupino3 | make dont-optimize seems broken.. it doesn't compile at all! :( |
| 15:21.42 | lupino3 | I'm running asterisk-1.2.15 |
| 15:22.52 | *** join/#asterisk bobbytux (n=bobbytux@LNeuilly-152-21-159-81.w193-253.abo.wanadoo.fr) |
| 15:23.14 | bobbytux | lo |
| 15:23.20 | *** join/#asterisk cpatry (n=junky@modemcable105.205-56-74.mc.videotron.ca) |
| 15:23.43 | *** join/#asterisk marv[work] (n=timr@24.214.206.254) |
| 15:24.30 | cpatry | when i ran ./configure, i see: checking for ZT_TONE_DTMF_BASE in zaptel.h... yes but app_meetme.c doesnt want to compile, any suggestion? |
| 15:24.39 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 15:24.44 | ManxPower | Looks like I may have been ripped off on an eBay auction for the first time ever and I've been using eBay for years. |
| 15:24.49 | lupino3 | I get lots of errors like |
| 15:24.57 | lupino3 | /usr/include/unistd.h:256: error: syntax error before "__nonnull" |
| 15:25.10 | lupino3 | compiling editline.c |
| 15:25.10 | mercestes | I hate end users. |
| 15:25.11 | lupino3 | :( |
| 15:25.11 | ManxPower | cpatry: did you do a "make clean" before "make"? |
| 15:28.01 | ManxPower | lupino3: my guess is either an old version of zaptel or a missing library |
| 15:28.11 | iCEBrkr | [TK]D-Fender: Oh, it's $$$$, but the dev SDK is only $50. Gives ya one channel. |
| 15:28.35 | iCEBrkr | [TK]D-Fender: But it's actually a bit cheaper than other speech recognition systems which are about $500 per line |
| 15:29.04 | iCEBrkr | ManxPower: Is he in Florida? I'll hunt him down for ya. |
| 15:29.19 | *** part/#asterisk m0nst3r (n=kvirc@c-71-206-116-36.hsd1.mi.comcast.net) |
| 15:29.54 | *** join/#asterisk _VoicePulse (n=contact@unaffiliated/voicepulse) |
| 15:31.27 | lupino3 | ManxPower, if you can help me, I can pastebin the output of make |
| 15:31.54 | lupino3 | ManxPower, I'm having some trouble with this debug, Asterisk doesn't behave properly when it's run under gdb |
| 15:32.08 | ThoMe | [TK]D-Fender: so.. now :-) |
| 15:32.14 | ThoMe | [TK]D-Fender: exten=> 2011,1,AgentCallbackLogin(${CALLERIDNUM}||${CALLERIDNUM}@intern) |
| 15:32.54 | ThoMe | <PROTECTED> |
| 15:33.04 | ManxPower | [TK]D-Fender: the amount was $10. His feedback was low, but not bad (he got lots of bad feedback after the audtion closed) so I took a chance. |
| 15:33.59 | *** join/#asterisk pardove (n=pardove@80.191.117.98) |
| 15:34.36 | [TK]D-Fender | ThoMe: That looks a lot better... |
| 15:34.58 | ThoMe | [TK]D-Fender: hm. can i logon also without a password? |
| 15:35.30 | awannabe | how do you get MOH to always be playing and when someone it put on hold it just starts where the music filing is currently being played at |
| 15:35.49 | *** join/#asterisk mivck (i=1000@ip-70-228.telesat.com.co) |
| 15:36.22 | ThoMe | ah. ok. agent => 51,,Thomas Mebes |
| 15:36.24 | ThoMe | agents.conf |
| 15:38.55 | giasai68 | <giasai68> another think i got another warning : [Feb 21 16:02:19] WARNING[22591]: src/chan_h323.c:977 ooh323_indicate: Don't know how to indicate condition -1 on ooh323c_4187 |
| 15:42.01 | *** join/#asterisk pdt (n=ptinsley@c-68-53-40-50.hsd1.tn.comcast.net) |
| 15:43.44 | [TK]D-Fender | awannabe: Use a streaming source, or use MPG123. |
| 15:44.10 | [TK]D-Fender | giasai68: *'s H323 support is weak and doesn't support all of the status indications you might want. |
| 15:46.07 | *** join/#asterisk pdt (n=ptinsley@c-68-53-40-50.hsd1.tn.comcast.net) |
| 15:46.25 | *** join/#asterisk jcast2 (n=jcast@adsl-76-211-179-129.dsl.wlfrct.sbcglobal.net) |
| 15:48.04 | lupino3 | is it possible that asterisk 1.2.15 doesn't compile without -O3 -fomit-frame-pointer? |
| 15:48.24 | ManxPower | lupino3: I compiled 1.2.15 last week with no issues |
| 15:48.47 | lupino3 | ManxPower, in dont-optimize mode? |
| 15:48.57 | lupino3 | uhm |
| 15:49.08 | lupino3 | strange thing, the vanilla tarball doesn't compile |
| 15:49.09 | ManxPower | lupino3: just "make" |
| 15:49.17 | lupino3 | plain "make" |
| 15:49.23 | lupino3 | but I already compiled it... |
| 15:49.45 | awannabe | [TK]D-Fender: ok, so i gotta use mpg123 then |
| 15:49.47 | e-ddie | you just said it doesnt compile |
| 15:49.49 | mercestes | When my retarded end users ineptly try to forward a voicemail to someone else because they are too damned lazy to do their jobs, they get a voicemail notification that says the voicemail is "0 duration" which is freaking them out like frightened cows and they suddenly think that the voicemail didnt' forward (but to my every test it did). Is there a way to fix the duration in the voicemail notification when you forward a voicemail so |
| 15:50.06 | lupino3 | e-ddie, yes, but some time ago |
| 15:50.07 | mercestes | I'm using ${vm_dur} in my email thingy but that's not working for forwarded voicemails |
| 15:50.11 | lupino3 | the first error is |
| 15:50.17 | ManxPower | mercestes: using what version of Asterisk? |
| 15:50.22 | mercestes | 1.2.13 |
| 15:50.23 | lupino3 | In file included from /usr/include/stdio.h:828, |
| 15:50.23 | lupino3 | <PROTECTED> |
| 15:50.23 | lupino3 | <PROTECTED> |
| 15:50.23 | lupino3 | <PROTECTED> |
| 15:50.23 | lupino3 | <PROTECTED> |
| 15:50.23 | lupino3 | /usr/include/bits/stdio.h:112: error: syntax error before '(' token |
| 15:50.24 | giasai68 | there aren' t solution to my problem rigth? |
| 15:50.28 | lupino3 | doesn't seem asterisk-related :( |
| 15:50.30 | [TK]D-Fender | mercestes: Have you confirmed if its case sensitive? |
| 15:50.41 | ManxPower | lupino3: you did a "make clean" |
| 15:50.44 | mercestes | It works if I don't forward the voiecmail |
| 15:50.52 | lupino3 | yes of course |
| 15:50.57 | giasai68 | or i need to have other version of *? |
| 15:50.58 | lupino3 | I even re-downloaded the tarball |
| 15:50.59 | [TK]D-Fender | mercestes: Ok... time to post a bug.... |
| 15:51.03 | lupino3 | and extracted it in another dir |
| 15:51.04 | lupino3 | :( |
| 15:51.14 | giasai68 | now i'm using *1.4 |
| 15:51.17 | mercestes | alrighty |
| 15:51.23 | jcast2 | Hello all. I just received my newly-purchased TDM400P card with one FXO and 1 FXS module. I plopped it into the machine and got the CHANCONFIG error when loading the wctdm module. Should I see the card when I do a `lspci'? |
| 15:51.25 | ManxPower | 2007-01-15 20:44 +0000 [r50946] Matt O'Gorman <mogorman@digium.com> |
| 15:51.26 | ManxPower | <PROTECTED> |
| 15:51.26 | ManxPower | <PROTECTED> |
| 15:51.32 | awannabe | [TK]D-Fender: isnt that the "pre 1.2" days? but still i need to do it that way your saying? |
| 15:51.49 | ManxPower | mercestes: there are at least a few VM updates in recent Asterisk versions |
| 15:52.20 | mercestes | 1.2.15? |
| 15:52.52 | ManxPower | mercestes: Don''t the know. |
| 15:52.57 | mercestes | That bears testing. |
| 15:53.02 | *** join/#asterisk pdt (n=ptinsley@c-68-53-40-50.hsd1.tn.comcast.net) |
| 15:53.59 | ThoMe | [TK]D-Fender: huhu? :-) |
| 15:54.14 | *** join/#asterisk pdt (n=ptinsley@c-68-53-40-50.hsd1.tn.comcast.net) |
| 15:55.26 | [TK]D-Fender | awannabe: Well since 1.2 you don't NED it, and its not SUGGESTED |
| 15:55.31 | mercestes | zOMG! Manx, I love you! |
| 15:55.35 | kippi | please could some have a look why i am getting this error http://www.pastebin.ca/366195 |
| 15:55.46 | lupino3 | well, it's MY problem, sorry for bothering you |
| 15:55.51 | lupino3 | the machine is not clean |
| 15:55.52 | *** join/#asterisk zmef420 (n=zmef420@metarb3-pool3-57.mtco.com) |
| 15:55.52 | lupino3 | :( |
| 15:56.30 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 15:56.47 | Dr-Linux | what could be wrong : http://phpfi.com/205717 |
| 15:57.01 | ThoMe | [TK]D-Fender: http://tm.muc.de/temp/a.txt |
| 15:57.17 | [TK]D-Fender | kippi: DBGet is deprectaded ; you're calling it wrong ; and your GotoIF is looking at a variablewith a double $$, and that is NOT an expression in your GogoIf. |
| 15:57.36 | [TK]D-Fender | kippi: You've seriously exceeded your error quota on that one :) |
| 15:58.46 | *** join/#asterisk dshrout (n=dshrout@c-76-18-159-29.hsd1.tn.comcast.net) |
| 16:01.41 | jcast2 | Hello all. I just received my newly-purchased TDM400P card with one FXO and 1 FXS module. I plopped it into the machine and got the CHANCONFIG error when loading the wctdm module. Should I see the card when I do a `lspci'? |
| 16:02.49 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
| 16:02.59 | ManxPower | jcast2: I think those modules ship in ports 3 and 4, not 1 and 2. make SURE you know which port is which. if you plug a phone line into and fxs port it will blow the module as soon as a call comes in |
| 16:03.11 | file | [TK]D-Fender: someone asked for a consultant in the Montreal area on -biz... thought of you :P |
| 16:03.51 | Dr-Linux | jcast2: lspci should show you the card |
| 16:04.16 | wunderkin | is HPEC applicable for PRIs? |
| 16:04.36 | file | wunderkin: if you've got echo |
| 16:04.51 | file | echo... echo... echo... |
| 16:05.00 | jcast2 | ManxPower and Dr-Linux, lspci doesn't show me the card and I haven't plugged anything into it yet. I was thinking that maybe I needed to in order for the modules to load, but thought better of it and decided to just plug the card in and attempt the modules first before connecting phones/lines to it. Perhaps I have a bad card. :-( |
| 16:05.09 | wunderkin | ah k i'll try it, we can't get the software echo cancel to work :/ echo echo echo file file file |
| 16:05.25 | ManxPower | wunderkin: from the mailing list messages I'll bet HPEC would take too much CPU to run 23 channels |
| 16:05.38 | Dr-Linux | :S |
| 16:05.40 | file | it depends on how many taps too |
| 16:06.07 | wunderkin | ah yeah that was the other problem, i forgot, we skimped and are using an athlon 750 because the intel pentium d or whatever it is now motherboard was not compatible with the t1 card.. ugh |
| 16:06.16 | Dr-Linux | jcast2: simply plug the card to the PCI slow and start the box and do "lspci" |
| 16:06.45 | *** join/#asterisk smace (n=smace_br@200.149.32.180) |
| 16:06.52 | ManxPower | wunderkin: there are external hardware EC available for T-1s |
| 16:07.08 | wunderkin | yeah thats what i was going to say.. darndit.. |
| 16:07.08 | tzafrir | ManxPower, it depends how many buckets you want per channel as well |
| 16:07.24 | smace | while compiling zaptel i'm getting one error: You do not appear to have the sources for the 2.6.9-42.EL kernel installed. |
| 16:07.41 | smace | how do I solve it? |
| 16:07.51 | ManxPower | smace: install the sources |
| 16:07.59 | smace | of zaptel? |
| 16:08.04 | angryuser | smace: kernel-sources |
| 16:08.09 | ManxPower | smace: no the linux kernel sources |
| 16:08.10 | wunderkin | so you have to do some rigging to get those tellab echo cans up? hmm sounds like fun |
| 16:08.10 | tzafrir | smace, yum install kernel-devel |
| 16:08.13 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 16:08.22 | kippi | could someone help me with this night service please? |
| 16:08.22 | Dr-Linux | jcast2: have a look here : http://www.syednetworks.com/asterisk-zaptelconf-configuration-with-x100p-fxo-and-tdm04b-cards |
| 16:08.27 | sweeper | where's a good reference on using sms? |
| 16:08.45 | Dr-Linux | sweeper: what sms? |
| 16:08.59 | sweeper | us-based, primarily |
| 16:09.06 | jcast2 | Dr-Linux, I have tried that and still no listing with lspci. Should I try it without the daughter-boards attached to see if that works? One other small observation is that the card doesn't get warm like the other cards in the machine. (but that may not mean a whole lot) |
| 16:09.11 | smace | after "yum install kernel-devel", what should I do? just "make"? |
| 16:09.17 | sweeper | I just want to see if I'm better off buying from a reseller or what |
| 16:09.26 | sweeper | aha |
| 16:09.47 | ManxPower | Dr-Linux: SMS is a GSM standard. Not really used in the USA except for on GSM networks. The term "SMS" is used for non-GSM networks, but they use standards that are not called "SMS". It is text messaging for cell phones |
| 16:09.57 | Dr-Linux | jcast2: try to connect molex 4 port connector to the card |
| 16:10.18 | smace | I keep getting the same erro. "You do not appear to have the sources for the 2.6.9-42.EL kernel installed." even after "yum" |
| 16:10.33 | sweeper | ManxPower: hmm, but if I get a link to, say, cingular, will they ONLY handle talking to cingular phones? |
| 16:10.38 | ManxPower | smace: put the output of "ls /usr/src" on pasebin.ca |
| 16:10.50 | ManxPower | sweeper: correct |
| 16:10.54 | sweeper | ah |
| 16:11.05 | sweeper | then I'm probablybetter off using a reseller |
| 16:11.08 | Dr-Linux | well, i'm using my nokia phone as SMS gateway thru opensource Gammu |
| 16:11.09 | smace | [ManxPower]: 2.6.9-42.0.8.EL-i686 |
| 16:11.13 | ManxPower | if you want multiple networks then you need a reseller |
| 16:11.41 | ManxPower | smace: as you can see the kernel sources you have installed do not match the running kernel |
| 16:11.53 | Dr-Linux | ManxPower: i'd like to ask you a question |
| 16:12.04 | Dr-Linux | ManxPower: what could be wrong : http://phpfi.com/205717 |
| 16:12.11 | jcast2 | Dr-Linux, I initially had the molex4 power connected. I dont' now, but the results are the same. Thanks for the URL, BTW. |
| 16:12.38 | angryuser | bye everybody have a nice day/evening/night |
| 16:12.46 | Dr-Linux | ManxPower: i'm sending calls to MultiTech VOIP gateway, but no luck, but i can recieve calls thru that gateways :S |
| 16:13.20 | tzafrir | smace, yum install kernel-devel |
| 16:13.22 | Dr-Linux | jcast2: i wrote that, but sorry for bad english, |
| 16:13.38 | Dr-Linux | jcast2: you can also call digium support, they will help you |
| 16:13.55 | Dr-Linux | angler: wa slaam :) |
| 16:14.00 | smace | I've just done: "yum install kernel-devel.i686"... but keeps the erro: You do not appear to have the sources for the 2.6.9-42.EL kernel installed. |
| 16:14.03 | Dr-Linux | angryuser: :) |
| 16:15.24 | jcast2 | Dr-Linux: I think I may call Digium Support, but I also feel I may have a bad card. |
| 16:16.38 | tzafrir | ls -l /lib/modules/`uname -r`/build |
| 16:16.39 | Dr-Linux | jcast2: maybe, but it can be problem with your board, or pci slots |
| 16:16.42 | jcast2 | Dr-Linux: Thanks for your help, though. |
| 16:17.02 | *** join/#asterisk poppo (n=adas@S0106004063d8e527.ed.shawcable.net) |
| 16:17.08 | Dr-Linux | jcast2: call digium and they tried and no luck then try on other machine still no luck, go to vendor and replace the card |
| 16:17.16 | smace | [tzafrir]: not found. |
| 16:17.32 | jcast2 | Dr-Linux: Perhaps. I had to take out a USB Controler card to make room for my new TDM card. The USB Controler worked fine... Hmm... |
| 16:17.41 | smace | tzafrir - should I LN to it ? |
| 16:18.08 | wunderkin | interesting.. i just got an echo report on a sip-sip call, they were on a handset... on ip430s ... hmm... |
| 16:18.11 | tzafrir | smace, hmmm ... you don't have kernel-devel for your current kernel version |
| 16:18.13 | jcast2 | Dr-Linux: Oh yeah, and I tried moving the card to different slots, but still no luck. |
| 16:18.22 | *** join/#asterisk russellb (i=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 16:18.22 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:20.33 | wunderkin | this again seems to be pointing back to the phones... i don't get it... someone had real bad echo (loud) on a new ip501 too but that was on the pri, they are hearing themselves |
| 16:20.50 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 16:21.09 | [TK]D-Fender | wunderkin: If PRI is involved, there's your problem. |
| 16:21.31 | Bobthehunter | is gsm g723 ? |
| 16:21.40 | puzzled | nope |
| 16:22.41 | wunderkin | [TK]D-Fender, i thought so too yes, but sip to sip.. no |
| 16:22.51 | ManxPower | g723 is a patented codec and not available for Asterisk for transcoding, only for passthru |
| 16:22.53 | wunderkin | they go across the same server |
| 16:23.08 | Bobthehunter | wahts the name of gsm then gXXX ? |
| 16:23.10 | Bobthehunter | also... |
| 16:23.16 | ManxPower | wunderkin: PRI will not eliminate echo |
| 16:23.24 | wunderkin | yeah i know that |
| 16:23.31 | ManxPower | Bobthehunter: GSM is called, oddly enough, "gsm" |
| 16:23.34 | wunderkin | i'm just saying that they have echo on sip to sip too even on a headset |
| 16:23.39 | wunderkin | er handset |
| 16:23.44 | ManxPower | wunderkin: then it is a phone issue. |
| 16:23.47 | Bobthehunter | anyone familiar with a device from nch.com.au no accepting the options request... its sends a 501 not implemented and therefore thiks the call failign and redirtcs to FXO |
| 16:23.49 | wunderkin | exactly |
| 16:23.56 | ManxPower | the PHONE is generating echo |
| 16:24.06 | wunderkin | we've had millions of phone issues *looks at fender* |
| 16:24.15 | ManxPower | wunderkin: what phone? |
| 16:24.17 | [TK]D-Fender | wunderkin: You are doing something horrifically wrong then. You are the best virgin sacrifice to the Polycom Gods EVER. Thanks ;) |
| 16:24.18 | *** join/#asterisk cubber (n=chatzill@static-74-41-185-190.br1.glv.ny.frontiernet.net) |
| 16:24.25 | wunderkin | bah! |
| 16:24.43 | wunderkin | ip430s and now ip501s, running 2.1.0, we have run everything from 1.6.something all the way up |
| 16:25.11 | cubber | I need to connect my asterisk box with our legacy PBX system which is a nortel norstar system. Could I use an FXO card on the asterisk box connected to an anolog extension on the nortel PBX to accomplish this? |
| 16:25.47 | [TK]D-Fender | cubber: Yes |
| 16:25.56 | ManxPower | wunderkin: take the sample sip.cfg and phone1.cfg that came with the version of the firmware you are running. apply any customizations to them and use them |
| 16:26.11 | ManxPower | the 430s were not supported in older firmware releases and will not work correctly. |
| 16:26.26 | cubber | D-Fender: so by doing that would the users on the PBX system be able to dial and reach all the extensions on the asterisk box and vice/versa? |
| 16:26.27 | ManxPower | with sip.cfg and phone1.cfgs from older releases |
| 16:26.33 | wunderkin | yeah i didnt use anything that the ip430 didn't support of course |
| 16:26.57 | wunderkin | i am using the sip and phone config that came with the firmware |
| 16:27.05 | wunderkin | i start fresh everytime |
| 16:27.15 | [TK]D-Fender | cubber: That would eb a way. The configuration of which is up to you. |
| 16:27.52 | cubber | D-Fender: would multiple users be able to use the connection or would I need multiple connections to support that? |
| 16:28.13 | *** join/#asterisk edison\ (n=edison@c-67-176-75-179.hsd1.wa.comcast.net) |
| 16:28.28 | edison\ | hey all |
| 16:28.38 | ManxPower | wunderkin: for example when we use the sip.cfg and phone1.cfg from a firmware release that did not support the 430s, the 430's gains were all screwed up and there were no icons next to the line buttons. |
| 16:28.45 | [TK]D-Fender | cubber: Well 1 port = 1 call. So if you expect multiple simultaneous connection, you'll need multiple ports... |
| 16:28.53 | *** join/#asterisk af_ (n=getsmart@ip-202-133.sn2.eutelia.it) |
| 16:28.55 | ManxPower | cubber: NO! NO! NO! |
| 16:29.10 | edison\ | is there an authoritative app_valetparking? I've found versions from Anthony Minessale and bkw |
| 16:29.14 | wunderkin | ManxPower, yeah i know better |
| 16:29.20 | ManxPower | The nortel will not indicate to Asterisk when the call should hang up. the nortel will also not send DTMF out analog station ports |
| 16:29.22 | cubber | ManxPower: ?? |
| 16:29.44 | cubber | ManxPower: so it would not work then? |
| 16:29.44 | ManxPower | your best bet is use FXS cards in Asterisk and plug it into the analog FXO/CO ports on the |
| 16:29.46 | ManxPower | nortel |
| 16:30.03 | tzanger | ManxPower: yep, or a DTI and use CAS T1 if you've already got it |
| 16:30.19 | ManxPower | tzanger: we ordered a DT1 last week for our nortel |
| 16:30.31 | [TK]D-Fender | cubber: I used the aforementioned FXS method personally using ATA's, but its a little more kludgy. WOrks though. |
| 16:30.33 | cubber | any suggestions on hardware? |
| 16:30.47 | puzzled | does anyone with experience updating the firmware of a 7961 have some notes to share? |
| 16:30.54 | [TK]D-Fender | cubber: How many ports do you think you'll need? |
| 16:31.20 | wunderkin | i know in sip 2.1.0 or shortly before there, they had to adjust the sidetone on the ip430, said it was too hot, but hmm what else could i be doing that is different :/ |
| 16:31.24 | cubber | right now the pbx is used for all of our phone calls and I have an asterisk box set up for an intercom system |
| 16:31.44 | cubber | they want me to set it up so they can dial an extension off the pbx phones to use the intercom |
| 16:31.58 | cubber | but in the future we are looking to migrate everything to asterisk |
| 16:32.18 | [TK]D-Fender | wunderkin: Less = More |
| 16:32.45 | [TK]D-Fender | cubber: Why is * your intercom and not using the standard Nortel paging? |
| 16:32.46 | cubber | At the moment I have the asterisk box set up with 2 voip phone lines provided by telesip and they can just dial those numbers to reach the intercom from the PBX, a cheap and easy solution |
| 16:32.54 | wunderkin | maybe i mucked something up in the config but i used what came with it and went from there.. shrug |
| 16:32.55 | Bobthehunter | nch.com.au/pbx/index.html |
| 16:33.01 | Bobthehunter | that the app not liking options.. |
| 16:33.39 | cubber | because our phone company provides us the pbx through a rental contract and they wanted over 4000 dollars to hook up the intercom |
| 16:33.44 | [TK]D-Fender | cubber: Then you'll want to use an ata like the Linksys SPA-2002 and connect to 2 line ports (not phone ports). Cost = $70 |
| 16:33.46 | cubber | I set up the voip solution for under 300 |
| 16:33.56 | wunderkin | they have been reporting a 'sticky key' problem too, where they press a number and they hear a long dtmf tone for a long time, they can press the other buttons but they dont register... happening on the new phones too.. ugh |
| 16:34.09 | *** join/#asterisk topping (n=topping@204.152.96.238) |
| 16:34.09 | [TK]D-Fender | cubber: How big is your system? |
| 16:34.25 | cubber | the pbx has about 20 extension on it now and has 8 lines |
| 16:34.53 | wunderkin | they are also reporting on one of the new phones, sometimes when they press the number it does not register either so she keeps pressing it and then after a bit they all show up at once |
| 16:34.57 | ManxPower | wunderkin: set dialplan.impossibleMatchHandling="1" in sip.cfg and phone1.cfg it may be listed multiple times |
| 16:35.25 | ManxPower | I think the default tries to do early dial stuff and that caused what you are describing with buttons |
| 16:36.09 | *** join/#asterisk Schreiber1337 (i=cee4b403@gateway/web/cgi-irc/ircatwork.com/x-f22ed3f578ccbf93) |
| 16:36.26 | [TK]D-Fender | You could replace your entire system for about that much.... |
| 16:36.47 | cubber | yah I would love to replace the entire system but my boss got us locked into a 5 year contract with them before I got here |
| 16:37.42 | [TK]D-Fender | wunderkin: set dialplan.impossibleMatchHandling="2" <- much more fun. While you're at is set your dialplan to : "x.T|*.T|#.T", and removeendofdial="0" |
| 16:37.48 | edison\ | anyone know if there's a valet parking that works for 1.4? |
| 16:37.53 | [TK]D-Fender | cubber: Poor chumps. |
| 16:37.58 | wunderkin | ManxPower, i was thinking about setting it to 1 but i wanted it to dial the asterisk server and tell them that they are stupid and entered the wrong thing oh well... i dont want to set it to 2, they will bitch at me |
| 16:38.09 | [TK]D-Fender | edison\: Anthm recently modded it for 1.4 |
| 16:38.38 | [TK]D-Fender | wunderkin: Why would they bithc? Works great when paired with the rest of the settings I gave. |
| 16:38.41 | cubber | I am looking at this linksys device now, to me it looks like you would hook the pbx into one of the phone ports and then connect that to a network switch? How does it communicate with the asterisk box to bridge the 2 systems? |
| 16:38.53 | edison\ | [TK]D-Fender: cool, thx. Know where I can find it? |
| 16:39.00 | *** join/#asterisk RoyK (n=roy@81.191.113.238) |
| 16:39.02 | wunderkin | [TK]D-Fender, because pressing send or # is more work for them :P |
| 16:39.07 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 16:39.14 | [TK]D-Fender | edison\: I don't think he formally linked it. ZLet me get back to him on that. |
| 16:39.22 | ManxPower | wunderkin: my users are not required to press SEND or # |
| 16:39.25 | [TK]D-Fender | wunderkin: You don't have to.. thats what the "T" is for... |
| 16:39.40 | wunderkin | ok, so they have to wait 3 seconds :) |
| 16:39.50 | [TK]D-Fender | wunderkin: Dial whatever you want + wait 3S or hit send. |
| 16:39.57 | wunderkin | yeah i know :D |
| 16:40.02 | [TK]D-Fender | wunderkin: I'm sure they'd prefer that it WORKED :) |
| 16:40.06 | ManxPower | wunderkin: a well designed dialplan will send the call as soon as the user is finished dialing for anything except international calls |
| 16:40.09 | wunderkin | lol it does when they dial right |
| 16:40.10 | *** part/#asterisk smace (n=smace_br@200.149.32.180) |
| 16:40.38 | [TK]D-Fender | wunderkin: My way lets them correct as they go and doesn't bitch at them super-fast or mangle their entry. |
| 16:40.52 | wunderkin | ManxPower, thats what i'm trying to do, i did not require 9 before and now they are still getting used to it i guess, but it just threw me off that sometimes they get a long dtmf, it ends up hitting the dial timeout and sends the call through |
| 16:40.56 | [TK]D-Fender | wunderkin: And ensures you aren't chasing to keep * & Polycom synched |
| 16:41.37 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 16:41.52 | wunderkin | i can do either way, but also what confuses me is why one person is reporting that other problem that hitting the number key does not register for several seconds later |
| 16:42.03 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 16:44.34 | *** join/#asterisk Inetmaster (n=Inetmast@cblmdm72-240-249-212.buckeyecom.net) |
| 16:45.30 | Schreiber1337 | Any help here for a noob? |
| 16:46.46 | [TK]D-Fender | Schreiber1337: Ask a specific question and you might get a specific answer... |
| 16:47.21 | *** join/#asterisk gr1ncheux_ (n=devine@unaffiliated/gr1ncheux) |
| 16:47.52 | mercestes | The only help for noobs is education and experience. |
| 16:48.54 | Schreiber1337 | I have worked with TrixBox and a few other canned installs before, but this is my first manual install on a debian system. I installed zaptel, libpri, asterisk, asterisk sounds, and asterisk addons... Everything seems to work like I'm used to, but I can't get any GSM files to playback |
| 16:48.54 | *** join/#asterisk amdtech (i=amdtech@nat/digium/x-1e4566282909e068) |
| 16:50.25 | Schreiber1337 | The CLI shows it playing, but I can't hear anything on the phones. |
| 16:50.38 | [TK]D-Fender | Schreiber1337: Pastebin the CLI output of a failed call in it entirety. |
| 16:50.40 | [TK]D-Fender | ~pb |
| 16:50.49 | jbot | i heard pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 16:50.50 | mercestes | Schreiber1337: Are you sure it's not one way audio? Any nat / firewall.? |
| 16:51.47 | Schreiber1337 | No, firewall... on the same LAN. |
| 16:52.13 | *** join/#asterisk friedrich| (n=friedric@e177250126.adsl.alicedsl.de) |
| 16:52.48 | Schreiber1337 | Phones are registering correclty with SIP, but you can't hear any sounds generated by the Asterisk system, not even beeps. |
| 16:53.11 | mercestes | Schreiber1337: Can they call each other and hear voice? |
| 16:53.21 | mercestes | you should ~pb cli output too |
| 16:53.21 | Schreiber1337 | Yes, |
| 16:54.19 | Schreiber1337 | The CLI shows : -- Executing [2799@unrestricted:1] Playback("SIP/2700-b6b53e78", "/var/lib/asterisk/sounds/vm-theperson") in new stack |
| 16:54.40 | Schreiber1337 | <PROTECTED> |
| 16:54.53 | Schreiber1337 | But nothing is heard on the phone |
| 16:55.12 | mercestes | Schreiber1337: Pastebin. Pastebin. paste........biiiin. |
| 16:55.35 | ManxPower | Schreiber1337: in sip.conf disallow=all and allow=ulaw |
| 16:55.59 | Schreiber1337 | Sorry, what are you asking me to do "Pastbin" |
| 16:56.18 | ManxPower | Schreiber1337: if you need to paste more than 1 or 2 lines, use pastebin |
| 16:56.21 | [TK]D-Fender | ~pb |
| 16:56.35 | jbot | pb is probably a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 16:57.53 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 16:57.58 | Inetmaster | Where might I be able to adjust the number of rings before a Zaptel channel is answered? |
| 16:58.50 | Qwell[] | Inetmaster: does your line have callerid? |
| 16:59.00 | ManxPower | Inetmaster: usercallerid=no |
| 16:59.16 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 16:59.32 | [TK]D-Fender | Inetmaster: in your dialplan. |
| 16:59.43 | [TK]D-Fender | Inetmaster: "show application wait" |
| 17:01.05 | *** join/#asterisk J4k3 (i=J4k3@dhcp-12-197-128-58.intrastar.net) |
| 17:01.06 | Schreiber1337 | Here is what my current sip.conf looks like http://channels.debian.net/paste/5449. |
| 17:01.35 | ManxPower | Schreiber1337: I can't help you if you don't follow my advice. |
| 17:01.53 | ManxPower | now put in disallow=all and allow=ulaw either in [general] or in the [sipdevice] section |
| 17:01.54 | Schreiber1337 | ManxPower: should I add disallow=all and allow=ulaw into sip.conf |
| 17:02.05 | Schreiber1337 | k |
| 17:02.50 | Inetmaster | Yes, I have callerid. If I add usercallerid=no, will that disable callerid functionality (will it be passed to SIP phones)? |
| 17:03.40 | Schreiber1337 | ManxPower: I did that... same result. |
| 17:03.46 | ManxPower | Inetmaster: what specifically do you want to do? reduce the time or increase the time? |
| 17:03.54 | ManxPower | Schreiber1337: did you do a reload in the CLI? |
| 17:04.20 | Inetmaster | I need to increase the time (add rings) |
| 17:04.43 | Schreiber1337 | I did a /etc/init.d/./asterisk reload |
| 17:04.46 | ManxPower | Inetmaster: Ah! You should have said so in the first place. put in a Wait(timeout) in your dialplan |
| 17:05.04 | ManxPower | Schreiber1337: no way to know if that worked. "asterisk -rvvv" then issue "reload" |
| 17:05.18 | Schreiber1337 | OK |
| 17:06.39 | Schreiber1337 | ManxPower:OK, did that... |
| 17:06.56 | Inetmaster | ManxPower: Sorry - Well this make make it simpler: I'd prefer to configure one of my channels not to answer at all...is there a way to do this? |
| 17:07.21 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 17:08.11 | ManxPower | Inetmaster: put the channel in a context all by iteself (context=noanswer for example) in zapata.conf, then do exten => s,1,Wait(999) |
| 17:08.14 | ManxPower | in that context |
| 17:10.46 | Inetmaster | ManxPower: Thanks...I'll be trying that! |
| 17:13.12 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 17:13.41 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 17:13.47 | [TK]D-Fender | ManxPower: Whats the point of that? Better jsut to toss it into a context that has no "s" and just let it fall through |
| 17:14.02 | *** join/#asterisk edison\ (n=edison@c-67-176-75-179.hsd1.wa.comcast.net) |
| 17:14.23 | ManxPower | [TK]D-Fender: stop interupting me with logic! 8-) |
| 17:14.43 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 17:14.53 | edison\ | hey guys, is there a way to always execute dialplan code, regardless of the extension? |
| 17:14.59 | ManxPower | edison\: no |
| 17:15.05 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 17:15.09 | Qwell[] | _. :P |
| 17:15.11 | [TK]D-Fender | ManxPower: And the fact you were barking up the CID tree aimlessly :) Answer his question, then question his INTENT ;) chan_psychic.so is currently buggy! |
| 17:15.33 | *** join/#asterisk qdk (n=qdk@90.184.3.249) |
| 17:15.47 | *** join/#asterisk bmd (n=bmd@72.54.252.34) |
| 17:15.54 | *** join/#asterisk apardo (n=apardo@87.217.145.247) |
| 17:16.21 | edison\ | ManxPower: how about an 'autoexec' on dialplan load to prodecurally create entries (or hints in my case)? |
| 17:16.55 | [TK]D-Fender | edison\: Sure. Use a catch all extension like "exten _.,1,Goto(${EXTEN},1)" and you'll execute all sorts of dialplan code regardless of extension. Oh and don't forget to point all channel definitions towards that context :) |
| 17:17.17 | [TK]D-Fender | edison\: No such thing as dynamic hints. Hardcode or else.... |
| 17:17.34 | Qwell[] | #exec |
| 17:17.46 | [TK]D-Fender | ? |
| 17:18.04 | [TK]D-Fender | Qwell :addendum for me? |
| 17:18.09 | Qwell[] | yeah |
| 17:18.17 | Qwell[] | You *could* use #exec to do something like that |
| 17:18.31 | [TK]D-Fender | Qwell : in the dialplan? |
| 17:18.58 | wunderkin | they seem to have a lot more echo problems with MG2 than KB1 also.. i thought MG2 was a modified KB1 and supposted to be better ;D |
| 17:19.06 | Qwell[] | in extensions.conf |
| 17:19.19 | Qwell[] | #exec myhintsscript.sh |
| 17:19.24 | edison\ | [TK]D-Fender won't _. *always* match? Doesn't that extension continually loop? |
| 17:19.34 | Qwell[] | then in myhintsscript.sh, just do some database magic, and put some hints on stdout |
| 17:19.49 | *** join/#asterisk Dovid (n=Dovid@85.159.160.207) |
| 17:19.56 | [TK]D-Fender | edison\: Well you said you wanted to execute dialplan code regardless of extension. Don't go whining at the solution! |
| 17:20.00 | ManxPower | edison\: _. will match anything, everything and in my opinion should amost never be used. |
| 17:20.17 | Schreiber1337 | ManxPower: Do you have any other suggestions of what to try? |
| 17:20.18 | wunderkin | anything including h, t, etc |
| 17:20.38 | Qwell[] | I've only ever seen one valid usage of _. |
| 17:20.39 | [TK]D-Fender | ManxPower: But... fulfills his request in an absolute manner ;) |
| 17:20.44 | ManxPower | Schreiber1337: no. |
| 17:20.55 | Qwell[] | and that was just a noop at priority 1 |
| 17:20.57 | *** join/#asterisk hellojoe_ (n=hellojoe@nat-service4.juniper.net) |
| 17:21.36 | hellojoe_ | <PROTECTED> |
| 17:21.43 | [TK]D-Fender | Qwell : I've had another. on a generic inbound context that throws absolutely ever inbound connectio to another context whereby you leave the problems behind |
| 17:22.06 | Schreiber1337 | Anyone have any idea, why I can't hear playback on my phones. |
| 17:22.23 | edison\ | Qwell: I suppose you could rebuild the entire dialplan flow with ._ |
| 17:22.35 | [TK]D-Fender | Schreiber1337: Do we really have to ask you like 10 times for the full CLI out of that defective call? |
| 17:22.36 | Schreiber1337 | Or any sounds from the Asterisk system |
| 17:22.44 | Dovid | can anyone help me with this ? |
| 17:22.44 | edison\ | so if you don't like how * handles priorities or whatever, make it flow your own way |
| 17:22.45 | Dovid | http://www.pastebin.ca/366375 |
| 17:22.49 | Dovid | happend when i upgraded the kernel |
| 17:22.52 | mafkees | Qwell[]: hhmm, nice trick :) |
| 17:23.13 | wunderkin | Schreiber1337, do you have a t1 card installed? |
| 17:24.43 | [TK]D-Fender | Dovid: Make sure you have the matching HEADERS, not just the kernel source |
| 17:24.51 | edison\ | Qwell[] where's #exec documented? |
| 17:25.06 | Qwell[] | edison\: dunno |
| 17:25.10 | Schreiber1337 | Sorry, here is what I see in the CLI http://channels.debian.net/paste/5451?tx=on&submit=Format+it%21 |
| 17:25.18 | Qwell[] | wiki? book? |
| 17:25.21 | Schreiber1337 | Yes, I have a T1 card in it. |
| 17:25.28 | edison\ | wiki search doesn't pull anything. |
| 17:25.39 | wunderkin | Schreiber1337, and you are taking timing from something that is not connected? i've had this problem before |
| 17:25.53 | [TK]D-Fender | Dovid: And FC5 has had a host of problems with * |
| 17:26.02 | Dovid | TK: ehat do umena by exact headers ? |
| 17:26.08 | Dovid | tk: what do u reccomend ? |
| 17:26.20 | *** join/#asterisk thekidrio (n=thekidri@66.107.42.13) |
| 17:26.23 | [TK]D-Fender | Dovid: There is kernel source, and then a seperate headers package IIRC |
| 17:26.50 | Dovid | TK: but isnt the sources gona be called something like kernel-source ? |
| 17:26.52 | Schreiber1337 | wunderkin: Not sure, I havn't even started working with the card. Other than to install zaptel. |
| 17:26.56 | Dovid | i didnt see any with yum as int he pb |
| 17:27.24 | Schreiber1337 | wunderkin: Where should I look? |
| 17:28.00 | wunderkin | Schreiber1337, i think you need to either setup the t1 and connect it, set timing to 0, or rmmod the t1 modules and use ztdummy.. i guess |
| 17:28.12 | Schreiber1337 | OK, let me look into those options. |
| 17:28.16 | Schreiber1337 | thx |
| 17:29.43 | Dovid | TK: what would the headers be called ? |
| 17:31.30 | [TK]D-Fender | Schreiber1337: You are having problems directly between SIP phones & * you said, so could you please provide the output requested. T1 issues are barking up the wrong tree. |
| 17:32.22 | [TK]D-Fender | Schreiber1337: Is that single like all there is? You should also be issuing an "Answer" before trying to play back into a channel. |
| 17:32.33 | wunderkin | well, its not like he configured the t1, he wouldn't have known any different |
| 17:32.38 | [TK]D-Fender | Schreiber1337: And while you're at it, please pastebin the config for that SIP peer.... |
| 17:32.45 | Dovid | there are my avail updates: http://www.pastebin.ca/366399 |
| 17:32.50 | [TK]D-Fender | wunderkin: T1 is not an issue here.... |
| 17:32.56 | wunderkin | i've had the problem before |
| 17:33.04 | wunderkin | oh.. |
| 17:33.09 | wunderkin | the t1 isnt configured though? |
| 17:33.14 | [TK]D-Fender | wunderkin: His SIP phone can't ehar a simple audio file from *. |
| 17:33.18 | wunderkin | i know |
| 17:33.21 | [TK]D-Fender | wunderkin: What would that have to do with T1? |
| 17:33.48 | wunderkin | i've had a problem before where if you had a t1 setup and set for timing from the telco and the line isn't connected, you can't hear anything even on a sip call when trying to playback a file.. i know.. weird |
| 17:34.10 | wunderkin | i don't know what that has to do with timing |
| 17:35.17 | [TK]D-Fender | wunderkin: I wouldn't bother thinking about that right now. Aim for the pieces that we can really test and examine before aiming for the "obscure" and "hard to prove" bits. |
| 17:38.17 | wunderkin | i forgot to ask if it was configured, my mind is on my problems right now, i need to pack the jump to conclusions board, he said phone to phone was ok |
| 17:38.35 | Schreiber1337 | OK, here is an example of the problem and it's output. http://channels.debian.net/paste/5452?tx=on&submit=Format+it%21 |
| 17:38.54 | Schreiber1337 | wunderkin: I'm also looking at the timing. |
| 17:38.59 | [TK]D-Fender | wunderkin: And users are NEVER EVER wrong... especially here ;) |
| 17:39.14 | *** join/#asterisk bkw__ (i=brian@adsl-70-143-62-84.dsl.tul2ok.sbcglobal.net) |
| 17:39.41 | [TK]D-Fender | Schreiber1337: I just finished telling you that you should issue an "answer first, and then I asked to see your phones config...... |
| 17:40.09 | [TK]D-Fender | Schreiber1337: Please pay attention |
| 17:41.23 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 17:41.29 | Schreiber1337 | Sorry... What would you like me to provide for the phone configs? |
| 17:41.36 | sweeper | what expression would I use to trim the first digit off of ${EXTEN}? |
| 17:41.42 | Qwell[] | :! |
| 17:41.43 | Qwell[] | :1 |
| 17:41.57 | ManxPower | sweeper: read README.variables |
| 17:42.19 | Schreiber1337 | D-Fender: The sip.conf contents? |
| 17:43.29 | [TK]D-Fender | Schreiber1337: *sigh* YES |
| 17:43.29 | lupino3 | Hello there... |
| 17:43.40 | lupino3 | is it possible that I get a crash when Asterisk is compiled using "make" |
| 17:43.55 | lupino3 | and then I don't get it if I compile it with "make dont-opt"? |
| 17:44.06 | lupino3 | s/opt/optimize/ |
| 17:44.08 | [TK]D-Fender | Schreiber1337: Then cahnge your diaplan test to do Answer first, then playback that sound file 3-4 times at least before hangining up. |
| 17:44.13 | lupino3 | yes |
| 17:44.22 | lupino3 | sorry for the typo |
| 17:44.25 | [TK]D-Fender | sweeper: ${EXTEN:1} |
| 17:44.38 | [TK]D-Fender | sweeper: Go read up on "asterisk variables" on the WIKI |
| 17:46.32 | sweeper | yay \o\ |
| 17:47.07 | Schreiber1337 | [TK]D-Fender: http://channels.debian.net/paste/5454?tx=on&submit=Format+it%21 |
| 17:48.34 | lupino3 | jbot: so, any suggestions? |
| 17:48.45 | [TK]D-Fender | Schreiber1337: And you can hear SOMETHING else through that soft-phone? |
| 17:49.06 | Dr-Linux | question, can i edit cisco 79xx phone setting from telnet access? |
| 17:49.08 | [TK]D-Fender | lupino3: Stop talking to the bot.... it'll only ignore you :) |
| 17:50.37 | lupino3 | lol that's a bot? |
| 17:50.37 | [TK]D-Fender | ... |
| 17:50.37 | [TK]D-Fender | ~lart lupino3 |
| 17:50.37 | terrapen | anyone here know a Justin J Newman? |
| 17:50.37 | lupino3 | didn't know, I don't come here often |
| 17:50.37 | terrapen | or is he here? |
| 17:50.37 | BrianR___ | When doing a agentcallbacklogin, where is the context that the outbound calls are placed from configured? |
| 17:50.37 | Schreiber1337 | [TK]D-Fender: It's a physical phone, but yes. I can call other extensions and carry on conversations. |
| 17:50.38 | Bobthehunter | anyone have a valid TFTP config ?for spa2100 ? |
| 17:50.38 | [TK]D-Fender | brianIn that line itself |
| 17:50.38 | Bobthehunter | dont ahve the pasword for dEFAULT RESET |
| 17:50.38 | Bobthehunter | so ill hack it |
| 17:50.38 | terrapen | i just got a resume from this guy...has lots of asterisk SMRTs |
| 17:51.18 | [TK]D-Fender | Schreiber1337: do an echo test : "exten => 2798,1,Answer" . "exten => 2798,2,Echo" |
| 17:51.27 | lupino3 | O_O |
| 17:51.58 | [TK]D-Fender | jbot : c'mere boy! |
| 17:52.00 | [TK]D-Fender | ~areyouadog? |
| 17:52.01 | jbot | [TK]D-Fender is a nub! |
| 17:52.08 | [TK]D-Fender | z0mg! |
| 17:52.17 | lupino3 | however, should I file a bug in Mantis? |
| 17:52.19 | [TK]D-Fender | Who has perverted my little pooch! |
| 17:53.15 | [TK]D-Fender | ~areyouadog |
| 17:53.16 | jbot | well, areyouadog is <response>Bark! Bark! |
| 17:53.23 | Schreiber1337 | [TK]D-Fender: Yes, the echo test on 2798 works correctly. |
| 17:53.24 | [TK]D-Fender | darn... almost ;) |
| 17:53.41 | [TK]D-Fender | Schreiber1337: Try picking another sound file. |
| 17:53.56 | [TK]D-Fender | Schreiber1337: make a recording and play it back. |
| 17:54.35 | *** join/#asterisk zillao (n=medinaor@190.40.18.195) |
| 17:54.55 | Dr-Linux | question, can i edit cisco 79xx phone setting from telnet access? |
| 17:57.20 | *** join/#asterisk MarkWD (n=Mark@rrcs-67-78-88-186.sw.biz.rr.com) |
| 17:57.32 | phearless | Dr-Linux: yes |
| 17:57.36 | phearless | cf voip info |
| 17:58.27 | [TK]D-Fender | ~areyouadog |
| 17:58.29 | jbot | Bark! Bark! |
| 17:58.31 | [TK]D-Fender | :D |
| 17:58.34 | [TK]D-Fender | there we go! |
| 17:58.42 | Schreiber1337 | [TK]D-Fender: I just tried both suggestions and neither worked. I didn't even get the beep that I normally here when using this routine http://channels.debian.net/paste/5455?tx=on&submit=Format+it%21 |
| 17:59.29 | [TK]D-Fender | Schreiber1337: For sanity's sake can you please start putting Answer's at the top of things like that? |
| 17:59.46 | Dr-Linux | phearless: how? can you give me any guide link? i wanna edit "directory_url" from the telnet |
| 18:00.13 | phearless | ~wiki |
| 18:00.26 | phearless | ~voip-info |
| 18:00.27 | jbot | hmm... voip-info is the Voice Over IP wiki. It is a community resource which will answer all of your questions, from Asterisk to ZTDummy. You can find it over at http://www.voip-info.org - well worth bookmarking |
| 18:00.36 | phearless | Dr-Linux here it is :) |
| 18:00.54 | phearless | you will find all the good stuff for 79xx phones |
| 18:00.55 | *** join/#asterisk friedrich| (n=friedric@e177255050.adsl.alicedsl.de) |
| 18:01.17 | Dr-Linux | phearless: i know, but i can't find any help to edit phone's setting via telnet |
| 18:01.26 | phearless | ah shit |
| 18:01.44 | [TK]D-Fender | Dr-Linux: Try www.cisco.com |
| 18:02.16 | Bobthehunter | http://www.pastebin.ca/366446 |
| 18:02.19 | Bobthehunter | anyidea ? |
| 18:02.36 | Dr-Linux | hhm.... :S |
| 18:02.50 | [TK]D-Fender | Bobthehunter: They don't like "Qualify" |
| 18:03.00 | Bobthehunter | seems like it and im not even sedning it or shoudnt |
| 18:03.38 | [TK]D-Fender | Bobthehunter: Go look and prove it. |
| 18:03.52 | Bobthehunter | yes weird |
| 18:03.54 | Bobthehunter | hmm |
| 18:06.51 | Schreiber1337 | [TK]D-Fender: http://channels.debian.net/paste/5456?tx=on&submit=Format+it%21 |
| 18:07.25 | *** join/#asterisk hellojoe (n=hellojoe@nat-service4.juniper.net) |
| 18:08.51 | giasai68 | i heve this worning : [Feb 21 19:08:19] WARNING[3261]: src/chan_h323.c:977 ooh323_indicate: Don't know how to indicate condition -1 on ooh323c_6206 |
| 18:09.06 | giasai68 | [Feb 21 19:08:19] WARNING[3261]: src/chan_h323.c:977 ooh323_indicate: Don't know how to indicate condition -1 on ooh323c_6206 |
| 18:09.08 | giasai68 | [Feb 21 19:08:19] WARNING[3261]: src/chan_h323.c:977 ooh323_indicate: Don't know how to indicate condition -1 on ooh323c_6206 |
| 18:09.10 | [TK]D-Fender | Schreiber1337: You system is looking pretty screwed. I'd suggest a complete reinstall and recompile |
| 18:09.30 | giasai68 | sorry for multiple msg my mistake |
| 18:09.35 | [TK]D-Fender | giasai68: I've answered this TWICE aready. Stop with the broken-record syndrome... |
| 18:10.04 | giasai68 | yes i know i insert in wrong box my mistake |
| 18:10.58 | Schreiber1337 | Sh*t, I'll try again from scratch. |
| 18:13.00 | [TK]D-Fender | Schreiber1337: Sorry, but those warnings are just nasty. Do back up your configs though |
| 18:15.23 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 18:19.07 | MaartenB | hello everyone |
| 18:19.17 | MaartenB | I am having trouble with my Asterisk installation |
| 18:19.26 | MaartenB | making local calls to other SIP accounts is perfect |
| 18:19.48 | MaartenB | but making outside calls with my voip providers, works, but my own speech sounds very garbled |
| 18:20.09 | MaartenB | I have switched to different protocols, but I am unable to solve it |
| 18:20.31 | MaartenB | connectivity between my asterisk box, and the providers gateway is perfect, 0 % packetloss and good bandwidth |
| 18:20.54 | MaartenB | any ideas what to try next? is there something available to debug or checks what is going wrong? |
| 18:27.03 | MaartenB | anyone? |
| 18:27.17 | Schreiber1337 | [TK]D-Fender: Do you recommend using AsteriskNow, or is it just another watered down install? |
| 18:27.53 | [TK]D-Fender | Schreiber1337: No I don't , and its a GUI. I wouldn't touch it personally. |
| 18:29.41 | *** join/#asterisk mavior (n=Miranda@88-149-162-164.f5.ngi.it) |
| 18:29.41 | Schreiber1337 | [TK]D-Fender: thx |
| 18:30.14 | mercestes | MaartenB: do an mtr from your * box to your providers interface, measuring jitter. |
| 18:30.29 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 18:30.34 | mavior | hello nice people, anybody knows how to delete my recordered name in mailbox? |
| 18:31.36 | *** join/#asterisk anthonyl (n=anthonyl@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 18:31.58 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 18:33.43 | mavior | anyone? |
| 18:34.17 | MaartenB | mercestes, mtr shows a Avg ping of 5 ms and a 0 % loss |
| 18:34.18 | Maveric | cd /var/spool/asterisk/vm/vmmailboxhere/ |
| 18:34.28 | Maveric | that is for mavior |
| 18:34.35 | MaartenB | mercestes, which is excellent imho :) |
| 18:35.42 | mercestes | MaartenB: 350ms ping is excellent too, if it stays at 350ms and never changes. What is your jitter? |
| 18:36.25 | mavior | maveric: got this list busy.gsm greet.gsm greet.wav greet.WAV INBOX Old tmp |
| 18:36.27 | mavior | from |
| 18:36.49 | mavior | cd /var/spool/asterisk/voicemail/default/1234/ that is my mailbox |
| 18:37.33 | Maveric | i'm not positive on which one contains your recorded name |
| 18:37.36 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:37.38 | Maveric | but if you want to just reset |
| 18:37.42 | Maveric | your voicemail prompts |
| 18:37.55 | mercestes | Maveric: "Greet.wav" |
| 18:37.55 | Maveric | just delete the busy.gsm greet.gsm greet.wav and greet.WAV |
| 18:38.04 | anthonyl | how mang g729 liceneses do i need |
| 18:39.13 | MaartenB | mercestes, how do I measure Jitter with mtr? |
| 18:39.30 | [TK]D-Fender | anthonyl: None |
| 18:39.43 | mavior | ok ty |
| 18:39.57 | MaartenB | mercestes, if it is StDev, then it is 0.6 |
| 18:40.19 | *** join/#asterisk RoyK (n=roy@ti211310a080-9075.bb.online.no) |
| 18:41.32 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 18:43.59 | SheriF_SpacE | <PROTECTED> |
| 18:44.57 | *** part/#asterisk pythos (i=lanebob@unaffiliated/pythos) |
| 18:45.48 | [TK]D-Fender | SheriF_SpacE: Joine #asterisk-gui for support on it |
| 18:45.59 | mercestes | MaartenB: Should be "O" for "Order of Fields." |
| 18:47.07 | SheriF_SpacE | [TK]D-Fender: i'm there but no one there :P |
| 18:47.11 | MaartenB | mercestes, found it, it is around 1 |
| 18:47.42 | mercestes | MaartenB: Hrm. Probably provider end. |
| 18:48.03 | [TK]D-Fender | SheriF_SpacE: This fals squarely in the "TFB" category..... |
| 18:48.33 | MaartenB | mercestes, actually a test call was fine now too |
| 18:48.36 | MaartenB | strange |
| 18:48.46 | MaartenB | what is the maximum Jitter I should have to work normally? |
| 18:50.15 | mercestes | MaartenB: Hard to say, really< I hate to throw out a number. I just know when it looks "not right." |
| 18:50.30 | mercestes | MaartenB: I would monitor the jitter value and continue to test calls and see if it varies when a call is bad. |
| 18:50.32 | Nugget | MaartenB: I suggest you stop after the third cup of coffee. |
| 18:50.52 | MaartenB | Nugget, :) |
| 18:51.04 | MaartenB | mercestes, ok, I think I have to try tomorrow again |
| 18:51.10 | MaartenB | it is mostly bad during the day times |
| 18:51.12 | Bobthehunter | anyone kjnow default paswords for sp2100 ? |
| 18:51.18 | MaartenB | in the evenings it is okay again |
| 18:51.42 | [TK]D-Fender | Bobthehunter: "admin/[blank]"IIRC |
| 18:52.12 | mercestes | MaartenB: I suspect jitter. Could be a bad switch or router. |
| 18:52.15 | wunderkin | administrator/password, ha ha ha |
| 18:52.25 | MaartenB | mercestes, the problem is that my provider says everything is fine |
| 18:52.39 | MaartenB | mercestes, they handle more calls, obviously |
| 18:53.12 | mercestes | MaartenB: yea, AT&T told me that once too. |
| 18:53.44 | MaartenB | and? |
| 18:53.50 | mercestes | I made them cry |
| 18:53.55 | MaartenB | good :) |
| 18:54.04 | MaartenB | how did you manage to do that? :) |
| 18:54.11 | mercestes | :) |
| 18:54.14 | [TK]D-Fender | mercestes: NO SINGING! |
| 18:54.15 | mercestes | it's not IRC appropriate. |
| 18:55.23 | MaartenB | how can I solve this? go for another voip provider? |
| 18:55.28 | MaartenB | or is there an easier solution? |
| 18:55.51 | mercestes | MaartenB: Catch the problem happening and observe what is wrnog. It could be your end. |
| 18:55.53 | MaartenB | I would like to proof my voip provider the fault is on their site :) |
| 18:55.59 | mercestes | MaartenB: Could be a bad switch, a bad wire, or a bad router. |
| 18:56.12 | Bobthehunter | from the phone ivr |
| 18:56.15 | MaartenB | okay |
| 18:56.20 | mercestes | MaartenB: If all of those things are good and *your* network is stable, you show them jitter going from your site to theirs. |
| 18:56.33 | MaartenB | is iax different from sip? |
| 18:56.41 | cpm | err, yup |
| 18:56.56 | MaartenB | hmm. let me ask that in another way :) |
| 18:57.19 | [TK]D-Fender | MaartenB: If you have jitter you're likely jsut as screwed |
| 18:57.21 | MaartenB | I use sip to connect to my own * box, my * box is connected to my voip provider |
| 18:57.29 | mercestes | IAX still travels over your network in an IP to IP fashion, so it is still susceptible to jitter |
| 18:57.59 | MaartenB | if I make a sip call to another sip phone, everything is alright |
| 18:58.19 | mercestes | MaartenB: Is your asterisk box the same version as your provider's asterisk box? |
| 18:58.26 | MaartenB | it is only outgoing messages wich uses iax that fail |
| 18:58.29 | mercestes | MaartenB: there is some IAX to IAX compatibility issues as well |
| 18:58.33 | MaartenB | I have no idea |
| 18:58.47 | MaartenB | is there a way to check that? |
| 19:00.35 | MaartenB | anyway, is it correct that I only have to check the route my * <> voip gateway |
| 19:00.46 | MaartenB | or do I have to monitor the jitter betwee me and * too? |
| 19:00.46 | mercestes | MaartenB: Ask your provider |
| 19:00.53 | cpm | does thought pretty much everyone used iax2 these days |
| 19:01.13 | tzanger | what? |
| 19:01.24 | tzanger | I used to be a huge fan of IAX2 |
| 19:01.35 | tzanger | but then Olle's nat traversal stuff made it in |
| 19:01.40 | tzanger | and IAX2 got too experimental for me |
| 19:01.41 | MaartenB | I am running 1.4.0, I think my provider uses a previous version |
| 19:01.46 | MaartenB | but I mail them about them |
| 19:01.55 | MaartenB | it could explain my problem anyway |
| 19:02.07 | MaartenB | is there anything documented about this iax incompatibility? |
| 19:02.16 | tzanger | when I can plug in four IP501s behind a bog-standard Linksys WRT54 connected to an ADSL or cable modem and have them "just work" with my Asterisk box... there's no reason for IAX2 for me. :-) |
| 19:03.08 | *** join/#asterisk J4k3 (i=J4k3@dhcp-12-197-128-58.intrastar.net) |
| 19:04.09 | mercestes | MaartenB: Dunno. could google it. but IAX stuff changes from version to version and has sync issues. |
| 19:05.28 | *** join/#asterisk elriah (i=elriah@adsl-072-149-159-016.sip.bhm.bellsouth.net) |
| 19:06.07 | elriah | Hi all. On Polycom 501+ phones, how do I get asterisk to play it's "busy" message when a user is on a call on a specific extension? |
| 19:08.06 | *** join/#asterisk trelane_ (n=trelane@unaffiliated/trelane) |
| 19:11.37 | MaartenB | while debugging, I ran into chan_iax2.c:3792 iax2_send: No private structure for packet? |
| 19:11.40 | mercestes | elriah: I hear that ChanIsAvail() works, otherwise you can set the callsperlinekey in the polycom configs. |
| 19:11.44 | MaartenB | anyone know what that means? |
| 19:17.37 | *** join/#asterisk drega (n=drega@80-47-225-31.lond-th.dynamic.dial.as9105.com) |
| 19:20.55 | wunderkin | fucking damnit ugh |
| 19:21.16 | wunderkin | no more mass reboots during business hours, i have like 4 phones that did not come back up, after 13+ minutes |
| 19:25.38 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 19:26.18 | wunderkin | the ip430s did not like my changes, i see.. hmm.. i wonder why |
| 19:26.22 | *** join/#asterisk Schreiber1337 (i=cee4b403@gateway/web/cgi-irc/ircatwork.com/x-fd491028043e8159) |
| 19:27.00 | wunderkin | i tried changing the chord stuff so that there is no more voicemail beep on the speaker |
| 19:27.11 | *** join/#asterisk juancri (n=juancri@65.111.169.15) |
| 19:27.38 | juancri | hello I need some help with AGI.. I wnt to record the call... but in the background |
| 19:27.55 | juancri | I don't find any unction to do that |
| 19:28.03 | ManxPower | juancri: why do you need an AGI for that? |
| 19:28.19 | juancri | because I want to define the filename with sme parameters from a database |
| 19:28.40 | ManxPower | juancri: "show application monitor" |
| 19:29.00 | wunderkin | it worked on my ip430 and ip501 but none of their ip430s.. uhh ok |
| 19:29.05 | SheriF_SpacE | [TK]D-Fender: thx :-) |
| 19:29.21 | juancri | ManxPower is that a command? |
| 19:30.01 | SheriF_SpacE | i have another problem asterisk don'st make transcoding .... for the codes between gsm / ulaw / alaw ! i hear silnce i have to put the grandstream on one codecs :-s all of it :( |
| 19:30.20 | mercestes | juancri: It's an asterisk CLI command and I do exactly that without the use of an AGI> |
| 19:30.32 | juancri | ahm ok |
| 19:30.44 | juancri | I'll try that |
| 19:31.05 | juancri | but... there's not anyway to do that through AGI? |
| 19:31.06 | ManxPower | juancri: it is an asterisk application you would run just like any other asterisk application via AGI EXEC |
| 19:31.41 | juancri | aha ok... |
| 19:32.13 | *** join/#asterisk lba (n=lba@user-0ccsqlo.cable.mindspring.com) |
| 19:33.10 | juancri | thank you all :) |
| 19:36.40 | elriah | mercestes: Thanks! |
| 19:36.55 | mercestes | elriah: your welcome! ....what did I do? |
| 19:37.10 | elriah | Answered my question above ... |
| 19:37.17 | mercestes | oh. your welcome. |
| 19:37.26 | mercestes | :) |
| 19:37.49 | ManxPower | SheriF_SpacE: disallow=all and alllow= each of the codecs you want in sip.conf, but don't allow g729 or g723.1 |
| 19:39.02 | ManxPower | also I would not allow both ulaw and alaw, there is no reason to. allow the *law that is for your country. |
| 19:39.13 | ManxPower | ulaw for usa/canada and alaw for the rest of the world |
| 19:40.50 | J4k3 | what if you make lots of international calls? |
| 19:40.52 | cpatry | except japan?:) |
| 19:41.07 | J4k3 | ;) |
| 19:41.27 | J4k3 | no matter what codec I use, when I call japan all they speak is jibberish |
| 19:41.29 | J4k3 | ;) |
| 19:42.05 | *** join/#asterisk thinwires (n=thinwire@24-49-196-96.kntnny.adelphia.net) |
| 19:43.11 | ManxPower | J4k3: if you make international calls your codec to your provider does not change |
| 19:43.14 | *** join/#asterisk SomethingISODD (n=Dan@NTL208H101-91-124.nt.net) |
| 19:43.33 | SomethingISODD | good day all |
| 19:44.23 | wunderkin | i had to revert that change and manual power cycle to get them back up |
| 19:44.46 | wunderkin | i sware that place has a curse, i test everything at home on an ip430 and ip501 and they work fine, i use the same config as they do |
| 19:44.50 | *** join/#asterisk anthonyl (n=anthonyl@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 19:46.45 | *** join/#asterisk saftsack (n=oliver@pD9E068AA.dip.t-dialin.net) |
| 19:48.07 | *** join/#asterisk mega (n=mega@217.201.234.253) |
| 19:49.46 | *** join/#asterisk ltd (n=z@202-161-1-26.dyn.iinet.net.au) |
| 19:50.29 | *** join/#asterisk lenne_dk (n=Miranda@83.72.129.7.ip.tele2adsl.dk) |
| 19:51.21 | poppo | Who makes the best TTS Engine out there? |
| 19:52.26 | SwK[Work] | probably cepstral |
| 19:52.34 | SwK[Work] | for use w/ asterisk |
| 19:52.44 | SwK[Work] | anyone even tried using VSX 7000 (from polycom) w/ asteirsk? |
| 19:53.00 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 19:53.56 | thinwires | has anyone here used AsteriskNOW? |
| 19:54.01 | dansmith | with polycom phones, tone.dtmf.viaRtp=1 means inband, right? |
| 19:54.10 | [TK]D-Fender | SwK[Work]: I'm likely to soon. Head office wants to head that way for video conferencing |
| 19:54.32 | mercestes | I have to compile from source in gentoo. how sad. No emerge for me. |
| 19:55.54 | *** join/#asterisk Bobthehunter (n=Bobthehu@145-27.mc.cite.net) |
| 19:58.32 | thinwires | hey guys, if I switch over to an asterisk install is there any program that will give my a GUI to use? Or would I be stuck to CLI? |
| 19:59.06 | thekidrio | asterisk 1.4 has a gui i think |
| 19:59.17 | elriah | thinwires: Try freepbx or asteriskgui, but if you want real control, learn the (easy to use) command line... |
| 20:00.00 | thinwires | xok, yeah the CLI is ok for me, but there are people at my office that need gui-warm-fuzzy-feeling |
| 20:00.12 | thekidrio | hehe |
| 20:00.44 | trelane_ | nofuzzyforyou! |
| 20:00.50 | J4k3 | ewwie gui |
| 20:00.57 | thinwires | I'm going to try Astierisk 1.4, I have an AsteriskNOW build but I can't find any help with this problem so I'm just going to dump and burn |
| 20:03.21 | SomethingISODD | Hello all anyone here good with h323 and pwlib? |
| 20:04.42 | SomethingISODD | when compiling pwlib i keep getting this error http://www.pastebin.ca/366617 |
| 20:04.51 | *** part/#asterisk Juppers (i=imagine@phantom.blueone.net) |
| 20:07.35 | *** join/#asterisk xo8ox (n=pride_32@wsip-66-210-250-2.ph.ph.cox.net) |
| 20:07.48 | xo8ox | guys does asterisk 1.4 come with a GUI interface ? |
| 20:07.57 | Nugget | Not really. |
| 20:08.13 | xo8ox | can we add a gui to it though ? |
| 20:08.30 | thekidrio | i think it does nugget, atleast they say it does heh |
| 20:08.32 | Nugget | If you're afraid of text files then you should look into AsteriskNOW. |
| 20:08.52 | thekidrio | why do people hate gui heh |
| 20:08.53 | xo8ox | asterisknow is very very buggy |
| 20:09.00 | thekidrio | some pride thing i am guessing |
| 20:09.14 | Nugget | Because GUIs are inadequate for running and configuring Asterisk. |
| 20:09.49 | pigpen | Ok..I think I have a good one....with voice distortion with an iax trunk. |
| 20:09.51 | SplasPood | thekidrio: its because most (all?) control panels lock you into doing things THEIR way and no other |
| 20:10.35 | pigpen | Now and then we get voice chopping/distortion on an iax trunk between two systems both running 1.2.14 |
| 20:11.12 | xo8ox | where can I get freebpx from ? |
| 20:11.12 | *** join/#asterisk drega (n=drega@80-47-225-31.lond-th.dynamic.dial.as9105.com) |
| 20:11.25 | pigpen | topology: PRI -- Digium -- Asterisk --- 45MB Ethernet (WAN) -- Asterisk -- SIP -- Polycom 60x |
| 20:11.38 | Nugget | xo8ox: there's a really neat site called google.com |
| 20:11.52 | xo8ox | no so neat after all cuz i can't find it |
| 20:11.58 | pigpen | I see this when I run iax2 show netstats: |
| 20:11.58 | Nugget | Then you're a fucking moron. |
| 20:11.58 | pigpen | Channel RTT Jit Del Lost % Drop OOO Kpkts Jit Del Lost % Drop OOO Kpkts |
| 20:11.59 | pigpen | IAX2/path-16385 6 1 41 144 11 0 0 0 1 41 0 0 0 0 1 |
| 20:12.02 | JoNate | Nugget: I heard you need licenses to use that site... |
| 20:12.10 | thinwires | Nugget: there is no help for these problems, it's called beta |
| 20:12.28 | pigpen | now...how and the hell am I getting 144 lost packets? |
| 20:12.38 | pigpen | Low usage...line stats are great. |
| 20:12.48 | thinwires | 144 is a bit to much :-p |
| 20:13.03 | ryant | anyone using dundi internally in their company for routeing calls efficiently? |
| 20:13.15 | pigpen | it is right? I mean..I would think so for a $2000/per month line... |
| 20:13.37 | pigpen | the funny thing is that it jumps in bursts. |
| 20:13.38 | xo8ox | Nugget fuck you |
| 20:13.58 | Nugget | seriously, if you can't manage to use google you're doomed when it comes to VoIP. |
| 20:14.05 | Nugget | just hire someone. |
| 20:14.16 | pigpen | xo8ox, trixbox.org |
| 20:14.16 | xo8ox | get lost i'm done with u |
| 20:14.39 | Nugget | You're the one who couldn't find "freepbx.org" when looking for freepbx. |
| 20:14.43 | pigpen | xo8ox, freepbx.org |
| 20:14.53 | xo8ox | trixbox is just like asterisknow they install os,*,gui.. I just need a good working gui |
| 20:15.03 | Nugget | No such thing exists. |
| 20:15.05 | xo8ox | pigpen thank u |
| 20:15.12 | [TK]D-Fender | xo8ox: http://www.google.ca/search?hl=en&sa=X&oi=spell&resnum=0&ct=result&cd=1&q=FREE+PBX+HOME&spell=1 |
| 20:15.25 | [TK]D-Fender | xo8ox: How about the FIRST LINK. Yeah you tried REALLY hard.... |
| 20:15.59 | pigpen | xo8ox, and /join #freepbx ..they can help there. |
| 20:16.05 | *** join/#asterisk backblue (n=moo@87-196-47-206.net.novis.pt) |
| 20:16.07 | [TK]D-Fender | Nugget: Good. Working. GUI. Pick TWO ;) |
| 20:16.23 | wunderkin | even just freepbx works |
| 20:16.26 | *** part/#asterisk xo8ox (n=pride_32@wsip-66-210-250-2.ph.ph.cox.net) |
| 20:16.27 | pigpen | yeah...we wrote our own. |
| 20:16.44 | Nugget | heh |
| 20:16.54 | [TK]D-Fender | pigpen: 11% loss? EEK |
| 20:17.10 | pigpen | [TK]D-Fender, yeah....I don't get it! |
| 20:17.28 | pigpen | the line screems, no errors on any of the interfaces... |
| 20:17.39 | [TK]D-Fender | pigpen: With your amazing T3 connection you can lose pckets faster than any kid on the block! |
| 20:17.41 | pigpen | great latency. |
| 20:17.50 | pigpen | hahaha! |
| 20:17.53 | thinwires | lol comawn, it's only 11% of your conversation, that can't be important :-O |
| 20:18.00 | pigpen | that was good. |
| 20:18.17 | pigpen | the funny thing is that not all the calls are that bad...just now and then. |
| 20:18.21 | pigpen | even at the same time. |
| 20:18.24 | [TK]D-Fender | pigpen: Yeah, since most subject matter is only 5% useful, who'd have though your loss would hit something important ;) |
| 20:18.56 | pigpen | yeah. |
| 20:19.14 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 20:19.14 | pigpen | the same stats I posted I have: |
| 20:19.15 | pigpen | IAX2/path-16386 6 0 60 14 0 0 0 0 0 40 0 0 0 0 0 |
| 20:19.19 | pigpen | Sorry, no headder. |
| 20:19.25 | pigpen | sp? |
| 20:19.29 | thinwires | maybe someone in your office is using the interwebs to download torrents and pulls in the third floor of hell and gobbes all your traffic |
| 20:20.28 | pigpen | Well, the * box is the gateway to the DS3.... |
| 20:20.35 | pigpen | but still..the line usage is a joke. |
| 20:20.54 | thinwires | yeah it would be pretty hard to jam that baby up |
| 20:21.01 | pigpen | so maybe the 5min average stats I am grabbing from mrtg is useless in this case. |
| 20:21.42 | pigpen | (big pipe..things come down quick...) |
| 20:21.53 | pigpen | (theory) |
| 20:22.30 | thinwires | only thing I could think of looking for would be something that is causing your *box to freeze up... are there any patterns in the data loss? or is it just happening at random? |
| 20:23.12 | pigpen | not really. It is a Dell 6850, quad proc, 8GB ram....no real pauses... |
| 20:23.17 | *** join/#asterisk drega (n=dforeman@80-47-225-31.lond-th.dynamic.dial.as9105.com) |
| 20:23.18 | pigpen | runs great actually. |
| 20:25.46 | pigpen | Channel RTT Jit Del Lost % Drop OOO Kpkts Jit Del Lost % Drop OOO Kpkts |
| 20:25.46 | pigpen | IAX2/path-16386 6 0 47 24 0 0 0 0 0 40 0 0 0 0 0 |
| 20:25.52 | pigpen | ^^ at least no loss. |
| 20:26.04 | pigpen | sorry...low % |
| 20:27.09 | pigpen | I will elaborate...I just remembered a few things. |
| 20:28.04 | pigpen | 20 VPN's, 3 interfaces, Primary router/firewall |
| 20:28.05 | pigpen | :) |
| 20:28.26 | pigpen | It is passing traffic to an inside firewall, who is terminating 1008 vpns. |
| 20:28.27 | pigpen | :) |
| 20:28.52 | pigpen | so the iax traffic has to fight it |
| 20:29.03 | pigpen | so the iax traffic has to fight it's own traffic out the pipe. |
| 20:29.11 | pigpen | thoughts? |
| 20:29.52 | *** join/#asterisk saftsack (n=oliver@pD9E068AA.dip.t-dialin.net) |
| 20:29.55 | pigpen | Reverse direction: |
| 20:30.04 | pigpen | Channel RTT Jit Del Lost % Drop OOO Kpkts Jit Del Lost % Drop OOO Kpkts |
| 20:30.04 | pigpen | IAX2/path-6 6 0 40 0 0 0 0 0 0 48 24 0 0 0 17 |
| 20:30.36 | pigpen | So am I correct the loss is at the remote? ^^ |
| 20:33.13 | SomethingISODD | Hello all anyone here good with h323 and pwlib? when compiling pwlib i keep getting this error http://www.pastebin.ca/366617 |
| 20:33.31 | SomethingISODD | can anyone help plz |
| 20:41.00 | *** join/#asterisk backblue (n=moo@87-196-47-206.net.novis.pt) |
| 20:41.27 | *** join/#asterisk RoyK (n=roy@ti211310a080-9075.bb.online.no) |
| 20:47.19 | lba | Dialplan question. How do you force outgoing calls to go to an extensions.conf context? |
| 20:47.27 | *** join/#asterisk smace (n=smace@200.149.32.180) |
| 20:47.54 | lba | I have contexts with extens that never get called. |
| 20:47.56 | smace | My asterisk is running but it not working and configurations seems fine, logs seem ok. any idea? |
| 20:48.45 | [TK]D-Fender | lba: Its up to you have INCLUDE contexts properly and assign them to the appropriate devices. |
| 20:49.14 | [TK]D-Fender | smace: Since you gave no details about whats not working.... NO. |
| 20:49.39 | smace | I am not sure. I have no more ideas :( |
| 20:50.02 | lba | [TK]D-Fender: Well sip.conf handles incoming calls with the context=. How do I call other extens? I don't want to mix outgoing calls in with incoming context. |
| 20:50.04 | smace | i have installed asterisk and copied extensions.conf and sip.conf to the new server. |
| 20:50.33 | smace | i have asterisk -p it is running when I ps aux |
| 20:50.33 | [TK]D-Fender | smace: If you can't even describe the problem, go get some sleep and/or coffee and come back with a clue. |
| 20:51.22 | smace | maybe you are right. |
| 20:51.36 | [TK]D-Fender | lba: What do you mean "call other extens"? I think you need to re-read the bit about "include"ing other contexts for mixed access. And if you have contexts that are not linked to anything you should know what to do with them.... |
| 20:51.41 | lba | [TK]D-Fender: To amplify. I used to have context=default in sip.conf. Last night I discovered that my internal calls were in the default context and I want to separate them out. |
| 20:52.05 | [TK]D-Fender | lba: Go right ahead. You have my permission. |
| 20:52.37 | lba | [TK]D-Fender: I know about includes but I don't want to include them in [default]. extens in contexts that are not called don't get executed with an include to somewhere, but where should they go? |
| 20:53.21 | [TK]D-Fender | lba: Maybe you shouldn |
| 20:53.34 | [TK]D-Fender | 't be assigning "default" to all of your devices then... |
| 20:53.47 | [TK]D-Fender | lba: Go grab some coffee and rethink your heirarchy |
| 20:54.17 | *** join/#asterisk flenders (n=fserto@unaffiliated/flenders) |
| 20:54.21 | edison\ | okay, this is bugging me |
| 20:54.23 | lba | [TK]D-Fender: In this case I need a clue. Can I have two context lines in a sip device? |
| 20:54.36 | *** join/#asterisk prikolus (n=PircBot@84.94.111.109.cable.012.net.il) |
| 20:54.51 | [TK]D-Fender | lba: Small tip : you should never name a context [default] or [general]. Seriously, what the hell does that say about its purpose or priveledges? |
| 20:54.57 | edison\ | how the heck do you check to see if a variable is set to something other than null |
| 20:54.58 | *** join/#asterisk pjdaley_ (n=guiguigu@208.180.232.232) |
| 20:54.59 | [TK]D-Fender | lba: No. |
| 20:55.09 | *** join/#asterisk jcool (n=zoro@58.69.18.66) |
| 20:55.18 | [TK]D-Fender | edison\: "show application gotoif" |
| 20:55.20 | edison\ | in the dialplan, that is. |
| 20:55.27 | [TK]D-Fender | edison\: And read up on * expressions. |
| 20:55.35 | smace | the X-Lite here is not registering in Asterisk server, I am sure the * server is running one it appears in "ps aux" but It does not show anything in logs. X-Lite reports: "Registration error 408 - Request Timeout" |
| 20:55.38 | lba | [TK]D-Fender: I agree about using default and am in the process of changing it to something meaningful. Thats when I found out that my outbound calls were in there. |
| 20:55.46 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 20:55.53 | [TK]D-Fender | edison\: http://www.voip-info.org/wiki/view/Asterisk+Expressions |
| 20:55.59 | pjdaley_ | where can I find the hardware that is supported by asterisk, or the hardware that is most reccomended? |
| 20:56.18 | edison\ | [TK]D-Fender I have read up. And IF() would be that I want, but it's not working as expected. |
| 20:56.32 | jcool | Hello good day guys, can i ask a question please, i have been task to integrate legacy pbx via FXO interface alcatel 4400 to be exact |
| 20:56.37 | [TK]D-Fender | smace: Likely a networking error between the 2. Could be NAT. Could be a wrong port #. et |
| 20:56.49 | smace | yes, there is NAT |
| 20:56.55 | [TK]D-Fender | edison\: Perhaps you could actually show us your attempt |
| 20:56.58 | smace | but asterisk server has a valid ip. |
| 20:57.11 | smace | I am pinging there. |
| 20:57.16 | jcool | is there a way to get a prefix number send by alcatel for proper routing ? |
| 20:57.19 | [TK]D-Fender | smace: Then I'd go double-check all your NAT settings, both on *'s side, and in the peer setup |
| 20:57.30 | edison\ | noop(IF(${FOO}?bar:baz)) works if FOO is a non-null value, but breaks if FOO is empty |
| 20:57.50 | [TK]D-Fender | edison\: that is NOT an expression. go read the link I gave you. |
| 20:57.53 | edison\ | complaints about wrong syntax amount |
| 20:58.16 | edison\ | expressions are in $[]. Is that what you're getting at? |
| 20:58.24 | [TK]D-Fender | edison\: Nor ar you calling that FUNCTION properly |
| 20:58.35 | [TK]D-Fender | edison\: Thats only the beginning.... |
| 20:58.58 | [TK]D-Fender | edison\: Go read that link.... |
| 20:59.16 | edison\ | please tell me what you want me to see on that page. |
| 20:59.19 | edison\ | I'm already reading it. |
| 20:59.25 | [TK]D-Fender | smace: Pinging doesn't mean ANYTHING. go check all the sip.conf setting you need for NAT to function. |
| 20:59.27 | edison\ | I've been using it as a reference. |
| 20:59.55 | smace | [TK]D-Fender, ill get a valid ip for my notebook. for testing. |
| 21:00.24 | *** join/#asterisk prikolus (n=PircBot@84.94.111.109.cable.012.net.il) |
| 21:02.38 | thinwires | lol ask again, your going to get a better response |
| 21:02.45 | thinwires | oops, wrong window, ignore that |
| 21:02.52 | edison\ | [TK]D-Fender "condition" is just a string. If the string is empty or "0", the condition is considered to be false, if it's anything else, the condition is true. |
| 21:03.28 | [TK]D-Fender | edison\: And guess what. if its BLANK (null), you're DOA. |
| 21:03.53 | [TK]D-Fender | edison\: * will see the "?" and nothing on the left hand side and bomb |
| 21:03.55 | edison\ | doesn't that equate to a empty string? |
| 21:04.00 | edison\ | right. |
| 21:04.04 | MarkWD | Heres a new question... what would you say is the best TISP in the market? |
| 21:04.05 | edison\ | that's what's happening. |
| 21:04.09 | edison\ | I see that. |
| 21:04.15 | [TK]D-Fender | edison\: Then we fall back to the problem of your not issuing an expression in the proper format. |
| 21:05.15 | edison\ | what's the right format, then? |
| 21:05.25 | *** join/#asterisk wonko2 (n=wonko@static-70-104-30-172.dllstx.fios.verizon.net) |
| 21:05.37 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 21:05.56 | [TK]D-Fender | edison\: http://www.voip-info.org/wiki/index.php?page=Asterisk%20Expressions |
| 21:06.16 | [TK]D-Fender | edison\: You could TRY looking on the WIKI you know.... I'm basiclly a human proxy at this point.... |
| 21:06.48 | edison\ | I'm sorry. I thought you had an idea as to what I would need to do |
| 21:06.50 | ManxPower | And don't use [TK]D-Fender as an SSL proxy, the performance sucks! |
| 21:06.54 | [TK]D-Fender | edison\: And this is now the second time I've linked it and you clain to be reading it..... |
| 21:06.57 | wonko2 | is there every any chance of Asterisk ever nicely building on non-linux and/or non-gcc? |
| 21:07.06 | edison\ | olklacajjkfm,fsfxf |
| 21:07.19 | [TK]D-Fender | ManxPower: No more cheap shots about my performce...... BITCH ;) |
| 21:07.20 | edison\ | obviously I'm missing something and therefore need some further help |
| 21:07.22 | [TK]D-Fender | huzzah! |
| 21:07.25 | edison\ | that's why I'm here. |
| 21:07.42 | ManxPower | wonko2: ASTERISK builds on non-linux currently |
| 21:07.48 | [TK]D-Fender | ManxPower: Oh, and you've been lined up for a "best ITSP" commentary just a few lines up.... |
| 21:07.55 | edison\ | so if someone would be so kind as to tell me WHAT I'm missing, I'll fix things and happily go away |
| 21:07.57 | wonko2 | with a lot of hand-holding, yes. :) |
| 21:08.22 | edison\ | But to continue to say, "Your[sic] missing something. Go find out what." isn't helping. |
| 21:08.28 | *** join/#asterisk tonycarstens (n=tony@206.135.21.162) |
| 21:08.31 | ManxPower | The Nortel T-1 card we ordered arrived today! |
| 21:08.38 | [TK]D-Fender | edison\: Perhaps you could show me how you've changed your attempt to better reflect what the WIKI told you was needed for proper formatting... |
| 21:08.59 | edison\ | perhaps you could show me WHAT I need to change? |
| 21:09.12 | edison\ | $[FOO] doesn't work as the expr in IF() |
| 21:09.20 | edison\ | $[${FOO}] doesn't work. |
| 21:09.22 | [TK]D-Fender | edison\: I'd like to see how what you've read has impacted the way you first showed me you tried... |
| 21:09.47 | [TK]D-Fender | edison\: an expression needs an OPERATOR. You know a COMPARISON? like " |
| 21:09.48 | edison\ | I've already read that page a dozen times |
| 21:09.52 | tonycarstens | hello all, i've encountered a problem with the grandstream BT-100 phones, I have SIP configured correctly and I can call the phones from the console but through the phone there is no dial tone, I have scoured the internet for a few days with no luck |
| 21:09.59 | *** join/#asterisk CunningPike (n=CunningP@204.239.12.189) |
| 21:09.59 | edison\ | ah |
| 21:10.02 | edison\ | Okay. |
| 21:10.16 | [TK]D-Fender | $["${variable}"="SHIT"] |
| 21:10.44 | ManxPower | does 1.2 still require spaces around the = ? |
| 21:10.51 | jcool | [TK]D-Fender: can i ask a single question please, in sip you get get the right prefix of incoming sip call for proper routing by exten => 999121,Goto(ivr,s,1) right |
| 21:10.56 | [TK]D-Fender | edison\: And heck they even had an entire SECTION dedicated to checking for null entries! |
| 21:11.03 | Corydon-w | ManxPower: 1.2 does not |
| 21:11.18 | jcool | [TK]D-Fender: do you think in TDM not PRI can i do the same thing ? |
| 21:11.19 | edison\ | right, but only using GotoIf. |
| 21:11.21 | [TK]D-Fender | jcool: HUH!?!? |
| 21:11.45 | [TK]D-Fender | edison\: IF/GotoIF both use the same expression formatting.... |
| 21:12.09 | ManxPower | jcool: once a call gets to extensions.conf it's just a call, no real difference in how the call got there, it's still just a call |
| 21:12.19 | Corydon-w | Actually, it's the expression parser which does that. GotoIf and IF know nothing of the expression parsing syntax |
| 21:12.27 | pjdaley_ | where can I find the hardware list that is supported by asterisk, or the hardware that is most reccomended? |
| 21:12.35 | [TK]D-Fender | edison\: Did you lookup the "IF" function's WIKI page? |
| 21:12.40 | edison\ | yep |
| 21:13.10 | edison\ | Please excuse my ignorance. It's not intentional. |
| 21:13.11 | [TK]D-Fender | CorydonYes, duly noted, was looking from the "simplify for the masses POV, but you are correct. |
| 21:13.14 | jcool | ManxPower: you mean to say in SIP that's possible on TDM call it doesn't right ? |
| 21:13.28 | [TK]D-Fender | edison\: http://www.voip-info.org/wiki/index.php?page=Asterisk%20func%20if |
| 21:13.35 | tonycarstens | does anyone have any expierence with grandstream budge phones? |
| 21:13.47 | [TK]D-Fender | edison\: And from there they give a very explicit example : exten => s,1,Set(foo=${IF($[ ${x} = 7]?tval:fval)}) |
| 21:13.57 | edison\ | sure. |
| 21:14.02 | edison\ | I'm using IF() in a number of other places |
| 21:14.03 | [TK]D-Fender | edison\: Which should give you a clue about how functions are to be called. |
| 21:14.46 | [TK]D-Fender | <edison\> noop(IF(${FOO}?bar:baz)) works if FOO is a non-null value, but breaks if FOO is empty |
| 21:14.50 | ManxPower | jcool: it is not a "sip call", it is a call |
| 21:15.13 | jcool | ManxPower: all right i understand thanks man |
| 21:15.19 | *** part/#asterisk [shodan] (n=shodan@ip117.96-113-216.pppoe1.joliette.intermonde.net) |
| 21:15.45 | [TK]D-Fender | <edison\> Just to speed this along..... noop(${IF($["${FOO}"=""]?bar:baz)) |
| 21:15.53 | edison\ | And the syntax for expressions is: $[ expr1 operator expr2 ] |
| 21:16.09 | edison\ | what's expr1? Expression1? |
| 21:16.24 | [TK]D-Fender | edison\: Expression 1. Like A = B. |
| 21:16.30 | jcool | ManxPower: last question if not that so much to ask, if i'm going to implement a legacy pbx to * you mean to say, there's no way from their phone to transfer a call to a sip extension without going to IVR ? |
| 21:16.45 | edison\ | okay, so expressions contain expressions? |
| 21:16.45 | [TK]D-Fender | edison\: "A" is one expression, "B" is another, and the operator (test) is "=" |
| 21:16.53 | edison\ | Right. |
| 21:16.57 | edison\ | I understand that. |
| 21:17.12 | [TK]D-Fender | edison\: Sure, you can nest expressions. |
| 21:17.34 | [TK]D-Fender | edison\: Its rare to find a reason to, but you can. |
| 21:17.43 | *** join/#asterisk fr33bi3 (n=fr33bi3@125.22.65.165) |
| 21:17.46 | edison\ | I extrapolated the A-type expression to be valid as the <expr> in "IF(<expr>?[<true>][:<false>])" |
| 21:17.50 | [TK]D-Fender | edison\: As long as you keep in mind what the RESULT of an expression will be. |
| 21:18.25 | [TK]D-Fender | edison\: Actaully looking at that, the term WAS used somewhat redundently. |
| 21:18.31 | edison\ | so therefore if A is an <expr> then IF(A?<true>:<false>) |
| 21:18.36 | edison\ | okay. |
| 21:18.36 | [TK]D-Fender | edison\: Do you have any programming experience? |
| 21:18.40 | edison\ | I'm not losing my head |
| 21:18.42 | edison\ | sure. tons |
| 21:18.49 | edison\ | that's what's frustrating |
| 21:18.51 | wonko2 | is there a way to tell make install to use bash instead of /bin/sh? |
| 21:19.00 | [TK]D-Fender | edison\: This sort of thing really should have been apparent, esp with the samples :) |
| 21:19.10 | edison\ | I use if($var){do something} else {do soemthing else} all the time |
| 21:19.29 | [TK]D-Fender | edison\: Read the sample code and description of the output, and ignore the rest. Reverse-engineer=success! |
| 21:19.34 | ManxPower | edison\: you are using AEL? |
| 21:19.41 | edison\ | if $var exists and is not null, then the first branch is taken. |
| 21:19.48 | edison\ | AEL? no. |
| 21:19.56 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 21:20.05 | edison\ | C, perl, PHP, name it. |
| 21:20.06 | ManxPower | Well then you can do if($var){do something} else {do soemthing else} in asterisk |
| 21:20.07 | [TK]D-Fender | edison\: Like I said, if VAR is blank * wil bitch at you, so its not safe to use on its own. do a test like the one I showed you |
| 21:20.12 | thekidrio | fortran |
| 21:20.13 | thekidrio | cobol |
| 21:20.16 | ManxPower | .e.r.. you CAN'T do |
| 21:20.17 | thekidrio | pascal? |
| 21:20.33 | ManxPower | extensions.conf is not a programming language. it is a hacked togather bastardized config file. |
| 21:20.41 | thekidrio | haha |
| 21:20.42 | edison\ | thekidrio heh. back in the day on my Apple ][ |
| 21:20.56 | thekidrio | hehe then i bet you are frustrated |
| 21:21.02 | thekidrio | i hate when simple stuff stumps me |
| 21:21.16 | [TK]D-Fender | ManxPower: Some of the syntax is at least close enough that anyone with programming experience should not get ulcers |
| 21:21.16 | *** join/#asterisk BPJ (n=jan@rrcs-67-78-88-186.sw.biz.rr.com) |
| 21:21.21 | ManxPower | if you insist on treating extensons.conf as a programming or scripting language you will eventually grow to hate asterisk and all it stands for. |
| 21:21.36 | *** join/#asterisk trixman (n=andy@rrcs-67-53-168-147.west.biz.rr.com) |
| 21:21.43 | thekidrio | i am having a bitch of a time dialing two numbers and patching them together |
| 21:21.45 | trixman | anybody familiar with trixbox |
| 21:21.49 | edison\ | [TK]D-Fender: the IF($[foo${FOO} != foo]?bar:baz) will work |
| 21:21.50 | Qwell[] | trixman: see topic |
| 21:21.51 | *** join/#asterisk tsurko (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 21:22.00 | thekidrio | trixman go to freepbx for trix help |
| 21:22.03 | ManxPower | trixman: I'll be the trixbox channel has people damiliar with tricvox |
| 21:22.04 | trixman | ok |
| 21:22.06 | tonycarstens | dialtone issues PLEASE HELP |
| 21:22.21 | edison\ | ManxPower I just like doing things procedurally if at all possible |
| 21:22.28 | [TK]D-Fender | edison\: Avoid whitespace in your expressions, and I'd suggest the double ended quotes for cleanliness sake |
| 21:22.59 | bcnl | is there a easy way to have a callee be prompted to enter a series of digits before the call is connected to them? |
| 21:23.00 | thekidrio | [TK]D-Fender: dont you need white space between the test? |
| 21:23.10 | thekidrio | i know that 3+1 != 3 + 1 |
| 21:23.11 | [TK]D-Fender | edison\: Extensions.conf is not a place to try making thisng pretty.... it will kill things.... |
| 21:23.18 | bcnl | i want to do this within a Dial() with multiple recipiants though |
| 21:23.18 | edison\ | [TK]D-Fender: would $['${FOO}!='] work, or is a single quote special? |
| 21:24.33 | ManxPower | edison\: why not just use the LEN function? |
| 21:24.34 | [TK]D-Fender | edison\: :encapsulated wrong... |
| 21:24.34 | edison\ | ManxPower: that's what I'm using now, but it's long and ugly. |
| 21:24.34 | [TK]D-Fender | edison\: $['${FOO}"!="'] |
| 21:24.34 | ManxPower | exten => s,3,GotoIf($[${LEN(${FAX_DEST})} = 0]?9:4) |
| 21:24.34 | [TK]D-Fender | errr... |
| 21:24.34 | edison\ | yes, I know it works and I shouldn't complain, but I *like* pretty things |
| 21:24.34 | ManxPower | edison\: it is the best way |
| 21:24.34 | [TK]D-Fender | edison\: $['${FOO}'!=''] |
| 21:24.34 | ManxPower | edison\: there is no such thing as a pretty extensions.conf. |
| 21:24.45 | [TK]D-Fender | ManxPower: Yeah, LEN is a decent way |
| 21:24.50 | edison\ | actually, ${LEN(${FOO})} by itself works since LEN() will always return a value |
| 21:25.09 | *** join/#asterisk ChicagoBud (n=Bud@adsl-70-228-35-78.dsl.chcgil.ameritech.net) |
| 21:25.12 | [TK]D-Fender | edison\: Intelligent dirty trick. You're learning :) |
| 21:25.15 | thinwires | I've found that line in the voicemail.conf before but I never thought to try to -i because the dude said sendmail doesn't support the -i |
| 21:25.24 | thinwires | garr, wrong chat again.. |
| 21:25.35 | edison\ | Heh. I had that *before* I came in here asking questions. |
| 21:25.50 | edison\ | I probably should have said, "is there a better way than..." |
| 21:27.10 | [TK]D-Fender | edison\: Hindsight is 20/20..... |
| 21:27.48 | edison\ | yeah, but you frustrated me with the "you're missing something" bit. I already knew I was missing something. |
| 21:27.57 | edison\ | I just didn't know *WHAT* I was missing. |
| 21:28.02 | tonycarstens | no dialtone, can ring phone through SIP from ast console... anybody have any ideas? |
| 21:28.18 | ManxPower | tonycarstens: what does not have a dialtone? |
| 21:28.48 | tonycarstens | my ip phone |
| 21:28.55 | tonycarstens | its a grandstream bt100 |
| 21:29.03 | [TK]D-Fender | edison\: When in doubt look at brackets/braces & $'s first, then move on to quote marks, and other seperators :) |
| 21:29.05 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-154-6-99.red.bezeqint.net) |
| 21:29.05 | tonycarstens | i've tried every combo of options |
| 21:29.09 | r0d3nt | http://consumerist.com/consumer/vonage/verizon-sues-to-shut-down-vonage-238460.php |
| 21:29.20 | r0d3nt | whatcha guys & gals think about that ?? |
| 21:29.32 | jcool | tonycarstens: could you check if your phone was already registered? |
| 21:29.37 | r0d3nt | that would be pretty shitty for the entire industry. =( |
| 21:29.39 | [TK]D-Fender | edison\: Just compare against another stament, even if it is pretty different the key factors of variable references, etc, announce themselves pretty fast |
| 21:29.43 | ManxPower | tonycarstens: On SIP devices Asterisk does not even know the phone goes off hook, the phone does everything up to and including sending the whole number to asterisk before asterisk knows anything. So the phone should be what is generating the dialtone |
| 21:30.14 | [TK]D-Fender | r0d3nt: EXCELLENT. No more Vongae morons coming in here trying to learn * in 10s or less :) |
| 21:30.37 | r0d3nt | ahahhAh |
| 21:30.39 | r0d3nt | yeah =) |
| 21:30.53 | r0d3nt | but one of their claims is they have patented call forwarding.... :-/ |
| 21:30.54 | J4k3 | ok |
| 21:31.01 | J4k3 | what exactly is the patent suit trying to cover? |
| 21:31.08 | r0d3nt | i think if verizon wins this, they'd go after everyone else offering telco grade services... |
| 21:31.08 | J4k3 | are all other ITSP's going to get nailed by the same nonsense? |
| 21:31.15 | J4k3 | well |
| 21:31.23 | [TK]D-Fender | ok, heading home, BBIAB |
| 21:31.25 | JoNate | thats classic...its from my local paper... |
| 21:31.27 | J4k3 | it'll create a huge ridge between the PSTN (Verizon) and everything else. |
| 21:31.34 | r0d3nt | Verizon claims have been violated cover, among other things, the "gateway interfaces" . critical for providing VoIP phone services that mimic traditional phone service . fraud detection, billing and features such as call-forwarding. |
| 21:31.51 | J4k3 | uh |
| 21:32.03 | r0d3nt | yeah |
| 21:32.04 | r0d3nt | it's stupid. |
| 21:32.05 | pigpen | ManxPower, using iax2 show netstats, would 100 lost packets across 172 Kpkts be bad? |
| 21:32.11 | J4k3 | Verizon doesn't even own that |
| 21:32.16 | J4k3 | GTE doesn't own the phone switch, at all. |
| 21:32.43 | *** join/#asterisk gatuno (n=gatuno@145.red-82-158-215.user.auna.net) |
| 21:32.43 | r0d3nt | well.. take a look @ this lawsuit.. cuz it'll affect us all in a very bad way.... |
| 21:32.47 | r0d3nt | if they win.... |
| 21:32.49 | J4k3 | yeah |
| 21:32.50 | J4k3 | well |
| 21:32.52 | J4k3 | yes and no |
| 21:33.00 | r0d3nt | cause if vonage goes down, all voip providers and software makers go too |
| 21:33.01 | J4k3 | maybe it'll affect the general public in the "fuck the phone company" sort of way |
| 21:33.04 | J4k3 | which is about 50 years late in the USA. |
| 21:33.08 | r0d3nt | yeah sure. |
| 21:33.13 | J4k3 | I dunno |
| 21:33.16 | r0d3nt | most people aren't intelligent enough to know they're getting fucked. |
| 21:33.19 | tonycarstens | manxpower: sorry my boss came in and i had to update him on this asterisk project, so it is the phone that is the problem? |
| 21:33.26 | J4k3 | the average american is starting to notice the rest of the world ran past them, even urban america is behind. |
| 21:33.27 | r0d3nt | but anyways.. thanks for the thoughts... |
| 21:33.29 | r0d3nt | =) |
| 21:33.31 | J4k3 | rural is rural, no matter where you are. |
| 21:33.33 | r0d3nt | i'll be watching it.... |
| 21:34.25 | ManxPower | tonycarstens: the phone is where I would start looking. Just remember that Grandstream phones are some of the worst in the entire industry. Their firmware is buggier that Windows 1.0 and their support is on par with the airlines. |
| 21:34.50 | tonycarstens | jcool: are you talking about sip show peers? |
| 21:34.56 | J4k3 | thinwires: memtest86 is your friend, too. |
| 21:35.04 | tonycarstens | manx: i've got that impression the last few days |
| 21:35.05 | J4k3 | use it while you aquire a new optical drive. |
| 21:35.16 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 21:35.25 | jcool | tonycarstens: yes |
| 21:35.46 | tonycarstens | yes its registered to the right user and extension with the right IP |
| 21:35.59 | jcool | jcool: so it means the problem is your phone :) |
| 21:36.04 | tonycarstens | i can dial the phone through the console |
| 21:36.09 | thinwires | naw, i'm sure it's the cd rom going out the window, it can't read the CD's good... it hasn't finished the install yet.... it's just that this thing is an old NetFinitiy from IBM, cd-rom's are a bit more difficult |
| 21:36.20 | jcool | tonycarstens: how about dialing to PSTN or other sip extension?? |
| 21:36.59 | tonycarstens | well right now i don't have PSTN setup because I am just trying to get the network phones going |
| 21:37.29 | ManxPower | tonycarstens: update your phone firmeware |
| 21:37.34 | tonycarstens | the company i work for is opening a new place and they don't even have phone lines installed yet, all i'm trying to do is network calls |
| 21:37.36 | jcool | tonycarstens: first what you can do is to get a softphone if your softphone works then |
| 21:37.40 | ManxPower | GS frequently ships nonworking firmware on their phones |
| 21:38.20 | jcool | tonycarstens: * has nothing to do with it right, then try to isolate the problem on the phone itself |
| 21:38.23 | tonycarstens | manx: i'll try that |
| 21:38.36 | *** part/#asterisk wonko2 (n=wonko@static-70-104-30-172.dllstx.fios.verizon.net) |
| 21:42.36 | *** join/#asterisk s1gny|wrk (n=s1gny@p54915119.dip.t-dialin.net) |
| 21:42.56 | *** part/#asterisk s1gny|wrk (n=s1gny@p54915119.dip.t-dialin.net) |
| 21:44.28 | BrianR___ | Is there a way to dump all of the variables for a channel from the CLI? |
| 21:44.30 | *** join/#asterisk Telemac (n=cchantep@ANantes-256-1-59-75.w86-199.abo.wanadoo.fr) |
| 21:44.35 | Telemac | Hello |
| 21:44.51 | tonycarstens | jcool/manx: i tried a brand new phone and i got a dial tone but as soon as I configure it for sip it goes away |
| 21:46.39 | jcool | tonycarstens: what do you mean it goes away? |
| 21:46.45 | *** join/#asterisk hematitec (n=cratz@adsl-71-159-206-4.dsl.pltn13.sbcglobal.net) |
| 21:46.52 | tonycarstens | on the console it says Registration from '<sip:wes@10.4.0.21>' failed for '10.4.0.40' - no matching peer found |
| 21:47.08 | bkruse | tonycarstens: is that not obvious? |
| 21:47.17 | bkruse | in sip/users.conf do you have a [wes] in there? |
| 21:47.33 | bkruse | its trying to match a peer from in there.... if you dont have [wes] |
| 21:47.39 | tonycarstens | no i don't |
| 21:47.43 | tonycarstens | im an idiot |
| 21:47.45 | *** join/#asterisk netsurfer (n=bbjunkie@user-514d1e74.l3.c1.dsl.pol.co.uk) |
| 21:47.56 | tonycarstens | i've never seen anything about users.conf in all the docs i have been reading |
| 21:48.07 | jcool | tonycarstens: could you pastebin your sip.conf ? please go to pastebin.ca |
| 21:50.52 | Telemac | What should I add in extensions.conf so that when a phone doesn't answer after 15sec the call go to voicemail ? I've tried just by adding the timeout parameter to Dial but it goes directly to voicemail |
| 21:50.52 | jcool | Telemac: e.g exten => 200,1,Dial(Sip/200|15) |
| 21:50.52 | tonycarstens | jcool: sorry im not using the same machine that * is installed on |
| 21:50.52 | jcool | Telemac: exten => 200,n,Voicemail(200|u) something like that |
| 21:50.59 | tonycarstens | [wes] |
| 21:51.06 | jcool | tonycarstens: just get a putty ssh client then |
| 21:51.16 | jcool | tonycarstens: ok pm me |
| 21:51.35 | jcool | tonycarstens: then pastes your sip.conf there not here. |
| 21:51.45 | tonycarstens | yeah that was an accidnet |
| 21:51.57 | thekidrio | anyone here use nerd vittles click 2 dial? |
| 21:51.58 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 21:52.10 | [hC] | Anyone know why i might be seeing this on some g729 audio files? |
| 21:52.11 | [hC] | format_g729.c:150 g729_read: Short read (8) (Interrupted system call)! |
| 21:52.30 | [hC] | Is this indicative of a transcoding error where the EOF was not done properly? |
| 21:53.23 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 21:59.58 | [hC] | Can I use asterisk to transcode a wav to a g729 audio file? This GX::Transcoder seems to be dumping garbage at the end of my files or something. |
| 22:00.24 | Schreiber1337 | [TK]D-Fender: 3 hours later after a complete reload, everything is working great. Thanks for all your help a few hours ago.. |
| 22:00.40 | [TK]D-Fender | Schreiber1337 : np |
| 22:02.34 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 22:04.04 | BrianR___ | The default settings for pridialplan/prilocaldialplan are retarded... |
| 22:04.38 | *** join/#asterisk xF|DarkSC (n=dark@server.vinylkind.de) |
| 22:04.58 | BPJ | is there a less flaky provider than TELIAX? |
| 22:05.00 | ManxPower | BrianR___: don't use any of the pri dialplan stuff unless you need them |
| 22:05.26 | ManxPower | [hC]: if you have g729 license it should |
| 22:05.42 | [hC] | ManxPower: yes it does it automatically, but i want to convert and store once, so i dont have to transcode every time. |
| 22:05.51 | ManxPower | [hC]: ah |
| 22:06.04 | *** join/#asterisk Zaw (i=zaw@unaffiliated/zaw) |
| 22:08.30 | |ryan| | Can anyone suguest a place to get a low monthly fee toll-free DID? |
| 22:10.06 | mercestes | <PROTECTED> |
| 22:11.57 | *** join/#asterisk [[blah]asfd (n=ckwall@63.149.122.93) |
| 22:12.15 | |ryan| | mercestes: They'll send it to SIP or IAX2? Or just forward to another PSTN number? |
| 22:12.41 | *** part/#asterisk thinwires (n=thinwire@24-49-196-96.kntnny.adelphia.net) |
| 22:12.49 | [[blah]asfd | is there any way collect data historically from meetme, like what rooms have been used and for how long rather than doing it from the CLI>? |
| 22:13.17 | *** join/#asterisk Mavvie (n=edwin@ppp252-220.lns3.syd6.internode.on.net) |
| 22:15.56 | cpatry | [[blah]asfd: log files? |
| 22:16.48 | [[blah]asfd | hmmm.. would want something more intuitive. I am trying to build a billing app for my company |
| 22:18.36 | [TK]D-Fender | [[blah]asfd Get coding... contributions are welcome I'm sure. |
| 22:20.22 | flenders | holy crap! sip providers are a pain in the ass! |
| 22:20.27 | thekidrio | heh |
| 22:20.45 | flenders | my * box has a dedicated adsl2 line for voip, so on my router, I have rules to send all traffic to and from * through that pipe |
| 22:21.01 | flenders | 3 of my sip providers work, but 1 doesn't |
| 22:21.40 | flenders | if I create an extra rule on the router to send traffic to/from that sip gateway through our main gateway (different pipe), it works |
| 22:21.58 | *** join/#asterisk X-Rob (n=Rob@CPE-58-169-113-201.vic.bigpond.net.au) |
| 22:22.14 | flenders | so now 3 of my sip accounts are running through one pipe, and the other one on the pipe with all our internet traffic |
| 22:22.46 | flenders | great eh? |
| 22:23.22 | *** join/#asterisk j0 (n=dan@S01060016b6b541d2.va.shawcable.net) |
| 22:25.14 | *** join/#asterisk Pepse (n=pepse@ip68-109-169-37.ph.ph.cox.net) |
| 22:29.41 | *** join/#asterisk netik (n=jna@204-16-159-132-static.ipnetworksinc.net) |
| 22:29.44 | netik | hello |
| 22:29.51 | netik | anyone here using merlin magix with asterisk? |
| 22:30.07 | netik | I'm trying to figure out which DCD card I should purchase. |
| 22:31.26 | netik | hello? |
| 22:32.40 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 22:33.28 | *** join/#asterisk drako (n=ljd@unaffiliated/luisjose) |
| 22:37.19 | *** join/#asterisk tonycarstens (n=tony@206.135.21.162) |
| 22:37.55 | *** join/#asterisk darken_darken (n=marco@250.151.76.83.cust.bluewin.ch) |
| 22:43.20 | *** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net) |
| 22:48.50 | *** join/#asterisk diclophis-work (n=jbardin@65.203.37.58) |
| 22:48.57 | diclophis-work | hello all |
| 22:48.59 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 22:49.13 | diclophis-work | in my "timeout" extension, how do i prevent it from falling into a infinite loop? |
| 22:51.04 | [TK]D-Fender | diclophis-work : Show us what you're doing now. |
| 22:51.15 | diclophis-work | exten => t,1,Goto(1) |
| 22:51.22 | diclophis-work | thats it for the "timeout" extension |
| 22:51.32 | diclophis-work | i noticed for somereason a call got "trapped" in there |
| 22:51.41 | diclophis-work | like it was supposed to be hung up, but it wasnt |
| 22:52.08 | diclophis-work | i would like to loop the main menu, say 3 times, then hangup |
| 22:52.24 | thekidrio | you could do a count |
| 22:52.38 | thekidrio | start at 1 play main menu then increment count |
| 22:52.48 | thekidrio | and if count is less then target ideal goto play |
| 22:54.54 | thekidrio | exten => 345,1,Set(COUNT=0) |
| 22:56.38 | thekidrio | exten => 345,2,GotoIf($[{COUNT} < 3]?:10) |
| 22:57.01 | *** join/#asterisk vlt (n=dm@p54B31E83.dip0.t-ipconnect.de) |
| 22:57.03 | *** join/#asterisk Bobthehunter (n=Bobthehu@145-27.mc.cite.net) |
| 22:57.03 | diclophis-work | how would i increment? |
| 22:57.13 | diclophis-work | Set(COUNT=${COUNT}+1) ? |
| 22:57.13 | Bobthehunter | any implementation ov video h323 up and stable ? |
| 22:57.21 | Bobthehunter | http://www.inaccessnetworks.com/projects/asterisk-oh323 |
| 22:57.25 | Bobthehunter | checkign this out but not sure |
| 22:57.32 | thekidrio | almost diclophis-work |
| 22:57.48 | diclophis-work | ${${COUNT}+1} |
| 22:58.30 | vlt | Hello. I replaced my ubuntu packaged asterisk with a gresh self compiled one. No wsounds aren't playing. Where is the default sound directory (now /var/lib/asterisk/sounds) defined? |
| 22:58.33 | Corydon-w | diclophis-work: $[] |
| 22:58.34 | thekidrio | its Set(COUNT=$[${COUNT} + 1]) |
| 22:58.56 | diclophis-work | gah, what does the $[] mean? |
| 22:59.00 | diclophis-work | evaluate? |
| 22:59.04 | Corydon-w | yes |
| 22:59.38 | Bobthehunter | jerjer ? |
| 23:03.05 | JerJer | eh? |
| 23:03.13 | JerJer | i'm about to gnaw on some dinner |
| 23:03.22 | JerJer | fresh kill |
| 23:03.24 | Bobthehunter | h323 |
| 23:03.34 | vlt | I tried to use the whole path in extensions.conf: Background(/var/lib/asterisk/sounds/cd1/track1). But it doesn't play. What could be wrong now? |
| 23:03.34 | Bobthehunter | just wondering if and wich versions wrork with 1.2.15 |
| 23:04.19 | JerJer | Bobthehunter: i would presume the versions listed in the readme - newer should work, if they haven't made API changes |
| 23:04.27 | Bobthehunter | kk |
| 23:04.29 | Bobthehunter | ill test |
| 23:05.07 | *** join/#asterisk eald (n=eald@189.157.105.134) |
| 23:05.43 | vlt | ... and the file name is /var/lib/asterisk/sounds/cd1/track1.gsm |
| 23:05.46 | diclophis-work | cool, now whats up with this error message: does not support indication 3, emulating it |
| 23:06.22 | diclophis-work | vlt: any error messages?, i have noticed sometimes with gsm files they are badly formatted |
| 23:06.31 | diclophis-work | and theres like "data header" errors |
| 23:06.49 | [TK]D-Fender | diclophis-work : Your problem is your goto. formatted as is it goes to "t,1", hence the looping |
| 23:07.08 | [TK]D-Fender | diclophis-work : Go read up on "asterisk expressions" on the WIKI |
| 23:07.10 | [TK]D-Fender | ~wikis |
| 23:07.13 | jbot | wikis is probably http://www.voip-info.org |
| 23:07.35 | diclophis-work | hmm, i will look into it |
| 23:08.04 | [TK]D-Fender | vlt : And what message do you get on that attempt? |
| 23:08.22 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 23:08.35 | vlt | diclophis-work, [TK]D-Fender: No error messages on -vvvvvvvvv CLI. It worked before with the same sound files and the ubuntu packaged version. Now I compiled my own and don't hear anything (no rtp packets sent out). |
| 23:10.35 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 23:10.40 | *** join/#asterisk Pepse (n=pepse@ip68-109-169-37.ph.ph.cox.net) |
| 23:10.53 | litage | i just put a second TDM400P with 4 FXO ports into my box a couple of days ago. when i answer an incoming call, i hear a huge amount of echo for up to a minute. it's been getting worse each day. this didn't exist before i installed the second TDM400P. what can i do to fix this? |
| 23:11.33 | *** join/#asterisk TaiSHi (i=WinNT@host19.190-31-19.telecom.net.ar) |
| 23:11.45 | [TK]D-Fender | vlt : And what format is it in? |
| 23:11.51 | TaiSHi | Hello everyone |
| 23:11.57 | vlt | [TK]D-Fender: gsm |
| 23:12.12 | vlt | [TK]D-Fender: converted by sox |
| 23:12.16 | [TK]D-Fender | litage : pastebin the output of "cat /proc/interrupts" |
| 23:12.18 | [TK]D-Fender | ~pb |
| 23:12.20 | jbot | from memory, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 23:12.32 | [TK]D-Fender | vlt : Pastebin the CLI output of the entire test call to it |
| 23:13.26 | vlt | [TK]D-Fender: Using the whole path or the old setting (just the file w/o ".gsm")? |
| 23:14.19 | litage | [TK]D-Fender: /proc/interrupts : http://rafb.net/p/bmzKYX76.html |
| 23:14.33 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 23:16.11 | *** join/#asterisk teknoprep (n=Chris@unaffiliated/teknoprep) |
| 23:17.52 | TaiSHi | Can I replace a HT with Asterisk ? |
| 23:18.01 | diclophis-work | is it possible for an Agent of a queue, to have multiple interfaces? |
| 23:18.02 | Qwell[] | ~ht |
| 23:18.03 | jbot | from memory, ht is Haiti. HyperThreading, a technology from intel included in processors from the P4 3.05 GHz, that emulates two processors in a single cpu machine |
| 23:18.11 | TaiSHi | HandyTone |
| 23:18.16 | TaiSHi | ~handytone |
| 23:18.20 | Qwell[] | a grandstream phone? |
| 23:18.25 | TaiSHi | Yes |
| 23:18.32 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 23:18.34 | Qwell[] | asterisk isn't a phone |
| 23:18.56 | florz | sure is it =:-) |
| 23:19.12 | litage | [TK]D-Fender: hrmmm, line 8. does that mean wctdm is sharing with hci_hcd:usb3 ? |
| 23:19.28 | [TK]D-Fender | litage : one of them is sharing interrupts. that is typically very bad. Upgrade to the latest * & zaptel |
| 23:19.50 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 23:19.53 | florz | I mean, a bit over-dimensioned maybe, but there is chan_oss and chan_alsa, which makes together with the CLI a phone, definitely :-) |
| 23:19.55 | TaiSHi | Actually, my idea is to remove (my boss' idea tbh) all physical stuff like hardphones and handytones |
| 23:20.06 | vlt | http://rafb.net/p/WqIsNY37.html |
| 23:20.11 | TaiSHi | And use softphone + bluetooth headphones |
| 23:20.23 | vlt | [TK]D-Fender: ^^^ |
| 23:20.45 | [TK]D-Fender | vlt : how long did you let it play? |
| 23:21.02 | vlt | [TK]D-Fender: ~10 sec |
| 23:22.17 | vlt | It surely worked before with the very same extensions.conf and the the same sound files. |
| 23:23.31 | litage | [TK]D-Fender: i upgraded zaptel to 1.2.14 when i installed the 2nd TDM400P. i'm still running asterisk 1.2.12.1 though. will upgrading asterisk help this interrupt issue? |
| 23:25.16 | [TK]D-Fender | litage : Possibly |
| 23:25.17 | TaiSHi | florz: Maybe I gave you the wrong idea ( Qwell[] this is for you aswell :) ) |
| 23:25.31 | litage | [TK]D-Fender: what should i do if it doesn't? |
| 23:26.18 | [TK]D-Fender | litage : Get a better card with HWEC |
| 23:26.19 | TaiSHi | I ment that instead of going like: network cable --> Grandstream --> Hard phone |
| 23:26.19 | vlt | Where does asterisk look for the sounds to play on a Playback() or Background() command? Which conf file? |
| 23:26.19 | *** join/#asterisk Grnd-Wire (n=groundwi@71-217-104-226.tukw.qwest.net) |
| 23:26.19 | litage | [TK]D-Fender: "a better card"? |
| 23:26.23 | TaiSHi | It would have to go network cable -> * -> Softphone |
| 23:26.36 | tzafrir_laptop | litage, you can probably ignore uhci_hcd. itnterrupts will be on the ehci_hcd |
| 23:26.57 | Grnd-Wire | So I've got a Digium TDM400P, and I'm having a weird problem when trying to dial outbound. It's actually picking up the line, but QWest isn't recognizing all of the digits it is sending.. Any ideas? |
| 23:27.33 | ManxPower | Grnd-Wire: put a "w" or two before the EXTEN in the Dial line |
| 23:27.48 | litage | tzafrir: so you're saying that it doesn't matter that wctdm and uhci_hcd:usb3 are sharing the same interrupt? |
| 23:28.01 | Grnd-Wire | ManxPower: Seriously? So it was a timing problem!? |
| 23:28.46 | [TK]D-Fender | litage : Well obviously you should try to get it on its own IRQ by disabling everything you don't need, and shifting slots, etc, but in the end you may still have echo issues |
| 23:28.47 | ManxPower | Grnd-Wire: your symptom is classic of a timing issue. Remember other then the X100P, Digium cards do NOT wait for dialtone before dialing |
| 23:29.21 | Grnd-Wire | ManxPower: mother ***, that's what I wa safraid of.. |
| 23:29.45 | |ryan| | what zone does TXTCIDNAME use for lookups? |
| 23:29.51 | litage | [TK]D-Fender: when you say "disabling everything", what are you referring to? |
| 23:30.05 | Bobthehunter | h323ep.cxx:3211: error: ‘class PNatStrategy’ has no member named ‘GetNATList’ |
| 23:30.06 | Bobthehunter | crap |
| 23:30.18 | Grnd-Wire | ManxPower: I contacted Digium support, and he wanted me to upgrade to the newest Asterisk, and all of this.. I told him it had to be a timing problem.. Why didn't he just tell me that? :( |
| 23:30.40 | ManxPower | Grnd-Wire: I don't want to comment about the quality of Digium support at this time. |
| 23:31.08 | techie | haha |
| 23:31.13 | Grnd-Wire | heh.. |
| 23:31.39 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 23:32.03 | ManxPower | I SUSPECT Digium has been hiring many new support people and they are not very well trained yet, but I have no actual proof of that. |
| 23:33.06 | Grnd-Wire | ManxPower: oooh, cause it used to be? |
| 23:33.43 | ManxPower | Grnd-Wire: I've been using Asterisk for like 5 years. I have a bit of an advantage of most tech support people. |
| 23:35.56 | pigpen | I am trying to do realtime voicemail with postgresql..., with the setting in extconfig.conf set...but where do I set the odbc settings? external odbc? |
| 23:36.07 | pigpen | sorry, new to odbc stuff. |
| 23:36.11 | ManxPower | Grnd-Wire: the tech support person should be shot for not knowing this classic, common issue. |
| 23:36.19 | [TK]D-Fender | litage : I'd start with unneeded USB port |
| 23:36.44 | litage | [TK]D-Fender: so disable stuff in the bios or kernel? |
| 23:37.04 | [TK]D-Fender | Bios |
| 23:37.08 | litage | thanks |
| 23:39.53 | TaiSHi | ManxPower |
| 23:40.07 | TaiSHi | Would you like to review my "problem" ? |
| 23:41.01 | ManxPower | TaiSHi: what is your problem>? |
| 23:41.12 | TaiSHi | Let me copy paste :P |
| 23:41.27 | ManxPower | TaiSHi: if it is more then 2 or 3 lines use pastebin.ca insteas |
| 23:41.29 | *** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com) |
| 23:41.30 | ManxPower | instead |
| 23:41.51 | TaiSHi | I would like to replace my office's networkk, right now it goes like: network cable --> Grandstream HandyTone--> Hard phone |
| 23:42.02 | TaiSHi | It would have to go like: network cable -> * -> Softphone |
| 23:42.37 | [TK]D-Fender | TaiSHi : So just installa bunch of softphones on your PC's |
| 23:42.50 | ManxPower | TaiSHi: no, I don't want to help you right now. |
| 23:43.04 | TaiSHi | Ok |
| 23:43.18 | TaiSHi | [TK]D-Fender: Main problem is not the soft phones |
| 23:43.33 | [TK]D-Fender | TaiSHi : So how about you tell us WHAT the problem is... |
| 23:43.33 | TaiSHi | Is taking the HT486 off the middle |
| 23:44.26 | [TK]D-Fender | TaiSHi : What do you mean "off the middle"? You unplug them. Completely. Period. You put them in a box. You light the box on fire. Then you install a fotphone on each of your PC's and have them auth to * using the same user & pass the the HT's used. |
| 23:44.35 | [TK]D-Fender | softphone* |
| 23:44.49 | [TK]D-Fender | TaiSHi : so whats the problem? |
| 23:44.59 | TaiSHi | Im reading and thinking what you just said |
| 23:46.14 | litage | i just ran `make install` in the asterisk source directory and it issued a warning, saying that some modules weren't installed by this version of asterisk. what should i do to fix this? |
| 23:46.51 | TaiSHi | The HT is not logging in on * |
| 23:47.05 | [TK]D-Fender | TaiSHi : What are they doing then? |
| 23:47.05 | CunningPike | litage: What modules? |
| 23:47.27 | TaiSHi | [TK]D-Fender: They probably log in @ Imspat (the provider) |
| 23:47.32 | litage | CunningPike: app_saycountpl.so , codec_g729a.so , format_mp3.so |
| 23:48.04 | [TK]D-Fender | TaiSHi : fine, so what do you expect * to do for you at all? |
| 23:48.22 | TaiSHi | Can't * log in @ Impsat |
| 23:48.29 | CunningPike | litage: OK- all it's doing is warning you that, because they are add-ons, you need to make sure their version is compatible with the version of Asterisk you are running |
| 23:48.30 | TaiSHi | Or receive and deliver the call? |
| 23:48.56 | litage | CunningPike: how do i ensure those modules' versions are compatible with asterisk 1.2.15? |
| 23:49.28 | CunningPike | litage: Well, install the latest 1.2.x asterisk-addons for a start, and re-install the modules from that |
| 23:49.36 | litage | k |
| 23:50.16 | ManxPower | TaiSHi: so what you are REALLY asking is how to get Asteirsk and your VOIP provider to work togather. |
| 23:50.23 | CunningPike | litage: Generally speaking, if the version of asterisk-addons matches the version of asterisk, you're good |
| 23:50.29 | TaiSHi | Wow, you really put it simple |
| 23:50.30 | litage | thanks |
| 23:50.41 | TaiSHi | Yes, that's what I'm asking, sorry for being quite dumb :) |
| 23:52.28 | litage | CunningPike: the newest version of asterisk-addons is 1.2.5, which obviously doesn't match 1.2.15 .... |
| 23:53.33 | CunningPike | litage: asterisk.latest + asterisk-addons.latest |
| 23:53.41 | litage | ah :) |
| 23:53.46 | CunningPike | litage: Bit of a literalist, aren't we? ;-) |
| 23:54.07 | litage | CunningPike: heh yeah, most of the time |
| 23:54.17 | CunningPike | :) |
| 23:54.19 | litage | CunningPike: i prefer not to assume stuff; gets you into trouble sometimes =P |
| 23:54.30 | CunningPike | Well, that's true..... |
| 23:54.41 | [TK]D-Fender | TaiSHi : You are completely unclear. Whtat do you expect Asterisk to DO for you? |
| 23:55.29 | TaiSHi | Receive the calls from the provider, and deliver to the operators @ the office |
| 23:56.41 | [TK]D-Fender | TaiSHi : And the soft-phones? |
| 23:57.03 | TaiSHi | The operators have 1 computer each, with softphones (soon-to-be) installed |
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| 23:58.09 | [TK]D-Fender | TaiSHi : So then what I said before still stands. You pull out the HT's. You install * and have it register to your provider. you install soft-phones on each of your PC's and have them auth to *. You configure your dialplan to route calls the way you like. The End. |
| 23:58.35 | TaiSHi | Great |
| 23:58.54 | TaiSHi | I didn't knew if * could login with my provider |
| 23:59.32 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |