| 00:00.11 | _paulos_ | (asterisk 1,2,12) |
| 00:01.17 | lenne_dk | So I just have to register the numbers with the same friend, and separate the numbers by giving different extensions. Workable, but a little hackish, if I wanted different accounting groups. |
| 00:01.29 | *** part/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 00:01.34 | JT_ | i could be leading you down the wrong road, but you could probably have a small context that checks the sup URI for the number, then redirects to the correct context based on that, lenne_dk |
| 00:01.41 | JT_ | a bit of a hack, but it should work |
| 00:01.48 | *** join/#asterisk FaithX (n=faithful@ns.linuxterminal.com) |
| 00:02.20 | lenne_dk | JT. Yes, I did that. After our main number was unreachable for half a day... :-( |
| 00:02.42 | JT_ | it reads the sup uri? |
| 00:02.56 | JT_ | well you'd think you'd check your main number :/ |
| 00:03.26 | lenne_dk | _paulos_ you should put the t option on when you queue: exten => 1,n,queue(paradiskunde,t) |
| 00:04.14 | _paulos_ | when the callee pres "*", the call hangs. |
| 00:04.44 | _paulos_ | weird... |
| 00:05.00 | _paulos_ | I'm usng *2 to attended transfers. |
| 00:05.42 | lenne_dk | JT: The call came in in the wrong context, where the s extension weren't. It dodn't occur to me that the new numbers would affect the old, but they did... |
| 00:06.10 | JT_ | ah, never assume anything will keep working after a change like that |
| 00:07.07 | lenne_dk | Yeah, there is as saying with "assume", which splits the word after "ass" :-) |
| 00:07.42 | *** join/#asterisk saftsack (n=saftsack@pD9E058AF.dip.t-dialin.net) |
| 00:08.24 | JT_ | heh |
| 00:11.31 | *** join/#asterisk Avochelm (n=damien@gw-morphett.koalatelecom.com.au) |
| 00:13.15 | *** join/#asterisk russellb (i=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 00:13.15 | *** mode/#asterisk [+o russellb] by ChanServ |
| 00:13.37 | ThoMe | a good asterisk gui? |
| 00:13.44 | JT_ | doesn't exist |
| 00:13.52 | ThoMe | JT_: hmm? |
| 00:13.57 | ThoMe | and a little good? ;) |
| 00:14.37 | JT_ | what's wrong with the cli? |
| 00:14.50 | ThoMe | ;( |
| 00:14.55 | ThoMe | wrong |
| 00:15.28 | russellb | ThoMe: check out asterisknow.org |
| 00:15.46 | russellb | ThoMe: and join #asterisk-gui or #asterisknow with questions about it .. |
| 00:15.52 | JT_ | ThoMe: well? |
| 00:17.56 | _paulos_ | oh... blind transfers works for agents, attended dont... |
| 00:17.59 | _paulos_ | I guess * is used to hangup. |
| 00:18.21 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
| 00:21.13 | has_many_questio | anyone have any comments about stanaphone? |
| 00:21.22 | has_many_questio | u guys think they'll be around for a while? |
| 00:21.45 | *** join/#asterisk k-man (n=jason@unaffiliated/k-man) |
| 00:21.54 | has_many_questio | i got a free number from them, i'll be pumping some cash into promoting that number..god knows if they come crashing..im screwed |
| 00:22.01 | k-man | anyone know of a debian+asterisk vmware image available anywhere? |
| 00:22.02 | lenne_dk | _paulos_ then change "*" in features.conf |
| 00:22.29 | sbingner | has_many_questio, if you're worried about that .... I suggest getting a number from somebody with a long standing reputation |
| 00:22.39 | k-man | also, anyone know where i could get an amplifier for a linksys phone and a plantronics headset? |
| 00:23.04 | has_many_questio | sbingner, basically AT&T? |
| 00:23.20 | has_many_questio | anyone else u recommend? |
| 00:23.42 | has_many_questio | anyone = anything |
| 00:24.16 | neddy | I just built and installed asterisk which went well |
| 00:24.30 | _paulos_ | thanks lenne_dk, will try that |
| 00:24.32 | neddy | how do I install the web server stuff in the screenshots on asterisknow.org |
| 00:24.59 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 00:25.21 | robl^ | neddy: join #asterisk-gui they are the pros for that |
| 00:25.24 | Supaplex | there's a gimp extension for screenshots that provide a webserver? |
| 00:25.25 | neddy | k |
| 00:26.03 | *** join/#asterisk ManxPower (n=manxpowe@56.sub-75-202-11.myvzw.com) |
| 00:26.17 | [TK]D-Fender | k-man : I did warn you the headset was wimpy on them didn't I? :) |
| 00:26.29 | k-man | [TK]D-Fender, yeah you did |
| 00:26.36 | k-man | [TK]D-Fender, hence i am asking about amp options |
| 00:27.36 | [TK]D-Fender | k-man : I have never seen one for 2.5mm ...... |
| 00:27.37 | k-man | [TK]D-Fender, actually, i haven't even tried the headset yet as i don't have a headset that has the 2.5mm plug on it |
| 00:27.37 | k-man | hmm |
| 00:27.37 | k-man | i wonder if it is electricaly equivalent to any other kind of headset socket |
| 00:27.37 | k-man | and if i could just wire up a converter of some sort |
| 00:30.18 | k-man | which OS is trixbox based on? any idea? |
| 00:30.35 | [TK]D-Fender | k-man : CentOS |
| 00:31.15 | k-man | oh, what is centos based on? redhat? |
| 00:32.04 | k-man | is trixbox worth downloading? |
| 00:32.11 | [TK]D-Fender | k-man : CentOS = RHEL - logos |
| 00:32.20 | k-man | ah, thanks |
| 00:32.35 | k-man | i'm a debian boy... is there a debian based trixbox equivalent? |
| 00:32.43 | [TK]D-Fender | Trixkbox is just a ready-made distro w/ * + Freepbx + other crap |
| 00:32.51 | k-man | oh |
| 00:32.58 | k-man | trixbox isn't asterisk? |
| 00:33.00 | [TK]D-Fender | hardly better than doing it yourself if your competant |
| 00:33.11 | sfinxen | I lite trixbox, but now i like asterisknow better |
| 00:33.11 | k-man | i mean, doesn;t have asterisk installed on it? |
| 00:33.14 | [TK]D-Fender | it owns your ass in is therefor a mistake |
| 00:33.22 | sfinxen | like |
| 00:33.34 | [TK]D-Fender | k-man : it formats your WHOLE DRIVE and dumps a FULL PBX onto it. |
| 00:33.51 | k-man | [TK]D-Fender, i was going to run it in a vmware image (for testing only) |
| 00:34.08 | sfinxen | [TK]D-Fender, i think trixbox asks you if you want to use automated partitioning? |
| 00:36.44 | ping2921 | I'm trying to have xlite connected to asterisk. anyone knows which rtp port I need to open in the router? |
| 00:37.22 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 00:41.10 | JT_ | CrashHD: i worked out what it was, it was generated by zaptel |
| 00:41.21 | JT_ | the indications for congestion sound right in asterisk |
| 00:41.25 | JT_ | they are wrong in zaptel |
| 00:44.13 | ManxPower | ping2921: all of them |
| 00:44.29 | mmlj4 | hey manx |
| 00:44.30 | robin_sz | out of interest, do you think I should try the latest gxp2000 firmware to se if the display blanking problem is finally solved, or is it futile? |
| 00:45.01 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 00:45.17 | mmlj4 | ManxPower: once I get over being sick, I'll finish the wiring in covington UMC |
| 00:45.26 | mmlj4 | just need to punch down in the closet |
| 00:45.34 | robin_sz | hmmm |
| 00:45.56 | robin_sz | "need to punch down in the closet" ... is that a euphamism? |
| 00:46.29 | mmlj4 | yes, i'm really talking about cocaine and bananas |
| 00:46.33 | *** join/#asterisk predder (n=predder@203.220.55.70) |
| 00:46.55 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 00:46.55 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 00:47.05 | robin_sz | funny, i thought you were talking about taking a dump ;) |
| 00:47.36 | Bobthehunter | thinks congestion is 37 |
| 00:50.17 | JT_ | cause code 37? |
| 00:53.41 | *** join/#asterisk jarro (n=jarek@h-87-199-9-2.dolsat.pl) |
| 00:58.14 | *** join/#asterisk coppice (n=chatzill@55.157.17.210.dyn.pacific.net.hk) |
| 01:06.47 | *** join/#asterisk sahafeez (n=sahafeez@ip68-6-215-70.sd.sd.cox.net) |
| 01:12.21 | *** join/#asterisk stridernzl (n=neville@222-155-172-130.jetstream.xtra.co.nz) |
| 01:14.00 | *** join/#asterisk RoyK (n=roy@217-175-39.100710.adsl.tele2.no) |
| 01:15.08 | *** join/#asterisk RoyK (n=roy@217-175-39.100710.adsl.tele2.no) |
| 01:15.25 | *** join/#asterisk MRH2 (n=Mr_happy@host-83-146-30-242.bulldogdsl.com) |
| 01:16.36 | MRH2 | hi silly question - where are the zaptel modules actually located again? |
| 01:20.26 | *** join/#asterisk sevard (i=kynan@24-179-181-160.dhcp.dlth.mn.charter.com) |
| 01:22.20 | MRH2 | never mind found it - curse the change of zaptel module paths causing problems with zaptel downgrades |
| 01:22.21 | Supaplex | MRH2: /lib/modules/... |
| 01:22.57 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 01:25.49 | MRH2 | yeah was looking under the wrong kernel - someone please send food as soon as possible |
| 01:29.44 | perd | http://ccrow.net/uploaded_images/will_code_html_for_food.jpg |
| 01:30.53 | sevard | MRH2: I made shells and peas in butter and garlic salt with slow skillet grilled garlic and herb spiced chicken breast |
| 01:31.19 | sevard | manderin oranges in a glass container, so you don't get that tinny aftertaste on the side of the plate |
| 01:31.26 | MRH2 | u tease |
| 01:31.51 | sevard | welcome to #asterisk |
| 01:37.34 | *** join/#asterisk dhill (i=dhill@fog.mindcry.org) |
| 01:38.37 | dhill | sipura -> asterisk sip gateway -> as5300 (pri) .. if i send a fax from a machine attached to the sipura.. does the asterisk box even care if it is a fax or not? |
| 01:38.54 | dhill | doesn't the sipura and as5300 talk t38? |
| 01:38.59 | dhill | or am i wrong? |
| 01:39.11 | dhill | i guess i am wrong :P |
| 01:40.19 | dhill | i guess it is considered pass-through? |
| 01:40.53 | *** join/#asterisk ariel_ (n=ariel_@dsl-20-177.cofs.net) |
| 01:44.32 | *** join/#asterisk BitBandit (n=polx@68-116-238-170.dhcp.stgr.ut.charter.com) |
| 01:46.52 | mquin | dhill: depends on the sipura - from what I can see the 2100 does, the others don't (but will do passthrough over g.711) |
| 01:46.55 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 01:46.55 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 01:49.26 | dhill | i am thinking the as5300 and the sipura are talking ulaw.. not t.38 |
| 01:50.30 | robl^ | the windover also has an bathroom enterance not connected to a bedroom.. |
| 01:50.35 | robl^ | oops. |
| 01:50.55 | robl^ | wrong window. ;-) |
| 01:51.07 | mquin | :) |
| 01:51.21 | robl^ | s/hids/hides |
| 02:03.39 | [hC] | Is there a manager API command to return the same data as 'show voicemail users' ? |
| 02:03.47 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
| 02:03.56 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-155-21-116.red.bezeqint.net) |
| 02:04.08 | [hC] | problem is currently, show voicemail users output is pretty mangled and not easily parseable. |
| 02:05.48 | k-man | if asterisk 1.2 is apt-getable, should i use that? or should i compile 1.4 (for debian)? |
| 02:06.04 | *** join/#asterisk topping (n=topping@204.152.96.50) |
| 02:10.31 | *** part/#asterisk stridernzl (n=neville@222-155-172-130.jetstream.xtra.co.nz) |
| 02:11.45 | *** join/#asterisk jimmy_deanPB (n=jhodapp@cpe-65-29-94-202.indy.res.rr.com) |
| 02:21.03 | *** join/#asterisk grandy (n=chatzill@c-71-198-130-108.hsd1.ca.comcast.net) |
| 02:21.17 | tzafrir_laptop | anyway, 1.2 is generally more stable ;-) |
| 02:21.20 | *** join/#asterisk stkn__ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 02:21.25 | Qwell | [hC]: there is something on the tracker |
| 02:22.06 | [hC] | I just found it |
| 02:22.09 | grandy | hello... Trying to figure out how to set the caller ID that is shown when a call from the outside world comes into my incoming context and is redirected to a local iax2 softphone |
| 02:22.09 | [hC] | Looks like its going into 1.4? |
| 02:32.48 | *** join/#asterisk osiris (n=osiris@c-71-205-27-131.hsd1.mi.comcast.net) |
| 02:33.25 | *** join/#asterisk AceStrider1 (n=AceStrid@216-150-34-249.netscorp.net) |
| 02:35.06 | AceStrider1 | Hi all |
| 02:39.23 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-231-237.dc.res.rr.com) |
| 02:41.00 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 02:41.44 | BitBandit | to dial an internal extention is it just #ext or is it *ext |
| 02:43.01 | grandy | hello... i'm trying to figure out how to set the caller id that displays on the softphone receiving the call... if the call is from the outside... |
| 02:45.45 | battini | grandy Set(CALLERID(all)=boobs <1337>) |
| 02:45.55 | grandy | battini: haha thanks... ! |
| 02:46.07 | grandy | battini: how do I get the thing it's currently set to? |
| 02:46.34 | battini | CALLERID(all) variable |
| 02:46.37 | battini | before you set it |
| 02:46.55 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 02:46.55 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 02:47.00 | grandy | battini: ahh! |
| 02:48.52 | battini | grandy http://www.asteriskguru.com/tutorials/dialplan_functions.html |
| 02:50.07 | grandy | battini: thanks |
| 02:55.53 | *** join/#asterisk emphyrio (n=stryfe@ool-4570f460.dyn.optonline.net) |
| 03:01.31 | AceStrider1 | has anybody ever experience the phone answering itself as you place a call out? |
| 03:03.08 | ping2921 | do i need to specify a stun server for connecting xlite to asterisk? |
| 03:03.39 | grandy | hello... how would i know whether there is actually caller ID on the incoming call? It says unknown iax caller? |
| 03:05.47 | *** join/#asterisk abatista (n=ariel_@dsl-20-177.cofs.net) |
| 03:06.33 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqia.cable.mindspring.com) |
| 03:07.29 | *** join/#asterisk fnordus (n=dnall@24.85.128.203) |
| 03:08.45 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 03:08.53 | blitzrage | evening all! |
| 03:09.47 | blitzrage | anyone know where I can get a service contract from Cisco for firmware? |
| 03:11.58 | ManxPower | blitzrage: Canada or USA? |
| 03:12.07 | blitzrage | Canada |
| 03:12.16 | ariel_ | hi |
| 03:12.16 | ManxPower | No idea |
| 03:12.22 | ariel_ | ManxPower, long time no see |
| 03:12.26 | ariel_ | blitzrage, hi |
| 03:12.58 | blitzrage | ariel_: hey hey |
| 03:13.02 | blitzrage | ManxPower: heh |
| 03:13.15 | blitzrage | ManxPower: does it really matter? |
| 03:13.27 | blitzrage | I have a US address if that helps :) |
| 03:13.55 | ariel_ | I think that voipsupply sells them now. But I could be wrong |
| 03:14.00 | ManxPower | blitzrage: I think Cisco cares. |
| 03:14.27 | ManxPower | For USA: Gaston Dureau, Avenue Computer Supply, 504-523-7874 |
| 03:16.54 | ManxPower | He is also an authorized Polycom reseller. |
| 03:17.18 | ariel_ | hummm polycom good... |
| 03:17.27 | ManxPower | Avenue doesn't do well if you don't know what you want, but if you have a part number, they are good. |
| 03:17.59 | blitzrage | ManxPower: got a part number for me? :) |
| 03:18.09 | blitzrage | ariel_: yah, I have a polycom too, and prefer them, but I also have a few cisco phones here |
| 03:18.21 | blitzrage | actually, I prefer my 7960 over everything |
| 03:18.33 | ManxPower | blitzrage: no idea. I have a CCO ID that lets me get any firmware. |
| 03:18.47 | blitzrage | ManxPower: that's what I'm trying to get :) |
| 03:19.46 | ManxPower | blitzrage: We have SmartNet on our routers |
| 03:20.18 | blitzrage | ahhh, that's handy |
| 03:21.00 | ariel_ | sure are heavy to put into a rack |
| 03:23.01 | *** join/#asterisk jesselang (n=jesse@h69-130-162-134.69-130.unk.tds.net) |
| 03:26.42 | *** join/#asterisk dongc (n=dongc@203.117.206.249) |
| 03:27.34 | blitzrage | ariel_: those things are monsters. We had two in our classroom, and put out a ton of heat. |
| 03:28.26 | *** join/#asterisk jimmy_deanPB (n=jhodapp@cpe-65-29-94-202.indy.res.rr.com) |
| 03:29.18 | ariel_ | yes they do. But 288 poe ports with full manangent asa router n board. |
| 03:31.02 | *** join/#asterisk Splat (n=splat@220-253-98-56.TAS.netspace.net.au) |
| 03:31.30 | k-man | can someone point me to a getting started guide for asterisk please? |
| 03:37.29 | *** join/#asterisk Globetrotter (n=eric@205.211.214.167) |
| 03:37.45 | Globetrotter | hi guys,, how do i enable call witing?? |
| 03:37.52 | Globetrotter | call waiting |
| 03:46.55 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 03:46.55 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 03:47.15 | blitzrage | ~book |
| 03:47.25 | jbot | extra, extra, read all about it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 03:47.27 | blitzrage | k-man: ^^^ |
| 03:48.57 | k-man | oh thanks blitzrage |
| 03:50.24 | blitzrage | np |
| 03:50.45 | k-man | is that book published yet? |
| 03:51.04 | *** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com) |
| 03:51.39 | blitzrage | k-man: has been for a year -- 2nd edition in progress |
| 03:51.48 | blitzrage | http://www.oreilly.com/catalog/asterisk |
| 03:52.08 | JT_ | will the 2nd edition also be released in pdf? |
| 03:52.26 | k-man | thanks blitzrage |
| 03:52.42 | blitzrage | JT_: yep |
| 03:52.43 | k-man | oh, is there a second edition coming out soon? |
| 03:52.51 | blitzrage | a few months |
| 03:52.52 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 03:53.40 | JT_ | will it be very different from the first? |
| 03:54.06 | blitzrage | yah -- lots of changes between 1.2 and 1.4 |
| 03:54.18 | Globetrotter | hi guys,, how do i enable call witing?? |
| 03:54.24 | Globetrotter | <PROTECTED> |
| 03:54.36 | k-man | hmmm |
| 03:54.38 | k-man | interesting |
| 03:54.46 | k-man | is the source for the book available, or only the pdfs? |
| 03:55.30 | *** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
| 03:55.43 | ManxPower | Globetrotter: that would depend on the phone you have |
| 03:55.59 | blitzrage | Globetrotter: on an IP phone, you just need to send it two calls typically |
| 03:56.10 | blitzrage | k-man: only the PDFs |
| 03:56.30 | k-man | its a bit annoying how they included the crop marks |
| 03:56.32 | k-man | oh well |
| 03:56.36 | *** part/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
| 03:56.39 | k-man | i was going to buy it anyway |
| 03:56.53 | blitzrage | nice |
| 03:57.31 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.94) |
| 04:02.57 | ez` | any idea how enable call waiting on polycom ip500 ?? |
| 04:04.09 | blitzrage | ez`: allow more than one call per line |
| 04:04.32 | blitzrage | callsPerLine I think is the option in the config file |
| 04:04.39 | ez` | in tftp file ? which one |
| 04:04.39 | ez` | k |
| 04:04.46 | blitzrage | the phone.cfg file |
| 04:04.51 | ez` | kewl |
| 04:05.04 | blitzrage | you should see it near the top in the section for your registration information |
| 04:05.14 | ez` | i try it |
| 04:05.14 | blitzrage | or rather, line information |
| 04:06.02 | ez` | callsPerLineKey="" |
| 04:06.07 | *** join/#asterisk SECGOD (i=SECGOD@c-71-57-36-106.hsd1.il.comcast.net) |
| 04:07.32 | blitzrage | set it to 2+ |
| 04:07.56 | ez` | k then i reboot the phone only ? |
| 04:08.00 | file | blitzrage: are you helping people? 'tsk 'tsk |
| 04:08.11 | ez` | heh |
| 04:09.06 | blitzrage | file: blasphemy. Yes I know I'll be scolded by the developers tomorrow. |
| 04:09.17 | file | :D |
| 04:13.16 | ez` | blitzrage sould i also enable cw to this extension in CLI ? |
| 04:13.23 | ez` | sould = should |
| 04:14.23 | blitzrage | huh? |
| 04:14.35 | blitzrage | in the CLI? |
| 04:14.41 | ez` | yes |
| 04:14.47 | blitzrage | if the phone can accept more than one call per line, it just works |
| 04:14.50 | blitzrage | what CLI? |
| 04:15.18 | JT_ | k-man: what do you mean by source to the book? |
| 04:15.19 | ez` | i get a busy tone ; |
| 04:15.39 | k-man | JT_, presumably they wrote it in dockbook or latex or some other tool |
| 04:15.46 | JT_ | blitzrage: do you guys needs any help for error checking or anything? |
| 04:15.48 | blitzrage | JT_: he means the DocBook |
| 04:15.51 | JT_ | right |
| 04:15.53 | k-man | JT_, and not in raw pdf |
| 04:15.59 | blitzrage | k-man: correct |
| 04:16.30 | JT_ | blitzrage: i remember seeing a few errors in the book, i can't remember what they are now :( |
| 04:16.30 | k-man | although, if the author wrote it in raw pdf, i would be pretty impressed ;) |
| 04:16.32 | blitzrage | JT_: when we get that far, I might post something in here |
| 04:16.43 | JT_ | if i notice some of the old errors, is it worth pointing them out? |
| 04:17.08 | k-man | blitzrage, are you the author? |
| 04:17.20 | blitzrage | sure -- http://www.oreilly.com/catalog/asterisk has errata |
| 04:17.37 | blitzrage | see that it hasn't already been fixed in a later edition -- that PDF is from the first printing (i.e. old) |
| 04:18.58 | blitzrage | k-man: I may or may not be a co-author |
| 04:19.16 | k-man | very non commital ;) |
| 04:19.44 | blitzrage | I may or may not be commital |
| 04:21.01 | JT_ | blitzrage's commital status is still subject to a number of allegations |
| 04:21.23 | k-man | you remind me of Broomfondle and Magicthighs in hhgttg |
| 04:23.33 | ez` | my polycom a re working fine ; but there is a online step by step recommanded config tutarial ; ?/ |
| 04:23.40 | *** join/#asterisk thoughtpolice (n=austin@ip70-185-140-61.lu.dl.cox.net) |
| 04:24.04 | ez` | blitzrage thanks you solve my problem by the way |
| 04:24.10 | blitzrage | w00t |
| 04:24.34 | blitzrage | ez`: step by step documentation is rare |
| 04:24.49 | ez` | baddest / worst thing in my system ; voice quality using sipura 3000 ; pstn line is not very clear ,,, |
| 04:25.01 | SwK | at&t pwnt j00! |
| 04:25.58 | ez` | blitzrage i mean i can see there is a way to hook these phone to 2 server ; may be interresting in case a server is unreacble .. yes ?? |
| 04:26.18 | blitzrage | there should be a backup proxy setting in the config files |
| 04:26.28 | blitzrage | if it can't register to one server, it'll try to register to the backup |
| 04:27.41 | ez` | so if i got 2 server or 2 internet provider ; it can switch automticly ?? |
| 04:28.56 | ez` | i would need a pdf talking about these function ; anyone got a url ? |
| 04:30.29 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 04:32.11 | *** join/#asterisk PhilKC (i=greece@freenode/staff/about.linux.philkc) |
| 04:34.52 | L|NUX | hello every one |
| 04:35.03 | L|NUX | can some one tell me how can i generate sip 503 error ? |
| 04:38.57 | james_ | a computer's probably a good start |
| 04:39.29 | L|NUX | james_ : but how ? |
| 04:39.55 | james_ | you just want to generate a 503 once-off? |
| 04:40.05 | james_ | you can use sipsak to construct a message and send it out |
| 04:40.39 | L|NUX | james_ : humm |
| 04:42.50 | L|NUX | but if we want to generate it using asterisk / |
| 04:43.27 | james_ | ahh, in that case, no idea |
| 04:43.29 | JT_ | ez`: you mean 2 sip providers? |
| 04:45.05 | L|NUX | well |
| 04:45.08 | L|NUX | basically |
| 04:45.31 | L|NUX | we have did provider who need me to generate sip 503 error on their asterisk |
| 04:46.55 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 04:46.56 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 04:46.57 | ez` | yesjt |
| 04:47.40 | *** join/#asterisk BugKhaM (n=LAMER@ppp-58.8.9.189.revip2.asianet.co.th) |
| 04:48.20 | BugKhaM | when upgrading to 1.4.0, do I need to remove the 1.2.x? |
| 04:49.54 | ManxPower | BugKhaM: then end of "make install" should list the modules that were not built with that version of asterisk, you can remove them manually |
| 04:53.06 | Globetrotter | So I dont have to do anyhting speacil to enable call waiting?? |
| 04:53.21 | Globetrotter | i have grandstream gxp2000 |
| 04:53.58 | ez` | Globetrotter if you using freepbx you gat to enable it *XX |
| 04:54.32 | Globetrotter | i am not using freepbx.. |
| 04:54.47 | Globetrotter | are there any settings fro call waiting? |
| 04:55.20 | ez` | sorry i using polycom ; dunno about gxp2000 |
| 04:55.21 | BugKhaM | ManxPower: the ones in /usr/lib/asterisk/modules/? |
| 04:55.31 | JT_ | ez`: sorry missed that until then |
| 04:55.43 | JT_ | ez`: people will find it easier if you address things to them like JT: |
| 04:55.47 | *** join/#asterisk FaithX (n=faithful@ns.linuxterminal.com) |
| 04:55.50 | naftali5 | Globetrotter, the gxp 2000 has an option under advanced Disable Call-Waiting - default enabled |
| 04:56.38 | ez` | JT_: ok |
| 04:57.00 | JT_ | ez`: multiple routes to the PSTN is simply a matter of setting up your dialplan correctly |
| 04:57.20 | JT_ | it's pretty easy though |
| 04:57.22 | JT_ | eg |
| 04:57.52 | JT_ | exten => _XXXX,1,Dial(SIP/Provider1/${EXTEN}) |
| 04:57.58 | JT_ | exten => _XXXX,n,Dial(SIP/Provider2/${EXTEN}) |
| 04:58.06 | JT_ | at its simplest |
| 04:58.20 | ez` | JT_ k |
| 04:59.20 | ez` | thanks |
| 04:59.20 | JT_ | np |
| 04:59.59 | JT_ | make sure qualify is switched on for your sip peers, it will save time dialling in the case that the first provider is completely unreachable |
| 05:01.13 | *** join/#asterisk Zand3r (n=Zand3r@spc2-bolt7-0-0-cust301.bagu.broadband.ntl.com) |
| 05:01.59 | Corydon76-home | JT_: I certainly have |
| 05:02.16 | Zand3r | Hi all... I'm looking at the S101I IAXy and notice the data sheet says it can do transfers (and blind transfers) - I wondered how this worked when the phone I plug in to it is analog and therefore has no IAX specific controls? |
| 05:02.26 | Zand3r | Is there an instruction manaual available at all? |
| 05:02.34 | Corydon76-home | Zand3r: # transfers |
| 05:03.08 | Corydon76-home | I think it can do flash hook transfers, too |
| 05:04.13 | Zand3r | corydon76-home - I see, seems obvious now, so the user would press # and then the extension - thanks |
| 05:04.53 | Zand3r | I'm looking for a wireless phone -strugling - and thinking an ATA and a DECT phone might do the job. |
| 05:04.57 | JT_ | Corydon76-home: wondered, or received donations? |
| 05:05.08 | Corydon76-home | JT_: donations |
| 05:05.13 | JT_ | ah ok |
| 05:05.15 | Corydon76-home | Twice, even |
| 05:05.38 | JT_ | heh |
| 05:05.50 | Corydon76-home | although with the second one, I gave him the option of making a tax-deductible gift to a nonprofit charity I run |
| 05:05.57 | JT_ | i never /quit, so putting anything in my quit message would be pretty pointless |
| 05:06.55 | *** join/#asterisk Igbothom_3rd (n=Hilton@office.quarkit.com.au) |
| 05:07.21 | Corydon76-home | I accept deep tissue massages and sexual favors, too. ;-) |
| 05:07.59 | Corydon76-home | although those tend not to be tax deductible |
| 05:11.13 | *** join/#asterisk InHisName (n=Administ@68.38.105.1) |
| 05:13.44 | JT_ | heh |
| 05:14.49 | InHisName | who is using sunrocket with *, I changed something and cant get it back. |
| 05:18.51 | *** join/#asterisk topping (n=topping@204.152.96.50) |
| 05:19.36 | *** join/#asterisk topping (n=topping@204.152.96.50) |
| 05:19.47 | *** join/#asterisk topping (n=topping@204.152.96.50) |
| 05:20.21 | *** join/#asterisk topping (n=topping@204.152.96.50) |
| 05:34.49 | InHisName | I am not quitting yet. Where are * conversations ? |
| 05:45.22 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 05:46.55 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 05:46.55 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 05:47.18 | InHisName | hello Bhaal and sbingner |
| 05:47.34 | k-man | InHisName, just be patient |
| 05:47.47 | k-man | if someone knows the answer to your question they will answer it, if not, they wont |
| 05:48.04 | k-man | go and have a cup of tea while you are waiting |
| 05:48.39 | InHisName | k-man, where are the other conversations ? seems so dead |
| 05:48.54 | sbingner | has_many_questio, no I don't really know any well enough to reccomend |
| 05:49.02 | k-man | InHisName, nowhere else afaik |
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| 05:55.32 | zeeesh | hi |
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| 06:00.30 | InHisName | hi zeesh |
| 06:06.57 | *** part/#asterisk InHisName (n=Administ@68.38.105.1) |
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| 06:30.15 | variable_office | any ideas on this -> http://forums.digium.com/viewtopic.php?p=41995#41995 ?? |
| 06:40.28 | *** join/#asterisk Jason[1] (n=jason@61.6.65.226) |
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| 06:47.59 | Jason[1] | hi... |
| 06:48.09 | Jason[1] | I hv a problem with PRI... |
| 06:48.15 | Jason[1] | anyone care to help? |
| 06:48.35 | JT_ | care to provider more details? |
| 06:48.41 | JT_ | provide, even |
| 06:48.43 | Jason[1] | ^^ |
| 06:49.05 | Jason[1] | ok. I hv an up & running PSTN PRI connected to Digium card |
| 06:49.53 | Jason[1] | however, whenever I reboot or restart Asterisk, I need to unplug and plug the cable in order for the PRI to up again |
| 06:50.40 | Jason[1] | this is giving me some remote support problem as whenever I restart *, I need to be on-site to unplug the PRI |
| 06:50.44 | JT_ | what sort of PRI? |
| 06:50.54 | Jason[1] | euroISDN |
| 06:51.00 | JT_ | so E1? |
| 06:51.05 | Jason[1] | ya |
| 06:51.27 | JT_ | have you tried rmmoding the relevant driver, then modprobing it, then ztcfg, then starting asterisk? |
| 06:52.17 | Jason[1] | done with modprobing & ztcfg...but whta do you mean by rmmoding? |
| 06:52.48 | JT_ | to unload it from the running kernel of course |
| 06:53.44 | Jason[1] | ohhh...new to Linux...do u mind telling me the command so I can explore ove rthe net? |
| 06:53.56 | JT_ | rmmod |
| 06:54.05 | JT_ | rmmod <driver> |
| 06:54.21 | Jason[1] | ok. will explore on my own and come back if still hv problem...Thanks |
| 06:54.31 | JT_ | it needs to not be in use in order to rmmod, so asterisk not running |
| 06:54.32 | JT_ | well |
| 06:54.35 | JT_ | if that fixes it |
| 06:54.42 | JT_ | it's likely a bug in the relevant driver |
| 06:54.46 | Jason[1] | btw....what dirvers should I rmmod? |
| 06:55.00 | JT_ | you should know what driver runs your pri card |
| 06:55.06 | JT_ | depends on what pri card you have |
| 06:55.23 | Jason[1] | i.e wct4xxp ? |
| 06:55.41 | JT_ | yes |
| 06:55.56 | Jason[1] | ic ic..will try it now... |
| 06:56.03 | Jason[1] | thanks again |
| 07:03.37 | *** join/#asterisk Flinx (n=Flinx@c-69-250-66-154.hsd1.md.comcast.net) |
| 07:03.51 | Flinx | *grin* I really need to learn how to spell better... |
| 07:05.20 | Flinx | Question: If I wanted to play with VoIP, would I have to subscribe to a VoIP server / In Addition to / running an Asterisk server? |
| 07:05.46 | nick125_lappy | Flinx: Usually, yes |
| 07:05.53 | Flinx | *nods* |
| 07:06.15 | Flinx | I wasn't sure If I could just use my existing broadband connection. |
| 07:06.31 | *** join/#asterisk tengulre (n=tengulre@61.185.220.249) |
| 07:06.31 | nick125_lappy | You should be able to, least for playing around |
| 07:07.24 | Flinx | *nods* |
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| 07:08.34 | Flinx | what would the limiting factor be? |
| 07:09.27 | JT_ | Flinx: voip is a general technology that can be used in a number of ways |
| 07:09.51 | JT_ | you can use voip softphones or physical hardphones or ATAs, they require not voip provider accounts |
| 07:09.56 | JT_ | s/not/no/ |
| 07:10.10 | Flinx | *nod* |
| 07:10.20 | JT_ | you can connect asterisk to another asterisk server, or a free voip server like FWD |
| 07:11.07 | *** join/#asterisk osiris (n=osiris@c-71-205-27-131.hsd1.mi.comcast.net) |
| 07:11.32 | Flinx | I know that work used to do a lot of work with Asterisk, I was also able to borrow one of the phones for a few weeks to play with. |
| 07:11.52 | Flinx | Any suggestions on a starting point to learn more? |
| 07:12.26 | JT_ | ~thebook |
| 07:12.37 | jbot | thebook is probably a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 07:12.38 | Flinx | the book? ruh roh.... |
| 07:12.41 | nick125_lappy | Flinx: Setup a small asterisk box, hook it up to a few voip providers (FWD, iaxtel, etc), etc |
| 07:12.44 | Flinx | oh :D |
| 07:13.07 | Flinx | I've printed/read an earlier version of that ... two years ago *thinks* |
| 07:13.13 | nick125_lappy | That's a really good book |
| 07:13.20 | nick125_lappy | I have the print copy of it somewhere around here |
| 07:13.27 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 07:13.38 | Flinx | *looks to see if it's different.* |
| 07:14.10 | Flinx | I know that one of the other engineers has a copy of the lime green book of Asterisk... but that's prolly ages old |
| 07:14.44 | JT_ | the online version hasn't changed |
| 07:14.51 | JT_ | 2nd revision is coming out eventually |
| 07:15.08 | Flinx | *nod* |
| 07:15.16 | Flinx | I'll take a look-see |
| 07:15.43 | JT_ | it's easier to buy that print |
| 07:15.46 | JT_ | s/that/than/ |
| 07:15.50 | Flinx | well yeah |
| 07:16.01 | JT_ | also cheaper if you actually pay all your printing costs |
| 07:16.12 | Flinx | In looking through this, I only printed up the "this is what Asterisk is..." documentation. |
| 07:16.26 | Flinx | what I have a hard copy of is only 15 pages or so. |
| 07:16.34 | JT_ | ah heh |
| 07:16.40 | JT_ | slightly different |
| 07:16.41 | tengulre | JT_: do you using asterisk for what? |
| 07:16.55 | JT_ | the wiki is the other great resource |
| 07:16.58 | JT_ | ~thewiki |
| 07:17.14 | jbot | rumour has it, thewiki is at http://www.voip-info.org/wiki-Asterisk |
| 07:17.14 | JT_ | ~wiki |
| 07:17.36 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 07:17.40 | nick125_lappy | Asterisk is like a box of legos, you can build whatever you want in it |
| 07:18.02 | JT_ | tengulre: a few different things, eg. isdn voip gateway |
| 07:18.21 | nick125_lappy | It's kind of hard to actually find something to do since there is so much to do. |
| 07:18.45 | Flinx | hehe |
| 07:18.53 | Flinx | right now I'm just trying to get one phone to work |
| 07:18.59 | Flinx | but soon... the world! |
| 07:19.15 | tengulre | JT_: does the telecom corp disable your voip protocl in your country? |
| 07:19.17 | Defraz | I have two asterisk boxes, on the asterisk box I have a sip trunk to my provider, I caonnect with a softphone(xlite) and I can make calls. I connect with my second asteirsk box then I dial out from that with a softphone connected with my second box and it starts to dial out the sip provider but then returns a 480 response. |
| 07:19.20 | Defraz | am I mjissing something |
| 07:19.40 | JT_ | tengulre: no, there is not just one telco either for everything |
| 07:20.53 | tengulre | JT_: but owner country disabled the SIP protocl, because owen using the internet resource. |
| 07:21.14 | JT_ | owen using?? |
| 07:21.38 | tengulre | sorry! s/owen/we |
| 07:22.10 | JT_ | you only have one telco and internet provider? |
| 07:23.29 | tengulre | JT_: yes. |
| 07:24.48 | tengulre | JT_: in my country have 3 internet providers. CHINA TELECOM, CHINE CNC, CHINA MOBILE. |
| 07:25.12 | tengulre | but the CHINA MOBILE only provide wireless services. |
| 07:25.17 | tengulre | so... |
| 07:25.20 | JT_ | well that sucks |
| 07:27.18 | tengulre | JT_: since I building a asterisk in a company , that in city A. the have many part company in different place. but using SIP protocl can not communication . |
| 07:27.26 | tengulre | only IAX2. |
| 07:28.52 | JT_ | tengulre: err you can probably just use a different port to run SIP |
| 07:28.52 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 07:29.47 | tengulre | JT_: that check the data packate. |
| 07:30.25 | JT_ | on every single port? |
| 07:30.34 | JT_ | that'd be a lot of cpu time |
| 07:32.41 | tengulre | anybody using asterisk under debian linux? |
| 07:32.47 | Flinx | damn... I guess my phone isn't FWD certified.. *chuckles* |
| 07:32.53 | Flinx | anyways.. thanks for all the help :D |
| 07:32.55 | JT_ | tengulre: yes |
| 07:33.30 | tengulre | JT_: do you using debian os? |
| 07:33.35 | JT_ | yes |
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| 07:34.01 | tengulre | JT_: how many processors in you box? |
| 07:35.07 | tengulre | s/you/your |
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| 07:36.28 | *** join/#asterisk Vec (n=Vector@dsl-243-70-208.telkomadsl.co.za) |
| 07:38.51 | JT_ | tengulre: most of them have 2 |
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| 08:05.32 | angryuser | good morning;) |
| 08:06.58 | angryuser | i have searched al over the web, posted on forums, bur i have not found to monitor FXS chanells with the 'Hint' or 'blf', maybe somebody here knows how? |
| 08:07.15 | Jason[1] | JT... |
| 08:07.35 | Jason[1] | I hv tested the PRI problem |
| 08:08.12 | Jason[1] | it seem that the problem only occure only if I rebooted the server |
| 08:08.49 | Jason[1] | and i solve it by issue ztcfg -vvvv while Asterisk is running... |
| 08:09.09 | Jason[1] | that somehow reset the Digium PRI card and up my links |
| 08:09.31 | Jason[1] | JT...u there? |
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| 08:21.17 | JT_ | Jason[1]: sorry, that doesn't really answer my question of whether rmmodding helped |
| 08:22.10 | Jason[1] | JT_..I think it does not...as the problem only occur if I rebooted the server |
| 08:22.15 | *** join/#asterisk JT (n=jon@unaffiliated/jt) |
| 08:22.31 | Jason[1] | so rmmodding does not apply here |
| 08:23.28 | Jason[1] | I believe the problem might due to Asterisk does not reset the PRI links during startup |
| 08:24.05 | JT_ | sorry, you're making little sense and are not being consistant |
| 08:24.14 | JT_ | does this problem occur when rebooting whole computer |
| 08:24.15 | Jason[1] | thus..when I do a ztcfg while * is running, it reseted the links |
| 08:24.22 | Jason[1] | ya.. |
| 08:24.22 | JT_ | or just restating the asterisk server? |
| 08:24.33 | Jason[1] | only cold reboot |
| 08:24.45 | Jason[1] | restarting * is ok. |
| 08:24.48 | JT_ | err so you don't have ztcfg in startup scripts |
| 08:25.03 | Jason[1] | I hv. |
| 08:25.11 | sevard | hiv? |
| 08:25.13 | Jason[1] | it run properly |
| 08:25.27 | *** join/#asterisk reber (i=reber@gateway/tor/x-96b8f5e4e9e7eab7) |
| 08:25.28 | Jason[1] | I even do manual ztcfg. |
| 08:25.35 | JT_ | it must run AFTER the kernel module is insmodded or modprobed |
| 08:25.51 | Jason[1] | then only I start *...is still the same |
| 08:27.39 | JT_ | i'm sorry, i don't understand |
| 08:27.39 | Jason[1] | yes. this is the steps..Linux up the I login as root, the run modprobe zaptel, wct4xxp,ztcfg -vvvv then asterisk -vvvvvc |
| 08:27.39 | Jason[1] | then zap show status...the link is NOP |
| 08:28.10 | JT_ | modprobing zaptel is unnecessary, modprobing wct4xxp is all you should need |
| 08:28.14 | Jason[1] | then I issue ztcfg -vvvv on another screen, it reset the links and up the link |
| 08:28.34 | Jason[1] | ohh...ok |
| 08:28.37 | JT_ | is there any messages in dmesg about it? |
| 08:28.50 | gfraysse | <PROTECTED> |
| 08:29.07 | Jason[1] | sry...dmesg? |
| 08:29.08 | *** join/#asterisk inspired (n=mikael@cl-330.sto-01.se.sixxs.net) |
| 08:29.15 | JT_ | dmesg |
| 08:29.35 | JT_ | when you bring up the card it should get some lines added to it |
| 08:30.34 | Jason[1] | oh...yes..the normal msg...nothing un-usual.. |
| 08:30.38 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 08:30.46 | Jason[1] | u need to see? |
| 08:30.50 | JT_ | ~pb |
| 08:31.01 | jbot | i guess pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 08:31.01 | *** join/#asterisk dj015 (n=damjan@dsl-243-129-165.telkomadsl.co.za) |
| 08:31.02 | JT_ | sure, in pastebin |
| 08:31.32 | Jason[1] | ok. |
| 08:31.49 | dj015 | why is the RTPAUDIOQOS variable set only when the sip phone dialling asterisk hangs up, but not when asterisk hangs up? |
| 08:32.11 | Jason[1] | just to confirm...u need the display when I run modprobe wct4xxp. right? |
| 08:32.17 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
| 08:32.27 | JT_ | yes |
| 08:32.34 | JT_ | and when you run ztcfg |
| 08:32.35 | Jason[1] | k |
| 08:32.45 | Jason[1] | k |
| 08:32.47 | JT_ | so the end of the dmesg, from whenever related stuff starts |
| 08:32.47 | Jason[1] | brb |
| 08:35.23 | JT | likewise, dinner time |
| 08:36.13 | dj015 | are the hangup functions (eg. sip_hangup() ) called before or after the h extension? |
| 08:37.42 | *** join/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
| 08:37.44 | *** join/#asterisk hellop (n=hellop@udp115314uds.hawaiiantel.net) |
| 08:38.02 | *** part/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
| 08:41.59 | *** join/#asterisk hads (n=hads@reef80.anchor.net.au) |
| 08:42.20 | *** part/#asterisk hads (n=hads@reef80.anchor.net.au) |
| 08:46.08 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 08:46.55 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 08:46.55 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 08:49.07 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 08:57.22 | JT_ | back |
| 08:58.04 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 09:02.27 | *** join/#asterisk mitcheloc (n=mitchelo@titaniumsoft.net) |
| 09:03.16 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 09:09.09 | angryuser | how to use BLF for fxs Channels? |
| 09:13.50 | Jason[1] | hi JT. http://www.pastebin.ca/325503 |
| 09:13.56 | *** part/#asterisk inspired (n=mikael@cl-330.sto-01.se.sixxs.net) |
| 09:14.04 | Jason[1] | only has msg from ztcfg -vvvv |
| 09:14.17 | Jason[1] | modprobe does not display any msg |
| 09:14.50 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 09:15.38 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 09:15.50 | sevard | you should try lsmod |
| 09:17.28 | *** join/#asterisk T42X (n=tadas@omega.vic.lt) |
| 09:18.19 | Jason[1] | lsmod? |
| 09:18.52 | *** join/#asterisk Aces1Up (n=really@ip68-96-235-198.lv.lv.cox.net) |
| 09:23.56 | *** part/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 09:24.48 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 09:25.03 | Jason[1] | JT. got to go. bb |
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| 09:38.31 | *** join/#asterisk yassine (n=yassine@dsl.voicint.com) |
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| 09:50.25 | Aces1Up | i had a crazy idea the other day just wondering if its possible with asterisk. |
| 09:50.57 | JT | probably, or maybe not |
| 09:51.02 | JT | depends how it is |
| 09:51.28 | *** join/#asterisk The_Ball (n=alex@58.163.133.108) |
| 09:52.12 | The_Ball | I want my parents to be able to dial my * box by putting in my domain name, which softphone should I use which is very simple in use and setup? |
| 09:52.12 | Aces1Up | if you had multiple broadband connections spread out around a city, each running an asterisk pbx, if you designated say 1 main asterisk box to recieve all incoming calls. could you transfer an incoming call to another box and then call out on that box with the just transferred call? Basically this would allow you to share all these broadbands connections together. |
| 09:52.31 | JT | Aces1Up: yes |
| 09:53.21 | Aces1Up | jt wow, and on the box that initially recieved the incoming call, one it transferred the call, that bandwidth would be free to recieve additional calls? |
| 09:54.05 | JT | sorry, your question is not clear enough to me |
| 09:54.14 | Aces1Up | hrmm. |
| 09:54.25 | Aces1Up | can i im you? |
| 09:54.44 | The_Ball | JT, i think what he's asking if the first asterisk box will proxy the call |
| 09:55.03 | Aces1Up | the_ball yeh thats sounds bout right. |
| 09:55.09 | JT | is it necessary? try and explain, especially with an ascii diagram |
| 09:55.11 | *** join/#asterisk jm|laptop (n=jamie@dilbert.jamiem.com) |
| 09:55.28 | Aces1Up | but i don't want the first box to bandwidth to be tied at all to that call once its transferred to another box. |
| 09:56.05 | Aces1Up | i want to be able to share the bandwidth of multiple broadband connections to push calls out. |
| 09:56.10 | JT | how will the first box receive incoming calls? |
| 09:56.49 | Aces1Up | DID from carrier terminating calls at the * box. |
| 09:58.01 | JT | sorry, not at all specific enough |
| 09:58.11 | JT | DIDs can be delivered over tonnes of technologies |
| 10:00.55 | Aces1Up | SIP |
| 10:02.45 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.94) |
| 10:06.43 | Aces1Up | it will be delivered via sip to the first box. |
| 10:07.00 | JT | yes quite possibly with SIP reinvites set to yes |
| 10:08.40 | L|NUX | hello |
| 10:08.47 | L|NUX | I have a DID vendor that wants me to be able to generate specific SIP |
| 10:08.47 | L|NUX | error messages under certain conditions and I'm completely stumped on |
| 10:08.47 | L|NUX | how to do these: |
| 10:09.04 | L|NUX | They want to see a SIP 503 error response(service unavailable) when they send the call in to an active extension and and the service is not available, I don't have a clue on how to simulate this. |
| 10:09.10 | L|NUX | any idea |
| 10:09.14 | Aces1Up | hey jt, you ever heard of a no usage free DID provider? |
| 10:09.34 | L|NUX | When they send in a call to an extension that doesn't exist they want to see a SIP 100 TRYING message before the receive the 404 NOT FOUND error. Currently I have only been able to generate the 404 error. |
| 10:10.35 | JT | Aces1Up: no |
| 10:11.10 | L|NUX | any one have any idea |
| 10:12.03 | Aces1Up | jt i have this provider that says he can get me DID's throughout the world for a fixed price, no usage costa, they just drop the calls off at my ^ box, does that sound right? |
| 10:13.37 | JT | Aces1Up: sure if there's only incoming calls, sounds right |
| 10:13.40 | JT | makes sense |
| 10:14.18 | *** join/#asterisk SoftIce (n=newbie@vc-196-207-45-253.3g.vodacom.co.za) |
| 10:14.45 | SoftIce | hi, can somebody tell me can I use an isdn modem on my asterisk box ? |
| 10:14.51 | SoftIce | or do I need to purchace some bri card? |
| 10:14.52 | JT | modem? |
| 10:15.01 | SoftIce | isdn modem yes |
| 10:15.01 | Aces1Up | jt hrmm... guess it seems weird to me, i mean if someone is making a call from mexico using one of these did's and it just drops the call to me into my box, that means i don't pay for it at all? |
| 10:15.02 | JT | card probably |
| 10:15.23 | JT | Aces1Up: except for the monthly fee and bandwidth usage, yes |
| 10:15.29 | SoftIce | JT people say you can use an isdm modem |
| 10:15.32 | *** join/#asterisk vgster (n=vgster@88-96-191-174.dsl.zen.co.uk) |
| 10:15.33 | Aces1Up | jt wow. |
| 10:15.35 | SoftIce | are you 100% sure I cant use an isdn modem |
| 10:15.43 | Aces1Up | how the heck can they get away with that? |
| 10:16.25 | JT | Aces1Up: err, they use the Internet to deliver calls, it's cheaper, but not as reliable as using the PSTN |
| 10:17.05 | Aces1Up | jt so they just have a POP in those locations then/ |
| 10:17.17 | JT | yes, a few PRIs and servers |
| 10:17.24 | Aces1Up | hrmm ok. |
| 10:17.25 | *** join/#asterisk Dibbler_XP (n=Dibbler@host217-45-198-229.in-addr.btopenworld.com) |
| 10:19.31 | Aces1Up | so if i wanted to use one of these DID's to bring calls into my box from Mexico and then transfer them onto say an end user's cell phone i could do that? |
| 10:20.24 | JT | yes |
| 10:21.07 | Aces1Up | jt what would be the cheapest delivery system to the cell phone, i'm at a loss here... would a voip line be the cheapest? |
| 10:21.37 | JT | depends on your country, but most likely |
| 10:22.23 | Aces1Up | jt would u use an unlimited plan or use a per-min plan? just curious. |
| 10:22.44 | JT | that really depends on your country |
| 10:22.59 | JT | here in australia we never pay for incoming calls unless they're 1800 |
| 10:23.16 | SoftIce | hmm, anyone know if I can order an ISDN line with 2 B channels, and have 1 used for a say phone number and use the other B channel for adsl ? or are those mixed technoligies? |
| 10:23.28 | JT | SoftIce: no |
| 10:23.42 | JT | i hear some countries support isdn and adsl on the same line |
| 10:23.56 | JT | he whole idea of B channels is virtual |
| 10:24.08 | JT | it's just one digitally modulated data stream |
| 10:24.24 | SoftIce | ahh, i see |
| 10:30.06 | *** join/#asterisk Bazy (n=bazy@eclipse.upcnet.ro) |
| 10:34.19 | SoftIce | what is the cheapest card i can use then JT? |
| 10:34.23 | SoftIce | a cheap hfc card? |
| 10:43.05 | *** join/#asterisk angryuser (n=Miranda@i03v-213-44-169-43.d4.club-internet.fr) |
| 10:44.33 | tzafrir | right |
| 10:44.33 | angryuser | so after 2 days of searching over the net, i have not found any info about Asterisk BLF(led blinking) support for FXS ports, i am very deceived of asterisk |
| 10:45.19 | angryuser | i mean the mous basic pbx has this option |
| 10:45.26 | angryuser | moust* |
| 10:46.55 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 10:46.55 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 10:47.48 | tzafrir | angryuser, what are you angry about? |
| 10:48.00 | tzafrir | ;-) |
| 10:48.41 | jeremy_g | tzafrir:which school do you go to? |
| 10:48.45 | jeremy_g | oh damn |
| 10:48.51 | angryuser | Busy lamp field is not working for FXS"s |
| 10:48.59 | jeremy_g | angryuser:which school do you go to? |
| 10:49.36 | angryuser | jeremy_g: why? |
| 10:49.51 | jeremy_g | angryuser: just wondering |
| 10:50.58 | angryuser | jeremy_g:i dont know its equivalent on english;) |
| 10:51.50 | *** join/#asterisk J4k3 (i=jsuter@dhcp-12-197-128-58.intrastar.net) |
| 10:51.53 | angryuser | but its IT lan/network most of the time |
| 10:52.18 | jeremy_g | angryuser:i thought you would say something else |
| 10:52.46 | angryuser | jeremy_g:like what? |
| 10:53.58 | jeremy_g | angryuser:why are you angry all the time? |
| 10:54.30 | tzafrir | angryuser, FXSs (phones) or FXOs (telco lines)? |
| 10:54.33 | angryuser | tzafrir: yea |
| 10:54.41 | tzafrir | both? |
| 10:55.09 | angryuser | tzafrir: i need BLF only for fxs's |
| 10:55.39 | angryuser | jeremy_g:not all the time;) |
| 10:55.53 | tzafrir | I don't have a phone with such a lamp, but do have quite a few FXSs. How can I test the Asterisk part? |
| 10:56.27 | SoftIce | please can somebody tell me the cheapest bri card I can buy |
| 10:56.42 | tzafrir | hfc-s based? |
| 10:56.51 | SoftIce | yes |
| 10:57.07 | SoftIce | there was a link to an italian site |
| 10:57.09 | SoftIce | but its dead now |
| 10:58.16 | angryuser | tzafrir: after some research i think is it impossible, but anyway, for SIP it is simple you need like exten=>200,hint,SIP/200 exten=>200,1,Dial(SIP/200) in your extensions.conf |
| 10:58.57 | SoftIce | what is the .hint ? |
| 10:59.01 | SoftIce | err ,hint ? |
| 10:59.02 | angryuser | in sip.conf if you have aster 1.4 call-limit=n where n limit of device |
| 10:59.34 | angryuser | SoftIce:google for 'blf asterik' |
| 11:01.56 | angryuser | tzafrir:and after that in [general] >>sip.conf subscribecontext=context where sip users configured |
| 11:03.33 | angryuser | angryuser: forgot to tell you that cal-limit=n must be added to each sip account for blf use |
| 11:04.03 | angryuser | angryuser: so after, reboot, then 'core show hints' |
| 11:04.35 | tzafrir | I saw quite a few model names (e.g: bilion). I believe that Billion. serach ebay for "HFC" for starters. This will give you some names. |
| 11:06.29 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 11:06.29 | J4k3 | hfc gets you airsoft guns... isdn and hfc gets you one quad isdn card thats like $300 |
| 11:06.33 | moon06 | hi everyone |
| 11:07.04 | J4k3 | the quad card seems kinda silly to me. I dunno about everywhere else, but its bordering on cheaper to get a PRI with 8 channels lit than it is four BRIs with 8 channels lit |
| 11:07.55 | JT | err |
| 11:08.24 | *** join/#asterisk Murdock__ (n=brian@dsl-217-155-77-209.zen.co.uk) |
| 11:08.40 | JT | in countries where bri is common, it's usually 10chans minimum for fractional E1 |
| 11:09.18 | JT | and multiport cards can be used for more than connecting tonnes of PSTN lines |
| 11:09.33 | JT | you can also make them act as the exchange so you can connect and existing ISDN pabx |
| 11:10.23 | moon06 | I got a problem with isdn / capi |
| 11:10.35 | J4k3 | well lack of a cheap ISDN card caused me to do more looking for a cheap incoming voip provider in my area, which I eventually found. by my math it works out cheaper than doing it the ISDN way. |
| 11:10.40 | moon06 | my hard/soft sip phones show "ringing" but i don't have any ringtone |
| 11:11.09 | moon06 | what's weird is that it works when using an IAX softphone ... |
| 11:11.30 | moon06 | I mean I can hear the ringtone when calling thru the same isdn line |
| 11:12.03 | JT | J4k3: HFC cards are cheap as all hell |
| 11:12.20 | JT | yes voip is usually cheaper |
| 11:12.26 | JT | not usually more reliable |
| 11:12.29 | J4k3 | JT: I never found any cheap ones... url me, I'd like a backup |
| 11:12.52 | SoftIce | yes, please can somebody show me where I can get a cheap hfc card |
| 11:13.27 | JT | J4k3: search ebay for isdn, look at something like an avm fritz or anything else that is hfc cologne based |
| 11:13.29 | J4k3 | all I use the x101p's for in my box is two pots lines for emergency outgoing and backup incoming... they almost never get hit, but I notice they generate a HUGE count of interrupts/second. |
| 11:13.49 | JT | x101p, heh |
| 11:15.27 | J4k3 | junk! |
| 11:15.42 | J4k3 | they generate those interrupts when idle. |
| 11:15.49 | J4k3 | the box seems.... easily annoyed |
| 11:15.50 | JT | hmm |
| 11:16.17 | J4k3 | if top shows less than 85% idle, I notice hiccups in the sound quality |
| 11:16.24 | J4k3 | especially when using the voicemail |
| 11:16.39 | J4k3 | doubly if I call in using my cdma phone |
| 11:16.46 | JT | hah |
| 11:17.04 | SoftIce | how much does x101p go for? |
| 11:17.10 | J4k3 | SoftIce: $6? $7? |
| 11:17.17 | J4k3 | beads and trinkets? |
| 11:18.06 | SoftIce | what 6 dollars for a hfc card? |
| 11:18.23 | *** join/#asterisk aload (n=user@mail.vwes.pl) |
| 11:18.24 | JT | no, analogue |
| 11:18.37 | SoftIce | oh, those are analogue cards |
| 11:18.47 | SoftIce | i need the cheapest hfc card |
| 11:19.03 | JT | just see what you can find |
| 11:19.21 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 11:19.31 | SoftIce | well i cant find anything |
| 11:19.33 | SoftIce | thats why im asking |
| 11:19.52 | JT | search ebay for isdn |
| 11:19.59 | JT | look at the cheap pci cards |
| 11:29.18 | *** join/#asterisk E-bola (i=bola@rbii-valhalla.mrseb.co.uk) |
| 11:29.21 | E-bola | Hello |
| 11:29.35 | E-bola | Do anybody use the X-lite client with asterisk? Im wondering how you transfer calls? |
| 11:30.05 | SoftIce | x-lite softphone/ |
| 11:30.18 | E-bola | ya |
| 11:30.19 | SoftIce | use the FOP panel :) |
| 11:30.37 | anonymouz666 | FOP is nice |
| 11:30.50 | SoftIce | customize it the way you want it. |
| 11:31.04 | E-bola | thats not too userfriendly |
| 11:31.10 | E-bola | to have to have a webpage open |
| 11:31.13 | SoftIce | what isn't |
| 11:31.19 | SoftIce | you drag and drop |
| 11:31.23 | SoftIce | how hard is that/ |
| 11:31.46 | E-bola | its not hard to use |
| 11:31.49 | E-bola | i said its not userfriendly |
| 11:31.56 | E-bola | its very intrusive to have to have that big a window open |
| 11:32.00 | SoftIce | in what sense? |
| 11:32.01 | *** join/#asterisk florz (i=nobody@2002:58c6:2592:1:0:0:0:2) |
| 11:32.03 | E-bola | it cant minimize to system tray? |
| 11:32.27 | SoftIce | that is all how you configure your browser setting |
| 11:32.32 | SoftIce | some can toggle to a tray |
| 11:32.42 | SoftIce | you can customine the fop pannel to look like you want it |
| 11:32.48 | SoftIce | if you want the operator to have flowers, put it there |
| 11:32.55 | SoftIce | change the buttons, add extension you want, etc |
| 11:32.57 | E-bola | i still think its very counter intuitive to use 2 programs |
| 11:32.59 | E-bola | to handle ur calls |
| 11:33.12 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 11:33.29 | E-bola | hmm seems you have to buy x-lite to get transfer functionality |
| 11:33.34 | SoftIce | well then setup extension to forward calls, etc |
| 11:33.52 | SoftIce | and if you want to do call pick up's add that to your dialplan to |
| 11:34.10 | E-bola | that doesnt work if a call is already in progress? |
| 11:34.11 | SoftIce | exten => *12 .... extension number |
| 11:34.18 | E-bola | im talking ont he clientside |
| 11:34.43 | SoftIce | what do you mean, the clientside works how you setup the server side |
| 11:34.52 | SoftIce | client is restricted to what you allow it to do |
| 11:35.01 | E-bola | So say i got a call going, on my softphone |
| 11:35.03 | SoftIce | its not a 'phone' functionality its how you configure your dialplan |
| 11:35.10 | E-bola | if i just press a number asterisk will intercept that and forward it? |
| 11:35.21 | E-bola | i dont think a dialplan is active when u got a call going? |
| 11:35.35 | angryuser | yes,yes yes, blf is working for FXS, asterisk is great;) |
| 11:35.45 | zoa | E-bola: try zoiper :p |
| 11:36.09 | E-bola | zoa: ok googling that... |
| 11:36.22 | E-bola | ahh looks nice |
| 11:36.24 | E-bola | beta though |
| 11:36.33 | E-bola | cool that it supports iax though |
| 11:36.35 | E-bola | do you use it? |
| 11:36.40 | zoa | i made it :) |
| 11:36.54 | E-bola | nice :) |
| 11:36.59 | zoa | it will go out of beta in a week or 2 max |
| 11:37.46 | E-bola | but its commercial too? |
| 11:37.57 | zoa | yes |
| 11:38.06 | zoa | it will have a free and a paid version |
| 11:38.27 | E-bola | what will the difference be? i cant find it in the faq |
| 11:39.02 | zoa | i dont know for sure yet |
| 11:39.20 | zoa | paid version will have outlook integration, t.38 fax, g729, and some other things |
| 11:39.34 | zoa | possibly also the native conferencing |
| 11:39.35 | E-bola | if u pull out the transfer call functionality of the free versions, its no better than xlite |
| 11:39.35 | E-bola | hehe |
| 11:39.39 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 11:39.47 | zoa | actually it will be better :) |
| 11:39.52 | E-bola | zoa: have you tested it with snap? |
| 11:39.55 | *** join/#asterisk Ebola (n=Ebola@host81-152-204-157.range81-152.btcentralplus.com) |
| 11:40.03 | zoa | nopez |
| 11:42.29 | *** join/#asterisk RoyK (n=roy@213.160.242.90) |
| 11:46.55 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 11:46.55 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 11:52.41 | JT | why are all the "better" softphones non-free? :( |
| 11:52.50 | zoa | because it takes a lot of money to make them |
| 11:52.55 | JT | (i say better because none are really good) |
| 11:53.05 | JT | uhuh, you could say the same about any software |
| 11:53.12 | JT | the only licensing issue is g.729 |
| 11:53.16 | zoa | yes |
| 11:53.30 | JT | i'm sure it's not just a g.729 fee |
| 11:53.32 | zoa | well, i have 5 developpers working on hours |
| 11:53.36 | zoa | ours i mean |
| 11:53.46 | JT | sure |
| 11:54.12 | zoa | so its either no softphone or a commercial one :) |
| 11:54.47 | JT | err there's plenty out there already |
| 11:54.59 | zoa | yesw |
| 11:55.00 | zoa | yes |
| 11:55.06 | aload | exit |
| 11:55.08 | zoa | we are now aiming to become nr 2 |
| 11:55.18 | *** part/#asterisk aload (n=user@mail.vwes.pl) |
| 11:55.24 | zoa | the counterpath people spend 5 million $ a year on operational costs |
| 11:55.28 | JT | where's the big market? |
| 11:55.29 | zoa | it will be hard to compete with that |
| 11:55.42 | zoa | there is no big market |
| 11:55.47 | JT | right |
| 11:58.27 | *** join/#asterisk n0rus (i=n0rus@tunnel-45-129.vpn.uib.no) |
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| 12:28.55 | moon06 | Got a *big* problem with my Asterisk : my sip soft/hardphone shows "ringing" when calling when I call someone but no ringtone .... |
| 12:28.55 | moon06 | I got the ringtone when calling with IAX, and when using a sip account to call |
| 12:28.55 | *** join/#asterisk inspired (n=mikael@cl-330.sto-01.se.sixxs.net) |
| 12:29.18 | *** part/#asterisk inspired (n=mikael@cl-330.sto-01.se.sixxs.net) |
| 12:32.18 | *** join/#asterisk chiang_sg (i=kodok@bb121-6-186-250.singnet.com.sg) |
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| 12:33.07 | *** part/#asterisk amarok77 (n=alex@a80-126-72-122.adsl.xs4all.nl) |
| 12:33.50 | tzafrir | moon06, can you call a test extension on Asterisk? e.g: an echo test? |
| 12:34.14 | tzafrir | Is this soft phone on the same LAN as Asterisk? |
| 12:34.18 | moon06 | yep |
| 12:34.28 | moon06 | for the 2nd question |
| 12:35.26 | moon06 | tzafrir, when calling a local sip phone I got the ringtone |
| 12:35.37 | moon06 | it seems only to be when using my ISDN line |
| 12:35.45 | chiang_sg | hi, i just setup 2 * servers, and connected via iax2. I have inserted switch => IAX/xx:xx@1.2.3.4/default, how do i confirm that the "switch" statement is correct from the CLI? |
| 12:35.53 | moon06 | but the thing is that it happened without changing anything |
| 12:36.55 | tzafrir | moon06, for starters, separate the SIP issue and the ISDN issue: can you call from both to echo test in Asterisk? |
| 12:37.14 | tzafrir | or demonstrate bidirectional communication with Asterisk in any other way |
| 12:37.20 | moon06 | don't have set up an echo test in asterisk |
| 12:38.42 | moon06 | tzafrir, when I make a call w\ using my ISDN phone line, I do have the ringtone |
| 12:39.47 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 12:40.06 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqia.cable.mindspring.com) |
| 12:42.35 | Murdock__ | Hi all. I would like to know if it is possible for a SIP phone to be passed the caller ID when you steal a call (using the pickupexten). They display **3, and my app that watches the phones doesnt get the caller id of the stolen call |
| 12:43.52 | moon06 | tzafrir, here's what I got : |
| 12:43.53 | moon06 | <PROTECTED> |
| 12:43.53 | moon06 | <PROTECTED> |
| 12:43.53 | moon06 | <PROTECTED> |
| 12:43.53 | moon06 | <PROTECTED> |
| 12:43.53 | moon06 | <PROTECTED> |
| 12:44.20 | moon06 | but I don't hear a ringtone ... |
| 12:46.55 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 12:46.55 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
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| 12:48.14 | chiang_sg | is there any command i can confirm "switch" is working fine in extensions.conf ? |
| 12:49.22 | backblue | moon06: use misdn. |
| 12:50.53 | moon06 | backblue, instead of ? |
| 12:52.43 | *** join/#asterisk shodan (n=shodan@ip022.99-113-216.pppoe4.joliette.intermonde.net) |
| 12:56.02 | *** join/#asterisk J4k3- (i=jsuter@dhcp-12-197-128-58.intrastar.net) |
| 13:01.48 | backblue | moon06: capi. |
| 13:04.12 | *** join/#asterisk Makenshi (n=chaz@kenshi.fox.furry.be) |
| 13:06.43 | JT | backblue: lol, use bristuff, even better |
| 13:06.48 | JT | stuff misdn for a joke |
| 13:07.21 | *** join/#asterisk sfinxen (i=sfinx@148-168-233-83.relakks.com) |
| 13:08.01 | sfinxen | I know this is a asterisknow specifik question, but i have some problems, http://rafb.net/p/2VzAJs95.html |
| 13:08.37 | JT | moon06: btw, use pastebin to paste in future |
| 13:08.39 | JT | ~pb |
| 13:08.47 | jbot | somebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 13:28.08 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 13:35.01 | *** join/#asterisk AstaWerksDotCom (n=doug@63.161.96.170) |
| 13:35.40 | AstaWerksDotCom | ok so i am meeting with cisco this morning. my boss has his mind set on cisco call manager. what questions should i ask cisco to make them feel stupid compaing to asterisk ? |
| 13:40.39 | *** join/#asterisk reber (i=reber@gateway/tor/x-141f0e2a8638ef0b) |
| 13:41.03 | *** join/#asterisk FaithX (n=faithful@ns.linuxterminal.com) |
| 13:41.07 | tzafrir | AstaWerksDotCom, how many phones? |
| 13:41.50 | *** join/#asterisk jimmy_deanPB (n=jhodapp@209.131.196.174) |
| 13:42.10 | robl^ | AstaWerksDotCom: ask them if they support IAX ;-) |
| 13:42.59 | AstaWerksDotCom | :) |
| 13:43.06 | AstaWerksDotCom | dunno yet |
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| 14:01.23 | Murdock__ | Hi all. I would like to know if it is possible for a SIP phone to be passed the caller ID when you steal a call (using the pickupexten). They display **3, and my app that watches the phones doesnt get the caller id of the stolen call |
| 14:02.36 | Corydon76-home | No it is not possible |
| 14:02.38 | *** join/#asterisk dasenjo (n=dasenjo@190.5.196.149) |
| 14:02.44 | dhill | [Jan 23 09:01:51] WARNING[24930]: chan_sip.c:4791 process_sdp: Unsupported SDP media type in offer: image 18494 udptl t38 |
| 14:02.49 | dhill | should i be getting that on 1.4.0 ? |
| 14:03.15 | dhill | i thought 1.4.0 support t38 |
| 14:03.19 | dhill | supported |
| 14:03.30 | oej | dhill: Only if you enable it in sip.conf |
| 14:03.59 | dhill | ahhh |
| 14:05.33 | *** join/#asterisk mut (n=ana@65.111.222.120) |
| 14:05.53 | mut | doesn't dial jump if the line is busy or something |
| 14:05.57 | mut | or just it just go next? |
| 14:06.16 | mut | like.. how do i do a dial() and if it returns busy, execute Busy() |
| 14:07.09 | *** join/#asterisk bkw_ (n=brian@adsl-70-143-45-86.dsl.tul2ok.sbcglobal.net) |
| 14:10.38 | dhill | t38pt_udptl=yes under sip peer config |
| 14:10.41 | dhill | still ssame issue |
| 14:11.00 | *** join/#asterisk ez` (n=Ez@c66.203.210-59.clta.globetrotter.net) |
| 14:12.07 | *** join/#asterisk phsultan (n=phsultan@PO-47165.rocqadm.inria.fr) |
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| 14:15.47 | *** join/#asterisk doolph (n=doo@200.124.21.169) |
| 14:16.20 | doolph | If I want to connect to another pbx, like a panasonic what hardware do i Need FXO or FXS? |
| 14:16.30 | doolph | or both |
| 14:19.05 | phsultan | Id like to know if RTP packetization time setting (in sip.conf or other) works under P2P bridging mode |
| 14:20.01 | DrukenLPY | doolph: would depend on the pbx |
| 14:20.18 | doolph | depending on what ? |
| 14:20.38 | doolph | its analog |
| 14:20.55 | doolph | I wish call from asterisk and call to asterisk |
| 14:22.22 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-221.ph.ph.cox.net) |
| 14:22.48 | The_Ball | does the "s" extension mean "start"? |
| 14:24.17 | tzafrir | doolph, can you connect a normal phone there and get a dialtone? |
| 14:24.23 | doolph | yes |
| 14:24.43 | tzafrir | so you need an FXO |
| 14:24.55 | tzafrir | A phone is an FXO |
| 14:25.09 | doolph | really |
| 14:25.32 | *** join/#asterisk unik-rados (n=rados@c-68-62-71-239.hsd1.mi.comcast.net) |
| 14:25.48 | doolph | but if I want to call from panasonic to an asterisk extension? |
| 14:26.04 | doolph | I'll need a FXS port too eh |
| 14:27.25 | tzafrir | doolph, or an IVR menu :-( |
| 14:27.34 | *** part/#asterisk naitram (n=danny@216.77.58.40) |
| 14:27.35 | doolph | ummmmm |
| 14:27.38 | doolph | ok i got it now |
| 14:27.43 | doolph | i send the call to an ivr |
| 14:27.49 | doolph | thats right |
| 14:27.50 | doolph | lol |
| 14:27.53 | doolph | I dont need fxs |
| 14:27.59 | backblue | JT: no, misdn it's far better than bristuff. |
| 14:30.12 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:30.12 | *** mode/#asterisk [+o anthm] by ChanServ |
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| 14:35.19 | *** join/#asterisk s1gny|wrk (n=s1gny@p54915C3A.dip.t-dialin.net) |
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| 14:38.09 | *** part/#asterisk s1gny|wrk (n=s1gny@p54915C3A.dip.t-dialin.net) |
| 14:40.27 | dhill | have there been any udptl/t38 patches since 1.4.0? |
| 14:40.46 | dhill | sipura 2100 -> asterisk passthrough -> cisco as5300 |
| 14:41.01 | dhill | t38 is seen on asterisk and as5300 |
| 14:41.10 | dhill | fax just doesn't go through.. or even attempt to send |
| 14:46.16 | brad_mssw | dhill: are you going through NAT? do you have the 3.3 firmware for the sipura? |
| 14:46.44 | dhill | through nat, yes |
| 14:46.48 | dhill | 3.3.6 on the 2100 |
| 14:46.54 | dhill | nat is bad news? :) |
| 14:46.55 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 14:46.56 | brad_mssw | ok, NAT is not supported with the 2100 afaik |
| 14:46.56 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 14:47.05 | brad_mssw | for t38 |
| 14:47.16 | dhill | ok, thanks for the tip.. let me go try without nat :) |
| 14:47.19 | brad_mssw | dhill: http://www.voip-info.org/tiki-index.php?page=Asterisk%20T.38 there's a table here |
| 14:48.04 | brad_mssw | dhill: also, I think t38pt_udptl might be a global sip.conf option only (e.g. under the general section) ... don't know if it works under a peer config |
| 14:48.16 | dhill | ok, thanks |
| 14:48.24 | dhill | bbiab :) |
| 14:54.20 | Gido-E | [Jan 23 15:46:20] WARNING[30719]: chan_misdn.c:5171 chan_misdn_log: Extension can never match, so disconnecting |
| 14:54.30 | Gido-E | why sould my extension never match? |
| 14:54.44 | *** join/#asterisk ^BanDiT^ (n=_BanDiT_@bzq-219-170-2.static.bezeqint.net) |
| 15:00.19 | *** join/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 15:01.54 | brodiem | Does anyone know how * handles an agent that is logged into two different queues? Say if the agent has a penalty of 1 in one queue and penalty of 2 in another -- does it take this into consideration when routing calls or do queues not "talk" to each other to determine any routing logic? |
| 15:02.56 | in-pt | Hi all |
| 15:03.12 | in-pt | Does anyone knows if cisco 7940 supports sip ??? |
| 15:03.27 | unik-rados | I'm trying to build an app that would automatically conference people in based on an alert of system outage. so basically asterisk gets notified that there's a problem, goes out and gets a list of people to call, calls them and then puts them all in MeetMe. What's the best approach to do that? |
| 15:03.32 | brodiem | in-pt yes |
| 15:03.33 | *** join/#asterisk jmls (n=asterisk@62.49.235.130) |
| 15:03.54 | in-pt | brodiem: thanks |
| 15:03.56 | jmls | anyone been able to use sipp to test voicemail ? |
| 15:04.07 | brodiem | unik-rados you could use .call files to call each person, with the destination channel a meetme ext |
| 15:04.11 | jmls | I want to see how many simultaneous voicemails can be left |
| 15:05.08 | unik-rados | brodiem: thanks! I'll take a look at this. I'm sort of new to asterisk so I'll read up on .call files first. |
| 15:05.25 | brodiem | unik-rados yep there's lots of info on it at voip-info.org, it's pretty easy |
| 15:05.42 | backblue | Gido-E: because your dialplan dont match?! (maybe...) |
| 15:06.21 | unik-rados | brodiem: yep, just found something. Thanks for pointing me in this direction! I really appreciate it! |
| 15:06.26 | brodiem | no problem |
| 15:09.38 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:09.43 | dhill | still no go |
| 15:09.50 | dhill | SIP/2.0 489 Bad event |
| 15:09.53 | dhill | i am seeing |
| 15:12.35 | *** join/#asterisk djflux (n=djflux@mm.shermfin.com) |
| 15:13.26 | *** join/#asterisk mercestes (n=merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 15:25.16 | Gido-E | backblue ! |
| 15:26.12 | Gido-E | backblue it goes into the right context but cant find the extension. |
| 15:26.16 | Gido-E | OK! |
| 15:33.34 | *** join/#asterisk h0 (i=fakhir@unaffiliated/fakhir) |
| 15:34.48 | *** join/#asterisk jimmy_deanPB (n=jhodapp@209.131.196.174) |
| 15:39.10 | *** join/#asterisk jamessan (n=jamessan@debian/developer/jamessan) |
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| 15:42.34 | jamessan | does anyone know if there's a way to hook into app_voicemail (Comedian Mail)? I'm working on a framework built around Asterisk and we provide other means of accessing voicemail aside from dialing the voicemail app. unfortunately, that means we've been re-writing our own way to forward/delete/play/etc voicemail |
| 15:44.06 | *** join/#asterisk ManxPower (n=manxpowe@191.sub-75-202-49.myvzw.com) |
| 15:44.24 | *** join/#asterisk supjigatr (n=syslod@152.53.16.10) |
| 15:44.32 | brodiem | jamessan if you already wrote code for fwd/del/playing msgs what are you trying to do now? |
| 15:44.46 | ManxPower | Today In Trenton's Life: When Good Upgrades Go Bad. |
| 15:45.32 | jamessan | brodiem: I'd rather just be able to tell app_voicemail to do it so I'm sure that I'm not going to accidentally step on Asterisk's toes if we're trying to do the same thing at the same time |
| 15:46.23 | *** join/#asterisk adop (n=adonis@ppp131-090.dsl.hol.gr) |
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| 15:46.55 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 15:47.03 | supjigatr | Anyone used asterisk or another device/software for GR-303 <-> SIP gateway? |
| 15:47.10 | *** join/#asterisk sahafeez (n=sahafeez@ip68-6-215-70.sd.sd.cox.net) |
| 15:48.19 | *** join/#asterisk Manfish1 (n=themanfi@usere117.dsl.pipex.com) |
| 15:49.41 | Manfish1 | is it possible to configure asterisk to use mpg123 as its default, its driving me mad that everytime an announcement is made in a queue it jumps to the begining of the track |
| 15:50.12 | Manfish1 | sorry version is 1.2.14 |
| 15:50.18 | supjigatr | I just converted everything to native Manfish1 |
| 15:51.07 | Manfish1 | but in native I cant get it to pause the mp3 just jump back to the start each time |
| 15:51.25 | *** join/#asterisk Deeewayne (i=dwayne@nat/digium/x-cf3747d4ec7b2f46) |
| 15:51.40 | *** part/#asterisk adop (n=adonis@ppp131-090.dsl.hol.gr) |
| 15:51.52 | supjigatr | Like queue announcments causing MOH to jump? |
| 15:52.02 | Manfish1 | yes |
| 15:52.05 | *** join/#asterisk reber (i=reber@gateway/tor/x-0c8411b5e3e6131a) |
| 15:52.11 | supjigatr | I don't think ours does that. |
| 15:52.13 | brodiem | Does anyone know how * handles an agent that is logged into two different queues? Say if the agent has a penalty of 1 in one queue and penalty of 2 in another -- does it take this into consideration when routing calls or do queues not "talk" to each other to determine any routing logic? |
| 15:52.46 | supjigatr | U running 1.4? |
| 15:53.05 | Manfish1 | an announcement in a queue, will cause moh to restart the track, similar with putting someone on hold for the second time it also starts a new track |
| 15:53.15 | Manfish1 | no 1.2.14 |
| 15:53.50 | *** join/#asterisk LeadCrayon (n=leadcray@flpn-sentry.natcotech.com) |
| 15:53.59 | LeadCrayon | Good morning. |
| 15:54.26 | brodiem | supjigatr nope |
| 15:54.27 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 15:54.44 | LeadCrayon | I'm having trouble dialing out on my PRI line. I've seen some documents that say adding a pause to the outbound dial prefix 'w+' will help but I'm not using AAH or TrixBox. What does that translate to in a standard Asterisk install? |
| 15:55.29 | brodiem | LeadCrayon normally the wait is used on a POTS type trunk |
| 15:55.31 | ManxPower | native music on hold plays the MOH from the start for each channel. mpg123 MOH uses 1 process for all calls in a MOH class |
| 15:55.33 | brodiem | shouldn't need with a PRI |
| 15:55.50 | ManxPower | LeadCrayon: w is not needed on PRI |
| 15:55.51 | ManxPower | EVER |
| 15:55.59 | mercestes | I hate AT&T |
| 15:56.15 | ManxPower | LeadCrayon: check the value of HANGUPCAUSE as the priority after the Dial |
| 15:56.34 | |Vulture| | a w will actually hurt on a PRI |
| 15:56.36 | brodiem | LeadCrayon its basically to allow a pots line to stabalize before dialing digits.. |
| 15:57.19 | mercestes | I think hell was created specifically for people like AT&T |
| 15:58.21 | Manfish1 | so asterisk 1.2.14 uses Format_MP3? |
| 15:58.41 | brodiem | ManxPower maybe you know the answer to this question? : |
| 15:58.44 | brodiem | Does anyone know how * handles an agent that is logged into two different queues? Say if the agent has a penalty of 1 in one queue and penalty of 2 in another -- does it take this into consideration when routing calls or do queues not "talk" to each other to determine any routing logic? |
| 15:58.49 | LeadCrayon | ManxPower: A pri debug shows a Disconnect Request |
| 15:59.03 | ManxPower | LeadCrayon: That is not what I said to watch. |
| 15:59.13 | ManxPower | brodiem: I try to not deal with agent stuff |
| 15:59.16 | JunK-Y | Manfish1: no, its in asterisk-addons |
| 15:59.59 | ManxPower | LeadCrayon: ALL calls, even failed ones will end with a Disconnect Request |
| 16:00.18 | ManxPower | What you want is the CAUSE CODE of the disconnect, which is stored in HANGUPCAUSE |
| 16:00.40 | LeadCrayon | How do I dump that so I can see it? |
| 16:01.27 | Manfish1 | so is it possible for me to tell asterisk to use mpg123 player instead of Format_MP3? |
| 16:01.44 | Manfish1 | or change the behaviour of the MOH? |
| 16:03.02 | JunK-Y | Manfish1: see musiconhold.conf |
| 16:03.15 | Manfish1 | thats what I am looking at |
| 16:03.33 | JunK-Y | in 1.2, the default in mp123 |
| 16:03.39 | *** join/#asterisk marv[work] (n=timr@24.214.206.254) |
| 16:05.15 | Manfish1 | ok so if I install mpg123 it should start to work as it did before? sorry but I have only just moved to asterisk from trixbox and I am still learning |
| 16:05.38 | *** join/#asterisk RoyKa (n=roy@213.160.242.90) |
| 16:06.35 | LeadCrayon | The problem may be different than I described it originally anyway. On my ISDN phone, when I dial a certain 888 number it goes through, on Asterisk I get the tri-tone and "Your call did not go through" |
| 16:07.01 | ManxPower | LeadCrayon: Asterisk does not by default send a tri-tone |
| 16:07.12 | ManxPower | LeadCrayon: You need to look at your dialplan |
| 16:07.44 | LeadCrayon | I figured so but when I had dialplan settings in zapata.conf, they said they were being ignored on load. |
| 16:07.44 | Murdock__ | When dialing the pickupexten, is it possible to get the callerid of the call you're picking up onto the phone you picked it up from? |
| 16:08.22 | ManxPower | LeadCrayon: that message means that "you cannot change that setting on a reload, only on a restart" |
| 16:08.24 | ManxPower | it is not a problem |
| 16:08.39 | LeadCrayon | Ah. |
| 16:08.45 | *** join/#asterisk atapi (n=virgill4@c-65-34-182-167.hsd1.fl.comcast.net) |
| 16:09.44 | ManxPower | LeadCrayon: you have an extensions.conf problem |
| 16:10.08 | ManxPower | in the priority after Dial put a Noop(HANGUPCAUSE is ${HANGUPCAUSE}) then watch the console during a failed call |
| 16:10.29 | *** join/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue) |
| 16:10.30 | LeadCrayon | Okay, doing that now. |
| 16:15.18 | *** part/#asterisk jamessan (n=jamessan@debian/developer/jamessan) |
| 16:19.59 | LeadCrayon | Well, nuts. Now I'm getting a "Unable to reconfigure channel 1-23" and no zaptel |
| 16:20.14 | *** part/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
| 16:26.26 | Vec | If I use G711a on my voip phones will no tronscoding take place when the data leaves my asterisk box(through the zap interface, digium card) on an E1 link ? |
| 16:26.47 | Vec | My E1 link uses G711a as far as I understand. |
| 16:27.12 | docelmo | No no transcodeing will take place |
| 16:27.21 | docelmo | TDM links are only Ulaw |
| 16:28.55 | Vec | docelmo : I think mine is Alaw (in europe) ? |
| 16:28.55 | docelmo | yes it is |
| 16:28.55 | docelmo | but even if you wanted to convert to ULAW it doesnt transcode much |
| 16:28.55 | docelmo | it just changes the structure |
| 16:29.04 | *** join/#asterisk cygar (n=cygar@201.216.200.33) |
| 16:29.12 | cygar | hello |
| 16:29.19 | docelmo | goodbye |
| 16:29.21 | Vec | docelmo : so it would only consume a very small cpu usage |
| 16:29.42 | docelmo | actually not many more than if you ran ALAW -> ALAW |
| 16:29.58 | docelmo | type show translations |
| 16:30.14 | docelmo | you will see how long it takes to transcode 1 frame from a single codec to another |
| 16:30.29 | coppice | the voice quality will be slightly better if you don't transcode |
| 16:31.43 | docelmo | Thank you coppice this is very ture.. |
| 16:32.08 | cygar | One question, I have a grandstream FXS well registered in my asterisk box that makes outbound/inbound calls perfectly. The problem is that it does not generate its dialtone. As far as i know this dialtone is generated in its own device and not asterisk sending RTP. Can anyone give me a hint on what's going on? [ i have seen the grandstream conf and there's nothing like 'dialtone' ]. If there's the posibility of generating the fake dialtone from * sending as |
| 16:32.24 | JunK-Y | coppice: ive a quick question related to dsp.c, for talk detection, how it works exactly? is it a certain amount of dB? |
| 16:32.44 | docelmo | I thought 1.4 was supposed to have t.38 pass thru.. Anyone know about this? |
| 16:33.21 | JunK-Y | it is, apparently. |
| 16:33.37 | docelmo | cygar the only time asterisk generages dialtone its for tdm fxs links not voip |
| 16:33.45 | coppice | JunK-Y: i don't know what dsp.c does. its probably just energy measurement, which isn't talk detection at all. its just noise detection |
| 16:33.47 | *** join/#asterisk montag___ (n=montag@h213-217-178-161.mi2.albacom.net) |
| 16:33.54 | Vec | thanks docelmo |
| 16:33.57 | docelmo | and cant help you otherwise soory.. grandstream ATA's suck |
| 16:34.15 | docelmo | Well not suck they do a good job but there are alot of better ones out there |
| 16:34.23 | docelmo | no prob vec |
| 16:34.25 | cygar | docelmo: I know, but i have a client who is complaining about that... so if i can fake a dial tone it would help anyway :) |
| 16:34.27 | *** join/#asterisk reber (i=reber@gateway/tor/x-90c0d6985edb567f) |
| 16:34.43 | JunK-Y | coppice: and how that noise detection occurs exacly? |
| 16:34.45 | coppice | the grandstreams are good for showing up the bugs in other people's kit :-) |
| 16:34.58 | montag___ | hi, i've a TDM400 connected to my analogic PABX, i've a sip client, it's possibile to use a codec like GSM for calls ? When i try to force the GSM codec on SIP client i got codec error, i can only use ulaw....it's a TDM400 limit ? |
| 16:35.16 | docelmo | I dont know of anyway to force a dialtone.. in throry the ATA is just supposed to make dialtone on registration |
| 16:35.20 | Qwell[] | coppice: if the phone crashes, there's a problem with the other SIP stack? :p |
| 16:35.45 | docelmo | montag___ no its a limit of your configuration |
| 16:35.53 | *** part/#asterisk jmls (n=asterisk@62.49.235.130) |
| 16:35.58 | docelmo | readup on the wiki about codecs and sip |
| 16:36.22 | coppice | i've been finding the grandstream ATAs good for flushing out bugs in my code and other people's. they tend to follow the specs accurately, but push them further than other people |
| 16:36.35 | docelmo | Qwell say do you know anything about asterisk 1.4 supporting t38? |
| 16:36.41 | Qwell[] | in passthrough |
| 16:36.48 | docelmo | I dont see a module for it |
| 16:36.55 | cygar | second question, last saturday I have a problem chaning from R2 to ISDN signalling [ what should be really easy and have done several times ], I could get inbound calls through the E1 but when trying to make outbound calls i got Q.931 error 31 [ normal, unspecified ] . I haven't seen any stuff wrong either in zapata/zaptel/dialplan... any idea ? |
| 16:36.56 | docelmo | is it built in? |
| 16:37.38 | Qwell[] | docelmo: it's not a module |
| 16:37.41 | Qwell[] | it's just part of SIP |
| 16:37.48 | docelmo | ohh ok cool |
| 16:37.51 | JunK-Y | docelmo: look for a struct ast_udptl |
| 16:38.04 | docelmo | do I need to do an allow=t38 or anything? |
| 16:38.09 | docelmo | or just force t38? |
| 16:38.30 | Qwell[] | I think just allow=t38, yeah |
| 16:39.30 | docelmo | ok sweet |
| 16:40.29 | montag___ | docelmo: i've tried to force gsm use on sip.conf site, but when i dial zaptel extension i got ulaw call... |
| 16:41.06 | *** join/#asterisk kannan (n=kannan@58.68.25.67) |
| 16:41.09 | *** join/#asterisk CrossRoad (n=SilentVa@207.47.18.18.static.nextweb.net) |
| 16:41.47 | LeadCrayon | I don't believe this. I ran ztcfg and now my zap interfaces don't show up in Asterisk anymore. What the heck? |
| 16:43.11 | cygar | Has anyone used Digivoice cards with R2 ? |
| 16:43.26 | *** join/#asterisk jarrod (i=nobody@dont.juniperyour.net) |
| 16:43.37 | jarrod | anyway on a cisco7960 to have it display caller-id on a second call? |
| 16:43.44 | jarrod | thats coming in |
| 16:43.55 | coppice | ctgar: are you in .br? |
| 16:44.07 | coppice | cygar: are you in .br? |
| 16:44.16 | cygar | coppice: yes |
| 16:44.37 | coppice | cygar: how much does a digivoice card cost there? |
| 16:45.28 | *** join/#asterisk santibiotico (n=santi@169.Red-88-15-142.dynamicIP.rima-tde.net) |
| 16:46.01 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
| 16:46.32 | cygar | coppice: 600 dollars aproximately |
| 16:46.55 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 16:46.55 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 16:47.08 | coppice | cygar: and how much is a digium or sangoma card? I heard there are huge import duties |
| 16:49.10 | cygar | coppice: yes, even if I am from Argentina and living in brazil just a couple of months ago I have always bought in digium so it's not too much a matter of money difference. Here [ in .ar too ] is still used r2 signalling so the good stuff about this cards is that it support its directly through the card without the need of using unicall. |
| 16:49.40 | *** join/#asterisk naitram (n=danny@216.77.58.40) |
| 16:49.43 | coppice | it seems quite cheap for a card with onboard processing |
| 16:49.45 | caio1982 | there's the brazilian Khomp board with R2 as well, which seems a little cheaper/better than digivoice's... although digivoice stuff is open source |
| 16:50.43 | cygar | caio1982: I've never used it. |
| 16:51.03 | cygar | caio1982: are you from .br? I am having a weird problem with a digivoice and r2 . |
| 16:51.13 | coppice | yeah, i've seen the khomp card as well |
| 16:51.52 | caio1982 | I successfully installed a board from them in a customer asterisk box... running for more then a month without much problems. the people from khomp are very inclined to solve every problem we face... but, stil... digivoice's stuff are 100% open source, different from khomp |
| 16:52.09 | caio1982 | coppice: what do you think about both cards? |
| 16:52.50 | *** join/#asterisk joelsolanki (i=joelsola@220.224.77.140) |
| 16:52.59 | coppice | i have no comment about them. they claim to do echo cancellation and other processing on the card. if they do that for US$600 its sounds like a good approach |
| 16:54.08 | coppice | the digium and sangoma cards could be much cheaper, if they wanted them to be. Its certainly easy to make a single E1 intelligent card and sell it for <$600 |
| 16:54.23 | cygar | caio1982: do you know how it works the dtmf stuff when sending through SIP and then how the r2 driver handles the dtmf that was sent through SIP to send it in r2? Sometimes it works and sometimes it doesn't when sending through r2 [ the dtmf ]. |
| 16:55.14 | caio1982 | coppice: when the sangoma folks came to brazil for the estacaovoip.com.br event, people were talking about they head-hunting people from khomp to perhaps embbed some technology for R2 in their boards |
| 16:55.19 | caio1982 | but i could not confirm that |
| 16:55.30 | caio1982 | unfortunately :) |
| 16:56.30 | coppice | the R2 stuff can be acquired from various places, like me. their pricing for a card with on board processing would be the issue. there is actually no reason at all to put the R2 on the card. its doesn't take much processing |
| 16:56.41 | caio1982 | cygar: people in the asteriskbrasil.org mailing list faced issues like that. I'd look in its archives as a place to start from |
| 16:57.47 | caio1982 | coppice: aren't you supposed to sell R2 stuff for a lower price since you living in china? :-) |
| 16:58.06 | coppice | actually, I think it would be stupid for sangoma to put R2 onto their board. its things like EC that need to be there |
| 16:58.09 | cygar | caio1982: I'd take a look, thanks. |
| 16:58.20 | caio1982 | I mean, you have better access for microchips and eletronic components than brazilian companies |
| 16:58.27 | coppice | I can't cut the price you get my R2 code for now |
| 16:59.35 | caio1982 | weren't you talking about boards and chips for R2 via hardware? i didnt get the code part |
| 17:01.19 | coppice | you could have an E1 card without echo cancellation for <$100 in medium volumes |
| 17:03.06 | caio1982 | after having many issues with regular boards, i believe that quality comes first and that the price of the board is (sad to say) secondary :/ |
| 17:03.31 | coppice | quality and price are not closely related |
| 17:03.31 | caio1982 | agreed |
| 17:03.55 | coppice | how much per month is E1 rental? |
| 17:04.54 | mut | fie dolla! |
| 17:05.11 | mut | anyting u want |
| 17:05.28 | *** join/#asterisk RoyK (n=roy@213.160.242.49) |
| 17:05.38 | RoyK | rotfl. "Note: This does not work in 0.7.1 (maybe other versions)" (http://www.voip-info.org/wiki/index.php?page=Asterisk+variable+hangupcause) |
| 17:05.47 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 17:05.49 | caio1982 | i dont have a clear idea on how much brazilian telcos ask for a E1 monthly |
| 17:06.13 | coppice | its probably enough to make the card cost seem less important |
| 17:06.14 | anonymouz666 | how can I measure the latency of a Digium card? |
| 17:07.20 | coppice | now, when you are supplying E1 ports to the telco, they act like every dollar is sacred, and screw you down to a few dollars per port :-) |
| 17:07.57 | caio1982 | hehe |
| 17:08.45 | docelmo | anonymouz666 take it to the tallest building you can find and drop it. then measure how long to takes to hit bottom |
| 17:09.37 | anonymouz666 | funny |
| 17:09.43 | docelmo | I try |
| 17:14.28 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 17:14.52 | *** join/#asterisk ClydeGoffe (n=cgoffe@ool-4351f665.dyn.optonline.net) |
| 17:16.43 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 17:20.39 | *** join/#asterisk wltjr (n=wltjr@gentoo/developer/wltjr) |
| 17:21.18 | *** join/#asterisk grandy (n=chatzill@c-71-198-130-108.hsd1.ca.comcast.net) |
| 17:21.26 | wltjr | anyone know off hand what Received Reserved LoopBack code 04! means from a Sangoma A101, connected to a PRI for * |
| 17:23.29 | *** join/#asterisk Ritalin2 (n=dave@c-68-47-199-178.hsd1.tn.comcast.net) |
| 17:24.16 | wltjr | here is a paste if it helps at all http://rafb.net/p/DOFWgt37.html |
| 17:24.21 | fetcher | wltjr: telco doing tests on your circuit? They can send a message through the FDL channel requesting a remote loopback |
| 17:24.45 | fetcher | but normally would only do that after opening a trouble ticket |
| 17:25.14 | wunderkin | i requested testing of a circuit, and they put the wrong circuit number in the ticket, so another customer got dinged :P |
| 17:25.57 | Ritalin2 | any idea what would cause a call to drop when I try and flash for callwaiting? |
| 17:26.23 | Ritalin2 | (on call, call waiting beeps, flash the hook to switch over, first call drops and it answers the second call) |
| 17:27.46 | wltjr | fetcher: I am not sure as to the status of the circuit, trying to get in touch with ISP on that, it's a new one, and we are at an odd point in install |
| 17:28.03 | wltjr | fetcher: they freak when customers provide their own equipment, more so when it's not a channel bank ;) |
| 17:28.40 | wltjr | fetcher: but also not sure if it's a cabling problem, I am not on site, but client ran their own cable, should be ok, and at first was not using a cross over cable, one is in place now, but error message is the same |
| 17:29.05 | wltjr | ISP, funny, Telco I meant, CLEC in this case :) |
| 17:29.08 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 17:30.45 | slima | hmm, czy ja co¶ ¼le robie? http://pastebin.ca/325846 |
| 17:32.50 | slima | oops, wrong window, sorry. |
| 17:35.14 | dhill | does * support modem relay? |
| 17:35.41 | Nivex | owww... my terminal aches at the 8859-1 |
| 17:37.24 | nakeee | I'm having hard time finding docs about asterisk authentication capabilities, can I get the user/password/ip in the module? |
| 17:41.01 | ClydeGoffe | Hey does anyone use sipphone as their Sip provider and using virtual numbers? |
| 17:42.07 | ClydeGoffe | I'm having an issue where when i recieve calls using my virtual number my softphone rings and although the person can hear me I can't hear them |
| 17:42.10 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 17:42.43 | ClydeGoffe | I'm not sure if this is related to sipphone, virtual numbers or asterisk but that's the issue i'm having...any ideas? |
| 17:42.46 | kannan | how select default ringing tone for outbound calls to standard USA style? i hope anyone can guide me with this |
| 17:44.58 | *** join/#asterisk nvicf (n=nvicf@201.250.166.200) |
| 17:45.36 | *** part/#asterisk Aces1Up (n=really@ip68-96-235-198.lv.lv.cox.net) |
| 17:46.28 | *** join/#asterisk jimmy_deanPB (n=jhodapp@209.131.196.174) |
| 17:46.55 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 17:46.55 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 17:47.53 | kannan | i mean some calls outbound gives a ring tone that sounds like a long busy tone |
| 17:48.18 | kannan | which i would like to change all to ring .. ring style as per all the other calls style |
| 17:52.32 | Dr-Linux | file: |
| 17:55.43 | Ritalin2 | I'm looking at info on keywords in the zapata.conf documentation and it mentions "these are for use only with (non-PRI) T1 lines" what's the diff between a T1 and a non-PRI T1 ? |
| 17:56.52 | wltjr | Ritalin2: pri will have a data channel, t1 won't, t1 all channels are used for data |
| 17:57.26 | dhill | for modem passthrough.. what payload-type should i set on the as5300? |
| 17:58.13 | Ritalin2 | wltjr: thanks. |
| 17:58.22 | dhill | to work through as5300 |
| 18:01.50 | *** join/#asterisk topping (n=topping@204.152.96.50) |
| 18:14.12 | Ritalin2 | call waiting seems like it should just work |
| 18:18.13 | *** join/#asterisk high-rez (i=gus@carrera.bourg.net) |
| 18:18.28 | high-rez | Question: Can anyone recommend a good software SIP Phone (for windows) ? |
| 18:19.54 | mquin | x-lite / eyebeam seems pretty decent |
| 18:20.25 | zoa | www.zoiper.com :p |
| 18:20.37 | zoa | tell me what you don't like about it and i will have it changed :) |
| 18:20.47 | mquin | ooh, IAX2 - cute :) |
| 18:20.59 | zoa | it has sip too |
| 18:21.21 | high-rez | zoa: My buggest concern is proper echo cancelation. |
| 18:21.33 | zoa | i need one more week for that |
| 18:21.38 | zoa | working on it |
| 18:21.44 | zoa | but mine will be the best :) |
| 18:22.17 | high-rez | zoa: Also, a nice feature is the ability to preconfigure the client and send it as a bundle, so when someone on the other end installs it, everything is configured :D |
| 18:23.00 | file | oh no it's zoa |
| 18:24.02 | *** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
| 18:24.02 | FuriousGeorge | hey all |
| 18:24.59 | Supaplex | zoa: will it allow me to choose which sound card/alsa device I want to use? |
| 18:25.01 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-155-21-116.red.bezeqint.net) |
| 18:25.25 | sevard | zoa: does your softphone chew 182mb of ram like xLITE |
| 18:25.29 | zoa | high-rez: working on that |
| 18:25.30 | sevard | LITE is a misnomer. |
| 18:25.31 | *** join/#asterisk angryuser (n=Miranda@i03v-213-44-169-43.d4.club-internet.fr) |
| 18:25.40 | angryuser | good evening;) |
| 18:25.47 | zoa | sevard, it runs good on a very tiny pc |
| 18:25.52 | sevard | oh, fun, trial beta. |
| 18:25.53 | zoa | with a via cpu |
| 18:26.02 | zoa | i will remove that tomorrow maybe |
| 18:26.10 | zoa | it was to avoid people from sending old bug reports |
| 18:26.23 | sevard | what will it cost after the trial? |
| 18:26.30 | angryuser | is it possible to change language on voicemail and meetme from default EN to something els? |
| 18:26.31 | zoa | there will be 2 versions |
| 18:26.33 | zoa | free and paid |
| 18:26.50 | sevard | differences? |
| 18:27.13 | zoa | outlook integration, conferencing, g729, t.38 |
| 18:27.13 | angryuser | else* |
| 18:27.14 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 18:27.45 | sevard | this isn't another sip client written in java, is it? |
| 18:28.04 | zoa | nopez |
| 18:28.12 | zoa | its delphi with C library |
| 18:28.20 | zoa | and the linux / mac are in C++ |
| 18:29.02 | Supaplex | so where's the mac/linux downloads? |
| 18:29.27 | zoa | coming |
| 18:29.40 | zoa | they dont produce audio atm so thats not really worth putting online :) |
| 18:30.07 | Supaplex | ahh, busticatedwares. gotcha. |
| 18:30.09 | zoa | hopefully that will be fixed next week |
| 18:30.34 | *** join/#asterisk k-man__ (n=jason@ppp244-232.static.internode.on.net) |
| 18:30.45 | zoa | some deadlock in the linux part :/ |
| 18:33.42 | nakeee | tzafrir_laptop, hey I haven't noticed you are on the channel:) |
| 18:34.07 | tzafrir_laptop | nakeee, welcome! |
| 18:34.25 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:34.36 | blitzrage | hey all! Suppose no one here is using a 7970 w/ Asterisk? |
| 18:35.04 | blitzrage | My issue is that I can place a call, but the phone is not trying to register. "Register with Proxy" is set to Yes. |
| 18:35.27 | tzafrir_laptop | nakeee, I feel quite alone in asterisk-il |
| 18:35.30 | blitzrage | I've looked at tethereal, and it really isn't sending anything to port 5060, but if I try to make a call, it does |
| 18:36.54 | angryuser | is it possiblr to change default language for meetme() application? |
| 18:38.12 | *** join/#asterisk clyrrad (n=ddd@CPE001195f553c7-CM0011aea484a4.cpe.net.cable.rogers.com) |
| 18:39.27 | robin_sz | so, where best to ask about mISDN ? |
| 18:39.34 | robin_sz | here or isdn4linux? |
| 18:41.39 | *** join/#asterisk Nobbie (n=no@mtngprs1.mtn.co.za) |
| 18:41.42 | Nobbie | heya =) |
| 18:42.27 | Nobbie | is it possible to use the microphone jack of a soundcard as the source of music on hold ? |
| 18:42.54 | Nobbie | does one use /dev/dsp as an input file ? |
| 18:43.05 | Qwell[] | Nobbie: it might work... try it out |
| 18:43.14 | Nobbie | don't have access to a box right now ... =( |
| 18:44.00 | angryuser | Nobbie: hehe asterisk disco dance;) |
| 18:45.16 | tzafrir_laptop | Nobbie, you can always use sox as a mediator |
| 18:45.22 | Nobbie | anything will be bettter then the current |
| 18:45.38 | Nobbie | mmm, what's this. chan_oss, chan_alsa. maybe that could help ? |
| 18:46.12 | sevard | i've seen people use line-in as a source for MoH, but for the life of me I can't recall what method they used. |
| 18:46.21 | sevard | have you tried voip-info? |
| 18:46.25 | *** join/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
| 18:46.49 | Nobbie | busy checking there now |
| 18:46.55 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 18:46.55 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 18:47.43 | *** join/#asterisk chandi (n=burni13@modemcable085.149-201-24.mc.videotron.ca) |
| 18:47.50 | sevard | http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf |
| 18:47.55 | sevard | under: Using a sound card as the source. |
| 18:49.21 | sevard | heh |
| 18:49.35 | blitzrage | heh |
| 18:49.50 | Nobbie | thanks |
| 18:49.54 | Nobbie | lol |
| 18:49.57 | chandi | Hi guys, I've just upgraded to 1.4.0 from 1.2 and my .call are not dialed anymore. I get those 2 errors : cdr.c:509 ast_cdr_disposition: Cause not handled |||||||||| and : pbx_spool.c:341 attempt_thread: Call failed to go through, reason 1. |
| 18:50.42 | Nobbie | chandi: try the new "originate" command from the manager interface |
| 18:50.45 | *** join/#asterisk arkadi (n=arkadi@87.246.143.140) |
| 18:50.59 | chandi | Nobbie I'm trying but I can't get it to do something else than show me the help ;) |
| 18:51.12 | wwalker | "No application 'Monitor' for extension" - I get this when MixMonitor() is hit in the dialplan. I've loaded app_mixmonitor.so already... ??? |
| 18:51.20 | Nobbie | chandi: have you tried reading it then ? ;) |
| 18:51.49 | blitzrage | wwalker: and show application mixmonitor gives you something? |
| 18:52.06 | chandi | Nobbie hahaha yup! my syntax is : originate SIP/ab002414 15142331123 cellout |
| 18:52.19 | blitzrage | wwalker: and Monitor() and MixMonitor() are separate applications |
| 18:52.53 | Nobbie | chandi: try: originate SIP/ab002414 extensions 15142331123 cellout |
| 18:52.56 | blitzrage | wwalker: it seems to me you are calling the Monitor() application in your dialplan. Perhaps you didn't run 'extensions reload' |
| 18:53.06 | Nobbie | chandi: or: originate SIP/ab002414 extension 15142331123@cellout |
| 18:53.08 | chandi | Nobbie ahh, it was the "extension" word that I didn't get |
| 18:53.13 | Nobbie | same here |
| 18:53.16 | Nobbie | took me a while |
| 18:53.18 | chandi | Nobbie it says : WARNING[5270]: cdr.c:509 ast_cdr_disposition: Cause not handled |
| 18:54.00 | Nobbie | cdr error. *shrug* |
| 18:54.10 | Nobbie | maybe you can ignore that since it's only an error |
| 18:54.14 | Nobbie | a warning i mean |
| 18:54.45 | Nobbie | will the new "core" commands in 1.4.0 cause problems for FreePBX 2.1.2 ? |
| 18:54.47 | chandi | Nobbie that's what I thought but the calls don't get through |
| 18:55.17 | wwalker | blitzrage: yup, I started out wanting to use Monitor (because it builds the filename instead of me having to provide it) and I changed 2 out of 3 locs in the dialplan |
| 18:55.32 | *** join/#asterisk mcab (n=mb@66.195.193.151) |
| 18:55.52 | wwalker | is Monitor part of core asterisk? I'd rather use it, but there is no app_monitor.so :( |
| 18:57.04 | *** join/#asterisk h0 (i=fakhir@unaffiliated/fakhir) |
| 18:57.09 | blitzrage | Monitor is just an application in Asterisk, like MixMonitor, yes |
| 18:57.13 | chandi | Nobbie have you got an idea ? |
| 18:58.36 | wwalker | nm, figured it out (res_monitor vs app_monitor) |
| 19:00.18 | LeadCrayon | Anyone have an idea why 800, 866 and 877 numbers would dial fine but 888 numbers would not. They're all configured exactly the samein the dialplan and I've poked pridialplan as many ways as I can think of and nothing helps. |
| 19:01.38 | *** join/#asterisk dlynes_laptop (n=dlynes@S0106001346f7843f.vc.shawcable.net) |
| 19:01.50 | clyrrad | ok quick question |
| 19:01.51 | clyrrad | I know there is a way to send a SIP NOTIFY command to a remote extension to instruct it to RESYNC with a provisioning server - I am not sure how you actually send that command via the CML in asterisk - does anyone here know how to do that? |
| 19:02.11 | blitzrage | ClydeGoffe: sip notify |
| 19:02.16 | blitzrage | errr... clyrrad |
| 19:02.44 | clyrrad | ah so just sip notify peer, and that will tell it to rysnc? |
| 19:03.06 | clyrrad | oh i see n/m |
| 19:03.10 | Nobbie | chandi: busy trying to get mine working ... |
| 19:03.13 | clyrrad | there are differnt messages that can be sent |
| 19:03.18 | chandi | Nobbie ahah ok |
| 19:03.24 | clyrrad | blitzrage: thanks |
| 19:03.34 | Nobbie | chandi: are you loading oss or alsa at all ? |
| 19:03.43 | chandi | Nobbie nope, not supposed to |
| 19:08.55 | Nobbie | chandi: it provides a console channel and a command: dial which may be an alternative ? |
| 19:11.40 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 19:13.48 | Nobbie | chandi: any detials why it's failing ? |
| 19:13.54 | dlynes_laptop | ~log |
| 19:13.59 | jbot | well, log is as piece of wood, or a record, or the opposite of exponentiation |
| 19:14.05 | dlynes_laptop | ~irclog |
| 19:14.07 | jbot | well, irclog is apt/ibot/jbot/purl all log to http://ibot.rikers.org/<channelname>/ where channelname is html encoded ie: %23debian | lines that start with a space are not shown | some channels have stats at http://ibot.rikers.org/stats/<channelname>.html.gz, or updated "nightly" |
| 19:14.41 | chandi | Nobbie I don't understand your question : it provides a console channel and a command: dial which may be an alternative ? |
| 19:15.05 | chandi | Nobbie I've no idea why it fails. I've got an ATA that uses the same dial plan/context and it works fine |
| 19:15.44 | chandi | what's the command in CLI to see all the active calls ? |
| 19:15.47 | chandi | in 1.4 |
| 19:15.56 | Nobbie | chandi: core show channels |
| 19:17.02 | *** join/#asterisk JerJer[mobile] (n=jj@208.65.54.162) |
| 19:17.15 | chandi | Nobbie if a call is made through asterisk but from a SIP phone and a SIP provider. Should it show on core show channels ? |
| 19:17.51 | Nobbie | chandi: i would think so. or you can use "sip show channels" to shows sip channels only |
| 19:18.10 | chandi | Nobbie nothing shows.. |
| 19:18.20 | Nobbie | when you try the originate ? |
| 19:18.22 | blitzrage | the channel has to be active |
| 19:18.53 | *** join/#asterisk J4k3 (i=jsuter@dhcp-12-197-128-58.intrastar.net) |
| 19:19.00 | Nobbie | but ringing channels will show tooo |
| 19:19.13 | dlynes_laptop | ~dell |
| 19:19.15 | jbot | Dude! Are you getting a Dell? |
| 19:19.24 | dlynes_laptop | sigh |
| 19:20.06 | chandi | Nobbie : calls don't get through with originate |
| 19:20.13 | chandi | Nobbie and nothing shows |
| 19:20.20 | chandi | on core show channels |
| 19:20.20 | chandi | never ever ;) |
| 19:20.22 | chandi | sad |
| 19:20.36 | Nobbie | i'm also having that problem, trying to figure it out |
| 19:21.17 | Nobbie | ahh, here we go |
| 19:22.58 | Nobbie | the originate command dials the tech/data first and requires a pickup, then it calls the extensions or application specified and if it answers, bridges the 2 |
| 19:23.21 | chandi | what'S a tech/data ? |
| 19:23.33 | Nobbie | chandi: so, in your example, SIP/ab002414 was probably ringing yes ? and once you answer it, you will see the channels in core show channels |
| 19:23.52 | Nobbie | chandi: is that what you require ? or do you want to be able to dial from the CLI directly ? |
| 19:24.04 | Nobbie | chandi: in your case, tech/data is SIP/ab002414 |
| 19:24.15 | chandi | nobbie : SIP/ab002414 is the sip provider/channel I use to dial to PSTN |
| 19:24.19 | in-pt | Hi all |
| 19:24.25 | Nobbie | chandi: what are you trying to do |
| 19:24.32 | in-pt | i have a little problem with voicemail configuration |
| 19:24.37 | chandi | Nobbie make automatic calls with a call file |
| 19:25.09 | in-pt | i had configured maxmessage length of voicemail to be 600 seconds but it gets diconnect all the recordings in 16 seconds |
| 19:25.13 | chandi | Nobbie : I was working in * 1.2 |
| 19:25.17 | in-pt | what could be the reason |
| 19:27.29 | *** join/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 19:27.50 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqu6.cable.mindspring.com) |
| 19:28.15 | Nobbie | chandi: what's your SIP channel for the ATA ? |
| 19:28.28 | EmleyMoor | Has anyone here got VoIPtalk working over IAX with ability to call others' VoIPtalk IDs directly, receive such calls yourself and use facilities like "withhold number"?# |
| 19:28.48 | chandi | Nobbie :chandi123c |
| 19:28.58 | chandi | sip/chandi123c |
| 19:29.25 | Nobbie | chandi: try originate SIP/chandi123c extension 15142331123@cellout |
| 19:29.44 | chandi | k |
| 19:30.02 | clyrrad | in sip_notify.conf does anyone know if Event=>resync is the proper syntax to have a Sipura ATA resync? |
| 19:31.15 | blitzrage | clyrrad: the only one I've ever got working was the Polycom |
| 19:31.46 | clyrrad | blitzrage: yea becase Event=>resync does not cause the SPA to resync |
| 19:31.52 | clyrrad | I wish there was documentation on this |
| 19:31.59 | *** join/#asterisk angom (n=angom@red-corp-201.130.139.211.telnor.net) |
| 19:33.10 | clyrrad | what happened? |
| 19:33.34 | blitzrage | clyrrad: there probably is from Sipura somewhere |
| 19:33.42 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:34.03 | Nobbie | chandi: something other then the callfiles must be wrong with your setup. seems to be a CDR issue, i suggest you start looking there. or set verbose way up and paste the flow at pastebin.ca |
| 19:34.43 | chandi | Nobbie I've just found out that I had setAccount commands in extension.conf which are not valid in 1.4 |
| 19:35.23 | Nobbie | are they replaced by something new ? |
| 19:35.30 | chandi | I don't know |
| 19:35.47 | *** join/#asterisk sfinxen (i=sfinx@148-168-233-83.relakks.com) |
| 19:36.02 | Nobbie | you can set the CDR info in your call file though |
| 19:36.39 | *** join/#asterisk jimmy_deanPB (n=jhodapp@209.131.196.174) |
| 19:39.49 | chandi | Nobbie : have you got an idea about this error ? : pbx_spool.c:341 attempt_thread: Call failed to go through, reason 1 |
| 19:40.03 | chandi | I'd really like to know what reason 1 is |
| 19:40.17 | chandi | I've read the source code but it doesn't help understanding the issue |
| 19:41.07 | *** join/#asterisk sjwilliamson (i=swilliam@209.42.110.46) |
| 19:41.17 | EmleyMoor | The only thing they seem to tell you how to handle with IAX is PSTN calls |
| 19:41.54 | Nobbie | gotta go, sorry |
| 19:41.56 | EmleyMoor | The "withhold number" method they get doesn't work with IAX either |
| 19:42.02 | *** join/#asterisk pekdon (n=pekdon@85.196.83.77) |
| 19:42.11 | EmleyMoor | er, give, even |
| 19:43.21 | pekdon | Hello! Straight onto a question, is it possible to get the number of calls in a queue in the dialplan? Want to skip an initial playback if it is empty |
| 19:46.55 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 19:46.55 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 19:49.55 | *** join/#asterisk Dibbler_ (n=Dibbler@dsl-217-155-254-174.zen.co.uk) |
| 19:50.35 | *** join/#asterisk nicklinn (n=ratfink@mrtc-dsl-610214.mis.net) |
| 19:50.55 | *** join/#asterisk Drac0 (n=kconley@66.184.203.18) |
| 19:51.58 | Drac0 | Hello... Anyone alive here? ;) |
| 19:52.00 | *** join/#asterisk riddlebox (n=riddlebo@24-207-167-95.dhcp.stls.mo.charter.com) |
| 19:52.43 | *** join/#asterisk ru_wing (n=wing@82.179.129.5) |
| 19:53.07 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
| 19:53.22 | *** join/#asterisk kgx (n=kgx@60.234.20.178) |
| 19:55.29 | clyrrad | Has anyone got sip notify working with Sipura / Linksys products? |
| 19:56.00 | clyrrad | the sipura documentation states you can send an Event: resync header to the device to cause it to resync |
| 19:56.30 | clyrrad | in sip_notify.conf i have this Event=>resync |
| 19:57.06 | clyrrad | Asterisk sais that it sends the notify command to the ATA - but the ATA never acutally does anything |
| 19:57.13 | *** join/#asterisk EricDGlobeTroote (n=eric@200.122.149.57) |
| 19:57.50 | dlynes_laptop | clyrrad: does the device already have the same current info? |
| 19:58.01 | dlynes_laptop | clyrrad: if it does, it shouldn't reboot |
| 19:58.12 | clyrrad | no I made a change to the configuration profile |
| 19:58.19 | clyrrad | so it should do the resync |
| 19:58.25 | dlynes_laptop | clyrrad: and you recompiled it? |
| 19:58.32 | clyrrad | its not compiled |
| 19:58.35 | clyrrad | its xml |
| 19:58.37 | clyrrad | over https |
| 19:58.46 | dlynes_laptop | clyrrad: oh...never done xml on those |
| 19:58.54 | dlynes_laptop | clyrrad: I've only ever done the compiled cfg files |
| 19:58.57 | clyrrad | its the same as the compiled profiles |
| 19:58.59 | Drac0 | I've been instructed to find out what the minimal cost would be to connect two branch offices using Asterisk, both locations use digital lines (T1) for voice. Is there a specific digital interace card that I should be looking at? |
| 19:59.08 | clyrrad | just differnt way of getting the data there |
| 19:59.16 | clyrrad | how are you forcing your resync? |
| 19:59.25 | dlynes_laptop | clyrrad: i'm not |
| 19:59.25 | clyrrad | IE.... what do you have in your sip_notify.conf? |
| 19:59.28 | clyrrad | ah |
| 19:59.52 | *** part/#asterisk rpm (n=russell@66.119.170.34) |
| 19:59.54 | Drac0 | Also, I understand that Asterisk can run on lower end systems, but would would be considered an average system to realistically use in a business environment? |
| 20:00.11 | clyrrad | it just seem as if Asterisk does not actually send the notify header.... strange indeed |
| 20:00.43 | dlynes_laptop | Drac0: you want either a sangoma a101u (single port pri, no hwec), sangoma a102u (dual port pri, no hwec), sangoma a102d (dual port pri, hwec), or a digium te110p/te105p/te111p/... |
| 20:01.01 | dlynes_laptop | Drac0: you'll have to read up on the specs so that you know the specifics of what you want |
| 20:01.02 | sfinxen | is it easy to setup asterisk on a debian freshinstall? |
| 20:01.14 | sfinxen | perhaps an easy howto is in order, tip? |
| 20:01.27 | dlynes_laptop | sfinxen: it's all available via aptget, albeit not the latest versions |
| 20:01.34 | EmleyMoor | sfinxen: Yes, very. I'd start by installing the packages you need (easy) then use the book |
| 20:01.36 | EmleyMoor | ~book |
| 20:01.38 | jbot | methinks book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 20:01.54 | dlynes_laptop | clyrrad: did you do a sip debug? |
| 20:02.03 | sfinxen | nice, is it hard to configure? i seems to have som problems with asterisknow and trixbox :/ |
| 20:02.08 | dlynes_laptop | clyrrad: to make sure of your assumption? |
| 20:02.11 | *** join/#asterisk signius (n=IceChat7@dsl-217-155-69-102.zen.co.uk) |
| 20:02.11 | Drac0 | Thanks dlynes_laptop, I've downloaded the book, but since I'm supposed to have all this info for a 9am meeting, that's a lot of book to read over night! lol |
| 20:02.25 | dlynes_laptop | Drac0: i never told you to read a book |
| 20:02.44 | dlynes_laptop | Drac0: besides...it's not my fault you're leaving everything to the last minute |
| 20:02.45 | Drac0 | That is correct. |
| 20:03.15 | dlynes_laptop | Drac0: you should've done your research a long time ago |
| 20:03.23 | Drac0 | I was informed this morning, so I wish I could blame myself. Thank you for the interface card suggestiongs. |
| 20:03.31 | clyrrad | dylnes_laptop: yes I did and I do not see any NOTFIY EVENT: resync in the sip debug |
| 20:03.33 | sfinxen | 376 pages :/ |
| 20:03.55 | dlynes_laptop | Drac0: give your boss bloody hell, then |
| 20:04.17 | clyrrad | dlynes_laptop: its as if the message is never sent |
| 20:04.26 | dlynes_laptop | sfinxen: pretty small book...should take you about 3 or 4 hours |
| 20:04.38 | signius | I know this is a question that will get everyone shouting i am flame baiting but i am not i am planning on setting a sinlge box purely for asterisk this box will have no other uses at all i an wondering which is the best distro to base this setup up on i am currently considering CentOS Debian Slackware can any of you guys give your preference |
| 20:04.42 | dlynes_laptop | sfinxen: assuming you read the whole thing, and not just the parts you need to read |
| 20:04.52 | Drac0 | LOL I wish I could. He's the CEO and just read an article somewhere about Asterisk, since we are looking at buying a phone system, he would like to check out Asterisk before we sink $150,000 into a new system. |
| 20:05.02 | dlynes_laptop | signius: centos, debian and slackware are all good choices |
| 20:05.32 | dlynes_laptop | signius: I don't personally like centos, myself, but other than the one kernel patch problem it has, I haven't heard of any issues with it and asterisk |
| 20:05.40 | dlynes_laptop | signius: and centos is well supported in the industry |
| 20:05.49 | dlynes_laptop | signius: debian is easy to administer |
| 20:06.03 | Drac0 | signius, CentOS is nice and stable, I use it for most of my server applications. I know Debian has Asterisk in it's apt-get repository as well. |
| 20:06.04 | dlynes_laptop | signius: and slackware is extremely stable, and easy to build your own packages for |
| 20:06.30 | dlynes_laptop | signius: I use Slackware myself, and lots of people in here use Debian, Gentoo and Centos |
| 20:06.31 | sfinxen | :/ |
| 20:07.24 | wltjr | anyone around familiar with sangoma t1 cards and the wanpipe stuff? still getting Received Reserved LoopBack code 04, telco is not seeing anything, and I think there is a problem on my end |
| 20:07.37 | wltjr | I wish sangoma would document some of this stuff better |
| 20:08.43 | signius | I am more familar with debian but i have heard good things about centos i persoannyl not so keen on fedoras yum and i assume centos uses this also gentoo ia hev played with in the past but OMG life is too short slackware i think could be good for stability and also i am not constantly going to be installing stuff as it s asingle purpose box this may be a good choice |
| 20:08.56 | signius | so i am more leaning toward debian and slackware currently |
| 20:09.11 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 20:10.51 | Drac0 | Life is too short... lol That describes Gentoo very well... I used to use it exclusively, but got impatient one day and switched back to Debian... |
| 20:11.14 | wltjr | AH, ok, that was a local issue, damn client plugged into wrong port on dmark |
| 20:11.58 | perd | holy shit signius, that's one hell of an unpuctuated rant |
| 20:12.20 | perd | when you read that do you sound like the micromachines guy? |
| 20:12.54 | signius | i seemt o have gone back to debian every time especially on server stuff ubuntu is the only thing thats tempted me away on the desktop front and on the server i am not tried but it may be worth considering and as its debian based its more familiar to me |
| 20:13.17 | sfinxen | does asterisk work on ubuntu easy? |
| 20:13.39 | signius | I have no idea |
| 20:16.16 | yassine | i have nat=yes for each sip user, and each time i do forward calls comming from Zap extention to a sip user no one can hear the other. any idea what could be the problem ? |
| 20:17.03 | Drac0 | Don't know how easy it is on Ubuntu since I've not set up Asterisk before, but again, they have the files in the apt-get repository. I've seen it mentioned a few times on the ubuntu forums but never looked into it much until today... lol |
| 20:17.49 | signius | Does anyone here use the linksys SPA942 if so whats your experiences with it i have just purchased one and i am sort of regretting the decision and with that i had gone with a grandstream gxp2000 now |
| 20:18.29 | sfinxen | i´m running with debian for now because i have it on a machine |
| 20:18.44 | signius | To my shock Linksys refuse to support these phones and say its down to who sold you the phone to provide the support and the tech support they have online is worse than useless and the phone support as i found out was non exisitant |
| 20:19.51 | J4k3 | signius: sue linksys for the cost of replacing the phones with something that actually gets support. |
| 20:20.04 | J4k3 | thats the only way anyone is going to get linksys to cut it with the suck. |
| 20:20.27 | signius | If anyone is interested there is a gapping security issue in the standard config if you set any passwords to protect the configs this can simply be over ridden by the user on the handset by selecting restore factory settings to me this is unaccpetable does anyone else agree |
| 20:20.36 | *** join/#asterisk HushPe (n=arron@mail.kamar.co.nz) |
| 20:21.39 | signius | luckily for me it got me out of the hole i was in by the phone locking me out but not good enough really |
| 20:23.36 | wltjr | ok, now not having any communication on data channel? I had 24 specified in config, trying now without a data channel configured |
| 20:23.59 | wltjr | are others specifying the data channel in their wanpipe*.conf files? |
| 20:27.07 | dlynes_laptop | wltjr: do you have a channelized t1/e1? |
| 20:27.45 | wltjr | dlynes_laptop: yes, pri full voice atm, miss allocation, will need it to be voice/data, but right not just working on voice portion |
| 20:27.52 | HushPe | i'm having a little problem getting my digium TDM400P working (22), modules are installed so it's not kernel related as the driver is all going, but I can't get zap working in asterisk, it might be the order I compiled things but i seem to not get zap.so in either case, i have no idea what the hell i'm missing :( |
| 20:28.27 | dlynes_laptop | wltjr: if it's a pri, you won't be able to do data, I don't think, but I could be wrong |
| 20:28.43 | wltjr | dlynes_laptop: pri has a reserved channel for data |
| 20:28.46 | *** join/#asterisk viperdudeuk (n=viperdud@philspc.enta.net) |
| 20:28.46 | dlynes_laptop | wltjr: I know you need a channelized t1...i don't think you can get that with a pri |
| 20:29.05 | dlynes_laptop | wltjr: but regardless with a pri, you need to configure it as 23 b channels and 1 d channel |
| 20:29.06 | wltjr | dlynes_laptop: it's that channel we are not communicating on, ISP is not seeing anything on the d channel, 24 |
| 20:29.19 | viperdudeuk | hi is anyone here good with h264 video on 1.4? |
| 20:29.19 | dlynes_laptop | wltjr: with a regular non-pri t1, you need 24 voice channels |
| 20:29.20 | wltjr | dlynes_laptop: yes, its the 1 d channel there is no communication |
| 20:29.38 | wltjr | dlynes_laptop: yes, I have those as well, this is not a full data t1 |
| 20:30.00 | sevard | hey dlynes |
| 20:30.02 | sevard | welcome back |
| 20:30.02 | yassine | no sounds in both directions and nat=yes is set for each user any ideas please ? |
| 20:30.19 | dlynes_laptop | wltjr: are you sure your ilec/clec has turned the d channel on? |
| 20:30.24 | dlynes_laptop | heya sevard |
| 20:30.34 | wltjr | dlynes_laptop: yes, on the phone with them now |
| 20:30.34 | sevard | where you been, poopie kins? |
| 20:30.41 | dlynes_laptop | around |
| 20:30.47 | dlynes_laptop | just been busy |
| 20:30.56 | wltjr | dlynes_laptop: I had the dchannel configured as 24 in wanpipe*.conf now I am trying without and * won't start |
| 20:32.12 | wltjr | dlynes_laptop: but when I did have d channel configured, * bitched about not being able to use it or etc |
| 20:32.13 | dlynes_laptop | wltjr: try asterisk -vvvvvvvvvvvvvvvvvvvvvb to find out why asterisk isn't starting |
| 20:32.15 | dlynes_laptop | erm |
| 20:32.15 | dlynes_laptop | s/b/g/ |
| 20:32.31 | dlynes_laptop | wltjr: also, did you pick option 2 or 3 for the type of driver you wanted to install during the initial setup? |
| 20:32.42 | dlynes_laptop | wltjr: i.e. one of the options that specified asterisk support? |
| 20:32.49 | dlynes_laptop | wltjr: erm zaptel i mean? |
| 20:33.13 | dlynes_laptop | wltjr: and which version of wanpipe are you using? |
| 20:33.20 | wltjr | dlynes_laptop: * could not bind to zap channels |
| 20:33.29 | wltjr | dlynes_laptop: when a dchannel was not specified in wanpipe*.conf |
| 20:33.31 | dlynes_laptop | wltjr: did you do a ztcfg -vvvvvvvvvvvvv before loading asterisk? |
| 20:33.42 | wltjr | dlynes_laptop: the latest version of wanpipe from wiki |
| 20:33.52 | wltjr | 2.3.4-4 |
| 20:33.52 | dlynes_laptop | wltjr: release 3.3.4-3? |
| 20:33.57 | dlynes_laptop | erm yeah |
| 20:33.59 | dlynes_laptop | that's what it was |
| 20:34.02 | dlynes_laptop | ok |
| 20:34.10 | wltjr | zaptel 1.2.12 |
| 20:34.14 | dlynes_laptop | the last beta that they had on there was quite buggy |
| 20:34.26 | *** join/#asterisk Dr-Linux|home (n=Dreamer@DSL-202-59-73-131.nexlinx.net.pk) |
| 20:34.32 | wltjr | the * autoconf did not work for me :( |
| 20:34.41 | wltjr | no wancfg zaptel or what ever |
| 20:34.49 | dlynes_laptop | asterisk autoconf? |
| 20:34.55 | dlynes_laptop | what's asterisk autoconf? |
| 20:34.55 | wltjr | per the wiki |
| 20:35.06 | dlynes_laptop | no idea what you're talking about |
| 20:35.06 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 20:35.07 | dlynes_laptop | normally |
| 20:35.08 | wltjr | dlynes_laptop: some auto wanpipe configuration for zaptel |
| 20:35.12 | dlynes_laptop | you do wanrouter start |
| 20:35.17 | wltjr | http://wiki.sangoma.com/wanpipe-asterisk-configure |
| 20:35.21 | dlynes_laptop | then do ztcfg -vvvvvvvvvvv |
| 20:35.23 | wltjr | #2 |
| 20:35.25 | dlynes_laptop | then /usr/sbin/safe_asterisk |
| 20:35.40 | wltjr | uses zaptel.conf to base the wanpipe config on |
| 20:35.48 | dlynes_laptop | ah...no idea |
| 20:35.52 | wltjr | but * starts again with dchannel defined in wanpipe1 just bitches about it |
| 20:35.52 | dlynes_laptop | i never use that method |
| 20:35.57 | wltjr | it seems new |
| 20:36.02 | *** join/#asterisk h0 (i=fakhir@unaffiliated/fakhir) |
| 20:36.13 | *** join/#asterisk jimmy_deanPB (n=jhodapp@209.131.196.174) |
| 20:36.44 | wltjr | damit, what is going on, I am having to restart this box to clear out stuff :( |
| 20:36.47 | dlynes_laptop | i dunno...doesn't mention anything about zaptel.conf there |
| 20:37.16 | dlynes_laptop | oh...nvm....further up |
| 20:37.16 | wltjr | restarting wanrouter causes some funkyness to where * can't bind to zap channels |
| 20:37.30 | dlynes_laptop | and that's ONLY APPLICABLE to 2.3.4 BETA, not 2.3.4-4 RELEASE |
| 20:37.51 | dlynes_laptop | 2.3.4-4 is a release, not a beta |
| 20:38.06 | dlynes_laptop | 2.3.4 beta is quite old |
| 20:38.09 | wltjr | dlynes_laptop: yeah, I realized that afterward :) |
| 20:38.12 | dlynes_laptop | probably about a year old |
| 20:38.26 | dlynes_laptop | well...not quite |
| 20:38.32 | dlynes_laptop | 2.3.4 beta came out in April of last year I think |
| 20:38.46 | wltjr | dlynes_laptop: yes, I was around when it first started :) Nedad added a noilmi option to get my data T1 up and running, Copper Mountain DSLAMs don't use or recognize ILMI and was causing interface to hang on start |
| 20:39.03 | *** join/#asterisk clive- (n=pirch@dsl-242-172-113.telkomadsl.co.za) |
| 20:39.10 | wltjr | dlynes_laptop: yes exactly the time I replaced my Netopia T1 router with a Sangoma A101 :) |
| 20:39.40 | wltjr | dlynes_laptop: I still have an inbetween version as nedad tweaked and changed somethings on one of my boxes, so it's pre-beta :) |
| 20:39.50 | dlynes_laptop | ah |
| 20:39.57 | dlynes_laptop | yeah...beta 2.3.4 was pretty stable, though |
| 20:40.07 | wltjr | dlynes_laptop: but this is a totally different box, not really sure what version I should be using |
| 20:40.11 | dlynes_laptop | I just haven't run it on any of my boxes for a long ass time, though |
| 20:40.16 | dlynes_laptop | because it didn't support the a200 |
| 20:40.26 | dlynes_laptop | so i only used it on my main server |
| 20:41.10 | dlynes_laptop | the thing that sucks about the new versions though |
| 20:41.15 | dlynes_laptop | is that they don't work on 2.4 kernels |
| 20:41.20 | *** join/#asterisk luke-jr_work (n=luke-jr@adsl-70-128-250-253.dsl.ksc2mo.swbell.net) |
| 20:41.28 | luke-jr_work | why is lists.digium.com always messed up? :/ |
| 20:42.03 | dlynes_laptop | it isn't |
| 20:42.23 | *** join/#asterisk tsurko (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 20:42.26 | dlynes_laptop | wltjr: 2.3.4-4...it's pretty stable |
| 20:42.30 | dlynes_laptop | wltjr: as is 2.3.4-3 |
| 20:42.52 | wltjr | ok, so it's not a driver stability version, one of a million vars I can rule out :) |
| 20:45.40 | wltjr | dlynes_laptop: do you know anything about the true encoding ? on/off |
| 20:46.22 | dlynes_laptop | true encoding? |
| 20:46.29 | dlynes_laptop | I don't recall ever coming across such an option |
| 20:46.54 | wltjr | dlynes_laptop: it's in the advanced options above HDLC engine |
| 20:46.55 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 20:46.55 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 20:47.21 | wltjr | dlynes_laptop: the config seems pretty simple and straight forward, so can't understand why the data channel is no go |
| 20:47.37 | dlynes_laptop | in wancfg? |
| 20:47.42 | wltjr | dlynes_laptop: yes |
| 20:48.21 | wltjr | dlynes_laptop: interface setup, advanced optoins |
| 20:48.25 | luke-jr_work | dlynes_laptop: it almost never works |
| 20:48.44 | *** part/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
| 20:48.55 | luke-jr_work | if ever |
| 20:49.05 | *** join/#asterisk dasenjo (n=dasenjo@190.5.196.149) |
| 20:49.32 | dlynes_laptop | luke-jr_work: what never works? the whole website? |
| 20:49.52 | luke-jr_work | dlynes_laptop: as far as I can tell |
| 20:50.03 | luke-jr_work | I can even reproduce it manually |
| 20:50.13 | *** join/#asterisk fiber0pti (n=John@207.114.199.107) |
| 20:50.33 | dlynes_laptop | wltjr: here's what I've got on interface setup: name: w1g1, operation mode: TDM_VOICE, span: 1, hdlc timeslot: 24, hwec: no |
| 20:50.50 | fiber0pti | Is there a way to allow 1.2 to have multiple parking lots? |
| 20:51.05 | dlynes_laptop | luke-jr_work: I just tried it...comes up just fine for me |
| 20:51.14 | luke-jr_work | dlynes_laptop: http://lists.digium.com/pipermail/asterisk-biz/2006-August/016614.html |
| 20:51.24 | dlynes_laptop | fiber0pti: there's a patch for that |
| 20:51.27 | fiber0pti | ohh..where? |
| 20:51.28 | luke-jr_work | it times out |
| 20:51.36 | dlynes_laptop | luke-jr_work: comes up just fine for me |
| 20:51.44 | dlynes_laptop | luke-jr_work: sure it isn't your proxy server? |
| 20:51.48 | luke-jr_work | dlynes_laptop: no proxy |
| 20:51.52 | *** join/#asterisk justinc- (n=samblack@host-64-179-18-177.spr.choiceone.net) |
| 20:51.58 | *** join/#asterisk darkskiez (i=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 20:51.59 | dlynes_laptop | luke-jr_work: sure you're not going through a transparent proxy server? |
| 20:52.06 | luke-jr_work | just NAT, and lists.digium.com is the only one with the problem |
| 20:52.12 | fiber0pti | dlynes_laptop: and is the multi parking lot functionality built into 1.4? |
| 20:52.26 | dlynes_laptop | fiber0pti: don't know...the best person to ask would be oej |
| 20:52.33 | dlynes_laptop | fiber0pti: he's the one that was working on that functionality |
| 20:52.40 | wltjr | dlynes_laptop: hmm, I am seeing unchannelized, I think that might be my problem |
| 20:52.56 | dlynes_laptop | wltjr: can you even do channelized on a pri? |
| 20:53.12 | wltjr | dlynes_laptop: no clue |
| 20:53.28 | wltjr | here is wanpipe output at start/load http://rafb.net/p/l6eqgU65.html |
| 20:53.35 | dlynes_laptop | wltjr: afaik, pri is a different line protocol from channelized t1 |
| 20:54.06 | wltjr | dlynes_laptop: ok |
| 20:54.18 | dlynes_laptop | wltjr: ummm...no wonder it's not working |
| 20:54.19 | wltjr | here is my config http://rafb.net/p/qsaK1147.html |
| 20:54.27 | dlynes_laptop | wltjr: looks like you don't have it configured for zaptel correctly |
| 20:54.37 | wltjr | dlynes_laptop: I was selecting the options per the wiki |
| 20:54.39 | dlynes_laptop | wltjr: you've got it configured for wanpipe api instead |
| 20:54.53 | dlynes_laptop | wltjr: that'll only work on yate and freeswitch, not asterisk |
| 20:55.18 | dlynes_laptop | wltjr: which part of the wiki? |
| 20:55.54 | wltjr | http://wiki.sangoma.com/wanpipe-asterisk-configure#a104 |
| 20:56.02 | dlynes_laptop | wltjr: your wanpipe config looks fine |
| 20:56.11 | dlynes_laptop | wltjr: but your driver startup doesn't |
| 20:56.20 | wltjr | for zaptel |
| 20:56.22 | dlynes_laptop | wltjr: i think maybe you compiled the driver incorrectly |
| 20:56.30 | wltjr | you want to see zapata.conf or zaptel.conf |
| 20:56.45 | wltjr | dlynes_laptop: I compiled it for option 3, voice/data |
| 20:56.46 | dlynes_laptop | wltjr: do lsmod |
| 20:56.56 | dlynes_laptop | wltjr: do you see 'zaptel' loaded? |
| 20:57.20 | wltjr | http://rafb.net/p/qC02Xx86.html |
| 20:57.23 | wltjr | yes of course |
| 20:57.30 | wltjr | I have zap interfaces that udev creates and all |
| 20:57.35 | JT | err i thought channelised t1 was the same as pri |
| 20:57.44 | wltjr | I was able to start *, it would just bitch about no d channel |
| 20:57.50 | JT | i guess channelised t1 could also be CAS RBS, but still |
| 20:57.54 | JT | it is channelised |
| 20:58.05 | dlynes_laptop | no idea...only telecom stuff I know quite well is analog, not digital |
| 20:58.07 | wltjr | atm I can't start * because it can't bind to channel 1, some sort of hw lock, not sure if on my end or isp |
| 20:58.22 | dlynes_laptop | It was just an understanding I had...nothing for sure |
| 20:58.44 | dlynes_laptop | ah...nvm...the ppp is for your channelizing |
| 20:58.46 | JT | unchannelised would be where the t1 is used for data, afaik |
| 20:59.28 | dlynes_laptop | well, unchannelized as I understand it |
| 20:59.34 | kgx | does anyone know a solution to my problem here: http://forums.digium.com/viewtopic.php?t=12974 . i've been trying to figure it our for days but have no idea whats wrong |
| 20:59.38 | dlynes_laptop | is where it's 24 voice channels, and you can't change that |
| 21:00.03 | *** join/#asterisk hohum (n=dcorbe@mercury.sunrocket.com) |
| 21:00.40 | JT | i think both pri and CAS are channelised |
| 21:00.41 | dlynes_laptop | ic |
| 21:00.42 | justinc- | kgx: did you try disabling the buffering to cdr's? |
| 21:03.13 | naitram | anyone know if there are any 1.4 packages available for debian. Dont really want to build it. |
| 21:04.47 | *** part/#asterisk naitram (n=danny@216.77.58.40) |
| 21:05.19 | kgx | justinc-: thanks..i'll try that. im just a developer and our network admin is away...my asterisk knowledge is only 3 days old :p someone's on the phone right now so i'll try that in a minute |
| 21:05.28 | justinc- | We're having a problem where our * server has severe CPU spikes randomly throughout the day. I've been fighting this for a while now and there's no issue with the dial plan (loops, etc.) and I've found nothing unusual with in the debug log. Only thing I really found was high load during AGI execution of perl scripts, but the scripts are incredibly simplistic ... anyone have any ideas? Maybe a pointer to where to look for what's |
| 21:05.28 | justinc- | <PROTECTED> |
| 21:05.43 | justinc- | kgx: no problem man... gl! :) |
| 21:06.02 | justinc- | kgx: you should be able to reload the cdr stuff by itself on the fly |
| 21:07.06 | kgx | justinc-: how? 'extensions reload' is it? |
| 21:07.15 | justinc- | kgx: no no :) ... lemme check |
| 21:07.45 | signius | why aint icechat opening links in a browser window when i click on them :-( |
| 21:07.54 | justinc- | kgx: I think it's 'reload cdr_manager.so' |
| 21:08.11 | kgx | justinc-: cheers |
| 21:08.29 | justinc- | kgx: and to you sir! :) |
| 21:09.26 | HushPe | i can't get chan_zap.so to compile, i have libpri, and zaptel installed (in that order), what else am I missing? |
| 21:09.40 | justinc- | HushPe: What errors are you getting? |
| 21:10.24 | JT | the errors are what you're missing, yes :) |
| 21:10.42 | HushPe | i don't have errors, it's just not compiling :) |
| 21:10.47 | justinc- | so no one has had any personal experiences with * giving large cpu spikes? :( |
| 21:10.54 | JT | there will be compile errors |
| 21:10.54 | naftali5 | HushPe, do you have your kernel source? |
| 21:10.58 | justinc- | um ... huh? heh |
| 21:11.05 | HushPe | should do i have recompiled the kernel :) |
| 21:11.10 | justinc- | no you don't |
| 21:11.11 | HushPe | unless I missed something there |
| 21:11.16 | justinc- | but you DO need the kernel devel stuff |
| 21:11.24 | justinc- | what distro are you running? |
| 21:11.27 | HushPe | slackware |
| 21:11.48 | sevard | i also run slackware. |
| 21:11.49 | justinc- | hmm .. been a while .. I don't remember if that comes with the kernel devel stuff installed by default |
| 21:12.02 | HushPe | the zaptel driver is installed (and working), but i can't get the channel in asterisk |
| 21:12.30 | justinc- | but .. ok I'm confused |
| 21:12.37 | justinc- | it's installed? so it compiled fine then .. |
| 21:13.00 | JT | justinc-: chan_zap is part of asterisk afaik |
| 21:13.12 | JT | not zaptel |
| 21:13.14 | HushPe | zaptel is compiled fine, but i can't get chan_zap to compile (with asterisk) |
| 21:13.20 | HushPe | i know, that's why i'm a bit puzzled |
| 21:13.23 | justinc- | oh .. ok |
| 21:13.39 | justinc- | I think we're back to square 1 heh ... |
| 21:13.44 | justinc- | what's the error you're getting? :) |
| 21:13.48 | HushPe | chan_skinny is compiling (in the make listing as it goes along), then i doesn't do chan_zap |
| 21:13.52 | JT | HushPe: so what compile errors are you getting? |
| 21:13.58 | HushPe | no compile errors |
| 21:14.08 | HushPe | i just don't get zap.so |
| 21:14.24 | JT | you'll need to provide more info on what you're doing then |
| 21:14.25 | justinc- | check your modules.conf? |
| 21:14.53 | justinc- | it sounds like he's compiling and compilation goes without error .. yet there's no chan_zap available in his working asterisk processes |
| 21:15.15 | JT | HushPe: is there a chan_zap.so somewhere on the filesystem? |
| 21:15.26 | justinc- | /usr/lib/asterisk/modules |
| 21:15.28 | justinc- | is where it should be |
| 21:15.30 | HushPe | JT: no :( |
| 21:15.42 | JT | updatedb;locate chan_zap |
| 21:16.16 | HushPe | nothing JT i already know that :) |
| 21:16.28 | justinc- | that's pretty weird ... |
| 21:16.30 | JT | how are you compiling? |
| 21:16.37 | justinc- | oo and what version? |
| 21:16.44 | HushPe | ./configure; make clean; make; make install |
| 21:16.48 | HushPe | 1.4.0 |
| 21:16.48 | justinc- | 1.4 lets you disable compilation of channel types |
| 21:16.53 | justinc- | aha :) |
| 21:17.23 | JT | 1.4.0 should only be for non-production systems imho |
| 21:17.23 | *** part/#asterisk pekdon (n=pekdon@85.196.83.77) |
| 21:17.30 | justinc- | I agree |
| 21:17.34 | justinc- | to early to use it |
| 21:17.58 | *** join/#asterisk generalhan (n=asd@67.90.64.2) |
| 21:18.07 | justinc- | at the source tree |
| 21:18.11 | HushPe | ok, to 1.2 i go, i sure it wasn't this hard last time (when i used 1.2 probably) |
| 21:18.11 | justinc- | type: 'make menuselect' |
| 21:18.24 | justinc- | or go back to 1.2 hehe :) |
| 21:18.55 | HushPe | has a * in the chan_zap box? odd! |
| 21:19.01 | generalhan | Anyone here using a cisco 7960 with the SipDefault.cfg file ?? |
| 21:19.08 | justinc- | It should by default ... |
| 21:19.22 | HushPe | i think 1.2 would be my better bet |
| 21:19.32 | *** join/#asterisk yacc (n=andreas@h1634.serverkompetenz.net) |
| 21:19.37 | justinc- | I agree |
| 21:19.42 | justinc- | 1.4 is still pretty young |
| 21:19.58 | dlynes_laptop | JT: nah...it should be thrown onto your main server that runs your whole business :) |
| 21:20.08 | justinc- | heh |
| 21:20.12 | dlynes_laptop | JT: make sure there's no redundancy, too :) |
| 21:20.20 | JT | of course |
| 21:20.40 | yacc | Anyone got an idea? |
| 21:20.55 | justinc- | so at the expense of sounding really stupid ... is there a way to really see what's utilizing the CPU within asterisk beyond the 'ps' command? |
| 21:21.02 | sevard | 'top' |
| 21:21.11 | justinc- | lol top is ps automated :) |
| 21:21.16 | sevard | exactly :) |
| 21:21.16 | justinc- | but thanks :) |
| 21:21.20 | justinc- | doesn't help bro :( |
| 21:21.28 | sevard | you want _within asterisk_ |
| 21:21.33 | justinc- | yeah pretty much |
| 21:21.36 | sevard | asterisk is one process |
| 21:21.48 | justinc- | specific function calls or applications like AGI ... |
| 21:21.50 | JT | justinc-: is it asterisk that's spiking for sure? |
| 21:21.57 | justinc- | but many threads :) |
| 21:22.05 | justinc- | um .. yeah pretty sure |
| 21:22.16 | justinc- | I've got scripts logging 'ps' output |
| 21:22.25 | sevard | doesn't pstree help? |
| 21:22.29 | JT | i know you can compile in options so you can see what part of asterisk uses what memory |
| 21:22.30 | justinc- | it's always an agi script and/or asterisk with the highest cpu usage |
| 21:22.45 | HushPe | thanks, i'll try 1.2 |
| 21:22.53 | HushPe | hopefully more luck there |
| 21:22.55 | justinc- | ps has stuff that can show threads |
| 21:22.58 | justinc- | like pstree |
| 21:23.08 | HushPe | i'll need all the 1.2 stuff eh, remove all 1.4 zaptel etc? |
| 21:23.09 | justinc- | only thing is ... I really can't find anything that looks out of order .. |
| 21:23.12 | justinc- | I'm really stumped |
| 21:23.21 | justinc- | no dial plan issues ... no debug issues ... |
| 21:24.11 | justinc- | hmm |
| 21:24.26 | justinc- | I guess I'll try to isloate asterisk as much as possible ... |
| 21:24.33 | justinc- | can't think of anything else to do :( |
| 21:24.46 | justinc- | HushPe: Yeah remove all the 1.4 stuff |
| 21:25.09 | justinc- | HushPe: Usually all I do is 'rm -f /usr/lib/asterisk/modules/*' :) |
| 21:25.24 | HushPe | :) |
| 21:25.55 | HushPe | i'm a bit pissed that i kinda lost a day |
| 21:26.04 | justinc- | HushPe: heh. You also probably want to back up the 1.4 conf files and move them out of /etc/asterisk .. there's been a lot of functionality/settings changes between 1.2 and 1.4 |
| 21:26.07 | HushPe | lasttime i played i had a bad card, but all the channels worked fine :) |
| 21:26.22 | HushPe | justinc-: thanks for the heads up |
| 21:26.31 | justinc- | HushPe: Anytime bro :) |
| 21:26.47 | *** part/#asterisk essaredee (i=srd@24-182-113-208.dhcp.sprn.tx.charter.com) |
| 21:26.51 | HushPe | a day is a day, don't get a replacement one or an RMA haha |
| 21:27.08 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 21:27.33 | justinc- | heck when digium finally linked the channel lockup issue with the agent channel and got rid of agentcallbacklogin .. I lost a week scambling to upgrade to the "new" paradigm :) |
| 21:28.35 | *** join/#asterisk hittop (n=Miranda@CPE001839bd0eb2-CM0014f8c23dbe.cpe.net.cable.rogers.com) |
| 21:28.58 | justinc- | has anyone had any problems running perl scripts through agi? |
| 21:29.06 | justinc- | perhaps perl startup overhead? |
| 21:30.56 | JT | do you know if the spikes come when agi is active? |
| 21:31.07 | *** join/#asterisk hads (n=hads@reef80.anchor.net.au) |
| 21:31.35 | justinc- | I have a log of 'ps' every time load average goes above 5 running ever 5 seconds ... |
| 21:31.36 | hads | mitcheloc: ping |
| 21:31.40 | justinc- | most of the time there's an agi script running |
| 21:31.45 | dlynes_laptop | wltjr: btw...in your dmesg output |
| 21:31.48 | justinc- | check this out: |
| 21:31.53 | dlynes_laptop | wltjr: i never did see where you ran ztcfg |
| 21:31.57 | justinc- | root 10643 2.2 1.6 57164 34848 pts/0 Sl Jan20 96:43 /usr/sbin/asterisk -gv -vvvg -c |
| 21:32.06 | justinc- | root 17292 16.0 0.1 7488 3600 pts/0 R 15:15 0:00 /usr/bin/perl /var/lib/asterisk/agi-bin/check_open_status.pl |
| 21:32.26 | justinc- | this server really only runs asterisk .. so it's not any other "major" daemons doing bad stuff :( |
| 21:32.43 | justinc- | oh and the check_open_status.pl file is about 5 lines :) |
| 21:32.45 | justinc- | very very simple :) |
| 21:32.46 | wltjr | dlynes_laptop: I am not running ztcfg, do I need to? |
| 21:32.51 | JT | justinc-: you should have that ps log also print a datestamp |
| 21:32.52 | dlynes_laptop | wltjr: yes |
| 21:33.09 | dlynes_laptop | wltjr: run it as ztcfg -vvvvvvvvv so you can see a bit more info when it initializes |
| 21:33.09 | JT | and make your agi write datestamps somewhere whenever it, or various stages of it runs |
| 21:33.11 | justinc- | JT: it creates a directory based off the output of 'date' |
| 21:33.20 | dlynes_laptop | wltjr: after it's finished, then run asterisk |
| 21:33.27 | justinc- | hmm not a bad idea |
| 21:33.30 | dlynes_laptop | wltjr: without ztcfg, asterisk doesn't see the zaptel device |
| 21:33.58 | JT | justinc-: what is your load average normally? |
| 21:34.01 | wltjr | http://rafb.net/p/s15K7w21.html |
| 21:34.03 | JT | and it spikes to over 5? |
| 21:34.16 | justinc- | lol .04 to .15 :) |
| 21:34.21 | justinc- | yep |
| 21:34.30 | dlynes_laptop | wltjr: looks fine, assuming they're all supposed to be clear |
| 21:34.31 | JT | sounds like I/O issues |
| 21:34.33 | justinc- | we've had spikes to around 15 and one to 43 today |
| 21:34.40 | justinc- | network perhaps? |
| 21:34.43 | dlynes_laptop | wltjr: now run asterisk? |
| 21:34.47 | justinc- | or device io? |
| 21:34.48 | JT | what does the script do? |
| 21:34.57 | JT | usually device |
| 21:35.08 | justinc- | some of our scripts do database connectivity for customer information and credit card processing |
| 21:35.15 | wltjr | dlynes_laptop: ok, ty that explains why it was not starting ;) |
| 21:35.19 | justinc- | database is on a different server |
| 21:35.23 | wltjr | dlynes_laptop: but back to chan_zap.c: No D-channels available! Using Primary channel 24 as D-channel anyway! |
| 21:35.25 | wltjr | wtf :( |
| 21:35.28 | dlynes_laptop | wltjr: hehe...i mentioned ztcfg a few times |
| 21:35.35 | dlynes_laptop | wltjr: but i guess you must not have seen it |
| 21:35.40 | dlynes_laptop | wltjr: you might get that at startup |
| 21:35.44 | justinc- | is there a way to monitor io usage? |
| 21:35.56 | justinc- | <- programmer more than an admin <blush> |
| 21:35.57 | dlynes_laptop | wltjr: it's only because of a timing/race condition in asterisk startup, i think |
| 21:36.05 | JT | justinc-: not sure |
| 21:36.11 | justinc- | I'll google it :) |
| 21:36.13 | wltjr | dlynes_laptop: yeah our zaptel init script calls that |
| 21:36.16 | dlynes_laptop | wltjr: it doesn't happen every time...just sometimes |
| 21:36.19 | justinc- | thanks for the ideas though .. :) |
| 21:36.24 | HushPe | no i have chan_zap :) with errors, but it's better than not having it! |
| 21:36.31 | bagpuss_thecat | anyone know why (or how to fix) Asterisk when it doesn't seem to detect the hangup tone? I get voicemail that lasts for the whole silence timeout period, and calls that aren't ended properly until timeout either. |
| 21:36.40 | wltjr | dlynes_laptop: that likely explains why I had no luck with no d channel specified, might have to try that again |
| 21:36.40 | dlynes_laptop | wltjr: depending on your system, it might happen often, rarely, or somewhere in between |
| 21:36.56 | wltjr | dlynes_laptop: still got to find out what's up with the d channel |
| 21:37.14 | JT | assuming your load average is usually <1, then if the load average goes above 1 for every cpu that the os sees, it's usually i/o |
| 21:37.14 | HushPe | JT: undefined symbol: ast_pickup_call (i'm googling now) |
| 21:37.15 | wltjr | * bitches about it, and the telco is not seeing any data on the d channel |
| 21:37.18 | dlynes_laptop | wltjr: that 'error' you're getting above I think is actually a warning |
| 21:37.26 | JT | justinc-: not running short on ram? |
| 21:37.44 | wltjr | dlynes_laptop: sure, and I would ignore but guy a telco said he was not seeing any data on d channel |
| 21:37.51 | dlynes_laptop | wltjr: i c |
| 21:37.53 | justinc- | actually yeah |
| 21:37.57 | justinc- | just noticed that |
| 21:37.59 | dlynes_laptop | wltjr: even after it's been running for a while? |
| 21:38.02 | wltjr | dlynes_laptop: not sure if the true hw encoding would effect that |
| 21:38.05 | dlynes_laptop | wltjr: and even after you did ztcfg? |
| 21:38.09 | justinc- | <PROTECTED> |
| 21:38.10 | wltjr | dlynes_laptop: no clue, never left it up long enough |
| 21:38.23 | JT | justinc-: can you show the output of free? |
| 21:38.24 | dlynes_laptop | wltjr: yeah..i'd let it run for at least a minute before testing it |
| 21:38.25 | wltjr | dlynes_laptop: well I did the ztcfg this time, and * still bitches about d-channel |
| 21:38.32 | dlynes_laptop | wltjr: yeah...like i said |
| 21:38.36 | wltjr | let me call telco and see if they are getthing anything on d channel |
| 21:38.37 | dlynes_laptop | wltjr: it's a startup issue |
| 21:38.43 | justinc- | <PROTECTED> |
| 21:38.44 | justinc- | Mem: 2074780 2020468 54312 0 95428 1582188 |
| 21:38.44 | justinc- | -/+ buffers/cache: 342852 1731928 |
| 21:38.44 | justinc- | Swap: 2031608 624 2030984 |
| 21:38.50 | dlynes_laptop | wltjr: i've seen that happen with both digium and sangoma cards |
| 21:38.58 | dlynes_laptop | wltjr: but everything still works just fine |
| 21:39.10 | justinc- | arg what's hogging my mem? |
| 21:39.16 | wltjr | dlynes_laptop: going to owe you some beer for the help, ty, if nothing else a little rubber ducking that has helped |
| 21:39.17 | JT | justinc-: plenty of free memory |
| 21:39.24 | dlynes_laptop | justinc-: keyword: CACHE |
| 21:39.27 | justinc- | 54meg? |
| 21:39.31 | justinc- | oh right 2.6 kernel |
| 21:39.32 | JT | 1.7GB |
| 21:39.34 | JT | no |
| 21:39.35 | justinc- | I always forget that :) |
| 21:39.40 | JT | linux has always been like that |
| 21:39.47 | JT | unused memory is wasted memory |
| 21:39.53 | dlynes_laptop | justinc-: it's your cache...nothing to do with 2.6 kernel |
| 21:39.54 | justinc- | lol then I just started noticing it around 2.6 |
| 21:40.04 | JT | if you're got heaps of buffers and cache, you're fine |
| 21:40.04 | wltjr | dlynes_laptop: getting those warnings ever 5 seconds |
| 21:40.09 | justinc- | <- needs to be more observant :) |
| 21:40.11 | dlynes_laptop | wltjr: oh...then it's a problem |
| 21:40.18 | wltjr | dlynes_laptop: yes :) |
| 21:40.19 | dlynes_laptop | wltjr: i thought you were only getting it once during startup |
| 21:40.30 | dlynes_laptop | wltjr: if it's only once during startup, then it's not a problem |
| 21:40.32 | wltjr | dlynes_laptop: no, again telco is not seeing anything on their end either |
| 21:40.35 | justinc- | ok .. so *that's* not the problem |
| 21:41.08 | dlynes_laptop | wltjr: try configuring it as an ordinary pri first |
| 21:41.13 | dlynes_laptop | wltjr: without the channelization |
| 21:41.17 | wltjr | dlynes_laptop: ok, no d channel |
| 21:41.21 | JT | well it'd be good to check ram usage during a memory spike, but it's probably not the prob |
| 21:41.22 | dlynes_laptop | wltjr: make sure it's working in a simple configuration first |
| 21:41.24 | wltjr | dlynes_laptop: or no tdm_voice ? |
| 21:41.31 | dlynes_laptop | wltjr: no...with d channel, but lose the ppp |
| 21:41.32 | wltjr | dlynes_laptop: this should be a simple config, not much to it :) |
| 21:41.48 | wltjr | dlynes_laptop: the wanrouter loads the ppp stuff, I don't have it configured at all |
| 21:41.48 | dlynes_laptop | wltjr: make sure your d channel is working correctly first |
| 21:41.56 | wltjr | no clue why it loads all modules, instead of just the ones it needs |
| 21:42.02 | dlynes_laptop | wltjr: nah...wanrouter only loads it if you told it to |
| 21:42.03 | *** part/#asterisk arkadi (n=arkadi@87.246.143.140) |
| 21:42.03 | justinc- | JT: I'll add that to the monitor .. |
| 21:42.10 | dlynes_laptop | wltjr: you told it to when you compiled the driver |
| 21:42.12 | wltjr | dlynes_laptop: I am surely not telling it to load ppp |
| 21:42.18 | dlynes_laptop | wltjr: because you told it you wanted to do channelized tdm |
| 21:42.21 | wltjr | dlynes_laptop: well I compiled in ppp support |
| 21:42.27 | dlynes_laptop | wltjr: telling it to do that, compiles in ppp support |
| 21:42.37 | wltjr | dlynes_laptop: sure, but what should that matter? |
| 21:42.38 | dlynes_laptop | wltjr: if ppp support is compiled in, it'll load it |
| 21:42.44 | dlynes_laptop | wltjr: Just try the basics |
| 21:42.46 | wltjr | dlynes_laptop: your saying I need to recompile? |
| 21:42.49 | dlynes_laptop | wltjr: make sure the basics are working first |
| 21:42.49 | wltjr | ok |
| 21:42.55 | JT | start from first principles |
| 21:43.05 | dlynes_laptop | wltjr: ./Setup install --linux-vanilla |
| 21:43.28 | dlynes_laptop | wltjr: no other parameters |
| 21:43.41 | wltjr | does the vanilla matter? |
| 21:43.44 | dlynes_laptop | wltjr: make sure you compile and install zaptel first |
| 21:43.51 | dlynes_laptop | wltjr: then do the wanpipe setup |
| 21:43.57 | wltjr | dlynes_laptop: yep, on gentoo here :) we compile it all |
| 21:44.08 | dlynes_laptop | wltjr: no idea, but that's the method i use, and i know it works |
| 21:44.08 | wltjr | dlynes_laptop: but just recompiled * and zap for fun, but it wasn't any |
| 21:44.25 | dlynes_laptop | wltjr: yeah, but what i'm saying is zaptel needs to be compiled before wanpipe |
| 21:44.37 | dlynes_laptop | wltjr: then wanpipe will compile itself and then recompile zaptel |
| 21:44.56 | dlynes_laptop | wltjr: before you needed to zaptel, then wanpipe, and then zaptel again |
| 21:45.04 | dlynes_laptop | wltjr: the third step isn't needed with the new wanpipes |
| 21:45.20 | dlynes_laptop | wltjr: it takes care of that internally now |
| 21:45.20 | wltjr | dlynes_laptop: ok so you want tdm voice only, not voice+data as I had it |
| 21:45.28 | dlynes_laptop | wltjr: one second |
| 21:45.38 | JT | clearly |
| 21:45.42 | JT | do you even want data? |
| 21:46.00 | data23 | oh, not me :) |
| 21:46.28 | wltjr | yes, the telco has this pri miss provisioned, atm it's full 23 channels for voice, but we will never use that many lines, order was supposed to be split voice and data |
| 21:46.49 | wltjr | once i get the voice portion down, I will get back in touch with telco and get them to re-provision the line properly as voice and data |
| 21:46.54 | JT | now's a good time to get voice working :) |
| 21:46.54 | dlynes_laptop | wltjr: just rerunning my setup from a fresh install, so i can see what options there were again |
| 21:46.55 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 21:46.55 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 21:47.04 | wunderkin | you probably want to confirm what is the d channel too |
| 21:47.09 | wltjr | dlynes_laptop: yeah curious about the Enable TDMV DCHAN Native HDLC Support & Patch Zaptel |
| 21:47.12 | wltjr | I said yes |
| 21:47.23 | *** join/#asterisk Axelatino (n=alexlsf@200.12.46.58) |
| 21:47.30 | dlynes_laptop | wltjr: TDM voice support only |
| 21:47.30 | wltjr | wunderkin: yeah confirmed that to be channel 24, not 16 or etc |
| 21:47.41 | wltjr | dlynes_laptop: after that |
| 21:47.43 | dlynes_laptop | wltjr: TDMV (Asterisk) on AFT adapters |
| 21:47.52 | wltjr | dlynes_laptop: yes # 2 I selected that one |
| 21:47.55 | dlynes_laptop | ok |
| 21:48.12 | dlynes_laptop | wltjr: so continue on then |
| 21:48.13 | wltjr | dlynes_laptop: next propmpt after zap sources is the one about the d channel |
| 21:48.20 | dlynes_laptop | wltjr: what's the prompt again? |
| 21:48.23 | wltjr | dlynes_laptop: I did, I said yes to channel patch of zaptel |
| 21:48.27 | wltjr | Enable TDMV DCHAN Native HDLC Support & Patch Zaptel |
| 21:48.32 | dlynes_laptop | ' Yeah...go ahead |
| 21:48.35 | dlynes_laptop | that's fine |
| 21:48.36 | wltjr | ok I did |
| 21:48.41 | sfinxen | I´m following the book of asterisk, page 72. i have entered this in my sip.conf http://pastebin.ca/326139 and made a successful connection with x-lite. however it says "The person you are calling are unavaible" when dialing 100 and 611, tips? |
| 21:48.52 | wltjr | compiling |
| 21:48.58 | dlynes_laptop | I'm just not able to continue further with the install on that amchine cause i'm too lazy to install zaptel on it |
| 21:49.04 | Axelatino | anyone can help me with ooh323 |
| 21:49.05 | dlynes_laptop | it's just my desktop at home :0 |
| 21:49.20 | wltjr | dlynes_laptop: np, I had to unpack a copy of zaptel in /usr/src, since it did not like just the headers :) |
| 21:49.31 | dlynes_laptop | and i don't want to screw up any of my existing installs |
| 21:49.36 | dlynes_laptop | because they're all on production boxes |
| 21:49.38 | wltjr | dlynes_laptop: understandably :) |
| 21:49.47 | wltjr | dlynes_laptop: yeah it's nice this one is not in production yet |
| 21:49.55 | wltjr | I still have another * box with a wildcard in it |
| 21:50.01 | Axelatino | my asterisk reset when i place a h323 call |
| 21:50.05 | Axelatino | anyideas? |
| 21:50.06 | dlynes_laptop | so i like to leave everything in a known state on those boxes |
| 21:50.12 | dlynes_laptop | Axelatino: yeah...don't use h323 |
| 21:50.15 | wltjr | at that location :) I have my own here with my el cheapo x100p clones ): |
| 21:50.20 | dlynes_laptop | Axelatino: it's quite unstable and unusable on asterisk |
| 21:50.26 | dlynes_laptop | Axelatino: only use it if you absolutely have to |
| 21:50.39 | Axelatino | i have to |
| 21:50.40 | dlynes_laptop | Axelatino: and make sure you've got a script to automatically start asterisk back up, if it core dumps |
| 21:51.05 | dlynes_laptop | wltjr: yeah...those x100p's really suck :) |
| 21:51.23 | dlynes_laptop | i've got three of them |
| 21:51.27 | wltjr | dlynes_laptop: mine aren't to bad, actually work quite well, but might have gotten a decent clone brand :) |
| 21:51.28 | dlynes_laptop | they're more or less just door stops |
| 21:51.31 | Supaplex | Axelatino: how many hours of talk time a week do you plan to use? |
| 21:51.38 | dlynes_laptop | wltjr: well, they're better than nothing |
| 21:51.45 | dlynes_laptop | wltjr: but i've got the x100p.com ones |
| 21:51.46 | wltjr | dlynes_laptop: at times I have seen them work better than the tdm400s :) |
| 21:51.57 | wltjr | I just got some $7 ones of ebay quite some time ago |
| 21:52.04 | Axelatino | a lot of hours |
| 21:52.11 | wltjr | I purchased them to experiment with, but worked so well I kept them in use ;) |
| 21:52.18 | Axelatino | i receive traffic form h323 provider |
| 21:52.33 | dlynes_laptop | Supaplex: i'm guessing if he's wanting to use h323, he's probably having to deal with a carrier, not an ITSP |
| 21:52.34 | JT | xheliox: ? |
| 21:52.43 | Supaplex | I see. |
| 21:52.52 | dlynes_laptop | Supaplex: so he'll probably be pushing 1M+ minutes per month |
| 21:53.16 | sfinxen | anyone has some ideas for me? |
| 21:53.17 | dlynes_laptop | Axelatino: right? |
| 21:53.32 | wltjr | dlynes_laptop: :( tried to start wanrouter after compiling the new mods, and must have panic'd the kernel or something :( |
| 21:53.39 | dlynes_laptop | wltjr: doh |
| 21:53.43 | wltjr | no more remote access, have to get client to hard re-boot the box |
| 21:53.50 | Axelatino | i will try to change to sip |
| 21:54.05 | dlynes_laptop | Axelatino: another option you can try |
| 21:54.06 | Supaplex | sfinxen: not involving duct tape and violating every other ASPCA law and regulation? - no. :) |
| 21:54.12 | dlynes_laptop | Axelatino: if you absolutely must have h323 |
| 21:54.23 | wltjr | dlynes_laptop: I really do not like the wanpipe build system |
| 21:54.26 | dlynes_laptop | Axelatino: is use yate, freeswitch, or openh323 for your h323 traffic |
| 21:54.29 | wltjr | I would much prefer configure and make |
| 21:54.38 | dlynes_laptop | Axelatino: they can all do sip, so you can just pass your traffic off as sip to them |
| 21:54.48 | dlynes_laptop | Axelatino: and then they can pass it off as h323 to your upstream |
| 21:54.54 | Grnd-Wire | So I need to know what options I have for troubleshooting some IAX trunking and context issues.. I' |
| 21:55.11 | Supaplex | but death2h323.com only accepts sip. ;) |
| 21:55.12 | dlynes_laptop | wltjr: yeah...wanpipe build system really sucks |
| 21:55.13 | Grnd-Wire | I've got verbosity turned up crazy high - and I've got IAX debugging on.. Is there anything else I don't know about? |
| 21:55.17 | Axelatino | i'm receiving the traffic now with cisco |
| 21:55.29 | Axelatino | but i need to have a bill the traffic |
| 21:55.33 | dlynes_laptop | wltjr: but, i've tried digium cards |
| 21:55.34 | wltjr | dlynes_laptop: and I need to see how I can rip it apart and make a gentoo ebuild for it |
| 21:55.46 | dlynes_laptop | wltjr: and the stability i get from sangoma is worth the hassle of dealing with wanpipe |
| 21:55.49 | wltjr | there is one, but it's older and not sure how they hacked the build system |
| 21:55.57 | wltjr | dlynes_laptop: yes, I have heard nothing but good stuff about them |
| 21:56.13 | dlynes_laptop | wltjr: otoh |
| 21:56.14 | wltjr | dlynes_laptop: I loved the support I got when I bought my first a101, for a data t1 |
| 21:56.16 | wltjr | it's still in use :) |
| 21:56.22 | dlynes_laptop | wltjr: Digium's tech support is pretty good |
| 21:56.32 | wltjr | yes, I have had great support from them as well |
| 21:56.37 | dlynes_laptop | wltjr: i'm not sure who's got better tech support...sangoma or digium |
| 21:56.47 | dlynes_laptop | wltjr: they're both pretty damned good |
| 21:56.55 | wltjr | only reason I went wanpipe, is I have more data uses than voice uses, so it's an ideal card |
| 21:57.05 | dlynes_laptop | wltjr: digium can also do channelized t1 |
| 21:57.16 | wltjr | dlynes_laptop: yeah but when I called digium, they weren't as up to date on like data t1's via their cards as Sangoma is |
| 21:57.23 | dlynes_laptop | wltjr: that's what the setdhlc utility is for |
| 21:57.32 | dlynes_laptop | wltjr: just weren't talking to the right people |
| 21:57.37 | wltjr | dlynes_laptop: give all the varying protocols and stuff for data t1' |
| 21:57.48 | dlynes_laptop | wltjr: Try Corydon-w ... he seems to be well-versed on it |
| 21:58.01 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 21:58.16 | wltjr | dlynes_laptop: eh well to late now, I am knee deep in Sangoma cards, and they are $25 less so :) |
| 21:58.17 | dlynes_laptop | bonjour m'sieu |
| 21:58.48 | dlynes_laptop | wltjr: yeah... [TK]D-Fender is the one that got me started on sangoma |
| 21:58.53 | dlynes_laptop | wltjr: haven't looked back since |
| 21:59.08 | wltjr | dlynes_laptop: plus given nedad @ sangoma had to modify the software for frame relay to work with Copper Mountain DSLAMS, that was a win in my book, not sure if Digium would have done the same |
| 21:59.33 | dlynes_laptop | wltjr: i'm sure they would |
| 21:59.39 | wltjr | of course they have been MIA with my current situation :) but I can't blame that entirely on them, my telco has some issues |
| 21:59.48 | dlynes_laptop | wltjr: whether they have anybody in house that is capable or not...I don't know |
| 21:59.55 | wltjr | not the current PRI one, two other data t1 upgrades |
| 22:00.01 | *** join/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
| 22:00.41 | wltjr | dlynes_laptop: at one point I had a digium guy log into one of my clients * boxes years ago, was impressed, he was at work after hours I believe of the clock, just hanging in here for some abuse |
| 22:00.59 | hoobastooba | i broke something with my meetme extension. Everything works correctly, except now when user join, the other users in the room dont hear the chime. What would have caused that? I see where you can turn the chime off... but can i explicitly make it chime when users enter the room? |
| 22:01.41 | dlynes_laptop | wltjr: yeah...Alex at Sangoma has done the same for me :) |
| 22:01.58 | dlynes_laptop | wltjr: as has David Yat Sin |
| 22:02.17 | dlynes_laptop | wltjr: those guys are both pretty cool |
| 22:02.19 | wltjr | sure, but Nedads name is all over the code, I realized that after he helped me, it humbled me a bit |
| 22:02.32 | generalhan | Anyone in here using Cisco7960s ? i need to adjust the SNTP configuration for the time zone but i dont think there is a listing for "Arizona" as a timezone ... and when i use MST or PST i have to set the clocks back for DST to make the time right again. |
| 22:02.41 | dlynes_laptop | wltjr: yeah...Alex's name is all over the code, too |
| 22:02.49 | dlynes_laptop | wltjr: He's one of their main driver developers |
| 22:02.52 | wltjr | dlynes_laptop: haven't seen that one, but will keep an eye out for it |
| 22:03.00 | wltjr | I try to avoid the code when I can :) |
| 22:03.01 | dlynes_laptop | wltjr: David is a tech support guy |
| 22:03.04 | wunderkin | generalhan, there isn't a no daylight setting? |
| 22:03.18 | dlynes_laptop | wltjr: he's currently working on the drivers for Solaris |
| 22:03.24 | wltjr | dlynes_laptop: no clue, still waiting to hear back on a couple of emails to techdesk or what ever |
| 22:03.36 | dlynes_laptop | wltjr: intel first, and then he'll be working on some for solaris on ultrasparc |
| 22:03.56 | wltjr | damit I think the wanpipe rc stuff is loading it at boot time and causing the panic to happen on boot :( |
| 22:03.57 | generalhan | wunderkin: there is ... but what does that mean when it goes to the ntp server to check for MST or PST, its going to take that time, so i still have to go in and change it from Mountain to Pacific twice a year |
| 22:04.02 | dlynes_laptop | wltjr: they might be putting out a miniPCI card by november or so, too |
| 22:04.10 | wltjr | I saw a pcie one |
| 22:04.19 | wunderkin | generalhan, -7 no daylight |
| 22:04.32 | dlynes_laptop | that would totally rock if they had a PC/104+ card though |
| 22:04.35 | generalhan | "-7" ? |
| 22:04.42 | dlynes_laptop | I'd love to see one of those for analog |
| 22:04.48 | JT | -7 from UTC, obviously |
| 22:04.57 | wunderkin | it works |
| 22:05.01 | dlynes_laptop | that would mean I could build an asterisk box that would look like a panasonic keysystem |
| 22:05.06 | generalhan | no i understand but they are asking me for a Time Zone not a time difference ... |
| 22:05.08 | unik-rados | I've got a werid problem... i've got asterisk connected to Avaya through ooh323. for the most part it works really well but certain IP Phones have only one way audio. so for example, when they call in to MeetMe, we can hear them but they can't hear us. digital sets users and most ip phone users don't have that issue. all ip phones use the same codec, so I was told my the telecom team. any ideas what could be going wrong? |
| 22:05.19 | generalhan | im not meaning to be condicending or anything i just want to be sure i understand |
| 22:05.35 | JT | you can't set that as a zone? |
| 22:05.40 | wltjr | dlynes_laptop: yeah this guy on gentoo-voip had mbx building some custom boxes for his uses :) |
| 22:05.42 | *** join/#asterisk thoughtpolice (n=austin@ip70-185-140-61.lu.dl.cox.net) |
| 22:05.45 | wunderkin | well it is mountain |
| 22:05.55 | dlynes_laptop | wltjr: mbx? |
| 22:06.13 | [TK]D-Fender | I've dealt with David I think once, and Nenad for my first experience. Great team over there. Still some things to be worked out in the longer run, but the follow-through has been exemplery |
| 22:06.17 | wltjr | http://mbx.com/ |
| 22:06.36 | wltjr | dlynes_laptop: tell them your specs, they will custom build one or more for you :) pre-assembled and all |
| 22:06.41 | dlynes_laptop | wltjr: cool |
| 22:06.43 | generalhan | wunderkin: so if i put it on mountain and take off the DST then it will stay the same distance from UTC ? and not follow mountain when they change ? |
| 22:06.48 | dlynes_laptop | wltjr: thanks for the link |
| 22:06.49 | wltjr | dlynes_laptop: they were tiny, fanless units |
| 22:07.05 | dlynes_laptop | wltjr: i've been looking at WRAP and Soekris systems |
| 22:07.15 | wunderkin | yeah... |
| 22:07.17 | j0 | is 400mhz fast enough for 4-6 g729 channels? |
| 22:07.48 | [TK]D-Fender | j0 : Not in *. |
| 22:08.05 | generalhan | i wonder why its not working ... half of my phones are an hour behind the others, and they all use the same SipDefault config ... ok thanks for the help ... ill see if i can get more info on why they are switching |
| 22:08.23 | j0 | [TK]D-Fender: ok thanks.. any idea how many i could support? |
| 22:08.38 | [TK]D-Fender | j0 : You'd be lucky to even get 1. |
| 22:08.43 | Supaplex | !flame generalhan's phones |
| 22:08.45 | j0 | what?! |
| 22:08.47 | Supaplex | ;) |
| 22:09.00 | JT | j0: you don't have a faster computer? |
| 22:09.06 | j0 | [TK]D-Fender: i was getting 3 fine, but due to some problens need to rebuild it |
| 22:09.17 | Grnd-Wire | [TK]D-Fender: Seriously!? wow.. What kind of CPU requirements are on that codec? |
| 22:09.19 | j0 | JT: i just have a nice fanless system running at 400mhz that would be nice to use |
| 22:09.20 | wunderkin | jt, 8088 isn't fast enough? or trs80? |
| 22:09.24 | [TK]D-Fender | j0 : Aside from the rest of processes a server will end up running, that is not an effecient processor, and G.729 is very taxing. |
| 22:09.38 | j0 | [TK]D-Fender: it will be dedicated to * |
| 22:09.56 | [TK]D-Fender | wunderkin : Cleary the 8088 would do better, no? it has like *3* "eights" in it! |
| 22:09.59 | JT | yeah well you need a lot of that power just to run things with g.729 |
| 22:10.14 | JT | without i meant |
| 22:10.22 | JT | so running asterisk is a base load |
| 22:10.23 | generalhan | ok how about this one ... is there documentation that explains how to make one of the line buttons on the 7960 do a kind of line status so that i can see when this person is on the phone ? |
| 22:10.36 | Grnd-Wire | hmm.. I've seen bandwidth studies, and all that sorts of stuff.. I've never seen processor power requirements, or rather, recommendations. |
| 22:10.53 | JT | Grnd-Wire: there's stuff on the wiki |
| 22:11.00 | hardwire | heya! |
| 22:11.03 | hardwire | echocan on t1 cards |
| 22:11.05 | Supaplex | just grab a cray3 |
| 22:11.06 | Grnd-Wire | generalhan: You're looking for "SIP HINTS", and "BLF buttons" |
| 22:11.18 | generalhan | Grnd-Wire: thanks ill take a look now |
| 22:11.23 | hardwire | is it better to buy echocan on a 4 port t1 card or just add an extra CPU? |
| 22:11.32 | hardwire | CPU = $700.. echocan = $1400+ |
| 22:11.41 | Grnd-Wire | heh |
| 22:12.02 | voidans | i have never had issus with the te110p, regarding echo |
| 22:12.11 | hardwire | I had massive issues with that board up here |
| 22:12.12 | voidans | is that the question? |
| 22:12.12 | hardwire | Alaska! |
| 22:12.16 | hardwire | the great stupid! |
| 22:12.17 | voidans | ah |
| 22:12.23 | voidans | they sell a two port one with a dsp now |
| 22:12.25 | [TK]D-Fender | hardwire : Umm... EC on a 4 port T1 card does not cost $1400 more.... |
| 22:12.41 | hardwire | [TK]D-Fender: 8 port sangoma.. sosorry! |
| 22:12.45 | voidans | hardware echo cancollation |
| 22:12.46 | hardwire | its around there |
| 22:12.49 | hardwire | just throwing around numbers |
| 22:12.58 | voidans | 8 port? |
| 22:13.02 | hardwire | sisi |
| 22:13.03 | [TK]D-Fender | hardwire : Forget about CPU, it wouldn't SURVIVE 8 T1's of EC. |
| 22:13.04 | Supaplex | 530582048! |
| 22:13.06 | JT | yes angoma makes 8 port |
| 22:13.11 | hardwire | [TK]D-Fender: ok |
| 22:13.13 | *** part/#asterisk ru_wing (n=wing@82.179.129.5) |
| 22:13.17 | [TK]D-Fender | Supaplex : 8675309! |
| 22:13.18 | JT | sangoma |
| 22:13.20 | hardwire | JT: know anybody thats used it with *? |
| 22:13.26 | JT | nup |
| 22:13.36 | hardwire | I am bulding a monitoring system for a call center |
| 22:13.38 | clive- | JT thats got a dsp for ec |
| 22:13.41 | hardwire | so 4 t1's worth of recordings |
| 22:13.51 | clive- | heard good things abt sangoma |
| 22:13.52 | [TK]D-Fender | hardwire : Just like the 4-port, oly, 4 more :) |
| 22:14.00 | hardwire | I never used sangoma with asterisk |
| 22:14.03 | hardwire | always digium |
| 22:14.16 | [TK]D-Fender | hardwire : Many here have, and you jsut missed the discussion. |
| 22:14.20 | hardwire | I would love to use a faster bus however :) So I am gonna try sangoma |
| 22:14.22 | JT | yeah 192 to 240 channels of software echo cancellation, forget it |
| 22:14.31 | hardwire | JT: damnit |
| 22:14.40 | justinc- | hardwire: amsuite.sourceforge.net .. been done :) |
| 22:14.50 | hardwire | I was really hoping I could vampire tap a bunch of t1's :) |
| 22:14.52 | [TK]D-Fender | hardwire : Work very well with 1 exception, and we're counting him like our "virgin sacrifice" to ensure our future properity :) |
| 22:15.19 | hardwire | justinc-: I was just going to make a dialplan that records both sides and feeds that to a sorting program |
| 22:15.34 | justinc- | oh .. heh |
| 22:15.45 | justinc- | thought you wanted like .. a gui tool and statistics and such .. |
| 22:15.51 | JT | it intercepts comms to an existing pbx? |
| 22:15.54 | hardwire | I dunno how you got that :) |
| 22:16.11 | hardwire | justinc-: I haven't seen that before |
| 22:16.18 | voidans | describe needs to be removed from irc. |
| 22:16.36 | justinc- | "I am bulding a monitoring system for a call center" |
| 22:16.37 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 22:16.38 | voidans | its like an antiquated emoticon. |
| 22:16.41 | JT | hardwire: so it will be intercepting? |
| 22:16.45 | justinc- | that's how I got it hehe .. .guess I misunderstood :) |
| 22:16.45 | voidans | hah. |
| 22:16.57 | justinc- | take a look .. it's pretty cool |
| 22:17.07 | hardwire | JT: the idea is span 1 and 2 are bridged |
| 22:17.10 | hardwire | thats about it |
| 22:17.19 | hardwire | span1 emulates the network feeding span 2 |
| 22:17.25 | JT | via the dialplan |
| 22:17.26 | JT | right |
| 22:17.29 | hardwire | indeed |
| 22:17.36 | hardwire | ah. |
| 22:17.40 | JT | yeah i've done similar |
| 22:17.44 | hardwire | it will be intercepting inbound calls as well |
| 22:17.54 | JT | sure |
| 22:17.58 | hardwire | JT: There is a BRI product that is 100% failsafe |
| 22:18.18 | hardwire | it just listens on the wires and decodes |
| 22:18.25 | JT | since it sounds like you'll be doing massive call rates, just a tip, use Monitor not MixMonitor |
| 22:18.32 | JT | dunno about 100% :) |
| 22:18.34 | JT | junghanns? |
| 22:18.43 | justinc- | JT: why monitor over mixmonitor? |
| 22:18.57 | JT | mixmonitor chokes under extreme volume i've heard |
| 22:18.58 | hardwire | JT: it taps the line |
| 22:19.02 | justinc- | ah |
| 22:19.12 | hardwire | the only way its not failsafe is if the equipment literally catches on fire and melts |
| 22:19.12 | JT | so you're better to post mix it yourself some other way |
| 22:19.21 | justinc- | if you tell monitor to join the channels with the 'm' switch tho you get pretty bad CPU spikes |
| 22:19.26 | hardwire | JT: yeh monitor. |
| 22:19.26 | justinc- | <- already fixed that problem :) |
| 22:19.41 | hardwire | I was going to run a niced process encoding all the raw to ogg's |
| 22:19.46 | JT | m switch =? |
| 22:19.57 | hardwire | 2 channel |
| 22:20.00 | justinc- | yeah the m (mix) switch |
| 22:20.12 | JT | right |
| 22:20.14 | JT | postmix |
| 22:20.18 | justinc- | it runs soxmix when monitor is done recording and merges the -in.gsm and -out.gsm files |
| 22:20.21 | JT | would be better to postmix on another machine |
| 22:20.30 | justinc- | yep |
| 22:20.34 | hardwire | yeh.. |
| 22:20.38 | JT | eww gsm, better to convert to mp3 or ogg |
| 22:20.47 | JT | from wav |
| 22:20.47 | justinc- | heh |
| 22:20.53 | justinc- | all depends on yer needs :) |
| 22:20.59 | justinc- | my call monitor ppl aren't picky :) |
| 22:21.03 | hardwire | yeh. like if you want to play it back |
| 22:21.03 | JT | i need to be able to hear clearly |
| 22:21.05 | JT | heh |
| 22:21.18 | hardwire | I chose ogg because you can load that into winamp |
| 22:21.24 | justinc- | nah we hire legally deaf people to listen to our calls :) |
| 22:21.25 | hardwire | which almost every manager descretely has on their system |
| 22:21.40 | voidans | winamp |
| 22:21.42 | voidans | in 2007 |
| 22:21.44 | voidans | ? |
| 22:21.45 | voidans | !?! |
| 22:21.51 | JT | voidans: yeah, so? |
| 22:21.54 | hardwire | I am sort of thinking about writing a flash app to isolate channels and do the mixing itself via your browser |
| 22:22.02 | JT | still better than WMP for audio |
| 22:22.02 | voidans | i havent seen winamp for 10 years |
| 22:22.05 | *** part/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
| 22:22.06 | justinc- | ouch |
| 22:22.08 | wunderkin | holy crap, speaking of monitor, my monitor directory is almost 4 gig, from about sep, only personal calls, recording in ulaw.. hehe :D |
| 22:22.13 | *** join/#asterisk tecnico (n=tecnico@24.96.146.69) |
| 22:22.16 | justinc- | that sounds like a lot more work than soxmix file1 file2 outfile |
| 22:22.47 | HushPe | WinAmp - it really whips the llama's ass |
| 22:22.52 | JT | hardwire: you should load the monitoring machine up with ram |
| 22:22.53 | justinc- | lol |
| 22:22.54 | JT | and cpu |
| 22:22.56 | HushPe | atleast when i was using windows and it was 2.* |
| 22:23.13 | *** join/#asterisk tecnico (n=tecnico@24.96.146.69) |
| 22:23.15 | JT | voidans: well that's your problem isn't it? :) |
| 22:23.20 | hardwire | JT: the recording machine? |
| 22:23.25 | JT | yes |
| 22:23.34 | voidans | what? |
| 22:23.42 | voidans | that i havent had windows for 10 years? :) |
| 22:23.43 | wltjr | dlynes_laptop: well ty for your help, I am stuck in the mud for today |
| 22:23.49 | hardwire | its a Dual Core Xeon 5130 with 4 gigs of ram |
| 22:23.49 | JT | that you haven't seen winamp |
| 22:23.53 | JT | ah ok |
| 22:24.00 | wltjr | box is down, I assume kernel panic, can't come back up, and likely need to livecd it :( |
| 22:24.05 | hardwire | still think I can't echocan everything in software? |
| 22:24.13 | JT | of course not |
| 22:24.17 | JT | it's 200 damn channels |
| 22:24.20 | JT | are you insane |
| 22:24.24 | hardwire | well not all at once |
| 22:24.28 | hardwire | but it would be nice right :) |
| 22:24.37 | *** join/#asterisk GaVak (n=denniso@adsl-074-228-124-003.sip.sav.bellsouth.net) |
| 22:24.44 | hardwire | hehe |
| 22:24.48 | hardwire | one of them is an autodialer |
| 22:24.50 | hardwire | fucker |
| 22:24.51 | JT | you can afford an 8 port pri card, you can afford hardware ec |
| 22:25.07 | *** join/#asterisk essenza (i=xtc@unaffiliated/essenza) |
| 22:25.16 | GaVak | I just switchted over from my own build of * to a Trixbox release. I've got it all working great except that it is building out my server-to-server IAX calls in GSM instead of g729. |
| 22:25.17 | hardwire | ok ram |
| 22:25.21 | hardwire | whats dual ranked fully buffered |
| 22:25.22 | GaVak | Anyone know where that is set in Trix? |
| 22:25.24 | hardwire | vs single ranked? |
| 22:25.34 | JT | ~trixbox |
| 22:25.36 | jbot | from memory, trixbox is unable to be supported here. Try joining #freepbx and asking there, or use the trixbox forums at http://www.trixbox.org/modules/newbb/ |
| 22:25.37 | hardwire | its been a while since I priced a server out that wasn't made form leftover 70's gear |
| 22:25.43 | GaVak | thnx |
| 22:25.52 | generalhan | ok so i got the sip hints working ... and when i "show hints" in CLI it shows the status of that line ... im a bit confused as to how i am to get that status to display on my 7960 ... can anyone push me in the right direction ? |
| 22:26.02 | voidans | ~aah |
| 22:26.03 | jbot | i heard aah is AMP and amp is, like, NOT supported here! people using it should join #freepbx (the new name of amp) |
| 22:26.14 | [TK]D-Fender | generalhan : Cisco does not support Presence. You are up a creek. |
| 22:26.16 | hardwire | JT: I would think 2GB ram would be fine |
| 22:26.36 | JT | i thought you said yhou had 4 |
| 22:26.39 | *** join/#asterisk CPSK (n=CPSK@c6.ars.ba.nextra.sk) |
| 22:26.40 | wunderkin | o rly? |
| 22:26.46 | hardwire | I am speccing it out |
| 22:26.50 | generalhan | [TK]D-Fender: i know i have done this some how before ... not quite sure i remember ... but it wasnt with "hints" so now i have to go through all the changes i ever made to see if i can find it again i guess |
| 22:26.51 | hardwire | I am in the cut corners phase |
| 22:27.04 | JT | what an unusual phase |
| 22:27.15 | [TK]D-Fender | hardwire : And I think Bush & Cheney should be shot, preferrably at close range as to only require 1 bullet and save on costs. Guess we're both to remain disappointed! |
| 22:27.39 | wunderkin | hillary clinton for president! it will happen |
| 22:27.56 | hardwire | [TK]D-Fender: heh |
| 22:27.57 | [TK]D-Fender | generalhan : only place I've suggested for it it to make an XML browser script for it. |
| 22:28.00 | JT | the us presidential election looks funny |
| 22:28.00 | CPSK | Hi everybody.. Anobody experienced in Tenovis Integral vs. Digium E1 card? Thank for any help! |
| 22:28.03 | hardwire | wunderkin: then she can cheat on bill |
| 22:28.05 | hardwire | its about revenge |
| 22:28.08 | wunderkin | heh heh |
| 22:28.11 | JT | first woman, or first black man, or first hispanic man |
| 22:28.24 | hardwire | first illegal immigrant |
| 22:28.36 | JT | the woman is white, so she will probably win |
| 22:28.47 | hardwire | can you get an un-tan? |
| 22:29.04 | [TK]D-Fender | hardwire : Check in to Neverland.... |
| 22:29.05 | voidans | yeah |
| 22:29.10 | voidans | edwards for president. |
| 22:30.05 | anonymouz666 | bill gates for president. |
| 22:30.35 | *** join/#asterisk Globetrotter (n=eric@205.211.214.167) |
| 22:30.37 | JT | no thanks |
| 22:30.45 | CPSK | Hi everybody.. Anobody experienced in Tenovis Integral vs. Digium E1 card? Thank for any help! |
| 22:30.45 | wunderkin | the us would crash |
| 22:31.04 | [TK]D-Fender | You guys would have just fine if Ross Perot wone back in the day. He was ready to pay off 10% of the national debt out of his own purse! Now look what's become of it... |
| 22:31.26 | J4k3 | [TK]D-Fender: we got bill clinton instead, who dropped it to about 1/3rd of the starting size. |
| 22:31.35 | wltjr | we had a surplus at end of clinton years, oh yeah and a over valued stock market and bubble ;) |
| 22:31.41 | hardwire | heh |
| 22:31.44 | Globetrotter | hi guy how do i give the astereisk user permission to a particular folder?? (/etc/asterisk) i am getting a permissions erro when trying to run * |
| 22:31.46 | Globetrotter | <PROTECTED> |
| 22:31.49 | sevard | and ROss Perot didn't have any balls either, that means that we wouldn't be in this shitty war either! |
| 22:31.52 | wltjr | who cares about govt debt, it's not like some other country will come break our knees to collect :) |
| 22:31.57 | hardwire | I think I could write a flash app to access .wav files recorded to the drive |
| 22:32.04 | wltjr | or repo our nukes :) |
| 22:32.13 | JT | eww flash |
| 22:32.26 | JT | the only flash app worth writing is one that deletes itself |
| 22:32.27 | hardwire | JT: it would make deploying a product easier |
| 22:32.38 | JT | across windows perhaps |
| 22:32.40 | hardwire | they could either download the audio or hit play and it plays |
| 22:32.43 | sevard | Globetrotter: you should read up on linux and the chown and chmod commands. |
| 22:32.47 | hardwire | JT: You got an amiga? |
| 22:32.52 | JT | linux |
| 22:32.55 | JT | this little os |
| 22:33.00 | JT | you may have heard of it |
| 22:33.02 | [TK]D-Fender | wltjr : You mean like China's threats tos ell off their cash USD inventory for euros and cause the US to start a new Depression? :) |
| 22:33.04 | hardwire | JT: yeh maybe |
| 22:33.20 | hardwire | JT: and you only install free software right? |
| 22:33.31 | wltjr | [TK]D-Fender: sure and when the US shuts down tech stuff with China, where will they be? |
| 22:33.32 | JT | not always |
| 22:33.34 | hardwire | so downloading adobe flash player plugins for firefox is against your code right? |
| 22:33.34 | sevard | I only install free software. |
| 22:33.46 | J4k3 | [TK]D-Fender: money is a bullshit confidence game anyways... |
| 22:33.46 | wltjr | no MS, Cisco, yahoo, ebay, rh, sun, IBM etc :) |
| 22:33.47 | sevard | it's aginst mine. |
| 22:33.48 | hardwire | heh |
| 22:33.49 | JT | hardwire: wtf, adobe flash player is fucking shit |
| 22:33.50 | hardwire | yeh |
| 22:33.52 | Globetrotter | ok thanks for the advice.. but can you give me the syntax please? |
| 22:33.54 | JT | and hardly works with anything but IE |
| 22:34.00 | J4k3 | american dollars are backed by hype and bullshit... |
| 22:34.00 | hardwire | weird |
| 22:34.00 | sevard | Globetrotter: you should read up on linux and the chown and chmod commands. |
| 22:34.07 | hardwire | JT you have a very different view on this than I do :) |
| 22:34.08 | JT | it's just trash |
| 22:34.14 | hardwire | I don't like flash much. |
| 22:34.15 | JT | it's reality |
| 22:34.17 | Globetrotter | gracias guys |
| 22:34.19 | hardwire | but it works well on my linux machine |
| 22:34.26 | sevard | If you need help for chmod or chown syntax, you shouldn't be using asterisk. sorry buddy, this is linux. |
| 22:34.27 | JT | to guarantee flash to work, you pretty much need windows |
| 22:34.32 | JT | well is an overstatement |
| 22:34.39 | JT | they never keep up with new versions |
| 22:34.44 | hardwire | guarantee maybe |
| 22:34.46 | JT | they work on limited browsers |
| 22:34.46 | sevard | apparently flash9 is almost out of beta |
| 22:34.46 | J4k3 | JT: no, you just code for an ancient version of flash that all the linux folks can't argue about. |
| 22:34.50 | JT | and usually crash |
| 22:34.51 | sevard | for linux that is |
| 22:35.04 | hardwire | on linux |
| 22:35.10 | sevard | sehr gut |
| 22:35.14 | hardwire | its the only way to get my favorite porno site to work |
| 22:35.21 | hardwire | sigh |
| 22:35.24 | JT | it's a proprietry standard anyway, and not one that is well supported like pdf |
| 22:35.26 | sevard | I don't even run X windows anymore. |
| 22:35.27 | wltjr | flash sucks MM refused to properly port to other OS back in the day |
| 22:35.35 | sevard | WELL SUPPORTED LIKE A PDF? |
| 22:35.37 | wltjr | the Flash standalone player should be ported to all OS |
| 22:35.40 | sevard | try a fucking HTML document. |
| 22:35.43 | *** join/#asterisk af_ (n=af@ip-173-157.sn1.eutelia.it) |
| 22:35.49 | JT | sevard: hey, if you compare the two |
| 22:35.51 | wltjr | sevard: agreed, Adobe is no better |
| 22:35.56 | JT | i agree pdf is a bit of a abomination |
| 22:36.02 | wltjr | sevard: they had a booth at LWE in 05 it was a joke |
| 22:36.04 | sevard | PDF is ass. |
| 22:36.04 | JT | oh it's better in terms of readability |
| 22:36.06 | hardwire | you know they are leading the way on free and open document systems right ? |
| 22:36.08 | J4k3 | sevard: hey, at least the 3rd party pdf viewers kinda work... try coding a website with css and getting different browsers to view it properly. |
| 22:36.12 | hardwire | it just took a while |
| 22:36.15 | JT | and there are free writers |
| 22:36.16 | hardwire | and a lot of hurt feelings I guess |
| 22:36.17 | hardwire | heh |
| 22:36.19 | JT | flash is just trash |
| 22:36.23 | sevard | J4k3: that's IE's fault. not mine. |
| 22:36.29 | wltjr | back in the day I read allot that said pdf was going to be html, but adobe refused to open the format |
| 22:36.30 | J4k3 | sevard: its everybody's fault equally. |
| 22:36.38 | wltjr | and thus HTML prevaled |
| 22:36.39 | J4k3 | firefox is no less broken than ie when it comes to css |
| 22:36.40 | sevard | I use telnet to browse the web. |
| 22:36.42 | J4k3 | its just broken differently. |
| 22:36.46 | *** join/#asterisk ardor (n=vircuser@las-static-66.18.135.148.mpowercom.net) |
| 22:36.54 | *** join/#asterisk xpot (n=jim@71-213-32-194.slkc.qwest.net) |
| 22:37.11 | hardwire | forget I said anything :) |
| 22:37.15 | sevard | haha. |
| 22:37.18 | ardor | Hi guys, i was wondering what is the voip service provider to use these days that support IAX and provide phone numbers for incoming calls? |
| 22:38.06 | hardwire | ardor: I am starting to think there needs to be a channel just for your question |
| 22:38.06 | hardwire | filled with eager sales people ready to answer that |
| 22:38.06 | xpot | I use gafachi |
| 22:38.08 | J4k3 | hardwire: I wish there was a #voipservice channel or something |
| 22:38.09 | hardwire | ardor: also teliax works well |
| 22:38.11 | ardor | but, i dont want to be answerd by sells people. =) |
| 22:38.21 | J4k3 | where all the IRC-friendly sip/iax/consultants/etc would hang out and be easy to find :) |
| 22:38.21 | sevard | ardor: there are HUGE articles on voip-info.org that list VoIP providers that offer DIDs |
| 22:38.25 | hardwire | ardor: and non sales people don't often want to answer :) |
| 22:38.41 | hardwire | bah |
| 22:38.48 | hardwire | lets talk about the DMCA some |
| 22:39.01 | sevard | teliax works well if you don't mind paying 5x what you should for VoIP |
| 22:39.01 | hardwire | heh |
| 22:39.03 | hardwire | no lets not |
| 22:39.07 | hardwire | sevard: haha |
| 22:39.17 | hardwire | I was AK terminiation/origination for him for a week |
| 22:39.19 | hardwire | it was a test |
| 22:39.28 | sevard | on the other hand, they were the ONLY voip carrier that could give me a rate center local to me |
| 22:39.34 | hardwire | it wasn't poorly priced, however they never offered IAX trunking |
| 22:39.38 | sevard | I did MN term for 5 months |
| 22:39.41 | JT | are there us based did providers who will do dids with no monthly fee? i'm guessing not :) |
| 22:39.59 | sevard | hardwire: when I had them I had IAX trunks |
| 22:40.23 | sevard | JT: you can get free DIDs in places like rural new york |
| 22:40.38 | JT | that's unusual, heh |
| 22:40.53 | sevard | if you're looking for just incoming it's not hard to find free access |
| 22:40.59 | hardwire | sevard: maybe it was just my codec they didn't like |
| 22:41.02 | hardwire | and the trunk parameters |
| 22:41.10 | clyrrad | Has anyone here found a way to have Asterisk respond to a Sipura device after sending a SIP NOTIFY from the CML? What I am trying to do is have a phone resync when instructed to from the CML, when I send the SIP NOTIFY resync command the ATA responds back with 401 (as its supposed to) - but then you need to You'll need to supply account password |
| 22:41.10 | clyrrad | information in authorization header (computed with MD5 digest) |
| 22:41.11 | sevard | they had codec issues for a long time |
| 22:41.11 | hardwire | speex :) |
| 22:41.15 | sevard | speex :( |
| 22:41.18 | hardwire | and set for uber distanc satellite stuff |
| 22:41.20 | JT | wouldn't outgoing to easier to find with no monthly fee? as in no did number |
| 22:41.32 | clyrrad | has anyone done this before? |
| 22:41.38 | dlynes_laptop | wltjr: no problem |
| 22:41.49 | dlynes_laptop | wltjr: thanks for the tip about that company too |
| 22:41.50 | sevard | you're on a sat connection? oh man, i forgot-- AK. Dude, your lag must be 800 freaking ms how were you able to complete a call? |
| 22:41.59 | dlynes_laptop | wltjr: just finished talking to ed jamison there about it |
| 22:42.10 | dlynes_laptop | wltjr: he's shipping me out some literature today and tomorrow on them |
| 22:42.27 | dlynes_laptop | wltjr: my boss is quite excited about it, because they do branding, too |
| 22:44.16 | clyrrad | any takers? |
| 22:44.29 | wltjr | dlynes_laptop: they are bad ass |
| 22:44.41 | *** join/#asterisk FaithX (n=faithful@ns.linuxterminal.com) |
| 22:44.44 | dlynes_laptop | wltjr: yeah...the branding's a bit on the expensive side |
| 22:44.49 | hardwire | sevard: speex was great |
| 22:44.54 | hardwire | in rural alaska its sat |
| 22:44.56 | dlynes_laptop | wltjr: but, otoh...it'll help sell our systems easier, too |
| 22:45.01 | JT | clyrrad: what the hell is a CML? |
| 22:45.04 | hardwire | mainland alaska usually isn't |
| 22:45.04 | sevard | speex was great in 1998 |
| 22:45.10 | hardwire | speex is still great |
| 22:45.11 | wltjr | dlynes_laptop: I have spoken with their peeps at LWE, and have the specs on irc logs from a system a dude had built specifically for * and voip, they were like remote boxes for satelite offices, some of which were oil rigs in the gulf :) |
| 22:45.20 | sevard | speex is bad, dude |
| 22:45.27 | wltjr | dlynes_laptop: oh if you are making and selling systems, it's the only way to go |
| 22:45.28 | JT | it's better than ilibc |
| 22:45.32 | hardwire | sevard: you think icons are bad. |
| 22:45.43 | wltjr | dlynes_laptop: they will do all kinds of stuff for you as well, crunch numbers show you alternatives etc |
| 22:45.46 | sevard | hell yes icons are bad. |
| 22:45.53 | hardwire | sevard: speex is good stuff |
| 22:45.55 | hardwire | its come a long way |
| 22:46.02 | hardwire | and it handled sat calls like a fizucking champ |
| 22:46.06 | sevard | it still sounds like you're talking undrewater with a dildo in your mouth |
| 22:46.09 | wltjr | dlynes_laptop: but on this driver front, not to sure :) |
| 22:46.12 | dlynes_laptop | wltjr: cool |
| 22:46.14 | hardwire | sevard: ahahaha. |
| 22:46.14 | k-man__ | so if i want to install asterisk on a debian system, what is the recommended approach? ie, build from source? find a prebuilt binary? etc |
| 22:46.24 | dlynes_laptop | wltjr: yeah...i don't care about the driver front |
| 22:46.25 | hardwire | sevard: yeh, when you forget to configure it, it can sound like crap. |
| 22:46.26 | wltjr | going to check logs, but pretty sure I am getting kernel panics, I will try to compile it again |
| 22:46.28 | JT | CPSK: stop repeating already |
| 22:46.32 | dlynes_laptop | wltjr: i'm planning to do all that shit myself, anyways |
| 22:46.43 | dlynes_laptop | wltjr: as long as the basics are supported |
| 22:46.50 | dlynes_laptop | wltjr: and i can upgrade the kernel version whenever i feel like it |
| 22:46.52 | clyrrad | k-mann___: easy enough to build from source |
| 22:46.53 | hardwire | sevard: it handled everything stupidly well |
| 22:46.54 | wltjr | dlynes_laptop: yeah I bet you don't :) I was refering to my driver problems, not them pre-loading :) |
| 22:46.55 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 22:46.55 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 22:47.01 | hardwire | better than gsm and ilbc |
| 22:47.08 | wltjr | dlynes_laptop: they don't get into the software stuff, BUT will preload your images if you provide them :) |
| 22:47.16 | dlynes_laptop | wltjr: yeah...forget that |
| 22:47.24 | dlynes_laptop | wltjr: i was talking to him about an ethernet LOM port |
| 22:47.33 | dlynes_laptop | wltjr: then i don't need to both with that shit :) |
| 22:47.37 | sevard | hardwire: g711 ulaw. |
| 22:47.49 | hardwire | sevard: over 12kbps available sat space? |
| 22:48.01 | sevard | use g729 then |
| 22:48.11 | JT | that wont fit in 12kbit.s |
| 22:48.19 | hardwire | g729 was nice, it worked well too.. why are you convincing me away from speex again? |
| 22:48.22 | EmleyMoor | Any of you lot good with voiptalk over IAX? |
| 22:48.26 | sevard | I thought it was supposed to be 9kbit/s with overhead |
| 22:48.30 | k-man__ | clyrrad, ok, ill give that a go |
| 22:48.43 | JT | not any voip overhead |
| 22:48.43 | hardwire | sevard: yeh but speex had better error recovery |
| 22:49.07 | wltjr | dlynes_laptop: by driver I was referring to my current situation, I guess I will try to recompile wanpipe driver with voice only again |
| 22:49.17 | sevard | i'm not doubting, i just strive for toll quality, if you don't have the bandwidth you shouldn't be doing voip. |
| 22:49.25 | kgx | ok a little bit of an asterisk noob here...if a phone comes in and someone picks it up, how i can trigger a script as soon as they pick it up? |
| 22:49.33 | hardwire | sevard: hahaha |
| 22:49.34 | wltjr | dlynes_laptop: will suck if I can't go that route, and can only build and load one with both voice and data support, which eventually I will have to since I will be doing voice+data ;) |
| 22:49.44 | JT | CPSK: FFS. |
| 22:49.47 | JT | CPSK: stop. |
| 22:49.47 | hardwire | sevard: your a hoot. |
| 22:49.52 | *** join/#asterisk dseeb_ (n=dcb@58.165.244.192) |
| 22:49.53 | sevard | :) |
| 22:50.00 | dlynes_laptop | hardwire: yeah...he's always pretty funny |
| 22:50.15 | hardwire | sevard: I won't be contacting you for anything that requires any sort of ingenuity I guess. |
| 22:50.20 | sevard | i'm hungry ;\ please sir, spare a fiver? |
| 22:50.23 | dlynes_laptop | hardwire: but he doesn't have a bandwidth bill, either :) |
| 22:50.32 | sevard | hardwire: I'm all about ingenuity bitch |
| 22:50.32 | hardwire | dlynes_laptop: sounds like it |
| 22:50.53 | hardwire | sevard: no, if you don't have the bandwidth you shouldn't be using voip right? |
| 22:50.54 | dlynes_laptop | hardwire: and he loves using 'bitch' a lot, too :0 |
| 22:50.58 | sevard | I make computers out of gum, toothpicks and paperclips |
| 22:51.02 | hardwire | what about if there are no options other than voip? |
| 22:51.13 | sevard | i'm fucking mac gyver |
| 22:51.13 | hardwire | because there is no telephone provider |
| 22:51.21 | sevard | then you use speex, silly! |
| 22:51.26 | hardwire | fu |
| 22:51.30 | sevard | hahaha. |
| 22:52.00 | hardwire | if I ever hear anybody talking like you on the street, I will kill them on the spot just hoping it was you. |
| 22:52.04 | sevard | hahahaha |
| 22:52.06 | sevard | awesome |
| 22:52.15 | hardwire | where do you live anyways? |
| 22:52.20 | sevard | no, i usually do voip over LANs, as that's where all the work I get is for |
| 22:52.20 | hardwire | just sos I knows |
| 22:52.30 | *** join/#asterisk Jingles (n=dfbarth@39.183.dowl.anc.borealisbroadband.net) |
| 22:52.30 | sevard | Minnesota, I was in AK last year, me loves sitka |
| 22:52.44 | hardwire | I haven't been to Sitka yet |
| 22:52.47 | Jingles | there's not much in sitka to love. |
| 22:52.51 | sevard | I almost went to school in Sitka |
| 22:52.53 | hardwire | but everybody seems to like naming their dogs after it |
| 22:53.04 | sevard | didn't know that was a trend |
| 22:53.08 | hardwire | augh |
| 22:53.13 | sevard | Jingles: what's not to love about sitka? |
| 22:53.13 | hardwire | Kenai/Sitka/Denali |
| 22:53.23 | hardwire | 3 most popular dog names for the unimaginative alaskan. |
| 22:53.30 | Jingles | lol |
| 22:53.34 | sevard | I spent a month there just fishing and living out in a boat |
| 22:53.39 | Jingles | are there more alaskans in here other than me? |
| 22:53.42 | Jingles | that's comedy. |
| 22:53.44 | [TK]D-Fender | sevard : I didn't know Richard Dean Anderson was gay...... |
| 22:53.56 | EmleyMoor | I who can hear a normal phone on the PSTN was having to use amplifiers on asterisk |
| 22:53.57 | sevard | [TK]D-Fender: what are you smoking |
| 22:53.58 | hardwire | http://www.google.com/maps?q=Sitka,+AK&ie=UTF8&z=9&ll=57.201759,-170.337524&spn=0.95366,3.47168&om=1 |
| 22:54.02 | hardwire | I was here |
| 22:54.22 | hardwire | http://www.google.com/maps?q=Sitka,+AK&ie=UTF8&om=1&z=12&ll=53.758454,-166.959229&spn=0.130103,0.43396 |
| 22:54.23 | hardwire | and there |
| 22:54.27 | hardwire | sorry for the misleading url |
| 22:54.29 | [TK]D-Fender | sevard : YOU said it :) |
| 22:54.37 | Bobthehunter | is therea known device that doesnt work on SRV |
| 22:54.38 | hardwire | the last one was difficult |
| 22:55.13 | dlynes_laptop | Bobthehunter: Azahell |
| 22:55.14 | sevard | [TK]D-Fender: where |
| 22:55.33 | dlynes_laptop | Bobthehunter: Grandscream Fidgetones |
| 22:55.44 | [TK]D-Fender | [17:51] <sevard> i'm fucking mac gyver |
| 22:55.44 | Bobthehunter | ok no one uses that lol |
| 22:55.50 | sevard | :D |
| 22:55.54 | hardwire | http://www.flickr.com/photos/skyefencer/148474203/ |
| 22:55.56 | hardwire | why I love alaska |
| 22:56.00 | hardwire | dogs chicks and snow |
| 22:56.04 | sevard | [TK], i don't say this much, and you'll probably never hear this again, but I love you, man. |
| 22:56.13 | dlynes_laptop | hardwire: and you have sex with them in that order? |
| 22:56.21 | sevard | hardwire: who knew alaskian's had sex with baby chickens |
| 22:56.27 | hardwire | dlynes_laptop: how do I respond!? |
| 22:56.30 | sevard | TWO SEXUAL JOKES IN ONE THIS MUST BE IRC |
| 22:56.31 | [TK]D-Fender | sevard : With that in context...... *I'm NOT gay* <--------- |
| 22:56.32 | Jingles | hardwire: *nods* I'm currently living in Anchorage. |
| 22:56.36 | hardwire | Jingles: yay! |
| 22:56.39 | hardwire | I am in Eagle River |
| 22:56.41 | sevard | [TK]D-Fender: oh girlfriend you so are |
| 22:56.44 | Jingles | no kidding? nice. |
| 22:56.46 | hardwire | working downtown at the historic train depot |
| 22:56.47 | sevard | hardwire: I loves me snow. |
| 22:56.57 | Jingles | we're having plenty of the snow at the moment. |
| 22:57.07 | sevard | Anchorage: The Anus of AK |
| 22:57.18 | Jingles | that's wasilla, baby. |
| 22:57.23 | sevard | haha |
| 22:57.26 | EmleyMoor | Is Cicely a real place or is it made up? |
| 22:57.30 | sevard | I really dislike anchorage |
| 22:57.31 | Jingles | totally made up. |
| 22:57.36 | wltjr | Anchorage only city physically larger than mine :) |
| 22:57.41 | hardwire | Jingles: hatchers pass |
| 22:57.53 | Jingles | Though there was an entry in the Milepost some years ago for 'Cicely, AK' |
| 22:58.00 | Jingles | the page that the entry pointed to was that index page. |
| 22:58.11 | sevard | hardwire: bring some of your wonderful snow back to minnesota, we had a YELLOW christmas and LA, CA got our damned snow |
| 22:58.38 | *** join/#asterisk Dovid (n=Dovid@l192-117-114-1.broadband.actcom.net.il) |
| 22:58.38 | *** join/#asterisk Ritalin2 (n=dave@74-34-103-241.dsl1.pwll.tn.frontiernet.net) |
| 22:58.40 | dlynes_laptop | sevard: yeah...we got a shitload of snow this year too |
| 22:58.40 | Dovid | evening all |
| 22:58.40 | dlynes_laptop | sevard: it was freaky |
| 22:58.40 | *** join/#asterisk grandy (n=chatzill@c-71-198-130-108.hsd1.ca.comcast.net) |
| 22:58.40 | Ritalin2 | hi Dovid |
| 22:58.40 | Dovid | been a loooooooooooong time since i have been here |
| 22:58.41 | dlynes_laptop | sevard: normally we get almost zero snow |
| 22:58.41 | sevard | laaaaaaaaaame, don't you live in newfoundland? |
| 22:58.45 | EmleyMoor | It's probably going to snow here in the next few days |
| 22:58.48 | dlynes_laptop | sevard: as freaking if |
| 22:58.52 | sevard | hhaha |
| 22:58.53 | wltjr | dlynes_laptop: you know now that I think about it, I might have to compile zaptel directly |
| 22:58.53 | dlynes_laptop | sevard: i'm not a damned newfie |
| 22:58.56 | sevard | BC? |
| 22:59.04 | Jingles | noooofie |
| 22:59.09 | dlynes_laptop | sevard: yeah...home of all that wonderful skunk weed |
| 22:59.13 | wltjr | dlynes_laptop: since the wanpipe setup says it's patching the zaptel sources for the dchan |
| 22:59.18 | Ritalin2 | My VoIP provider allows multiple calls at once. In theory I should be able to use an analog handset attached to an FXS port on a TDM400 and have call waiting functioning, correct? |
| 22:59.22 | sevard | BC is home of bucktoothed british wannabees |
| 22:59.27 | sevard | Winnipeg, MB is where it's at |
| 22:59.31 | [hC] | Anyone know of a Queue() example where people will be put in a holding queue for a certain amount of time (say, 10 minutes) if all available agents are currently on a call, but if the agents are FREE, it gives up after only 2 minutes of ringing? |
| 22:59.32 | dlynes_laptop | sevard: ummmmm |
| 22:59.39 | Jingles | call waiting is a function of the PSTN line itself, isn't it? |
| 22:59.41 | dlynes_laptop | sevard: it's almost all predominantly asian here |
| 22:59.46 | dlynes_laptop | sevard: just ask [hC] |
| 22:59.52 | sevard | i thought it was british imbreds |
| 23:00.09 | dlynes_laptop | sevard: nah...you're thinking of west virginia |
| 23:00.14 | sevard | hahaha |
| 23:00.16 | Ritalin2 | jingles: how so? |
| 23:00.18 | [hC] | dlynes_laptop: i would definitely disagree with the bucktoothed british wannabe's in BC, and claiming winterpeg to be better. |
| 23:00.24 | hardwire | Jingles: where you workin? |
| 23:00.27 | sevard | west virginia is full of black people who hate white people. |
| 23:00.30 | dlynes_laptop | [hC]: no doubt :) |
| 23:00.46 | sevard | winnipeg > most of CA |
| 23:00.47 | Jingles | hardwire: local tech company as the IT director. guess what I'm spending most of my time working on?! Our asterisk setup. |
| 23:00.51 | EmleyMoor | I used to live in a place called Edmonton and had trouble convincing people I wasn't in Alberta |
| 23:00.52 | dlynes_laptop | [hC]: i guess metrobridge doesn't like to wholesale, eh? |
| 23:00.57 | Ritalin2 | jingles: I have a grandstream phone. if I'm connected via SIP directly to my VoIP provider to a call and another call comes in, the next line indicator rings (it's a 4 line phone) |
| 23:01.10 | [hC] | dlynes_laptop: how so? |
| 23:01.25 | sevard | Jingles: would you like to hire a friendly and agreeable contractor? |
| 23:01.30 | [TK]D-Fender | CPSK : The odds of finding someone here who has used your particular unpopular PBX are between slim and NONE. Make you you should just state your PROBLEM. |
| 23:01.31 | dlynes_laptop | [hC]: shurg...had a meeting with them the other day...they seemed to be pretty reluctant |
| 23:01.39 | Jingles | Ritalin2: gotcha. you said analog phone initially. I thought PSTN type. |
| 23:01.54 | dlynes_laptop | [hC]: eventually they said they already had too many companies they were reselling to |
| 23:01.54 | [TK]D-Fender | "maybe you"* |
| 23:01.55 | Ritalin2 | if I try and use call waiting (either with calls to my VoIP provider or via extensions) asterisk drops one of the calls |
| 23:01.56 | Jingles | sevard: lols. if I need some asterisk expertise, I'll check here first. |
| 23:02.07 | EmleyMoor | ... and one of the three is rotary! |
| 23:02.13 | HushPe | i now have chan_zap, but not it's not working: chan_zap.so: undefined symbol: ast_pickup_call (from asterisk) the drivers are all installed (and working) |
| 23:02.14 | *** join/#asterisk FaithX (n=faithful@ns.linuxterminal.com) |
| 23:02.16 | Ritalin2 | Jingles: VoIP provider -> Asterisk -> Digium card with FXS port -> analog phone |
| 23:02.18 | sevard | Jingles: just looking for work, mate. can you blame a guy? |
| 23:02.28 | Jingles | sevard: not at all. |
| 23:02.33 | [hC] | dlynes_laptop: really? I'm at metrobridge right now. |
| 23:02.39 | [hC] | dlynes_laptop: did you come to the metrobridge office? |
| 23:02.40 | dlynes_laptop | [hC]: yeah, i know |
| 23:02.45 | dlynes_laptop | [hC]: that's why i asked you |
| 23:02.53 | dlynes_laptop | [hC]: nah...the owner of our company did |
| 23:03.02 | hardwire | Jingles: need help? |
| 23:03.03 | hardwire | heh |
| 23:03.06 | hardwire | and what company? |
| 23:03.12 | sevard | where's hmmhesays or as I like to call him "Mr. HatesEverything" |
| 23:03.18 | Jingles | Ritalin2: you know, I'm not sure how that would work. I have to configure the Zap channels, it seems like there's only one channel per signalling module available. |
| 23:03.34 | Dovid | has anyone ever messed with embeded asterisk ? |
| 23:03.35 | Jingles | I haven't actually looked to see if there's a way to shove multiple channels through one FXS module. |
| 23:03.36 | dlynes_laptop | sevard: don't you mean mr. i want to charge for every bit of help i offer on here? |
| 23:03.41 | dlynes_laptop | sevard: heh |
| 23:03.47 | Dovid | http://www.rowetel.com/ucasterisk/index.html |
| 23:03.58 | sevard | I didn't know hmmhesays did that |
| 23:04.07 | hardwire | Jingles: one channel per module |
| 23:04.11 | Dovid | what do u guys think of an IAX to SIP device? |
| 23:04.14 | hardwire | 2 wire only |
| 23:04.20 | dlynes_laptop | sevard: yeah...he learns somethign from somebody |
| 23:04.23 | hardwire | Dovid: why not just IAX? |
| 23:04.28 | dlynes_laptop | sevard: then sees someone else having the same problem |
| 23:04.35 | Ritalin2 | well I don't want two channels, I just want one channel but I want to use call waiting |
| 23:04.36 | wltjr | dlynes_laptop: how do you have your timeslots configured on your wanpipe? |
| 23:04.36 | dlynes_laptop | sevard: and offers to fix their problem for them, for a fee :0 |
| 23:04.44 | sevard | hahaha |
| 23:04.50 | wltjr | dlynes_laptop: I had mine set to all, should that have been all or 1-23? |
| 23:04.55 | Ritalin2 | hardwire: call waiting works on the zaptel modules doesn't it? I mean it's a flag to set in zapata.conf |
| 23:04.56 | Dovid | hardwire: a device for SIP only devices such as snom and polycom |
| 23:04.56 | sevard | sounds like a business man |
| 23:04.58 | Jingles | hardwire: that answers the question then, doesn't it? one call, one channel. and since the VoIP provide isn't providing call waiting, then no call waiting. |
| 23:05.09 | Jingles | VoIP *provider |
| 23:05.12 | Dovid | gona try to make an embeded device that will convert IAX to SIP |
| 23:05.24 | dlynes_laptop | wltjr: 24 |
| 23:05.25 | sevard | Dovid: like... asterisk? |
| 23:05.34 | Ritalin2 | jingles: when I was using a sipura device to connect to my voip provider call waiting worked |
| 23:05.44 | wltjr | dlynes_laptop: hmm, wonder if there is a difference between all and 24 |
| 23:05.45 | hardwire | Jingles: need more FXO? |
| 23:05.58 | dlynes_laptop | wltjr: 24 means put d channel on channel 24 |
| 23:06.01 | hardwire | I have a 4 port FXO SIP device and a lot of 1FXS/1FXO sip devices I am selling |
| 23:06.17 | Dovid | Asterisk Server ---> Internet ----> Router -----> [IAX----converted to ------ SIP] --> SIP Phone |
| 23:06.19 | Jingles | Ritalin2: the sipura is a pure voip phone though, right? |
| 23:06.19 | wltjr | dlynes_laptop: yes, I am not talking about that part, there is a part under the initial t1 config where you specify timeslots |
| 23:06.23 | Ritalin2 | jingles: when a call comes in I get a call waiting tone/beep. I do a flash and it dops the first call and switches to the second call |
| 23:06.25 | dlynes_laptop | wltjr: all i would imagine means use all channels for timeslots |
| 23:06.28 | Dovid | this way u save on the NAT headache |
| 23:06.28 | wltjr | dlynes_laptop: by default it says all |
| 23:06.41 | dlynes_laptop | wltjr: you're looking on the interface configuration? |
| 23:06.42 | Dovid | hardwire can I PM u ? |
| 23:06.47 | Ritalin2 | jingles: sorry. sipura 3000. has one fxs and one fxo port. I was just using the FXS port to connect an analog phone |
| 23:06.49 | sevard | Dovid: why don't you just register with SIP instead of creating a convuluted system that may or may not fail? |
| 23:06.51 | hardwire | Dovid: you gonna be kinky? |
| 23:07.05 | Dovid | sevard - cause SIP SUX |
| 23:07.08 | hardwire | Dovid: so this device is basically asterisk? |
| 23:07.14 | Dovid | especially over NAT |
| 23:07.15 | Dovid | yes |
| 23:07.20 | Dovid | but it will help with NAT issues |
| 23:07.24 | Dovid | emebded asterisk |
| 23:07.26 | wltjr | dlynes_laptop: yes, before protocol |
| 23:07.35 | sevard | why don't you put an end * box at the client location, hell you can run * on a WRT54G |
| 23:07.36 | Dovid | hardwire: I am a dude - dont know if u want me to go there.... |
| 23:07.38 | [TK]D-Fender | Ok, I'm off for a bit... later all |
| 23:07.50 | dlynes_laptop | wltjr: oh there |
| 23:07.57 | Dovid | never used an wrt |
| 23:07.58 | dlynes_laptop | wltjr: i don't do any advanced setup |
| 23:07.59 | wltjr | dlynes_laptop: just wondering if all used 24 as a regular channel instead of data, so doing 1-23 timeslots, and doing 24 for dchan |
| 23:08.12 | Dovid | and it cant handle the call volume that the other box can |
| 23:08.16 | hardwire | Dovid: then why ask if you aren't gonna be kinky! |
| 23:08.21 | Dovid | also this one u can and fxo/fxs to it |
| 23:08.26 | Dovid | haha |
| 23:08.31 | dlynes_laptop | wltjr: # of groups = 1, group 1 config = all |
| 23:08.33 | hardwire | Dovid: whats the device? |
| 23:08.33 | sevard | Dovid: it's quite easy, if you're servicing up to about four low traffic phones, go for it |
| 23:08.36 | Dovid | didnt know u swong that way |
| 23:08.37 | wltjr | dlynes_laptop: also this voice only for what ever reason is a no go, damn box keeps freezing on wanrouter start |
| 23:08.37 | hardwire | cause its basically asterisk in a box |
| 23:08.44 | fetcher | Ritalin2: your inbound VoIP provider has to support two simultaneous calls (even though they're not "simultaneous" at the phone) |
| 23:08.45 | Dovid | yup |
| 23:08.46 | wltjr | dlynes_laptop: yep so that's not in then |
| 23:08.54 | wltjr | dlynes_laptop: ty |
| 23:08.59 | [hC] | dlynes_laptop: so you were wanting to have metrobridge resell your voip services? |
| 23:09.02 | Dovid | i had a few clients, whose nat issues gave me some white hairs |
| 23:09.06 | dlynes_laptop | [hC]: nah |
| 23:09.13 | Dovid | sevard: u have flashed a wrt54g ? |
| 23:09.16 | dlynes_laptop | [hC]: we were wanting to resell their analog services I think |
| 23:09.19 | sevard | Dovid: so set up a 400mhz box at the client location |
| 23:09.21 | fetcher | Ritalin2: and without VAD / silence-suppression, both will still use bandwidth while connected |
| 23:09.22 | grandy | hello, i'm wondering if someone would be willing to help me figure out why my sip config isn't working.... i know this isn't fun but i'm stuck and i'm sure that someone who'se been through it before would quickly get to the bottom of it... |
| 23:09.23 | sevard | Dovid: yes I do, running asterisk |
| 23:09.24 | dlynes_laptop | [hC]: and their internet |
| 23:09.32 | Ritalin2 | Jingles: my provider supports unlimited (well, bandwidth limited) channels |
| 23:09.42 | Jingles | *nods* |
| 23:09.44 | Dovid | sevard: cant - no room for a box |
| 23:09.47 | Dovid | plus |
| 23:09.49 | [hC] | dlynes_laptop: they're only internet. no telephony. I do telephony for them currently. surprised that they wouldnt want you reselling their internet though. they are expanding into the US market right now so that might be why |
| 23:09.54 | Dovid | u cant store VM on it etc. |
| 23:09.58 | Jingles | well, I'm still new to Asterisk. So, so far I don't have any other questions that might help/shed some light. |
| 23:10.02 | sevard | if you're using business clients I'd suggest going for one of those fancy Asus routers with lots of cpu and ram, check the openwrt website |
| 23:10.07 | Ritalin2 | Jingles: I just wanted to make sure I could do call waiting with one ZAP channel before I wasted a bunch of time and effort |
| 23:10.08 | dlynes_laptop | [hC]: ah...maybe they're already spread too thin |
| 23:10.17 | Dovid | my bigest problem is that i cant compete with panasonic hybird systems |
| 23:10.27 | [hC] | dlynes_laptop: probably just looking to put all focus on the new market, they already ahve a pretty big dominance in vancouver. |
| 23:10.31 | sevard | the WRTSL54G has a USB port and so do those asus routers where you can attach an external HDD |
| 23:10.36 | grandy | I am getting a 488 error from my SIP origination/termination provider... I have the proper port ranges opened up in the firewall (my * box is behind one) and when I type sip show registry nothing is listed... |
| 23:10.38 | Dovid | after the PC and the fxo card my cost is thru the roof |
| 23:10.53 | dlynes_laptop | Dovid: a KX-TDA30? |
| 23:10.54 | Dovid | sevard: u have any links on where i can get one ? |
| 23:10.55 | *** join/#asterisk DrJolo (i=r3-36-15@r3.ists.pl) |
| 23:11.09 | Jingles | right now, I'm watching my server real close because I *was* having a 'mass deregestration' issue. |
| 23:11.10 | dlynes_laptop | Dovid: with KX-TVM50? |
| 23:11.23 | Jingles | one of my SIP providers was causing my entire SIP peer list to drop off. |
| 23:11.30 | kgx | does anyone know how i can trigger a script as soon as someone *answers* an incoming phone |
| 23:11.30 | Dovid | dlynes_laptop: not sure - in genedal |
| 23:11.42 | sevard | Dovid: check the openwrt website for compatibility issues, make sure it works well with white russian 4, (is 5 out? can't remember) then go down to your local walmart and buy one |
| 23:11.44 | Dovid | kgx: agi or a system script ? |
| 23:11.49 | sevard | once you know exactly what you want |
| 23:12.01 | Dovid | sevard: i am real scared of brickin it |
| 23:12.08 | Ritalin2 | I think I'm running WR5 |
| 23:12.14 | Dovid | i once did one - waited 8 minutes and gave up - bricked it |
| 23:12.21 | Dovid | went back to cc and it said it was doa |
| 23:12.23 | Dovid | ;) |
| 23:12.25 | sevard | Dovid: that's why you buy from walmart, take it back within 30 days if you're evil like my boss |
| 23:12.28 | dlynes_laptop | Dovid: KX-TDA30 with a 3 4-port PC/104 cards, and a KX-TVM50 is generally around $1300 or so before taxes and labor |
| 23:12.31 | Dovid | hahah |
| 23:12.40 | Dovid | dlynes: ur right |
| 23:12.42 | dlynes_laptop | Dovid: in Canadian dollars, and that's wholesale |
| 23:12.45 | sevard | if you're flashing openwrt to a wrt54 you should WAIT |
| 23:12.46 | Dovid | but i get decent machines |
| 23:12.54 | sevard | sometimes it takes upwords of 15 minutes to run firstboot |
| 23:12.57 | dlynes_laptop | Dovid: ummm |
| 23:13.02 | Dovid | sevard: my patience needs some work |
| 23:13.10 | sevard | Dovid: yes it does. |
| 23:13.12 | Dovid | by the time i get an echo can card with a box i am done |
| 23:13.13 | dlynes_laptop | Dovid: so basically you're trying to compare a BCM to a KX-TDA30 |
| 23:13.19 | Dovid | BCM ? |
| 23:13.22 | Dovid | BCM = ? |
| 23:13.23 | *** part/#asterisk CPSK (n=CPSK@c6.ars.ba.nextra.sk) |
| 23:13.26 | *** join/#asterisk CPSK (n=CPSK@c6.ars.ba.nextra.sk) |
| 23:13.27 | dlynes_laptop | Dovid: of course your solution is going to be more expensive |
| 23:13.31 | *** part/#asterisk CPSK (n=CPSK@c6.ars.ba.nextra.sk) |
| 23:13.48 | dlynes_laptop | Dovid: BCM == Nortel's VOIP solution for Small to medium sized businesses |
| 23:13.49 | Dovid | so i wana use an embeded device so i can lower my cost |
| 23:14.03 | kgx | Dovid: possibly agi (a php script). what im trying to do is run something similar to cdr_mysql except i wanna do it when the phone is picked up |
| 23:14.04 | sevard | there's the asterisk applicance |
| 23:14.07 | grandy | if I have a register line in sip.conf, should sip show registry in the console show it? |
| 23:14.24 | Dovid | sevard: but is it locked ? |
| 23:14.29 | dlynes_laptop | Dovid: why not just use something like a Dell SC430? |
| 23:14.34 | Dovid | i saw it - if its open i will go for it |
| 23:14.37 | sevard | i've never used one |
| 23:14.42 | Dovid | cause i dont get low ebd dell shit |
| 23:14.48 | dlynes_laptop | Dovid: erm... Dell SC440 I mean...the 440's replaced the 430 |
| 23:15.00 | dlynes_laptop | Dovid: you don't do low end dell shit |
| 23:15.00 | Dovid | grandy: whats ur question |
| 23:15.11 | dlynes_laptop | Dovid: but yet you want to use an even shittier linksys router? |
| 23:15.15 | Dovid | haha |
| 23:15.25 | dlynes_laptop | you're making absolutely zero sense |
| 23:15.27 | Dovid | dlynes_laptop: wasnt gona go for the linksys router |
| 23:15.33 | Dovid | was gona use the blackfin |
| 23:15.38 | FuriousGeorge | one of my users swears she would park a call and it would get disconnected by the next incoming call |
| 23:15.41 | sevard | Product Code: AADKSTD |
| 23:15.42 | sevard | Description: Asterisk Appliance Developer Kit Std |
| 23:15.42 | sevard | Price: $2195.00 |
| 23:15.43 | FuriousGeorge | then she couldnt call out |
| 23:15.44 | hardwire | I am having a hard time following here |
| 23:15.50 | FuriousGeorge | of course we couldnt reproduce it |
| 23:15.54 | grandy | dovid ... i'm trying to figure out if my sip.conf is actually connecting to the sip origination/termination provider... when I type sip show registry it doesn't show anything... I have a register line and a friend listing... |
| 23:15.55 | dlynes_laptop | Dovid: try checking out WRAP, Soekris, or MBX |
| 23:15.57 | Dovid | sevard: that price is with all the training |
| 23:16.06 | J4k3 | wrap, soekris, etc, are all very low performance. |
| 23:16.13 | hardwire | *bullshit* |
| 23:16.14 | sevard | Intended for Asterisk experts, those who simply want to preview the Asterisk Appliance, and others who do not require training or support, this package includes the Asterisk Appliance Developers Kit only. |
| 23:16.19 | hardwire | they are low speed :) |
| 23:16.31 | J4k3 | hardwire: they're overclocked 486s, when it comes to capability. |
| 23:16.32 | Dovid | grandy: when u do sip show registry what do u get ? |
| 23:16.38 | grandy | Dovid: nothing... |
| 23:16.46 | dlynes_laptop | J4k3: some of them are a lot better than that |
| 23:16.49 | Dovid | grandy: also pb ur sip.conf |
| 23:16.51 | Dovid | ~pb |
| 23:16.52 | 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 |
| 23:17.05 | hardwire | J4k3: I have a few sitting around, you wanna buy some :)? |
| 23:17.10 | Dovid | also did u reload asterisk after making ur changes ? |
| 23:17.15 | sevard | Dovid: a 500mhz clunker with 256mb of ram will process calls better than a wrt54gs |
| 23:17.18 | J4k3 | well... for about $200 you can get a C-M 1.3 ghz based laptop with a busted LCD off ebay... |
| 23:17.29 | hardwire | I wrote an LED driver for the GPIO that used sigma-delta modulation to fade in and out the led's |
| 23:17.30 | J4k3 | sometimes less |
| 23:17.34 | Dovid | sevard: wrts' are for a diffrent client |
| 23:17.34 | hardwire | ona 700hz timer |
| 23:17.35 | hardwire | it was neat |
| 23:17.36 | hardwire | :) |
| 23:17.44 | Ritalin2 | jingles: it looks lke the setup I hvae should work/do what I want. I just have to figure out why it's not :) |
| 23:17.53 | *** part/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 23:17.57 | Jingles | Ritalin2: *nods* |
| 23:18.17 | Dovid | wrt's are for a client that has phones behind nat which means that the rtp stream will go thru his server - he dosent want that |
| 23:18.29 | sevard | I need a laptop but I don't have any more $$, I'm still on this 133mhz 24mb of ram clunking laptop |
| 23:18.53 | Ritalin2 | what can you do with a 133mhz laptop? |
| 23:18.58 | Dovid | SIP Phones ---> Wrt ---> Asterisk server ---> Sip provider |
| 23:18.59 | Qwell[] | Ritalin2: lots |
| 23:19.06 | sevard | I can play an mp3! |
| 23:19.09 | Jingles | you don't need a lot of horsepower to sling linux. |
| 23:19.14 | Qwell[] | sevard: and...that's about it :P |
| 23:19.21 | sevard | i can ssh out :) |
| 23:19.21 | Dovid | this way the rtp can go from wrt directly to sip provider |
| 23:19.27 | J4k3 | Ritalin2: I have a 486DX2/66 laptop here with a 200mW senao 802.11b card in the side, runs FreeBSD. |
| 23:19.33 | Dovid | damn |
| 23:19.34 | J4k3 | 20MB ram, 12GB HD at the moment. |
| 23:19.37 | Qwell[] | sevard: You can bash people with it |
| 23:19.38 | Dovid | i got some old laptops out there |
| 23:19.40 | sevard | THis laptop would be great except I can't get it to support any newer PCMCIA cards |
| 23:19.49 | Jingles | cardbus itchews. |
| 23:19.52 | sevard | Qwell[]: yes, it's about 14lbs |
| 23:19.52 | J4k3 | my real laptop is a P-M 1.7 w/ 1.2GB + 60gb 7200 rpm |
| 23:19.54 | grandy | Dovid: ok... here it is: http://pastebin.ca/326234 |
| 23:19.54 | Dovid | sevard: i think we will start a paypal fund for ya |
| 23:19.59 | sevard | Dovid: hahaha |
| 23:20.00 | Ritalin2 | linux and irc .... what else does a person need? |
| 23:20.07 | Qwell[] | Dovid: hey, mine first :p |
| 23:20.09 | sevard | Will paypal for asterisk help |
| 23:20.09 | dlynes_laptop | J4k3: that's something to brag about? |
| 23:20.21 | Ritalin2 | wonder if my wrt54g has enough horsepower to run linux. I'm *really* impressed with openwrt |
| 23:20.22 | hardwire | J4k3: as opposed to your fake laptop, using Xen? |
| 23:20.22 | Dovid | Qwell:U work for the king - ask him for a new years present |
| 23:20.32 | wunderkin | Ritalin2, maybe some ritalin? |
| 23:20.36 | *** join/#asterisk manipura (n=mike@S01060011954c9c46.cg.shawcable.net) |
| 23:20.36 | hardwire | Ritalin2: openwrt == linux isn't it? |
| 23:20.41 | dlynes_laptop | hardwire: yes |
| 23:20.44 | sevard | dlynes_laptop: to me it is, it's 12x faster than my lapotpo |
| 23:20.46 | J4k3 | dlynes_laptop: no, but its several dozen times faster than the 486. |
| 23:20.47 | Ritalin2 | handwire: yes |
| 23:20.50 | sevard | hardwire: yes'm |
| 23:20.52 | dlynes_laptop | J4k3: heh |
| 23:20.56 | hardwire | Ritalin2: so you just wanna install debian on it then? |
| 23:20.58 | Dovid | my laptop is a dual core 2.0, gig of ram, 80 gig sata, dual layer dvd burner...... |
| 23:21.01 | hardwire | oh |
| 23:21.02 | hardwire | hah |
| 23:21.06 | Ritalin2 | handwire: what? |
| 23:21.07 | sevard | Dovid: can I have your laptop? |
| 23:21.08 | hardwire | i'm *really* impressed |
| 23:21.09 | manipura | Playback() doesn't play wav files or what? |
| 23:21.11 | dlynes_laptop | sevard: so i guess mine would probably be about 100x faster than your laptop :0 |
| 23:21.12 | J4k3 | Dovid: nice. |
| 23:21.14 | hardwire | not i'm not really impressed |
| 23:21.16 | sevard | manipura: It does. |
| 23:21.17 | hardwire | I misreaded |
| 23:21.17 | Dovid | grandyL are u behind nat ? |
| 23:21.22 | Qwell[] | manipura: remove the extension |
| 23:21.23 | Dovid | grandy: |
| 23:21.34 | sevard | Let's start a "donate your old laptop" fund |
| 23:21.38 | j0 | anybody use multiple net connections for their asterisk box? i'm wondering how to do automatic failover |
| 23:21.39 | grandy | Dovid: yep but 5060-5070 and 8xxx through 35000 have been opened |
| 23:21.43 | sevard | who can upgrade my 133mhz beast |
| 23:21.45 | dlynes_laptop | Dovid: mine's the same, but 1G DDR-2, 120GB SATA |
| 23:21.50 | Ritalin2 | hardwire: you don't like openwrt? |
| 23:21.55 | sevard | my 133mhz laptop is hostnamed "Brick" |
| 23:21.57 | Dovid | grandy: ok u need to set nat=yes |
| 23:22.02 | dlynes_laptop | Dovid: oh...and original Core Duo, not Core Duo 2 |
| 23:22.11 | dlynes_laptop | J4k3: good deal |
| 23:22.11 | Dovid | core duo 2 |
| 23:22.16 | EmleyMoor | sevard: Do you have a machine called "outhouse"? |
| 23:22.23 | sevard | EmleyMoor: I COULD! |
| 23:22.28 | J4k3 | yeah... made upgrading it not painful at all. |
| 23:22.33 | sevard | I have one named "TheJesus" but he's even slower |
| 23:22.35 | Ritalin2 | my laptop was $1300 or $1400 about 4 months ago. core duo and then a month later the core 2 duo comes out for the same price... sigh |
| 23:22.36 | grandy | Dovid: ok... did that |
| 23:22.43 | dlynes_laptop | Dovid: toshiba satellite? |
| 23:22.47 | sevard | sorry Ritalin2 :\ |
| 23:23.02 | dlynes_laptop | Ritalin2: same thing happened to me |
| 23:23.10 | grandy | Dovid: I get a 488 not acceptable here error from voipstreet |
| 23:23.15 | dlynes_laptop | Ritalin2: but the price was $999 Cdn |
| 23:23.15 | EmleyMoor | My machines generally have Sheffield-related names - but then again, my domain has a Sheffield-related name too |
| 23:23.39 | Jingles | CDN? is that even real money? |
| 23:23.43 | Dovid | grandy: http://pastebin.ca/326236 |
| 23:23.48 | sevard | My machines are named after their purpose, like Kelso, named after Bob Kelso from Scrubs as it was a short, white machine that didn't do much. |
| 23:23.51 | Dovid | dlynes: Dell |
| 23:23.52 | *** join/#asterisk Tako-san (n=Tako-san@24.108.162.254) |
| 23:23.57 | dlynes_laptop | Jingles: hey...our money's better than greenbacks |
| 23:24.00 | Dovid | haha |
| 23:24.04 | Dovid | Kelso |
| 23:24.06 | Dovid | hate the ass |
| 23:24.07 | sevard | :) |
| 23:24.10 | Jingles | heh |
| 23:24.12 | dlynes_laptop | Jingles: at least when we're drunk we can't accidentally give the cab driver a hundred spot |
| 23:24.18 | sevard | I spoke to Bob Kelso on the phone, the real guy |
| 23:24.20 | Dovid | iw ould name it after the blond |
| 23:24.27 | j0 | lol dlynes_laptop |
| 23:24.37 | file | American currency forces you to think |
| 23:24.43 | Dovid | lol |
| 23:24.47 | sevard | you know that episode when Turk gets a cellphone Call-Tur(k)? It's a real phone they turn on when they're filming |
| 23:24.54 | Dovid | i need a well deserved smoke. brb |
| 23:24.59 | sevard | I got some lady that worked there then she threw the phone to Bob kelso, it was tight |
| 23:25.03 | Tako-san | Anyone have experience with using an Adtran 750? |
| 23:25.05 | EmleyMoor | My main Linux box is "topdeck", as in the M1. My Win98 box (no longer used) was "bottomdeck". My Asterisk box is "firthpark", simply because that's what it used to be when it ran OpenBSD and I couldn't find a more appropriate name... |
| 23:25.14 | EmleyMoor | As for my laptop, that's "wincobank" |
| 23:25.19 | Dovid | and i may be getting a billion are as a client........ |
| 23:25.23 | Dovid | we can all show off |
| 23:25.24 | dlynes_laptop | well, when you drink as much beer as your average canadian |
| 23:25.24 | sevard | wincobank? |
| 23:25.32 | dlynes_laptop | having different colored money is extremely important |
| 23:25.37 | EmleyMoor | (though it no longer appears as wincobank due to me using wireless) |
| 23:25.42 | *** join/#asterisk Shaun2222 (n=Shaun@ip68-4-212-221.oc.oc.cox.net) |
| 23:25.49 | EmleyMoor | Wincobank is a settlement near Tinsley Viaduct |
| 23:25.52 | sevard | dlynes_laptop: canadian money is nice for when you're hammered and in the dark |
| 23:25.54 | mercestes | My laptop is the DeltaFlier. My desktop is the defiant. My Server is the relativity, it has a HDD share named Kes. |
| 23:26.00 | dlynes_laptop | sevard: exactly |
| 23:26.08 | mercestes | My *other* desktop is called voyager |
| 23:26.10 | dlynes_laptop | sevard: with american money you go real broke real fast |
| 23:26.12 | sevard | dlynes_laptop: but you can't tip strippers loonies or twoonies |
| 23:26.16 | EmleyMoor | mercestes: Kes? |
| 23:26.17 | Shaun2222 | whats a good PCI card that works with asterisk that will allow me to plug in 2 or more phone lines? |
| 23:26.19 | mercestes | my password for IRC is "Cisco." |
| 23:26.25 | dlynes_laptop | sevard: sure you can...tuck it in their undies |
| 23:26.28 | mercestes | Kes is this hot vulcan chick. |
| 23:26.32 | Jingles | my computers are all named 'poop'. or some derivation thereof. |
| 23:26.34 | mercestes | well, she's not vulcan. |
| 23:26.35 | sevard | dlynes_laptop: hahaha, i think they might get ticked |
| 23:26.36 | mercestes | she's something else.' |
| 23:26.40 | dlynes_laptop | sevard: or get them to grab it with their titties from your teeth :) |
| 23:26.54 | EmleyMoor | "Kes" was the film version of Barry Hines' "A Kestrel for a Knave" |
| 23:26.57 | HushPe | http://pastie.caboo.se/35171 < i haven't been able to find a solution on google :( |
| 23:26.58 | mercestes | I thought she was vulcan. |
| 23:27.08 | mercestes | she worked wtih the crazy doctor on Voyager |
| 23:27.19 | dlynes_laptop | sevard: unfortunately you can't do the teeth trick in BC :( |
| 23:27.23 | sevard | dlynes_laptop: I don't think you understand whow strippers work |
| 23:27.28 | dlynes_laptop | sevard: no physical contact allowed here :( |
| 23:27.34 | sevard | heh |
| 23:27.36 | dlynes_laptop | sevard: but in ontario it worked just fine |
| 23:27.43 | sevard | don't you have a wife? |
| 23:27.47 | dlynes_laptop | nope |
| 23:27.49 | Ritalin2 | I don't run openwrt I actually run X-Wrt. I'm more impressed with it than openwrt fwiw |
| 23:27.59 | dlynes_laptop | although |
| 23:28.00 | sevard | swear you were married, long time girlfriend? |
| 23:28.01 | Dovid[AWAY] | HushPe: did u install zaptel ? |
| 23:28.07 | dlynes_laptop | just met a super nice korean girl recently |
| 23:28.12 | dlynes_laptop | nope |
| 23:28.13 | sevard | stripper? |
| 23:28.16 | dlynes_laptop | nah |
| 23:28.17 | HushPe | Dovid[AWAY]: i did :) |
| 23:28.20 | sevard | heh |
| 23:28.22 | dlynes_laptop | she actually goes to church regularly :0 |
| 23:28.26 | Dovid[AWAY] | lippri too ? |
| 23:28.29 | sevard | score |
| 23:28.33 | Shaun2222 | whats a good PCI card that works with asterisk that will allow me to plug in 2 or more phone lines? |
| 23:28.36 | sevard | v card for dlynes! |
| 23:28.37 | mercestes | dlynes_laptop: My wife is half-korean..;) good find. |
| 23:28.47 | HushPe | Dovid[AWAY]: the driver is working (i can hear when i dial on the 'real phone') |
| 23:28.47 | sevard | Shaun2222: check the digium website for TDM cards |
| 23:28.48 | Jingles | did you install them *both* (and the correct versions) prior to compiling asterisk itself? |
| 23:28.53 | EmleyMoor | Shaun2222: A TDM400P will support up to 4 with the right modules |
| 23:28.56 | dlynes_laptop | mercestes: yeah...she's super pretty, and sparkling personality |
| 23:29.02 | dlynes_laptop | mercestes: super funny |
| 23:29.06 | naftali5 | HushPe, had the problem a few times and used an older one i had lying around compiled :( but one thing to make sure is that your source directories are named zaptel asterisk etc not asterisk-1.4 or somthing like that |
| 23:29.06 | mercestes | nice |
| 23:29.18 | mercestes | ... |
| 23:29.19 | sevard | my gf is a polskie |
| 23:29.19 | dlynes_laptop | mercestes: i prefer chinese, but i'm starting to think i should switch to koreans :) |
| 23:29.20 | HushPe | Jingles: indeed, however the thing i'm not too sure about was removing 1.4 for zaptel :( |
| 23:29.24 | sevard | a hawt polskie |
| 23:29.40 | mercestes | mine crawls around onthe floor in her collar staring up at me waiting for me to tell her to do stuff. =. |
| 23:29.41 | mercestes | =/ |
| 23:29.45 | sevard | i likes me my germanic baby |
| 23:29.53 | sevard | mercestes: this isn't #bondage |
| 23:29.56 | HushPe | naftali5: i got no zap channel at all when i was using 1.4, so using 1.2 instead |
| 23:30.02 | mercestes | there is a channel for that?? |
| 23:30.08 | sevard | there's a NETWORK for that |
| 23:30.13 | Dovid[AWAY] | HushPe: http://lists.digium.com/pipermail/asterisk-users/2006-April/146804.html |
| 23:30.16 | grandy | Dovid[AWAY]: Thanks!!! it works! |
| 23:30.18 | sevard | my sicko buddy lives there |
| 23:30.23 | mercestes | wher eis it? |
| 23:30.30 | sevard | i think it's irc.bondage.net |
| 23:30.39 | *** join/#asterisk Daveb21 (n=daveb@eth2235.sa.adsl.internode.on.net) |
| 23:30.41 | Jingles | I'm using 1.2 for similar reasons - it was absolutely a bear to fight my way through the zap stuff in 1.4 |
| 23:30.41 | naftali5 | HushPe, is your src dir named /usr/src/asterisk or asterisk1.2.xx |
| 23:30.43 | sevard | he invited me there once, i stayed approx 4 seconds. |
| 23:30.46 | mercestes | woohoo] |
| 23:30.50 | sevard | sicko. |
| 23:31.34 | mercestes | can MIRC connect to more than on eserver at once? Hmm. |
| 23:31.38 | sevard | yes |
| 23:32.00 | sevard | I had to remember back to like 1995 to answer that question |
| 23:32.04 | HushPe | naftali5: asterisk-1.2.14 i've gotta zip out for a few mins, any help could PM me I would be greatly appreciative |
| 23:32.14 | HushPe | oh i have a TDM422P |
| 23:32.20 | naftali5 | HushPe, one sec |
| 23:32.30 | naftali5 | mv asterisk-1.2.14 asterisk |
| 23:32.39 | HushPe | into /usr/src ? |
| 23:32.43 | naftali5 | and the same goes for zaptel libpri etc |
| 23:32.44 | naftali5 | yes |
| 23:32.53 | naftali5 | don't leave it named with the version number |
| 23:32.58 | HushPe | i compiled from /storage/asterisk/1.2/asterisk-1.2.14 |
| 23:33.00 | Daveb21 | Hey all, quick question, does anyone know of some good asterisk resources on the net for dealing with multiple server implementations or setting up asterisk for maximum uptime. Had a bid of a google and couldnt find much. Cheers |
| 23:33.01 | HushPe | ah that might be it |
| 23:33.02 | generalhan | mercestes: just click the "open server in new window" check box when you connect |
| 23:33.02 | naftali5 | zaptel has to know where to find asterisk |
| 23:33.07 | HushPe | i'll try hwen i get back in :) |
| 23:33.10 | HushPe | cheers heaps! |
| 23:34.17 | mercestes | sevard: can't find it |
| 23:34.25 | kgx | hey what would be easiest way to check if a user is on call? |
| 23:34.41 | Jingles | show channels |
| 23:34.50 | *** join/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 23:35.08 | mercestes | sevard: It's .com not .net |
| 23:35.41 | kgx | Jingles: and how about checking the caller id of the person they're talking to? |
| 23:35.44 | sevard | alright, it was seriously like 3 years ago when he told me about it, i'm surprised i even remembered |
| 23:36.06 | Jingles | well, I just did a 'show channels' |
| 23:36.11 | *** join/#asterisk Mavvie (n=edwin@ppp30-46.lns1.syd6.internode.on.net) |
| 23:36.33 | kgx | doesn't show the cid of the other person though |
| 23:36.37 | Jingles | and it spat out what looks like an 'end to end' display of who was talking to who (as far as my system had the entries defined) |
| 23:37.02 | kgx | ok ...ta |
| 23:37.05 | sevard | how about a show channels verbose |
| 23:37.16 | Jingles | oooooer! |
| 23:37.23 | Jingles | *huzzah!* |
| 23:37.25 | Jingles | beer for sevard |
| 23:37.58 | sevard | fux yeah, lessee, average beer about 3.50/pint |
| 23:38.04 | sevard | paypal it? ;) |
| 23:38.11 | Jingles | lols |
| 23:38.11 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net) |
| 23:38.14 | sevard | heh |
| 23:38.29 | Jingles | you're so po you can't afford the last two letters? |
| 23:38.43 | sevard | true dat |
| 23:39.08 | kgx | sevard: yeah thats wht i tried eventually :) |
| 23:39.14 | Tako-san | Any suggestions for why analog phones won't ring when they receive a call? They have dialtone and can dial out but do not ring on incoming calls. |
| 23:39.31 | sevard | ringing voltage |
| 23:39.35 | Tako-san | Ok |
| 23:39.38 | sevard | are you using a tdm2400 or a tdm400p? |
| 23:39.40 | Tako-san | How do I fix that? |
| 23:39.48 | sevard | there's a fix, one second. |
| 23:39.55 | Tako-san | Adtran TA 750 |
| 23:40.28 | Tako-san | Asterisk->Channel Bank->Patch Panel->Analog phone |
| 23:40.47 | sevard | i have no idea about that particular hardware, but in your conf try boostringer=1 |
| 23:40.59 | Tako-san | Which conf file? |
| 23:41.00 | sevard | no wait |
| 23:41.15 | sevard | i think it's when you probe the module |
| 23:41.25 | sevard | like modprobe xxxmodulexxx options boostringer=1 |
| 23:41.44 | *** join/#asterisk Nobbie (n=corne@wbs-41-208-195-71.wbs.co.za) |
| 23:42.32 | Tako-san | Hmmm |
| 23:42.49 | sevard | or in your /etc/modutils/zaptel you do options xxxmodulexxx boostringer=1 |
| 23:42.49 | Tako-san | which module are you referencing? |
| 23:42.59 | Tako-san | ok let me check |
| 23:43.07 | sevard | the module that controls whatever hardware you're using |
| 23:43.24 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 23:44.04 | sevard | apparently that amps up the zaptel ringing voltage from 48VAC to 89VAC |
| 23:44.12 | Shaun2222 | anything special that need to be done when upgradeing from 1.2 to 1.4? |
| 23:44.29 | Nobbie | Shaun2222: i'm busy doing that right now, i'm a bit ~nervous~ |
| 23:44.43 | Tako-san | Ok, thanks sevard. Will look into it. |
| 23:44.49 | sevard | no problemo |
| 23:45.00 | sevard | that'll be four bucks baby |
| 23:45.15 | aptura | ha vitality is having issues with my number. Slow bussy when i call it. |
| 23:45.34 | *** part/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 23:45.41 | Nobbie | Shaun2222: so far i've only had issues with the G729 codecs, had to download the appropriate ones from http://kvin.lv/pub/Linux/Asterisk/built-for-asterisk-1.4/, and now the CDR's are messed up, probably becuase i didn't upgrade the addons |
| 23:45.54 | aptura | btw, what is the margin looking like on voip wholesalers? is it so thin thats why asterlink went out of business? |
| 23:45.56 | Nobbie | Shaun2222: the UPGRADE.txt file is pretty usefull too |
| 23:45.58 | Shaun2222 | ah ok |
| 23:46.36 | sevard | Tako-san: once again, that'd be like "modprobe wctdm boostringer=1" |
| 23:46.44 | sevard | but I'm only familar with digium hardware |
| 23:46.55 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 23:46.55 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 23:47.30 | aptura | sevard how many digium cards have you worked with now? |
| 23:47.32 | Tako-san | Ok |
| 23:47.40 | sevard | aptura: quite a few |
| 23:47.42 | Tako-san | I am currently using a Sangoma card. |
| 23:48.00 | sevard | Apparently Sangoma cards use the same zaptel modules, correct? |
| 23:48.03 | Tako-san | I should also mention that the phones will ring but only after hanging up the calling phone |
| 23:48.08 | Tako-san | Correct |
| 23:48.16 | sevard | then rmmod the module and modprobe it again with boostringer=1 |
| 23:48.17 | Tako-san | Then the receiving phone will ring once |
| 23:48.38 | sevard | I had the same issue at a hotel I did an install with, I boosted the ringing voltage and everything was candy |
| 23:48.48 | Tako-san | Ok will give it a go, |
| 23:48.56 | aptura | sevard cool |
| 23:48.57 | aptura | :) |
| 23:49.14 | Daveb21 | Sure enough right after I posted that I found a little bit of info in The Hitchhikers guide to Asterisk but not much - havent found anything on asterisk site or in the TFOT book tho |
| 23:51.45 | De_Mon | I have a dialplan that calls a numbers and puts the caller/callee in a conference. How do I determine when either person leaves the conference? |
| 23:52.17 | De_Mon | I tried adding a NoOp after the meetme command but it doesnt get executed, it says == Spawn extension (elephant-queues, goto-500, 3) exited non-zero on 'SIP/jon-08268540' |
| 23:52.29 | De_Mon | instaed of going to priority 4 |
| 23:52.51 | *** join/#asterisk k-man__ (n=jason@ppp244-232.static.internode.on.net) |
| 23:53.17 | De_Mon | I can put stuff in the hangup extension, but that seems rather overkill |
| 23:56.26 | *** join/#asterisk saftsack (n=saftsack@pD9E06BD7.dip.t-dialin.net) |
| 23:56.29 | *** join/#asterisk Globetrotter (n=eric@205.211.214.167) |
| 23:57.09 | Globetrotter | hi guys, can i have two network cards in my box and connect it to two differnt isps.. for redundancy.. |
| 23:57.16 | *** join/#asterisk neddy (n=js152033@192.18.43.225) |
| 23:57.36 | Globetrotter | also can i have an asterisk server cluster? |
| 23:57.54 | Ritalin2 | you can do anything if you have enough 1) time 2) money |
| 23:58.08 | Supaplex | but will it stay crispy in milk? |
| 23:58.21 | Ritalin2 | not without magic dust |
| 23:58.36 | sevard | BUT, WILL IT BLEND? |
| 23:58.44 | Ritalin2 | THAT IS THE QUESTION |
| 23:59.23 | Supaplex | *> reload app_chewey_end_fries |
